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Diffstat (limited to '1.4.23-rc4/channels/chan_sip.c')
| -rw-r--r-- | 1.4.23-rc4/channels/chan_sip.c | 18893 |
1 files changed, 18893 insertions, 0 deletions
diff --git a/1.4.23-rc4/channels/chan_sip.c b/1.4.23-rc4/channels/chan_sip.c new file mode 100644 index 000000000..004aaaabe --- /dev/null +++ b/1.4.23-rc4/channels/chan_sip.c @@ -0,0 +1,18893 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 1999 - 2006, Digium, Inc. + * + * Mark Spencer <markster@digium.com> + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +/*! + * \file + * \brief Implementation of Session Initiation Protocol + * + * \author Mark Spencer <markster@digium.com> + * + * See Also: + * \arg \ref AstCREDITS + * + * Implementation of RFC 3261 - without S/MIME, TCP and TLS support + * Configuration file \link Config_sip sip.conf \endlink + * + * + * \todo SIP over TCP + * \todo SIP over TLS + * \todo Better support of forking + * \todo VIA branch tag transaction checking + * \todo Transaction support + * + * \ingroup channel_drivers + * + * \par Overview of the handling of SIP sessions + * The SIP channel handles several types of SIP sessions, or dialogs, + * not all of them being "telephone calls". + * - Incoming calls that will be sent to the PBX core + * - Outgoing calls, generated by the PBX + * - SIP subscriptions and notifications of states and voicemail messages + * - SIP registrations, both inbound and outbound + * - SIP peer management (peerpoke, OPTIONS) + * - SIP text messages + * + * In the SIP channel, there's a list of active SIP dialogs, which includes + * all of these when they are active. "sip show channels" in the CLI will + * show most of these, excluding subscriptions which are shown by + * "sip show subscriptions" + * + * \par incoming packets + * Incoming packets are received in the monitoring thread, then handled by + * sipsock_read(). This function parses the packet and matches an existing + * dialog or starts a new SIP dialog. + * + * sipsock_read sends the packet to handle_request(), that parses a bit more. + * if it's a response to an outbound request, it's sent to handle_response(). + * If it is a request, handle_request sends it to one of a list of functions + * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc + * sipsock_read locks the ast_channel if it exists (an active call) and + * unlocks it after we have processed the SIP message. + * + * A new INVITE is sent to handle_request_invite(), that will end up + * starting a new channel in the PBX, the new channel after that executing + * in a separate channel thread. This is an incoming "call". + * When the call is answered, either by a bridged channel or the PBX itself + * the sip_answer() function is called. + * + * The actual media - Video or Audio - is mostly handled by the RTP subsystem + * in rtp.c + * + * \par Outbound calls + * Outbound calls are set up by the PBX through the sip_request_call() + * function. After that, they are activated by sip_call(). + * + * \par Hanging up + * The PBX issues a hangup on both incoming and outgoing calls through + * the sip_hangup() function + * + * \par Deprecated stuff + * This is deprecated and will be removed after the 1.4 release + * - the SIPUSERAGENT dialplan variable + * - the ALERT_INFO dialplan variable + */ + +/*** MODULEINFO + <depend>res_features</depend> + ***/ + + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include <stdio.h> +#include <ctype.h> +#include <string.h> +#include <unistd.h> +#include <sys/socket.h> +#include <sys/ioctl.h> +#include <net/if.h> +#include <errno.h> +#include <stdlib.h> +#include <fcntl.h> +#include <netdb.h> +#include <signal.h> +#include <sys/signal.h> +#include <netinet/in.h> +#include <netinet/in_systm.h> +#include <arpa/inet.h> +#include <netinet/ip.h> +#include <regex.h> + +#include "asterisk/lock.h" +#include "asterisk/channel.h" +#include "asterisk/config.h" +#include "asterisk/logger.h" +#include "asterisk/module.h" +#include "asterisk/pbx.h" +#include "asterisk/options.h" +#include "asterisk/sched.h" +#include "asterisk/io.h" +#include "asterisk/rtp.h" +#include "asterisk/udptl.h" +#include "asterisk/acl.h" +#include "asterisk/manager.h" +#include "asterisk/callerid.h" +#include "asterisk/cli.h" +#include "asterisk/app.h" +#include "asterisk/musiconhold.h" +#include "asterisk/dsp.h" +#include "asterisk/features.h" +#include "asterisk/srv.h" +#include "asterisk/astdb.h" +#include "asterisk/causes.h" +#include "asterisk/utils.h" +#include "asterisk/file.h" +#include "asterisk/astobj.h" +#include "asterisk/devicestate.h" +#include "asterisk/linkedlists.h" +#include "asterisk/stringfields.h" +#include "asterisk/monitor.h" +#include "asterisk/localtime.h" +#include "asterisk/abstract_jb.h" +#include "asterisk/compiler.h" +#include "asterisk/threadstorage.h" +#include "asterisk/translate.h" + +#ifndef FALSE +#define FALSE 0 +#endif + +#ifndef TRUE +#define TRUE 1 +#endif + +#define SIPBUFSIZE 512 + +#define XMIT_ERROR -2 + +#define VIDEO_CODEC_MASK 0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */ +#ifndef IPTOS_MINCOST +#define IPTOS_MINCOST 0x02 +#endif + +/* #define VOCAL_DATA_HACK */ + +#define DEFAULT_DEFAULT_EXPIRY 120 +#define DEFAULT_MIN_EXPIRY 60 +#define DEFAULT_MAX_EXPIRY 3600 +#define DEFAULT_REGISTRATION_TIMEOUT 20 +#define DEFAULT_MAX_FORWARDS "70" + +/* guard limit must be larger than guard secs */ +/* guard min must be < 1000, and should be >= 250 */ +#define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */ +#define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of + EXPIRY_GUARD_SECS */ +#define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If + GUARD_PCT turns out to be lower than this, it + will use this time instead. + This is in milliseconds. */ +#define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when + below EXPIRY_GUARD_LIMIT */ +#define DEFAULT_EXPIRY 900 /*!< Expire slowly */ + +static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */ +static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */ +static int default_expiry = DEFAULT_DEFAULT_EXPIRY; +static int expiry = DEFAULT_EXPIRY; + +#ifndef MAX +#define MAX(a,b) ((a) > (b) ? (a) : (b)) +#endif + +#define CALLERID_UNKNOWN "Unknown" + +#define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */ +#define DEFAULT_FREQ_OK 60 * 1000 /*!< Qualification: How often to check for the host to be up */ +#define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */ + +#define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */ +#define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */ +#define SIP_TRANS_TIMEOUT 32000 /*!< SIP request timeout (rfc 3261) 64*T1 + \todo Use known T1 for timeout (peerpoke) + */ +#define DEFAULT_TRANS_TIMEOUT -1 /* Use default SIP transaction timeout */ +#define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */ + +#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */ +#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */ +#define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */ + +#define SDP_MAX_RTPMAP_CODECS 32 /*!< Maximum number of codecs allowed in received SDP */ + +#define INITIAL_CSEQ 101 /*!< our initial sip sequence number */ + +/*! \brief Global jitterbuffer configuration - by default, jb is disabled */ +static struct ast_jb_conf default_jbconf = +{ + .flags = 0, + .max_size = -1, + .resync_threshold = -1, + .impl = "" +}; +static struct ast_jb_conf global_jbconf; + +static const char config[] = "sip.conf"; +static const char notify_config[] = "sip_notify.conf"; + +#define RTP 1 +#define NO_RTP 0 + +/*! \brief Authorization scheme for call transfers +\note Not a bitfield flag, since there are plans for other modes, + like "only allow transfers for authenticated devices" */ +enum transfermodes { + TRANSFER_OPENFORALL, /*!< Allow all SIP transfers */ + TRANSFER_CLOSED, /*!< Allow no SIP transfers */ +}; + + +enum sip_result { + AST_SUCCESS = 0, + AST_FAILURE = -1, +}; + +/*! \brief States for the INVITE transaction, not the dialog + \note this is for the INVITE that sets up the dialog +*/ +enum invitestates { + INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */ + INV_CALLING = 1, /*!< Invite sent, no answer */ + INV_PROCEEDING = 2, /*!< We got/sent 1xx message */ + INV_EARLY_MEDIA = 3, /*!< We got/sent 18x message with to-tag back */ + INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */ + INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */ + INV_TERMINATED = 6, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done + The only way out of this is a BYE from one side */ + INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */ +}; + +/* Do _NOT_ make any changes to this enum, or the array following it; + if you think you are doing the right thing, you are probably + not doing the right thing. If you think there are changes + needed, get someone else to review them first _before_ + submitting a patch. If these two lists do not match properly + bad things will happen. +*/ + +enum xmittype { + XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits. + If it fails, it's critical and will cause a teardown of the session */ + XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */ + XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */ +}; + +enum parse_register_result { + PARSE_REGISTER_FAILED, + PARSE_REGISTER_UPDATE, + PARSE_REGISTER_QUERY, +}; + +enum subscriptiontype { + NONE = 0, + XPIDF_XML, + DIALOG_INFO_XML, + CPIM_PIDF_XML, + PIDF_XML, + MWI_NOTIFICATION +}; + +static const struct cfsubscription_types { + enum subscriptiontype type; + const char * const event; + const char * const mediatype; + const char * const text; +} subscription_types[] = { + { NONE, "-", "unknown", "unknown" }, + /* RFC 4235: SIP Dialog event package */ + { DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" }, + { CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */ + { PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */ + { XPIDF_XML, "presence", "application/xpidf+xml", "xpidf+xml" }, /* Pre-RFC 3863 with MS additions */ + { MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */ +}; + +/*! \brief SIP Request methods known by Asterisk */ +enum sipmethod { + SIP_UNKNOWN, /* Unknown response */ + SIP_RESPONSE, /* Not request, response to outbound request */ + SIP_REGISTER, + SIP_OPTIONS, + SIP_NOTIFY, + SIP_INVITE, + SIP_ACK, + SIP_PRACK, /* Not supported at all */ + SIP_BYE, + SIP_REFER, + SIP_SUBSCRIBE, + SIP_MESSAGE, + SIP_UPDATE, /* We can send UPDATE; but not accept it */ + SIP_INFO, + SIP_CANCEL, + SIP_PUBLISH, /* Not supported at all */ + SIP_PING, /* Not supported at all, no standard but still implemented out there */ +}; + +/*! \brief Authentication types - proxy or www authentication + \note Endpoints, like Asterisk, should always use WWW authentication to + allow multiple authentications in the same call - to the proxy and + to the end point. +*/ +enum sip_auth_type { + PROXY_AUTH, + WWW_AUTH, +}; + +/*! \brief Authentication result from check_auth* functions */ +enum check_auth_result { + AUTH_SUCCESSFUL = 0, + AUTH_CHALLENGE_SENT = 1, + AUTH_SECRET_FAILED = -1, + AUTH_USERNAME_MISMATCH = -2, + AUTH_NOT_FOUND = -3, + AUTH_FAKE_AUTH = -4, + AUTH_UNKNOWN_DOMAIN = -5, + AUTH_PEER_NOT_DYNAMIC = -6, + AUTH_ACL_FAILED = -7, +}; + +/*! \brief States for outbound registrations (with register= lines in sip.conf */ +enum sipregistrystate { + REG_STATE_UNREGISTERED = 0, /*!< We are not registred */ + REG_STATE_REGSENT, /*!< Registration request sent */ + REG_STATE_AUTHSENT, /*!< We have tried to authenticate */ + REG_STATE_REGISTERED, /*!< Registred and done */ + REG_STATE_REJECTED, /*!< Registration rejected */ + REG_STATE_TIMEOUT, /*!< Registration timed out */ + REG_STATE_NOAUTH, /*!< We have no accepted credentials */ + REG_STATE_FAILED, /*!< Registration failed after several tries */ +}; + +#define CAN_NOT_CREATE_DIALOG 0 +#define CAN_CREATE_DIALOG 1 +#define CAN_CREATE_DIALOG_UNSUPPORTED_METHOD 2 + +/*! XXX Note that sip_methods[i].id == i must hold or the code breaks */ +static const struct cfsip_methods { + enum sipmethod id; + int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */ + char * const text; + int can_create; +} sip_methods[] = { + { SIP_UNKNOWN, RTP, "-UNKNOWN-", CAN_CREATE_DIALOG }, + { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG }, + { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG }, + { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG }, + { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG }, + { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG }, + { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG }, + { SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG }, + { SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG }, + { SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG }, + { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE", CAN_CREATE_DIALOG }, + { SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG }, + { SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG }, + { SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG }, + { SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG }, + { SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }, + { SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD } +}; + +/*! Define SIP option tags, used in Require: and Supported: headers + We need to be aware of these properties in the phones to use + the replace: header. We should not do that without knowing + that the other end supports it... + This is nothing we can configure, we learn by the dialog + Supported: header on the REGISTER (peer) or the INVITE + (other devices) + We are not using many of these today, but will in the future. + This is documented in RFC 3261 +*/ +#define SUPPORTED 1 +#define NOT_SUPPORTED 0 + +#define SIP_OPT_REPLACES (1 << 0) +#define SIP_OPT_100REL (1 << 1) +#define SIP_OPT_TIMER (1 << 2) +#define SIP_OPT_EARLY_SESSION (1 << 3) +#define SIP_OPT_JOIN (1 << 4) +#define SIP_OPT_PATH (1 << 5) +#define SIP_OPT_PREF (1 << 6) +#define SIP_OPT_PRECONDITION (1 << 7) +#define SIP_OPT_PRIVACY (1 << 8) +#define SIP_OPT_SDP_ANAT (1 << 9) +#define SIP_OPT_SEC_AGREE (1 << 10) +#define SIP_OPT_EVENTLIST (1 << 11) +#define SIP_OPT_GRUU (1 << 12) +#define SIP_OPT_TARGET_DIALOG (1 << 13) +#define SIP_OPT_NOREFERSUB (1 << 14) +#define SIP_OPT_HISTINFO (1 << 15) +#define SIP_OPT_RESPRIORITY (1 << 16) + +/*! \brief List of well-known SIP options. If we get this in a require, + we should check the list and answer accordingly. */ +static const struct cfsip_options { + int id; /*!< Bitmap ID */ + int supported; /*!< Supported by Asterisk ? */ + char * const text; /*!< Text id, as in standard */ +} sip_options[] = { /* XXX used in 3 places */ + /* RFC3891: Replaces: header for transfer */ + { SIP_OPT_REPLACES, SUPPORTED, "replaces" }, + /* One version of Polycom firmware has the wrong label */ + { SIP_OPT_REPLACES, SUPPORTED, "replace" }, + /* RFC3262: PRACK 100% reliability */ + { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" }, + /* RFC4028: SIP Session Timers */ + { SIP_OPT_TIMER, NOT_SUPPORTED, "timer" }, + /* RFC3959: SIP Early session support */ + { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" }, + /* RFC3911: SIP Join header support */ + { SIP_OPT_JOIN, NOT_SUPPORTED, "join" }, + /* RFC3327: Path support */ + { SIP_OPT_PATH, NOT_SUPPORTED, "path" }, + /* RFC3840: Callee preferences */ + { SIP_OPT_PREF, NOT_SUPPORTED, "pref" }, + /* RFC3312: Precondition support */ + { SIP_OPT_PRECONDITION, NOT_SUPPORTED, "precondition" }, + /* RFC3323: Privacy with proxies*/ + { SIP_OPT_PRIVACY, NOT_SUPPORTED, "privacy" }, + /* RFC4092: Usage of the SDP ANAT Semantics in the SIP */ + { SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp-anat" }, + /* RFC3329: Security agreement mechanism */ + { SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" }, + /* SIMPLE events: RFC4662 */ + { SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" }, + /* GRUU: Globally Routable User Agent URI's */ + { SIP_OPT_GRUU, NOT_SUPPORTED, "gruu" }, + /* RFC4538: Target-dialog */ + { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "tdialog" }, + /* Disable the REFER subscription, RFC 4488 */ + { SIP_OPT_NOREFERSUB, NOT_SUPPORTED, "norefersub" }, + /* ietf-sip-history-info-06.txt */ + { SIP_OPT_HISTINFO, NOT_SUPPORTED, "histinfo" }, + /* ietf-sip-resource-priority-10.txt */ + { SIP_OPT_RESPRIORITY, NOT_SUPPORTED, "resource-priority" }, +}; + + +/*! \brief SIP Methods we support */ +#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY" + +/*! \brief SIP Extensions we support */ +#define SUPPORTED_EXTENSIONS "replaces" + +/*! \brief Standard SIP port from RFC 3261. DO NOT CHANGE THIS */ +#define STANDARD_SIP_PORT 5060 +/* Note: in many SIP headers, absence of a port number implies port 5060, + * and this is why we cannot change the above constant. + * There is a limited number of places in asterisk where we could, + * in principle, use a different "default" port number, but + * we do not support this feature at the moment. + */ + +/* Default values, set and reset in reload_config before reading configuration */ +/* These are default values in the source. There are other recommended values in the + sip.conf.sample for new installations. These may differ to keep backwards compatibility, + yet encouraging new behaviour on new installations + */ +#define DEFAULT_CONTEXT "default" +#define DEFAULT_MOHINTERPRET "default" +#define DEFAULT_MOHSUGGEST "" +#define DEFAULT_VMEXTEN "asterisk" +#define DEFAULT_CALLERID "asterisk" +#define DEFAULT_NOTIFYMIME "application/simple-message-summary" +#define DEFAULT_MWITIME 10 +#define DEFAULT_ALLOWGUEST TRUE +#define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */ +#define DEFAULT_COMPACTHEADERS FALSE +#define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */ +#define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */ +#define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */ +#define DEFAULT_ALLOW_EXT_DOM TRUE +#define DEFAULT_REALM "asterisk" +#define DEFAULT_NOTIFYRINGING TRUE +#define DEFAULT_PEDANTIC FALSE +#define DEFAULT_AUTOCREATEPEER FALSE +#define DEFAULT_QUALIFY FALSE +#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */ +#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */ +#ifndef DEFAULT_USERAGENT +#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */ +#endif + + +/* Default setttings are used as a channel setting and as a default when + configuring devices */ +static char default_context[AST_MAX_CONTEXT]; +static char default_subscribecontext[AST_MAX_CONTEXT]; +static char default_language[MAX_LANGUAGE]; +static char default_callerid[AST_MAX_EXTENSION]; +static char default_fromdomain[AST_MAX_EXTENSION]; +static char default_notifymime[AST_MAX_EXTENSION]; +static int default_qualify; /*!< Default Qualify= setting */ +static char default_vmexten[AST_MAX_EXTENSION]; +static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */ +static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting + * a bridged channel on hold */ +static int default_maxcallbitrate; /*!< Maximum bitrate for call */ +static struct ast_codec_pref default_prefs; /*!< Default codec prefs */ + +/* Global settings only apply to the channel */ +static int global_directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */ +static int global_limitonpeers; /*!< Match call limit on peers only */ +static int global_rtautoclear; +static int global_notifyringing; /*!< Send notifications on ringing */ +static int global_notifyhold; /*!< Send notifications on hold */ +static int global_alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */ +static int srvlookup; /*!< SRV Lookup on or off. Default is on */ +static int pedanticsipchecking; /*!< Extra checking ? Default off */ +static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */ +static int global_relaxdtmf; /*!< Relax DTMF */ +static int global_rtptimeout; /*!< Time out call if no RTP */ +static int global_rtpholdtimeout; +static int global_rtpkeepalive; /*!< Send RTP keepalives */ +static int global_reg_timeout; +static int global_regattempts_max; /*!< Registration attempts before giving up */ +static int global_allowguest; /*!< allow unauthenticated users/peers to connect? */ +static int global_allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE + the global setting is in globals_flags[1] */ +static int global_mwitime; /*!< Time between MWI checks for peers */ +static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */ +static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */ +static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */ +static int compactheaders; /*!< send compact sip headers */ +static int recordhistory; /*!< Record SIP history. Off by default */ +static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */ +static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */ +static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */ +static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */ +static int allow_external_domains; /*!< Accept calls to external SIP domains? */ +static int global_callevents; /*!< Whether we send manager events or not */ +static int global_t1min; /*!< T1 roundtrip time minimum */ +static int global_autoframing; /*!< Turn autoframing on or off. */ +static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */ + +static int global_matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */ + +/*! \brief Codecs that we support by default: */ +static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263; + +/*! \brief Global list of addresses dynamic peers are not allowed to use */ +static struct ast_ha *global_contact_ha = NULL; +static int global_dynamic_exclude_static = 0; + +/* Object counters */ +static int suserobjs = 0; /*!< Static users */ +static int ruserobjs = 0; /*!< Realtime users */ +static int speerobjs = 0; /*!< Statis peers */ +static int rpeerobjs = 0; /*!< Realtime peers */ +static int apeerobjs = 0; /*!< Autocreated peer objects */ +static int regobjs = 0; /*!< Registry objects */ + +static struct ast_flags global_flags[2] = {{0}}; /*!< global SIP_ flags */ + +/*! \brief Protect the SIP dialog list (of sip_pvt's) */ +AST_MUTEX_DEFINE_STATIC(iflock); + +/*! \brief Protect the monitoring thread, so only one process can kill or start it, and not + when it's doing something critical. */ +AST_MUTEX_DEFINE_STATIC(netlock); + +AST_MUTEX_DEFINE_STATIC(monlock); + +AST_MUTEX_DEFINE_STATIC(sip_reload_lock); + +/*! \brief This is the thread for the monitor which checks for input on the channels + which are not currently in use. */ +static pthread_t monitor_thread = AST_PTHREADT_NULL; + +static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */ +static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */ + +static struct sched_context *sched; /*!< The scheduling context */ +static struct io_context *io; /*!< The IO context */ +static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */ + +#define DEC_CALL_LIMIT 0 +#define INC_CALL_LIMIT 1 +#define DEC_CALL_RINGING 2 +#define INC_CALL_RINGING 3 + +/*! \brief sip_request: The data grabbed from the UDP socket */ +struct sip_request { + char *rlPart1; /*!< SIP Method Name or "SIP/2.0" protocol version */ + char *rlPart2; /*!< The Request URI or Response Status */ + int len; /*!< Length */ + int headers; /*!< # of SIP Headers */ + int method; /*!< Method of this request */ + int lines; /*!< Body Content */ + unsigned int flags; /*!< SIP_PKT Flags for this packet */ + char *header[SIP_MAX_HEADERS]; + char *line[SIP_MAX_LINES]; + char data[SIP_MAX_PACKET]; + unsigned int sdp_start; /*!< the line number where the SDP begins */ + unsigned int sdp_end; /*!< the line number where the SDP ends */ + AST_LIST_ENTRY(sip_request) next; +}; + +/* + * A sip packet is stored into the data[] buffer, with the header followed + * by an empty line and the body of the message. + * On outgoing packets, data is accumulated in data[] with len reflecting + * the next available byte, headers and lines count the number of lines + * in both parts. There are no '\0' in data[0..len-1]. + * + * On received packet, the input read from the socket is copied into data[], + * len is set and the string is NUL-terminated. Then a parser fills up + * the other fields -header[] and line[] to point to the lines of the + * message, rlPart1 and rlPart2 parse the first lnie as below: + * + * Requests have in the first line METHOD URI SIP/2.0 + * rlPart1 = method; rlPart2 = uri; + * Responses have in the first line SIP/2.0 code description + * rlPart1 = SIP/2.0; rlPart2 = code + description; + * + */ + +/*! \brief structure used in transfers */ +struct sip_dual { + struct ast_channel *chan1; /*!< First channel involved */ + struct ast_channel *chan2; /*!< Second channel involved */ + struct sip_request req; /*!< Request that caused the transfer (REFER) */ + int seqno; /*!< Sequence number */ +}; + +struct sip_pkt; + +/*! \brief Parameters to the transmit_invite function */ +struct sip_invite_param { + const char *distinctive_ring; /*!< Distinctive ring header */ + int addsipheaders; /*!< Add extra SIP headers */ + const char *uri_options; /*!< URI options to add to the URI */ + const char *vxml_url; /*!< VXML url for Cisco phones */ + char *auth; /*!< Authentication */ + char *authheader; /*!< Auth header */ + enum sip_auth_type auth_type; /*!< Authentication type */ + const char *replaces; /*!< Replaces header for call transfers */ + int transfer; /*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */ +}; + +/*! \brief Structure to save routing information for a SIP session */ +struct sip_route { + struct sip_route *next; + char hop[0]; +}; + +/*! \brief Modes for SIP domain handling in the PBX */ +enum domain_mode { + SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */ + SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */ +}; + +/*! \brief Domain data structure. + \note In the future, we will connect this to a configuration tree specific + for this domain +*/ +struct domain { + char domain[MAXHOSTNAMELEN]; /*!< SIP domain we are responsible for */ + char context[AST_MAX_EXTENSION]; /*!< Incoming context for this domain */ + enum domain_mode mode; /*!< How did we find this domain? */ + AST_LIST_ENTRY(domain) list; /*!< List mechanics */ +}; + +static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */ + + +/*! \brief sip_history: Structure for saving transactions within a SIP dialog */ +struct sip_history { + AST_LIST_ENTRY(sip_history) list; + char event[0]; /* actually more, depending on needs */ +}; + +AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */ + +/*! \brief sip_auth: Credentials for authentication to other SIP services */ +struct sip_auth { + char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */ + char username[256]; /*!< Username */ + char secret[256]; /*!< Secret */ + char md5secret[256]; /*!< MD5Secret */ + struct sip_auth *next; /*!< Next auth structure in list */ +}; + +/*--- Various flags for the flags field in the pvt structure */ +#define SIP_ALREADYGONE (1 << 0) /*!< Whether or not we've already been destroyed by our peer */ +#define SIP_NEEDDESTROY (1 << 1) /*!< if we need to be destroyed by the monitor thread */ +#define SIP_NOVIDEO (1 << 2) /*!< Didn't get video in invite, don't offer */ +#define SIP_RINGING (1 << 3) /*!< Have sent 180 ringing */ +#define SIP_PROGRESS_SENT (1 << 4) /*!< Have sent 183 message progress */ +#define SIP_NEEDREINVITE (1 << 5) /*!< Do we need to send another reinvite? */ +#define SIP_PENDINGBYE (1 << 6) /*!< Need to send bye after we ack? */ +#define SIP_GOTREFER (1 << 7) /*!< Got a refer? */ +#define SIP_PROMISCREDIR (1 << 8) /*!< Promiscuous redirection */ +#define SIP_TRUSTRPID (1 << 9) /*!< Trust RPID headers? */ +#define SIP_USEREQPHONE (1 << 10) /*!< Add user=phone to numeric URI. Default off */ +#define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */ +#define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */ +#define SIP_OUTGOING (1 << 13) /*!< Direction of the last transaction in this dialog */ +#define SIP_FREE_BIT (1 << 14) /*!< ---- */ +#define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */ +#define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */ +#define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */ +#define SIP_DTMF_INBAND (1 << 16) /*!< DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */ +#define SIP_DTMF_INFO (2 << 16) /*!< DTMF Support: SIP Info messages - "info" */ +#define SIP_DTMF_AUTO (3 << 16) /*!< DTMF Support: AUTO switch between rfc2833 and in-band DTMF */ +/* NAT settings */ +#define SIP_NAT (3 << 18) /*!< four settings, uses two bits */ +#define SIP_NAT_NEVER (0 << 18) /*!< No nat support */ +#define SIP_NAT_RFC3581 (1 << 18) /*!< NAT RFC3581 */ +#define SIP_NAT_ROUTE (2 << 18) /*!< NAT Only ROUTE */ +#define SIP_NAT_ALWAYS (3 << 18) /*!< NAT Both ROUTE and RFC3581 */ +/* re-INVITE related settings */ +#define SIP_REINVITE (7 << 20) /*!< three bits used */ +#define SIP_CAN_REINVITE (1 << 20) /*!< allow peers to be reinvited to send media directly p2p */ +#define SIP_CAN_REINVITE_NAT (2 << 20) /*!< allow media reinvite when new peer is behind NAT */ +#define SIP_REINVITE_UPDATE (4 << 20) /*!< use UPDATE (RFC3311) when reinviting this peer */ +/* "insecure" settings */ +#define SIP_INSECURE_PORT (1 << 23) /*!< don't require matching port for incoming requests */ +#define SIP_INSECURE_INVITE (1 << 24) /*!< don't require authentication for incoming INVITEs */ +/* Sending PROGRESS in-band settings */ +#define SIP_PROG_INBAND (3 << 25) /*!< three settings, uses two bits */ +#define SIP_PROG_INBAND_NEVER (0 << 25) +#define SIP_PROG_INBAND_NO (1 << 25) +#define SIP_PROG_INBAND_YES (2 << 25) +#define SIP_NO_HISTORY (1 << 27) /*!< Suppress recording request/response history */ +#define SIP_CALL_LIMIT (1 << 28) /*!< Call limit enforced for this call */ +#define SIP_SENDRPID (1 << 29) /*!< Remote Party-ID Support */ +#define SIP_INC_COUNT (1 << 30) /*!< Did this connection increment the counter of in-use calls? */ +#define SIP_G726_NONSTANDARD (1 << 31) /*!< Use non-standard packing for G726-32 data */ + +#define SIP_FLAGS_TO_COPY \ + (SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \ + SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \ + SIP_USEREQPHONE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE) + +/*--- a new page of flags (for flags[1] */ +/* realtime flags */ +#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0) +#define SIP_PAGE2_RTUPDATE (1 << 1) +#define SIP_PAGE2_RTAUTOCLEAR (1 << 2) +#define SIP_PAGE2_RT_FROMCONTACT (1 << 4) +#define SIP_PAGE2_RTSAVE_SYSNAME (1 << 5) +/* Space for addition of other realtime flags in the future */ +#define SIP_PAGE2_STATECHANGEQUEUE (1 << 9) /*!< D: Unsent state pending change exists */ +#define SIP_PAGE2_IGNOREREGEXPIRE (1 << 10) +#define SIP_PAGE2_DEBUG (3 << 11) +#define SIP_PAGE2_DEBUG_CONFIG (1 << 11) +#define SIP_PAGE2_DEBUG_CONSOLE (1 << 12) +#define SIP_PAGE2_DYNAMIC (1 << 13) /*!< Dynamic Peers register with Asterisk */ +#define SIP_PAGE2_SELFDESTRUCT (1 << 14) /*!< Automatic peers need to destruct themselves */ +#define SIP_PAGE2_VIDEOSUPPORT (1 << 15) +#define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 16) /*!< Allow subscriptions from this peer? */ +#define SIP_PAGE2_ALLOWOVERLAP (1 << 17) /*!< Allow overlap dialing ? */ +#define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 18) /*!< Only issue MWI notification if subscribed to */ +#define SIP_PAGE2_INC_RINGING (1 << 19) /*!< Did this connection increment the counter of in-use calls? */ +#define SIP_PAGE2_T38SUPPORT (7 << 20) /*!< T38 Fax Passthrough Support */ +#define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 20) /*!< 20: T38 Fax Passthrough Support */ +#define SIP_PAGE2_T38SUPPORT_RTP (2 << 20) /*!< 21: T38 Fax Passthrough Support (not implemented) */ +#define SIP_PAGE2_T38SUPPORT_TCP (4 << 20) /*!< 22: T38 Fax Passthrough Support (not implemented) */ +#define SIP_PAGE2_CALL_ONHOLD (3 << 23) /*!< Call states */ +#define SIP_PAGE2_CALL_ONHOLD_ACTIVE (1 << 23) /*!< 23: Active hold */ +#define SIP_PAGE2_CALL_ONHOLD_ONEDIR (2 << 23) /*!< 23: One directional hold */ +#define SIP_PAGE2_CALL_ONHOLD_INACTIVE (3 << 23) /*!< 23: Inactive hold */ +#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< 25: ???? */ +#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< 26: Buggy CISCO MWI fix */ +#define SIP_PAGE2_OUTGOING_CALL (1 << 27) /*!< 27: Is this an outgoing call? */ +#define SIP_PAGE2_UDPTL_DESTINATION (1 << 28) /*!< 28: Use source IP of RTP as destination if NAT is enabled */ +#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 29) /*!< 29: Has a dialog been established? */ + +#define SIP_PAGE2_FLAGS_TO_COPY \ + (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \ + SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_UDPTL_DESTINATION) + +/* SIP packet flags */ +#define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */ +#define SIP_PKT_WITH_TOTAG (1 << 1) /*!< This packet has a to-tag */ +#define SIP_PKT_IGNORE (1 << 2) /*!< This is a re-transmit, ignore it */ +#define SIP_PKT_IGNORE_RESP (1 << 3) /*!< Resp ignore - ??? */ +#define SIP_PKT_IGNORE_REQ (1 << 4) /*!< Req ignore - ??? */ + +/* T.38 set of flags */ +#define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0 (unset)*/ +#define T38FAX_TRANSCODING_MMR (1 << 1) /*!< Default: 0 (unset)*/ +#define T38FAX_TRANSCODING_JBIG (1 << 2) /*!< Default: 0 (unset)*/ +/* Rate management */ +#define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF (0 << 3) +#define T38FAX_RATE_MANAGEMENT_LOCAL_TCF (1 << 3) /*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */ +/* UDP Error correction */ +#define T38FAX_UDP_EC_NONE (0 << 4) /*!< two bits, if unset NO t38UDPEC field in T38 SDP*/ +#define T38FAX_UDP_EC_FEC (1 << 4) /*!< Set for t38UDPFEC */ +#define T38FAX_UDP_EC_REDUNDANCY (2 << 4) /*!< Set for t38UDPRedundancy */ +/* T38 Spec version */ +#define T38FAX_VERSION (3 << 6) /*!< two bits, 2 values so far, up to 4 values max */ +#define T38FAX_VERSION_0 (0 << 6) /*!< Version 0 */ +#define T38FAX_VERSION_1 (1 << 6) /*!< Version 1 */ +/* Maximum Fax Rate */ +#define T38FAX_RATE_2400 (1 << 8) /*!< 2400 bps t38FaxRate */ +#define T38FAX_RATE_4800 (1 << 9) /*!< 4800 bps t38FaxRate */ +#define T38FAX_RATE_7200 (1 << 10) /*!< 7200 bps t38FaxRate */ +#define T38FAX_RATE_9600 (1 << 11) /*!< 9600 bps t38FaxRate */ +#define T38FAX_RATE_12000 (1 << 12) /*!< 12000 bps t38FaxRate */ +#define T38FAX_RATE_14400 (1 << 13) /*!< 14400 bps t38FaxRate */ + +/*!< This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */ +static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600; + +#define sipdebug ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG) +#define sipdebug_config ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG) +#define sipdebug_console ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE) + +/*! \brief T38 States for a call */ +enum t38state { + T38_DISABLED = 0, /*!< Not enabled */ + T38_LOCAL_DIRECT, /*!< Offered from local */ + T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */ + T38_PEER_DIRECT, /*!< Offered from peer */ + T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */ + T38_ENABLED /*!< Negotiated (enabled) */ +}; + +/*! \brief T.38 channel settings (at some point we need to make this alloc'ed */ +struct t38properties { + struct ast_flags t38support; /*!< Flag for udptl, rtp or tcp support for this session */ + int capability; /*!< Our T38 capability */ + int peercapability; /*!< Peers T38 capability */ + int jointcapability; /*!< Supported T38 capability at both ends */ + enum t38state state; /*!< T.38 state */ +}; + +/*! \brief Parameters to know status of transfer */ +enum referstatus { + REFER_IDLE, /*!< No REFER is in progress */ + REFER_SENT, /*!< Sent REFER to transferee */ + REFER_RECEIVED, /*!< Received REFER from transferer */ + REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING */ + REFER_ACCEPTED, /*!< Accepted by transferee */ + REFER_RINGING, /*!< Target Ringing */ + REFER_200OK, /*!< Answered by transfer target */ + REFER_FAILED, /*!< REFER declined - go on */ + REFER_NOAUTH /*!< We had no auth for REFER */ +}; + +static const struct c_referstatusstring { + enum referstatus status; + char *text; +} referstatusstrings[] = { + { REFER_IDLE, "<none>" }, + { REFER_SENT, "Request sent" }, + { REFER_RECEIVED, "Request received" }, + { REFER_ACCEPTED, "Accepted" }, + { REFER_RINGING, "Target ringing" }, + { REFER_200OK, "Done" }, + { REFER_FAILED, "Failed" }, + { REFER_NOAUTH, "Failed - auth failure" } +} ; + +/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed */ +/* OEJ: Should be moved to string fields */ +struct sip_refer { + char refer_to[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO extension */ + char refer_to_domain[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO domain */ + char refer_to_urioption[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO uri options */ + char refer_to_context[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO context */ + char referred_by[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */ + char referred_by_name[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */ + char refer_contact[AST_MAX_EXTENSION]; /*!< Place to store Contact info from a REFER extension */ + char replaces_callid[SIPBUFSIZE]; /*!< Replace info: callid */ + char replaces_callid_totag[SIPBUFSIZE/2]; /*!< Replace info: to-tag */ + char replaces_callid_fromtag[SIPBUFSIZE/2]; /*!< Replace info: from-tag */ + struct sip_pvt *refer_call; /*!< Call we are referring */ + int attendedtransfer; /*!< Attended or blind transfer? */ + int localtransfer; /*!< Transfer to local domain? */ + enum referstatus status; /*!< REFER status */ +}; + +/*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe */ +static struct sip_pvt { + ast_mutex_t lock; /*!< Dialog private lock */ + int method; /*!< SIP method that opened this dialog */ + enum invitestates invitestate; /*!< The state of the INVITE transaction only */ + AST_DECLARE_STRING_FIELDS( + AST_STRING_FIELD(callid); /*!< Global CallID */ + AST_STRING_FIELD(randdata); /*!< Random data */ + AST_STRING_FIELD(accountcode); /*!< Account code */ + AST_STRING_FIELD(realm); /*!< Authorization realm */ + AST_STRING_FIELD(nonce); /*!< Authorization nonce */ + AST_STRING_FIELD(opaque); /*!< Opaque nonsense */ + AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */ + AST_STRING_FIELD(domain); /*!< Authorization domain */ + AST_STRING_FIELD(from); /*!< The From: header */ + AST_STRING_FIELD(useragent); /*!< User agent in SIP request */ + AST_STRING_FIELD(exten); /*!< Extension where to start */ + AST_STRING_FIELD(context); /*!< Context for this call */ + AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */ + AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */ + AST_STRING_FIELD(fromdomain); /*!< Domain to show in the from field */ + AST_STRING_FIELD(fromuser); /*!< User to show in the user field */ + AST_STRING_FIELD(fromname); /*!< Name to show in the user field */ + AST_STRING_FIELD(tohost); /*!< Host we should put in the "to" field */ + AST_STRING_FIELD(language); /*!< Default language for this call */ + AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */ + AST_STRING_FIELD(mohsuggest); /*!< MOH class to suggest when putting a peer on hold */ + AST_STRING_FIELD(rdnis); /*!< Referring DNIS */ + AST_STRING_FIELD(theirtag); /*!< Their tag */ + AST_STRING_FIELD(username); /*!< [user] name */ + AST_STRING_FIELD(peername); /*!< [peer] name, not set if [user] */ + AST_STRING_FIELD(authname); /*!< Who we use for authentication */ + AST_STRING_FIELD(uri); /*!< Original requested URI */ + AST_STRING_FIELD(okcontacturi); /*!< URI from the 200 OK on INVITE */ + AST_STRING_FIELD(peersecret); /*!< Password */ + AST_STRING_FIELD(peermd5secret); + AST_STRING_FIELD(cid_num); /*!< Caller*ID number */ + AST_STRING_FIELD(cid_name); /*!< Caller*ID name */ + AST_STRING_FIELD(via); /*!< Via: header */ + AST_STRING_FIELD(fullcontact); /*!< The Contact: that the UA registers with us */ + AST_STRING_FIELD(our_contact); /*!< Our contact header */ + AST_STRING_FIELD(rpid); /*!< Our RPID header */ + AST_STRING_FIELD(rpid_from); /*!< Our RPID From header */ + ); + unsigned int ocseq; /*!< Current outgoing seqno */ + unsigned int icseq; /*!< Current incoming seqno */ + ast_group_t callgroup; /*!< Call group */ + ast_group_t pickupgroup; /*!< Pickup group */ + int lastinvite; /*!< Last Cseq of invite */ + int lastnoninvite; /*!< Last Cseq of non-invite */ + struct ast_flags flags[2]; /*!< SIP_ flags */ + int timer_t1; /*!< SIP timer T1, ms rtt */ + unsigned int sipoptions; /*!< Supported SIP options on the other end */ + struct ast_codec_pref prefs; /*!< codec prefs */ + int capability; /*!< Special capability (codec) */ + int jointcapability; /*!< Supported capability at both ends (codecs) */ + int peercapability; /*!< Supported peer capability */ + int prefcodec; /*!< Preferred codec (outbound only) */ + int noncodeccapability; /*!< DTMF RFC2833 telephony-event */ + int jointnoncodeccapability; /*!< Joint Non codec capability */ + int redircodecs; /*!< Redirect codecs */ + int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */ + struct t38properties t38; /*!< T38 settings */ + struct sockaddr_in udptlredirip; /*!< Where our T.38 UDPTL should be going if not to us */ + struct ast_udptl *udptl; /*!< T.38 UDPTL session */ + int callingpres; /*!< Calling presentation */ + int authtries; /*!< Times we've tried to authenticate */ + int expiry; /*!< How long we take to expire */ + long branch; /*!< The branch identifier of this session */ + long invite_branch; /*!< The branch used when we sent the initial INVITE */ + char tag[11]; /*!< Our tag for this session */ + int sessionid; /*!< SDP Session ID */ + int sessionversion; /*!< SDP Session Version */ + struct sockaddr_in sa; /*!< Our peer */ + struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */ + struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */ + time_t lastrtprx; /*!< Last RTP received */ + time_t lastrtptx; /*!< Last RTP sent */ + int rtptimeout; /*!< RTP timeout time */ + struct sockaddr_in recv; /*!< Received as */ + struct in_addr ourip; /*!< Our IP */ + struct ast_channel *owner; /*!< Who owns us (if we have an owner) */ + struct sip_route *route; /*!< Head of linked list of routing steps (fm Record-Route) */ + int route_persistant; /*!< Is this the "real" route? */ + struct sip_auth *peerauth; /*!< Realm authentication */ + int noncecount; /*!< Nonce-count */ + char lastmsg[256]; /*!< Last Message sent/received */ + int amaflags; /*!< AMA Flags */ + int pendinginvite; /*!< Any pending INVITE or state NOTIFY (in subscribe pvt's) ? (seqno of this) */ + struct sip_request initreq; /*!< Request that opened the latest transaction + within this SIP dialog */ + + int maxtime; /*!< Max time for first response */ + int initid; /*!< Auto-congest ID if appropriate (scheduler) */ + int waitid; /*!< Wait ID for scheduler after 491 or other delays */ + int autokillid; /*!< Auto-kill ID (scheduler) */ + enum transfermodes allowtransfer; /*!< REFER: restriction scheme */ + struct sip_refer *refer; /*!< REFER: SIP transfer data structure */ + enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */ + int stateid; /*!< SUBSCRIBE: ID for devicestate subscriptions */ + int laststate; /*!< SUBSCRIBE: Last known extension state */ + int dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */ + + struct ast_dsp *vad; /*!< Voice Activation Detection dsp */ + + struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one + Used in peerpoke, mwi subscriptions */ + struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */ + struct ast_rtp *rtp; /*!< RTP Session */ + struct ast_rtp *vrtp; /*!< Video RTP session */ + struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */ + struct sip_history_head *history; /*!< History of this SIP dialog */ + size_t history_entries; /*!< Number of entires in the history */ + struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */ + AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */ + int request_queue_sched_id; /*!< Scheduler ID of any scheduled action to process queued requests */ + struct sip_pvt *next; /*!< Next dialog in chain */ + struct sip_invite_param *options; /*!< Options for INVITE */ + int autoframing; +} *iflist = NULL; + +/*! Max entires in the history list for a sip_pvt */ +#define MAX_HISTORY_ENTRIES 50 + +#define FLAG_RESPONSE (1 << 0) +#define FLAG_FATAL (1 << 1) + +/*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission */ +struct sip_pkt { + struct sip_pkt *next; /*!< Next packet in linked list */ + int retrans; /*!< Retransmission number */ + int method; /*!< SIP method for this packet */ + int seqno; /*!< Sequence number */ + unsigned int flags; /*!< non-zero if this is a response packet (e.g. 200 OK) */ + struct sip_pvt *owner; /*!< Owner AST call */ + int retransid; /*!< Retransmission ID */ + int timer_a; /*!< SIP timer A, retransmission timer */ + int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */ + int packetlen; /*!< Length of packet */ + char data[0]; +}; + +/*! \brief Structure for SIP user data. User's place calls to us */ +struct sip_user { + /* Users who can access various contexts */ + ASTOBJ_COMPONENTS(struct sip_user); + char secret[80]; /*!< Password */ + char md5secret[80]; /*!< Password in md5 */ + char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */ + char subscribecontext[AST_MAX_CONTEXT]; /* Default context for subscriptions */ + char cid_num[80]; /*!< Caller ID num */ + char cid_name[80]; /*!< Caller ID name */ + char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */ + char language[MAX_LANGUAGE]; /*!< Default language for this user */ + char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */ + char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */ + char useragent[256]; /*!< User agent in SIP request */ + struct ast_codec_pref prefs; /*!< codec prefs */ + ast_group_t callgroup; /*!< Call group */ + ast_group_t pickupgroup; /*!< Pickup Group */ + unsigned int sipoptions; /*!< Supported SIP options */ + struct ast_flags flags[2]; /*!< SIP_ flags */ + int amaflags; /*!< AMA flags for billing */ + int callingpres; /*!< Calling id presentation */ + int capability; /*!< Codec capability */ + int inUse; /*!< Number of calls in use */ + int call_limit; /*!< Limit of concurrent calls */ + enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */ + struct ast_ha *ha; /*!< ACL setting */ + struct ast_variable *chanvars; /*!< Variables to set for channel created by user */ + int maxcallbitrate; /*!< Maximum Bitrate for a video call */ + int autoframing; +}; + +/*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host) */ +/* XXX field 'name' must be first otherwise sip_addrcmp() will fail */ +struct sip_peer { + ASTOBJ_COMPONENTS(struct sip_peer); /*!< name, refcount, objflags, object pointers */ + /*!< peer->name is the unique name of this object */ + char secret[80]; /*!< Password */ + char md5secret[80]; /*!< Password in MD5 */ + struct sip_auth *auth; /*!< Realm authentication list */ + char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */ + char subscribecontext[AST_MAX_CONTEXT]; /*!< Default context for subscriptions */ + char username[80]; /*!< Temporary username until registration */ + char accountcode[AST_MAX_ACCOUNT_CODE]; /*!< Account code */ + int amaflags; /*!< AMA Flags (for billing) */ + char tohost[MAXHOSTNAMELEN]; /*!< If not dynamic, IP address */ + char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */ + char fromuser[80]; /*!< From: user when calling this peer */ + char fromdomain[MAXHOSTNAMELEN]; /*!< From: domain when calling this peer */ + char fullcontact[256]; /*!< Contact registered with us (not in sip.conf) */ + char cid_num[80]; /*!< Caller ID num */ + char cid_name[80]; /*!< Caller ID name */ + int callingpres; /*!< Calling id presentation */ + int inUse; /*!< Number of calls in use */ + int inRinging; /*!< Number of calls ringing */ + int onHold; /*!< Peer has someone on hold */ + int call_limit; /*!< Limit of concurrent calls */ + enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */ + char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/ + char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */ + char language[MAX_LANGUAGE]; /*!< Default language for prompts */ + char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */ + char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */ + char useragent[256]; /*!< User agent in SIP request (saved from registration) */ + struct ast_codec_pref prefs; /*!< codec prefs */ + int lastmsgssent; + time_t lastmsgcheck; /*!< Last time we checked for MWI */ + unsigned int sipoptions; /*!< Supported SIP options */ + struct ast_flags flags[2]; /*!< SIP_ flags */ + int expire; /*!< When to expire this peer registration */ + int capability; /*!< Codec capability */ + int rtptimeout; /*!< RTP timeout */ + int rtpholdtimeout; /*!< RTP Hold Timeout */ + int rtpkeepalive; /*!< Send RTP packets for keepalive */ + ast_group_t callgroup; /*!< Call group */ + ast_group_t pickupgroup; /*!< Pickup group */ + struct sockaddr_in addr; /*!< IP address of peer */ + int maxcallbitrate; /*!< Maximum Bitrate for a video call */ + + /* Qualification */ + struct sip_pvt *call; /*!< Call pointer */ + int pokeexpire; /*!< When to expire poke (qualify= checking) */ + int lastms; /*!< How long last response took (in ms), or -1 for no response */ + int maxms; /*!< Max ms we will accept for the host to be up, 0 to not monitor */ + struct timeval ps; /*!< Ping send time */ + + struct sockaddr_in defaddr; /*!< Default IP address, used until registration */ + struct ast_ha *ha; /*!< Access control list */ + struct ast_ha *contactha; /*!< Restrict what IPs are allowed in the Contact header (for registration) */ + struct ast_variable *chanvars; /*!< Variables to set for channel created by user */ + struct sip_pvt *mwipvt; /*!< Subscription for MWI */ + int lastmsg; + int autoframing; +}; + + + +/*! \brief Registrations with other SIP proxies */ +struct sip_registry { + ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1); + AST_DECLARE_STRING_FIELDS( + AST_STRING_FIELD(callid); /*!< Global Call-ID */ + AST_STRING_FIELD(realm); /*!< Authorization realm */ + AST_STRING_FIELD(nonce); /*!< Authorization nonce */ + AST_STRING_FIELD(opaque); /*!< Opaque nonsense */ + AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */ + AST_STRING_FIELD(domain); /*!< Authorization domain */ + AST_STRING_FIELD(username); /*!< Who we are registering as */ + AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */ + AST_STRING_FIELD(hostname); /*!< Domain or host we register to */ + AST_STRING_FIELD(secret); /*!< Password in clear text */ + AST_STRING_FIELD(md5secret); /*!< Password in md5 */ + AST_STRING_FIELD(contact); /*!< Contact extension */ + AST_STRING_FIELD(random); + ); + int portno; /*!< Optional port override */ + int expire; /*!< Sched ID of expiration */ + int regattempts; /*!< Number of attempts (since the last success) */ + int timeout; /*!< sched id of sip_reg_timeout */ + int refresh; /*!< How often to refresh */ + struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */ + enum sipregistrystate regstate; /*!< Registration state (see above) */ + time_t regtime; /*!< Last succesful registration time */ + int callid_valid; /*!< 0 means we haven't chosen callid for this registry yet. */ + unsigned int ocseq; /*!< Sequence number we got to for REGISTERs for this registry */ + struct sockaddr_in us; /*!< Who the server thinks we are */ + int noncecount; /*!< Nonce-count */ + char lastmsg[256]; /*!< Last Message sent/received */ +}; + +/* --- Linked lists of various objects --------*/ + +/*! \brief The user list: Users and friends */ +static struct ast_user_list { + ASTOBJ_CONTAINER_COMPONENTS(struct sip_user); +} userl; + +/*! \brief The peer list: Peers and Friends */ +static struct ast_peer_list { + ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer); +} peerl; + +/*! \brief The register list: Other SIP proxys we register with and place calls to */ +static struct ast_register_list { + ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry); + int recheck; +} regl; + +static void temp_pvt_cleanup(void *); + +/*! \brief A per-thread temporary pvt structure */ +AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup); + +#ifdef LOW_MEMORY +static void ts_ast_rtp_destroy(void *); + +AST_THREADSTORAGE_CUSTOM(ts_audio_rtp, ts_audio_rtp_init, ts_ast_rtp_destroy); +AST_THREADSTORAGE_CUSTOM(ts_video_rtp, ts_video_rtp_init, ts_ast_rtp_destroy); +#endif + +/*! \todo Move the sip_auth list to AST_LIST */ +static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */ + + +/* --- Sockets and networking --------------*/ +static int sipsock = -1; /*!< Main socket for SIP network communication */ +static struct sockaddr_in bindaddr = { 0, }; /*!< The address we bind to */ +static struct sockaddr_in externip; /*!< External IP address if we are behind NAT */ +static char externhost[MAXHOSTNAMELEN]; /*!< External host name (possibly with dynamic DNS and DHCP */ +static time_t externexpire = 0; /*!< Expiration counter for re-resolving external host name in dynamic DNS */ +static int externrefresh = 10; +static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */ +static struct in_addr __ourip; +static struct sockaddr_in outboundproxyip; +static int ourport; +static struct sockaddr_in debugaddr; + +static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */ + +/*---------------------------- Forward declarations of functions in chan_sip.c */ +/*! \note This is added to help splitting up chan_sip.c into several files + in coming releases */ + +/*--- PBX interface functions */ +static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause); +static int sip_devicestate(void *data); +static int sip_sendtext(struct ast_channel *ast, const char *text); +static int sip_call(struct ast_channel *ast, char *dest, int timeout); +static int sip_hangup(struct ast_channel *ast); +static int sip_answer(struct ast_channel *ast); +static struct ast_frame *sip_read(struct ast_channel *ast); +static int sip_write(struct ast_channel *ast, struct ast_frame *frame); +static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen); +static int sip_transfer(struct ast_channel *ast, const char *dest); +static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan); +static int sip_senddigit_begin(struct ast_channel *ast, char digit); +static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration); + +/*--- Transmitting responses and requests */ +static int sipsock_read(int *id, int fd, short events, void *ignore); +static int __sip_xmit(struct sip_pvt *p, char *data, int len); +static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod); +static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable); +static int retrans_pkt(const void *data); +static int transmit_sip_request(struct sip_pvt *p, struct sip_request *req); +static int transmit_response_using_temp(ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg); +static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req); +static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req); +static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req); +static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable); +static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported); +static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale); +static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable); +static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, int reliable); +static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch); +static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch); +static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init); +static int transmit_reinvite_with_sdp(struct sip_pvt *p); +static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration); +static int transmit_info_with_vidupdate(struct sip_pvt *p); +static int transmit_message_with_text(struct sip_pvt *p, const char *text); +static int transmit_refer(struct sip_pvt *p, const char *dest); +static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten); +static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate); +static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader); +static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno); +static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno); +static void copy_request(struct sip_request *dst, const struct sip_request *src); +static void receive_message(struct sip_pvt *p, struct sip_request *req); +static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req); +static int sip_send_mwi_to_peer(struct sip_peer *peer); +static int does_peer_need_mwi(struct sip_peer *peer); + +/*--- Dialog management */ +static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin, + int useglobal_nat, const int intended_method); +static int __sip_autodestruct(const void *data); +static void sip_scheddestroy(struct sip_pvt *p, int ms); +static int sip_cancel_destroy(struct sip_pvt *p); +static void sip_destroy(struct sip_pvt *p); +static int __sip_destroy(struct sip_pvt *p, int lockowner); +static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod); +static void __sip_pretend_ack(struct sip_pvt *p); +static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod); +static int auto_congest(const void *nothing); +static int update_call_counter(struct sip_pvt *fup, int event); +static int hangup_sip2cause(int cause); +static const char *hangup_cause2sip(int cause); +static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method); +static void free_old_route(struct sip_route *route); +static void list_route(struct sip_route *route); +static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards); +static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin, + struct sip_request *req, char *uri); +static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag); +static void check_pendings(struct sip_pvt *p); +static void *sip_park_thread(void *stuff); +static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno); +static int sip_sipredirect(struct sip_pvt *p, const char *dest); + +/*--- Codec handling / SDP */ +static void try_suggested_sip_codec(struct sip_pvt *p); +static const char* get_sdp_iterate(int* start, struct sip_request *req, const char *name); +static const char *get_sdp(struct sip_request *req, const char *name); +static int find_sdp(struct sip_request *req); +static int process_sdp(struct sip_pvt *p, struct sip_request *req); +static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate, + char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, + int debug, int *min_packet_size); +static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate, + char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, + int debug); +static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p); +static void stop_media_flows(struct sip_pvt *p); + +/*--- Authentication stuff */ +static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len); +static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len); +static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username, + const char *secret, const char *md5secret, int sipmethod, + char *uri, enum xmittype reliable, int ignore); +static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req, + int sipmethod, char *uri, enum xmittype reliable, + struct sockaddr_in *sin, struct sip_peer **authpeer); +static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin); + +/*--- Domain handling */ +static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */ +static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context); +static void clear_sip_domains(void); + +/*--- SIP realm authentication */ +static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); +static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */ +static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm); + +/*--- Misc functions */ +static int sip_do_reload(enum channelreloadreason reason); +static int reload_config(enum channelreloadreason reason); +static int expire_register(const void *data); +static void *do_monitor(void *data); +static int restart_monitor(void); +static int sip_send_mwi_to_peer(struct sip_peer *peer); +static int sip_addrcmp(char *name, struct sockaddr_in *sin); /* Support for peer matching */ +static int sip_refer_allocate(struct sip_pvt *p); +static void ast_quiet_chan(struct ast_channel *chan); +static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target); + +/*--- Device monitoring and Device/extension state handling */ +static int cb_extensionstate(char *context, char* exten, int state, void *data); +static int sip_devicestate(void *data); +static int sip_poke_noanswer(const void *data); +static int sip_poke_peer(struct sip_peer *peer); +static void sip_poke_all_peers(void); +static void sip_peer_hold(struct sip_pvt *p, int hold); + +/*--- Applications, functions, CLI and manager command helpers */ +static const char *sip_nat_mode(const struct sip_pvt *p); +static int sip_show_inuse(int fd, int argc, char *argv[]); +static char *transfermode2str(enum transfermodes mode) attribute_const; +static char *nat2str(int nat) attribute_const; +static int peer_status(struct sip_peer *peer, char *status, int statuslen); +static int sip_show_users(int fd, int argc, char *argv[]); +static int _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]); +static int sip_show_peers(int fd, int argc, char *argv[]); +static int sip_show_objects(int fd, int argc, char *argv[]); +static void print_group(int fd, ast_group_t group, int crlf); +static const char *dtmfmode2str(int mode) attribute_const; +static const char *insecure2str(int port, int invite) attribute_const; +static void cleanup_stale_contexts(char *new, char *old); +static void print_codec_to_cli(int fd, struct ast_codec_pref *pref); +static const char *domain_mode_to_text(const enum domain_mode mode); +static int sip_show_domains(int fd, int argc, char *argv[]); +static int _sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]); +static int sip_show_peer(int fd, int argc, char *argv[]); +static int sip_show_user(int fd, int argc, char *argv[]); +static int sip_show_registry(int fd, int argc, char *argv[]); +static int sip_show_settings(int fd, int argc, char *argv[]); +static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure; +static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype); +static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions); +static int sip_show_channels(int fd, int argc, char *argv[]); +static int sip_show_subscriptions(int fd, int argc, char *argv[]); +static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions); +static char *complete_sipch(const char *line, const char *word, int pos, int state); +static char *complete_sip_peer(const char *word, int state, int flags2); +static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state); +static char *complete_sip_debug_peer(const char *line, const char *word, int pos, int state); +static char *complete_sip_user(const char *word, int state, int flags2); +static char *complete_sip_show_user(const char *line, const char *word, int pos, int state); +static char *complete_sipnotify(const char *line, const char *word, int pos, int state); +static char *complete_sip_prune_realtime_peer(const char *line, const char *word, int pos, int state); +static char *complete_sip_prune_realtime_user(const char *line, const char *word, int pos, int state); +static int sip_show_channel(int fd, int argc, char *argv[]); +static int sip_show_history(int fd, int argc, char *argv[]); +static int sip_do_debug_ip(int fd, int argc, char *argv[]); +static int sip_do_debug_peer(int fd, int argc, char *argv[]); +static int sip_do_debug(int fd, int argc, char *argv[]); +static int sip_no_debug(int fd, int argc, char *argv[]); +static int sip_notify(int fd, int argc, char *argv[]); +static int sip_do_history(int fd, int argc, char *argv[]); +static int sip_no_history(int fd, int argc, char *argv[]); +static int func_header_read(struct ast_channel *chan, char *function, char *data, char *buf, size_t len); +static int func_check_sipdomain(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len); +static int function_sippeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len); +static int function_sipchaninfo_read(struct ast_channel *chan, 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); +static int sip_do_reload(enum channelreloadreason reason); +static int sip_reload(int fd, int argc, char *argv[]); +static int acf_channel_read(struct ast_channel *chan, char *funcname, char *preparse, char *buf, size_t buflen); + +/*--- Debugging + Functions for enabling debug per IP or fully, or enabling history logging for + a SIP dialog +*/ +static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to LOG_DEBUG at end of dialog, before destroying data */ +static inline int sip_debug_test_addr(const struct sockaddr_in *addr); +static inline int sip_debug_test_pvt(struct sip_pvt *p); +static void append_history_full(struct sip_pvt *p, const char *fmt, ...); +static void sip_dump_history(struct sip_pvt *dialog); + +/*--- Device object handling */ +static struct sip_peer *temp_peer(const char *name); +static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime); +static struct sip_user *build_user(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime); +static int update_call_counter(struct sip_pvt *fup, int event); +static void sip_destroy_peer(struct sip_peer *peer); +static void sip_destroy_user(struct sip_user *user); +static int sip_poke_peer(struct sip_peer *peer); +static int sip_poke_peer_s(const void *data); +static void set_peer_defaults(struct sip_peer *peer); +static struct sip_peer *temp_peer(const char *name); +static void register_peer_exten(struct sip_peer *peer, int onoff); +static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime, int devstate_only); +static struct sip_user *find_user(const char *name, int realtime); +static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req); +static int expire_register(const void *data); +static void reg_source_db(struct sip_peer *peer); +static void destroy_association(struct sip_peer *peer); +static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v); + +/* Realtime device support */ +static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey); +static struct sip_user *realtime_user(const char *username); +static void update_peer(struct sip_peer *p, int expiry); +static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin, int devstate_only); +static int sip_prune_realtime(int fd, int argc, char *argv[]); + +/*--- Internal UA client handling (outbound registrations) */ +static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us); +static void sip_registry_destroy(struct sip_registry *reg); +static int sip_register(char *value, int lineno); +static char *regstate2str(enum sipregistrystate regstate) attribute_const; +static int sip_reregister(const void *data); +static int __sip_do_register(struct sip_registry *r); +static int sip_reg_timeout(const void *data); +static void sip_send_all_registers(void); + +/*--- Parsing SIP requests and responses */ +static void append_date(struct sip_request *req); /* Append date to SIP packet */ +static int determine_firstline_parts(struct sip_request *req); +static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype); +static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize); +static void set_insecure_flags(struct ast_flags *flags, const char *value, int lineno); +static int find_sip_method(const char *msg); +static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported); +static int parse_request(struct sip_request *req); +static const char *get_header(const struct sip_request *req, const char *name); +static char *referstatus2str(enum referstatus rstatus) attribute_pure; +static int method_match(enum sipmethod id, const char *name); +static void parse_copy(struct sip_request *dst, const struct sip_request *src); +static char *get_in_brackets(char *tmp); +static const char *find_alias(const char *name, const char *_default); +static const char *__get_header(const struct sip_request *req, const char *name, int *start); +static int lws2sws(char *msgbuf, int len); +static void extract_uri(struct sip_pvt *p, struct sip_request *req); +static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req); +static int get_also_info(struct sip_pvt *p, struct sip_request *oreq); +static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req); +static int set_address_from_contact(struct sip_pvt *pvt); +static void check_via(struct sip_pvt *p, const struct sip_request *req); +static char *get_calleridname(const char *input, char *output, size_t outputsize); +static int get_rpid_num(const char *input, char *output, int maxlen); +static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq); +static int get_destination(struct sip_pvt *p, struct sip_request *oreq); +static int get_msg_text(char *buf, int len, struct sip_request *req); +static void free_old_route(struct sip_route *route); +static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout); + +/*--- Constructing requests and responses */ +static void initialize_initreq(struct sip_pvt *p, struct sip_request *req); +static int init_req(struct sip_request *req, int sipmethod, const char *recip); +static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch); +static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod); +static int init_resp(struct sip_request *resp, const char *msg); +static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req); +static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p); +static void build_via(struct sip_pvt *p); +static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer); +static int create_addr(struct sip_pvt *dialog, const char *opeer); +static char *generate_random_string(char *buf, size_t size); +static void build_callid_pvt(struct sip_pvt *pvt); +static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain); +static void make_our_tag(char *tagbuf, size_t len); +static int add_header(struct sip_request *req, const char *var, const char *value); +static int add_header_contentLength(struct sip_request *req, int len); +static int add_line(struct sip_request *req, const char *line); +static int add_text(struct sip_request *req, const char *text); +static int add_digit(struct sip_request *req, char digit, unsigned int duration); +static int add_vidupdate(struct sip_request *req); +static void add_route(struct sip_request *req, struct sip_route *route); +static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field); +static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field); +static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field); +static void set_destination(struct sip_pvt *p, char *uri); +static void append_date(struct sip_request *req); +static void build_contact(struct sip_pvt *p); +static void build_rpid(struct sip_pvt *p); + +/*------Request handling functions */ +static int handle_request(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock); +static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e, int *nounlock); +static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, int *nounlock); +static int handle_request_bye(struct sip_pvt *p, struct sip_request *req); +static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e); +static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req); +static int handle_request_message(struct sip_pvt *p, struct sip_request *req); +static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e); +static void handle_request_info(struct sip_pvt *p, struct sip_request *req); +static int handle_request_options(struct sip_pvt *p, struct sip_request *req); +static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin); +static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e); +static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno); + +/*------Response handling functions */ +static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno); +static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno); +static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno); +static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno); + +/*----- RTP interface functions */ +static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active); +static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp); +static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp); +static int sip_get_codec(struct ast_channel *chan); +static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect); + +/*------ T38 Support --------- */ +static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */ +static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans); +static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p); +static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan); +static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl); + +/*! \brief Definition of this channel for PBX channel registration */ +static const struct ast_channel_tech sip_tech = { + .type = "SIP", + .description = "Session Initiation Protocol (SIP)", + .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1), + .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER, + .requester = sip_request_call, + .devicestate = sip_devicestate, + .call = sip_call, + .hangup = sip_hangup, + .answer = sip_answer, + .read = sip_read, + .write = sip_write, + .write_video = sip_write, + .indicate = sip_indicate, + .transfer = sip_transfer, + .fixup = sip_fixup, + .send_digit_begin = sip_senddigit_begin, + .send_digit_end = sip_senddigit_end, + .bridge = ast_rtp_bridge, + .send_text = sip_sendtext, + .func_channel_read = acf_channel_read, +}; + +/*! \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, + .devicestate = sip_devicestate, + .call = sip_call, + .hangup = sip_hangup, + .answer = sip_answer, + .read = sip_read, + .write = sip_write, + .write_video = sip_write, + .indicate = sip_indicate, + .transfer = sip_transfer, + .fixup = sip_fixup, + .send_digit_end = sip_senddigit_end, + .bridge = ast_rtp_bridge, + .send_text = sip_sendtext, + .func_channel_read = acf_channel_read, +}; + +/**--- some list management macros. **/ + +#define UNLINK(element, head, prev) do { \ + if (prev) \ + (prev)->next = (element)->next; \ + else \ + (head) = (element)->next; \ + } while (0) + +/*! \brief Interface structure with callbacks used to connect to RTP module */ +static struct ast_rtp_protocol sip_rtp = { + type: "SIP", + get_rtp_info: sip_get_rtp_peer, + get_vrtp_info: sip_get_vrtp_peer, + set_rtp_peer: sip_set_rtp_peer, + get_codec: sip_get_codec, +}; + +/*! \brief Interface structure with callbacks used to connect to UDPTL module*/ +static struct ast_udptl_protocol sip_udptl = { + type: "SIP", + get_udptl_info: sip_get_udptl_peer, + set_udptl_peer: sip_set_udptl_peer, +}; + +/*! \brief Convert transfer status to string */ +static char *referstatus2str(enum referstatus rstatus) +{ + int i = (sizeof(referstatusstrings) / sizeof(referstatusstrings[0])); + int x; + + for (x = 0; x < i; x++) { + if (referstatusstrings[x].status == rstatus) + return (char *) referstatusstrings[x].text; + } + return ""; +} + +/*! \brief Initialize the initital request packet in the pvt structure. + This packet is used for creating replies and future requests in + a dialog */ +static void initialize_initreq(struct sip_pvt *p, struct sip_request *req) +{ + if (p->initreq.headers && option_debug) { + ast_log(LOG_DEBUG, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid); + } + /* Use this as the basis */ + copy_request(&p->initreq, req); + parse_request(&p->initreq); + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); +} + +static void sip_alreadygone(struct sip_pvt *dialog) +{ + if (option_debug > 2) + ast_log(LOG_DEBUG, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid); + ast_set_flag(&dialog->flags[0], SIP_ALREADYGONE); +} + + +/*! \brief returns true if 'name' (with optional trailing whitespace) + * matches the sip method 'id'. + * Strictly speaking, SIP methods are case SENSITIVE, but we do + * a case-insensitive comparison to be more tolerant. + * following Jon Postel's rule: Be gentle in what you accept, strict with what you send + */ +static int method_match(enum sipmethod id, const char *name) +{ + int len = strlen(sip_methods[id].text); + int l_name = name ? strlen(name) : 0; + /* true if the string is long enough, and ends with whitespace, and matches */ + return (l_name >= len && name[len] < 33 && + !strncasecmp(sip_methods[id].text, name, len)); +} + +/*! \brief find_sip_method: Find SIP method from header */ +static int find_sip_method(const char *msg) +{ + int i, res = 0; + + if (ast_strlen_zero(msg)) + return 0; + for (i = 1; i < (sizeof(sip_methods) / sizeof(sip_methods[0])) && !res; i++) { + if (method_match(i, msg)) + res = sip_methods[i].id; + } + return res; +} + +/*! \brief Parse supported header in incoming packet */ +static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported) +{ + char *next, *sep; + char *temp; + unsigned int profile = 0; + int i, found; + + if (ast_strlen_zero(supported) ) + return 0; + temp = ast_strdupa(supported); + + if (option_debug > 2 && sipdebug) + ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported); + + for (next = temp; next; next = sep) { + found = FALSE; + if ( (sep = strchr(next, ',')) != NULL) + *sep++ = '\0'; + next = ast_skip_blanks(next); + if (option_debug > 2 && sipdebug) + ast_log(LOG_DEBUG, "Found SIP option: -%s-\n", next); + for (i=0; i < (sizeof(sip_options) / sizeof(sip_options[0])); i++) { + if (!strcasecmp(next, sip_options[i].text)) { + profile |= sip_options[i].id; + found = TRUE; + if (option_debug > 2 && sipdebug) + ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next); + break; + } + } + if (!found && option_debug > 2 && sipdebug) { + if (!strncasecmp(next, "x-", 2)) + ast_log(LOG_DEBUG, "Found private SIP option, not supported: %s\n", next); + else + ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next); + } + } + + if (pvt) + pvt->sipoptions = profile; + return profile; +} + +/*! \brief See if we pass debug IP filter */ +static inline int sip_debug_test_addr(const struct sockaddr_in *addr) +{ + if (!sipdebug) + return 0; + if (debugaddr.sin_addr.s_addr) { + if (((ntohs(debugaddr.sin_port) != 0) + && (debugaddr.sin_port != addr->sin_port)) + || (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr)) + return 0; + } + return 1; +} + +/*! \brief The real destination address for a write */ +static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p) +{ + return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? &p->recv : &p->sa; +} + +/*! \brief Display SIP nat mode */ +static const char *sip_nat_mode(const struct sip_pvt *p) +{ + return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? "NAT" : "no NAT"; +} + +/*! \brief Test PVT for debugging output */ +static inline int sip_debug_test_pvt(struct sip_pvt *p) +{ + if (!sipdebug) + return 0; + return sip_debug_test_addr(sip_real_dst(p)); +} + +/*! \brief Transmit SIP message */ +static int __sip_xmit(struct sip_pvt *p, char *data, int len) +{ + int res; + const struct sockaddr_in *dst = sip_real_dst(p); + res = sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in)); + + if (res == -1) { + switch (errno) { + case EBADF: /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */ + case EHOSTUNREACH: /* Host can't be reached */ + case ENETDOWN: /* Inteface down */ + case ENETUNREACH: /* Network failure */ + case ECONNREFUSED: /* ICMP port unreachable */ + res = XMIT_ERROR; /* Don't bother with trying to transmit again */ + } + } + if (res != len) + ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno)); + return res; +} + + +/*! \brief Build a Via header for a request */ +static void build_via(struct sip_pvt *p) +{ + /* Work around buggy UNIDEN UIP200 firmware */ + const char *rport = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : ""; + + /* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */ + ast_string_field_build(p, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s", + ast_inet_ntoa(p->ourip), ourport, (int) p->branch, rport); +} + +/*! \brief NAT fix - decide which IP address to use for ASterisk server? + * + * Using the localaddr structure built up with localnet statements in sip.conf + * apply it to their address to see if we need to substitute our + * externip or can get away with our internal bindaddr + */ +static enum sip_result ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us) +{ + struct sockaddr_in theirs, ours; + + /* Get our local information */ + ast_ouraddrfor(them, us); + theirs.sin_addr = *them; + ours.sin_addr = *us; + + if (localaddr && externip.sin_addr.s_addr && + (ast_apply_ha(localaddr, &theirs)) && + (!global_matchexterniplocally || !ast_apply_ha(localaddr, &ours))) { + if (externexpire && time(NULL) >= externexpire) { + struct ast_hostent ahp; + struct hostent *hp; + + externexpire = time(NULL) + externrefresh; + if ((hp = ast_gethostbyname(externhost, &ahp))) { + memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr)); + } else + ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost); + } + *us = externip.sin_addr; + if (option_debug) { + ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n", + ast_inet_ntoa(*(struct in_addr *)&them->s_addr)); + } + } else if (bindaddr.sin_addr.s_addr) + *us = bindaddr.sin_addr; + return AST_SUCCESS; +} + +/*! \brief Append to SIP dialog history + \return Always returns 0 */ +#define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args) + +static void append_history_full(struct sip_pvt *p, const char *fmt, ...) + __attribute__((format(printf, 2, 3))); + +/*! \brief Append to SIP dialog history with arg list */ +static void __attribute__((format(printf, 2, 0))) append_history_va(struct sip_pvt *p, const char *fmt, va_list ap) +{ + char buf[80], *c = buf; /* max history length */ + struct sip_history *hist; + int l; + + vsnprintf(buf, sizeof(buf), fmt, ap); + strsep(&c, "\r\n"); /* Trim up everything after \r or \n */ + l = strlen(buf) + 1; + if (!(hist = ast_calloc(1, sizeof(*hist) + l))) + return; + if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) { + free(hist); + return; + } + memcpy(hist->event, buf, l); + if (p->history_entries == MAX_HISTORY_ENTRIES) { + struct sip_history *oldest; + oldest = AST_LIST_REMOVE_HEAD(p->history, list); + p->history_entries--; + free(oldest); + } + AST_LIST_INSERT_TAIL(p->history, hist, list); + p->history_entries++; +} + +/*! \brief Append to SIP dialog history with arg list */ +static void append_history_full(struct sip_pvt *p, const char *fmt, ...) +{ + va_list ap; + + if (!p) + return; + + if (ast_test_flag(&p->flags[0], SIP_NO_HISTORY) + && !recordhistory && !dumphistory) { + return; + } + + va_start(ap, fmt); + append_history_va(p, fmt, ap); + va_end(ap); + + return; +} + +/*! \brief Retransmit SIP message if no answer (Called from scheduler) */ +static int retrans_pkt(const void *data) +{ + struct sip_pkt *pkt = (struct sip_pkt *)data, *prev, *cur = NULL; + int reschedule = DEFAULT_RETRANS; + int xmitres = 0; + + /* Lock channel PVT */ + ast_mutex_lock(&pkt->owner->lock); + + if (pkt->retrans < MAX_RETRANS) { + pkt->retrans++; + if (!pkt->timer_t1) { /* Re-schedule using timer_a and timer_t1 */ + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_methods[pkt->method].text, pkt->method); + } else { + int siptimer_a; + + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method); + if (!pkt->timer_a) + pkt->timer_a = 2 ; + else + pkt->timer_a = 2 * pkt->timer_a; + + /* For non-invites, a maximum of 4 secs */ + siptimer_a = pkt->timer_t1 * pkt->timer_a; /* Double each time */ + if (pkt->method != SIP_INVITE && siptimer_a > 4000) + siptimer_a = 4000; + + /* Reschedule re-transmit */ + reschedule = siptimer_a; + if (option_debug > 3) + ast_log(LOG_DEBUG, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid); + } + + if (sip_debug_test_pvt(pkt->owner)) { + const struct sockaddr_in *dst = sip_real_dst(pkt->owner); + ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n", + pkt->retrans, sip_nat_mode(pkt->owner), + ast_inet_ntoa(dst->sin_addr), + ntohs(dst->sin_port), pkt->data); + } + + append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data); + xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); + ast_mutex_unlock(&pkt->owner->lock); + if (xmitres == XMIT_ERROR) + ast_log(LOG_WARNING, "Network error on retransmit in dialog %s\n", pkt->owner->callid); + else + return reschedule; + } + /* Too many retries */ + if (pkt->owner && pkt->method != SIP_OPTIONS && xmitres == 0) { + if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */ + ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s) -- See doc/sip-retransmit.txt.\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request"); + } else if ((pkt->method == SIP_OPTIONS) && sipdebug) { + ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) -- See doc/sip-retransmit.txt.\n", pkt->owner->callid); + } + if (xmitres == XMIT_ERROR) { + ast_log(LOG_WARNING, "Transmit error :: Cancelling transmission of transaction in call id %s \n", pkt->owner->callid); + append_history(pkt->owner, "XmitErr", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)"); + } else + append_history(pkt->owner, "MaxRetries", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)"); + + pkt->retransid = -1; + + if (ast_test_flag(pkt, FLAG_FATAL)) { + while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) { + DEADLOCK_AVOIDANCE(&pkt->owner->lock); /* SIP_PVT, not channel */ + } + + if (pkt->owner->owner && !pkt->owner->owner->hangupcause) + pkt->owner->owner->hangupcause = AST_CAUSE_NO_USER_RESPONSE; + + if (pkt->owner->owner) { + sip_alreadygone(pkt->owner); + ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet (see doc/sip-retransmit.txt).\n", pkt->owner->callid); + ast_queue_hangup(pkt->owner->owner); + ast_channel_unlock(pkt->owner->owner); + } else { + /* If no channel owner, destroy now */ + + /* Let the peerpoke system expire packets when the timer expires for poke_noanswer */ + if (pkt->method != SIP_OPTIONS) { + ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY); + sip_alreadygone(pkt->owner); + if (option_debug) + append_history(pkt->owner, "DialogKill", "Killing this failed dialog immediately"); + } + } + } + + if (pkt->method == SIP_BYE) { + /* We're not getting answers on SIP BYE's. Tear down the call anyway. */ + if (pkt->owner->owner) + ast_channel_unlock(pkt->owner->owner); + append_history(pkt->owner, "ByeFailure", "Remote peer doesn't respond to bye. Destroying call anyway."); + ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY); + } + + /* In any case, go ahead and remove the packet */ + for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) { + if (cur == pkt) + break; + } + if (cur) { + if (prev) + prev->next = cur->next; + else + pkt->owner->packets = cur->next; + ast_mutex_unlock(&pkt->owner->lock); + free(cur); + pkt = NULL; + } else + ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n"); + if (pkt) + ast_mutex_unlock(&pkt->owner->lock); + return 0; +} + +/*! \brief Transmit packet with retransmits + \return 0 on success, -1 on failure to allocate packet +*/ +static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod) +{ + struct sip_pkt *pkt; + int siptimer_a = DEFAULT_RETRANS; + int xmitres = 0; + + if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1))) + return AST_FAILURE; + memcpy(pkt->data, data, len); + pkt->method = sipmethod; + pkt->packetlen = len; + pkt->next = p->packets; + pkt->owner = p; + pkt->seqno = seqno; + if (resp) + ast_set_flag(pkt, FLAG_RESPONSE); + pkt->data[len] = '\0'; + pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */ + pkt->retransid = -1; + if (fatal) + ast_set_flag(pkt, FLAG_FATAL); + if (pkt->timer_t1) + siptimer_a = pkt->timer_t1 * 2; + + if (option_debug > 3 && sipdebug) + ast_log(LOG_DEBUG, "*** SIP TIMER: Initializing retransmit timer on packet: Id #%d\n", pkt->retransid); + pkt->retransid = -1; + pkt->next = p->packets; + p->packets = pkt; + if (sipmethod == SIP_INVITE) { + /* Note this is a pending invite */ + p->pendinginvite = seqno; + } + + xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */ + + if (xmitres == XMIT_ERROR) { /* Serious network trouble, no need to try again */ + append_history(pkt->owner, "XmitErr", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)"); + return AST_FAILURE; + } else { + /* Schedule retransmission */ + pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1); + return AST_SUCCESS; + } +} + +/*! \brief Kill a SIP dialog (called by scheduler) */ +static int __sip_autodestruct(const void *data) +{ + struct sip_pvt *p = (struct sip_pvt *)data; + + /* If this is a subscription, tell the phone that we got a timeout */ + if (p->subscribed) { + transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, TRUE); /* Send last notification */ + p->subscribed = NONE; + append_history(p, "Subscribestatus", "timeout"); + if (option_debug > 2) + ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP subsription %s\n", p->callid ? p->callid : "<unknown>"); + return 10000; /* Reschedule this destruction so that we know that it's gone */ + } + + /* If there are packets still waiting for delivery, delay the destruction */ + /* via bug 12101, the two usages of SIP_NEEDDESTROY in the following block + * of code make a sort of "safety relief valve", that allows sip channels + * that were created via INVITE, then thru some sequence were CANCELED, + * to die, rather than infinitely be rescheduled */ + if (p->packets && !ast_test_flag(&p->flags[0], SIP_NEEDDESTROY)) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP call %s\n", p->callid ? p->callid : "<unknown>"); + append_history(p, "ReliableXmit", "timeout"); + if (p->method == SIP_CANCEL || p->method == SIP_BYE) { + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + return 10000; + } + + /* If we're destroying a subscription, dereference peer object too */ + if (p->subscribed == MWI_NOTIFICATION && p->relatedpeer) + ASTOBJ_UNREF(p->relatedpeer,sip_destroy_peer); + + /* Reset schedule ID */ + p->autokillid = -1; + + if (option_debug) + ast_log(LOG_DEBUG, "Auto destroying SIP dialog '%s'\n", p->callid); + append_history(p, "AutoDestroy", "%s", p->callid); + if (p->owner) { + ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text); + ast_queue_hangup(p->owner); + } else if (p->refer && !ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Finally hanging up channel after transfer: %s\n", p->callid); + transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } else + sip_destroy(p); + return 0; +} + +/*! \brief Schedule destruction of SIP dialog */ +static void sip_scheddestroy(struct sip_pvt *p, int ms) +{ + if (ms < 0) { + if (p->timer_t1 == 0) + p->timer_t1 = 500; /* Set timer T1 if not set (RFC 3261) */ + ms = p->timer_t1 * 64; + } + if (sip_debug_test_pvt(p)) + ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text); + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) + append_history(p, "SchedDestroy", "%d ms", ms); + + AST_SCHED_DEL(sched, p->autokillid); + p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, p); +} + +/*! \brief Cancel destruction of SIP dialog */ +static int sip_cancel_destroy(struct sip_pvt *p) +{ + int res = 0; + if (p->autokillid > -1) { + if (!(res = ast_sched_del(sched, p->autokillid))) { + append_history(p, "CancelDestroy", ""); + p->autokillid = -1; + } + } + return res; +} + +/*! \brief Acknowledges receipt of a packet and stops retransmission + * called with p locked*/ +static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod) +{ + struct sip_pkt *cur, *prev = NULL; + + /* Just in case... */ + char *msg; + int res = FALSE; + + msg = sip_methods[sipmethod].text; + + for (cur = p->packets; cur; prev = cur, cur = cur->next) { + if ((cur->seqno == seqno) && ((ast_test_flag(cur, FLAG_RESPONSE)) == resp) && + ((ast_test_flag(cur, FLAG_RESPONSE)) || + (!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) { + if (!resp && (seqno == p->pendinginvite)) { + if (option_debug) + ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite); + p->pendinginvite = 0; + } + /* this is our baby */ + res = TRUE; + UNLINK(cur, p->packets, prev); + if (cur->retransid > -1) { + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid); + } + /* This odd section is designed to thwart a + * race condition in the packet scheduler. There are + * two conditions under which deleting the packet from the + * scheduler can fail. + * + * 1. The packet has been removed from the scheduler because retransmission + * is being attempted. The problem is that if the packet is currently attempting + * retransmission and we are at this point in the code, then that MUST mean + * that retrans_pkt is waiting on p's lock. Therefore we will relinquish the + * lock temporarily to allow retransmission. + * + * 2. The packet has reached its maximum number of retransmissions and has + * been permanently removed from the packet scheduler. If this is the case, then + * the packet's retransid will be set to -1. The atomicity of the setting and checking + * of the retransid to -1 is ensured since in both cases p's lock is held. + */ + while (cur->retransid > -1 && ast_sched_del(sched, cur->retransid)) { + DEADLOCK_AVOIDANCE(&p->lock); + } + free(cur); + break; + } + } + if (option_debug) + ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n", p->callid, resp ? "Response" : "Request", seqno, res == FALSE ? "Not Found" : "Found"); +} + +/*! \brief Pretend to ack all packets + * called with p locked */ +static void __sip_pretend_ack(struct sip_pvt *p) +{ + struct sip_pkt *cur = NULL; + + while (p->packets) { + int method; + if (cur == p->packets) { + ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text); + return; + } + cur = p->packets; + method = (cur->method) ? cur->method : find_sip_method(cur->data); + __sip_ack(p, cur->seqno, ast_test_flag(cur, FLAG_RESPONSE), method); + } +} + +/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */ +static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod) +{ + struct sip_pkt *cur; + int res = -1; + + for (cur = p->packets; cur; cur = cur->next) { + if (cur->seqno == seqno && ast_test_flag(cur, FLAG_RESPONSE) == resp && + (ast_test_flag(cur, FLAG_RESPONSE) || method_match(sipmethod, cur->data))) { + /* this is our baby */ + if (cur->retransid > -1) { + if (option_debug > 3 && sipdebug) + ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text); + } + AST_SCHED_DEL(sched, cur->retransid); + res = 0; + break; + } + } + if (option_debug) + ast_log(LOG_DEBUG, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res == -1 ? "Not Found" : "Found"); + return res; +} + + +/*! \brief Copy SIP request, parse it */ +static void parse_copy(struct sip_request *dst, const struct sip_request *src) +{ + memset(dst, 0, sizeof(*dst)); + memcpy(dst->data, src->data, sizeof(dst->data)); + dst->len = src->len; + parse_request(dst); +} + +/*! \brief add a blank line if no body */ +static void add_blank(struct sip_request *req) +{ + if (!req->lines) { + /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */ + snprintf(req->data + req->len, sizeof(req->data) - req->len, "\r\n"); + req->len += strlen(req->data + req->len); + } +} + +/*! \brief Transmit response on SIP request*/ +static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno) +{ + int res; + + add_blank(req); + if (sip_debug_test_pvt(p)) { + const struct sockaddr_in *dst = sip_real_dst(p); + + ast_verbose("\n<--- %sTransmitting (%s) to %s:%d --->\n%s\n<------------>\n", + reliable ? "Reliably " : "", sip_nat_mode(p), + ast_inet_ntoa(dst->sin_addr), + ntohs(dst->sin_port), req->data); + } + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) { + struct sip_request tmp; + parse_copy(&tmp, req); + append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), + (tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? tmp.rlPart2 : sip_methods[tmp.method].text); + } + res = (reliable) ? + __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) : + __sip_xmit(p, req->data, req->len); + if (res > 0) + return 0; + return res; +} + +/*! \brief Send SIP Request to the other part of the dialogue */ +static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno) +{ + int res; + + add_blank(req); + if (sip_debug_test_pvt(p)) { + if (ast_test_flag(&p->flags[0], SIP_NAT_ROUTE)) + ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port), req->data); + else + ast_verbose("%sTransmitting (no NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->sa.sin_addr), ntohs(p->sa.sin_port), req->data); + } + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) { + struct sip_request tmp; + parse_copy(&tmp, req); + append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text); + } + res = (reliable) ? + __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) : + __sip_xmit(p, req->data, req->len); + return res; +} + +/*! \brief Locate closing quote in a string, skipping escaped quotes. + * optionally with a limit on the search. + * start must be past the first quote. + */ +static const char *find_closing_quote(const char *start, const char *lim) +{ + char last_char = '\0'; + const char *s; + for (s = start; *s && s != lim; last_char = *s++) { + if (*s == '"' && last_char != '\\') + break; + } + return s; +} + +/*! \brief Pick out text in brackets from character string + \return pointer to terminated stripped string + \param tmp input string that will be modified + Examples: + + "foo" <bar> valid input, returns bar + foo returns the whole string + < "foo ... > returns the string between brackets + < "foo... bogus (missing closing bracket), returns the whole string + XXX maybe should still skip the opening bracket + */ +static char *get_in_brackets(char *tmp) +{ + const char *parse = tmp; + char *first_bracket; + + /* + * Skip any quoted text until we find the part in brackets. + * On any error give up and return the full string. + */ + while ( (first_bracket = strchr(parse, '<')) ) { + char *first_quote = strchr(parse, '"'); + + if (!first_quote || first_quote > first_bracket) + break; /* no need to look at quoted part */ + /* the bracket is within quotes, so ignore it */ + parse = find_closing_quote(first_quote + 1, NULL); + if (!*parse) { /* not found, return full string ? */ + /* XXX or be robust and return in-bracket part ? */ + ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp); + break; + } + parse++; + } + if (first_bracket) { + char *second_bracket = strchr(first_bracket + 1, '>'); + if (second_bracket) { + *second_bracket = '\0'; + tmp = first_bracket + 1; + } else { + ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp); + } + } + return tmp; +} + +/*! \brief Send SIP MESSAGE text within a call + Called from PBX core sendtext() application */ +static int sip_sendtext(struct ast_channel *ast, const char *text) +{ + struct sip_pvt *p = ast->tech_pvt; + int debug = sip_debug_test_pvt(p); + + if (debug) + ast_verbose("Sending text %s on %s\n", text, ast->name); + if (!p) + return -1; + if (ast_strlen_zero(text)) + return 0; + if (debug) + ast_verbose("Really sending text %s on %s\n", text, ast->name); + transmit_message_with_text(p, text); + return 0; +} + +/*! \brief Update peer object in realtime storage + If the Asterisk system name is set in asterisk.conf, we will use + that name and store that in the "regserver" field in the sippeers + table to facilitate multi-server setups. +*/ +static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey) +{ + char port[10]; + char ipaddr[INET_ADDRSTRLEN]; + char regseconds[20]; + + char *sysname = ast_config_AST_SYSTEM_NAME; + char *syslabel = NULL; + + time_t nowtime = time(NULL) + expirey; + const char *fc = fullcontact ? "fullcontact" : NULL; + + snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime); /* Expiration time */ + ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr)); + snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port)); + + if (ast_strlen_zero(sysname)) /* No system name, disable this */ + sysname = NULL; + else if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTSAVE_SYSNAME)) + syslabel = "regserver"; + + if (fc) + ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr, + "port", port, "regseconds", regseconds, + "username", username, fc, fullcontact, syslabel, sysname, NULL); /* note fc and syslabel _can_ be NULL */ + else + ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr, + "port", port, "regseconds", regseconds, + "username", username, syslabel, sysname, NULL); /* note syslabel _can_ be NULL */ +} + +/*! \brief Automatically add peer extension to dial plan */ +static void register_peer_exten(struct sip_peer *peer, int onoff) +{ + char multi[256]; + char *stringp, *ext, *context; + + /* XXX note that global_regcontext is both a global 'enable' flag and + * the name of the global regexten context, if not specified + * individually. + */ + if (ast_strlen_zero(global_regcontext)) + return; + + ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi)); + stringp = multi; + while ((ext = strsep(&stringp, "&"))) { + if ((context = strchr(ext, '@'))) { + *context++ = '\0'; /* split ext@context */ + if (!ast_context_find(context)) { + ast_log(LOG_WARNING, "Context %s must exist in regcontext= in sip.conf!\n", context); + continue; + } + } else { + context = global_regcontext; + } + if (onoff) { + if (!ast_exists_extension(NULL, context, ext, 1, NULL)) { + ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop", + ast_strdup(peer->name), ast_free, "SIP"); + } + } else { + ast_context_remove_extension(context, ext, 1, NULL); + } + } +} + +/*! \brief Destroy peer object from memory */ +static void sip_destroy_peer(struct sip_peer *peer) +{ + if (option_debug > 2) + ast_log(LOG_DEBUG, "Destroying SIP peer %s\n", peer->name); + + /* Delete it, it needs to disappear */ + if (peer->call) + sip_destroy(peer->call); + + if (peer->mwipvt) /* We have an active subscription, delete it */ + sip_destroy(peer->mwipvt); + + if (peer->chanvars) { + ast_variables_destroy(peer->chanvars); + peer->chanvars = NULL; + } + + register_peer_exten(peer, FALSE); + ast_free_ha(peer->ha); + if (ast_test_flag(&peer->flags[1], SIP_PAGE2_SELFDESTRUCT)) + apeerobjs--; + else if (ast_test_flag(&peer->flags[0], SIP_REALTIME)) + rpeerobjs--; + else + speerobjs--; + clear_realm_authentication(peer->auth); + peer->auth = NULL; + free(peer); +} + +/*! \brief Update peer data in database (if used) */ +static void update_peer(struct sip_peer *p, int expiry) +{ + int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS); + if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTUPDATE) && + (ast_test_flag(&p->flags[0], SIP_REALTIME) || rtcachefriends)) { + realtime_update_peer(p->name, &p->addr, p->username, rtcachefriends ? p->fullcontact : NULL, expiry); + } +} + + +/*! \brief realtime_peer: Get peer from realtime storage + * Checks the "sippeers" realtime family from extconfig.conf + * \todo Consider adding check of port address when matching here to follow the same + * algorithm as for static peers. Will we break anything by adding that? +*/ +static struct sip_peer *realtime_peer(const char *newpeername, struct sockaddr_in *sin, int devstate_only) +{ + struct sip_peer *peer=NULL; + struct ast_variable *var = NULL; + struct ast_config *peerlist = NULL; + struct ast_variable *tmp; + struct ast_flags flags = {0}; + const char *iabuf = NULL; + char portstring[6]; /*up to five digits plus null terminator*/ + const char *insecure; + char *cat = NULL; + unsigned short portnum; + + /* First check on peer name */ + if (newpeername) { + var = ast_load_realtime("sippeers", "name", newpeername, "host", "dynamic", NULL); + if (!var && sin) + var = ast_load_realtime("sippeers", "name", newpeername, "host", ast_inet_ntoa(sin->sin_addr), NULL); + if (!var) { + var = ast_load_realtime("sippeers", "name", newpeername, NULL); + /*!\note + * If this one loaded something, then we need to ensure that the host + * field matched. The only reason why we can't have this as a criteria + * is because we only have the IP address and the host field might be + * set as a name (and the reverse PTR might not match). + */ + if (var && sin) { + for (tmp = var; tmp; tmp = tmp->next) { + if (!strcasecmp(tmp->name, "host")) { + struct hostent *hp; + struct ast_hostent ahp; + if (!(hp = ast_gethostbyname(tmp->value, &ahp)) || (memcmp(&hp->h_addr, &sin->sin_addr, sizeof(hp->h_addr)))) { + /* No match */ + ast_variables_destroy(var); + var = NULL; + } + break; + } + } + } + } + } + + if (!var && sin) { /* Then check on IP address */ + iabuf = ast_inet_ntoa(sin->sin_addr); + portnum = ntohs(sin->sin_port); + sprintf(portstring, "%d", portnum); + var = ast_load_realtime("sippeers", "host", iabuf, "port", portstring, NULL); /* First check for fixed IP hosts */ + if (!var) + var = ast_load_realtime("sippeers", "ipaddr", iabuf, "port", portstring, NULL); /* Then check for registered hosts */ + if (!var) { + peerlist = ast_load_realtime_multientry("sippeers", "host", iabuf, NULL); /*No exact match, see if port is insecure, try host match first*/ + if(peerlist){ + while((cat = ast_category_browse(peerlist, cat))) + { + insecure = ast_variable_retrieve(peerlist, cat, "insecure"); + set_insecure_flags(&flags, insecure, -1); + if(ast_test_flag(&flags, SIP_INSECURE_PORT)) { + var = ast_category_root(peerlist, cat); + break; + } + } + } + if(!var) { + ast_config_destroy(peerlist); + peerlist = NULL; /*for safety's sake*/ + cat = NULL; + peerlist = ast_load_realtime_multientry("sippeers", "ipaddr", iabuf, NULL); /*No exact match, see if port is insecure, now try ip address match*/ + if(peerlist) { + while((cat = ast_category_browse(peerlist, cat))) + { + insecure = ast_variable_retrieve(peerlist, cat, "insecure"); + set_insecure_flags(&flags, insecure, -1); + if(ast_test_flag(&flags, SIP_INSECURE_PORT)) { + var = ast_category_root(peerlist, cat); + break; + } + } + } + } + } + } + + if (!var) { + if(peerlist) + ast_config_destroy(peerlist); + return NULL; + } + + for (tmp = var; tmp; tmp = tmp->next) { + /* If this is type=user, then skip this object. */ + if (!strcasecmp(tmp->name, "type") && + !strcasecmp(tmp->value, "user")) { + ast_variables_destroy(var); + return NULL; + } else if (!newpeername && !strcasecmp(tmp->name, "name")) { + newpeername = tmp->value; + } + } + + if (!newpeername) { /* Did not find peer in realtime */ + ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", iabuf); + if(peerlist) + ast_config_destroy(peerlist); + else + ast_variables_destroy(var); + return NULL; + } + + /* Peer found in realtime, now build it in memory */ + peer = build_peer(newpeername, var, NULL, 1); + if (!peer) { + if(peerlist) + ast_config_destroy(peerlist); + else + ast_variables_destroy(var); + return NULL; + } + + if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && !devstate_only) { + /* Cache peer */ + ast_copy_flags(&peer->flags[1],&global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS); + if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) { + if (!AST_SCHED_DEL(sched, peer->expire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + peer->expire = ast_sched_add(sched, (global_rtautoclear) * 1000, expire_register, ASTOBJ_REF(peer)); + if (peer->expire == -1) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + } + ASTOBJ_CONTAINER_LINK(&peerl,peer); + } + ast_set_flag(&peer->flags[0], SIP_REALTIME); + if(peerlist) + ast_config_destroy(peerlist); + else + ast_variables_destroy(var); + return peer; +} + +/*! \brief Support routine for find_peer */ +static int sip_addrcmp(char *name, struct sockaddr_in *sin) +{ + /* We know name is the first field, so we can cast */ + struct sip_peer *p = (struct sip_peer *) name; + return !(!inaddrcmp(&p->addr, sin) || + (ast_test_flag(&p->flags[0], SIP_INSECURE_PORT) && + (p->addr.sin_addr.s_addr == sin->sin_addr.s_addr))); +} + +/*! \brief Locate peer by name or ip address + * This is used on incoming SIP message to find matching peer on ip + or outgoing message to find matching peer on name */ +static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime, int devstate_only) +{ + struct sip_peer *p = NULL; + + if (peer) + p = ASTOBJ_CONTAINER_FIND(&peerl, peer); + else + p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp); + + if (!p && (realtime || devstate_only)) + p = realtime_peer(peer, sin, devstate_only); + + return p; +} + +/*! \brief Remove user object from in-memory storage */ +static void sip_destroy_user(struct sip_user *user) +{ + if (option_debug > 2) + ast_log(LOG_DEBUG, "Destroying user object from memory: %s\n", user->name); + ast_free_ha(user->ha); + if (user->chanvars) { + ast_variables_destroy(user->chanvars); + user->chanvars = NULL; + } + if (ast_test_flag(&user->flags[0], SIP_REALTIME)) + ruserobjs--; + else + suserobjs--; + free(user); +} + +/*! \brief Load user from realtime storage + * Loads user from "sipusers" category in realtime (extconfig.conf) + * Users are matched on From: user name (the domain in skipped) */ +static struct sip_user *realtime_user(const char *username) +{ + struct ast_variable *var; + struct ast_variable *tmp; + struct sip_user *user = NULL; + + var = ast_load_realtime("sipusers", "name", username, NULL); + + if (!var) + return NULL; + + for (tmp = var; tmp; tmp = tmp->next) { + if (!strcasecmp(tmp->name, "type") && + !strcasecmp(tmp->value, "peer")) { + ast_variables_destroy(var); + return NULL; + } + } + + user = build_user(username, var, NULL, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)); + + if (!user) { /* No user found */ + ast_variables_destroy(var); + return NULL; + } + + if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) { + ast_set_flag(&user->flags[1], SIP_PAGE2_RTCACHEFRIENDS); + suserobjs++; + ASTOBJ_CONTAINER_LINK(&userl,user); + } else { + /* Move counter from s to r... */ + suserobjs--; + ruserobjs++; + } + ast_set_flag(&user->flags[0], SIP_REALTIME); + ast_variables_destroy(var); + return user; +} + +/*! \brief Locate user by name + * Locates user by name (From: sip uri user name part) first + * from in-memory list (static configuration) then from + * realtime storage (defined in extconfig.conf) */ +static struct sip_user *find_user(const char *name, int realtime) +{ + struct sip_user *u = ASTOBJ_CONTAINER_FIND(&userl, name); + if (!u && realtime) + u = realtime_user(name); + return u; +} + +/*! \brief Set nat mode on the various data sockets */ +static void do_setnat(struct sip_pvt *p, int natflags) +{ + const char *mode = natflags ? "On" : "Off"; + + if (p->rtp) { + if (option_debug) + ast_log(LOG_DEBUG, "Setting NAT on RTP to %s\n", mode); + ast_rtp_setnat(p->rtp, natflags); + } + if (p->vrtp) { + if (option_debug) + ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", mode); + ast_rtp_setnat(p->vrtp, natflags); + } + if (p->udptl) { + if (option_debug) + ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", mode); + ast_udptl_setnat(p->udptl, natflags); + } +} + +/*! \brief Create address structure from peer reference. + * return -1 on error, 0 on success. + */ +static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer) +{ + if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) && + (!peer->maxms || ((peer->lastms >= 0) && (peer->lastms <= peer->maxms)))) { + dialog->sa = (peer->addr.sin_addr.s_addr) ? peer->addr : peer->defaddr; + dialog->recv = dialog->sa; + } else + return -1; + + ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY); + ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY); + dialog->capability = peer->capability; + if ((!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) || !(dialog->capability & AST_FORMAT_VIDEO_MASK)) && dialog->vrtp) { + ast_rtp_destroy(dialog->vrtp); + dialog->vrtp = NULL; + } + dialog->prefs = peer->prefs; + if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) { + dialog->t38.capability = global_t38_capability; + if (dialog->udptl) { + if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_FEC ) + dialog->t38.capability |= T38FAX_UDP_EC_FEC; + else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY ) + dialog->t38.capability |= T38FAX_UDP_EC_REDUNDANCY; + else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_NONE ) + dialog->t38.capability |= T38FAX_UDP_EC_NONE; + dialog->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF; + if (option_debug > 1) + ast_log(LOG_DEBUG,"Our T38 capability (%d)\n", dialog->t38.capability); + } + dialog->t38.jointcapability = dialog->t38.capability; + } else if (dialog->udptl) { + ast_udptl_destroy(dialog->udptl); + dialog->udptl = NULL; + } + do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE ); + + if (dialog->rtp) { + ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833); + ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE)); + ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout); + ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout); + ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive); + /* Set Frame packetization */ + ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs); + dialog->autoframing = peer->autoframing; + } + if (dialog->vrtp) { + ast_rtp_setdtmf(dialog->vrtp, 0); + ast_rtp_setdtmfcompensate(dialog->vrtp, 0); + ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout); + ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout); + ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive); + } + + ast_string_field_set(dialog, peername, peer->name); + ast_string_field_set(dialog, authname, peer->username); + ast_string_field_set(dialog, username, peer->username); + ast_string_field_set(dialog, peersecret, peer->secret); + ast_string_field_set(dialog, peermd5secret, peer->md5secret); + ast_string_field_set(dialog, mohsuggest, peer->mohsuggest); + ast_string_field_set(dialog, mohinterpret, peer->mohinterpret); + ast_string_field_set(dialog, tohost, peer->tohost); + ast_string_field_set(dialog, fullcontact, peer->fullcontact); + if (!dialog->initreq.headers && !ast_strlen_zero(peer->fromdomain)) { + char *tmpcall; + char *c; + tmpcall = ast_strdupa(dialog->callid); + c = strchr(tmpcall, '@'); + if (c) { + *c = '\0'; + ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain); + } + } + if (ast_strlen_zero(dialog->tohost)) + ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr)); + if (!ast_strlen_zero(peer->fromdomain)) + ast_string_field_set(dialog, fromdomain, peer->fromdomain); + if (!ast_strlen_zero(peer->fromuser)) + ast_string_field_set(dialog, fromuser, peer->fromuser); + if (!ast_strlen_zero(peer->language)) + ast_string_field_set(dialog, language, peer->language); + dialog->maxtime = peer->maxms; + dialog->callgroup = peer->callgroup; + dialog->pickupgroup = peer->pickupgroup; + dialog->peerauth = peer->auth; + dialog->allowtransfer = peer->allowtransfer; + /* Set timer T1 to RTT for this peer (if known by qualify=) */ + /* Minimum is settable or default to 100 ms */ + if (peer->maxms && peer->lastms) + dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms; + if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) || + (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) + dialog->noncodeccapability |= AST_RTP_DTMF; + else + dialog->noncodeccapability &= ~AST_RTP_DTMF; + dialog->jointnoncodeccapability = dialog->noncodeccapability; + ast_string_field_set(dialog, context, peer->context); + dialog->rtptimeout = peer->rtptimeout; + if (peer->call_limit) + ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT); + dialog->maxcallbitrate = peer->maxcallbitrate; + + return 0; +} + +/*! \brief create address structure from peer name + * Or, if peer not found, find it in the global DNS + * returns TRUE (-1) on failure, FALSE on success */ +static int create_addr(struct sip_pvt *dialog, const char *opeer) +{ + struct hostent *hp; + struct ast_hostent ahp; + struct sip_peer *p; + char *port; + int portno; + char host[MAXHOSTNAMELEN], *hostn; + char peer[256]; + + ast_copy_string(peer, opeer, sizeof(peer)); + port = strchr(peer, ':'); + if (port) + *port++ = '\0'; + dialog->sa.sin_family = AF_INET; + dialog->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */ + p = find_peer(peer, NULL, 1, 0); + + if (p) { + int res = create_addr_from_peer(dialog, p); + if (port) { + portno = atoi(port); + dialog->sa.sin_port = dialog->recv.sin_port = htons(portno); + } + ASTOBJ_UNREF(p, sip_destroy_peer); + return res; + } + hostn = peer; + portno = port ? atoi(port) : STANDARD_SIP_PORT; + if (srvlookup) { + char service[MAXHOSTNAMELEN]; + int tportno; + int ret; + + snprintf(service, sizeof(service), "_sip._udp.%s", peer); + ret = ast_get_srv(NULL, host, sizeof(host), &tportno, service); + if (ret > 0) { + hostn = host; + portno = tportno; + } + } + hp = ast_gethostbyname(hostn, &ahp); + if (!hp) { + ast_log(LOG_WARNING, "No such host: %s\n", peer); + return -1; + } + ast_string_field_set(dialog, tohost, peer); + memcpy(&dialog->sa.sin_addr, hp->h_addr, sizeof(dialog->sa.sin_addr)); + dialog->sa.sin_port = htons(portno); + dialog->recv = dialog->sa; + return 0; +} + +/*! \brief Scheduled congestion on a call */ +static int auto_congest(const void *nothing) +{ + struct sip_pvt *p = (struct sip_pvt *)nothing; + + ast_mutex_lock(&p->lock); + p->initid = -1; + if (p->owner) { + /* XXX fails on possible deadlock */ + if (!ast_channel_trylock(p->owner)) { + ast_log(LOG_NOTICE, "Auto-congesting %s\n", p->owner->name); + append_history(p, "Cong", "Auto-congesting (timer)"); + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_channel_unlock(p->owner); + } + } + ast_mutex_unlock(&p->lock); + return 0; +} + + +/*! \brief Initiate SIP call from PBX + * used from the dial() application */ +static int sip_call(struct ast_channel *ast, char *dest, int timeout) +{ + int res, xmitres = 0; + struct sip_pvt *p; + struct varshead *headp; + struct ast_var_t *current; + const char *referer = NULL; /* SIP refererer */ + + p = ast->tech_pvt; + if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) { + ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name); + return -1; + } + + /* Check whether there is vxml_url, distinctive ring variables */ + headp=&ast->varshead; + AST_LIST_TRAVERSE(headp,current,entries) { + /* Check whether there is a VXML_URL variable */ + if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) { + p->options->vxml_url = ast_var_value(current); + } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) { + p->options->uri_options = ast_var_value(current); + } else if (!p->options->distinctive_ring && !strcasecmp(ast_var_name(current), "ALERT_INFO")) { + /* Check whether there is a ALERT_INFO variable */ + p->options->distinctive_ring = ast_var_value(current); + } else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) { + /* Check whether there is a variable with a name starting with SIPADDHEADER */ + p->options->addsipheaders = 1; + } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) { + /* This is a transfered call */ + p->options->transfer = 1; + } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) { + /* This is the referer */ + referer = ast_var_value(current); + } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) { + /* We're replacing a call. */ + p->options->replaces = ast_var_value(current); + } else if (!strcasecmp(ast_var_name(current), "T38CALL")) { + p->t38.state = T38_LOCAL_DIRECT; + if (option_debug) + ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name); + } + + } + + res = 0; + ast_set_flag(&p->flags[0], SIP_OUTGOING); + + if (p->options->transfer) { + char buf[SIPBUFSIZE/2]; + + if (referer) { + if (sipdebug && option_debug > 2) + ast_log(LOG_DEBUG, "Call for %s transfered by %s\n", p->username, referer); + snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer); + } else + snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name); + ast_string_field_set(p, cid_name, buf); + } + if (option_debug) + ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username); + + res = update_call_counter(p, INC_CALL_RINGING); + if ( res != -1 ) { + p->callingpres = ast->cid.cid_pres; + p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec); + p->jointnoncodeccapability = p->noncodeccapability; + + /* If there are no audio formats left to offer, punt */ + if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) { + ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username); + res = -1; + } else { + p->t38.jointcapability = p->t38.capability; + if (option_debug > 1) + ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability); + xmitres = transmit_invite(p, SIP_INVITE, 1, 2); + if (xmitres == XMIT_ERROR) + return -1; /* Transmission error */ + + p->invitestate = INV_CALLING; + + /* Initialize auto-congest time */ + AST_SCHED_DEL(sched, p->initid); + p->initid = ast_sched_add(sched, p->maxtime ? (p->maxtime * 4) : SIP_TRANS_TIMEOUT, auto_congest, p); + } + } else { + ast->hangupcause = AST_CAUSE_USER_BUSY; + } + return res; +} + +/*! \brief Destroy registry object + Objects created with the register= statement in static configuration */ +static void sip_registry_destroy(struct sip_registry *reg) +{ + /* Really delete */ + if (option_debug > 2) + ast_log(LOG_DEBUG, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname); + + if (reg->call) { + /* Clear registry before destroying to ensure + we don't get reentered trying to grab the registry lock */ + reg->call->registry = NULL; + if (option_debug > 2) + ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname); + sip_destroy(reg->call); + } + AST_SCHED_DEL(sched, reg->expire); + AST_SCHED_DEL(sched, reg->timeout); + ast_string_field_free_memory(reg); + regobjs--; + free(reg); + +} + +/*! \brief Execute destruction of SIP dialog structure, release memory */ +static int __sip_destroy(struct sip_pvt *p, int lockowner) +{ + struct sip_pvt *cur, *prev = NULL; + struct sip_pkt *cp; + struct sip_request *req; + + /* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */ + if (p->rtp && ast_rtp_get_bridged(p->rtp)) { + ast_verbose("Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text); + return -1; + } + + if (p->vrtp && ast_rtp_get_bridged(p->vrtp)) { + ast_verbose("Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text); + return -1; + } + + if (sip_debug_test_pvt(p) || option_debug > 2) + ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text); + + if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) { + update_call_counter(p, DEC_CALL_LIMIT); + if (option_debug > 1) + ast_log(LOG_DEBUG, "This call did not properly clean up call limits. Call ID %s\n", p->callid); + } + + /* Unlink us from the owner if we have one */ + if (p->owner) { + if (lockowner) + ast_channel_lock(p->owner); + if (option_debug) + ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name); + p->owner->tech_pvt = NULL; + /* Make sure that the channel knows its backend is going away */ + p->owner->_softhangup |= AST_SOFTHANGUP_DEV; + if (lockowner) + ast_channel_unlock(p->owner); + /* Give the channel a chance to react before deallocation */ + usleep(1); + } + + /* Remove link from peer to subscription of MWI */ + if (p->relatedpeer) { + if (p->relatedpeer->mwipvt == p) { + p->relatedpeer->mwipvt = NULL; + } + ASTOBJ_UNREF(p->relatedpeer, sip_destroy_peer); + } + + if (dumphistory) + sip_dump_history(p); + + if (p->options) + free(p->options); + + if (p->stateid > -1) + ast_extension_state_del(p->stateid, NULL); + AST_SCHED_DEL(sched, p->initid); + AST_SCHED_DEL(sched, p->waitid); + AST_SCHED_DEL(sched, p->autokillid); + AST_SCHED_DEL(sched, p->request_queue_sched_id); + + if (p->rtp) { + ast_rtp_destroy(p->rtp); + } + if (p->vrtp) { + ast_rtp_destroy(p->vrtp); + } + if (p->udptl) + ast_udptl_destroy(p->udptl); + if (p->refer) + free(p->refer); + if (p->route) { + free_old_route(p->route); + p->route = NULL; + } + if (p->registry) { + if (p->registry->call == p) + p->registry->call = NULL; + ASTOBJ_UNREF(p->registry, sip_registry_destroy); + } + + /* Clear history */ + if (p->history) { + struct sip_history *hist; + while ( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) ) { + free(hist); + p->history_entries--; + } + free(p->history); + p->history = NULL; + } + + while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) { + ast_free(req); + } + + for (prev = NULL, cur = iflist; cur; prev = cur, cur = cur->next) { + if (cur == p) { + UNLINK(cur, iflist, prev); + break; + } + } + if (!cur) { + ast_log(LOG_WARNING, "Trying to destroy \"%s\", not found in dialog list?!?! \n", p->callid); + return 0; + } + + /* remove all current packets in this dialog */ + while((cp = p->packets)) { + p->packets = p->packets->next; + AST_SCHED_DEL(sched, cp->retransid); + free(cp); + } + if (p->chanvars) { + ast_variables_destroy(p->chanvars); + p->chanvars = NULL; + } + ast_mutex_destroy(&p->lock); + + ast_string_field_free_memory(p); + + free(p); + return 0; +} + +/*! \brief update_call_counter: Handle call_limit for SIP users + * Setting a call-limit will cause calls above the limit not to be accepted. + * + * Remember that for a type=friend, there's one limit for the user and + * another for the peer, not a combined call limit. + * This will cause unexpected behaviour in subscriptions, since a "friend" + * is *two* devices in Asterisk, not one. + * + * Thought: For realtime, we should propably update storage with inuse counter... + * + * \return 0 if call is ok (no call limit, below treshold) + * -1 on rejection of call + * + */ +static int update_call_counter(struct sip_pvt *fup, int event) +{ + char name[256]; + int *inuse = NULL, *call_limit = NULL, *inringing = NULL; + int outgoing = ast_test_flag(&fup->flags[1], SIP_PAGE2_OUTGOING_CALL); + struct sip_user *u = NULL; + struct sip_peer *p = NULL; + + if (option_debug > 2) + ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming"); + + /* Test if we need to check call limits, in order to avoid + realtime lookups if we do not need it */ + if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT) && !ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD)) + return 0; + + ast_copy_string(name, fup->username, sizeof(name)); + + /* Check the list of users only for incoming calls */ + if (global_limitonpeers == FALSE && !outgoing && (u = find_user(name, 1))) { + inuse = &u->inUse; + call_limit = &u->call_limit; + inringing = NULL; + } else if ( (p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, 1, 0) ) ) { /* Try to find peer */ + inuse = &p->inUse; + call_limit = &p->call_limit; + inringing = &p->inRinging; + ast_copy_string(name, fup->peername, sizeof(name)); + } + if (!p && !u) { + if (option_debug > 1) + ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name); + return 0; + } + + switch(event) { + /* incoming and outgoing affects the inUse counter */ + case DEC_CALL_LIMIT: + if ( *inuse > 0 ) { + if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) { + (*inuse)--; + ast_clear_flag(&fup->flags[0], SIP_INC_COUNT); + } + } else { + *inuse = 0; + } + if (inringing) { + if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) { + if (*inringing > 0) + (*inringing)--; + else if (!ast_test_flag(&p->flags[0], SIP_REALTIME) || ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) + ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", fup->peername); + ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING); + } + } + if (ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD) && global_notifyhold) { + ast_clear_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD); + sip_peer_hold(fup, 0); + } + if (option_debug > 1 || sipdebug) { + ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit); + } + break; + + case INC_CALL_RINGING: + case INC_CALL_LIMIT: + if (*call_limit > 0 ) { + if (*inuse >= *call_limit) { + ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit); + if (u) + ASTOBJ_UNREF(u, sip_destroy_user); + else + ASTOBJ_UNREF(p, sip_destroy_peer); + return -1; + } + } + if (inringing && (event == INC_CALL_RINGING)) { + if (!ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) { + (*inringing)++; + ast_set_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING); + } + } + /* Continue */ + (*inuse)++; + ast_set_flag(&fup->flags[0], SIP_INC_COUNT); + if (option_debug > 1 || sipdebug) { + ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit); + } + break; + + case DEC_CALL_RINGING: + if (inringing) { + if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) { + if (*inringing > 0) + (*inringing)--; + else if (!ast_test_flag(&p->flags[0], SIP_REALTIME) || ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) + ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", p->name); + ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING); + } + } + break; + + default: + ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event); + } + if (p) { + ast_device_state_changed("SIP/%s", p->name); + ASTOBJ_UNREF(p, sip_destroy_peer); + } else /* u must be set */ + ASTOBJ_UNREF(u, sip_destroy_user); + return 0; +} + +/*! \brief Destroy SIP call structure */ +static void sip_destroy(struct sip_pvt *p) +{ + ast_mutex_lock(&iflock); + if (option_debug > 2) + ast_log(LOG_DEBUG, "Destroying SIP dialog %s\n", p->callid); + __sip_destroy(p, 1); + ast_mutex_unlock(&iflock); +} + +/*! \brief Convert SIP hangup causes to Asterisk hangup causes */ +static int hangup_sip2cause(int cause) +{ + /* Possible values taken from causes.h */ + + switch(cause) { + case 401: /* Unauthorized */ + return AST_CAUSE_CALL_REJECTED; + case 403: /* Not found */ + return AST_CAUSE_CALL_REJECTED; + case 404: /* Not found */ + return AST_CAUSE_UNALLOCATED; + case 405: /* Method not allowed */ + return AST_CAUSE_INTERWORKING; + case 407: /* Proxy authentication required */ + return AST_CAUSE_CALL_REJECTED; + case 408: /* No reaction */ + return AST_CAUSE_NO_USER_RESPONSE; + case 409: /* Conflict */ + return AST_CAUSE_NORMAL_TEMPORARY_FAILURE; + case 410: /* Gone */ + return AST_CAUSE_UNALLOCATED; + case 411: /* Length required */ + return AST_CAUSE_INTERWORKING; + case 413: /* Request entity too large */ + return AST_CAUSE_INTERWORKING; + case 414: /* Request URI too large */ + return AST_CAUSE_INTERWORKING; + case 415: /* Unsupported media type */ + return AST_CAUSE_INTERWORKING; + case 420: /* Bad extension */ + return AST_CAUSE_NO_ROUTE_DESTINATION; + case 480: /* No answer */ + return AST_CAUSE_NO_ANSWER; + case 481: /* No answer */ + return AST_CAUSE_INTERWORKING; + case 482: /* Loop detected */ + return AST_CAUSE_INTERWORKING; + case 483: /* Too many hops */ + return AST_CAUSE_NO_ANSWER; + case 484: /* Address incomplete */ + return AST_CAUSE_INVALID_NUMBER_FORMAT; + case 485: /* Ambigous */ + return AST_CAUSE_UNALLOCATED; + case 486: /* Busy everywhere */ + return AST_CAUSE_BUSY; + case 487: /* Request terminated */ + return AST_CAUSE_INTERWORKING; + case 488: /* No codecs approved */ + return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; + case 491: /* Request pending */ + return AST_CAUSE_INTERWORKING; + case 493: /* Undecipherable */ + return AST_CAUSE_INTERWORKING; + case 500: /* Server internal failure */ + return AST_CAUSE_FAILURE; + case 501: /* Call rejected */ + return AST_CAUSE_FACILITY_REJECTED; + case 502: + return AST_CAUSE_DESTINATION_OUT_OF_ORDER; + case 503: /* Service unavailable */ + return AST_CAUSE_CONGESTION; + case 504: /* Gateway timeout */ + return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE; + case 505: /* SIP version not supported */ + return AST_CAUSE_INTERWORKING; + case 600: /* Busy everywhere */ + return AST_CAUSE_USER_BUSY; + case 603: /* Decline */ + return AST_CAUSE_CALL_REJECTED; + case 604: /* Does not exist anywhere */ + return AST_CAUSE_UNALLOCATED; + case 606: /* Not acceptable */ + return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; + default: + return AST_CAUSE_NORMAL; + } + /* Never reached */ + return 0; +} + +/*! \brief Convert Asterisk hangup causes to SIP codes +\verbatim + Possible values from causes.h + AST_CAUSE_NOTDEFINED AST_CAUSE_NORMAL AST_CAUSE_BUSY + AST_CAUSE_FAILURE AST_CAUSE_CONGESTION AST_CAUSE_UNALLOCATED + + In addition to these, a lot of PRI codes is defined in causes.h + ...should we take care of them too ? + + Quote RFC 3398 + + ISUP Cause value SIP response + ---------------- ------------ + 1 unallocated number 404 Not Found + 2 no route to network 404 Not found + 3 no route to destination 404 Not found + 16 normal call clearing --- (*) + 17 user busy 486 Busy here + 18 no user responding 408 Request Timeout + 19 no answer from the user 480 Temporarily unavailable + 20 subscriber absent 480 Temporarily unavailable + 21 call rejected 403 Forbidden (+) + 22 number changed (w/o diagnostic) 410 Gone + 22 number changed (w/ diagnostic) 301 Moved Permanently + 23 redirection to new destination 410 Gone + 26 non-selected user clearing 404 Not Found (=) + 27 destination out of order 502 Bad Gateway + 28 address incomplete 484 Address incomplete + 29 facility rejected 501 Not implemented + 31 normal unspecified 480 Temporarily unavailable +\endverbatim +*/ +static const char *hangup_cause2sip(int cause) +{ + switch (cause) { + case AST_CAUSE_UNALLOCATED: /* 1 */ + case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */ + case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */ + return "404 Not Found"; + case AST_CAUSE_CONGESTION: /* 34 */ + case AST_CAUSE_SWITCH_CONGESTION: /* 42 */ + return "503 Service Unavailable"; + case AST_CAUSE_NO_USER_RESPONSE: /* 18 */ + return "408 Request Timeout"; + case AST_CAUSE_NO_ANSWER: /* 19 */ + case AST_CAUSE_UNREGISTERED: /* 20 */ + return "480 Temporarily unavailable"; + case AST_CAUSE_CALL_REJECTED: /* 21 */ + return "403 Forbidden"; + case AST_CAUSE_NUMBER_CHANGED: /* 22 */ + return "410 Gone"; + case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */ + return "480 Temporarily unavailable"; + case AST_CAUSE_INVALID_NUMBER_FORMAT: + return "484 Address incomplete"; + case AST_CAUSE_USER_BUSY: + return "486 Busy here"; + case AST_CAUSE_FAILURE: + return "500 Server internal failure"; + case AST_CAUSE_FACILITY_REJECTED: /* 29 */ + return "501 Not Implemented"; + case AST_CAUSE_CHAN_NOT_IMPLEMENTED: + return "503 Service Unavailable"; + /* Used in chan_iax2 */ + case AST_CAUSE_DESTINATION_OUT_OF_ORDER: + return "502 Bad Gateway"; + case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */ + return "488 Not Acceptable Here"; + + case AST_CAUSE_NOTDEFINED: + default: + if (option_debug) + ast_log(LOG_DEBUG, "AST hangup cause %d (no match found in SIP)\n", cause); + return NULL; + } + + /* Never reached */ + return 0; +} + + +/*! \brief sip_hangup: Hangup SIP call + * Part of PBX interface, called from ast_hangup */ +static int sip_hangup(struct ast_channel *ast) +{ + struct sip_pvt *p = ast->tech_pvt; + int needcancel = FALSE; + int needdestroy = 0; + struct ast_channel *oldowner = ast; + + if (!p) { + if (option_debug) + ast_log(LOG_DEBUG, "Asked to hangup channel that was not connected\n"); + return 0; + } + + if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) { + if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) { + if (option_debug && sipdebug) + ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username); + update_call_counter(p, DEC_CALL_LIMIT); + } + if (option_debug >3) + ast_log(LOG_DEBUG, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid); + if (p->autokillid > -1 && sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Really hang up next time */ + ast_clear_flag(&p->flags[0], SIP_NEEDDESTROY); + p->owner->tech_pvt = NULL; + p->owner = NULL; /* Owner will be gone after we return, so take it away */ + return 0; + } + if (option_debug) { + if (ast_test_flag(ast, AST_FLAG_ZOMBIE) && p->refer && option_debug) + ast_log(LOG_DEBUG, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid); + else { + if (option_debug) + ast_log(LOG_DEBUG, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid); + } + } + if (option_debug && ast_test_flag(ast, AST_FLAG_ZOMBIE)) + ast_log(LOG_DEBUG, "Hanging up zombie call. Be scared.\n"); + + ast_mutex_lock(&p->lock); + if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) { + if (option_debug && sipdebug) + ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username); + update_call_counter(p, DEC_CALL_LIMIT); + } + + /* Determine how to disconnect */ + if (p->owner != ast) { + ast_log(LOG_WARNING, "Huh? We aren't the owner? Can't hangup call.\n"); + ast_mutex_unlock(&p->lock); + return 0; + } + /* If the call is not UP, we need to send CANCEL instead of BYE */ + if (ast->_state == AST_STATE_RING || ast->_state == AST_STATE_RINGING || (p->invitestate < INV_COMPLETED && ast->_state != AST_STATE_UP)) { + needcancel = TRUE; + if (option_debug > 3) + ast_log(LOG_DEBUG, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state)); + } + + stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */ + + append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", p->owner ? ast_cause2str(p->owner->hangupcause) : "Unknown"); + + /* Disconnect */ + if (p->vad) + ast_dsp_free(p->vad); + + p->owner = NULL; + ast->tech_pvt = NULL; + + ast_module_unref(ast_module_info->self); + + /* Do not destroy this pvt until we have timeout or + get an answer to the BYE or INVITE/CANCEL + If we get no answer during retransmit period, drop the call anyway. + (Sorry, mother-in-law, you can't deny a hangup by sending + 603 declined to BYE...) + */ + if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) + needdestroy = 1; /* Set destroy flag at end of this function */ + else if (p->invitestate != INV_CALLING) + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + + /* Start the process if it's not already started */ + if (!ast_test_flag(&p->flags[0], SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) { + if (needcancel) { /* Outgoing call, not up */ + if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) { + /* stop retransmitting an INVITE that has not received a response */ + __sip_pretend_ack(p); + p->invitestate = INV_CANCELLED; + + /* if we can't send right now, mark it pending */ + if (p->invitestate == INV_CALLING) { + /* We can't send anything in CALLING state */ + ast_set_flag(&p->flags[0], SIP_PENDINGBYE); + /* Do we need a timer here if we don't hear from them at all? Yes we do or else we will get hung dialogs and those are no fun. */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + append_history(p, "DELAY", "Not sending cancel, waiting for timeout"); + } else { + /* Send a new request: CANCEL */ + transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE); + /* Actually don't destroy us yet, wait for the 487 on our original + INVITE, but do set an autodestruct just in case we never get it. */ + needdestroy = 0; + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } + if ( p->initid != -1 ) { + /* channel still up - reverse dec of inUse counter + only if the channel is not auto-congested */ + update_call_counter(p, INC_CALL_LIMIT); + } + } else { /* Incoming call, not up */ + const char *res; + if (ast->hangupcause && (res = hangup_cause2sip(ast->hangupcause))) + transmit_response_reliable(p, res, &p->initreq); + else + transmit_response_reliable(p, "603 Declined", &p->initreq); + p->invitestate = INV_TERMINATED; + } + } else { /* Call is in UP state, send BYE */ + if (!p->pendinginvite) { + char *audioqos = ""; + char *videoqos = ""; + if (p->rtp) + audioqos = ast_rtp_get_quality(p->rtp, NULL); + if (p->vrtp) + videoqos = ast_rtp_get_quality(p->vrtp, NULL); + /* Send a hangup */ + transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1); + + /* Get RTCP quality before end of call */ + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) { + if (p->rtp) + append_history(p, "RTCPaudio", "Quality:%s", audioqos); + if (p->vrtp) + append_history(p, "RTCPvideo", "Quality:%s", videoqos); + } + if (p->rtp && oldowner) + pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", audioqos); + if (p->vrtp && oldowner) + pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", videoqos); + } else { + /* Note we will need a BYE when this all settles out + but we can't send one while we have "INVITE" outstanding. */ + ast_set_flag(&p->flags[0], SIP_PENDINGBYE); + ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE); + AST_SCHED_DEL(sched, p->waitid); + if (sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + } + } + } + if (needdestroy) + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + ast_mutex_unlock(&p->lock); + return 0; +} + +/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */ +static void try_suggested_sip_codec(struct sip_pvt *p) +{ + int fmt; + const char *codec; + + codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC"); + if (!codec) + return; + + fmt = ast_getformatbyname(codec); + if (fmt) { + ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC} variable\n", codec); + if (p->jointcapability & fmt) { + p->jointcapability &= fmt; + p->capability &= fmt; + } else + ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n"); + } else + ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec); + return; +} + +/*! \brief sip_answer: Answer SIP call , send 200 OK on Invite + * Part of PBX interface */ +static int sip_answer(struct ast_channel *ast) +{ + int res = 0; + struct sip_pvt *p = ast->tech_pvt; + + ast_mutex_lock(&p->lock); + if (ast->_state != AST_STATE_UP) { + try_suggested_sip_codec(p); + + ast_setstate(ast, AST_STATE_UP); + if (option_debug) + ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name); + if (p->t38.state == T38_PEER_DIRECT) { + p->t38.state = T38_ENABLED; + if (option_debug > 1) + ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name); + res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL); + ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + } else { + res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL); + ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + } + } + ast_mutex_unlock(&p->lock); + return res; +} + +/*! \brief Send frame to media channel (rtp) */ +static int sip_write(struct ast_channel *ast, struct ast_frame *frame) +{ + struct sip_pvt *p = ast->tech_pvt; + int res = 0; + + switch (frame->frametype) { + case AST_FRAME_VOICE: + if (!(frame->subclass & ast->nativeformats)) { + char s1[512], s2[512], s3[512]; + ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %s(%d) read/write = %s(%d)/%s(%d)\n", + frame->subclass, + ast_getformatname_multiple(s1, sizeof(s1) - 1, ast->nativeformats & AST_FORMAT_AUDIO_MASK), + ast->nativeformats & AST_FORMAT_AUDIO_MASK, + ast_getformatname_multiple(s2, sizeof(s2) - 1, ast->readformat), + ast->readformat, + ast_getformatname_multiple(s3, sizeof(s3) - 1, ast->writeformat), + ast->writeformat); + return 0; + } + if (p) { + ast_mutex_lock(&p->lock); + if (p->rtp) { + /* If channel is not up, activate early media session */ + if ((ast->_state != AST_STATE_UP) && + !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) && + !ast_test_flag(&p->flags[0], SIP_OUTGOING)) { + ast_rtp_new_source(p->rtp); + p->invitestate = INV_EARLY_MEDIA; + transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE); + ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT); + } + p->lastrtptx = time(NULL); + res = ast_rtp_write(p->rtp, frame); + } + ast_mutex_unlock(&p->lock); + } + break; + case AST_FRAME_VIDEO: + if (p) { + ast_mutex_lock(&p->lock); + if (p->vrtp) { + /* Activate video early media */ + if ((ast->_state != AST_STATE_UP) && + !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) && + !ast_test_flag(&p->flags[0], SIP_OUTGOING)) { + p->invitestate = INV_EARLY_MEDIA; + transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE); + ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT); + } + p->lastrtptx = time(NULL); + res = ast_rtp_write(p->vrtp, frame); + } + ast_mutex_unlock(&p->lock); + } + break; + case AST_FRAME_IMAGE: + return 0; + break; + case AST_FRAME_MODEM: + if (p) { + ast_mutex_lock(&p->lock); + /* UDPTL requires two-way communication, so early media is not needed here. + we simply forget the frames if we get modem frames before the bridge is up. + Fax will re-transmit. + */ + if (p->udptl && ast->_state == AST_STATE_UP) + res = ast_udptl_write(p->udptl, frame); + ast_mutex_unlock(&p->lock); + } + break; + default: + ast_log(LOG_WARNING, "Can't send %d type frames with SIP write\n", frame->frametype); + return 0; + } + + return res; +} + +/*! \brief sip_fixup: Fix up a channel: If a channel is consumed, this is called. + Basically update any ->owner links */ +static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan) +{ + int ret = -1; + struct sip_pvt *p; + + if (newchan && ast_test_flag(newchan, AST_FLAG_ZOMBIE) && option_debug) + ast_log(LOG_DEBUG, "New channel is zombie\n"); + if (oldchan && ast_test_flag(oldchan, AST_FLAG_ZOMBIE) && option_debug) + ast_log(LOG_DEBUG, "Old channel is zombie\n"); + + if (!newchan || !newchan->tech_pvt) { + if (!newchan) + ast_log(LOG_WARNING, "No new channel! Fixup of %s failed.\n", oldchan->name); + else + ast_log(LOG_WARNING, "No SIP tech_pvt! Fixup of %s failed.\n", oldchan->name); + return -1; + } + p = newchan->tech_pvt; + + if (!p) { + ast_log(LOG_WARNING, "No pvt after masquerade. Strange things may happen\n"); + return -1; + } + + ast_mutex_lock(&p->lock); + append_history(p, "Masq", "Old channel: %s\n", oldchan->name); + append_history(p, "Masq (cont)", "...new owner: %s\n", newchan->name); + if (p->owner != oldchan) + ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner); + else { + p->owner = newchan; + /* Re-invite RTP back to Asterisk. Needed if channel is masqueraded out of a native + RTP bridge (i.e., RTP not going through Asterisk): RTP bridge code might not be + able to do this if the masquerade happens before the bridge breaks (e.g., AMI + redirect of both channels). Note that a channel can not be masqueraded *into* + a native bridge. So there is no danger that this breaks a native bridge that + should stay up. */ + sip_set_rtp_peer(newchan, NULL, NULL, 0, 0); + ret = 0; + } + if (option_debug > 2) + ast_log(LOG_DEBUG, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, p->owner->name, oldchan->name); + + ast_mutex_unlock(&p->lock); + return ret; +} + +static int sip_senddigit_begin(struct ast_channel *ast, char digit) +{ + struct sip_pvt *p = ast->tech_pvt; + int res = 0; + + ast_mutex_lock(&p->lock); + switch (ast_test_flag(&p->flags[0], SIP_DTMF)) { + case SIP_DTMF_INBAND: + res = -1; /* Tell Asterisk to generate inband indications */ + break; + case SIP_DTMF_RFC2833: + if (p->rtp) + ast_rtp_senddigit_begin(p->rtp, digit); + break; + default: + break; + } + ast_mutex_unlock(&p->lock); + + return res; +} + +/*! \brief Send DTMF character on SIP channel + within one call, we're able to transmit in many methods simultaneously */ +static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration) +{ + struct sip_pvt *p = ast->tech_pvt; + int res = 0; + + ast_mutex_lock(&p->lock); + switch (ast_test_flag(&p->flags[0], SIP_DTMF)) { + case SIP_DTMF_INFO: + transmit_info_with_digit(p, digit, duration); + break; + case SIP_DTMF_RFC2833: + if (p->rtp) + ast_rtp_senddigit_end(p->rtp, digit); + break; + case SIP_DTMF_INBAND: + res = -1; /* Tell Asterisk to stop inband indications */ + break; + } + ast_mutex_unlock(&p->lock); + + return res; +} + +/*! \brief Transfer SIP call */ +static int sip_transfer(struct ast_channel *ast, const char *dest) +{ + struct sip_pvt *p = ast->tech_pvt; + int res; + + if (dest == NULL) /* functions below do not take a NULL */ + dest = ""; + ast_mutex_lock(&p->lock); + if (ast->_state == AST_STATE_RING) + res = sip_sipredirect(p, dest); + else + res = transmit_refer(p, dest); + ast_mutex_unlock(&p->lock); + return res; +} + +/*! \brief Play indication to user + * With SIP a lot of indications is sent as messages, letting the device play + the indication - busy signal, congestion etc + \return -1 to force ast_indicate to send indication in audio, 0 if SIP can handle the indication by sending a message +*/ +static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen) +{ + struct sip_pvt *p = ast->tech_pvt; + int res = 0; + + ast_mutex_lock(&p->lock); + switch(condition) { + case AST_CONTROL_RINGING: + if (ast->_state == AST_STATE_RING) { + p->invitestate = INV_EARLY_MEDIA; + if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) || + (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) { + /* Send 180 ringing if out-of-band seems reasonable */ + transmit_response(p, "180 Ringing", &p->initreq); + ast_set_flag(&p->flags[0], SIP_RINGING); + if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_YES) + break; + } else { + /* Well, if it's not reasonable, just send in-band */ + } + } + res = -1; + break; + case AST_CONTROL_BUSY: + if (ast->_state != AST_STATE_UP) { + transmit_response_reliable(p, "486 Busy Here", &p->initreq); + p->invitestate = INV_COMPLETED; + sip_alreadygone(p); + ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV); + break; + } + res = -1; + break; + case AST_CONTROL_CONGESTION: + if (ast->_state != AST_STATE_UP) { + transmit_response_reliable(p, "503 Service Unavailable", &p->initreq); + p->invitestate = INV_COMPLETED; + sip_alreadygone(p); + ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV); + break; + } + res = -1; + break; + case AST_CONTROL_PROCEEDING: + if ((ast->_state != AST_STATE_UP) && + !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) && + !ast_test_flag(&p->flags[0], SIP_OUTGOING)) { + transmit_response(p, "100 Trying", &p->initreq); + p->invitestate = INV_PROCEEDING; + break; + } + res = -1; + break; + case AST_CONTROL_PROGRESS: + if ((ast->_state != AST_STATE_UP) && + !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) && + !ast_test_flag(&p->flags[0], SIP_OUTGOING)) { + p->invitestate = INV_EARLY_MEDIA; + transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE); + ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT); + break; + } + res = -1; + break; + case AST_CONTROL_HOLD: + ast_rtp_new_source(p->rtp); + ast_moh_start(ast, data, p->mohinterpret); + break; + case AST_CONTROL_UNHOLD: + ast_rtp_new_source(p->rtp); + ast_moh_stop(ast); + break; + case AST_CONTROL_VIDUPDATE: /* Request a video frame update */ + if (p->vrtp && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) { + transmit_info_with_vidupdate(p); + /* ast_rtcp_send_h261fur(p->vrtp); */ + } else + res = -1; + break; + case AST_CONTROL_SRCUPDATE: + ast_rtp_new_source(p->rtp); + break; + case -1: + res = -1; + break; + default: + ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition); + res = -1; + break; + } + ast_mutex_unlock(&p->lock); + return res; +} + + +/*! \brief Initiate a call in the SIP channel + called from sip_request_call (calls from the pbx ) for outbound channels + and from handle_request_invite for inbound channels + +*/ +static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *title) +{ + struct ast_channel *tmp; + struct ast_variable *v = NULL; + int fmt; + int what; + int needvideo = 0, video = 0; + char *decoded_exten; + { + const char *my_name; /* pick a good name */ + + if (title) + my_name = title; + else if ( (my_name = strchr(i->fromdomain,':')) ) + my_name++; /* skip ':' */ + else + my_name = i->fromdomain; + + ast_mutex_unlock(&i->lock); + /* Don't hold a sip pvt lock while we allocate a channel */ + tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, i->amaflags, "SIP/%s-%08x", my_name, (int)(long) i); + + } + if (!tmp) { + ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n"); + ast_mutex_lock(&i->lock); + return NULL; + } + ast_mutex_lock(&i->lock); + + if (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INFO) + tmp->tech = &sip_tech_info; + else + tmp->tech = &sip_tech; + + /* Select our native format based on codec preference until we receive + something from another device to the contrary. */ + if (i->jointcapability) { /* The joint capabilities of us and peer */ + what = i->jointcapability; + video = i->jointcapability & AST_FORMAT_VIDEO_MASK; + } else if (i->capability) { /* Our configured capability for this peer */ + what = i->capability; + video = i->capability & AST_FORMAT_VIDEO_MASK; + } else { + what = global_capability; /* Global codec support */ + video = global_capability & AST_FORMAT_VIDEO_MASK; + } + + /* Set the native formats for audio and merge in video */ + tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video; + if (option_debug > 2) { + char buf[SIPBUFSIZE]; + ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmp->nativeformats)); + ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability)); + ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability)); + ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_codec_choose(&i->prefs, what, 1))); + if (i->prefcodec) + ast_log(LOG_DEBUG, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcodec)); + } + + /* XXX Why are we choosing a codec from the native formats?? */ + fmt = ast_best_codec(tmp->nativeformats); + + /* If we have a prefcodec setting, we have an inbound channel that set a + preferred format for this call. Otherwise, we check the jointcapability + We also check for vrtp. If it's not there, we are not allowed do any video anyway. + */ + if (i->vrtp) { + if (i->prefcodec) + needvideo = i->prefcodec & AST_FORMAT_VIDEO_MASK; /* Outbound call */ + else + needvideo = i->jointcapability & AST_FORMAT_VIDEO_MASK; /* Inbound call */ + } + + if (option_debug > 2) { + if (needvideo) + ast_log(LOG_DEBUG, "This channel can handle video! HOLLYWOOD next!\n"); + else + ast_log(LOG_DEBUG, "This channel will not be able to handle video.\n"); + } + + + + if (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) { + i->vad = ast_dsp_new(); + ast_dsp_set_features(i->vad, DSP_FEATURE_DTMF_DETECT); + if (global_relaxdtmf) + ast_dsp_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF); + } + if (i->rtp) { + tmp->fds[0] = ast_rtp_fd(i->rtp); + tmp->fds[1] = ast_rtcp_fd(i->rtp); + } + if (needvideo && i->vrtp) { + tmp->fds[2] = ast_rtp_fd(i->vrtp); + tmp->fds[3] = ast_rtcp_fd(i->vrtp); + } + if (i->udptl) { + tmp->fds[5] = ast_udptl_fd(i->udptl); + } + if (state == AST_STATE_RING) + tmp->rings = 1; + tmp->adsicpe = AST_ADSI_UNAVAILABLE; + tmp->writeformat = fmt; + tmp->rawwriteformat = fmt; + tmp->readformat = fmt; + tmp->rawreadformat = fmt; + tmp->tech_pvt = i; + + tmp->callgroup = i->callgroup; + tmp->pickupgroup = i->pickupgroup; + tmp->cid.cid_pres = i->callingpres; + if (!ast_strlen_zero(i->accountcode)) + ast_string_field_set(tmp, accountcode, i->accountcode); + if (i->amaflags) + tmp->amaflags = i->amaflags; + if (!ast_strlen_zero(i->language)) + ast_string_field_set(tmp, language, i->language); + i->owner = tmp; + ast_module_ref(ast_module_info->self); + ast_copy_string(tmp->context, i->context, sizeof(tmp->context)); + /*Since it is valid to have extensions in the dialplan that have unescaped characters in them + * we should decode the uri before storing it in the channel, but leave it encoded in the sip_pvt + * structure so that there aren't issues when forming URI's + */ + decoded_exten = ast_strdupa(i->exten); + ast_uri_decode(decoded_exten); + ast_copy_string(tmp->exten, decoded_exten, sizeof(tmp->exten)); + + /* Don't use ast_set_callerid() here because it will + * generate an unnecessary NewCallerID event */ + tmp->cid.cid_ani = ast_strdup(i->cid_num); + if (!ast_strlen_zero(i->rdnis)) + tmp->cid.cid_rdnis = ast_strdup(i->rdnis); + + if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s")) + tmp->cid.cid_dnid = ast_strdup(i->exten); + + tmp->priority = 1; + if (!ast_strlen_zero(i->uri)) + pbx_builtin_setvar_helper(tmp, "SIPURI", i->uri); + if (!ast_strlen_zero(i->domain)) + pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain); + if (!ast_strlen_zero(i->useragent)) + pbx_builtin_setvar_helper(tmp, "SIPUSERAGENT", i->useragent); + if (!ast_strlen_zero(i->callid)) + pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid); + if (i->rtp) + ast_jb_configure(tmp, &global_jbconf); + + /* If the INVITE contains T.38 SDP information set the proper channel variable so a created outgoing call will also have T.38 */ + if (i->udptl && i->t38.state == T38_PEER_DIRECT) + pbx_builtin_setvar_helper(tmp, "_T38CALL", "1"); + + /* Set channel variables for this call from configuration */ + for (v = i->chanvars ; v ; v = v->next) + pbx_builtin_setvar_helper(tmp, v->name, v->value); + + if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) { + ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name); + tmp->hangupcause = AST_CAUSE_SWITCH_CONGESTION; + ast_hangup(tmp); + tmp = NULL; + } + + if (!ast_test_flag(&i->flags[0], SIP_NO_HISTORY)) + append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid); + + return tmp; +} + +/*! \brief Reads one line of SIP message body */ +static char *get_body_by_line(const char *line, const char *name, int nameLen) +{ + if (strncasecmp(line, name, nameLen) == 0 && line[nameLen] == '=') + return ast_skip_blanks(line + nameLen + 1); + + return ""; +} + +/*! \brief Lookup 'name' in the SDP starting + * at the 'start' line. Returns the matching line, and 'start' + * is updated with the next line number. + */ +static const char *get_sdp_iterate(int *start, struct sip_request *req, const char *name) +{ + int len = strlen(name); + + while (*start < req->sdp_end) { + const char *r = get_body_by_line(req->line[(*start)++], name, len); + if (r[0] != '\0') + return r; + } + + return ""; +} + +/*! \brief Get a line from an SDP message body */ +static const char *get_sdp(struct sip_request *req, const char *name) +{ + int dummy = 0; + + return get_sdp_iterate(&dummy, req, name); +} + +/*! \brief Get a specific line from the message body */ +static char *get_body(struct sip_request *req, char *name) +{ + int x; + int len = strlen(name); + char *r; + + for (x = 0; x < req->lines; x++) { + r = get_body_by_line(req->line[x], name, len); + if (r[0] != '\0') + return r; + } + + return ""; +} + +/*! \brief Find compressed SIP alias */ +static const char *find_alias(const char *name, const char *_default) +{ + /*! \brief Structure for conversion between compressed SIP and "normal" SIP */ + static const struct cfalias { + char * const fullname; + char * const shortname; + } aliases[] = { + { "Content-Type", "c" }, + { "Content-Encoding", "e" }, + { "From", "f" }, + { "Call-ID", "i" }, + { "Contact", "m" }, + { "Content-Length", "l" }, + { "Subject", "s" }, + { "To", "t" }, + { "Supported", "k" }, + { "Refer-To", "r" }, + { "Referred-By", "b" }, + { "Allow-Events", "u" }, + { "Event", "o" }, + { "Via", "v" }, + { "Accept-Contact", "a" }, + { "Reject-Contact", "j" }, + { "Request-Disposition", "d" }, + { "Session-Expires", "x" }, + { "Identity", "y" }, + { "Identity-Info", "n" }, + }; + int x; + + for (x=0; x<sizeof(aliases) / sizeof(aliases[0]); x++) + if (!strcasecmp(aliases[x].fullname, name)) + return aliases[x].shortname; + + return _default; +} + +static const char *__get_header(const struct sip_request *req, const char *name, int *start) +{ + int pass; + + /* + * Technically you can place arbitrary whitespace both before and after the ':' in + * a header, although RFC3261 clearly says you shouldn't before, and place just + * one afterwards. If you shouldn't do it, what absolute idiot decided it was + * a good idea to say you can do it, and if you can do it, why in the hell would. + * you say you shouldn't. + * Anyways, pedanticsipchecking controls whether we allow spaces before ':', + * and we always allow spaces after that for compatibility. + */ + for (pass = 0; name && pass < 2;pass++) { + int x, len = strlen(name); + for (x=*start; x<req->headers; x++) { + if (!strncasecmp(req->header[x], name, len)) { + char *r = req->header[x] + len; /* skip name */ + if (pedanticsipchecking) + r = ast_skip_blanks(r); + + if (*r == ':') { + *start = x+1; + return ast_skip_blanks(r+1); + } + } + } + if (pass == 0) /* Try aliases */ + name = find_alias(name, NULL); + } + + /* Don't return NULL, so get_header is always a valid pointer */ + return ""; +} + +/*! \brief Get header from SIP request */ +static const char *get_header(const struct sip_request *req, const char *name) +{ + int start = 0; + return __get_header(req, name, &start); +} + +/*! \brief Read RTP from network */ +static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect) +{ + /* Retrieve audio/etc from channel. Assumes p->lock is already held. */ + struct ast_frame *f; + + if (!p->rtp) { + /* We have no RTP allocated for this channel */ + return &ast_null_frame; + } + + switch(ast->fdno) { + case 0: + f = ast_rtp_read(p->rtp); /* RTP Audio */ + break; + case 1: + f = ast_rtcp_read(p->rtp); /* RTCP Control Channel */ + break; + case 2: + f = ast_rtp_read(p->vrtp); /* RTP Video */ + break; + case 3: + f = ast_rtcp_read(p->vrtp); /* RTCP Control Channel for video */ + break; + case 5: + f = ast_udptl_read(p->udptl); /* UDPTL for T.38 */ + break; + default: + f = &ast_null_frame; + } + /* Don't forward RFC2833 if we're not supposed to */ + if (f && (f->frametype == AST_FRAME_DTMF) && + (ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833)) + return &ast_null_frame; + + /* We already hold the channel lock */ + if (!p->owner || (f && f->frametype != AST_FRAME_VOICE)) + return f; + + if (f && f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) { + if (!(f->subclass & p->jointcapability)) { + if (option_debug) { + ast_log(LOG_DEBUG, "Bogus frame of format '%s' received from '%s'!\n", + ast_getformatname(f->subclass), p->owner->name); + } + return &ast_null_frame; + } + if (option_debug) + ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass); + p->owner->nativeformats = (p->owner->nativeformats & AST_FORMAT_VIDEO_MASK) | f->subclass; + ast_set_read_format(p->owner, p->owner->readformat); + ast_set_write_format(p->owner, p->owner->writeformat); + } + + if (f && (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) { + f = ast_dsp_process(p->owner, p->vad, f); + if (f && f->frametype == AST_FRAME_DTMF) { + if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && f->subclass == 'f') { + if (option_debug) + ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name); + *faxdetect = 1; + } else if (option_debug) { + ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass); + } + } + } + + return f; +} + +/*! \brief Read SIP RTP from channel */ +static struct ast_frame *sip_read(struct ast_channel *ast) +{ + struct ast_frame *fr; + struct sip_pvt *p = ast->tech_pvt; + int faxdetected = FALSE; + + ast_mutex_lock(&p->lock); + fr = sip_rtp_read(ast, p, &faxdetected); + p->lastrtprx = time(NULL); + + /* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */ + /* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */ + if (faxdetected && ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) { + if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) { + if (!p->pendinginvite) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name); + p->t38.state = T38_LOCAL_REINVITE; + transmit_reinvite_with_t38_sdp(p); + if (option_debug > 1) + ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name); + } + } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name); + ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); + } + } + + /* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */ + if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) { + fr = &ast_null_frame; + } + + ast_mutex_unlock(&p->lock); + return fr; +} + + +/*! \brief Generate 32 byte random string for callid's etc */ +static char *generate_random_string(char *buf, size_t size) +{ + long val[4]; + int x; + + for (x=0; x<4; x++) + val[x] = ast_random(); + snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]); + + return buf; +} + +/*! \brief Build SIP Call-ID value for a non-REGISTER transaction */ +static void build_callid_pvt(struct sip_pvt *pvt) +{ + char buf[33]; + + const char *host = S_OR(pvt->fromdomain, ast_inet_ntoa(pvt->ourip)); + + ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host); + +} + +/*! \brief Build SIP Call-ID value for a REGISTER transaction */ +static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain) +{ + char buf[33]; + + const char *host = S_OR(fromdomain, ast_inet_ntoa(ourip)); + + ast_string_field_build(reg, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host); +} + +/*! \brief Make our SIP dialog tag */ +static void make_our_tag(char *tagbuf, size_t len) +{ + snprintf(tagbuf, len, "as%08lx", ast_random()); +} + +/*! \brief Allocate SIP_PVT structure and set defaults */ +static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin, + int useglobal_nat, const int intended_method) +{ + struct sip_pvt *p; + + if (!(p = ast_calloc(1, sizeof(*p)))) + return NULL; + + if (ast_string_field_init(p, 512)) { + free(p); + return NULL; + } + + ast_mutex_init(&p->lock); + + p->method = intended_method; + p->initid = -1; + p->waitid = -1; + p->autokillid = -1; + p->request_queue_sched_id = -1; + p->subscribed = NONE; + p->stateid = -1; + p->prefs = default_prefs; /* Set default codecs for this call */ + + if (intended_method != SIP_OPTIONS) /* Peerpoke has it's own system */ + p->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */ + + if (sin) { + p->sa = *sin; + if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip)) + p->ourip = __ourip; + } else + p->ourip = __ourip; + + /* Copy global flags to this PVT at setup. */ + ast_copy_flags(&p->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY); + ast_copy_flags(&p->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY); + + ast_set2_flag(&p->flags[0], !recordhistory, SIP_NO_HISTORY); + + p->branch = ast_random(); + make_our_tag(p->tag, sizeof(p->tag)); + p->ocseq = INITIAL_CSEQ; + + if (sip_methods[intended_method].need_rtp) { + p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr); + /* If the global videosupport flag is on, we always create a RTP interface for video */ + if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT)) + p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr); + if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT)) + p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr); + if (!p->rtp || (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)) { + ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n", + ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video" : "", strerror(errno)); + ast_mutex_destroy(&p->lock); + if (p->chanvars) { + ast_variables_destroy(p->chanvars); + p->chanvars = NULL; + } + free(p); + return NULL; + } + ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833); + ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE)); + ast_rtp_settos(p->rtp, global_tos_audio); + ast_rtp_set_rtptimeout(p->rtp, global_rtptimeout); + ast_rtp_set_rtpholdtimeout(p->rtp, global_rtpholdtimeout); + ast_rtp_set_rtpkeepalive(p->rtp, global_rtpkeepalive); + if (p->vrtp) { + ast_rtp_settos(p->vrtp, global_tos_video); + ast_rtp_setdtmf(p->vrtp, 0); + ast_rtp_setdtmfcompensate(p->vrtp, 0); + ast_rtp_set_rtptimeout(p->vrtp, global_rtptimeout); + ast_rtp_set_rtpholdtimeout(p->vrtp, global_rtpholdtimeout); + ast_rtp_set_rtpkeepalive(p->vrtp, global_rtpkeepalive); + } + if (p->udptl) + ast_udptl_settos(p->udptl, global_tos_audio); + p->maxcallbitrate = default_maxcallbitrate; + p->autoframing = global_autoframing; + ast_rtp_codec_setpref(p->rtp, &p->prefs); + } + + if (useglobal_nat && sin) { + /* Setup NAT structure according to global settings if we have an address */ + ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT); + p->recv = *sin; + do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE); + } + + if (p->method != SIP_REGISTER) + ast_string_field_set(p, fromdomain, default_fromdomain); + build_via(p); + if (!callid) + build_callid_pvt(p); + else + ast_string_field_set(p, callid, callid); + /* Assign default music on hold class */ + ast_string_field_set(p, mohinterpret, default_mohinterpret); + ast_string_field_set(p, mohsuggest, default_mohsuggest); + p->capability = global_capability; + p->allowtransfer = global_allowtransfer; + if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) || + (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) + p->noncodeccapability |= AST_RTP_DTMF; + if (p->udptl) { + p->t38.capability = global_t38_capability; + if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY) + p->t38.capability |= T38FAX_UDP_EC_REDUNDANCY; + else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_FEC) + p->t38.capability |= T38FAX_UDP_EC_FEC; + else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_NONE) + p->t38.capability |= T38FAX_UDP_EC_NONE; + p->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF; + p->t38.jointcapability = p->t38.capability; + } + ast_string_field_set(p, context, default_context); + + AST_LIST_HEAD_INIT_NOLOCK(&p->request_queue); + + /* Add to active dialog list */ + ast_mutex_lock(&iflock); + p->next = iflist; + iflist = p; + ast_mutex_unlock(&iflock); + if (option_debug) + ast_log(LOG_DEBUG, "Allocating new SIP dialog for %s - %s (%s)\n", callid ? callid : "(No Call-ID)", sip_methods[intended_method].text, p->rtp ? "With RTP" : "No RTP"); + return p; +} + +/*! \brief Connect incoming SIP message to current dialog or create new dialog structure + Called by handle_request, sipsock_read */ +static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method) +{ + struct sip_pvt *p = NULL; + char *tag = ""; /* note, tag is never NULL */ + char totag[128]; + char fromtag[128]; + const char *callid = get_header(req, "Call-ID"); + const char *from = get_header(req, "From"); + const char *to = get_header(req, "To"); + const char *cseq = get_header(req, "Cseq"); + + /* Call-ID, to, from and Cseq are required by RFC 3261. (Max-forwards and via too - ignored now) */ + /* get_header always returns non-NULL so we must use ast_strlen_zero() */ + if (ast_strlen_zero(callid) || ast_strlen_zero(to) || + ast_strlen_zero(from) || ast_strlen_zero(cseq)) + return NULL; /* Invalid packet */ + + if (pedanticsipchecking) { + /* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy + we need more to identify a branch - so we have to check branch, from + and to tags to identify a call leg. + For Asterisk to behave correctly, you need to turn on pedanticsipchecking + in sip.conf + */ + if (gettag(req, "To", totag, sizeof(totag))) + ast_set_flag(req, SIP_PKT_WITH_TOTAG); /* Used in handle_request/response */ + gettag(req, "From", fromtag, sizeof(fromtag)); + + tag = (req->method == SIP_RESPONSE) ? totag : fromtag; + + if (option_debug > 4 ) + ast_log(LOG_DEBUG, "= Looking for Call ID: %s (Checking %s) --From tag %s --To-tag %s \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag); + } + + ast_mutex_lock(&iflock); + for (p = iflist; p; p = p->next) { + /* In pedantic, we do not want packets with bad syntax to be connected to a PVT */ + int found = FALSE; + if (ast_strlen_zero(p->callid)) + continue; + if (req->method == SIP_REGISTER) + found = (!strcmp(p->callid, callid)); + else { + found = !strcmp(p->callid, callid); + if (pedanticsipchecking && found) { + found = ast_strlen_zero(tag) || ast_strlen_zero(p->theirtag) || !ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED) || !strcmp(p->theirtag, tag); + } + } + + if (option_debug > 4) + ast_log(LOG_DEBUG, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag); + + /* If we get a new request within an existing to-tag - check the to tag as well */ + if (pedanticsipchecking && found && req->method != SIP_RESPONSE) { /* SIP Request */ + if (p->tag[0] == '\0' && totag[0]) { + /* We have no to tag, but they have. Wrong dialog */ + found = FALSE; + } else if (totag[0]) { /* Both have tags, compare them */ + if (strcmp(totag, p->tag)) { + found = FALSE; /* This is not our packet */ + } + } + if (!found && option_debug > 4) + ast_log(LOG_DEBUG, "= Being pedantic: This is not our match on request: Call ID: %s Ourtag <null> Totag %s Method %s\n", p->callid, totag, sip_methods[req->method].text); + } + if (found) { + /* Found the call */ + ast_mutex_lock(&p->lock); + ast_mutex_unlock(&iflock); + return p; + } + } + ast_mutex_unlock(&iflock); + + /* See if the method is capable of creating a dialog */ + if (sip_methods[intended_method].can_create == CAN_CREATE_DIALOG) { + if (intended_method == SIP_REFER) { + /* We do support REFER, but not outside of a dialog yet */ + transmit_response_using_temp(callid, sin, 1, intended_method, req, "603 Declined (no dialog)"); + } else if (intended_method == SIP_NOTIFY) { + /* We do not support out-of-dialog NOTIFY either, + like voicemail notification, so cancel that early */ + transmit_response_using_temp(callid, sin, 1, intended_method, req, "489 Bad event"); + } else { + /* Ok, time to create a new SIP dialog object, a pvt */ + if ((p = sip_alloc(callid, sin, 1, intended_method))) { + /* Ok, we've created a dialog, let's go and process it */ + ast_mutex_lock(&p->lock); + } else { + /* We have a memory or file/socket error (can't allocate RTP sockets or something) so we're not + getting a dialog from sip_alloc. + + Without a dialog we can't retransmit and handle ACKs and all that, but at least + send an error message. + + Sorry, we apologize for the inconvienience + */ + transmit_response_using_temp(callid, sin, 1, intended_method, req, "500 Server internal error"); + if (option_debug > 3) + ast_log(LOG_DEBUG, "Failed allocating SIP dialog, sending 500 Server internal error and giving up\n"); + } + } + return p; + } else if( sip_methods[intended_method].can_create == CAN_CREATE_DIALOG_UNSUPPORTED_METHOD) { + /* A method we do not support, let's take it on the volley */ + transmit_response_using_temp(callid, sin, 1, intended_method, req, "501 Method Not Implemented"); + } else if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) { + /* This is a request outside of a dialog that we don't know about + ...never reply to an ACK! + */ + transmit_response_using_temp(callid, sin, 1, intended_method, req, "481 Call leg/transaction does not exist"); + } + /* We do not respond to responses for dialogs that we don't know about, we just drop + the session quickly */ + + return p; +} + +/*! \brief Parse register=> line in sip.conf and add to registry */ +static int sip_register(char *value, int lineno) +{ + struct sip_registry *reg; + int portnum = 0; + char username[256] = ""; + char *hostname=NULL, *secret=NULL, *authuser=NULL; + char *porta=NULL; + char *contact=NULL; + + if (!value) + return -1; + ast_copy_string(username, value, sizeof(username)); + /* First split around the last '@' then parse the two components. */ + hostname = strrchr(username, '@'); /* allow @ in the first part */ + if (hostname) + *hostname++ = '\0'; + if (ast_strlen_zero(username) || ast_strlen_zero(hostname)) { + ast_log(LOG_WARNING, "Format for registration is user[:secret[:authuser]]@host[:port][/contact] at line %d\n", lineno); + return -1; + } + /* split user[:secret[:authuser]] */ + secret = strchr(username, ':'); + if (secret) { + *secret++ = '\0'; + authuser = strchr(secret, ':'); + if (authuser) + *authuser++ = '\0'; + } + /* split host[:port][/contact] */ + contact = strchr(hostname, '/'); + if (contact) + *contact++ = '\0'; + if (ast_strlen_zero(contact)) + contact = "s"; + porta = strchr(hostname, ':'); + if (porta) { + *porta++ = '\0'; + portnum = atoi(porta); + if (portnum == 0) { + ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno); + return -1; + } + } + if (!(reg = ast_calloc(1, sizeof(*reg)))) { + ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n"); + return -1; + } + + if (ast_string_field_init(reg, 256)) { + ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry strings\n"); + free(reg); + return -1; + } + + regobjs++; + ASTOBJ_INIT(reg); + ast_string_field_set(reg, contact, contact); + if (!ast_strlen_zero(username)) + ast_string_field_set(reg, username, username); + if (hostname) + ast_string_field_set(reg, hostname, hostname); + if (authuser) + ast_string_field_set(reg, authuser, authuser); + if (secret) + ast_string_field_set(reg, secret, secret); + reg->expire = -1; + reg->timeout = -1; + reg->refresh = default_expiry; + reg->portno = portnum; + reg->callid_valid = FALSE; + reg->ocseq = INITIAL_CSEQ; + ASTOBJ_CONTAINER_LINK(®l, reg); /* Add the new registry entry to the list */ + ASTOBJ_UNREF(reg,sip_registry_destroy); + return 0; +} + +/*! \brief Parse multiline SIP headers into one header + This is enabled if pedanticsipchecking is enabled */ +static int lws2sws(char *msgbuf, int len) +{ + int h = 0, t = 0; + int lws = 0; + + for (; h < len;) { + /* Eliminate all CRs */ + if (msgbuf[h] == '\r') { + h++; + continue; + } + /* Check for end-of-line */ + if (msgbuf[h] == '\n') { + /* Check for end-of-message */ + if (h + 1 == len) + break; + /* Check for a continuation line */ + if (msgbuf[h + 1] == ' ' || msgbuf[h + 1] == '\t') { + /* Merge continuation line */ + h++; + continue; + } + /* Propagate LF and start new line */ + msgbuf[t++] = msgbuf[h++]; + lws = 0; + continue; + } + if (msgbuf[h] == ' ' || msgbuf[h] == '\t') { + if (lws) { + h++; + continue; + } + msgbuf[t++] = msgbuf[h++]; + lws = 1; + continue; + } + msgbuf[t++] = msgbuf[h++]; + if (lws) + lws = 0; + } + msgbuf[t] = '\0'; + return t; +} + +/*! \brief Parse a SIP message + \note this function is used both on incoming and outgoing packets +*/ +static int parse_request(struct sip_request *req) +{ + /* Divide fields by NULL's */ + char *c; + int f = 0; + + c = req->data; + + /* First header starts immediately */ + req->header[f] = c; + while(*c) { + if (*c == '\n') { + /* We've got a new header */ + *c = 0; + + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG, "Header %d: %s (%d)\n", f, req->header[f], (int) strlen(req->header[f])); + if (ast_strlen_zero(req->header[f])) { + /* Line by itself means we're now in content */ + c++; + break; + } + if (f >= SIP_MAX_HEADERS - 1) { + ast_log(LOG_WARNING, "Too many SIP headers. Ignoring.\n"); + } else { + f++; + req->header[f] = c + 1; + } + } else if (*c == '\r') { + /* Ignore but eliminate \r's */ + *c = 0; + } + c++; + } + + req->headers = f; + + /* Check a non-newline-terminated last header */ + if (!ast_strlen_zero(req->header[f])) { + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG, "Header %d: %s (%d)\n", f, req->header[f], (int) strlen(req->header[f])); + req->headers++; + } + + /* Now we process any body content */ + f = 0; + req->line[f] = c; + while (*c) { + if (*c == '\n') { + /* We've got a new line */ + *c = 0; + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG, "Line: %s (%d)\n", req->line[f], (int) strlen(req->line[f])); + if (f == SIP_MAX_LINES - 1) { + ast_log(LOG_WARNING, "Too many SDP lines. Ignoring.\n"); + break; + } else { + f++; + req->line[f] = c + 1; + } + } else if (*c == '\r') { + /* Ignore and eliminate \r's */ + *c = 0; + } + c++; + } + + req->lines = f; + + /* Check a non-newline-terminated last line */ + if (!ast_strlen_zero(req->line[f])) { + req->lines++; + } + + if (*c) + ast_log(LOG_WARNING, "Odd content, extra stuff left over ('%s')\n", c); + + /* Split up the first line parts */ + return determine_firstline_parts(req); +} + +/*! + \brief Determine whether a SIP message contains an SDP in its body + \param req the SIP request to process + \return 1 if SDP found, 0 if not found + + Also updates req->sdp_start and req->sdp_end to indicate where the SDP + lives in the message body. +*/ +static int find_sdp(struct sip_request *req) +{ + const char *content_type; + const char *content_length; + const char *search; + char *boundary; + unsigned int x; + int boundaryisquoted = FALSE; + int found_application_sdp = FALSE; + int found_end_of_headers = FALSE; + + content_length = get_header(req, "Content-Length"); + + if (!ast_strlen_zero(content_length)) { + if (sscanf(content_length, "%ud", &x) != 1) { + ast_log(LOG_WARNING, "Invalid Content-Length: %s\n", content_length); + return 0; + } + + /* Content-Length of zero means there can't possibly be an + SDP here, even if the Content-Type says there is */ + if (x == 0) + return 0; + } + + content_type = get_header(req, "Content-Type"); + + /* if the body contains only SDP, this is easy */ + if (!strncasecmp(content_type, "application/sdp", 15)) { + req->sdp_start = 0; + req->sdp_end = req->lines; + return req->lines ? 1 : 0; + } + + /* if it's not multipart/mixed, there cannot be an SDP */ + if (strncasecmp(content_type, "multipart/mixed", 15)) + return 0; + + /* if there is no boundary marker, it's invalid */ + if ((search = strcasestr(content_type, ";boundary="))) + search += 10; + else if ((search = strcasestr(content_type, "; boundary="))) + search += 11; + else + return 0; + + if (ast_strlen_zero(search)) + return 0; + + /* If the boundary is quoted with ", remove quote */ + if (*search == '\"') { + search++; + boundaryisquoted = TRUE; + } + + /* make a duplicate of the string, with two extra characters + at the beginning */ + boundary = ast_strdupa(search - 2); + boundary[0] = boundary[1] = '-'; + /* Remove final quote */ + if (boundaryisquoted) + boundary[strlen(boundary) - 1] = '\0'; + + /* search for the boundary marker, the empty line delimiting headers from + sdp part and the end boundry if it exists */ + + for (x = 0; x < (req->lines ); x++) { + if(!strncasecmp(req->line[x], boundary, strlen(boundary))){ + if(found_application_sdp && found_end_of_headers){ + req->sdp_end = x-1; + return 1; + } + found_application_sdp = FALSE; + } + if(!strcasecmp(req->line[x], "Content-Type: application/sdp")) + found_application_sdp = TRUE; + + if(strlen(req->line[x]) == 0 ){ + if(found_application_sdp && !found_end_of_headers){ + req->sdp_start = x; + found_end_of_headers = TRUE; + } + } + } + if(found_application_sdp && found_end_of_headers) { + req->sdp_end = x; + return TRUE; + } + return FALSE; +} + +/*! \brief Change hold state for a call */ +static void change_hold_state(struct sip_pvt *dialog, struct sip_request *req, int holdstate, int sendonly) +{ + if (global_notifyhold && (!holdstate || !ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD))) + sip_peer_hold(dialog, holdstate); + if (global_callevents) + manager_event(EVENT_FLAG_CALL, holdstate ? "Hold" : "Unhold", + "Channel: %s\r\n" + "Uniqueid: %s\r\n", + dialog->owner->name, + dialog->owner->uniqueid); + append_history(dialog, holdstate ? "Hold" : "Unhold", "%s", req->data); + if (!holdstate) { /* Put off remote hold */ + ast_clear_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD); /* Clear both flags */ + return; + } + /* No address for RTP, we're on hold */ + + if (sendonly == 1) /* One directional hold (sendonly/recvonly) */ + ast_set_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD_ONEDIR); + else if (sendonly == 2) /* Inactive stream */ + ast_set_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD_INACTIVE); + else + ast_set_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD_ACTIVE); + return; +} + +/*! \brief Process SIP SDP offer, select formats and activate RTP channels + If offer is rejected, we will not change any properties of the call + Return 0 on success, a negative value on errors. + Must be called after find_sdp(). +*/ +static int process_sdp(struct sip_pvt *p, struct sip_request *req) +{ + const char *m; /* SDP media offer */ + const char *c; + const char *a; + char host[258]; + int len = -1; + int portno = -1; /*!< RTP Audio port number */ + int vportno = -1; /*!< RTP Video port number */ + int udptlportno = -1; + int peert38capability = 0; + char s[256]; + int old = 0; + + /* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */ + int peercapability = 0, peernoncodeccapability = 0; + int vpeercapability = 0, vpeernoncodeccapability = 0; + struct sockaddr_in sin; /*!< media socket address */ + struct sockaddr_in vsin; /*!< Video socket address */ + + const char *codecs; + struct hostent *hp; /*!< RTP Audio host IP */ + struct hostent *vhp = NULL; /*!< RTP video host IP */ + struct ast_hostent audiohp; + struct ast_hostent videohp; + int codec; + int destiterator = 0; + int iterator; + int sendonly = -1; + int numberofports; + struct ast_rtp *newaudiortp, *newvideortp; /* Buffers for codec handling */ + int newjointcapability; /* Negotiated capability */ + int newpeercapability; + int newnoncodeccapability; + int numberofmediastreams = 0; + int debug = sip_debug_test_pvt(p); + + int found_rtpmap_codecs[SDP_MAX_RTPMAP_CODECS]; + int last_rtpmap_codec=0; + + if (!p->rtp) { + ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n"); + return -1; + } + + /* Initialize the temporary RTP structures we use to evaluate the offer from the peer */ +#ifdef LOW_MEMORY + newaudiortp = ast_threadstorage_get(&ts_audio_rtp, ast_rtp_alloc_size()); +#else + newaudiortp = alloca(ast_rtp_alloc_size()); +#endif + memset(newaudiortp, 0, ast_rtp_alloc_size()); + ast_rtp_new_init(newaudiortp); + ast_rtp_pt_clear(newaudiortp); + +#ifdef LOW_MEMORY + newvideortp = ast_threadstorage_get(&ts_video_rtp, ast_rtp_alloc_size()); +#else + newvideortp = alloca(ast_rtp_alloc_size()); +#endif + memset(newvideortp, 0, ast_rtp_alloc_size()); + ast_rtp_new_init(newvideortp); + ast_rtp_pt_clear(newvideortp); + + /* Update our last rtprx when we receive an SDP, too */ + p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */ + + + /* Try to find first media stream */ + m = get_sdp(req, "m"); + destiterator = req->sdp_start; + c = get_sdp_iterate(&destiterator, req, "c"); + if (ast_strlen_zero(m) || ast_strlen_zero(c)) { + ast_log(LOG_WARNING, "Insufficient information for SDP (m = '%s', c = '%s')\n", m, c); + return -1; + } + + /* Check for IPv4 address (not IPv6 yet) */ + if (sscanf(c, "IN IP4 %256s", host) != 1) { + ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c); + return -1; + } + + /* XXX This could block for a long time, and block the main thread! XXX */ + hp = ast_gethostbyname(host, &audiohp); + if (!hp) { + ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c); + return -1; + } + vhp = hp; /* Copy to video address as default too */ + + iterator = req->sdp_start; + ast_set_flag(&p->flags[0], SIP_NOVIDEO); + + + /* Find media streams in this SDP offer */ + while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') { + int x; + int audio = FALSE; + + numberofports = 1; + len = -1; + if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) || + (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1 && len > 0)) { + audio = TRUE; + numberofmediastreams++; + /* Found audio stream in this media definition */ + portno = x; + /* Scan through the RTP payload types specified in a "m=" line: */ + for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) { + if (sscanf(codecs, "%d%n", &codec, &len) != 1) { + ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs); + return -1; + } + if (debug) + ast_verbose("Found RTP audio format %d\n", codec); + ast_rtp_set_m_type(newaudiortp, codec); + } + } else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) || + (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1 && len >= 0)) { + /* If it is not audio - is it video ? */ + ast_clear_flag(&p->flags[0], SIP_NOVIDEO); + numberofmediastreams++; + vportno = x; + /* Scan through the RTP payload types specified in a "m=" line: */ + for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) { + if (sscanf(codecs, "%d%n", &codec, &len) != 1) { + ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs); + return -1; + } + if (debug) + ast_verbose("Found RTP video format %d\n", codec); + ast_rtp_set_m_type(newvideortp, codec); + } + } else if (p->udptl && ( (sscanf(m, "image %d udptl t38%n", &x, &len) == 1 && len > 0) || + (sscanf(m, "image %d UDPTL t38%n", &x, &len) == 1 && len >= 0) )) { + if (debug) + ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid); + udptlportno = x; + numberofmediastreams++; + + if (p->owner && p->lastinvite) { + p->t38.state = T38_PEER_REINVITE; /* T38 Offered in re-invite from remote party */ + if (option_debug > 1) + ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>" ); + } else { + p->t38.state = T38_PEER_DIRECT; /* T38 Offered directly from peer in first invite */ + if (option_debug > 1) + ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + } + } else + ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m); + if (numberofports > 1) + ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports); + + + /* Check for Media-description-level-address for audio */ + c = get_sdp_iterate(&destiterator, req, "c"); + if (!ast_strlen_zero(c)) { + if (sscanf(c, "IN IP4 %256s", host) != 1) { + ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c); + } else { + /* XXX This could block for a long time, and block the main thread! XXX */ + if (audio) { + if ( !(hp = ast_gethostbyname(host, &audiohp))) { + ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in secondary c= line, '%s'\n", c); + return -2; + } + } else if (!(vhp = ast_gethostbyname(host, &videohp))) { + ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c); + return -2; + } + } + + } + } + if (portno == -1 && vportno == -1 && udptlportno == -1) + /* No acceptable offer found in SDP - we have no ports */ + /* Do not change RTP or VRTP if this is a re-invite */ + return -2; + + if (numberofmediastreams > 2) + /* We have too many fax, audio and/or video media streams, fail this offer */ + return -3; + + /* RTP addresses and ports for audio and video */ + sin.sin_family = AF_INET; + vsin.sin_family = AF_INET; + memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr)); + if (vhp) + memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr)); + + /* Setup UDPTL port number */ + if (p->udptl) { + if (udptlportno > 0) { + sin.sin_port = htons(udptlportno); + if (ast_test_flag(&p->flags[0], SIP_NAT) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) { + struct sockaddr_in peer; + ast_rtp_get_peer(p->rtp, &peer); + if (peer.sin_addr.s_addr) { + memcpy(&sin.sin_addr, &peer.sin_addr, sizeof(sin.sin_addr)); + if (debug) { + ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(sin.sin_addr)); + } + } + } + ast_udptl_set_peer(p->udptl, &sin); + if (debug) + ast_log(LOG_DEBUG,"Peer T.38 UDPTL is at port %s:%d\n",ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); + } else { + ast_udptl_stop(p->udptl); + if (debug) + ast_log(LOG_DEBUG, "Peer doesn't provide T.38 UDPTL\n"); + } + } + + + if (p->rtp) { + if (portno > 0) { + sin.sin_port = htons(portno); + ast_rtp_set_peer(p->rtp, &sin); + if (debug) + ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); + } else { + if (udptlportno > 0) { + if (debug) + ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid %s\n", p->callid); + } else { + ast_rtp_stop(p->rtp); + if (debug) + ast_verbose("Peer doesn't provide audio. Callid %s\n", p->callid); + } + } + } + /* Setup video port number */ + if (vportno != -1) + vsin.sin_port = htons(vportno); + + /* Next, scan through each "a=rtpmap:" line, noting each + * specified RTP payload type (with corresponding MIME subtype): + */ + /* XXX This needs to be done per media stream, since it's media stream specific */ + iterator = req->sdp_start; + while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') { + char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */ + if (option_debug > 1) { + int breakout = FALSE; + + /* If we're debugging, check for unsupported sdp options */ + if (!strncasecmp(a, "rtcp:", (size_t) 5)) { + if (debug) + ast_verbose("Got unsupported a:rtcp in SDP offer \n"); + breakout = TRUE; + } else if (!strncasecmp(a, "fmtp:", (size_t) 5)) { + /* Format parameters: Not supported */ + /* Note: This is used for codec parameters, like bitrate for + G722 and video formats for H263 and H264 + See RFC2327 for an example */ + if (debug) + ast_verbose("Got unsupported a:fmtp in SDP offer \n"); + breakout = TRUE; + } else if (!strncasecmp(a, "framerate:", (size_t) 10)) { + /* Video stuff: Not supported */ + if (debug) + ast_verbose("Got unsupported a:framerate in SDP offer \n"); + breakout = TRUE; + } else if (!strncasecmp(a, "maxprate:", (size_t) 9)) { + /* Video stuff: Not supported */ + if (debug) + ast_verbose("Got unsupported a:maxprate in SDP offer \n"); + breakout = TRUE; + } else if (!strncasecmp(a, "crypto:", (size_t) 7)) { + /* SRTP stuff, not yet supported */ + if (debug) + ast_verbose("Got unsupported a:crypto in SDP offer \n"); + breakout = TRUE; + } + if (breakout) /* We have a match, skip to next header */ + continue; + } + if (!strcasecmp(a, "sendonly")) { + if (sendonly == -1) + sendonly = 1; + continue; + } else if (!strcasecmp(a, "inactive")) { + if (sendonly == -1) + sendonly = 2; + continue; + } else if (!strcasecmp(a, "sendrecv")) { + if (sendonly == -1) + sendonly = 0; + continue; + } else if (strlen(a) > 5 && !strncasecmp(a, "ptime", 5)) { + char *tmp = strrchr(a, ':'); + long int framing = 0; + if (tmp) { + tmp++; + framing = strtol(tmp, NULL, 10); + if (framing == LONG_MIN || framing == LONG_MAX) { + framing = 0; + if (option_debug) + ast_log(LOG_DEBUG, "Can't read framing from SDP: %s\n", a); + } + } + if (framing && p->autoframing) { + struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp); + int codec_n; + int format = 0; + for (codec_n = 0; codec_n < MAX_RTP_PT; codec_n++) { + format = ast_rtp_codec_getformat(codec_n); + if (!format) /* non-codec or not found */ + continue; + if (option_debug) + ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing); + ast_codec_pref_setsize(pref, format, framing); + } + ast_rtp_codec_setpref(p->rtp, pref); + } + continue; + } else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) == 2) { + /* We have a rtpmap to handle */ + int found = FALSE; + /* We should propably check if this is an audio or video codec + so we know where to look */ + + if (last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) { + /* Note: should really look at the 'freq' and '#chans' params too */ + if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype, + ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) { + if (debug) + ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec); + found_rtpmap_codecs[last_rtpmap_codec] = codec; + last_rtpmap_codec++; + found = TRUE; + + } else if (p->vrtp) { + if(ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0) != -1) { + if (debug) + ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec); + found_rtpmap_codecs[last_rtpmap_codec] = codec; + last_rtpmap_codec++; + found = TRUE; + } + } + } else { + if (debug) + ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec); + } + + if (!found) { + /* Remove this codec since it's an unknown media type for us */ + /* XXX This is buggy since the media line for audio and video can have the + same numbers. We need to check as described above, but for testing this works... */ + ast_rtp_unset_m_type(newaudiortp, codec); + ast_rtp_unset_m_type(newvideortp, codec); + if (debug) + ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec); + } + } + } + + if (udptlportno != -1) { + int found = 0, x; + + old = 0; + + /* Scan trough the a= lines for T38 attributes and set apropriate fileds */ + iterator = req->sdp_start; + while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') { + if ((sscanf(a, "T38FaxMaxBuffer:%d", &x) == 1)) { + found = 1; + if (option_debug > 2) + ast_log(LOG_DEBUG, "MaxBufferSize:%d\n",x); + } else if ((sscanf(a, "T38MaxBitRate:%d", &x) == 1) || (sscanf(a, "T38FaxMaxRate:%d", &x) == 1)) { + found = 1; + if (option_debug > 2) + ast_log(LOG_DEBUG,"T38MaxBitRate: %d\n",x); + switch (x) { + case 14400: + peert38capability |= T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400; + break; + case 12000: + peert38capability |= T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400; + break; + case 9600: + peert38capability |= T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400; + break; + case 7200: + peert38capability |= T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400; + break; + case 4800: + peert38capability |= T38FAX_RATE_4800 | T38FAX_RATE_2400; + break; + case 2400: + peert38capability |= T38FAX_RATE_2400; + break; + } + } else if ((sscanf(a, "T38FaxVersion:%d", &x) == 1)) { + found = 1; + if (option_debug > 2) + ast_log(LOG_DEBUG, "FaxVersion: %d\n",x); + if (x == 0) + peert38capability |= T38FAX_VERSION_0; + else if (x == 1) + peert38capability |= T38FAX_VERSION_1; + } else if ((sscanf(a, "T38FaxMaxDatagram:%d", &x) == 1) || (sscanf(a, "T38MaxDatagram:%d", &x) == 1)) { + found = 1; + if (option_debug > 2) + ast_log(LOG_DEBUG, "FaxMaxDatagram: %d\n",x); + ast_udptl_set_far_max_datagram(p->udptl, x); + ast_udptl_set_local_max_datagram(p->udptl, x); + } else if ((strncmp(a, "T38FaxFillBitRemoval", 20) == 0)) { + found = 1; + if ((sscanf(a, "T38FaxFillBitRemoval:%d", &x) == 1)) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "FillBitRemoval: %d\n",x); + if (x == 1) + peert38capability |= T38FAX_FILL_BIT_REMOVAL; + } else { + if (option_debug > 2) + ast_log(LOG_DEBUG, "FillBitRemoval\n"); + peert38capability |= T38FAX_FILL_BIT_REMOVAL; + } + } else if ((strncmp(a, "T38FaxTranscodingMMR", 20) == 0)) { + found = 1; + if ((sscanf(a, "T38FaxTranscodingMMR:%d", &x) == 1)) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Transcoding MMR: %d\n",x); + if (x == 1) + peert38capability |= T38FAX_TRANSCODING_MMR; + } else { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Transcoding MMR\n"); + peert38capability |= T38FAX_TRANSCODING_MMR; + } + } else if ((strncmp(a, "T38FaxTranscodingJBIG", 21) == 0)) { + found = 1; + if ((sscanf(a, "T38FaxTranscodingJBIG:%d", &x) == 1)) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Transcoding JBIG: %d\n",x); + if (x == 1) + peert38capability |= T38FAX_TRANSCODING_JBIG; + } else { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Transcoding JBIG\n"); + peert38capability |= T38FAX_TRANSCODING_JBIG; + } + } else if ((sscanf(a, "T38FaxRateManagement:%255s", s) == 1)) { + found = 1; + if (option_debug > 2) + ast_log(LOG_DEBUG, "RateManagement: %s\n", s); + if (!strcasecmp(s, "localTCF")) + peert38capability |= T38FAX_RATE_MANAGEMENT_LOCAL_TCF; + else if (!strcasecmp(s, "transferredTCF")) + peert38capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF; + } else if ((sscanf(a, "T38FaxUdpEC:%255s", s) == 1)) { + found = 1; + if (option_debug > 2) + ast_log(LOG_DEBUG, "UDP EC: %s\n", s); + if (!strcasecmp(s, "t38UDPRedundancy")) { + peert38capability |= T38FAX_UDP_EC_REDUNDANCY; + ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY); + } else if (!strcasecmp(s, "t38UDPFEC")) { + peert38capability |= T38FAX_UDP_EC_FEC; + ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC); + } else { + peert38capability |= T38FAX_UDP_EC_NONE; + ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE); + } + } + } + if (found) { /* Some cisco equipment returns nothing beside c= and m= lines in 200 OK T38 SDP */ + p->t38.peercapability = peert38capability; + p->t38.jointcapability = (peert38capability & 255); /* Put everything beside supported speeds settings */ + peert38capability &= (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400); + p->t38.jointcapability |= (peert38capability & p->t38.capability); /* Put the lower of our's and peer's speed */ + } + if (debug) + ast_log(LOG_DEBUG, "Our T38 capability = (%d), peer T38 capability (%d), joint T38 capability (%d)\n", + p->t38.capability, + p->t38.peercapability, + p->t38.jointcapability); + } else { + p->t38.state = T38_DISABLED; + if (option_debug > 2) + ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + } + + /* Now gather all of the codecs that we are asked for: */ + ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability); + ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability); + + newjointcapability = p->capability & (peercapability | vpeercapability); + newpeercapability = (peercapability | vpeercapability); + newnoncodeccapability = p->noncodeccapability & peernoncodeccapability; + + + if (debug) { + /* shame on whoever coded this.... */ + char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE]; + + ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n", + ast_getformatname_multiple(s1, SIPBUFSIZE, p->capability), + ast_getformatname_multiple(s2, SIPBUFSIZE, newpeercapability), + ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability), + ast_getformatname_multiple(s4, SIPBUFSIZE, newjointcapability)); + + ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n", + ast_rtp_lookup_mime_multiple(s1, SIPBUFSIZE, p->noncodeccapability, 0, 0), + ast_rtp_lookup_mime_multiple(s2, SIPBUFSIZE, peernoncodeccapability, 0, 0), + ast_rtp_lookup_mime_multiple(s3, SIPBUFSIZE, newnoncodeccapability, 0, 0)); + } + if (!newjointcapability) { + /* If T.38 was not negotiated either, totally bail out... */ + if (!p->t38.jointcapability || !udptlportno) { + ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n"); + /* Do NOT Change current setting */ + return -1; + } else { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Have T.38 but no audio codecs, accepting offer anyway\n"); + return 0; + } + } + + /* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since + they are acceptable */ + p->jointcapability = newjointcapability; /* Our joint codec profile for this call */ + p->peercapability = newpeercapability; /* The other sides capability in latest offer */ + p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */ + + ast_rtp_pt_copy(p->rtp, newaudiortp); + if (p->vrtp) + ast_rtp_pt_copy(p->vrtp, newvideortp); + + if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) { + ast_clear_flag(&p->flags[0], SIP_DTMF); + if (newnoncodeccapability & AST_RTP_DTMF) { + /* XXX Would it be reasonable to drop the DSP at this point? XXX */ + ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833); + /* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */ + ast_rtp_setdtmf(p->rtp, 1); + ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE)); + } else { + ast_set_flag(&p->flags[0], SIP_DTMF_INBAND); + } + } + + /* Setup audio port number */ + if (p->rtp && sin.sin_port) { + ast_rtp_set_peer(p->rtp, &sin); + if (debug) + ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); + } + + /* Setup video port number */ + if (p->vrtp && vsin.sin_port) { + ast_rtp_set_peer(p->vrtp, &vsin); + if (debug) + ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port)); + } + + /* Ok, we're going with this offer */ + if (option_debug > 1) { + char buf[SIPBUFSIZE]; + ast_log(LOG_DEBUG, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability)); + } + + if (!p->owner) /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */ + return 0; + + if (option_debug > 3) + ast_log(LOG_DEBUG, "We have an owner, now see if we need to change this call\n"); + + if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) { + if (debug) { + char s1[SIPBUFSIZE], s2[SIPBUFSIZE]; + ast_log(LOG_DEBUG, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n", + ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcapability), + ast_getformatname_multiple(s2, SIPBUFSIZE, p->owner->nativeformats)); + } + p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability); + ast_set_read_format(p->owner, p->owner->readformat); + ast_set_write_format(p->owner, p->owner->writeformat); + } + + if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && sin.sin_addr.s_addr && (!sendonly || sendonly == -1)) { + ast_queue_control(p->owner, AST_CONTROL_UNHOLD); + /* Activate a re-invite */ + ast_queue_frame(p->owner, &ast_null_frame); + } else if (!sin.sin_addr.s_addr || (sendonly && sendonly != -1)) { + ast_queue_control_data(p->owner, AST_CONTROL_HOLD, + S_OR(p->mohsuggest, NULL), + !ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0); + if (sendonly) + ast_rtp_stop(p->rtp); + /* RTCP needs to go ahead, even if we're on hold!!! */ + /* Activate a re-invite */ + ast_queue_frame(p->owner, &ast_null_frame); + } + + /* Manager Hold and Unhold events must be generated, if necessary */ + if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && sin.sin_addr.s_addr && (!sendonly || sendonly == -1)) + change_hold_state(p, req, FALSE, sendonly); + else if (!sin.sin_addr.s_addr || (sendonly && sendonly != -1)) + change_hold_state(p, req, TRUE, sendonly); + return 0; +} + +#ifdef LOW_MEMORY +static void ts_ast_rtp_destroy(void *data) +{ + struct ast_rtp *tmp = data; + ast_rtp_destroy(tmp); +} +#endif + +/*! \brief Add header to SIP message */ +static int add_header(struct sip_request *req, const char *var, const char *value) +{ + int maxlen = sizeof(req->data) - 4 - req->len; /* 4 bytes are for two \r\n ? */ + + if (req->headers == SIP_MAX_HEADERS) { + ast_log(LOG_WARNING, "Out of SIP header space\n"); + return -1; + } + + if (req->lines) { + ast_log(LOG_WARNING, "Can't add more headers when lines have been added\n"); + return -1; + } + + if (maxlen <= 0) { + ast_log(LOG_WARNING, "Out of space, can't add anymore (%s:%s)\n", var, value); + return -1; + } + + req->header[req->headers] = req->data + req->len; + + if (compactheaders) + var = find_alias(var, var); + + snprintf(req->header[req->headers], maxlen, "%s: %s\r\n", var, value); + req->len += strlen(req->header[req->headers]); + req->headers++; + + return 0; +} + +/*! \brief Add 'Content-Length' header to SIP message */ +static int add_header_contentLength(struct sip_request *req, int len) +{ + char clen[10]; + + snprintf(clen, sizeof(clen), "%d", len); + return add_header(req, "Content-Length", clen); +} + +/*! \brief Add content (not header) to SIP message */ +static int add_line(struct sip_request *req, const char *line) +{ + if (req->lines == SIP_MAX_LINES) { + ast_log(LOG_WARNING, "Out of SIP line space\n"); + return -1; + } + if (!req->lines) { + /* Add extra empty return */ + snprintf(req->data + req->len, sizeof(req->data) - req->len, "\r\n"); + req->len += strlen(req->data + req->len); + } + if (req->len >= sizeof(req->data) - 4) { + ast_log(LOG_WARNING, "Out of space, can't add anymore\n"); + return -1; + } + req->line[req->lines] = req->data + req->len; + snprintf(req->line[req->lines], sizeof(req->data) - req->len, "%s", line); + req->len += strlen(req->line[req->lines]); + req->lines++; + return 0; +} + +/*! \brief Copy one header field from one request to another */ +static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field) +{ + const char *tmp = get_header(orig, field); + + if (!ast_strlen_zero(tmp)) /* Add what we're responding to */ + return add_header(req, field, tmp); + ast_log(LOG_NOTICE, "No field '%s' present to copy\n", field); + return -1; +} + +/*! \brief Copy all headers from one request to another */ +static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field) +{ + int start = 0; + int copied = 0; + for (;;) { + const char *tmp = __get_header(orig, field, &start); + + if (ast_strlen_zero(tmp)) + break; + /* Add what we're responding to */ + add_header(req, field, tmp); + copied++; + } + return copied ? 0 : -1; +} + +/*! \brief Copy SIP VIA Headers from the request to the response +\note If the client indicates that it wishes to know the port we received from, + it adds ;rport without an argument to the topmost via header. We need to + add the port number (from our point of view) to that parameter. + We always add ;received=<ip address> to the topmost via header. + Received: RFC 3261, rport RFC 3581 */ +static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field) +{ + int copied = 0; + int start = 0; + + for (;;) { + char new[512]; + const char *oh = __get_header(orig, field, &start); + + if (ast_strlen_zero(oh)) + break; + + if (!copied) { /* Only check for empty rport in topmost via header */ + char leftmost[512], *others, *rport; + + /* Only work on leftmost value */ + ast_copy_string(leftmost, oh, sizeof(leftmost)); + others = strchr(leftmost, ','); + if (others) + *others++ = '\0'; + + /* Find ;rport; (empty request) */ + rport = strstr(leftmost, ";rport"); + if (rport && *(rport+6) == '=') + rport = NULL; /* We already have a parameter to rport */ + + /* Check rport if NAT=yes or NAT=rfc3581 (which is the default setting) */ + if (rport && ((ast_test_flag(&p->flags[0], SIP_NAT) == SIP_NAT_ALWAYS) || (ast_test_flag(&p->flags[0], SIP_NAT) == SIP_NAT_RFC3581))) { + /* We need to add received port - rport */ + char *end; + + rport = strstr(leftmost, ";rport"); + + if (rport) { + end = strchr(rport + 1, ';'); + if (end) + memmove(rport, end, strlen(end) + 1); + else + *rport = '\0'; + } + + /* Add rport to first VIA header if requested */ + snprintf(new, sizeof(new), "%s;received=%s;rport=%d%s%s", + leftmost, ast_inet_ntoa(p->recv.sin_addr), + ntohs(p->recv.sin_port), + others ? "," : "", others ? others : ""); + } else { + /* We should *always* add a received to the topmost via */ + snprintf(new, sizeof(new), "%s;received=%s%s%s", + leftmost, ast_inet_ntoa(p->recv.sin_addr), + others ? "," : "", others ? others : ""); + } + oh = new; /* the header to copy */ + } /* else add the following via headers untouched */ + add_header(req, field, oh); + copied++; + } + if (!copied) { + ast_log(LOG_NOTICE, "No header field '%s' present to copy\n", field); + return -1; + } + return 0; +} + +/*! \brief Add route header into request per learned route */ +static void add_route(struct sip_request *req, struct sip_route *route) +{ + char r[SIPBUFSIZE*2], *p; + int n, rem = sizeof(r); + + if (!route) + return; + + p = r; + for (;route ; route = route->next) { + n = strlen(route->hop); + if (rem < n+3) /* we need room for ",<route>" */ + break; + if (p != r) { /* add a separator after fist route */ + *p++ = ','; + --rem; + } + *p++ = '<'; + ast_copy_string(p, route->hop, rem); /* cannot fail */ + p += n; + *p++ = '>'; + rem -= (n+2); + } + *p = '\0'; + add_header(req, "Route", r); +} + +/*! \brief Set destination from SIP URI */ +static void set_destination(struct sip_pvt *p, char *uri) +{ + char *h, *maddr, hostname[256]; + int port, hn; + struct hostent *hp; + struct ast_hostent ahp; + int debug=sip_debug_test_pvt(p); + + /* Parse uri to h (host) and port - uri is already just the part inside the <> */ + /* general form we are expecting is sip[s]:username[:password]@host[:port][;...] */ + + if (debug) + ast_verbose("set_destination: Parsing <%s> for address/port to send to\n", uri); + + /* Find and parse hostname */ + h = strchr(uri, '@'); + if (h) + ++h; + else { + h = uri; + if (strncasecmp(h, "sip:", 4) == 0) + h += 4; + else if (strncasecmp(h, "sips:", 5) == 0) + h += 5; + } + hn = strcspn(h, ":;>") + 1; + if (hn > sizeof(hostname)) + hn = sizeof(hostname); + ast_copy_string(hostname, h, hn); + /* XXX bug here if string has been trimmed to sizeof(hostname) */ + h += hn - 1; + + /* Is "port" present? if not default to STANDARD_SIP_PORT */ + if (*h == ':') { + /* Parse port */ + ++h; + port = strtol(h, &h, 10); + } + else + port = STANDARD_SIP_PORT; + + /* Got the hostname:port - but maybe there's a "maddr=" to override address? */ + maddr = strstr(h, "maddr="); + if (maddr) { + maddr += 6; + hn = strspn(maddr, "0123456789.") + 1; + if (hn > sizeof(hostname)) + hn = sizeof(hostname); + ast_copy_string(hostname, maddr, hn); + } + + hp = ast_gethostbyname(hostname, &ahp); + if (hp == NULL) { + ast_log(LOG_WARNING, "Can't find address for host '%s'\n", hostname); + return; + } + p->sa.sin_family = AF_INET; + memcpy(&p->sa.sin_addr, hp->h_addr, sizeof(p->sa.sin_addr)); + p->sa.sin_port = htons(port); + if (debug) + ast_verbose("set_destination: set destination to %s, port %d\n", ast_inet_ntoa(p->sa.sin_addr), port); +} + +/*! \brief Initialize SIP response, based on SIP request */ +static int init_resp(struct sip_request *resp, const char *msg) +{ + /* Initialize a response */ + memset(resp, 0, sizeof(*resp)); + resp->method = SIP_RESPONSE; + resp->header[0] = resp->data; + snprintf(resp->header[0], sizeof(resp->data), "SIP/2.0 %s\r\n", msg); + resp->len = strlen(resp->header[0]); + resp->headers++; + return 0; +} + +/*! \brief Initialize SIP request */ +static int init_req(struct sip_request *req, int sipmethod, const char *recip) +{ + /* Initialize a request */ + memset(req, 0, sizeof(*req)); + req->method = sipmethod; + req->header[0] = req->data; + snprintf(req->header[0], sizeof(req->data), "%s %s SIP/2.0\r\n", sip_methods[sipmethod].text, recip); + req->len = strlen(req->header[0]); + req->headers++; + return 0; +} + + +/*! \brief Prepare SIP response packet */ +static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req) +{ + char newto[256]; + const char *ot; + + init_resp(resp, msg); + copy_via_headers(p, resp, req, "Via"); + if (msg[0] == '1' || msg[0] == '2') + copy_all_header(resp, req, "Record-Route"); + copy_header(resp, req, "From"); + ot = get_header(req, "To"); + if (!strcasestr(ot, "tag=") && strncmp(msg, "100", 3)) { + /* Add the proper tag if we don't have it already. If they have specified + their tag, use it. Otherwise, use our own tag */ + if (!ast_strlen_zero(p->theirtag) && ast_test_flag(&p->flags[0], SIP_OUTGOING)) + snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag); + else if (p->tag && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) + snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag); + else + ast_copy_string(newto, ot, sizeof(newto)); + ot = newto; + } + add_header(resp, "To", ot); + copy_header(resp, req, "Call-ID"); + copy_header(resp, req, "CSeq"); + if (!ast_strlen_zero(global_useragent)) + add_header(resp, "User-Agent", global_useragent); + add_header(resp, "Allow", ALLOWED_METHODS); + add_header(resp, "Supported", SUPPORTED_EXTENSIONS); + if (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER)) { + /* For registration responses, we also need expiry and + contact info */ + char tmp[256]; + + snprintf(tmp, sizeof(tmp), "%d", p->expiry); + add_header(resp, "Expires", tmp); + if (p->expiry) { /* Only add contact if we have an expiry time */ + char contact[SIPBUFSIZE]; + snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, p->expiry); + add_header(resp, "Contact", contact); /* Not when we unregister */ + } + } else if (msg[0] != '4' && !ast_strlen_zero(p->our_contact)) { + add_header(resp, "Contact", p->our_contact); + } + return 0; +} + +/*! \brief Initialize a SIP request message (not the initial one in a dialog) */ +static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch) +{ + struct sip_request *orig = &p->initreq; + char stripped[80]; + char tmp[80]; + char newto[256]; + const char *c; + const char *ot, *of; + int is_strict = FALSE; /*!< Strict routing flag */ + + memset(req, 0, sizeof(struct sip_request)); + + snprintf(p->lastmsg, sizeof(p->lastmsg), "Tx: %s", sip_methods[sipmethod].text); + + if (!seqno) { + p->ocseq++; + seqno = p->ocseq; + } + + if (sipmethod == SIP_CANCEL) { + p->branch = p->invite_branch; + build_via(p); + } else if (newbranch) { + p->branch ^= ast_random(); + build_via(p); + } + + /* Check for strict or loose router */ + if (p->route && !ast_strlen_zero(p->route->hop) && strstr(p->route->hop,";lr") == NULL) { + is_strict = TRUE; + if (sipdebug) + ast_log(LOG_DEBUG, "Strict routing enforced for session %s\n", p->callid); + } + + if (sipmethod == SIP_CANCEL) + c = p->initreq.rlPart2; /* Use original URI */ + else if (sipmethod == SIP_ACK) { + /* Use URI from Contact: in 200 OK (if INVITE) + (we only have the contacturi on INVITEs) */ + if (!ast_strlen_zero(p->okcontacturi)) + c = is_strict ? p->route->hop : p->okcontacturi; + else + c = p->initreq.rlPart2; + } else if (!ast_strlen_zero(p->okcontacturi)) + c = is_strict ? p->route->hop : p->okcontacturi; /* Use for BYE or REINVITE */ + else if (!ast_strlen_zero(p->uri)) + c = p->uri; + else { + char *n; + /* We have no URI, use To: or From: header as URI (depending on direction) */ + ast_copy_string(stripped, get_header(orig, (ast_test_flag(&p->flags[0], SIP_OUTGOING)) ? "To" : "From"), + sizeof(stripped)); + n = get_in_brackets(stripped); + c = strsep(&n, ";"); /* trim ; and beyond */ + } + init_req(req, sipmethod, c); + + snprintf(tmp, sizeof(tmp), "%d %s", seqno, sip_methods[sipmethod].text); + + add_header(req, "Via", p->via); + if (p->route) { + set_destination(p, p->route->hop); + add_route(req, is_strict ? p->route->next : p->route); + } + + ot = get_header(orig, "To"); + of = get_header(orig, "From"); + + /* Add tag *unless* this is a CANCEL, in which case we need to send it exactly + as our original request, including tag (or presumably lack thereof) */ + if (!strcasestr(ot, "tag=") && sipmethod != SIP_CANCEL) { + /* Add the proper tag if we don't have it already. If they have specified + their tag, use it. Otherwise, use our own tag */ + if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_strlen_zero(p->theirtag)) + snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag); + else if (!ast_test_flag(&p->flags[0], SIP_OUTGOING)) + snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag); + else + snprintf(newto, sizeof(newto), "%s", ot); + ot = newto; + } + + if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) { + add_header(req, "From", of); + add_header(req, "To", ot); + } else { + add_header(req, "From", ot); + add_header(req, "To", of); + } + /* Do not add Contact for MESSAGE, BYE and Cancel requests */ + if (sipmethod != SIP_BYE && sipmethod != SIP_CANCEL && sipmethod != SIP_MESSAGE) + add_header(req, "Contact", p->our_contact); + + copy_header(req, orig, "Call-ID"); + add_header(req, "CSeq", tmp); + + if (!ast_strlen_zero(global_useragent)) + add_header(req, "User-Agent", global_useragent); + add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS); + + if (!ast_strlen_zero(p->rpid)) + add_header(req, "Remote-Party-ID", p->rpid); + + return 0; +} + +/*! \brief Base transmit response function */ +static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable) +{ + struct sip_request resp; + int seqno = 0; + + if (reliable && (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1)) { + ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq")); + return -1; + } + respprep(&resp, p, msg, req); + add_header_contentLength(&resp, 0); + /* If we are cancelling an incoming invite for some reason, add information + about the reason why we are doing this in clear text */ + if (p->method == SIP_INVITE && msg[0] != '1' && p->owner && p->owner->hangupcause) { + char buf[10]; + + add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause)); + snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause); + add_header(&resp, "X-Asterisk-HangupCauseCode", buf); + } + return send_response(p, &resp, reliable, seqno); +} + +static void temp_pvt_cleanup(void *data) +{ + struct sip_pvt *p = data; + + ast_string_field_free_memory(p); + + free(data); +} + +/*! \brief Transmit response, no retransmits, using a temporary pvt structure */ +static int transmit_response_using_temp(ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg) +{ + struct sip_pvt *p = NULL; + + if (!(p = ast_threadstorage_get(&ts_temp_pvt, sizeof(*p)))) { + ast_log(LOG_NOTICE, "Failed to get temporary pvt\n"); + return -1; + } + + /* if the structure was just allocated, initialize it */ + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) { + ast_set_flag(&p->flags[0], SIP_NO_HISTORY); + if (ast_string_field_init(p, 512)) + return -1; + } + + /* Initialize the bare minimum */ + p->method = intended_method; + + if (sin) { + p->sa = *sin; + if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip)) + p->ourip = __ourip; + } else + p->ourip = __ourip; + + p->branch = ast_random(); + make_our_tag(p->tag, sizeof(p->tag)); + p->ocseq = INITIAL_CSEQ; + + if (useglobal_nat && sin) { + ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT); + p->recv = *sin; + do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE); + } + check_via(p, req); + + ast_string_field_set(p, fromdomain, default_fromdomain); + build_via(p); + ast_string_field_set(p, callid, callid); + + /* Use this temporary pvt structure to send the message */ + __transmit_response(p, msg, req, XMIT_UNRELIABLE); + + /* Free the string fields, but not the pool space */ + ast_string_field_reset_all(p); + + return 0; +} + +/*! \brief Transmit response, no retransmits */ +static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req) +{ + return __transmit_response(p, msg, req, XMIT_UNRELIABLE); +} + +/*! \brief Transmit response, no retransmits */ +static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported) +{ + struct sip_request resp; + respprep(&resp, p, msg, req); + append_date(&resp); + add_header(&resp, "Unsupported", unsupported); + add_header_contentLength(&resp, 0); + return send_response(p, &resp, XMIT_UNRELIABLE, 0); +} + +/*! \brief Transmit response, Make sure you get an ACK + This is only used for responses to INVITEs, where we need to make sure we get an ACK +*/ +static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req) +{ + return __transmit_response(p, msg, req, XMIT_CRITICAL); +} + +/*! \brief Append date to SIP message */ +static void append_date(struct sip_request *req) +{ + char tmpdat[256]; + struct tm tm; + time_t t = time(NULL); + + gmtime_r(&t, &tm); + strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T GMT", &tm); + add_header(req, "Date", tmpdat); +} + +/*! \brief Append date and content length before transmitting response */ +static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req) +{ + struct sip_request resp; + respprep(&resp, p, msg, req); + append_date(&resp); + add_header_contentLength(&resp, 0); + return send_response(p, &resp, XMIT_UNRELIABLE, 0); +} + +/*! \brief Append Accept header, content length before transmitting response */ +static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable) +{ + struct sip_request resp; + respprep(&resp, p, msg, req); + add_header(&resp, "Accept", "application/sdp"); + add_header_contentLength(&resp, 0); + return send_response(p, &resp, reliable, 0); +} + +/*! \brief Respond with authorization request */ +static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *randdata, enum xmittype reliable, const char *header, int stale) +{ + struct sip_request resp; + char tmp[512]; + int seqno = 0; + + if (reliable && (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1)) { + ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq")); + return -1; + } + /* Stale means that they sent us correct authentication, but + based it on an old challenge (nonce) */ + snprintf(tmp, sizeof(tmp), "Digest algorithm=MD5, realm=\"%s\", nonce=\"%s\"%s", global_realm, randdata, stale ? ", stale=true" : ""); + respprep(&resp, p, msg, req); + add_header(&resp, header, tmp); + add_header_contentLength(&resp, 0); + append_history(p, "AuthChal", "Auth challenge sent for %s - nc %d", p->username, p->noncecount); + return send_response(p, &resp, reliable, seqno); +} + +/*! \brief Add text body to SIP message */ +static int add_text(struct sip_request *req, const char *text) +{ + /* XXX Convert \n's to \r\n's XXX */ + add_header(req, "Content-Type", "text/plain"); + add_header_contentLength(req, strlen(text)); + add_line(req, text); + return 0; +} + +/*! \brief Add DTMF INFO tone to sip message */ +/* Always adds default duration 250 ms, regardless of what came in over the line */ +static int add_digit(struct sip_request *req, char digit, unsigned int duration) +{ + char tmp[256]; + + snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=%u\r\n", digit, duration); + add_header(req, "Content-Type", "application/dtmf-relay"); + add_header_contentLength(req, strlen(tmp)); + add_line(req, tmp); + return 0; +} + +/*! \brief add XML encoded media control with update + \note XML: The only way to turn 0 bits of information into a few hundred. (markster) */ +static int add_vidupdate(struct sip_request *req) +{ + const char *xml_is_a_huge_waste_of_space = + "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n" + " <media_control>\r\n" + " <vc_primitive>\r\n" + " <to_encoder>\r\n" + " <picture_fast_update>\r\n" + " </picture_fast_update>\r\n" + " </to_encoder>\r\n" + " </vc_primitive>\r\n" + " </media_control>\r\n"; + add_header(req, "Content-Type", "application/media_control+xml"); + add_header_contentLength(req, strlen(xml_is_a_huge_waste_of_space)); + add_line(req, xml_is_a_huge_waste_of_space); + return 0; +} + +/*! \brief Add codec offer to SDP offer/answer body in INVITE or 200 OK */ +static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate, + char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, + int debug, int *min_packet_size) +{ + int rtp_code; + struct ast_format_list fmt; + + + if (debug) + ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec)); + if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1) + return; + + if (p->rtp) { + struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp); + fmt = ast_codec_pref_getsize(pref, codec); + } else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */ + return; + ast_build_string(m_buf, m_size, " %d", rtp_code); + ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code, + ast_rtp_lookup_mime_subtype(1, codec, + ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0), + sample_rate); + if (codec == AST_FORMAT_G729A) { + /* Indicate that we don't support VAD (G.729 annex B) */ + ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code); + } else if (codec == AST_FORMAT_G723_1) { + /* Indicate that we don't support VAD (G.723.1 annex A) */ + ast_build_string(a_buf, a_size, "a=fmtp:%d annexa=no\r\n", rtp_code); + } else if (codec == AST_FORMAT_ILBC) { + /* Add information about us using only 20/30 ms packetization */ + ast_build_string(a_buf, a_size, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms); + } + + if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size)) + *min_packet_size = fmt.cur_ms; + + /* Our first codec packetization processed cannot be less than zero */ + if ((*min_packet_size) == 0 && fmt.cur_ms) + *min_packet_size = fmt.cur_ms; +} + +/*! \brief Get Max T.38 Transmission rate from T38 capabilities */ +static int t38_get_rate(int t38cap) +{ + int maxrate = (t38cap & (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400)); + + if (maxrate & T38FAX_RATE_14400) { + if (option_debug > 1) + ast_log(LOG_DEBUG, "T38MaxBitRate 14400 found\n"); + return 14400; + } else if (maxrate & T38FAX_RATE_12000) { + if (option_debug > 1) + ast_log(LOG_DEBUG, "T38MaxBitRate 12000 found\n"); + return 12000; + } else if (maxrate & T38FAX_RATE_9600) { + if (option_debug > 1) + ast_log(LOG_DEBUG, "T38MaxBitRate 9600 found\n"); + return 9600; + } else if (maxrate & T38FAX_RATE_7200) { + if (option_debug > 1) + ast_log(LOG_DEBUG, "T38MaxBitRate 7200 found\n"); + return 7200; + } else if (maxrate & T38FAX_RATE_4800) { + if (option_debug > 1) + ast_log(LOG_DEBUG, "T38MaxBitRate 4800 found\n"); + return 4800; + } else if (maxrate & T38FAX_RATE_2400) { + if (option_debug > 1) + ast_log(LOG_DEBUG, "T38MaxBitRate 2400 found\n"); + return 2400; + } else { + if (option_debug > 1) + ast_log(LOG_DEBUG, "Strange, T38MaxBitRate NOT found in peers T38 SDP.\n"); + return 0; + } +} + +/*! \brief Add T.38 Session Description Protocol message */ +static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p) +{ + int len = 0; + int x = 0; + struct sockaddr_in udptlsin; + char v[256] = ""; + char s[256] = ""; + char o[256] = ""; + char c[256] = ""; + char t[256] = ""; + char m_modem[256]; + char a_modem[1024]; + char *m_modem_next = m_modem; + size_t m_modem_left = sizeof(m_modem); + char *a_modem_next = a_modem; + size_t a_modem_left = sizeof(a_modem); + struct sockaddr_in udptldest = { 0, }; + int debug; + + debug = sip_debug_test_pvt(p); + len = 0; + if (!p->udptl) { + ast_log(LOG_WARNING, "No way to add SDP without an UDPTL structure\n"); + return -1; + } + + if (!p->sessionid) { + p->sessionid = getpid(); + p->sessionversion = p->sessionid; + } else + p->sessionversion++; + + /* Our T.38 end is */ + ast_udptl_get_us(p->udptl, &udptlsin); + + /* Determine T.38 UDPTL destination */ + if (p->udptlredirip.sin_addr.s_addr) { + udptldest.sin_port = p->udptlredirip.sin_port; + udptldest.sin_addr = p->udptlredirip.sin_addr; + } else { + udptldest.sin_addr = p->ourip; + udptldest.sin_port = udptlsin.sin_port; + } + + if (debug) + ast_log(LOG_DEBUG, "T.38 UDPTL is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(udptlsin.sin_port)); + + /* We break with the "recommendation" and send our IP, in order that our + peer doesn't have to ast_gethostbyname() us */ + + if (debug) { + ast_log(LOG_DEBUG, "Our T38 capability (%d), peer T38 capability (%d), joint capability (%d)\n", + p->t38.capability, + p->t38.peercapability, + p->t38.jointcapability); + } + snprintf(v, sizeof(v), "v=0\r\n"); + snprintf(o, sizeof(o), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(udptldest.sin_addr)); + snprintf(s, sizeof(s), "s=session\r\n"); + snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(udptldest.sin_addr)); + snprintf(t, sizeof(t), "t=0 0\r\n"); + ast_build_string(&m_modem_next, &m_modem_left, "m=image %d udptl t38\r\n", ntohs(udptldest.sin_port)); + + if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_0) + ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxVersion:0\r\n"); + if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_1) + ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxVersion:1\r\n"); + if ((x = t38_get_rate(p->t38.jointcapability))) + ast_build_string(&a_modem_next, &a_modem_left, "a=T38MaxBitRate:%d\r\n",x); + if ((p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) == T38FAX_FILL_BIT_REMOVAL) + ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxFillBitRemoval\r\n"); + if ((p->t38.jointcapability & T38FAX_TRANSCODING_MMR) == T38FAX_TRANSCODING_MMR) + ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingMMR\r\n"); + if ((p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) == T38FAX_TRANSCODING_JBIG) + ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingJBIG\r\n"); + ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxRateManagement:%s\r\n", (p->t38.jointcapability & T38FAX_RATE_MANAGEMENT_LOCAL_TCF) ? "localTCF" : "transferredTCF"); + x = ast_udptl_get_local_max_datagram(p->udptl); + ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxMaxBuffer:%d\r\n",x); + ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxMaxDatagram:%d\r\n",x); + if (p->t38.jointcapability != T38FAX_UDP_EC_NONE) + ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxUdpEC:%s\r\n", (p->t38.jointcapability & T38FAX_UDP_EC_REDUNDANCY) ? "t38UDPRedundancy" : "t38UDPFEC"); + len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m_modem) + strlen(a_modem); + add_header(resp, "Content-Type", "application/sdp"); + add_header_contentLength(resp, len); + add_line(resp, v); + add_line(resp, o); + add_line(resp, s); + add_line(resp, c); + add_line(resp, t); + add_line(resp, m_modem); + add_line(resp, a_modem); + + /* Update lastrtprx when we send our SDP */ + p->lastrtprx = p->lastrtptx = time(NULL); + + return 0; +} + + +/*! \brief Add RFC 2833 DTMF offer to SDP */ +static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate, + char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, + int debug) +{ + int rtp_code; + + if (debug) + ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype(0, format, 0)); + if ((rtp_code = ast_rtp_lookup_code(p->rtp, 0, format)) == -1) + return; + + ast_build_string(m_buf, m_size, " %d", rtp_code); + ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code, + ast_rtp_lookup_mime_subtype(0, format, 0), + sample_rate); + if (format == AST_RTP_DTMF) + /* Indicate we support DTMF and FLASH... */ + ast_build_string(a_buf, a_size, "a=fmtp:%d 0-16\r\n", rtp_code); +} + +/*! + * \note G.722 actually is supposed to specified as 8 kHz, even though it is + * really 16 kHz. Update this macro for other formats as they are added in + * the future. + */ +#define SDP_SAMPLE_RATE(x) 8000 + +/*! \brief Add Session Description Protocol message */ +static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) +{ + int len = 0; + int alreadysent = 0; + + struct sockaddr_in sin; + struct sockaddr_in vsin; + struct sockaddr_in dest; + struct sockaddr_in vdest = { 0, }; + + /* SDP fields */ + char *version = "v=0\r\n"; /* Protocol version */ + char *subject = "s=session\r\n"; /* Subject of the session */ + char owner[256]; /* Session owner/creator */ + char connection[256]; /* Connection data */ + char *stime = "t=0 0\r\n"; /* Time the session is active */ + char bandwidth[256] = ""; /* Max bitrate */ + char *hold; + char m_audio[256]; /* Media declaration line for audio */ + char m_video[256]; /* Media declaration line for video */ + char a_audio[1024]; /* Attributes for audio */ + char a_video[1024]; /* Attributes for video */ + char *m_audio_next = m_audio; + char *m_video_next = m_video; + size_t m_audio_left = sizeof(m_audio); + size_t m_video_left = sizeof(m_video); + char *a_audio_next = a_audio; + char *a_video_next = a_video; + size_t a_audio_left = sizeof(a_audio); + size_t a_video_left = sizeof(a_video); + + int x; + int capability; + int needvideo = FALSE; + int debug = sip_debug_test_pvt(p); + int min_audio_packet_size = 0; + int min_video_packet_size = 0; + + m_video[0] = '\0'; /* Reset the video media string if it's not needed */ + + if (!p->rtp) { + ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n"); + return AST_FAILURE; + } + + /* Set RTP Session ID and version */ + if (!p->sessionid) { + p->sessionid = getpid(); + p->sessionversion = p->sessionid; + } else + p->sessionversion++; + + /* Get our addresses */ + ast_rtp_get_us(p->rtp, &sin); + if (p->vrtp) + ast_rtp_get_us(p->vrtp, &vsin); + + /* Is this a re-invite to move the media out, then use the original offer from caller */ + if (p->redirip.sin_addr.s_addr) { + dest.sin_port = p->redirip.sin_port; + dest.sin_addr = p->redirip.sin_addr; + } else { + dest.sin_addr = p->ourip; + dest.sin_port = sin.sin_port; + } + + capability = p->jointcapability; + + + if (option_debug > 1) { + char codecbuf[SIPBUFSIZE]; + ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False"); + ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec)); + } + +#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS + if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP)) { + ast_build_string(&m_audio_next, &m_audio_left, " %d", 191); + ast_build_string(&a_audio_next, &a_audio_left, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000); + } +#endif + + /* Check if we need video in this call */ + if ((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) { + if (p->vrtp) { + needvideo = TRUE; + if (option_debug > 1) + ast_log(LOG_DEBUG, "This call needs video offers!\n"); + } else if (option_debug > 1) + ast_log(LOG_DEBUG, "This call needs video offers, but there's no video support enabled!\n"); + } + + + /* Ok, we need video. Let's add what we need for video and set codecs. + Video is handled differently than audio since we can not transcode. */ + if (needvideo) { + /* Determine video destination */ + if (p->vredirip.sin_addr.s_addr) { + vdest.sin_addr = p->vredirip.sin_addr; + vdest.sin_port = p->vredirip.sin_port; + } else { + vdest.sin_addr = p->ourip; + vdest.sin_port = vsin.sin_port; + } + ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vdest.sin_port)); + + /* Build max bitrate string */ + if (p->maxcallbitrate) + snprintf(bandwidth, sizeof(bandwidth), "b=CT:%d\r\n", p->maxcallbitrate); + if (debug) + ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(vsin.sin_port)); + } + + if (debug) + ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(sin.sin_port)); + + /* Start building generic SDP headers */ + + /* We break with the "recommendation" and send our IP, in order that our + peer doesn't have to ast_gethostbyname() us */ + + snprintf(owner, sizeof(owner), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr)); + snprintf(connection, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr)); + ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port)); + + if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR) + hold = "a=recvonly\r\n"; + else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE) + hold = "a=inactive\r\n"; + else + hold = "a=sendrecv\r\n"; + + /* Now, start adding audio codecs. These are added in this order: + - First what was requested by the calling channel + - Then preferences in order from sip.conf device config for this peer/user + - Then other codecs in capabilities, including video + */ + + /* Prefer the audio codec we were requested to use, first, no matter what + Note that p->prefcodec can include video codecs, so mask them out + */ + if (capability & p->prefcodec) { + int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK; + + add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec), + &m_audio_next, &m_audio_left, + &a_audio_next, &a_audio_left, + debug, &min_audio_packet_size); + alreadysent |= codec; + } + + /* Start by sending our preferred audio codecs */ + for (x = 0; x < 32; x++) { + int codec; + + if (!(codec = ast_codec_pref_index(&p->prefs, x))) + break; + + if (!(capability & codec)) + continue; + + if (alreadysent & codec) + continue; + + add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec), + &m_audio_next, &m_audio_left, + &a_audio_next, &a_audio_left, + debug, &min_audio_packet_size); + alreadysent |= codec; + } + + /* Now send any other common audio and video codecs, and non-codec formats: */ + for (x = 1; x <= (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) { + if (!(capability & x)) /* Codec not requested */ + continue; + + if (alreadysent & x) /* Already added to SDP */ + continue; + + if (x <= AST_FORMAT_MAX_AUDIO) + add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x), + &m_audio_next, &m_audio_left, + &a_audio_next, &a_audio_left, + debug, &min_audio_packet_size); + else + add_codec_to_sdp(p, x, 90000, + &m_video_next, &m_video_left, + &a_video_next, &a_video_left, + debug, &min_video_packet_size); + } + + /* Now add DTMF RFC2833 telephony-event as a codec */ + for (x = 1; x <= AST_RTP_MAX; x <<= 1) { + if (!(p->jointnoncodeccapability & x)) + continue; + + add_noncodec_to_sdp(p, x, 8000, + &m_audio_next, &m_audio_left, + &a_audio_next, &a_audio_left, + debug); + } + + if (option_debug > 2) + ast_log(LOG_DEBUG, "-- Done with adding codecs to SDP\n"); + + if (!p->owner || !ast_internal_timing_enabled(p->owner)) + ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n"); + + if (min_audio_packet_size) + ast_build_string(&a_audio_next, &a_audio_left, "a=ptime:%d\r\n", min_audio_packet_size); + + if (min_video_packet_size) + ast_build_string(&a_video_next, &a_video_left, "a=ptime:%d\r\n", min_video_packet_size); + + if ((m_audio_left < 2) || (m_video_left < 2) || (a_audio_left == 0) || (a_video_left == 0)) + ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n"); + + ast_build_string(&m_audio_next, &m_audio_left, "\r\n"); + if (needvideo) + ast_build_string(&m_video_next, &m_video_left, "\r\n"); + + len = strlen(version) + strlen(subject) + strlen(owner) + strlen(connection) + strlen(stime) + strlen(m_audio) + strlen(a_audio) + strlen(hold); + if (needvideo) /* only if video response is appropriate */ + len += strlen(m_video) + strlen(a_video) + strlen(bandwidth) + strlen(hold); + + add_header(resp, "Content-Type", "application/sdp"); + add_header_contentLength(resp, len); + add_line(resp, version); + add_line(resp, owner); + add_line(resp, subject); + add_line(resp, connection); + if (needvideo) /* only if video response is appropriate */ + add_line(resp, bandwidth); + add_line(resp, stime); + add_line(resp, m_audio); + add_line(resp, a_audio); + add_line(resp, hold); + if (needvideo) { /* only if video response is appropriate */ + add_line(resp, m_video); + add_line(resp, a_video); + add_line(resp, hold); /* Repeat hold for the video stream */ + } + + /* Update lastrtprx when we send our SDP */ + p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */ + + if (option_debug > 2) { + char buf[SIPBUFSIZE]; + ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, capability)); + } + + return AST_SUCCESS; +} + +/*! \brief Used for 200 OK and 183 early media */ +static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans) +{ + struct sip_request resp; + int seqno; + + if (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1) { + ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", get_header(req, "CSeq")); + return -1; + } + respprep(&resp, p, msg, req); + if (p->udptl) { + ast_udptl_offered_from_local(p->udptl, 0); + add_t38_sdp(&resp, p); + } else + ast_log(LOG_ERROR, "Can't add SDP to response, since we have no UDPTL session allocated. Call-ID %s\n", p->callid); + if (retrans && !p->pendinginvite) + p->pendinginvite = seqno; /* Buggy clients sends ACK on RINGING too */ + return send_response(p, &resp, retrans, seqno); +} + +/*! \brief copy SIP request (mostly used to save request for responses) */ +static void copy_request(struct sip_request *dst, const struct sip_request *src) +{ + long offset; + int x; + offset = ((void *)dst) - ((void *)src); + /* First copy stuff */ + memcpy(dst, src, sizeof(*dst)); + /* Now fix pointer arithmetic */ + for (x=0; x < src->headers; x++) + dst->header[x] += offset; + for (x=0; x < src->lines; x++) + dst->line[x] += offset; + dst->rlPart1 += offset; + dst->rlPart2 += offset; +} + +/*! \brief Used for 200 OK and 183 early media + \return Will return XMIT_ERROR for network errors. +*/ +static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable) +{ + struct sip_request resp; + int seqno; + if (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1) { + ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", get_header(req, "CSeq")); + return -1; + } + respprep(&resp, p, msg, req); + if (p->rtp) { + if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) { + if (option_debug) + ast_log(LOG_DEBUG, "Setting framing from config on incoming call\n"); + ast_rtp_codec_setpref(p->rtp, &p->prefs); + } + try_suggested_sip_codec(p); + add_sdp(&resp, p); + } else + ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid); + if (reliable && !p->pendinginvite) + p->pendinginvite = seqno; /* Buggy clients sends ACK on RINGING too */ + return send_response(p, &resp, reliable, seqno); +} + +/*! \brief Parse first line of incoming SIP request */ +static int determine_firstline_parts(struct sip_request *req) +{ + char *e = ast_skip_blanks(req->header[0]); /* there shouldn't be any */ + + if (!*e) + return -1; + req->rlPart1 = e; /* method or protocol */ + e = ast_skip_nonblanks(e); + if (*e) + *e++ = '\0'; + /* Get URI or status code */ + e = ast_skip_blanks(e); + if ( !*e ) + return -1; + ast_trim_blanks(e); + + if (!strcasecmp(req->rlPart1, "SIP/2.0") ) { /* We have a response */ + if (strlen(e) < 3) /* status code is 3 digits */ + return -1; + req->rlPart2 = e; + } else { /* We have a request */ + if ( *e == '<' ) { /* XXX the spec says it must not be in <> ! */ + ast_log(LOG_WARNING, "bogus uri in <> %s\n", e); + e++; + if (!*e) + return -1; + } + req->rlPart2 = e; /* URI */ + e = ast_skip_nonblanks(e); + if (*e) + *e++ = '\0'; + e = ast_skip_blanks(e); + if (strcasecmp(e, "SIP/2.0") ) { + ast_log(LOG_WARNING, "Bad request protocol %s\n", e); + return -1; + } + } + return 1; +} + +/*! \brief Transmit reinvite with SDP +\note A re-invite is basically a new INVITE with the same CALL-ID and TAG as the + INVITE that opened the SIP dialogue + We reinvite so that the audio stream (RTP) go directly between + the SIP UAs. SIP Signalling stays with * in the path. +*/ +static int transmit_reinvite_with_sdp(struct sip_pvt *p) +{ + struct sip_request req; + + reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1); + + add_header(&req, "Allow", ALLOWED_METHODS); + add_header(&req, "Supported", SUPPORTED_EXTENSIONS); + if (sipdebug) + add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)"); + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) + append_history(p, "ReInv", "Re-invite sent"); + add_sdp(&req, p); + /* Use this as the basis */ + initialize_initreq(p, &req); + p->lastinvite = p->ocseq; + ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Change direction of this dialog */ + return send_request(p, &req, XMIT_CRITICAL, p->ocseq); +} + +/*! \brief Transmit reinvite with T38 SDP + We reinvite so that the T38 processing can take place. + SIP Signalling stays with * in the path. +*/ +static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p) +{ + struct sip_request req; + + reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1); + + add_header(&req, "Allow", ALLOWED_METHODS); + add_header(&req, "Supported", SUPPORTED_EXTENSIONS); + if (sipdebug) + add_header(&req, "X-asterisk-info", "SIP re-invite (T38 switchover)"); + ast_udptl_offered_from_local(p->udptl, 1); + add_t38_sdp(&req, p); + /* Use this as the basis */ + initialize_initreq(p, &req); + ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Change direction of this dialog */ + p->lastinvite = p->ocseq; + return send_request(p, &req, XMIT_CRITICAL, p->ocseq); +} + +/*! \brief Check Contact: URI of SIP message */ +static void extract_uri(struct sip_pvt *p, struct sip_request *req) +{ + char stripped[SIPBUFSIZE]; + char *c; + + ast_copy_string(stripped, get_header(req, "Contact"), sizeof(stripped)); + c = get_in_brackets(stripped); + c = strsep(&c, ";"); /* trim ; and beyond */ + if (!ast_strlen_zero(c)) + ast_string_field_set(p, uri, c); +} + +/*! \brief Build contact header - the contact header we send out */ +static void build_contact(struct sip_pvt *p) +{ + /* Construct Contact: header */ + if (ourport != STANDARD_SIP_PORT) + ast_string_field_build(p, our_contact, "<sip:%s%s%s:%d>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip), ourport); + else + ast_string_field_build(p, our_contact, "<sip:%s%s%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip)); +} + +/*! \brief Build the Remote Party-ID & From using callingpres options */ +static void build_rpid(struct sip_pvt *p) +{ + int send_pres_tags = TRUE; + const char *privacy=NULL; + const char *screen=NULL; + char buf[256]; + const char *clid = default_callerid; + const char *clin = NULL; + const char *fromdomain; + + if (!ast_strlen_zero(p->rpid) || !ast_strlen_zero(p->rpid_from)) + return; + + if (p->owner && p->owner->cid.cid_num) + clid = p->owner->cid.cid_num; + if (p->owner && p->owner->cid.cid_name) + clin = p->owner->cid.cid_name; + if (ast_strlen_zero(clin)) + clin = clid; + + switch (p->callingpres) { + case AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED: + privacy = "off"; + screen = "no"; + break; + case AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN: + privacy = "off"; + screen = "yes"; + break; + case AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN: + privacy = "off"; + screen = "no"; + break; + case AST_PRES_ALLOWED_NETWORK_NUMBER: + privacy = "off"; + screen = "yes"; + break; + case AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED: + privacy = "full"; + screen = "no"; + break; + case AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN: + privacy = "full"; + screen = "yes"; + break; + case AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN: + privacy = "full"; + screen = "no"; + break; + case AST_PRES_PROHIB_NETWORK_NUMBER: + privacy = "full"; + screen = "yes"; + break; + case AST_PRES_NUMBER_NOT_AVAILABLE: + send_pres_tags = FALSE; + break; + default: + ast_log(LOG_WARNING, "Unsupported callingpres (%d)\n", p->callingpres); + if ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) + privacy = "full"; + else + privacy = "off"; + screen = "no"; + break; + } + + fromdomain = S_OR(p->fromdomain, ast_inet_ntoa(p->ourip)); + + snprintf(buf, sizeof(buf), "\"%s\" <sip:%s@%s>", clin, clid, fromdomain); + if (send_pres_tags) + snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), ";privacy=%s;screen=%s", privacy, screen); + ast_string_field_set(p, rpid, buf); + + ast_string_field_build(p, rpid_from, "\"%s\" <sip:%s@%s>;tag=%s", clin, + S_OR(p->fromuser, clid), + fromdomain, p->tag); +} + +/*! \brief Initiate new SIP request to peer/user */ +static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod) +{ + char invite_buf[256] = ""; + char *invite = invite_buf; + size_t invite_max = sizeof(invite_buf); + char from[256]; + char to[256]; + char tmp[SIPBUFSIZE/2]; + char tmp2[SIPBUFSIZE/2]; + const char *l = NULL, *n = NULL; + const char *urioptions = ""; + + if (ast_test_flag(&p->flags[0], SIP_USEREQPHONE)) { + const char *s = p->username; /* being a string field, cannot be NULL */ + + /* Test p->username against allowed characters in AST_DIGIT_ANY + If it matches the allowed characters list, then sipuser = ";user=phone" + If not, then sipuser = "" + */ + /* + is allowed in first position in a tel: uri */ + if (*s == '+') + s++; + for (; *s; s++) { + if (!strchr(AST_DIGIT_ANYNUM, *s) ) + break; + } + /* If we have only digits, add ;user=phone to the uri */ + if (!*s) + urioptions = ";user=phone"; + } + + + snprintf(p->lastmsg, sizeof(p->lastmsg), "Init: %s", sip_methods[sipmethod].text); + + if (p->owner) { + l = p->owner->cid.cid_num; + n = p->owner->cid.cid_name; + } + /* if we are not sending RPID and user wants his callerid restricted */ + if (!ast_test_flag(&p->flags[0], SIP_SENDRPID) && + ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED)) { + l = CALLERID_UNKNOWN; + n = l; + } + if (ast_strlen_zero(l)) + l = default_callerid; + if (ast_strlen_zero(n)) + n = l; + /* Allow user to be overridden */ + if (!ast_strlen_zero(p->fromuser)) + l = p->fromuser; + else /* Save for any further attempts */ + ast_string_field_set(p, fromuser, l); + + /* Allow user to be overridden */ + if (!ast_strlen_zero(p->fromname)) + n = p->fromname; + else /* Save for any further attempts */ + ast_string_field_set(p, fromname, n); + + if (pedanticsipchecking) { + ast_uri_encode(n, tmp, sizeof(tmp), 0); + n = tmp; + ast_uri_encode(l, tmp2, sizeof(tmp2), 0); + l = tmp2; + } + + if (ourport != STANDARD_SIP_PORT && ast_strlen_zero(p->fromdomain)) + snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", n, l, S_OR(p->fromdomain, ast_inet_ntoa(p->ourip)), ourport, p->tag); + else + snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=%s", n, l, S_OR(p->fromdomain, ast_inet_ntoa(p->ourip)), p->tag); + + /* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */ + if (!ast_strlen_zero(p->fullcontact)) { + /* If we have full contact, trust it */ + ast_build_string(&invite, &invite_max, "%s", p->fullcontact); + } else { + /* Otherwise, use the username while waiting for registration */ + ast_build_string(&invite, &invite_max, "sip:"); + if (!ast_strlen_zero(p->username)) { + n = p->username; + if (pedanticsipchecking) { + ast_uri_encode(n, tmp, sizeof(tmp), 0); + n = tmp; + } + ast_build_string(&invite, &invite_max, "%s@", n); + } + ast_build_string(&invite, &invite_max, "%s", p->tohost); + if (ntohs(p->sa.sin_port) != STANDARD_SIP_PORT) + ast_build_string(&invite, &invite_max, ":%d", ntohs(p->sa.sin_port)); + ast_build_string(&invite, &invite_max, "%s", urioptions); + } + + /* If custom URI options have been provided, append them */ + if (p->options && !ast_strlen_zero(p->options->uri_options)) + ast_build_string(&invite, &invite_max, ";%s", p->options->uri_options); + + ast_string_field_set(p, uri, invite_buf); + + if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) { + /* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */ + snprintf(to, sizeof(to), "<%s%s>;tag=%s", (!strncasecmp(p->uri, "sip:", 4) ? "" : "sip:"), p->uri, p->theirtag); + } else if (p->options && p->options->vxml_url) { + /* If there is a VXML URL append it to the SIP URL */ + snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url); + } else + snprintf(to, sizeof(to), "<%s>", p->uri); + + init_req(req, sipmethod, p->uri); + snprintf(tmp, sizeof(tmp), "%d %s", ++p->ocseq, sip_methods[sipmethod].text); + + add_header(req, "Via", p->via); + /* SLD: FIXME?: do Route: here too? I think not cos this is the first request. + * OTOH, then we won't have anything in p->route anyway */ + /* Build Remote Party-ID and From */ + if (ast_test_flag(&p->flags[0], SIP_SENDRPID) && (sipmethod == SIP_INVITE)) { + build_rpid(p); + add_header(req, "From", p->rpid_from); + } else + add_header(req, "From", from); + add_header(req, "To", to); + ast_string_field_set(p, exten, l); + build_contact(p); + add_header(req, "Contact", p->our_contact); + add_header(req, "Call-ID", p->callid); + add_header(req, "CSeq", tmp); + if (!ast_strlen_zero(global_useragent)) + add_header(req, "User-Agent", global_useragent); + add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS); + if (!ast_strlen_zero(p->rpid)) + add_header(req, "Remote-Party-ID", p->rpid); +} + +/*! \brief Build REFER/INVITE/OPTIONS message and transmit it */ +static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init) +{ + struct sip_request req; + + req.method = sipmethod; + if (init) { /* Seems like init always is 2 */ + /* Bump branch even on initial requests */ + p->branch ^= ast_random(); + p->invite_branch = p->branch; + build_via(p); + if (init > 1) + initreqprep(&req, p, sipmethod); + else + reqprep(&req, p, sipmethod, 0, 1); + } else + reqprep(&req, p, sipmethod, 0, 1); + + if (p->options && p->options->auth) + add_header(&req, p->options->authheader, p->options->auth); + append_date(&req); + if (sipmethod == SIP_REFER) { /* Call transfer */ + if (p->refer) { + char buf[SIPBUFSIZE]; + if (!ast_strlen_zero(p->refer->refer_to)) + add_header(&req, "Refer-To", p->refer->refer_to); + if (!ast_strlen_zero(p->refer->referred_by)) { + snprintf(buf, sizeof(buf), "%s <%s>", p->refer->referred_by_name, p->refer->referred_by); + add_header(&req, "Referred-By", buf); + } + } + } + /* This new INVITE is part of an attended transfer. Make sure that the + other end knows and replace the current call with this new call */ + if (p->options && p->options->replaces && !ast_strlen_zero(p->options->replaces)) { + add_header(&req, "Replaces", p->options->replaces); + add_header(&req, "Require", "replaces"); + } + + add_header(&req, "Allow", ALLOWED_METHODS); + add_header(&req, "Supported", SUPPORTED_EXTENSIONS); + if (p->options && p->options->addsipheaders && p->owner) { + struct ast_channel *chan = p->owner; /* The owner channel */ + struct varshead *headp; + + ast_channel_lock(chan); + + headp = &chan->varshead; + + if (!headp) + ast_log(LOG_WARNING,"No Headp for the channel...ooops!\n"); + else { + const struct ast_var_t *current; + AST_LIST_TRAVERSE(headp, current, entries) { + /* SIPADDHEADER: Add SIP header to outgoing call */ + if (!strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) { + char *content, *end; + const char *header = ast_var_value(current); + char *headdup = ast_strdupa(header); + + /* Strip of the starting " (if it's there) */ + if (*headdup == '"') + headdup++; + if ((content = strchr(headdup, ':'))) { + *content++ = '\0'; + content = ast_skip_blanks(content); /* Skip white space */ + /* Strip the ending " (if it's there) */ + end = content + strlen(content) -1; + if (*end == '"') + *end = '\0'; + + add_header(&req, headdup, content); + if (sipdebug) + ast_log(LOG_DEBUG, "Adding SIP Header \"%s\" with content :%s: \n", headdup, content); + } + } + } + } + + ast_channel_unlock(chan); + } + if (sdp) { + if (p->udptl && (p->t38.state == T38_LOCAL_DIRECT || p->t38.state == T38_LOCAL_REINVITE)) { + ast_udptl_offered_from_local(p->udptl, 1); + if (option_debug) + ast_log(LOG_DEBUG, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + add_t38_sdp(&req, p); + } else if (p->rtp) + add_sdp(&req, p); + } else { + add_header_contentLength(&req, 0); + } + + if (!p->initreq.headers || init > 2) + initialize_initreq(p, &req); + p->lastinvite = p->ocseq; + return send_request(p, &req, init ? XMIT_CRITICAL : XMIT_RELIABLE, p->ocseq); +} + +/*! \brief Used in the SUBSCRIBE notification subsystem */ +static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout) +{ + char tmp[4000], from[256], to[256]; + char *t = tmp, *c, *mfrom, *mto; + size_t maxbytes = sizeof(tmp); + struct sip_request req; + char hint[AST_MAX_EXTENSION]; + char *statestring = "terminated"; + const struct cfsubscription_types *subscriptiontype; + enum state { NOTIFY_OPEN, NOTIFY_INUSE, NOTIFY_CLOSED } local_state = NOTIFY_OPEN; + char *pidfstate = "--"; + char *pidfnote= "Ready"; + + memset(from, 0, sizeof(from)); + memset(to, 0, sizeof(to)); + memset(tmp, 0, sizeof(tmp)); + + switch (state) { + case (AST_EXTENSION_RINGING | AST_EXTENSION_INUSE): + statestring = (global_notifyringing) ? "early" : "confirmed"; + local_state = NOTIFY_INUSE; + pidfstate = "busy"; + pidfnote = "Ringing"; + break; + case AST_EXTENSION_RINGING: + statestring = "early"; + local_state = NOTIFY_INUSE; + pidfstate = "busy"; + pidfnote = "Ringing"; + break; + case AST_EXTENSION_INUSE: + statestring = "confirmed"; + local_state = NOTIFY_INUSE; + pidfstate = "busy"; + pidfnote = "On the phone"; + break; + case AST_EXTENSION_BUSY: + statestring = "confirmed"; + local_state = NOTIFY_CLOSED; + pidfstate = "busy"; + pidfnote = "On the phone"; + break; + case AST_EXTENSION_UNAVAILABLE: + statestring = "terminated"; + local_state = NOTIFY_CLOSED; + pidfstate = "away"; + pidfnote = "Unavailable"; + break; + case AST_EXTENSION_ONHOLD: + statestring = "confirmed"; + local_state = NOTIFY_CLOSED; + pidfstate = "busy"; + pidfnote = "On Hold"; + break; + case AST_EXTENSION_NOT_INUSE: + default: + /* Default setting */ + break; + } + + subscriptiontype = find_subscription_type(p->subscribed); + + /* Check which device/devices we are watching and if they are registered */ + if (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, p->context, p->exten)) { + char *hint2 = hint, *individual_hint = NULL; + int hint_count = 0, unavailable_count = 0; + + while ((individual_hint = strsep(&hint2, "&"))) { + hint_count++; + + if (ast_device_state(individual_hint) == AST_DEVICE_UNAVAILABLE) + unavailable_count++; + } + + /* If none of the hinted devices are registered, we will + * override notification and show no availability. + */ + if (hint_count > 0 && hint_count == unavailable_count) { + local_state = NOTIFY_CLOSED; + pidfstate = "away"; + pidfnote = "Not online"; + } + } + + ast_copy_string(from, get_header(&p->initreq, "From"), sizeof(from)); + c = get_in_brackets(from); + if (strncasecmp(c, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c); + return -1; + } + mfrom = strsep(&c, ";"); /* trim ; and beyond */ + + ast_copy_string(to, get_header(&p->initreq, "To"), sizeof(to)); + c = get_in_brackets(to); + if (strncasecmp(c, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c); + return -1; + } + mto = strsep(&c, ";"); /* trim ; and beyond */ + + reqprep(&req, p, SIP_NOTIFY, 0, 1); + + + add_header(&req, "Event", subscriptiontype->event); + add_header(&req, "Content-Type", subscriptiontype->mediatype); + switch(state) { + case AST_EXTENSION_DEACTIVATED: + if (timeout) + add_header(&req, "Subscription-State", "terminated;reason=timeout"); + else { + add_header(&req, "Subscription-State", "terminated;reason=probation"); + add_header(&req, "Retry-After", "60"); + } + break; + case AST_EXTENSION_REMOVED: + add_header(&req, "Subscription-State", "terminated;reason=noresource"); + break; + default: + if (p->expiry) + add_header(&req, "Subscription-State", "active"); + else /* Expired */ + add_header(&req, "Subscription-State", "terminated;reason=timeout"); + } + switch (p->subscribed) { + case XPIDF_XML: + case CPIM_PIDF_XML: + ast_build_string(&t, &maxbytes, "<?xml version=\"1.0\"?>\n"); + ast_build_string(&t, &maxbytes, "<!DOCTYPE presence PUBLIC \"-//IETF//DTD RFCxxxx XPIDF 1.0//EN\" \"xpidf.dtd\">\n"); + ast_build_string(&t, &maxbytes, "<presence>\n"); + ast_build_string(&t, &maxbytes, "<presentity uri=\"%s;method=SUBSCRIBE\" />\n", mfrom); + ast_build_string(&t, &maxbytes, "<atom id=\"%s\">\n", p->exten); + ast_build_string(&t, &maxbytes, "<address uri=\"%s;user=ip\" priority=\"0.800000\">\n", mto); + ast_build_string(&t, &maxbytes, "<status status=\"%s\" />\n", (local_state == NOTIFY_OPEN) ? "open" : (local_state == NOTIFY_INUSE) ? "inuse" : "closed"); + ast_build_string(&t, &maxbytes, "<msnsubstatus substatus=\"%s\" />\n", (local_state == NOTIFY_OPEN) ? "online" : (local_state == NOTIFY_INUSE) ? "onthephone" : "offline"); + ast_build_string(&t, &maxbytes, "</address>\n</atom>\n</presence>\n"); + break; + case PIDF_XML: /* Eyebeam supports this format */ + ast_build_string(&t, &maxbytes, "<?xml version=\"1.0\" encoding=\"ISO-8859-1\"?>\n"); + ast_build_string(&t, &maxbytes, "<presence xmlns=\"urn:ietf:params:xml:ns:pidf\" \nxmlns:pp=\"urn:ietf:params:xml:ns:pidf:person\"\nxmlns:es=\"urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status\"\nxmlns:ep=\"urn:ietf:params:xml:ns:pidf:rpid:rpid-person\"\nentity=\"%s\">\n", mfrom); + ast_build_string(&t, &maxbytes, "<pp:person><status>\n"); + if (pidfstate[0] != '-') + ast_build_string(&t, &maxbytes, "<ep:activities><ep:%s/></ep:activities>\n", pidfstate); + ast_build_string(&t, &maxbytes, "</status></pp:person>\n"); + ast_build_string(&t, &maxbytes, "<note>%s</note>\n", pidfnote); /* Note */ + ast_build_string(&t, &maxbytes, "<tuple id=\"%s\">\n", p->exten); /* Tuple start */ + ast_build_string(&t, &maxbytes, "<contact priority=\"1\">%s</contact>\n", mto); + if (pidfstate[0] == 'b') /* Busy? Still open ... */ + ast_build_string(&t, &maxbytes, "<status><basic>open</basic></status>\n"); + else + ast_build_string(&t, &maxbytes, "<status><basic>%s</basic></status>\n", (local_state != NOTIFY_CLOSED) ? "open" : "closed"); + ast_build_string(&t, &maxbytes, "</tuple>\n</presence>\n"); + break; + case DIALOG_INFO_XML: /* SNOM subscribes in this format */ + ast_build_string(&t, &maxbytes, "<?xml version=\"1.0\"?>\n"); + ast_build_string(&t, &maxbytes, "<dialog-info xmlns=\"urn:ietf:params:xml:ns:dialog-info\" version=\"%d\" state=\"%s\" entity=\"%s\">\n", p->dialogver++, full ? "full":"partial", mto); + if ((state & AST_EXTENSION_RINGING) && global_notifyringing) + ast_build_string(&t, &maxbytes, "<dialog id=\"%s\" direction=\"recipient\">\n", p->exten); + else + ast_build_string(&t, &maxbytes, "<dialog id=\"%s\">\n", p->exten); + ast_build_string(&t, &maxbytes, "<state>%s</state>\n", statestring); + if (state == AST_EXTENSION_ONHOLD) { + ast_build_string(&t, &maxbytes, "<local>\n<target uri=\"%s\">\n" + "<param pname=\"+sip.rendering\" pvalue=\"no\"/>\n" + "</target>\n</local>\n", mto); + } + ast_build_string(&t, &maxbytes, "</dialog>\n</dialog-info>\n"); + break; + case NONE: + default: + break; + } + + if (t > tmp + sizeof(tmp)) + ast_log(LOG_WARNING, "Buffer overflow detected!! (Please file a bug report)\n"); + + add_header_contentLength(&req, strlen(tmp)); + add_line(&req, tmp); + p->pendinginvite = p->ocseq; /* Remember that we have a pending NOTIFY in order not to confuse the NOTIFY subsystem */ + + return send_request(p, &req, XMIT_RELIABLE, p->ocseq); +} + +/*! \brief Notify user of messages waiting in voicemail +\note - Notification only works for registered peers with mailbox= definitions + in sip.conf + - We use the SIP Event package message-summary + MIME type defaults to "application/simple-message-summary"; + */ +static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten) +{ + struct sip_request req; + char tmp[500]; + char *t = tmp; + size_t maxbytes = sizeof(tmp); + + initreqprep(&req, p, SIP_NOTIFY); + add_header(&req, "Event", "message-summary"); + add_header(&req, "Content-Type", default_notifymime); + + ast_build_string(&t, &maxbytes, "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no"); + ast_build_string(&t, &maxbytes, "Message-Account: sip:%s@%s\r\n", + S_OR(vmexten, default_vmexten), S_OR(p->fromdomain, ast_inet_ntoa(p->ourip))); + /* Cisco has a bug in the SIP stack where it can't accept the + (0/0) notification. This can temporarily be disabled in + sip.conf with the "buggymwi" option */ + ast_build_string(&t, &maxbytes, "Voice-Message: %d/%d%s\r\n", newmsgs, oldmsgs, (ast_test_flag(&p->flags[1], SIP_PAGE2_BUGGY_MWI) ? "" : " (0/0)")); + + if (p->subscribed) { + if (p->expiry) + add_header(&req, "Subscription-State", "active"); + else /* Expired */ + add_header(&req, "Subscription-State", "terminated;reason=timeout"); + } + + if (t > tmp + sizeof(tmp)) + ast_log(LOG_WARNING, "Buffer overflow detected!! (Please file a bug report)\n"); + + add_header_contentLength(&req, strlen(tmp)); + add_line(&req, tmp); + + if (!p->initreq.headers) + initialize_initreq(p, &req); + return send_request(p, &req, XMIT_RELIABLE, p->ocseq); +} + +/*! \brief Transmit SIP request unreliably (only used in sip_notify subsystem) */ +static int transmit_sip_request(struct sip_pvt *p, struct sip_request *req) +{ + if (!p->initreq.headers) /* Initialize first request before sending */ + initialize_initreq(p, req); + return send_request(p, req, XMIT_UNRELIABLE, p->ocseq); +} + +/*! \brief Notify a transferring party of the status of transfer */ +static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate) +{ + struct sip_request req; + char tmp[SIPBUFSIZE/2]; + + reqprep(&req, p, SIP_NOTIFY, 0, 1); + snprintf(tmp, sizeof(tmp), "refer;id=%d", cseq); + add_header(&req, "Event", tmp); + add_header(&req, "Subscription-state", terminate ? "terminated;reason=noresource" : "active"); + add_header(&req, "Content-Type", "message/sipfrag;version=2.0"); + add_header(&req, "Allow", ALLOWED_METHODS); + add_header(&req, "Supported", SUPPORTED_EXTENSIONS); + + snprintf(tmp, sizeof(tmp), "SIP/2.0 %s\r\n", message); + add_header_contentLength(&req, strlen(tmp)); + add_line(&req, tmp); + + if (!p->initreq.headers) + initialize_initreq(p, &req); + + p->lastnoninvite = p->ocseq; + + return send_request(p, &req, XMIT_RELIABLE, p->ocseq); +} + +/*! \brief Convert registration state status to string */ +static char *regstate2str(enum sipregistrystate regstate) +{ + switch(regstate) { + case REG_STATE_FAILED: + return "Failed"; + case REG_STATE_UNREGISTERED: + return "Unregistered"; + case REG_STATE_REGSENT: + return "Request Sent"; + case REG_STATE_AUTHSENT: + return "Auth. Sent"; + case REG_STATE_REGISTERED: + return "Registered"; + case REG_STATE_REJECTED: + return "Rejected"; + case REG_STATE_TIMEOUT: + return "Timeout"; + case REG_STATE_NOAUTH: + return "No Authentication"; + default: + return "Unknown"; + } +} + +/*! \brief Update registration with SIP Proxy */ +static int sip_reregister(const void *data) +{ + /* if we are here, we know that we need to reregister. */ + struct sip_registry *r= ASTOBJ_REF((struct sip_registry *) data); + + /* if we couldn't get a reference to the registry object, punt */ + if (!r) + return 0; + + if (r->call && !ast_test_flag(&r->call->flags[0], SIP_NO_HISTORY)) + append_history(r->call, "RegistryRenew", "Account: %s@%s", r->username, r->hostname); + /* Since registry's are only added/removed by the the monitor thread, this + may be overkill to reference/dereference at all here */ + if (sipdebug) + ast_log(LOG_NOTICE, " -- Re-registration for %s@%s\n", r->username, r->hostname); + + r->expire = -1; + __sip_do_register(r); + ASTOBJ_UNREF(r, sip_registry_destroy); + return 0; +} + +/*! \brief Register with SIP proxy */ +static int __sip_do_register(struct sip_registry *r) +{ + int res; + + res = transmit_register(r, SIP_REGISTER, NULL, NULL); + return res; +} + +/*! \brief Registration timeout, register again */ +static int sip_reg_timeout(const void *data) +{ + + /* if we are here, our registration timed out, so we'll just do it over */ + struct sip_registry *r = ASTOBJ_REF((struct sip_registry *) data); + struct sip_pvt *p; + int res; + + /* if we couldn't get a reference to the registry object, punt */ + if (!r) + return 0; + + ast_log(LOG_NOTICE, " -- Registration for '%s@%s' timed out, trying again (Attempt #%d)\n", r->username, r->hostname, r->regattempts); + if (r->call) { + /* Unlink us, destroy old call. Locking is not relevant here because all this happens + in the single SIP manager thread. */ + p = r->call; + ast_mutex_lock(&p->lock); + if (p->registry) + ASTOBJ_UNREF(p->registry, sip_registry_destroy); + r->call = NULL; + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + /* Pretend to ACK anything just in case */ + __sip_pretend_ack(p); + ast_mutex_unlock(&p->lock); + } + /* If we have a limit, stop registration and give up */ + if (global_regattempts_max && (r->regattempts > global_regattempts_max)) { + /* Ok, enough is enough. Don't try any more */ + /* We could add an external notification here... + steal it from app_voicemail :-) */ + ast_log(LOG_NOTICE, " -- Giving up forever trying to register '%s@%s'\n", r->username, r->hostname); + r->regstate = REG_STATE_FAILED; + } else { + r->regstate = REG_STATE_UNREGISTERED; + r->timeout = -1; + res=transmit_register(r, SIP_REGISTER, NULL, NULL); + } + manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelDriver: SIP\r\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate)); + ASTOBJ_UNREF(r, sip_registry_destroy); + return 0; +} + +/*! \brief Transmit register to SIP proxy or UA */ +static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader) +{ + struct sip_request req; + char from[256]; + char to[256]; + char tmp[80]; + char addr[80]; + struct sip_pvt *p; + char *fromdomain; + + /* exit if we are already in process with this registrar ?*/ + if ( r == NULL || ((auth==NULL) && (r->regstate==REG_STATE_REGSENT || r->regstate==REG_STATE_AUTHSENT))) { + if (r) { + ast_log(LOG_NOTICE, "Strange, trying to register %s@%s when registration already pending\n", r->username, r->hostname); + } + return 0; + } + + if (r->call) { /* We have a registration */ + if (!auth) { + ast_log(LOG_WARNING, "Already have a REGISTER going on to %s@%s?? \n", r->username, r->hostname); + return 0; + } else { + p = r->call; + make_our_tag(p->tag, sizeof(p->tag)); /* create a new local tag for every register attempt */ + ast_string_field_free(p, theirtag); /* forget their old tag, so we don't match tags when getting response */ + } + } else { + /* Build callid for registration if we haven't registered before */ + if (!r->callid_valid) { + build_callid_registry(r, __ourip, default_fromdomain); + r->callid_valid = TRUE; + } + /* Allocate SIP packet for registration */ + if (!(p = sip_alloc( r->callid, NULL, 0, SIP_REGISTER))) { + ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n"); + return 0; + } + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) + append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname); + /* Find address to hostname */ + if (create_addr(p, r->hostname)) { + /* we have what we hope is a temporary network error, + * probably DNS. We need to reschedule a registration try */ + sip_destroy(p); + + if (r->timeout > -1) + ast_log(LOG_WARNING, "Still have a registration timeout for %s@%s (create_addr() error), %d\n", r->username, r->hostname, r->timeout); + else + ast_log(LOG_WARNING, "Probably a DNS error for registration to %s@%s, trying REGISTER again (after %d seconds)\n", r->username, r->hostname, global_reg_timeout); + + AST_SCHED_DEL(sched, r->timeout); + r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, r); + r->regattempts++; + return 0; + } + /* Copy back Call-ID in case create_addr changed it */ + ast_string_field_set(r, callid, p->callid); + if (r->portno) { + p->sa.sin_port = htons(r->portno); + p->recv.sin_port = htons(r->portno); + } else /* Set registry port to the port set from the peer definition/srv or default */ + r->portno = ntohs(p->sa.sin_port); + ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Registration is outgoing call */ + r->call=p; /* Save pointer to SIP packet */ + p->registry = ASTOBJ_REF(r); /* Add pointer to registry in packet */ + if (!ast_strlen_zero(r->secret)) /* Secret (password) */ + ast_string_field_set(p, peersecret, r->secret); + if (!ast_strlen_zero(r->md5secret)) + ast_string_field_set(p, peermd5secret, r->md5secret); + /* User name in this realm + - if authuser is set, use that, otherwise use username */ + if (!ast_strlen_zero(r->authuser)) { + ast_string_field_set(p, peername, r->authuser); + ast_string_field_set(p, authname, r->authuser); + } else if (!ast_strlen_zero(r->username)) { + ast_string_field_set(p, peername, r->username); + ast_string_field_set(p, authname, r->username); + ast_string_field_set(p, fromuser, r->username); + } + if (!ast_strlen_zero(r->username)) + ast_string_field_set(p, username, r->username); + /* Save extension in packet */ + ast_string_field_set(p, exten, r->contact); + + /* + check which address we should use in our contact header + based on whether the remote host is on the external or + internal network so we can register through nat + */ + if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip)) + p->ourip = bindaddr.sin_addr; + build_contact(p); + } + + /* set up a timeout */ + if (auth == NULL) { + if (r->timeout > -1) + ast_log(LOG_WARNING, "Still have a registration timeout, #%d - deleting it\n", r->timeout); + AST_SCHED_DEL(sched, r->timeout); + r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, r); + if (option_debug) + ast_log(LOG_DEBUG, "Scheduled a registration timeout for %s id #%d \n", r->hostname, r->timeout); + } + + if ((fromdomain = strchr(r->username, '@'))) { + /* the domain name is just behind '@' */ + fromdomain++ ; + /* We have a domain in the username for registration */ + snprintf(from, sizeof(from), "<sip:%s>;tag=%s", r->username, p->tag); + if (!ast_strlen_zero(p->theirtag)) + snprintf(to, sizeof(to), "<sip:%s>;tag=%s", r->username, p->theirtag); + else + snprintf(to, sizeof(to), "<sip:%s>", r->username); + + /* If the registration username contains '@', then the domain should be used as + the equivalent of "fromdomain" for the registration */ + if (ast_strlen_zero(p->fromdomain)) { + ast_string_field_set(p, fromdomain, fromdomain); + } + } else { + snprintf(from, sizeof(from), "<sip:%s@%s>;tag=%s", r->username, p->tohost, p->tag); + if (!ast_strlen_zero(p->theirtag)) + snprintf(to, sizeof(to), "<sip:%s@%s>;tag=%s", r->username, p->tohost, p->theirtag); + else + snprintf(to, sizeof(to), "<sip:%s@%s>", r->username, p->tohost); + } + + /* Fromdomain is what we are registering to, regardless of actual + host name from SRV */ + if (!ast_strlen_zero(p->fromdomain)) { + if (r->portno && r->portno != STANDARD_SIP_PORT) + snprintf(addr, sizeof(addr), "sip:%s:%d", p->fromdomain, r->portno); + else + snprintf(addr, sizeof(addr), "sip:%s", p->fromdomain); + } else { + if (r->portno && r->portno != STANDARD_SIP_PORT) + snprintf(addr, sizeof(addr), "sip:%s:%d", r->hostname, r->portno); + else + snprintf(addr, sizeof(addr), "sip:%s", r->hostname); + } + ast_string_field_set(p, uri, addr); + + p->branch ^= ast_random(); + + init_req(&req, sipmethod, addr); + + /* Add to CSEQ */ + snprintf(tmp, sizeof(tmp), "%u %s", ++r->ocseq, sip_methods[sipmethod].text); + p->ocseq = r->ocseq; + + build_via(p); + add_header(&req, "Via", p->via); + add_header(&req, "From", from); + add_header(&req, "To", to); + add_header(&req, "Call-ID", p->callid); + add_header(&req, "CSeq", tmp); + if (!ast_strlen_zero(global_useragent)) + add_header(&req, "User-Agent", global_useragent); + add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS); + + + if (auth) /* Add auth header */ + add_header(&req, authheader, auth); + else if (!ast_strlen_zero(r->nonce)) { + char digest[1024]; + + /* We have auth data to reuse, build a digest header! */ + if (sipdebug) + ast_log(LOG_DEBUG, " >>> Re-using Auth data for %s@%s\n", r->username, r->hostname); + ast_string_field_set(p, realm, r->realm); + ast_string_field_set(p, nonce, r->nonce); + ast_string_field_set(p, domain, r->domain); + ast_string_field_set(p, opaque, r->opaque); + ast_string_field_set(p, qop, r->qop); + r->noncecount++; + p->noncecount = r->noncecount; + + memset(digest,0,sizeof(digest)); + if(!build_reply_digest(p, sipmethod, digest, sizeof(digest))) + add_header(&req, "Authorization", digest); + else + ast_log(LOG_NOTICE, "No authorization available for authentication of registration to %s@%s\n", r->username, r->hostname); + + } + + snprintf(tmp, sizeof(tmp), "%d", default_expiry); + add_header(&req, "Expires", tmp); + add_header(&req, "Contact", p->our_contact); + add_header(&req, "Event", "registration"); + add_header_contentLength(&req, 0); + + initialize_initreq(p, &req); + if (sip_debug_test_pvt(p)) + ast_verbose("REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines); + r->regstate = auth ? REG_STATE_AUTHSENT : REG_STATE_REGSENT; + r->regattempts++; /* Another attempt */ + if (option_debug > 3) + ast_verbose("REGISTER attempt %d to %s@%s\n", r->regattempts, r->username, r->hostname); + return send_request(p, &req, XMIT_CRITICAL, p->ocseq); +} + +/*! \brief Transmit text with SIP MESSAGE method */ +static int transmit_message_with_text(struct sip_pvt *p, const char *text) +{ + struct sip_request req; + + reqprep(&req, p, SIP_MESSAGE, 0, 1); + add_text(&req, text); + return send_request(p, &req, XMIT_RELIABLE, p->ocseq); +} + +/*! \brief Allocate SIP refer structure */ +static int sip_refer_allocate(struct sip_pvt *p) +{ + p->refer = ast_calloc(1, sizeof(struct sip_refer)); + return p->refer ? 1 : 0; +} + +/*! \brief Transmit SIP REFER message (initiated by the transfer() dialplan application + \note this is currently broken as we have no way of telling the dialplan + engine whether a transfer succeeds or fails. + \todo Fix the transfer() dialplan function so that a transfer may fail +*/ +static int transmit_refer(struct sip_pvt *p, const char *dest) +{ + struct sip_request req = { + .headers = 0, + }; + char from[256]; + const char *of; + char *c; + char referto[256]; + char *ttag, *ftag; + char *theirtag = ast_strdupa(p->theirtag); + + if (option_debug || sipdebug) + ast_log(LOG_DEBUG, "SIP transfer of %s to %s\n", p->callid, dest); + + /* Are we transfering an inbound or outbound call ? */ + if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) { + of = get_header(&p->initreq, "To"); + ttag = theirtag; + ftag = p->tag; + } else { + of = get_header(&p->initreq, "From"); + ftag = theirtag; + ttag = p->tag; + } + + ast_copy_string(from, of, sizeof(from)); + of = get_in_brackets(from); + ast_string_field_set(p, from, of); + if (strncasecmp(of, "sip:", 4)) + ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n"); + else + of += 4; + /* Get just the username part */ + if ((c = strchr(dest, '@'))) + c = NULL; + else if ((c = strchr(of, '@'))) + *c++ = '\0'; + if (c) + snprintf(referto, sizeof(referto), "<sip:%s@%s>", dest, c); + else + snprintf(referto, sizeof(referto), "<sip:%s>", dest); + + /* save in case we get 407 challenge */ + sip_refer_allocate(p); + ast_copy_string(p->refer->refer_to, referto, sizeof(p->refer->refer_to)); + ast_copy_string(p->refer->referred_by, p->our_contact, sizeof(p->refer->referred_by)); + p->refer->status = REFER_SENT; /* Set refer status */ + + reqprep(&req, p, SIP_REFER, 0, 1); + + add_header(&req, "Refer-To", referto); + add_header(&req, "Allow", ALLOWED_METHODS); + add_header(&req, "Supported", SUPPORTED_EXTENSIONS); + if (!ast_strlen_zero(p->our_contact)) + add_header(&req, "Referred-By", p->our_contact); + + return send_request(p, &req, XMIT_RELIABLE, p->ocseq); + /* We should propably wait for a NOTIFY here until we ack the transfer */ + /* Maybe fork a new thread and wait for a STATUS of REFER_200OK on the refer status before returning to app_transfer */ + + /*! \todo In theory, we should hang around and wait for a reply, before + returning to the dial plan here. Don't know really how that would + affect the transfer() app or the pbx, but, well, to make this + useful we should have a STATUS code on transfer(). + */ +} + + +/*! \brief Send SIP INFO dtmf message, see Cisco documentation on cisco.com */ +static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration) +{ + struct sip_request req; + + reqprep(&req, p, SIP_INFO, 0, 1); + add_digit(&req, digit, duration); + return send_request(p, &req, XMIT_RELIABLE, p->ocseq); +} + +/*! \brief Send SIP INFO with video update request */ +static int transmit_info_with_vidupdate(struct sip_pvt *p) +{ + struct sip_request req; + + reqprep(&req, p, SIP_INFO, 0, 1); + add_vidupdate(&req); + return send_request(p, &req, XMIT_RELIABLE, p->ocseq); +} + +/*! \brief Transmit generic SIP request + returns XMIT_ERROR if transmit failed with a critical error (don't retry) +*/ +static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch) +{ + struct sip_request resp; + + if (sipmethod == SIP_ACK) + p->invitestate = INV_CONFIRMED; + + reqprep(&resp, p, sipmethod, seqno, newbranch); + add_header_contentLength(&resp, 0); + return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq); +} + +/*! \brief Transmit SIP request, auth added */ +static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch) +{ + struct sip_request resp; + + reqprep(&resp, p, sipmethod, seqno, newbranch); + if (!ast_strlen_zero(p->realm)) { + char digest[1024]; + + memset(digest, 0, sizeof(digest)); + if(!build_reply_digest(p, sipmethod, digest, sizeof(digest))) { + if (p->options && p->options->auth_type == PROXY_AUTH) + add_header(&resp, "Proxy-Authorization", digest); + else if (p->options && p->options->auth_type == WWW_AUTH) + add_header(&resp, "Authorization", digest); + else /* Default, to be backwards compatible (maybe being too careful, but leaving it for now) */ + add_header(&resp, "Proxy-Authorization", digest); + } else + ast_log(LOG_WARNING, "No authentication available for call %s\n", p->callid); + } + /* If we are hanging up and know a cause for that, send it in clear text to make + debugging easier. */ + if (sipmethod == SIP_BYE && p->owner && p->owner->hangupcause) { + char buf[10]; + + add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause)); + snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause); + add_header(&resp, "X-Asterisk-HangupCauseCode", buf); + } + + add_header_contentLength(&resp, 0); + return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq); +} + +/*! \brief Remove registration data from realtime database or AST/DB when registration expires */ +static void destroy_association(struct sip_peer *peer) +{ + if (!ast_test_flag(&global_flags[1], SIP_PAGE2_IGNOREREGEXPIRE)) { + if (ast_test_flag(&peer->flags[1], SIP_PAGE2_RT_FROMCONTACT)) + ast_update_realtime("sippeers", "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", "username", "", "regserver", "", NULL); + else + ast_db_del("SIP/Registry", peer->name); + } +} + +/*! \brief Expire registration of SIP peer */ +static int expire_register(const void *data) +{ + struct sip_peer *peer = (struct sip_peer *)data; + + if (!peer) /* Hmmm. We have no peer. Weird. */ + return 0; + + memset(&peer->addr, 0, sizeof(peer->addr)); + + destroy_association(peer); /* remove registration data from storage */ + + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name); + register_peer_exten(peer, FALSE); /* Remove regexten */ + peer->expire = -1; + ast_device_state_changed("SIP/%s", peer->name); + + /* Do we need to release this peer from memory? + Only for realtime peers and autocreated peers + */ + if (ast_test_flag(&peer->flags[1], SIP_PAGE2_SELFDESTRUCT) || + ast_test_flag(&peer->flags[1], SIP_PAGE2_RTAUTOCLEAR)) { + struct sip_peer *peer_ptr = peer_ptr; + peer_ptr = ASTOBJ_CONTAINER_UNLINK(&peerl, peer); + if (peer_ptr) { + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + } + + ASTOBJ_UNREF(peer, sip_destroy_peer); + + return 0; +} + +/*! \brief Poke peer (send qualify to check if peer is alive and well) */ +static int sip_poke_peer_s(const void *data) +{ + struct sip_peer *peer = (struct sip_peer *) data; + + peer->pokeexpire = -1; + + sip_poke_peer(peer); + + ASTOBJ_UNREF(peer, sip_destroy_peer); + + return 0; +} + +/*! \brief Get registration details from Asterisk DB */ +static void reg_source_db(struct sip_peer *peer) +{ + char data[256]; + struct in_addr in; + int expiry; + int port; + char *scan, *addr, *port_str, *expiry_str, *username, *contact; + + if (ast_test_flag(&peer->flags[1], SIP_PAGE2_RT_FROMCONTACT)) + return; + if (ast_db_get("SIP/Registry", peer->name, data, sizeof(data))) + return; + + scan = data; + addr = strsep(&scan, ":"); + port_str = strsep(&scan, ":"); + expiry_str = strsep(&scan, ":"); + username = strsep(&scan, ":"); + contact = scan; /* Contact include sip: and has to be the last part of the database entry as long as we use : as a separator */ + + if (!inet_aton(addr, &in)) + return; + + if (port_str) + port = atoi(port_str); + else + return; + + if (expiry_str) + expiry = atoi(expiry_str); + else + return; + + if (username) + ast_copy_string(peer->username, username, sizeof(peer->username)); + if (contact) + ast_copy_string(peer->fullcontact, contact, sizeof(peer->fullcontact)); + + if (option_debug > 1) + ast_log(LOG_DEBUG, "SIP Seeding peer from astdb: '%s' at %s@%s:%d for %d\n", + peer->name, peer->username, ast_inet_ntoa(in), port, expiry); + + memset(&peer->addr, 0, sizeof(peer->addr)); + peer->addr.sin_family = AF_INET; + peer->addr.sin_addr = in; + peer->addr.sin_port = htons(port); + if (sipsock < 0) { + /* SIP isn't up yet, so schedule a poke only, pretty soon */ + if (!AST_SCHED_DEL(sched, peer->pokeexpire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + peer->pokeexpire = ast_sched_add(sched, ast_random() % 5000 + 1, sip_poke_peer_s, ASTOBJ_REF(peer)); + if (peer->pokeexpire == -1) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + } else + sip_poke_peer(peer); + if (!AST_SCHED_DEL(sched, peer->expire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + peer->expire = ast_sched_add(sched, (expiry + 10) * 1000, expire_register, ASTOBJ_REF(peer)); + if (peer->expire == -1) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + register_peer_exten(peer, TRUE); +} + +/*! \brief Save contact header for 200 OK on INVITE */ +static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req) +{ + char contact[SIPBUFSIZE]; + char *c; + + /* Look for brackets */ + ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact)); + c = get_in_brackets(contact); + + /* Save full contact to call pvt for later bye or re-invite */ + ast_string_field_set(pvt, fullcontact, c); + + /* Save URI for later ACKs, BYE or RE-invites */ + ast_string_field_set(pvt, okcontacturi, c); + + /* We should return false for URI:s we can't handle, + like sips:, tel:, mailto:,ldap: etc */ + return TRUE; +} + +static int __set_address_from_contact(const char *fullcontact, struct sockaddr_in *sin) +{ + struct hostent *hp; + struct ast_hostent ahp; + int port; + char *c, *host, *pt; + char contact_buf[256]; + char *contact; + + /* Work on a copy */ + ast_copy_string(contact_buf, fullcontact, sizeof(contact_buf)); + contact = contact_buf; + + /* Make sure it's a SIP URL */ + if (strncasecmp(contact, "sip:", 4)) { + ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact); + } else + contact += 4; + + /* Ditch arguments */ + /* XXX this code is replicated also shortly below */ + + /* Grab host */ + host = strchr(contact, '@'); + if (!host) { /* No username part */ + host = contact; + c = NULL; + } else { + *host++ = '\0'; + } + pt = strchr(host, ':'); + if (pt) { + *pt++ = '\0'; + port = atoi(pt); + } else + port = STANDARD_SIP_PORT; + + contact = strsep(&contact, ";"); /* trim ; and beyond in username part */ + host = strsep(&host, ";"); /* trim ; and beyond in host/domain part */ + + /* XXX This could block for a long time XXX */ + /* We should only do this if it's a name, not an IP */ + hp = ast_gethostbyname(host, &ahp); + if (!hp) { + ast_log(LOG_WARNING, "Invalid host name in Contact: (can't resolve in DNS) : '%s'\n", host); + return -1; + } + sin->sin_family = AF_INET; + memcpy(&sin->sin_addr, hp->h_addr, sizeof(sin->sin_addr)); + sin->sin_port = htons(port); + + return 0; +} + +/*! \brief Change the other partys IP address based on given contact */ +static int set_address_from_contact(struct sip_pvt *pvt) +{ + if (ast_test_flag(&pvt->flags[0], SIP_NAT_ROUTE)) { + /* NAT: Don't trust the contact field. Just use what they came to us + with. */ + pvt->sa = pvt->recv; + return 0; + } + + return __set_address_from_contact(pvt->fullcontact, &pvt->sa); +} + + +/*! \brief Parse contact header and save registration (peer registration) */ +static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req) +{ + char contact[SIPBUFSIZE]; + char data[SIPBUFSIZE]; + const char *expires = get_header(req, "Expires"); + int expiry = atoi(expires); + char *curi, *n, *pt; + int port; + const char *useragent; + struct hostent *hp; + struct ast_hostent ahp; + struct sockaddr_in oldsin, testsin; + + ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact)); + + if (ast_strlen_zero(expires)) { /* No expires header */ + expires = strcasestr(contact, ";expires="); + if (expires) { + /* XXX bug here, we overwrite the string */ + expires = strsep((char **) &expires, ";"); /* trim ; and beyond */ + if (sscanf(expires + 9, "%d", &expiry) != 1) + expiry = default_expiry; + } else { + /* Nothing has been specified */ + expiry = default_expiry; + } + } + + /* Look for brackets */ + curi = contact; + if (strchr(contact, '<') == NULL) /* No <, check for ; and strip it */ + strsep(&curi, ";"); /* This is Header options, not URI options */ + curi = get_in_brackets(contact); + + /* if they did not specify Contact: or Expires:, they are querying + what we currently have stored as their contact address, so return + it + */ + if (ast_strlen_zero(curi) && ast_strlen_zero(expires)) { + /* If we have an active registration, tell them when the registration is going to expire */ + if (peer->expire > -1 && !ast_strlen_zero(peer->fullcontact)) + pvt->expiry = ast_sched_when(sched, peer->expire); + return PARSE_REGISTER_QUERY; + } else if (!strcasecmp(curi, "*") || !expiry) { /* Unregister this peer */ + /* This means remove all registrations and return OK */ + memset(&peer->addr, 0, sizeof(peer->addr)); + if (!AST_SCHED_DEL(sched, peer->expire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + + destroy_association(peer); + + register_peer_exten(peer, 0); /* Add extension from regexten= setting in sip.conf */ + peer->fullcontact[0] = '\0'; + peer->useragent[0] = '\0'; + peer->sipoptions = 0; + peer->lastms = 0; + pvt->expiry = 0; + + if (option_verbose > 2) + ast_verbose(VERBOSE_PREFIX_3 "Unregistered SIP '%s'\n", peer->name); + + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unregistered\r\n", peer->name); + return PARSE_REGISTER_UPDATE; + } + + /* Store whatever we got as a contact from the client */ + ast_copy_string(peer->fullcontact, curi, sizeof(peer->fullcontact)); + + /* For the 200 OK, we should use the received contact */ + ast_string_field_build(pvt, our_contact, "<%s>", curi); + + /* Make sure it's a SIP URL */ + if (strncasecmp(curi, "sip:", 4)) { + ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", curi); + } else + curi += 4; + /* Ditch q */ + curi = strsep(&curi, ";"); + /* Grab host */ + n = strchr(curi, '@'); + if (!n) { + n = curi; + curi = NULL; + } else + *n++ = '\0'; + pt = strchr(n, ':'); + if (pt) { + *pt++ = '\0'; + port = atoi(pt); + } else + port = STANDARD_SIP_PORT; + oldsin = peer->addr; + + /* Check that they're allowed to register at this IP */ + /* XXX This could block for a long time XXX */ + hp = ast_gethostbyname(n, &ahp); + if (!hp) { + ast_log(LOG_WARNING, "Invalid host '%s'\n", n); + *peer->fullcontact = '\0'; + ast_string_field_set(pvt, our_contact, ""); + return PARSE_REGISTER_FAILED; + } + memcpy(&testsin.sin_addr, hp->h_addr, sizeof(testsin.sin_addr)); + if ( ast_apply_ha(global_contact_ha, &testsin) != AST_SENSE_ALLOW || + ast_apply_ha(peer->contactha, &testsin) != AST_SENSE_ALLOW) { + ast_log(LOG_WARNING, "Host '%s' disallowed by rule\n", n); + *peer->fullcontact = '\0'; + ast_string_field_set(pvt, our_contact, ""); + return PARSE_REGISTER_FAILED; + } + + if (!ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE)) { + peer->addr.sin_family = AF_INET; + memcpy(&peer->addr.sin_addr, hp->h_addr, sizeof(peer->addr.sin_addr)); + peer->addr.sin_port = htons(port); + } else { + /* Don't trust the contact field. Just use what they came to us + with */ + peer->addr = pvt->recv; + } + + /* Save SIP options profile */ + peer->sipoptions = pvt->sipoptions; + + if (curi && ast_strlen_zero(peer->username)) + ast_copy_string(peer->username, curi, sizeof(peer->username)); + + if (!AST_SCHED_DEL(sched, peer->expire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + if (expiry > max_expiry) + expiry = max_expiry; + if (expiry < min_expiry) + expiry = min_expiry; + if (ast_test_flag(&peer->flags[0], SIP_REALTIME) && !ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) { + peer->expire = -1; + } else { + peer->expire = ast_sched_add(sched, (expiry + 10) * 1000, expire_register, ASTOBJ_REF(peer)); + if (peer->expire == -1) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + } + pvt->expiry = expiry; + snprintf(data, sizeof(data), "%s:%d:%d:%s:%s", ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port), expiry, peer->username, peer->fullcontact); + if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_RT_FROMCONTACT)) + ast_db_put("SIP/Registry", peer->name, data); + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Registered\r\n", peer->name); + + /* Is this a new IP address for us? */ + if (option_verbose > 2 && inaddrcmp(&peer->addr, &oldsin)) { + ast_verbose(VERBOSE_PREFIX_3 "Registered SIP '%s' at %s port %d\n", peer->name, ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port)); + } + sip_poke_peer(peer); + register_peer_exten(peer, 1); + + /* Save User agent */ + useragent = get_header(req, "User-Agent"); + if (strcasecmp(useragent, peer->useragent)) { /* XXX copy if they are different ? */ + ast_copy_string(peer->useragent, useragent, sizeof(peer->useragent)); + if (option_verbose > 3) + ast_verbose(VERBOSE_PREFIX_3 "Saved useragent \"%s\" for peer %s\n", peer->useragent, peer->name); + } + return PARSE_REGISTER_UPDATE; +} + +/*! \brief Remove route from route list */ +static void free_old_route(struct sip_route *route) +{ + struct sip_route *next; + + while (route) { + next = route->next; + free(route); + route = next; + } +} + +/*! \brief List all routes - mostly for debugging */ +static void list_route(struct sip_route *route) +{ + if (!route) + ast_verbose("list_route: no route\n"); + else { + for (;route; route = route->next) + ast_verbose("list_route: hop: <%s>\n", route->hop); + } +} + +/*! \brief Build route list from Record-Route header */ +static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards) +{ + struct sip_route *thishop, *head, *tail; + int start = 0; + int len; + const char *rr, *contact, *c; + + /* Once a persistant route is set, don't fool with it */ + if (p->route && p->route_persistant) { + if (option_debug) + ast_log(LOG_DEBUG, "build_route: Retaining previous route: <%s>\n", p->route->hop); + return; + } + + if (p->route) { + free_old_route(p->route); + p->route = NULL; + } + + /* We only want to create the route set the first time this is called */ + p->route_persistant = 1; + + /* Build a tailq, then assign it to p->route when done. + * If backwards, we add entries from the head so they end up + * in reverse order. However, we do need to maintain a correct + * tail pointer because the contact is always at the end. + */ + head = NULL; + tail = head; + /* 1st we pass through all the hops in any Record-Route headers */ + for (;;) { + /* Each Record-Route header */ + rr = __get_header(req, "Record-Route", &start); + if (*rr == '\0') + break; + for (; (rr = strchr(rr, '<')) ; rr += len) { /* Each route entry */ + ++rr; + len = strcspn(rr, ">") + 1; + /* Make a struct route */ + if ((thishop = ast_malloc(sizeof(*thishop) + len))) { + /* ast_calloc is not needed because all fields are initialized in this block */ + ast_copy_string(thishop->hop, rr, len); + if (option_debug > 1) + ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop); + /* Link in */ + if (backwards) { + /* Link in at head so they end up in reverse order */ + thishop->next = head; + head = thishop; + /* If this was the first then it'll be the tail */ + if (!tail) + tail = thishop; + } else { + thishop->next = NULL; + /* Link in at the end */ + if (tail) + tail->next = thishop; + else + head = thishop; + tail = thishop; + } + } + } + } + + /* Only append the contact if we are dealing with a strict router */ + if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) { + /* 2nd append the Contact: if there is one */ + /* Can be multiple Contact headers, comma separated values - we just take the first */ + contact = get_header(req, "Contact"); + if (!ast_strlen_zero(contact)) { + if (option_debug > 1) + ast_log(LOG_DEBUG, "build_route: Contact hop: %s\n", contact); + /* Look for <: delimited address */ + c = strchr(contact, '<'); + if (c) { + /* Take to > */ + ++c; + len = strcspn(c, ">") + 1; + } else { + /* No <> - just take the lot */ + c = contact; + len = strlen(contact) + 1; + } + if ((thishop = ast_malloc(sizeof(*thishop) + len))) { + /* ast_calloc is not needed because all fields are initialized in this block */ + ast_copy_string(thishop->hop, c, len); + thishop->next = NULL; + /* Goes at the end */ + if (tail) + tail->next = thishop; + else + head = thishop; + } + } + } + + /* Store as new route */ + p->route = head; + + /* For debugging dump what we ended up with */ + if (sip_debug_test_pvt(p)) + list_route(p->route); +} + +AST_THREADSTORAGE(check_auth_buf, check_auth_buf_init); +#define CHECK_AUTH_BUF_INITLEN 256 + +/*! \brief Check user authorization from peer definition + Some actions, like REGISTER and INVITEs from peers require + authentication (if peer have secret set) + \return 0 on success, non-zero on error +*/ +static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username, + const char *secret, const char *md5secret, int sipmethod, + char *uri, enum xmittype reliable, int ignore) +{ + const char *response = "407 Proxy Authentication Required"; + const char *reqheader = "Proxy-Authorization"; + const char *respheader = "Proxy-Authenticate"; + const char *authtoken; + char a1_hash[256]; + char resp_hash[256]=""; + char *c; + int wrongnonce = FALSE; + int good_response; + const char *usednonce = p->randdata; + struct ast_dynamic_str *buf; + int res; + + /* table of recognised keywords, and their value in the digest */ + enum keys { K_RESP, K_URI, K_USER, K_NONCE, K_LAST }; + struct x { + const char *key; + const char *s; + } *i, keys[] = { + [K_RESP] = { "response=", "" }, + [K_URI] = { "uri=", "" }, + [K_USER] = { "username=", "" }, + [K_NONCE] = { "nonce=", "" }, + [K_LAST] = { NULL, NULL} + }; + + /* Always OK if no secret */ + if (ast_strlen_zero(secret) && ast_strlen_zero(md5secret)) + return AUTH_SUCCESSFUL; + if (sipmethod == SIP_REGISTER || sipmethod == SIP_SUBSCRIBE) { + /* On a REGISTER, we have to use 401 and its family of headers instead of 407 and its family + of headers -- GO SIP! Whoo hoo! Two things that do the same thing but are used in + different circumstances! What a surprise. */ + response = "401 Unauthorized"; + reqheader = "Authorization"; + respheader = "WWW-Authenticate"; + } + authtoken = get_header(req, reqheader); + if (ignore && !ast_strlen_zero(p->randdata) && ast_strlen_zero(authtoken)) { + /* This is a retransmitted invite/register/etc, don't reconstruct authentication + information */ + if (!reliable) { + /* Resend message if this was NOT a reliable delivery. Otherwise the + retransmission should get it */ + transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0); + /* Schedule auto destroy in 32 seconds (according to RFC 3261) */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } + return AUTH_CHALLENGE_SENT; + } else if (ast_strlen_zero(p->randdata) || ast_strlen_zero(authtoken)) { + /* We have no auth, so issue challenge and request authentication */ + ast_string_field_build(p, randdata, "%08lx", ast_random()); /* Create nonce for challenge */ + transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0); + /* Schedule auto destroy in 32 seconds */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return AUTH_CHALLENGE_SENT; + } + + /* --- We have auth, so check it */ + + /* Whoever came up with the authentication section of SIP can suck my %&#$&* for not putting + an example in the spec of just what it is you're doing a hash on. */ + + if (!(buf = ast_dynamic_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN))) + return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */ + + /* Make a copy of the response and parse it */ + res = ast_dynamic_str_thread_set(&buf, 0, &check_auth_buf, "%s", authtoken); + + if (res == AST_DYNSTR_BUILD_FAILED) + return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */ + + c = buf->str; + + while(c && *(c = ast_skip_blanks(c)) ) { /* lookup for keys */ + for (i = keys; i->key != NULL; i++) { + const char *separator = ","; /* default */ + + if (strncasecmp(c, i->key, strlen(i->key)) != 0) + continue; + /* Found. Skip keyword, take text in quotes or up to the separator. */ + c += strlen(i->key); + if (*c == '"') { /* in quotes. Skip first and look for last */ + c++; + separator = "\""; + } + i->s = c; + strsep(&c, separator); + break; + } + if (i->key == NULL) /* not found, jump after space or comma */ + strsep(&c, " ,"); + } + + /* Verify that digest username matches the username we auth as */ + if (strcmp(username, keys[K_USER].s)) { + ast_log(LOG_WARNING, "username mismatch, have <%s>, digest has <%s>\n", + username, keys[K_USER].s); + /* Oops, we're trying something here */ + return AUTH_USERNAME_MISMATCH; + } + + /* Verify nonce from request matches our nonce. If not, send 401 with new nonce */ + if (strcasecmp(p->randdata, keys[K_NONCE].s)) { /* XXX it was 'n'casecmp ? */ + wrongnonce = TRUE; + usednonce = keys[K_NONCE].s; + } + + if (!ast_strlen_zero(md5secret)) + ast_copy_string(a1_hash, md5secret, sizeof(a1_hash)); + else { + char a1[256]; + snprintf(a1, sizeof(a1), "%s:%s:%s", username, global_realm, secret); + ast_md5_hash(a1_hash, a1); + } + + /* compute the expected response to compare with what we received */ + { + char a2[256]; + char a2_hash[256]; + char resp[256]; + + snprintf(a2, sizeof(a2), "%s:%s", sip_methods[sipmethod].text, + S_OR(keys[K_URI].s, uri)); + ast_md5_hash(a2_hash, a2); + snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, usednonce, a2_hash); + ast_md5_hash(resp_hash, resp); + } + + good_response = keys[K_RESP].s && + !strncasecmp(keys[K_RESP].s, resp_hash, strlen(resp_hash)); + if (wrongnonce) { + if (good_response) { + if (sipdebug) + ast_log(LOG_NOTICE, "Correct auth, but based on stale nonce received from '%s'\n", get_header(req, "To")); + /* We got working auth token, based on stale nonce . */ + ast_string_field_build(p, randdata, "%08lx", ast_random()); + transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, TRUE); + } else { + /* Everything was wrong, so give the device one more try with a new challenge */ + if (!ast_test_flag(req, SIP_PKT_IGNORE)) { + if (sipdebug) + ast_log(LOG_NOTICE, "Bad authentication received from '%s'\n", get_header(req, "To")); + ast_string_field_build(p, randdata, "%08lx", ast_random()); + } else { + if (sipdebug) + ast_log(LOG_NOTICE, "Duplicate authentication received from '%s'\n", get_header(req, "To")); + } + transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, FALSE); + } + + /* Schedule auto destroy in 32 seconds */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return AUTH_CHALLENGE_SENT; + } + if (good_response) { + append_history(p, "AuthOK", "Auth challenge succesful for %s", username); + return AUTH_SUCCESSFUL; + } + + /* Ok, we have a bad username/secret pair */ + /* Tell the UAS not to re-send this authentication data, because + it will continue to fail + */ + + return AUTH_SECRET_FAILED; +} + +/*! \brief Change onhold state of a peer using a pvt structure */ +static void sip_peer_hold(struct sip_pvt *p, int hold) +{ + struct sip_peer *peer = find_peer(p->peername, NULL, 1, 0); + + if (!peer) + return; + + /* If they put someone on hold, increment the value... otherwise decrement it */ + if (hold) + peer->onHold++; + else + peer->onHold--; + + /* Request device state update */ + ast_device_state_changed("SIP/%s", peer->name); + + return; +} + +/*! \brief Callback for the devicestate notification (SUBSCRIBE) support subsystem +\note If you add an "hint" priority to the extension in the dial plan, + you will get notifications on device state changes */ +static int cb_extensionstate(char *context, char* exten, int state, void *data) +{ + struct sip_pvt *p = data; + + ast_mutex_lock(&p->lock); + + switch(state) { + case AST_EXTENSION_DEACTIVATED: /* Retry after a while */ + case AST_EXTENSION_REMOVED: /* Extension is gone */ + if (p->autokillid > -1 && sip_cancel_destroy(p)) /* Remove subscription expiry for renewals */ + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); /* Delete subscription in 32 secs */ + ast_verbose(VERBOSE_PREFIX_2 "Extension state: Watcher for hint %s %s. Notify User %s\n", exten, state == AST_EXTENSION_DEACTIVATED ? "deactivated" : "removed", p->username); + p->stateid = -1; + p->subscribed = NONE; + append_history(p, "Subscribestatus", "%s", state == AST_EXTENSION_REMOVED ? "HintRemoved" : "Deactivated"); + break; + default: /* Tell user */ + p->laststate = state; + break; + } + if (p->subscribed != NONE) { /* Only send state NOTIFY if we know the format */ + if (!p->pendinginvite) { + transmit_state_notify(p, state, 1, FALSE); + } else { + /* We already have a NOTIFY sent that is not answered. Queue the state up. + if many state changes happen meanwhile, we will only send a notification of the last one */ + ast_set_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE); + } + } + if (option_verbose > 1) + ast_verbose(VERBOSE_PREFIX_1 "Extension Changed %s[%s] new state %s for Notify User %s %s\n", exten, context, ast_extension_state2str(state), p->username, + ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE) ? "(queued)" : ""); + + + ast_mutex_unlock(&p->lock); + + return 0; +} + +/*! \brief Send a fake 401 Unauthorized response when the administrator + wants to hide the names of local users/peers from fishers + */ +static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, int reliable) +{ + ast_string_field_build(p, randdata, "%08lx", ast_random()); /* Create nonce for challenge */ + transmit_response_with_auth(p, "401 Unauthorized", req, p->randdata, reliable, "WWW-Authenticate", 0); +} + +/*! \brief Verify registration of user + - Registration is done in several steps, first a REGISTER without auth + to get a challenge (nonce) then a second one with auth + - Registration requests are only matched with peers that are marked as "dynamic" + */ +static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin, + struct sip_request *req, char *uri) +{ + enum check_auth_result res = AUTH_NOT_FOUND; + struct sip_peer *peer; + char tmp[256]; + char *name, *c; + char *t; + char *domain; + + /* Terminate URI */ + t = uri; + while(*t && (*t > 32) && (*t != ';')) + t++; + *t = '\0'; + + ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp)); + if (pedanticsipchecking) + ast_uri_decode(tmp); + + c = get_in_brackets(tmp); + c = strsep(&c, ";"); /* Ditch ;user=phone */ + + if (!strncasecmp(c, "sip:", 4)) { + name = c + 4; + } else { + name = c; + ast_log(LOG_NOTICE, "Invalid to address: '%s' from %s (missing sip:) trying to use anyway...\n", c, ast_inet_ntoa(sin->sin_addr)); + } + + /* Strip off the domain name */ + if ((c = strchr(name, '@'))) { + *c++ = '\0'; + domain = c; + if ((c = strchr(domain, ':'))) /* Remove :port */ + *c = '\0'; + if (!AST_LIST_EMPTY(&domain_list)) { + if (!check_sip_domain(domain, NULL, 0)) { + transmit_response(p, "404 Not found (unknown domain)", &p->initreq); + return AUTH_UNKNOWN_DOMAIN; + } + } + } + + ast_string_field_set(p, exten, name); + build_contact(p); + peer = find_peer(name, NULL, 1, 0); + if (!(peer && ast_apply_ha(peer->ha, sin))) { + /* Peer fails ACL check */ + if (peer) { + ASTOBJ_UNREF(peer, sip_destroy_peer); + res = AUTH_ACL_FAILED; + } else + res = AUTH_NOT_FOUND; + } + if (peer) { + /* Set Frame packetization */ + if (p->rtp) { + ast_rtp_codec_setpref(p->rtp, &peer->prefs); + p->autoframing = peer->autoframing; + } + if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)) { + ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name); + res = AUTH_PEER_NOT_DYNAMIC; + } else { + ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT); + transmit_response(p, "100 Trying", req); + if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri, XMIT_UNRELIABLE, ast_test_flag(req, SIP_PKT_IGNORE)))) { + if (sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + + /* We have a succesful registration attemp with proper authentication, + now, update the peer */ + switch (parse_register_contact(p, peer, req)) { + case PARSE_REGISTER_FAILED: + ast_log(LOG_WARNING, "Failed to parse contact info\n"); + transmit_response_with_date(p, "400 Bad Request", req); + peer->lastmsgssent = -1; + res = 0; + break; + case PARSE_REGISTER_QUERY: + transmit_response_with_date(p, "200 OK", req); + peer->lastmsgssent = -1; + res = 0; + break; + case PARSE_REGISTER_UPDATE: + update_peer(peer, p->expiry); + /* Say OK and ask subsystem to retransmit msg counter */ + transmit_response_with_date(p, "200 OK", req); + if (!ast_test_flag((&peer->flags[1]), SIP_PAGE2_SUBSCRIBEMWIONLY)) + peer->lastmsgssent = -1; + res = 0; + break; + } + } + } + } + if (!peer && autocreatepeer) { + /* Create peer if we have autocreate mode enabled */ + peer = temp_peer(name); + if (peer) { + ASTOBJ_CONTAINER_LINK(&peerl, peer); + if (sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + switch (parse_register_contact(p, peer, req)) { + case PARSE_REGISTER_FAILED: + ast_log(LOG_WARNING, "Failed to parse contact info\n"); + transmit_response_with_date(p, "400 Bad Request", req); + peer->lastmsgssent = -1; + res = 0; + break; + case PARSE_REGISTER_QUERY: + transmit_response_with_date(p, "200 OK", req); + peer->lastmsgssent = -1; + res = 0; + break; + case PARSE_REGISTER_UPDATE: + /* Say OK and ask subsystem to retransmit msg counter */ + transmit_response_with_date(p, "200 OK", req); + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Registered\r\n", peer->name); + peer->lastmsgssent = -1; + res = 0; + break; + } + } + } + if (!res) { + ast_device_state_changed("SIP/%s", peer->name); + } + if (res < 0) { + switch (res) { + case AUTH_SECRET_FAILED: + /* Wrong password in authentication. Go away, don't try again until you fixed it */ + transmit_response(p, "403 Forbidden (Bad auth)", &p->initreq); + break; + case AUTH_USERNAME_MISMATCH: + /* Username and digest username does not match. + Asterisk uses the From: username for authentication. We need the + users to use the same authentication user name until we support + proper authentication by digest auth name */ + transmit_response(p, "403 Authentication user name does not match account name", &p->initreq); + break; + case AUTH_NOT_FOUND: + case AUTH_PEER_NOT_DYNAMIC: + case AUTH_ACL_FAILED: + if (global_alwaysauthreject) { + transmit_fake_auth_response(p, &p->initreq, 1); + } else { + /* URI not found */ + if (res == AUTH_PEER_NOT_DYNAMIC) + transmit_response(p, "403 Forbidden", &p->initreq); + else + transmit_response(p, "404 Not found", &p->initreq); + } + break; + default: + break; + } + } + if (peer) + ASTOBJ_UNREF(peer, sip_destroy_peer); + + return res; +} + +/*! \brief Get referring dnis */ +static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq) +{ + char tmp[256], *c, *a; + struct sip_request *req; + + req = oreq; + if (!req) + req = &p->initreq; + ast_copy_string(tmp, get_header(req, "Diversion"), sizeof(tmp)); + if (ast_strlen_zero(tmp)) + return 0; + c = get_in_brackets(tmp); + if (strncasecmp(c, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not an RDNIS SIP header (%s)?\n", c); + return -1; + } + c += 4; + a = c; + strsep(&a, "@;"); /* trim anything after @ or ; */ + if (sip_debug_test_pvt(p)) + ast_verbose("RDNIS is %s\n", c); + ast_string_field_set(p, rdnis, c); + + return 0; +} + +/*! \brief Find out who the call is for + We use the INVITE uri to find out +*/ +static int get_destination(struct sip_pvt *p, struct sip_request *oreq) +{ + char tmp[256] = "", *uri, *a; + char tmpf[256] = "", *from; + struct sip_request *req; + char *colon; + char *decoded_uri; + + req = oreq; + if (!req) + req = &p->initreq; + + /* Find the request URI */ + if (req->rlPart2) + ast_copy_string(tmp, req->rlPart2, sizeof(tmp)); + + if (pedanticsipchecking) + ast_uri_decode(tmp); + + uri = get_in_brackets(tmp); + + if (strncasecmp(uri, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", uri); + return -1; + } + uri += 4; + + /* Now find the From: caller ID and name */ + ast_copy_string(tmpf, get_header(req, "From"), sizeof(tmpf)); + if (!ast_strlen_zero(tmpf)) { + if (pedanticsipchecking) + ast_uri_decode(tmpf); + from = get_in_brackets(tmpf); + } else { + from = NULL; + } + + if (!ast_strlen_zero(from)) { + if (strncasecmp(from, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", from); + return -1; + } + from += 4; + if ((a = strchr(from, '@'))) + *a++ = '\0'; + else + a = from; /* just a domain */ + from = strsep(&from, ";"); /* Remove userinfo options */ + a = strsep(&a, ";"); /* Remove URI options */ + ast_string_field_set(p, fromdomain, a); + } + + /* Skip any options and find the domain */ + + /* Get the target domain */ + if ((a = strchr(uri, '@'))) { + *a++ = '\0'; + } else { /* No username part */ + a = uri; + uri = "s"; /* Set extension to "s" */ + } + colon = strchr(a, ':'); /* Remove :port */ + if (colon) + *colon = '\0'; + + uri = strsep(&uri, ";"); /* Remove userinfo options */ + a = strsep(&a, ";"); /* Remove URI options */ + + ast_string_field_set(p, domain, a); + + if (!AST_LIST_EMPTY(&domain_list)) { + char domain_context[AST_MAX_EXTENSION]; + + domain_context[0] = '\0'; + if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) { + if (!allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) { + if (option_debug) + ast_log(LOG_DEBUG, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain); + return -2; + } + } + /* If we have a context defined, overwrite the original context */ + if (!ast_strlen_zero(domain_context)) + ast_string_field_set(p, context, domain_context); + } + + /* If the request coming in is a subscription and subscribecontext has been specified use it */ + if (req->method == SIP_SUBSCRIBE && !ast_strlen_zero(p->subscribecontext)) + ast_string_field_set(p, context, p->subscribecontext); + + if (sip_debug_test_pvt(p)) + ast_verbose("Looking for %s in %s (domain %s)\n", uri, p->context, p->domain); + + /* If this is a subscription we actually just need to see if a hint exists for the extension */ + if (req->method == SIP_SUBSCRIBE) { + char hint[AST_MAX_EXTENSION]; + return (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, p->context, p->exten) ? 0 : -1); + } else { + decoded_uri = ast_strdupa(uri); + ast_uri_decode(decoded_uri); + /* Check the dialplan for the username part of the request URI, + the domain will be stored in the SIPDOMAIN variable + Since extensions.conf can have unescaped characters, try matching a decoded + uri in addition to the non-decoded uri + Return 0 if we have a matching extension */ + if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from)) || ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from)) || + !strcmp(decoded_uri, ast_pickup_ext())) { + if (!oreq) + ast_string_field_set(p, exten, decoded_uri); + return 0; + } + } + + /* Return 1 for pickup extension or overlap dialling support (if we support it) */ + if((ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) && + ast_canmatch_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))) || + !strncmp(decoded_uri, ast_pickup_ext(), strlen(decoded_uri))) { + return 1; + } + + return -1; +} + +/*! \brief Lock interface lock and find matching pvt lock +*/ +static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag) +{ + struct sip_pvt *sip_pvt_ptr; + + ast_mutex_lock(&iflock); + + if (option_debug > 3 && totag) + ast_log(LOG_DEBUG, "Looking for callid %s (fromtag %s totag %s)\n", callid, fromtag ? fromtag : "<no fromtag>", totag ? totag : "<no totag>"); + + /* Search interfaces and find the match */ + for (sip_pvt_ptr = iflist; sip_pvt_ptr; sip_pvt_ptr = sip_pvt_ptr->next) { + if (!strcmp(sip_pvt_ptr->callid, callid)) { + int match = 1; + + /* Go ahead and lock it (and its owner) before returning */ + ast_mutex_lock(&sip_pvt_ptr->lock); + + /* Check if tags match. If not, this is not the call we want + (With a forking SIP proxy, several call legs share the + call id, but have different tags) + */ + if (pedanticsipchecking) { + const char *pvt_fromtag, *pvt_totag; + + if (ast_test_flag(&sip_pvt_ptr->flags[1], SIP_PAGE2_OUTGOING_CALL)) { + /* Outgoing call tags : from is "our", to is "their" */ + pvt_fromtag = sip_pvt_ptr->tag ; + pvt_totag = sip_pvt_ptr->theirtag ; + } else { + /* Incoming call tags : from is "their", to is "our" */ + pvt_fromtag = sip_pvt_ptr->theirtag ; + pvt_totag = sip_pvt_ptr->tag ; + } + if (ast_strlen_zero(fromtag) || strcmp(fromtag, pvt_fromtag) || (!ast_strlen_zero(totag) && strcmp(totag, pvt_totag))) + match = 0; + } + + if (!match) { + ast_mutex_unlock(&sip_pvt_ptr->lock); + continue; + } + + if (option_debug > 3 && totag) + ast_log(LOG_DEBUG, "Matched %s call - their tag is %s Our tag is %s\n", + ast_test_flag(&sip_pvt_ptr->flags[1], SIP_PAGE2_OUTGOING_CALL) ? "OUTGOING": "INCOMING", + sip_pvt_ptr->theirtag, sip_pvt_ptr->tag); + + /* deadlock avoidance... */ + while (sip_pvt_ptr->owner && ast_channel_trylock(sip_pvt_ptr->owner)) { + DEADLOCK_AVOIDANCE(&sip_pvt_ptr->lock); + } + break; + } + } + ast_mutex_unlock(&iflock); + if (option_debug > 3 && !sip_pvt_ptr) + ast_log(LOG_DEBUG, "Found no match for callid %s to-tag %s from-tag %s\n", callid, totag, fromtag); + return sip_pvt_ptr; +} + +/*! \brief Call transfer support (the REFER method) + * Extracts Refer headers into pvt dialog structure */ +static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req) +{ + + const char *p_referred_by = NULL; + char *h_refer_to = NULL; + char *h_referred_by = NULL; + char *refer_to; + const char *p_refer_to; + char *referred_by_uri = NULL; + char *ptr; + struct sip_request *req = NULL; + const char *transfer_context = NULL; + struct sip_refer *referdata; + + + req = outgoing_req; + referdata = transferer->refer; + + if (!req) + req = &transferer->initreq; + + p_refer_to = get_header(req, "Refer-To"); + if (ast_strlen_zero(p_refer_to)) { + ast_log(LOG_WARNING, "Refer-To Header missing. Skipping transfer.\n"); + return -2; /* Syntax error */ + } + h_refer_to = ast_strdupa(p_refer_to); + refer_to = get_in_brackets(h_refer_to); + if (pedanticsipchecking) + ast_uri_decode(refer_to); + + if (strncasecmp(refer_to, "sip:", 4)) { + ast_log(LOG_WARNING, "Can't transfer to non-sip: URI. (Refer-to: %s)?\n", refer_to); + return -3; + } + refer_to += 4; /* Skip sip: */ + + /* Get referred by header if it exists */ + p_referred_by = get_header(req, "Referred-By"); + if (!ast_strlen_zero(p_referred_by)) { + char *lessthan; + h_referred_by = ast_strdupa(p_referred_by); + if (pedanticsipchecking) + ast_uri_decode(h_referred_by); + + /* Store referrer's caller ID name */ + ast_copy_string(referdata->referred_by_name, h_referred_by, sizeof(referdata->referred_by_name)); + if ((lessthan = strchr(referdata->referred_by_name, '<'))) { + *(lessthan - 1) = '\0'; /* Space */ + } + + referred_by_uri = get_in_brackets(h_referred_by); + if(strncasecmp(referred_by_uri, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a sip: header (Referred-by: %s). Skipping.\n", referred_by_uri); + referred_by_uri = (char *) NULL; + } else { + referred_by_uri += 4; /* Skip sip: */ + } + } + + /* Check for arguments in the refer_to header */ + if ((ptr = strchr(refer_to, '?'))) { /* Search for arguments */ + *ptr++ = '\0'; + if (!strncasecmp(ptr, "REPLACES=", 9)) { + char *to = NULL, *from = NULL; + + /* This is an attended transfer */ + referdata->attendedtransfer = 1; + ast_copy_string(referdata->replaces_callid, ptr+9, sizeof(referdata->replaces_callid)); + ast_uri_decode(referdata->replaces_callid); + if ((ptr = strchr(referdata->replaces_callid, ';'))) /* Find options */ { + *ptr++ = '\0'; + } + + if (ptr) { + /* Find the different tags before we destroy the string */ + to = strcasestr(ptr, "to-tag="); + from = strcasestr(ptr, "from-tag="); + } + + /* Grab the to header */ + if (to) { + ptr = to + 7; + if ((to = strchr(ptr, '&'))) + *to = '\0'; + if ((to = strchr(ptr, ';'))) + *to = '\0'; + ast_copy_string(referdata->replaces_callid_totag, ptr, sizeof(referdata->replaces_callid_totag)); + } + + if (from) { + ptr = from + 9; + if ((to = strchr(ptr, '&'))) + *to = '\0'; + if ((to = strchr(ptr, ';'))) + *to = '\0'; + ast_copy_string(referdata->replaces_callid_fromtag, ptr, sizeof(referdata->replaces_callid_fromtag)); + } + + if (option_debug > 1) { + if (!pedanticsipchecking) + ast_log(LOG_DEBUG,"Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", referdata->replaces_callid ); + else + ast_log(LOG_DEBUG,"Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" ); + } + } + } + + if ((ptr = strchr(refer_to, '@'))) { /* Separate domain */ + char *urioption = NULL, *domain; + *ptr++ = '\0'; + + if ((urioption = strchr(ptr, ';'))) /* Separate urioptions */ + *urioption++ = '\0'; + + domain = ptr; + if ((ptr = strchr(domain, ':'))) /* Remove :port */ + *ptr = '\0'; + + /* Save the domain for the dial plan */ + ast_copy_string(referdata->refer_to_domain, domain, sizeof(referdata->refer_to_domain)); + if (urioption) + ast_copy_string(referdata->refer_to_urioption, urioption, sizeof(referdata->refer_to_urioption)); + } + + if ((ptr = strchr(refer_to, ';'))) /* Remove options */ + *ptr = '\0'; + ast_copy_string(referdata->refer_to, refer_to, sizeof(referdata->refer_to)); + + if (referred_by_uri) { + if ((ptr = strchr(referred_by_uri, ';'))) /* Remove options */ + *ptr = '\0'; + ast_copy_string(referdata->referred_by, referred_by_uri, sizeof(referdata->referred_by)); + } else { + referdata->referred_by[0] = '\0'; + } + + /* Determine transfer context */ + if (transferer->owner) /* Mimic behaviour in res_features.c */ + transfer_context = pbx_builtin_getvar_helper(transferer->owner, "TRANSFER_CONTEXT"); + + /* By default, use the context in the channel sending the REFER */ + if (ast_strlen_zero(transfer_context)) { + transfer_context = S_OR(transferer->owner->macrocontext, + S_OR(transferer->context, default_context)); + } + + ast_copy_string(referdata->refer_to_context, transfer_context, sizeof(referdata->refer_to_context)); + + /* Either an existing extension or the parking extension */ + if (ast_exists_extension(NULL, transfer_context, refer_to, 1, NULL) ) { + if (sip_debug_test_pvt(transferer)) { + ast_verbose("SIP transfer to extension %s@%s by %s\n", refer_to, transfer_context, referred_by_uri); + } + /* We are ready to transfer to the extension */ + return 0; + } + if (sip_debug_test_pvt(transferer)) + ast_verbose("Failed SIP Transfer to non-existing extension %s in context %s\n n", refer_to, transfer_context); + + /* Failure, we can't find this extension */ + return -1; +} + + +/*! \brief Call transfer support (old way, deprecated by the IETF)--*/ +static int get_also_info(struct sip_pvt *p, struct sip_request *oreq) +{ + char tmp[256] = "", *c, *a; + struct sip_request *req = oreq ? oreq : &p->initreq; + struct sip_refer *referdata = NULL; + const char *transfer_context = NULL; + + if (!p->refer && !sip_refer_allocate(p)) + return -1; + + referdata = p->refer; + + ast_copy_string(tmp, get_header(req, "Also"), sizeof(tmp)); + c = get_in_brackets(tmp); + + if (pedanticsipchecking) + ast_uri_decode(c); + + if (strncasecmp(c, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a SIP header in Also: transfer (%s)?\n", c); + return -1; + } + c += 4; + if ((a = strchr(c, ';'))) /* Remove arguments */ + *a = '\0'; + + if ((a = strchr(c, '@'))) { /* Separate Domain */ + *a++ = '\0'; + ast_copy_string(referdata->refer_to_domain, a, sizeof(referdata->refer_to_domain)); + } + + if (sip_debug_test_pvt(p)) + ast_verbose("Looking for %s in %s\n", c, p->context); + + if (p->owner) /* Mimic behaviour in res_features.c */ + transfer_context = pbx_builtin_getvar_helper(p->owner, "TRANSFER_CONTEXT"); + + /* By default, use the context in the channel sending the REFER */ + if (ast_strlen_zero(transfer_context)) { + transfer_context = S_OR(p->owner->macrocontext, + S_OR(p->context, default_context)); + } + if (ast_exists_extension(NULL, transfer_context, c, 1, NULL)) { + /* This is a blind transfer */ + if (option_debug) + ast_log(LOG_DEBUG,"SIP Bye-also transfer to Extension %s@%s \n", c, transfer_context); + ast_copy_string(referdata->refer_to, c, sizeof(referdata->refer_to)); + ast_copy_string(referdata->referred_by, "", sizeof(referdata->referred_by)); + ast_copy_string(referdata->refer_contact, "", sizeof(referdata->refer_contact)); + referdata->refer_call = NULL; + /* Set new context */ + ast_string_field_set(p, context, transfer_context); + return 0; + } else if (ast_canmatch_extension(NULL, p->context, c, 1, NULL)) { + return 1; + } + + return -1; +} +/*! \brief check Via: header for hostname, port and rport request/answer */ +static void check_via(struct sip_pvt *p, const struct sip_request *req) +{ + char via[512]; + char *c, *pt; + struct hostent *hp; + struct ast_hostent ahp; + + ast_copy_string(via, get_header(req, "Via"), sizeof(via)); + + /* Work on the leftmost value of the topmost Via header */ + c = strchr(via, ','); + if (c) + *c = '\0'; + + /* Check for rport */ + c = strstr(via, ";rport"); + if (c && (c[6] != '=')) /* rport query, not answer */ + ast_set_flag(&p->flags[0], SIP_NAT_ROUTE); + + c = strchr(via, ';'); + if (c) + *c = '\0'; + + c = strchr(via, ' '); + if (c) { + *c = '\0'; + c = ast_skip_blanks(c+1); + if (strcasecmp(via, "SIP/2.0/UDP")) { + ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via); + return; + } + pt = strchr(c, ':'); + if (pt) + *pt++ = '\0'; /* remember port pointer */ + hp = ast_gethostbyname(c, &ahp); + if (!hp) { + ast_log(LOG_WARNING, "'%s' is not a valid host\n", c); + return; + } + memset(&p->sa, 0, sizeof(p->sa)); + p->sa.sin_family = AF_INET; + memcpy(&p->sa.sin_addr, hp->h_addr, sizeof(p->sa.sin_addr)); + p->sa.sin_port = htons(pt ? atoi(pt) : STANDARD_SIP_PORT); + + if (sip_debug_test_pvt(p)) { + const struct sockaddr_in *dst = sip_real_dst(p); + ast_verbose("Sending to %s : %d (%s)\n", ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), sip_nat_mode(p)); + } + } +} + +/*! \brief Get caller id name from SIP headers */ +static char *get_calleridname(const char *input, char *output, size_t outputsize) +{ + const char *end = strchr(input,'<'); /* first_bracket */ + const char *tmp = strchr(input,'"'); /* first quote */ + int bytes = 0; + int maxbytes = outputsize - 1; + + if (!end || end == input) /* we require a part in brackets */ + return NULL; + + end--; /* move just before "<" */ + + if (tmp && tmp <= end) { + /* The quote (tmp) precedes the bracket (end+1). + * Find the matching quote and return the content. + */ + end = strchr(tmp+1, '"'); + if (!end) + return NULL; + bytes = (int) (end - tmp); + /* protect the output buffer */ + if (bytes > maxbytes) + bytes = maxbytes; + ast_copy_string(output, tmp + 1, bytes); + } else { + /* No quoted string, or it is inside brackets. */ + /* clear the empty characters in the begining*/ + input = ast_skip_blanks(input); + /* clear the empty characters in the end */ + while(*end && *end < 33 && end > input) + end--; + if (end >= input) { + bytes = (int) (end - input) + 2; + /* protect the output buffer */ + if (bytes > maxbytes) + bytes = maxbytes; + ast_copy_string(output, input, bytes); + } else + return NULL; + } + return output; +} + +/*! \brief Get caller id number from Remote-Party-ID header field + * Returns true if number should be restricted (privacy setting found) + * output is set to NULL if no number found + */ +static int get_rpid_num(const char *input, char *output, int maxlen) +{ + char *start; + char *end; + + start = strchr(input,':'); + if (!start) { + output[0] = '\0'; + return 0; + } + start++; + + /* we found "number" */ + ast_copy_string(output,start,maxlen); + output[maxlen-1] = '\0'; + + end = strchr(output,'@'); + if (end) + *end = '\0'; + else + output[0] = '\0'; + if (strstr(input,"privacy=full") || strstr(input,"privacy=uri")) + return AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED; + + return 0; +} + + +/*! \brief Check if matching user or peer is defined + Match user on From: user name and peer on IP/port + This is used on first invite (not re-invites) and subscribe requests + \return 0 on success, non-zero on failure +*/ +static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req, + int sipmethod, char *uri, enum xmittype reliable, + struct sockaddr_in *sin, struct sip_peer **authpeer) +{ + struct sip_user *user = NULL; + struct sip_peer *peer; + char from[256], *c; + char *of; + char rpid_num[50]; + const char *rpid; + enum check_auth_result res = AUTH_SUCCESSFUL; + char *t; + char calleridname[50]; + int debug=sip_debug_test_addr(sin); + struct ast_variable *tmpvar = NULL, *v = NULL; + char *uri2 = ast_strdupa(uri); + + /* Terminate URI */ + t = uri2; + while (*t && *t > 32 && *t != ';') + t++; + *t = '\0'; + ast_copy_string(from, get_header(req, "From"), sizeof(from)); /* XXX bug in original code, overwrote string */ + if (pedanticsipchecking) + ast_uri_decode(from); + /* XXX here tries to map the username for invite things */ + memset(calleridname, 0, sizeof(calleridname)); + get_calleridname(from, calleridname, sizeof(calleridname)); + if (calleridname[0]) + ast_string_field_set(p, cid_name, calleridname); + + rpid = get_header(req, "Remote-Party-ID"); + memset(rpid_num, 0, sizeof(rpid_num)); + if (!ast_strlen_zero(rpid)) + p->callingpres = get_rpid_num(rpid, rpid_num, sizeof(rpid_num)); + + of = get_in_brackets(from); + if (ast_strlen_zero(p->exten)) { + t = uri2; + if (!strncasecmp(t, "sip:", 4)) + t+= 4; + ast_string_field_set(p, exten, t); + t = strchr(p->exten, '@'); + if (t) + *t = '\0'; + if (ast_strlen_zero(p->our_contact)) + build_contact(p); + } + /* save the URI part of the From header */ + ast_string_field_set(p, from, of); + if (strncasecmp(of, "sip:", 4)) { + ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n"); + } else + of += 4; + /* Get just the username part */ + if ((c = strchr(of, '@'))) { + char *tmp; + *c = '\0'; + if ((c = strchr(of, ':'))) + *c = '\0'; + tmp = ast_strdupa(of); + /* We need to be able to handle auth-headers looking like + <sip:8164444422;phone-context=+1@1.2.3.4:5060;user=phone;tag=SDadkoa01-gK0c3bdb43> + */ + tmp = strsep(&tmp, ";"); + if (ast_is_shrinkable_phonenumber(tmp)) + ast_shrink_phone_number(tmp); + ast_string_field_set(p, cid_num, tmp); + } + + if (!authpeer) /* If we are looking for a peer, don't check the user objects (or realtime) */ + user = find_user(of, 1); + + /* Find user based on user name in the from header */ + if (user && ast_apply_ha(user->ha, sin)) { + ast_copy_flags(&p->flags[0], &user->flags[0], SIP_FLAGS_TO_COPY); + ast_copy_flags(&p->flags[1], &user->flags[1], SIP_PAGE2_FLAGS_TO_COPY); + if (sipmethod == SIP_INVITE) { + /* copy channel vars */ + for (v = user->chanvars ; v ; v = v->next) { + if ((tmpvar = ast_variable_new(v->name, v->value))) { + tmpvar->next = p->chanvars; + p->chanvars = tmpvar; + } + } + } + p->prefs = user->prefs; + /* Set Frame packetization */ + if (p->rtp) { + ast_rtp_codec_setpref(p->rtp, &p->prefs); + p->autoframing = user->autoframing; + } + /* replace callerid if rpid found, and not restricted */ + if (!ast_strlen_zero(rpid_num) && ast_test_flag(&p->flags[0], SIP_TRUSTRPID)) { + char *tmp; + if (*calleridname) + ast_string_field_set(p, cid_name, calleridname); + tmp = ast_strdupa(rpid_num); + if (ast_is_shrinkable_phonenumber(tmp)) + ast_shrink_phone_number(tmp); + ast_string_field_set(p, cid_num, tmp); + } + + do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT_ROUTE) ); + + if (!(res = check_auth(p, req, user->name, user->secret, user->md5secret, sipmethod, uri2, reliable, ast_test_flag(req, SIP_PKT_IGNORE)))) { + if (sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + ast_copy_flags(&p->flags[0], &user->flags[0], SIP_FLAGS_TO_COPY); + ast_copy_flags(&p->flags[1], &user->flags[1], SIP_PAGE2_FLAGS_TO_COPY); + /* Copy SIP extensions profile from INVITE */ + if (p->sipoptions) + user->sipoptions = p->sipoptions; + + /* If we have a call limit, set flag */ + if (user->call_limit) + ast_set_flag(&p->flags[0], SIP_CALL_LIMIT); + if (!ast_strlen_zero(user->context)) + ast_string_field_set(p, context, user->context); + if (!ast_strlen_zero(user->cid_num)) { + char *tmp = ast_strdupa(user->cid_num); + if (ast_is_shrinkable_phonenumber(tmp)) + ast_shrink_phone_number(tmp); + ast_string_field_set(p, cid_num, tmp); + } + if (!ast_strlen_zero(user->cid_name)) + ast_string_field_set(p, cid_name, user->cid_name); + ast_string_field_set(p, username, user->name); + ast_string_field_set(p, peername, user->name); + ast_string_field_set(p, peersecret, user->secret); + ast_string_field_set(p, peermd5secret, user->md5secret); + ast_string_field_set(p, subscribecontext, user->subscribecontext); + ast_string_field_set(p, accountcode, user->accountcode); + ast_string_field_set(p, language, user->language); + ast_string_field_set(p, mohsuggest, user->mohsuggest); + ast_string_field_set(p, mohinterpret, user->mohinterpret); + p->allowtransfer = user->allowtransfer; + p->amaflags = user->amaflags; + p->callgroup = user->callgroup; + p->pickupgroup = user->pickupgroup; + if (user->callingpres) /* User callingpres setting will override RPID header */ + p->callingpres = user->callingpres; + + /* Set default codec settings for this call */ + p->capability = user->capability; /* User codec choice */ + p->jointcapability = user->capability; /* Our codecs */ + if (p->peercapability) /* AND with peer's codecs */ + p->jointcapability &= p->peercapability; + if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) || + (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) + p->noncodeccapability |= AST_RTP_DTMF; + else + p->noncodeccapability &= ~AST_RTP_DTMF; + p->jointnoncodeccapability = p->noncodeccapability; + if (p->t38.peercapability) + p->t38.jointcapability &= p->t38.peercapability; + p->maxcallbitrate = user->maxcallbitrate; + /* If we do not support video, remove video from call structure */ + if ((!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) || !(p->capability & AST_FORMAT_VIDEO_MASK)) && p->vrtp) { + ast_rtp_destroy(p->vrtp); + p->vrtp = NULL; + } + } + if (user && debug) + ast_verbose("Found user '%s'\n", user->name); + } else { + if (user) { + if (!authpeer && debug) + ast_verbose("Found user '%s', but fails host access\n", user->name); + ASTOBJ_UNREF(user,sip_destroy_user); + } + user = NULL; + } + + if (!user) { + /* If we didn't find a user match, check for peers */ + if (sipmethod == SIP_SUBSCRIBE) + /* For subscribes, match on peer name only */ + peer = find_peer(of, NULL, 1, 0); + else + /* Look for peer based on the IP address we received data from */ + /* If peer is registered from this IP address or have this as a default + IP address, this call is from the peer + */ + peer = find_peer(NULL, &p->recv, 1, 0); + + if (peer) { + /* Set Frame packetization */ + if (p->rtp) { + ast_rtp_codec_setpref(p->rtp, &peer->prefs); + p->autoframing = peer->autoframing; + } + if (debug) + ast_verbose("Found peer '%s'\n", peer->name); + + /* Take the peer */ + ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY); + ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY); + + /* Copy SIP extensions profile to peer */ + if (p->sipoptions) + peer->sipoptions = p->sipoptions; + + /* replace callerid if rpid found, and not restricted */ + if (!ast_strlen_zero(rpid_num) && ast_test_flag(&p->flags[0], SIP_TRUSTRPID)) { + char *tmp = ast_strdupa(rpid_num); + if (*calleridname) + ast_string_field_set(p, cid_name, calleridname); + if (ast_is_shrinkable_phonenumber(tmp)) + ast_shrink_phone_number(tmp); + ast_string_field_set(p, cid_num, tmp); + } + do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT_ROUTE)); + + ast_string_field_set(p, peersecret, peer->secret); + ast_string_field_set(p, peermd5secret, peer->md5secret); + ast_string_field_set(p, subscribecontext, peer->subscribecontext); + ast_string_field_set(p, mohinterpret, peer->mohinterpret); + ast_string_field_set(p, mohsuggest, peer->mohsuggest); + if (peer->callingpres) /* Peer calling pres setting will override RPID */ + p->callingpres = peer->callingpres; + if (peer->maxms && peer->lastms) + p->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms; + if (ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE)) { + /* Pretend there is no required authentication */ + ast_string_field_free(p, peersecret); + ast_string_field_free(p, peermd5secret); + } + if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable, ast_test_flag(req, SIP_PKT_IGNORE)))) { + ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY); + ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY); + /* If we have a call limit, set flag */ + if (peer->call_limit) + ast_set_flag(&p->flags[0], SIP_CALL_LIMIT); + ast_string_field_set(p, peername, peer->name); + ast_string_field_set(p, authname, peer->name); + + if (sipmethod == SIP_INVITE) { + /* copy channel vars */ + for (v = peer->chanvars ; v ; v = v->next) { + if ((tmpvar = ast_variable_new(v->name, v->value))) { + tmpvar->next = p->chanvars; + p->chanvars = tmpvar; + } + } + } + if (authpeer) { + (*authpeer) = ASTOBJ_REF(peer); /* Add a ref to the object here, to keep it in memory a bit longer if it is realtime */ + } + + if (!ast_strlen_zero(peer->username)) { + ast_string_field_set(p, username, peer->username); + /* Use the default username for authentication on outbound calls */ + /* XXX this takes the name from the caller... can we override ? */ + ast_string_field_set(p, authname, peer->username); + } + if (!ast_strlen_zero(peer->cid_num)) { + char *tmp = ast_strdupa(peer->cid_num); + if (ast_is_shrinkable_phonenumber(tmp)) + ast_shrink_phone_number(tmp); + ast_string_field_set(p, cid_num, tmp); + } + if (!ast_strlen_zero(peer->cid_name)) + ast_string_field_set(p, cid_name, peer->cid_name); + ast_string_field_set(p, fullcontact, peer->fullcontact); + if (!ast_strlen_zero(peer->context)) + ast_string_field_set(p, context, peer->context); + ast_string_field_set(p, peersecret, peer->secret); + ast_string_field_set(p, peermd5secret, peer->md5secret); + ast_string_field_set(p, language, peer->language); + ast_string_field_set(p, accountcode, peer->accountcode); + p->amaflags = peer->amaflags; + p->callgroup = peer->callgroup; + p->pickupgroup = peer->pickupgroup; + p->capability = peer->capability; + p->prefs = peer->prefs; + p->jointcapability = peer->capability; + if (p->peercapability) + p->jointcapability &= p->peercapability; + p->maxcallbitrate = peer->maxcallbitrate; + if ((!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) || !(p->capability & AST_FORMAT_VIDEO_MASK)) && p->vrtp) { + ast_rtp_destroy(p->vrtp); + p->vrtp = NULL; + } + if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) || + (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) + p->noncodeccapability |= AST_RTP_DTMF; + else + p->noncodeccapability &= ~AST_RTP_DTMF; + p->jointnoncodeccapability = p->noncodeccapability; + if (p->t38.peercapability) + p->t38.jointcapability &= p->t38.peercapability; + } + ASTOBJ_UNREF(peer, sip_destroy_peer); + } else { + if (debug) + ast_verbose("Found no matching peer or user for '%s:%d'\n", ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port)); + + /* do we allow guests? */ + if (!global_allowguest) { + if (global_alwaysauthreject) + res = AUTH_FAKE_AUTH; /* reject with fake authorization request */ + else + res = AUTH_SECRET_FAILED; /* we don't want any guests, authentication will fail */ + } else if (!ast_strlen_zero(rpid_num) && ast_test_flag(&p->flags[0], SIP_TRUSTRPID)) { + char *tmp = ast_strdupa(rpid_num); + if (*calleridname) + ast_string_field_set(p, cid_name, calleridname); + if (ast_is_shrinkable_phonenumber(tmp)) + ast_shrink_phone_number(tmp); + ast_string_field_set(p, cid_num, tmp); + } + } + + } + + if (user) + ASTOBJ_UNREF(user, sip_destroy_user); + return res; +} + +/*! \brief Find user + If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced +*/ +static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin) +{ + return check_user_full(p, req, sipmethod, uri, reliable, sin, NULL); +} + +/*! \brief Get text out of a SIP MESSAGE packet */ +static int get_msg_text(char *buf, int len, struct sip_request *req) +{ + int x; + int y; + + buf[0] = '\0'; + y = len - strlen(buf) - 5; + if (y < 0) + y = 0; + for (x=0;x<req->lines;x++) { + strncat(buf, req->line[x], y); /* safe */ + y -= strlen(req->line[x]) + 1; + if (y < 0) + y = 0; + if (y != 0) + strcat(buf, "\n"); /* safe */ + } + return 0; +} + + +/*! \brief Receive SIP MESSAGE method messages +\note We only handle messages within current calls currently + Reference: RFC 3428 */ +static void receive_message(struct sip_pvt *p, struct sip_request *req) +{ + char buf[1024]; + struct ast_frame f; + const char *content_type = get_header(req, "Content-Type"); + + if (strncmp(content_type, "text/plain", strlen("text/plain"))) { /* No text/plain attachment */ + transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */ + if (!p->owner) + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return; + } + + if (get_msg_text(buf, sizeof(buf), req)) { + ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid); + transmit_response(p, "202 Accepted", req); + if (!p->owner) + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return; + } + + if (p->owner) { + if (sip_debug_test_pvt(p)) + ast_verbose("Message received: '%s'\n", buf); + memset(&f, 0, sizeof(f)); + f.frametype = AST_FRAME_TEXT; + f.subclass = 0; + f.offset = 0; + f.data = buf; + f.datalen = strlen(buf); + ast_queue_frame(p->owner, &f); + transmit_response(p, "202 Accepted", req); /* We respond 202 accepted, since we relay the message */ + } else { /* Message outside of a call, we do not support that */ + ast_log(LOG_WARNING,"Received message to %s from %s, dropped it...\n Content-Type:%s\n Message: %s\n", get_header(req,"To"), get_header(req,"From"), content_type, buf); + transmit_response(p, "405 Method Not Allowed", req); /* Good enough, or? */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } + return; +} + +/*! \brief CLI Command to show calls within limits set by call_limit */ +static int sip_show_inuse(int fd, int argc, char *argv[]) +{ +#define FORMAT "%-25.25s %-15.15s %-15.15s \n" +#define FORMAT2 "%-25.25s %-15.15s %-15.15s \n" + char ilimits[40]; + char iused[40]; + int showall = FALSE; + + if (argc < 3) + return RESULT_SHOWUSAGE; + + if (argc == 4 && !strcmp(argv[3],"all")) + showall = TRUE; + + ast_cli(fd, FORMAT, "* User name", "In use", "Limit"); + ASTOBJ_CONTAINER_TRAVERSE(&userl, 1, do { + ASTOBJ_RDLOCK(iterator); + if (iterator->call_limit) + snprintf(ilimits, sizeof(ilimits), "%d", iterator->call_limit); + else + ast_copy_string(ilimits, "N/A", sizeof(ilimits)); + snprintf(iused, sizeof(iused), "%d", iterator->inUse); + if (showall || iterator->call_limit) + ast_cli(fd, FORMAT2, iterator->name, iused, ilimits); + ASTOBJ_UNLOCK(iterator); + } while (0) ); + + ast_cli(fd, FORMAT, "* Peer name", "In use", "Limit"); + + ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do { + ASTOBJ_RDLOCK(iterator); + if (iterator->call_limit) + snprintf(ilimits, sizeof(ilimits), "%d", iterator->call_limit); + else + ast_copy_string(ilimits, "N/A", sizeof(ilimits)); + snprintf(iused, sizeof(iused), "%d/%d", iterator->inUse, iterator->inRinging); + if (showall || iterator->call_limit) + ast_cli(fd, FORMAT2, iterator->name, iused, ilimits); + ASTOBJ_UNLOCK(iterator); + } while (0) ); + + return RESULT_SUCCESS; +#undef FORMAT +#undef FORMAT2 +} + +/*! \brief Convert transfer mode to text string */ +static char *transfermode2str(enum transfermodes mode) +{ + if (mode == TRANSFER_OPENFORALL) + return "open"; + else if (mode == TRANSFER_CLOSED) + return "closed"; + return "strict"; +} + +/*! \brief Convert NAT setting to text string */ +static char *nat2str(int nat) +{ + switch(nat) { + case SIP_NAT_NEVER: + return "No"; + case SIP_NAT_ROUTE: + return "Route"; + case SIP_NAT_ALWAYS: + return "Always"; + case SIP_NAT_RFC3581: + return "RFC3581"; + default: + return "Unknown"; + } +} + +/*! \brief Report Peer status in character string + * \return 0 if peer is unreachable, 1 if peer is online, -1 if unmonitored + */ +static int peer_status(struct sip_peer *peer, char *status, int statuslen) +{ + int res = 0; + if (peer->maxms) { + if (peer->lastms < 0) { + ast_copy_string(status, "UNREACHABLE", statuslen); + } else if (peer->lastms > peer->maxms) { + snprintf(status, statuslen, "LAGGED (%d ms)", peer->lastms); + res = 1; + } else if (peer->lastms) { + snprintf(status, statuslen, "OK (%d ms)", peer->lastms); + res = 1; + } else { + ast_copy_string(status, "UNKNOWN", statuslen); + } + } else { + ast_copy_string(status, "Unmonitored", statuslen); + /* Checking if port is 0 */ + res = -1; + } + return res; +} + +/*! \brief CLI Command 'SIP Show Users' */ +static int sip_show_users(int fd, int argc, char *argv[]) +{ + regex_t regexbuf; + int havepattern = FALSE; + +#define FORMAT "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n" + + switch (argc) { + case 5: + if (!strcasecmp(argv[3], "like")) { + if (regcomp(®exbuf, argv[4], REG_EXTENDED | REG_NOSUB)) + return RESULT_SHOWUSAGE; + havepattern = TRUE; + } else + return RESULT_SHOWUSAGE; + case 3: + break; + default: + return RESULT_SHOWUSAGE; + } + + ast_cli(fd, FORMAT, "Username", "Secret", "Accountcode", "Def.Context", "ACL", "NAT"); + ASTOBJ_CONTAINER_TRAVERSE(&userl, 1, do { + ASTOBJ_RDLOCK(iterator); + + if (havepattern && regexec(®exbuf, iterator->name, 0, NULL, 0)) { + ASTOBJ_UNLOCK(iterator); + continue; + } + + ast_cli(fd, FORMAT, iterator->name, + iterator->secret, + iterator->accountcode, + iterator->context, + iterator->ha ? "Yes" : "No", + nat2str(ast_test_flag(&iterator->flags[0], SIP_NAT))); + ASTOBJ_UNLOCK(iterator); + } while (0) + ); + + if (havepattern) + regfree(®exbuf); + + return RESULT_SUCCESS; +#undef FORMAT +} + +static char mandescr_show_peers[] = +"Description: Lists SIP peers in text format with details on current status.\n" +"Variables: \n" +" ActionID: <id> Action ID for this transaction. Will be returned.\n"; + +/*! \brief Show SIP peers in the manager API */ +/* Inspired from chan_iax2 */ +static int manager_sip_show_peers(struct mansession *s, const struct message *m) +{ + const char *id = astman_get_header(m,"ActionID"); + const char *a[] = {"sip", "show", "peers"}; + char idtext[256] = ""; + int total = 0; + + if (!ast_strlen_zero(id)) + snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id); + + astman_send_ack(s, m, "Peer status list will follow"); + /* List the peers in separate manager events */ + _sip_show_peers(-1, &total, s, m, 3, a); + /* Send final confirmation */ + astman_append(s, + "Event: PeerlistComplete\r\n" + "ListItems: %d\r\n" + "%s" + "\r\n", total, idtext); + return 0; +} + +/*! \brief CLI Show Peers command */ +static int sip_show_peers(int fd, int argc, char *argv[]) +{ + return _sip_show_peers(fd, NULL, NULL, NULL, argc, (const char **) argv); +} + +/*! \brief _sip_show_peers: Execute sip show peers command */ +static int _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]) +{ + regex_t regexbuf; + int havepattern = FALSE; + +#define FORMAT2 "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %-10s\n" +#define FORMAT "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %-10s\n" + + char name[256]; + int total_peers = 0; + int peers_mon_online = 0; + int peers_mon_offline = 0; + int peers_unmon_offline = 0; + int peers_unmon_online = 0; + const char *id; + char idtext[256] = ""; + int realtimepeers; + + realtimepeers = ast_check_realtime("sippeers"); + + if (s) { /* Manager - get ActionID */ + id = astman_get_header(m,"ActionID"); + if (!ast_strlen_zero(id)) + snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id); + } + + switch (argc) { + case 5: + if (!strcasecmp(argv[3], "like")) { + if (regcomp(®exbuf, argv[4], REG_EXTENDED | REG_NOSUB)) + return RESULT_SHOWUSAGE; + havepattern = TRUE; + } else + return RESULT_SHOWUSAGE; + case 3: + break; + default: + return RESULT_SHOWUSAGE; + } + + if (!s) /* Normal list */ + ast_cli(fd, FORMAT2, "Name/username", "Host", "Dyn", "Nat", "ACL", "Port", "Status", (realtimepeers ? "Realtime" : "")); + + ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do { + char status[20] = ""; + char srch[2000]; + char pstatus; + + ASTOBJ_RDLOCK(iterator); + + if (havepattern && regexec(®exbuf, iterator->name, 0, NULL, 0)) { + ASTOBJ_UNLOCK(iterator); + continue; + } + + if (!ast_strlen_zero(iterator->username) && !s) + snprintf(name, sizeof(name), "%s/%s", iterator->name, iterator->username); + else + ast_copy_string(name, iterator->name, sizeof(name)); + + pstatus = peer_status(iterator, status, sizeof(status)); + if (pstatus == 1) + peers_mon_online++; + else if (pstatus == 0) + peers_mon_offline++; + else { + if (iterator->addr.sin_port == 0) + peers_unmon_offline++; + else + peers_unmon_online++; + } + + snprintf(srch, sizeof(srch), FORMAT, name, + iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iterator->addr.sin_addr) : "(Unspecified)", + ast_test_flag(&iterator->flags[1], SIP_PAGE2_DYNAMIC) ? " D " : " ", /* Dynamic or not? */ + ast_test_flag(&iterator->flags[0], SIP_NAT_ROUTE) ? " N " : " ", /* NAT=yes? */ + iterator->ha ? " A " : " ", /* permit/deny */ + ntohs(iterator->addr.sin_port), status, + realtimepeers ? (ast_test_flag(&iterator->flags[0], SIP_REALTIME) ? "Cached RT":"") : ""); + + if (!s) {/* Normal CLI list */ + ast_cli(fd, FORMAT, name, + iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iterator->addr.sin_addr) : "(Unspecified)", + ast_test_flag(&iterator->flags[1], SIP_PAGE2_DYNAMIC) ? " D " : " ", /* Dynamic or not? */ + ast_test_flag(&iterator->flags[0], SIP_NAT_ROUTE) ? " N " : " ", /* NAT=yes? */ + iterator->ha ? " A " : " ", /* permit/deny */ + + ntohs(iterator->addr.sin_port), status, + realtimepeers ? (ast_test_flag(&iterator->flags[0], SIP_REALTIME) ? "Cached RT":"") : ""); + } else { /* Manager format */ + /* The names here need to be the same as other channels */ + astman_append(s, + "Event: PeerEntry\r\n%s" + "Channeltype: SIP\r\n" + "ObjectName: %s\r\n" + "ChanObjectType: peer\r\n" /* "peer" or "user" */ + "IPaddress: %s\r\n" + "IPport: %d\r\n" + "Dynamic: %s\r\n" + "Natsupport: %s\r\n" + "VideoSupport: %s\r\n" + "ACL: %s\r\n" + "Status: %s\r\n" + "RealtimeDevice: %s\r\n\r\n", + idtext, + iterator->name, + iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iterator->addr.sin_addr) : "-none-", + ntohs(iterator->addr.sin_port), + ast_test_flag(&iterator->flags[1], SIP_PAGE2_DYNAMIC) ? "yes" : "no", /* Dynamic or not? */ + ast_test_flag(&iterator->flags[0], SIP_NAT_ROUTE) ? "yes" : "no", /* NAT=yes? */ + ast_test_flag(&iterator->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no", /* VIDEOSUPPORT=yes? */ + iterator->ha ? "yes" : "no", /* permit/deny */ + status, + realtimepeers ? (ast_test_flag(&iterator->flags[0], SIP_REALTIME) ? "yes":"no") : "no"); + } + + ASTOBJ_UNLOCK(iterator); + + total_peers++; + } while(0) ); + + if (!s) + ast_cli(fd, "%d sip peers [Monitored: %d online, %d offline Unmonitored: %d online, %d offline]\n", + total_peers, peers_mon_online, peers_mon_offline, peers_unmon_online, peers_unmon_offline); + + if (havepattern) + regfree(®exbuf); + + if (total) + *total = total_peers; + + + return RESULT_SUCCESS; +#undef FORMAT +#undef FORMAT2 +} + +/*! \brief List all allocated SIP Objects (realtime or static) */ +static int sip_show_objects(int fd, int argc, char *argv[]) +{ + char tmp[256]; + if (argc != 3) + return RESULT_SHOWUSAGE; + ast_cli(fd, "-= User objects: %d static, %d realtime =-\n\n", suserobjs, ruserobjs); + ASTOBJ_CONTAINER_DUMP(fd, tmp, sizeof(tmp), &userl); + ast_cli(fd, "-= Peer objects: %d static, %d realtime, %d autocreate =-\n\n", speerobjs, rpeerobjs, apeerobjs); + ASTOBJ_CONTAINER_DUMP(fd, tmp, sizeof(tmp), &peerl); + ast_cli(fd, "-= Registry objects: %d =-\n\n", regobjs); + ASTOBJ_CONTAINER_DUMP(fd, tmp, sizeof(tmp), ®l); + return RESULT_SUCCESS; +} +/*! \brief Print call group and pickup group */ +static void print_group(int fd, ast_group_t group, int crlf) +{ + char buf[256]; + ast_cli(fd, crlf ? "%s\r\n" : "%s\n", ast_print_group(buf, sizeof(buf), group) ); +} + +/*! \brief Convert DTMF mode to printable string */ +static const char *dtmfmode2str(int mode) +{ + switch (mode) { + case SIP_DTMF_RFC2833: + return "rfc2833"; + case SIP_DTMF_INFO: + return "info"; + case SIP_DTMF_INBAND: + return "inband"; + case SIP_DTMF_AUTO: + return "auto"; + } + return "<error>"; +} + +/*! \brief Convert Insecure setting to printable string */ +static const char *insecure2str(int port, int invite) +{ + if (port && invite) + return "port,invite"; + else if (port) + return "port"; + else if (invite) + return "invite"; + else + return "no"; +} + +/*! \brief Destroy disused contexts between reloads + Only used in reload_config so the code for regcontext doesn't get ugly +*/ +static void cleanup_stale_contexts(char *new, char *old) +{ + char *oldcontext, *newcontext, *stalecontext, *stringp, newlist[AST_MAX_CONTEXT]; + + while ((oldcontext = strsep(&old, "&"))) { + stalecontext = '\0'; + ast_copy_string(newlist, new, sizeof(newlist)); + stringp = newlist; + while ((newcontext = strsep(&stringp, "&"))) { + if (strcmp(newcontext, oldcontext) == 0) { + /* This is not the context you're looking for */ + stalecontext = '\0'; + break; + } else if (strcmp(newcontext, oldcontext)) { + stalecontext = oldcontext; + } + + } + if (stalecontext) + ast_context_destroy(ast_context_find(stalecontext), "SIP"); + } +} + +/*! \brief Remove temporary realtime objects from memory (CLI) */ +static int sip_prune_realtime(int fd, int argc, char *argv[]) +{ + struct sip_peer *peer; + struct sip_user *user; + int pruneuser = FALSE; + int prunepeer = FALSE; + int multi = FALSE; + char *name = NULL; + regex_t regexbuf; + + switch (argc) { + case 4: + if (!strcasecmp(argv[3], "user")) + return RESULT_SHOWUSAGE; + if (!strcasecmp(argv[3], "peer")) + return RESULT_SHOWUSAGE; + if (!strcasecmp(argv[3], "like")) + return RESULT_SHOWUSAGE; + if (!strcasecmp(argv[3], "all")) { + multi = TRUE; + pruneuser = prunepeer = TRUE; + } else { + pruneuser = prunepeer = TRUE; + name = argv[3]; + } + break; + case 5: + if (!strcasecmp(argv[4], "like")) + return RESULT_SHOWUSAGE; + if (!strcasecmp(argv[3], "all")) + return RESULT_SHOWUSAGE; + if (!strcasecmp(argv[3], "like")) { + multi = TRUE; + name = argv[4]; + pruneuser = prunepeer = TRUE; + } else if (!strcasecmp(argv[3], "user")) { + pruneuser = TRUE; + if (!strcasecmp(argv[4], "all")) + multi = TRUE; + else + name = argv[4]; + } else if (!strcasecmp(argv[3], "peer")) { + prunepeer = TRUE; + if (!strcasecmp(argv[4], "all")) + multi = TRUE; + else + name = argv[4]; + } else + return RESULT_SHOWUSAGE; + break; + case 6: + if (strcasecmp(argv[4], "like")) + return RESULT_SHOWUSAGE; + if (!strcasecmp(argv[3], "user")) { + pruneuser = TRUE; + name = argv[5]; + } else if (!strcasecmp(argv[3], "peer")) { + prunepeer = TRUE; + name = argv[5]; + } else + return RESULT_SHOWUSAGE; + break; + default: + return RESULT_SHOWUSAGE; + } + + if (multi && name) { + if (regcomp(®exbuf, name, REG_EXTENDED | REG_NOSUB)) + return RESULT_SHOWUSAGE; + } + + if (multi) { + if (prunepeer) { + int pruned = 0; + + ASTOBJ_CONTAINER_WRLOCK(&peerl); + ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do { + ASTOBJ_RDLOCK(iterator); + if (name && regexec(®exbuf, iterator->name, 0, NULL, 0)) { + ASTOBJ_UNLOCK(iterator); + continue; + }; + if (ast_test_flag(&iterator->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) { + ASTOBJ_MARK(iterator); + pruned++; + } + ASTOBJ_UNLOCK(iterator); + } while (0) ); + if (pruned) { + ASTOBJ_CONTAINER_PRUNE_MARKED(&peerl, sip_destroy_peer); + ast_cli(fd, "%d peers pruned.\n", pruned); + } else + ast_cli(fd, "No peers found to prune.\n"); + ASTOBJ_CONTAINER_UNLOCK(&peerl); + } + if (pruneuser) { + int pruned = 0; + + ASTOBJ_CONTAINER_WRLOCK(&userl); + ASTOBJ_CONTAINER_TRAVERSE(&userl, 1, do { + ASTOBJ_RDLOCK(iterator); + if (name && regexec(®exbuf, iterator->name, 0, NULL, 0)) { + ASTOBJ_UNLOCK(iterator); + continue; + }; + if (ast_test_flag(&iterator->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) { + ASTOBJ_MARK(iterator); + pruned++; + } + ASTOBJ_UNLOCK(iterator); + } while (0) ); + if (pruned) { + ASTOBJ_CONTAINER_PRUNE_MARKED(&userl, sip_destroy_user); + ast_cli(fd, "%d users pruned.\n", pruned); + } else + ast_cli(fd, "No users found to prune.\n"); + ASTOBJ_CONTAINER_UNLOCK(&userl); + } + } else { + if (prunepeer) { + if ((peer = ASTOBJ_CONTAINER_FIND_UNLINK(&peerl, name))) { + if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) { + ast_cli(fd, "Peer '%s' is not a Realtime peer, cannot be pruned.\n", name); + ASTOBJ_CONTAINER_LINK(&peerl, peer); + } else + ast_cli(fd, "Peer '%s' pruned.\n", name); + ASTOBJ_UNREF(peer, sip_destroy_peer); + } else + ast_cli(fd, "Peer '%s' not found.\n", name); + } + if (pruneuser) { + if ((user = ASTOBJ_CONTAINER_FIND_UNLINK(&userl, name))) { + if (!ast_test_flag(&user->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) { + ast_cli(fd, "User '%s' is not a Realtime user, cannot be pruned.\n", name); + ASTOBJ_CONTAINER_LINK(&userl, user); + } else + ast_cli(fd, "User '%s' pruned.\n", name); + ASTOBJ_UNREF(user, sip_destroy_user); + } else + ast_cli(fd, "User '%s' not found.\n", name); + } + } + + return RESULT_SUCCESS; +} + +/*! \brief Print codec list from preference to CLI/manager */ +static void print_codec_to_cli(int fd, struct ast_codec_pref *pref) +{ + int x, codec; + + for(x = 0; x < 32 ; x++) { + codec = ast_codec_pref_index(pref, x); + if (!codec) + break; + ast_cli(fd, "%s", ast_getformatname(codec)); + ast_cli(fd, ":%d", pref->framing[x]); + if (x < 31 && ast_codec_pref_index(pref, x + 1)) + ast_cli(fd, ","); + } + if (!x) + ast_cli(fd, "none"); +} + +/*! \brief Print domain mode to cli */ +static const char *domain_mode_to_text(const enum domain_mode mode) +{ + switch (mode) { + case SIP_DOMAIN_AUTO: + return "[Automatic]"; + case SIP_DOMAIN_CONFIG: + return "[Configured]"; + } + + return ""; +} + +/*! \brief CLI command to list local domains */ +static int sip_show_domains(int fd, int argc, char *argv[]) +{ + struct domain *d; +#define FORMAT "%-40.40s %-20.20s %-16.16s\n" + + if (AST_LIST_EMPTY(&domain_list)) { + ast_cli(fd, "SIP Domain support not enabled.\n\n"); + return RESULT_SUCCESS; + } else { + ast_cli(fd, FORMAT, "Our local SIP domains:", "Context", "Set by"); + AST_LIST_LOCK(&domain_list); + AST_LIST_TRAVERSE(&domain_list, d, list) + ast_cli(fd, FORMAT, d->domain, S_OR(d->context, "(default)"), + domain_mode_to_text(d->mode)); + AST_LIST_UNLOCK(&domain_list); + ast_cli(fd, "\n"); + return RESULT_SUCCESS; + } +} +#undef FORMAT + +static char mandescr_show_peer[] = +"Description: Show one SIP peer with details on current status.\n" +"Variables: \n" +" Peer: <name> The peer name you want to check.\n" +" ActionID: <id> Optional action ID for this AMI transaction.\n"; + +/*! \brief Show SIP peers in the manager API */ +static int manager_sip_show_peer(struct mansession *s, const struct message *m) +{ + const char *a[4]; + const char *peer; + int ret; + + peer = astman_get_header(m,"Peer"); + if (ast_strlen_zero(peer)) { + astman_send_error(s, m, "Peer: <name> missing."); + return 0; + } + a[0] = "sip"; + a[1] = "show"; + a[2] = "peer"; + a[3] = peer; + + ret = _sip_show_peer(1, -1, s, m, 4, a); + astman_append(s, "\r\n\r\n" ); + return ret; +} + + + +/*! \brief Show one peer in detail */ +static int sip_show_peer(int fd, int argc, char *argv[]) +{ + return _sip_show_peer(0, fd, NULL, NULL, argc, (const char **) argv); +} + +/*! \brief Show one peer in detail (main function) */ +static int _sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]) +{ + char status[30] = ""; + char cbuf[256]; + struct sip_peer *peer; + char codec_buf[512]; + struct ast_codec_pref *pref; + struct ast_variable *v; + struct sip_auth *auth; + int x = 0, codec = 0, load_realtime; + int realtimepeers; + + realtimepeers = ast_check_realtime("sippeers"); + + if (argc < 4) + return RESULT_SHOWUSAGE; + + load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE; + peer = find_peer(argv[3], NULL, load_realtime, 0); + if (s) { /* Manager */ + if (peer) { + const char *id = astman_get_header(m,"ActionID"); + + astman_append(s, "Response: Success\r\n"); + if (!ast_strlen_zero(id)) + astman_append(s, "ActionID: %s\r\n",id); + } else { + snprintf (cbuf, sizeof(cbuf), "Peer %s not found.", argv[3]); + astman_send_error(s, m, cbuf); + return 0; + } + } + if (peer && type==0 ) { /* Normal listing */ + ast_cli(fd,"\n\n"); + ast_cli(fd, " * Name : %s\n", peer->name); + if (realtimepeers) { /* Realtime is enabled */ + ast_cli(fd, " Realtime peer: %s\n", ast_test_flag(&peer->flags[0], SIP_REALTIME) ? "Yes, cached" : "No"); + } + ast_cli(fd, " Secret : %s\n", ast_strlen_zero(peer->secret)?"<Not set>":"<Set>"); + ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(peer->md5secret)?"<Not set>":"<Set>"); + for (auth = peer->auth; auth; auth = auth->next) { + ast_cli(fd, " Realm-auth : Realm %-15.15s User %-10.20s ", auth->realm, auth->username); + ast_cli(fd, "%s\n", !ast_strlen_zero(auth->secret)?"<Secret set>":(!ast_strlen_zero(auth->md5secret)?"<MD5secret set>" : "<Not set>")); + } + ast_cli(fd, " Context : %s\n", peer->context); + ast_cli(fd, " Subscr.Cont. : %s\n", S_OR(peer->subscribecontext, "<Not set>") ); + ast_cli(fd, " Language : %s\n", peer->language); + if (!ast_strlen_zero(peer->accountcode)) + ast_cli(fd, " Accountcode : %s\n", peer->accountcode); + ast_cli(fd, " AMA flags : %s\n", ast_cdr_flags2str(peer->amaflags)); + ast_cli(fd, " Transfer mode: %s\n", transfermode2str(peer->allowtransfer)); + ast_cli(fd, " CallingPres : %s\n", ast_describe_caller_presentation(peer->callingpres)); + if (!ast_strlen_zero(peer->fromuser)) + ast_cli(fd, " FromUser : %s\n", peer->fromuser); + if (!ast_strlen_zero(peer->fromdomain)) + ast_cli(fd, " FromDomain : %s\n", peer->fromdomain); + ast_cli(fd, " Callgroup : "); + print_group(fd, peer->callgroup, 0); + ast_cli(fd, " Pickupgroup : "); + print_group(fd, peer->pickupgroup, 0); + ast_cli(fd, " Mailbox : %s\n", peer->mailbox); + ast_cli(fd, " VM Extension : %s\n", peer->vmexten); + ast_cli(fd, " LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff); + ast_cli(fd, " Call limit : %d\n", peer->call_limit); + ast_cli(fd, " Dynamic : %s\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)?"Yes":"No")); + ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>")); + ast_cli(fd, " MaxCallBR : %d kbps\n", peer->maxcallbitrate); + ast_cli(fd, " Expire : %ld\n", ast_sched_when(sched, peer->expire)); + ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT), ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE))); + ast_cli(fd, " Nat : %s\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT))); + ast_cli(fd, " ACL : %s\n", (peer->ha?"Yes":"No")); + ast_cli(fd, " T38 pt UDPTL : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL)?"Yes":"No"); +#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS + ast_cli(fd, " T38 pt RTP : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_RTP)?"Yes":"No"); + ast_cli(fd, " T38 pt TCP : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_TCP)?"Yes":"No"); +#endif + ast_cli(fd, " CanReinvite : %s\n", ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)?"Yes":"No"); + ast_cli(fd, " PromiscRedir : %s\n", ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Yes":"No"); + ast_cli(fd, " User=Phone : %s\n", ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Yes":"No"); + ast_cli(fd, " Video Support: %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Yes":"No"); + ast_cli(fd, " Trust RPID : %s\n", ast_test_flag(&peer->flags[0], SIP_TRUSTRPID) ? "Yes" : "No"); + ast_cli(fd, " Send RPID : %s\n", ast_test_flag(&peer->flags[0], SIP_SENDRPID) ? "Yes" : "No"); + ast_cli(fd, " Subscriptions: %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No"); + ast_cli(fd, " Overlap dial : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP) ? "Yes" : "No"); + + /* - is enumerated */ + ast_cli(fd, " DTMFmode : %s\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF))); + ast_cli(fd, " LastMsg : %d\n", peer->lastmsg); + ast_cli(fd, " ToHost : %s\n", peer->tohost); + ast_cli(fd, " Addr->IP : %s Port %d\n", peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port)); + ast_cli(fd, " Defaddr->IP : %s Port %d\n", ast_inet_ntoa(peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port)); + if (!ast_strlen_zero(global_regcontext)) + ast_cli(fd, " Reg. exten : %s\n", peer->regexten); + ast_cli(fd, " Def. Username: %s\n", peer->username); + ast_cli(fd, " SIP Options : "); + if (peer->sipoptions) { + int lastoption = -1; + for (x=0 ; (x < (sizeof(sip_options) / sizeof(sip_options[0]))); x++) { + if (sip_options[x].id != lastoption) { + if (peer->sipoptions & sip_options[x].id) + ast_cli(fd, "%s ", sip_options[x].text); + lastoption = x; + } + } + } else + ast_cli(fd, "(none)"); + + ast_cli(fd, "\n"); + ast_cli(fd, " Codecs : "); + ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability); + ast_cli(fd, "%s\n", codec_buf); + ast_cli(fd, " Codec Order : ("); + print_codec_to_cli(fd, &peer->prefs); + ast_cli(fd, ")\n"); + + ast_cli(fd, " Auto-Framing: %s \n", peer->autoframing ? "Yes" : "No"); + ast_cli(fd, " Status : "); + peer_status(peer, status, sizeof(status)); + ast_cli(fd, "%s\n",status); + ast_cli(fd, " Useragent : %s\n", peer->useragent); + ast_cli(fd, " Reg. Contact : %s\n", peer->fullcontact); + if (peer->chanvars) { + ast_cli(fd, " Variables :\n"); + for (v = peer->chanvars ; v ; v = v->next) + ast_cli(fd, " %s = %s\n", v->name, v->value); + } + ast_cli(fd,"\n"); + ASTOBJ_UNREF(peer,sip_destroy_peer); + } else if (peer && type == 1) { /* manager listing */ + char buf[256]; + astman_append(s, "Channeltype: SIP\r\n"); + astman_append(s, "ObjectName: %s\r\n", peer->name); + astman_append(s, "ChanObjectType: peer\r\n"); + astman_append(s, "SecretExist: %s\r\n", ast_strlen_zero(peer->secret)?"N":"Y"); + astman_append(s, "MD5SecretExist: %s\r\n", ast_strlen_zero(peer->md5secret)?"N":"Y"); + astman_append(s, "Context: %s\r\n", peer->context); + astman_append(s, "Language: %s\r\n", peer->language); + if (!ast_strlen_zero(peer->accountcode)) + astman_append(s, "Accountcode: %s\r\n", peer->accountcode); + astman_append(s, "AMAflags: %s\r\n", ast_cdr_flags2str(peer->amaflags)); + astman_append(s, "CID-CallingPres: %s\r\n", ast_describe_caller_presentation(peer->callingpres)); + if (!ast_strlen_zero(peer->fromuser)) + astman_append(s, "SIP-FromUser: %s\r\n", peer->fromuser); + if (!ast_strlen_zero(peer->fromdomain)) + astman_append(s, "SIP-FromDomain: %s\r\n", peer->fromdomain); + astman_append(s, "Callgroup: "); + astman_append(s, "%s\r\n", ast_print_group(buf, sizeof(buf), peer->callgroup)); + astman_append(s, "Pickupgroup: "); + astman_append(s, "%s\r\n", ast_print_group(buf, sizeof(buf), peer->pickupgroup)); + astman_append(s, "VoiceMailbox: %s\r\n", peer->mailbox); + astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer)); + astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent); + astman_append(s, "Call-limit: %d\r\n", peer->call_limit); + astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate); + astman_append(s, "Dynamic: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)?"Y":"N")); + astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "")); + astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched,peer->expire)); + astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT), ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE))); + astman_append(s, "SIP-NatSupport: %s\r\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT))); + astman_append(s, "ACL: %s\r\n", (peer->ha?"Y":"N")); + astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)?"Y":"N")); + astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N")); + astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N")); + astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N")); + + /* - is enumerated */ + astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF))); + astman_append(s, "SIPLastMsg: %d\r\n", peer->lastmsg); + astman_append(s, "ToHost: %s\r\n", peer->tohost); + astman_append(s, "Address-IP: %s\r\nAddress-Port: %d\r\n", peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "", ntohs(peer->addr.sin_port)); + astman_append(s, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_inet_ntoa(peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port)); + astman_append(s, "Default-Username: %s\r\n", peer->username); + if (!ast_strlen_zero(global_regcontext)) + astman_append(s, "RegExtension: %s\r\n", peer->regexten); + astman_append(s, "Codecs: "); + ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability); + astman_append(s, "%s\r\n", codec_buf); + astman_append(s, "CodecOrder: "); + pref = &peer->prefs; + for(x = 0; x < 32 ; x++) { + codec = ast_codec_pref_index(pref,x); + if (!codec) + break; + astman_append(s, "%s", ast_getformatname(codec)); + if (x < 31 && ast_codec_pref_index(pref,x+1)) + astman_append(s, ","); + } + + astman_append(s, "\r\n"); + astman_append(s, "Status: "); + peer_status(peer, status, sizeof(status)); + astman_append(s, "%s\r\n", status); + astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent); + astman_append(s, "Reg-Contact : %s\r\n", peer->fullcontact); + if (peer->chanvars) { + for (v = peer->chanvars ; v ; v = v->next) { + astman_append(s, "ChanVariable:\n"); + astman_append(s, " %s,%s\r\n", v->name, v->value); + } + } + + ASTOBJ_UNREF(peer,sip_destroy_peer); + + } else { + ast_cli(fd,"Peer %s not found.\n", argv[3]); + ast_cli(fd,"\n"); + } + + return RESULT_SUCCESS; +} + +/*! \brief Show one user in detail */ +static int sip_show_user(int fd, int argc, char *argv[]) +{ + char cbuf[256]; + struct sip_user *user; + struct ast_variable *v; + int load_realtime; + + if (argc < 4) + return RESULT_SHOWUSAGE; + + /* Load from realtime storage? */ + load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE; + + user = find_user(argv[3], load_realtime); + if (user) { + ast_cli(fd,"\n\n"); + ast_cli(fd, " * Name : %s\n", user->name); + ast_cli(fd, " Secret : %s\n", ast_strlen_zero(user->secret)?"<Not set>":"<Set>"); + ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(user->md5secret)?"<Not set>":"<Set>"); + ast_cli(fd, " Context : %s\n", user->context); + ast_cli(fd, " Language : %s\n", user->language); + if (!ast_strlen_zero(user->accountcode)) + ast_cli(fd, " Accountcode : %s\n", user->accountcode); + ast_cli(fd, " AMA flags : %s\n", ast_cdr_flags2str(user->amaflags)); + ast_cli(fd, " Transfer mode: %s\n", transfermode2str(user->allowtransfer)); + ast_cli(fd, " MaxCallBR : %d kbps\n", user->maxcallbitrate); + ast_cli(fd, " CallingPres : %s\n", ast_describe_caller_presentation(user->callingpres)); + ast_cli(fd, " Call limit : %d\n", user->call_limit); + ast_cli(fd, " Callgroup : "); + print_group(fd, user->callgroup, 0); + ast_cli(fd, " Pickupgroup : "); + print_group(fd, user->pickupgroup, 0); + ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), user->cid_name, user->cid_num, "<unspecified>")); + ast_cli(fd, " ACL : %s\n", (user->ha?"Yes":"No")); + ast_cli(fd, " Codec Order : ("); + print_codec_to_cli(fd, &user->prefs); + ast_cli(fd, ")\n"); + + ast_cli(fd, " Auto-Framing: %s \n", user->autoframing ? "Yes" : "No"); + if (user->chanvars) { + ast_cli(fd, " Variables :\n"); + for (v = user->chanvars ; v ; v = v->next) + ast_cli(fd, " %s = %s\n", v->name, v->value); + } + ast_cli(fd,"\n"); + ASTOBJ_UNREF(user,sip_destroy_user); + } else { + ast_cli(fd,"User %s not found.\n", argv[3]); + ast_cli(fd,"\n"); + } + + return RESULT_SUCCESS; +} + +/*! \brief Show SIP Registry (registrations with other SIP proxies */ +static int sip_show_registry(int fd, int argc, char *argv[]) +{ +#define FORMAT2 "%-30.30s %-12.12s %8.8s %-20.20s %-25.25s\n" +#define FORMAT "%-30.30s %-12.12s %8d %-20.20s %-25.25s\n" + char host[80]; + char tmpdat[256]; + struct tm tm; + + + if (argc != 3) + return RESULT_SHOWUSAGE; + ast_cli(fd, FORMAT2, "Host", "Username", "Refresh", "State", "Reg.Time"); + ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do { + ASTOBJ_RDLOCK(iterator); + snprintf(host, sizeof(host), "%s:%d", iterator->hostname, iterator->portno ? iterator->portno : STANDARD_SIP_PORT); + if (iterator->regtime) { + ast_localtime(&iterator->regtime, &tm, NULL); + strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T", &tm); + } else { + tmpdat[0] = 0; + } + ast_cli(fd, FORMAT, host, iterator->username, iterator->refresh, regstate2str(iterator->regstate), tmpdat); + ASTOBJ_UNLOCK(iterator); + } while(0)); + return RESULT_SUCCESS; +#undef FORMAT +#undef FORMAT2 +} + +/*! \brief List global settings for the SIP channel */ +static int sip_show_settings(int fd, int argc, char *argv[]) +{ + int realtimepeers; + int realtimeusers; + char codec_buf[SIPBUFSIZE]; + + realtimepeers = ast_check_realtime("sippeers"); + realtimeusers = ast_check_realtime("sipusers"); + + if (argc != 3) + return RESULT_SHOWUSAGE; + ast_cli(fd, "\n\nGlobal Settings:\n"); + ast_cli(fd, "----------------\n"); + ast_cli(fd, " SIP Port: %d\n", ntohs(bindaddr.sin_port)); + ast_cli(fd, " Bindaddress: %s\n", ast_inet_ntoa(bindaddr.sin_addr)); + ast_cli(fd, " Videosupport: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "Yes" : "No"); + ast_cli(fd, " AutoCreatePeer: %s\n", autocreatepeer ? "Yes" : "No"); + ast_cli(fd, " Allow unknown access: %s\n", global_allowguest ? "Yes" : "No"); + ast_cli(fd, " Allow subscriptions: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No"); + ast_cli(fd, " Allow overlap dialing: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) ? "Yes" : "No"); + ast_cli(fd, " Promsic. redir: %s\n", ast_test_flag(&global_flags[0], SIP_PROMISCREDIR) ? "Yes" : "No"); + ast_cli(fd, " SIP domain support: %s\n", AST_LIST_EMPTY(&domain_list) ? "No" : "Yes"); + ast_cli(fd, " Call to non-local dom.: %s\n", allow_external_domains ? "Yes" : "No"); + ast_cli(fd, " URI user is phone no: %s\n", ast_test_flag(&global_flags[0], SIP_USEREQPHONE) ? "Yes" : "No"); + ast_cli(fd, " Our auth realm %s\n", global_realm); + ast_cli(fd, " Realm. auth: %s\n", authl ? "Yes": "No"); + ast_cli(fd, " Always auth rejects: %s\n", global_alwaysauthreject ? "Yes" : "No"); + ast_cli(fd, " Call limit peers only: %s\n", global_limitonpeers ? "Yes" : "No"); + ast_cli(fd, " Direct RTP setup: %s\n", global_directrtpsetup ? "Yes" : "No"); + ast_cli(fd, " User Agent: %s\n", global_useragent); + ast_cli(fd, " MWI checking interval: %d secs\n", global_mwitime); + ast_cli(fd, " Reg. context: %s\n", S_OR(global_regcontext, "(not set)")); + ast_cli(fd, " Caller ID: %s\n", default_callerid); + ast_cli(fd, " From: Domain: %s\n", default_fromdomain); + ast_cli(fd, " Record SIP history: %s\n", recordhistory ? "On" : "Off"); + ast_cli(fd, " Call Events: %s\n", global_callevents ? "On" : "Off"); + ast_cli(fd, " IP ToS SIP: %s\n", ast_tos2str(global_tos_sip)); + ast_cli(fd, " IP ToS RTP audio: %s\n", ast_tos2str(global_tos_audio)); + ast_cli(fd, " IP ToS RTP video: %s\n", ast_tos2str(global_tos_video)); + ast_cli(fd, " T38 fax pt UDPTL: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL) ? "Yes" : "No"); +#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS + ast_cli(fd, " T38 fax pt RTP: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP) ? "Yes" : "No"); + ast_cli(fd, " T38 fax pt TCP: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_TCP) ? "Yes" : "No"); +#endif + ast_cli(fd, " RFC2833 Compensation: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_RFC2833_COMPENSATE) ? "Yes" : "No"); + if (!realtimepeers && !realtimeusers) + ast_cli(fd, " SIP realtime: Disabled\n" ); + else + ast_cli(fd, " SIP realtime: Enabled\n" ); + + ast_cli(fd, "\nGlobal Signalling Settings:\n"); + ast_cli(fd, "---------------------------\n"); + ast_cli(fd, " Codecs: "); + ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, global_capability); + ast_cli(fd, "%s\n", codec_buf); + ast_cli(fd, " Codec Order: "); + print_codec_to_cli(fd, &default_prefs); + ast_cli(fd, "\n"); + ast_cli(fd, " T1 minimum: %d\n", global_t1min); + ast_cli(fd, " Relax DTMF: %s\n", global_relaxdtmf ? "Yes" : "No"); + ast_cli(fd, " Compact SIP headers: %s\n", compactheaders ? "Yes" : "No"); + ast_cli(fd, " RTP Keepalive: %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" ); + ast_cli(fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" ); + ast_cli(fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)"); + ast_cli(fd, " MWI NOTIFY mime type: %s\n", default_notifymime); + ast_cli(fd, " DNS SRV lookup: %s\n", srvlookup ? "Yes" : "No"); + ast_cli(fd, " Pedantic SIP support: %s\n", pedanticsipchecking ? "Yes" : "No"); + ast_cli(fd, " Reg. min duration %d secs\n", min_expiry); + ast_cli(fd, " Reg. max duration: %d secs\n", max_expiry); + ast_cli(fd, " Reg. default duration: %d secs\n", default_expiry); + ast_cli(fd, " Outbound reg. timeout: %d secs\n", global_reg_timeout); + ast_cli(fd, " Outbound reg. attempts: %d\n", global_regattempts_max); + ast_cli(fd, " Notify ringing state: %s\n", global_notifyringing ? "Yes" : "No"); + ast_cli(fd, " Notify hold state: %s\n", global_notifyhold ? "Yes" : "No"); + ast_cli(fd, " SIP Transfer mode: %s\n", transfermode2str(global_allowtransfer)); + ast_cli(fd, " Max Call Bitrate: %d kbps\r\n", default_maxcallbitrate); + ast_cli(fd, " Auto-Framing: %s \r\n", global_autoframing ? "Yes" : "No"); + ast_cli(fd, "\nDefault Settings:\n"); + ast_cli(fd, "-----------------\n"); + ast_cli(fd, " Context: %s\n", default_context); + ast_cli(fd, " Nat: %s\n", nat2str(ast_test_flag(&global_flags[0], SIP_NAT))); + ast_cli(fd, " DTMF: %s\n", dtmfmode2str(ast_test_flag(&global_flags[0], SIP_DTMF))); + ast_cli(fd, " Qualify: %d\n", default_qualify); + ast_cli(fd, " Use ClientCode: %s\n", ast_test_flag(&global_flags[0], SIP_USECLIENTCODE) ? "Yes" : "No"); + ast_cli(fd, " Progress inband: %s\n", (ast_test_flag(&global_flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER) ? "Never" : (ast_test_flag(&global_flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NO) ? "No" : "Yes" ); + ast_cli(fd, " Language: %s\n", S_OR(default_language, "(Defaults to English)")); + ast_cli(fd, " MOH Interpret: %s\n", default_mohinterpret); + ast_cli(fd, " MOH Suggest: %s\n", default_mohsuggest); + ast_cli(fd, " Voice Mail Extension: %s\n", default_vmexten); + + + if (realtimepeers || realtimeusers) { + ast_cli(fd, "\nRealtime SIP Settings:\n"); + ast_cli(fd, "----------------------\n"); + ast_cli(fd, " Realtime Peers: %s\n", realtimepeers ? "Yes" : "No"); + ast_cli(fd, " Realtime Users: %s\n", realtimeusers ? "Yes" : "No"); + ast_cli(fd, " Cache Friends: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) ? "Yes" : "No"); + ast_cli(fd, " Update: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_RTUPDATE) ? "Yes" : "No"); + ast_cli(fd, " Ignore Reg. Expire: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_IGNOREREGEXPIRE) ? "Yes" : "No"); + ast_cli(fd, " Save sys. name: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_RTSAVE_SYSNAME) ? "Yes" : "No"); + ast_cli(fd, " Auto Clear: %d\n", global_rtautoclear); + } + ast_cli(fd, "\n----\n"); + return RESULT_SUCCESS; +} + +/*! \brief Show subscription type in string format */ +static const char *subscription_type2str(enum subscriptiontype subtype) +{ + int i; + + for (i = 1; (i < (sizeof(subscription_types) / sizeof(subscription_types[0]))); i++) { + if (subscription_types[i].type == subtype) { + return subscription_types[i].text; + } + } + return subscription_types[0].text; +} + +/*! \brief Find subscription type in array */ +static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype) +{ + int i; + + for (i = 1; (i < (sizeof(subscription_types) / sizeof(subscription_types[0]))); i++) { + if (subscription_types[i].type == subtype) { + return &subscription_types[i]; + } + } + return &subscription_types[0]; +} + +/*! \brief Show active SIP channels */ +static int sip_show_channels(int fd, int argc, char *argv[]) +{ + return __sip_show_channels(fd, argc, argv, 0); +} + +/*! \brief Show active SIP subscriptions */ +static int sip_show_subscriptions(int fd, int argc, char *argv[]) +{ + return __sip_show_channels(fd, argc, argv, 1); +} + +/*! \brief SIP show channels CLI (main function) */ +static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions) +{ +#define FORMAT3 "%-15.15s %-10.10s %-11.11s %-15.15s %-13.13s %-15.15s %-10.10s\n" +#define FORMAT2 "%-15.15s %-10.10s %-11.11s %-11.11s %-15.15s %-7.7s %-15.15s\n" +#define FORMAT "%-15.15s %-10.10s %-11.11s %5.5d/%5.5d %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s\n" + struct sip_pvt *cur; + int numchans = 0; + char *referstatus = NULL; + + if (argc != 3) + return RESULT_SHOWUSAGE; + ast_mutex_lock(&iflock); + cur = iflist; + if (!subscriptions) + ast_cli(fd, FORMAT2, "Peer", "User/ANR", "Call ID", "Seq (Tx/Rx)", "Format", "Hold", "Last Message"); + else + ast_cli(fd, FORMAT3, "Peer", "User", "Call ID", "Extension", "Last state", "Type", "Mailbox"); + for (; cur; cur = cur->next) { + referstatus = ""; + if (cur->refer) { /* SIP transfer in progress */ + referstatus = referstatus2str(cur->refer->status); + } + if (cur->subscribed == NONE && !subscriptions) { + char formatbuf[SIPBUFSIZE/2]; + ast_cli(fd, FORMAT, ast_inet_ntoa(cur->sa.sin_addr), + S_OR(cur->username, S_OR(cur->cid_num, "(None)")), + cur->callid, + cur->ocseq, cur->icseq, + ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0), + ast_test_flag(&cur->flags[1], SIP_PAGE2_CALL_ONHOLD) ? "Yes" : "No", + ast_test_flag(&cur->flags[0], SIP_NEEDDESTROY) ? "(d)" : "", + cur->lastmsg , + referstatus + ); + numchans++; + } + if (cur->subscribed != NONE && subscriptions) { + ast_cli(fd, FORMAT3, ast_inet_ntoa(cur->sa.sin_addr), + S_OR(cur->username, S_OR(cur->cid_num, "(None)")), + cur->callid, + /* the 'complete' exten/context is hidden in the refer_to field for subscriptions */ + cur->subscribed == MWI_NOTIFICATION ? "--" : cur->subscribeuri, + cur->subscribed == MWI_NOTIFICATION ? "<none>" : ast_extension_state2str(cur->laststate), + subscription_type2str(cur->subscribed), + cur->subscribed == MWI_NOTIFICATION ? (cur->relatedpeer ? cur->relatedpeer->mailbox : "<none>") : "<none>" +); + numchans++; + } + } + ast_mutex_unlock(&iflock); + if (!subscriptions) + ast_cli(fd, "%d active SIP channel%s\n", numchans, (numchans != 1) ? "s" : ""); + else + ast_cli(fd, "%d active SIP subscription%s\n", numchans, (numchans != 1) ? "s" : ""); + return RESULT_SUCCESS; +#undef FORMAT +#undef FORMAT2 +#undef FORMAT3 +} + +/*! \brief Support routine for 'sip show channel' CLI */ +static char *complete_sipch(const char *line, const char *word, int pos, int state) +{ + int which=0; + struct sip_pvt *cur; + char *c = NULL; + int wordlen = strlen(word); + + if (pos != 3) { + return NULL; + } + + ast_mutex_lock(&iflock); + for (cur = iflist; cur; cur = cur->next) { + if (!strncasecmp(word, cur->callid, wordlen) && ++which > state) { + c = ast_strdup(cur->callid); + break; + } + } + ast_mutex_unlock(&iflock); + return c; +} + +/*! \brief Do completion on peer name */ +static char *complete_sip_peer(const char *word, int state, int flags2) +{ + char *result = NULL; + int wordlen = strlen(word); + int which = 0; + + ASTOBJ_CONTAINER_TRAVERSE(&peerl, !result, do { + /* locking of the object is not required because only the name and flags are being compared */ + if (!strncasecmp(word, iterator->name, wordlen) && + (!flags2 || ast_test_flag(&iterator->flags[1], flags2)) && + ++which > state) + result = ast_strdup(iterator->name); + } while(0) ); + return result; +} + +/*! \brief Support routine for 'sip show peer' CLI */ +static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state) +{ + if (pos == 3) + return complete_sip_peer(word, state, 0); + + return NULL; +} + +/*! \brief Support routine for 'sip debug peer' CLI */ +static char *complete_sip_debug_peer(const char *line, const char *word, int pos, int state) +{ + if (pos == 3) + return complete_sip_peer(word, state, 0); + + return NULL; +} + +/*! \brief Do completion on user name */ +static char *complete_sip_user(const char *word, int state, int flags2) +{ + char *result = NULL; + int wordlen = strlen(word); + int which = 0; + + ASTOBJ_CONTAINER_TRAVERSE(&userl, !result, do { + /* locking of the object is not required because only the name and flags are being compared */ + if (!strncasecmp(word, iterator->name, wordlen)) { + if (flags2 && !ast_test_flag(&iterator->flags[1], flags2)) + continue; + if (++which > state) { + result = ast_strdup(iterator->name); + } + } + } while(0) ); + return result; +} + +/*! \brief Support routine for 'sip show user' CLI */ +static char *complete_sip_show_user(const char *line, const char *word, int pos, int state) +{ + if (pos == 3) + return complete_sip_user(word, state, 0); + + return NULL; +} + +/*! \brief Support routine for 'sip notify' CLI */ +static char *complete_sipnotify(const char *line, const char *word, int pos, int state) +{ + char *c = NULL; + + if (pos == 2) { + int which = 0; + char *cat = NULL; + int wordlen = strlen(word); + + /* do completion for notify type */ + + if (!notify_types) + return NULL; + + while ( (cat = ast_category_browse(notify_types, cat)) ) { + if (!strncasecmp(word, cat, wordlen) && ++which > state) { + c = ast_strdup(cat); + break; + } + } + return c; + } + + if (pos > 2) + return complete_sip_peer(word, state, 0); + + return NULL; +} + +/*! \brief Support routine for 'sip prune realtime peer' CLI */ +static char *complete_sip_prune_realtime_peer(const char *line, const char *word, int pos, int state) +{ + if (pos == 4) + return complete_sip_peer(word, state, SIP_PAGE2_RTCACHEFRIENDS); + return NULL; +} + +/*! \brief Support routine for 'sip prune realtime user' CLI */ +static char *complete_sip_prune_realtime_user(const char *line, const char *word, int pos, int state) +{ + if (pos == 4) + return complete_sip_user(word, state, SIP_PAGE2_RTCACHEFRIENDS); + + return NULL; +} + +/*! \brief Show details of one active dialog */ +static int sip_show_channel(int fd, int argc, char *argv[]) +{ + struct sip_pvt *cur; + size_t len; + int found = 0; + + if (argc != 4) + return RESULT_SHOWUSAGE; + len = strlen(argv[3]); + ast_mutex_lock(&iflock); + for (cur = iflist; cur; cur = cur->next) { + if (!strncasecmp(cur->callid, argv[3], len)) { + char formatbuf[SIPBUFSIZE/2]; + ast_cli(fd,"\n"); + if (cur->subscribed != NONE) + ast_cli(fd, " * Subscription (type: %s)\n", subscription_type2str(cur->subscribed)); + else + ast_cli(fd, " * SIP Call\n"); + ast_cli(fd, " Curr. trans. direction: %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING) ? "Outgoing" : "Incoming"); + ast_cli(fd, " Call-ID: %s\n", cur->callid); + ast_cli(fd, " Owner channel ID: %s\n", cur->owner ? cur->owner->name : "<none>"); + ast_cli(fd, " Our Codec Capability: %d\n", cur->capability); + ast_cli(fd, " Non-Codec Capability (DTMF): %d\n", cur->noncodeccapability); + ast_cli(fd, " Their Codec Capability: %d\n", cur->peercapability); + ast_cli(fd, " Joint Codec Capability: %d\n", cur->jointcapability); + ast_cli(fd, " Format: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0) ); + ast_cli(fd, " MaxCallBR: %d kbps\n", cur->maxcallbitrate); + ast_cli(fd, " Theoretical Address: %s:%d\n", ast_inet_ntoa(cur->sa.sin_addr), ntohs(cur->sa.sin_port)); + ast_cli(fd, " Received Address: %s:%d\n", ast_inet_ntoa(cur->recv.sin_addr), ntohs(cur->recv.sin_port)); + ast_cli(fd, " SIP Transfer mode: %s\n", transfermode2str(cur->allowtransfer)); + ast_cli(fd, " NAT Support: %s\n", nat2str(ast_test_flag(&cur->flags[0], SIP_NAT))); + ast_cli(fd, " Audio IP: %s %s\n", ast_inet_ntoa(cur->redirip.sin_addr.s_addr ? cur->redirip.sin_addr : cur->ourip), cur->redirip.sin_addr.s_addr ? "(Outside bridge)" : "(local)" ); + ast_cli(fd, " Our Tag: %s\n", cur->tag); + ast_cli(fd, " Their Tag: %s\n", cur->theirtag); + ast_cli(fd, " SIP User agent: %s\n", cur->useragent); + if (!ast_strlen_zero(cur->username)) + ast_cli(fd, " Username: %s\n", cur->username); + if (!ast_strlen_zero(cur->peername)) + ast_cli(fd, " Peername: %s\n", cur->peername); + if (!ast_strlen_zero(cur->uri)) + ast_cli(fd, " Original uri: %s\n", cur->uri); + if (!ast_strlen_zero(cur->cid_num)) + ast_cli(fd, " Caller-ID: %s\n", cur->cid_num); + ast_cli(fd, " Need Destroy: %d\n", ast_test_flag(&cur->flags[0], SIP_NEEDDESTROY)); + ast_cli(fd, " Last Message: %s\n", cur->lastmsg); + ast_cli(fd, " Promiscuous Redir: %s\n", ast_test_flag(&cur->flags[0], SIP_PROMISCREDIR) ? "Yes" : "No"); + ast_cli(fd, " Route: %s\n", cur->route ? cur->route->hop : "N/A"); + ast_cli(fd, " DTMF Mode: %s\n", dtmfmode2str(ast_test_flag(&cur->flags[0], SIP_DTMF))); + ast_cli(fd, " SIP Options: "); + if (cur->sipoptions) { + int x; + for (x=0 ; (x < (sizeof(sip_options) / sizeof(sip_options[0]))); x++) { + if (cur->sipoptions & sip_options[x].id) + ast_cli(fd, "%s ", sip_options[x].text); + } + } else + ast_cli(fd, "(none)\n"); + ast_cli(fd, "\n\n"); + found++; + } + } + ast_mutex_unlock(&iflock); + if (!found) + ast_cli(fd, "No such SIP Call ID starting with '%s'\n", argv[3]); + return RESULT_SUCCESS; +} + +/*! \brief Show history details of one dialog */ +static int sip_show_history(int fd, int argc, char *argv[]) +{ + struct sip_pvt *cur; + size_t len; + int found = 0; + + if (argc != 4) + return RESULT_SHOWUSAGE; + if (!recordhistory) + ast_cli(fd, "\n***Note: History recording is currently DISABLED. Use 'sip history' to ENABLE.\n"); + len = strlen(argv[3]); + ast_mutex_lock(&iflock); + for (cur = iflist; cur; cur = cur->next) { + if (!strncasecmp(cur->callid, argv[3], len)) { + struct sip_history *hist; + int x = 0; + + ast_cli(fd,"\n"); + if (cur->subscribed != NONE) + ast_cli(fd, " * Subscription\n"); + else + ast_cli(fd, " * SIP Call\n"); + if (cur->history) + AST_LIST_TRAVERSE(cur->history, hist, list) + ast_cli(fd, "%d. %s\n", ++x, hist->event); + if (x == 0) + ast_cli(fd, "Call '%s' has no history\n", cur->callid); + found++; + } + } + ast_mutex_unlock(&iflock); + if (!found) + ast_cli(fd, "No such SIP Call ID starting with '%s'\n", argv[3]); + return RESULT_SUCCESS; +} + +/*! \brief Dump SIP history to debug log file at end of lifespan for SIP dialog */ +static void sip_dump_history(struct sip_pvt *dialog) +{ + int x = 0; + struct sip_history *hist; + static int errmsg = 0; + + if (!dialog) + return; + + if (!option_debug && !sipdebug) { + if (!errmsg) { + ast_log(LOG_NOTICE, "You must have debugging enabled (SIP or Asterisk) in order to dump SIP history.\n"); + errmsg = 1; + } + return; + } + + ast_log(LOG_DEBUG, "\n---------- SIP HISTORY for '%s' \n", dialog->callid); + if (dialog->subscribed) + ast_log(LOG_DEBUG, " * Subscription\n"); + else + ast_log(LOG_DEBUG, " * SIP Call\n"); + if (dialog->history) + AST_LIST_TRAVERSE(dialog->history, hist, list) + ast_log(LOG_DEBUG, " %-3.3d. %s\n", ++x, hist->event); + if (!x) + ast_log(LOG_DEBUG, "Call '%s' has no history\n", dialog->callid); + ast_log(LOG_DEBUG, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid); +} + + +/*! \brief Receive SIP INFO Message +\note Doesn't read the duration of the DTMF signal */ +static void handle_request_info(struct sip_pvt *p, struct sip_request *req) +{ + char buf[1024]; + unsigned int event; + const char *c = get_header(req, "Content-Type"); + + /* Need to check the media/type */ + if (!strcasecmp(c, "application/dtmf-relay") || + !strcasecmp(c, "application/vnd.nortelnetworks.digits")) { + unsigned int duration = 0; + + /* Try getting the "signal=" part */ + if (ast_strlen_zero(c = get_body(req, "Signal")) && ast_strlen_zero(c = get_body(req, "d"))) { + ast_log(LOG_WARNING, "Unable to retrieve DTMF signal from INFO message from %s\n", p->callid); + transmit_response(p, "200 OK", req); /* Should return error */ + return; + } else { + ast_copy_string(buf, c, sizeof(buf)); + } + + if (!ast_strlen_zero((c = get_body(req, "Duration")))) + duration = atoi(c); + if (!duration) + duration = 100; /* 100 ms */ + + if (!p->owner) { /* not a PBX call */ + transmit_response(p, "481 Call leg/transaction does not exist", req); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return; + } + + if (ast_strlen_zero(buf)) { + transmit_response(p, "200 OK", req); + return; + } + + if (buf[0] == '*') + event = 10; + else if (buf[0] == '#') + event = 11; + else if ((buf[0] >= 'A') && (buf[0] <= 'D')) + event = 12 + buf[0] - 'A'; + else + event = atoi(buf); + if (event == 16) { + /* send a FLASH event */ + struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH, }; + ast_queue_frame(p->owner, &f); + if (sipdebug) + ast_verbose("* DTMF-relay event received: FLASH\n"); + } else { + /* send a DTMF event */ + struct ast_frame f = { AST_FRAME_DTMF, }; + if (event < 10) { + f.subclass = '0' + event; + } else if (event < 11) { + f.subclass = '*'; + } else if (event < 12) { + f.subclass = '#'; + } else if (event < 16) { + f.subclass = 'A' + (event - 12); + } + f.len = duration; + ast_queue_frame(p->owner, &f); + if (sipdebug) + ast_verbose("* DTMF-relay event received: %c\n", f.subclass); + } + transmit_response(p, "200 OK", req); + return; + } else if (!strcasecmp(c, "application/media_control+xml")) { + /* Eh, we'll just assume it's a fast picture update for now */ + if (p->owner) + ast_queue_control(p->owner, AST_CONTROL_VIDUPDATE); + transmit_response(p, "200 OK", req); + return; + } else if (!ast_strlen_zero(c = get_header(req, "X-ClientCode"))) { + /* Client code (from SNOM phone) */ + if (ast_test_flag(&p->flags[0], SIP_USECLIENTCODE)) { + if (p->owner && p->owner->cdr) + ast_cdr_setuserfield(p->owner, c); + if (p->owner && ast_bridged_channel(p->owner) && ast_bridged_channel(p->owner)->cdr) + ast_cdr_setuserfield(ast_bridged_channel(p->owner), c); + transmit_response(p, "200 OK", req); + } else { + transmit_response(p, "403 Unauthorized", req); + } + return; + } else if (ast_strlen_zero(c = get_header(req, "Content-Length")) || !strcasecmp(c, "0")) { + /* This is probably just a packet making sure the signalling is still up, just send back a 200 OK */ + transmit_response(p, "200 OK", req); + return; + } + + /* Other type of INFO message, not really understood by Asterisk */ + /* if (get_msg_text(buf, sizeof(buf), req)) { */ + + ast_log(LOG_WARNING, "Unable to parse INFO message from %s. Content %s\n", p->callid, buf); + transmit_response(p, "415 Unsupported media type", req); + return; +} + +/*! \brief Enable SIP Debugging in CLI */ +static int sip_do_debug_ip(int fd, int argc, char *argv[]) +{ + struct hostent *hp; + struct ast_hostent ahp; + int port = 0; + char *p, *arg; + + /* sip set debug ip <ip> */ + if (argc != 5) + return RESULT_SHOWUSAGE; + p = arg = argv[4]; + strsep(&p, ":"); + if (p) + port = atoi(p); + hp = ast_gethostbyname(arg, &ahp); + if (hp == NULL) + return RESULT_SHOWUSAGE; + + debugaddr.sin_family = AF_INET; + memcpy(&debugaddr.sin_addr, hp->h_addr, sizeof(debugaddr.sin_addr)); + debugaddr.sin_port = htons(port); + if (port == 0) + ast_cli(fd, "SIP Debugging Enabled for IP: %s\n", ast_inet_ntoa(debugaddr.sin_addr)); + else + ast_cli(fd, "SIP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(debugaddr.sin_addr), port); + + ast_set_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE); + + return RESULT_SUCCESS; +} + +/*! \brief sip_do_debug_peer: Turn on SIP debugging with peer mask */ +static int sip_do_debug_peer(int fd, int argc, char *argv[]) +{ + struct sip_peer *peer; + if (argc != 5) + return RESULT_SHOWUSAGE; + peer = find_peer(argv[4], NULL, 1, 0); + if (peer) { + if (peer->addr.sin_addr.s_addr) { + debugaddr.sin_family = AF_INET; + debugaddr.sin_addr = peer->addr.sin_addr; + debugaddr.sin_port = peer->addr.sin_port; + ast_cli(fd, "SIP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(debugaddr.sin_addr), ntohs(debugaddr.sin_port)); + ast_set_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE); + } else + ast_cli(fd, "Unable to get IP address of peer '%s'\n", argv[4]); + ASTOBJ_UNREF(peer,sip_destroy_peer); + } else + ast_cli(fd, "No such peer '%s'\n", argv[4]); + return RESULT_SUCCESS; +} + +/*! \brief Turn on SIP debugging (CLI command) */ +static int sip_do_debug(int fd, int argc, char *argv[]) +{ + int oldsipdebug = sipdebug_console; + if (argc != 3) { + if (argc != 5) + return RESULT_SHOWUSAGE; + else if (strcmp(argv[3], "ip") == 0) + return sip_do_debug_ip(fd, argc, argv); + else if (strcmp(argv[3], "peer") == 0) + return sip_do_debug_peer(fd, argc, argv); + else + return RESULT_SHOWUSAGE; + } + ast_set_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE); + memset(&debugaddr, 0, sizeof(debugaddr)); + ast_cli(fd, "SIP Debugging %senabled\n", oldsipdebug ? "re-" : ""); + return RESULT_SUCCESS; +} + +static int sip_do_debug_deprecated(int fd, int argc, char *argv[]) +{ + int oldsipdebug = sipdebug_console; + char *newargv[6] = { "sip", "set", "debug", NULL }; + if (argc != 2) { + if (argc != 4) + return RESULT_SHOWUSAGE; + else if (strcmp(argv[2], "ip") == 0) { + newargv[3] = argv[2]; + newargv[4] = argv[3]; + return sip_do_debug_ip(fd, argc + 1, newargv); + } else if (strcmp(argv[2], "peer") == 0) { + newargv[3] = argv[2]; + newargv[4] = argv[3]; + return sip_do_debug_peer(fd, argc + 1, newargv); + } else + return RESULT_SHOWUSAGE; + } + ast_set_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE); + memset(&debugaddr, 0, sizeof(debugaddr)); + ast_cli(fd, "SIP Debugging %senabled\n", oldsipdebug ? "re-" : ""); + return RESULT_SUCCESS; +} + +/*! \brief Cli command to send SIP notify to peer */ +static int sip_notify(int fd, int argc, char *argv[]) +{ + struct ast_variable *varlist; + int i; + + if (argc < 4) + return RESULT_SHOWUSAGE; + + if (!notify_types) { + ast_cli(fd, "No %s file found, or no types listed there\n", notify_config); + return RESULT_FAILURE; + } + + varlist = ast_variable_browse(notify_types, argv[2]); + + if (!varlist) { + ast_cli(fd, "Unable to find notify type '%s'\n", argv[2]); + return RESULT_FAILURE; + } + + for (i = 3; i < argc; i++) { + struct sip_pvt *p; + struct sip_request req; + struct ast_variable *var; + + if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY))) { + ast_log(LOG_WARNING, "Unable to build sip pvt data for notify (memory/socket error)\n"); + return RESULT_FAILURE; + } + + if (create_addr(p, argv[i])) { + /* Maybe they're not registered, etc. */ + sip_destroy(p); + ast_cli(fd, "Could not create address for '%s'\n", argv[i]); + continue; + } + + initreqprep(&req, p, SIP_NOTIFY); + + for (var = varlist; var; var = var->next) + add_header(&req, var->name, ast_unescape_semicolon(var->value)); + + /* Recalculate our side, and recalculate Call ID */ + if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip)) + p->ourip = __ourip; + build_via(p); + build_callid_pvt(p); + ast_cli(fd, "Sending NOTIFY of type '%s' to '%s'\n", argv[2], argv[i]); + transmit_sip_request(p, &req); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } + + return RESULT_SUCCESS; +} + +/*! \brief Disable SIP Debugging in CLI */ +static int sip_no_debug(int fd, int argc, char *argv[]) +{ + if (argc != 4) + return RESULT_SHOWUSAGE; + ast_clear_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE); + ast_cli(fd, "SIP Debugging Disabled\n"); + return RESULT_SUCCESS; +} + +static int sip_no_debug_deprecated(int fd, int argc, char *argv[]) +{ + if (argc != 3) + return RESULT_SHOWUSAGE; + ast_clear_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE); + ast_cli(fd, "SIP Debugging Disabled\n"); + return RESULT_SUCCESS; +} + +/*! \brief Enable SIP History logging (CLI) */ +static int sip_do_history(int fd, int argc, char *argv[]) +{ + if (argc != 2) { + return RESULT_SHOWUSAGE; + } + recordhistory = TRUE; + ast_cli(fd, "SIP History Recording Enabled (use 'sip show history')\n"); + return RESULT_SUCCESS; +} + +/*! \brief Disable SIP History logging (CLI) */ +static int sip_no_history(int fd, int argc, char *argv[]) +{ + if (argc != 3) { + return RESULT_SHOWUSAGE; + } + recordhistory = FALSE; + ast_cli(fd, "SIP History Recording Disabled\n"); + return RESULT_SUCCESS; +} + +/*! \brief Authenticate for outbound registration */ +static int do_register_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader) +{ + char digest[1024]; + p->authtries++; + memset(digest,0,sizeof(digest)); + if (reply_digest(p, req, header, SIP_REGISTER, digest, sizeof(digest))) { + /* There's nothing to use for authentication */ + /* No digest challenge in request */ + if (sip_debug_test_pvt(p) && p->registry) + ast_verbose("No authentication challenge, sending blank registration to domain/host name %s\n", p->registry->hostname); + /* No old challenge */ + return -1; + } + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) + append_history(p, "RegistryAuth", "Try: %d", p->authtries); + if (sip_debug_test_pvt(p) && p->registry) + ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname); + return transmit_register(p->registry, SIP_REGISTER, digest, respheader); +} + +/*! \brief Add authentication on outbound SIP packet */ +static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, int sipmethod, int init) +{ + char digest[1024]; + + if (!p->options && !(p->options = ast_calloc(1, sizeof(*p->options)))) + return -2; + + p->authtries++; + if (option_debug > 1) + ast_log(LOG_DEBUG, "Auth attempt %d on %s\n", p->authtries, sip_methods[sipmethod].text); + memset(digest, 0, sizeof(digest)); + if (reply_digest(p, req, header, sipmethod, digest, sizeof(digest) )) { + /* No way to authenticate */ + return -1; + } + /* Now we have a reply digest */ + p->options->auth = digest; + p->options->authheader = respheader; + return transmit_invite(p, sipmethod, sipmethod == SIP_INVITE, init); +} + +/*! \brief reply to authentication for outbound registrations +\return Returns -1 if we have no auth +\note This is used for register= servers in sip.conf, SIP proxies we register + with for receiving calls from. */ +static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len) +{ + char tmp[512]; + char *c; + char oldnonce[256]; + + /* table of recognised keywords, and places where they should be copied */ + const struct x { + const char *key; + int field_index; + } *i, keys[] = { + { "realm=", ast_string_field_index(p, realm) }, + { "nonce=", ast_string_field_index(p, nonce) }, + { "opaque=", ast_string_field_index(p, opaque) }, + { "qop=", ast_string_field_index(p, qop) }, + { "domain=", ast_string_field_index(p, domain) }, + { NULL, 0 }, + }; + + ast_copy_string(tmp, get_header(req, header), sizeof(tmp)); + if (ast_strlen_zero(tmp)) + return -1; + if (strncasecmp(tmp, "Digest ", strlen("Digest "))) { + ast_log(LOG_WARNING, "missing Digest.\n"); + return -1; + } + c = tmp + strlen("Digest "); + ast_copy_string(oldnonce, p->nonce, sizeof(oldnonce)); + while (c && *(c = ast_skip_blanks(c))) { /* lookup for keys */ + for (i = keys; i->key != NULL; i++) { + char *src, *separator; + if (strncasecmp(c, i->key, strlen(i->key)) != 0) + continue; + /* Found. Skip keyword, take text in quotes or up to the separator. */ + c += strlen(i->key); + if (*c == '"') { + src = ++c; + separator = "\""; + } else { + src = c; + separator = ","; + } + strsep(&c, separator); /* clear separator and move ptr */ + ast_string_field_index_set(p, i->field_index, src); + break; + } + if (i->key == NULL) /* not found, try ',' */ + strsep(&c, ","); + } + /* Reset nonce count */ + if (strcmp(p->nonce, oldnonce)) + p->noncecount = 0; + + /* Save auth data for following registrations */ + if (p->registry) { + struct sip_registry *r = p->registry; + + if (strcmp(r->nonce, p->nonce)) { + ast_string_field_set(r, realm, p->realm); + ast_string_field_set(r, nonce, p->nonce); + ast_string_field_set(r, domain, p->domain); + ast_string_field_set(r, opaque, p->opaque); + ast_string_field_set(r, qop, p->qop); + r->noncecount = 0; + } + } + return build_reply_digest(p, sipmethod, digest, digest_len); +} + +/*! \brief Build reply digest +\return Returns -1 if we have no auth +\note Build digest challenge for authentication of peers (for registration) + and users (for calls). Also used for authentication of CANCEL and BYE +*/ +static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int digest_len) +{ + char a1[256]; + char a2[256]; + char a1_hash[256]; + char a2_hash[256]; + char resp[256]; + char resp_hash[256]; + char uri[256]; + char opaque[256] = ""; + char cnonce[80]; + const char *username; + const char *secret; + const char *md5secret; + struct sip_auth *auth = NULL; /* Realm authentication */ + + if (!ast_strlen_zero(p->domain)) + ast_copy_string(uri, p->domain, sizeof(uri)); + else if (!ast_strlen_zero(p->uri)) + ast_copy_string(uri, p->uri, sizeof(uri)); + else + snprintf(uri, sizeof(uri), "sip:%s@%s",p->username, ast_inet_ntoa(p->sa.sin_addr)); + + snprintf(cnonce, sizeof(cnonce), "%08lx", ast_random()); + + /* Check if we have separate auth credentials */ + if(!(auth = find_realm_authentication(p->peerauth, p->realm))) /* Start with peer list */ + auth = find_realm_authentication(authl, p->realm); /* If not, global list */ + + if (auth) { + ast_log(LOG_DEBUG, "use realm [%s] from peer [%s][%s]\n", auth->username, p->peername, p->username); + username = auth->username; + secret = auth->secret; + md5secret = auth->md5secret; + if (sipdebug) + ast_log(LOG_DEBUG,"Using realm %s authentication for call %s\n", p->realm, p->callid); + } else { + /* No authentication, use peer or register= config */ + username = p->authname; + secret = p->peersecret; + md5secret = p->peermd5secret; + } + if (ast_strlen_zero(username)) /* We have no authentication */ + return -1; + + /* Calculate SIP digest response */ + snprintf(a1,sizeof(a1),"%s:%s:%s", username, p->realm, secret); + snprintf(a2,sizeof(a2),"%s:%s", sip_methods[method].text, uri); + if (!ast_strlen_zero(md5secret)) + ast_copy_string(a1_hash, md5secret, sizeof(a1_hash)); + else + ast_md5_hash(a1_hash,a1); + ast_md5_hash(a2_hash,a2); + + p->noncecount++; + if (!ast_strlen_zero(p->qop)) + snprintf(resp,sizeof(resp),"%s:%s:%08x:%s:%s:%s", a1_hash, p->nonce, p->noncecount, cnonce, "auth", a2_hash); + else + snprintf(resp,sizeof(resp),"%s:%s:%s", a1_hash, p->nonce, a2_hash); + ast_md5_hash(resp_hash, resp); + + /* only include the opaque string if it's set */ + if (!ast_strlen_zero(p->opaque)) { + snprintf(opaque, sizeof(opaque), ", opaque=\"%s\"", p->opaque); + } + + /* XXX We hard code our qop to "auth" for now. XXX */ + if (!ast_strlen_zero(p->qop)) + snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s, qop=auth, cnonce=\"%s\", nc=%08x", username, p->realm, uri, p->nonce, resp_hash, opaque, cnonce, p->noncecount); + else + snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s", username, p->realm, uri, p->nonce, resp_hash, opaque); + + append_history(p, "AuthResp", "Auth response sent for %s in realm %s - nc %d", username, p->realm, p->noncecount); + + return 0; +} + +static char show_domains_usage[] = +"Usage: sip show domains\n" +" Lists all configured SIP local domains.\n" +" Asterisk only responds to SIP messages to local domains.\n"; + +static char notify_usage[] = +"Usage: sip notify <type> <peer> [<peer>...]\n" +" Send a NOTIFY message to a SIP peer or peers\n" +" Message types are defined in sip_notify.conf\n"; + +static char show_users_usage[] = +"Usage: sip show users [like <pattern>]\n" +" Lists all known SIP users.\n" +" Optional regular expression pattern is used to filter the user list.\n"; + +static char show_user_usage[] = +"Usage: sip show user <name> [load]\n" +" Shows all details on one SIP user and the current status.\n" +" Option \"load\" forces lookup of peer in realtime storage.\n"; + +static char show_inuse_usage[] = +"Usage: sip show inuse [all]\n" +" List all SIP users and peers usage counters and limits.\n" +" Add option \"all\" to show all devices, not only those with a limit.\n"; + +static char show_channels_usage[] = +"Usage: sip show channels\n" +" Lists all currently active SIP channels.\n"; + +static char show_channel_usage[] = +"Usage: sip show channel <channel>\n" +" Provides detailed status on a given SIP channel.\n"; + +static char show_history_usage[] = +"Usage: sip show history <channel>\n" +" Provides detailed dialog history on a given SIP channel.\n"; + +static char show_peers_usage[] = +"Usage: sip show peers [like <pattern>]\n" +" Lists all known SIP peers.\n" +" Optional regular expression pattern is used to filter the peer list.\n"; + +static char show_peer_usage[] = +"Usage: sip show peer <name> [load]\n" +" Shows all details on one SIP peer and the current status.\n" +" Option \"load\" forces lookup of peer in realtime storage.\n"; + +static char prune_realtime_usage[] = +"Usage: sip prune realtime [peer|user] [<name>|all|like <pattern>]\n" +" Prunes object(s) from the cache.\n" +" Optional regular expression pattern is used to filter the objects.\n"; + +static char show_reg_usage[] = +"Usage: sip show registry\n" +" Lists all registration requests and status.\n"; + +static char debug_usage[] = +"Usage: sip set debug\n" +" Enables dumping of SIP packets for debugging purposes\n\n" +" sip set debug ip <host[:PORT]>\n" +" Enables dumping of SIP packets to and from host.\n\n" +" sip set debug peer <peername>\n" +" Enables dumping of SIP packets to and from host.\n" +" Require peer to be registered.\n"; + +static char no_debug_usage[] = +"Usage: sip set debug off\n" +" Disables dumping of SIP packets for debugging purposes\n"; + +static char no_history_usage[] = +"Usage: sip history off\n" +" Disables recording of SIP dialog history for debugging purposes\n"; + +static char history_usage[] = +"Usage: sip history\n" +" Enables recording of SIP dialog history for debugging purposes.\n" +"Use 'sip show history' to view the history of a call number.\n"; + +static char sip_reload_usage[] = +"Usage: sip reload\n" +" Reloads SIP configuration from sip.conf\n"; + +static char show_subscriptions_usage[] = +"Usage: sip show subscriptions\n" +" Lists active SIP subscriptions for extension states\n"; + +static char show_objects_usage[] = +"Usage: sip show objects\n" +" Lists status of known SIP objects\n"; + +static char show_settings_usage[] = +"Usage: sip show settings\n" +" Provides detailed list of the configuration of the SIP channel.\n"; + +/*! \brief Read SIP header (dialplan function) */ +static int func_header_read(struct ast_channel *chan, char *function, char *data, char *buf, size_t len) +{ + struct sip_pvt *p; + const char *content = NULL; + AST_DECLARE_APP_ARGS(args, + AST_APP_ARG(header); + AST_APP_ARG(number); + ); + int i, number, start = 0; + + if (ast_strlen_zero(data)) { + ast_log(LOG_WARNING, "This function requires a header name.\n"); + return -1; + } + + ast_channel_lock(chan); + if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) { + ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n"); + ast_channel_unlock(chan); + return -1; + } + + AST_STANDARD_APP_ARGS(args, data); + if (!args.number) { + number = 1; + } else { + sscanf(args.number, "%d", &number); + if (number < 1) + number = 1; + } + + p = chan->tech_pvt; + + /* If there is no private structure, this channel is no longer alive */ + if (!p) { + ast_channel_unlock(chan); + return -1; + } + + for (i = 0; i < number; i++) + content = __get_header(&p->initreq, args.header, &start); + + if (ast_strlen_zero(content)) { + ast_channel_unlock(chan); + return -1; + } + + ast_copy_string(buf, content, len); + ast_channel_unlock(chan); + + return 0; +} + +static struct ast_custom_function sip_header_function = { + .name = "SIP_HEADER", + .synopsis = "Gets the specified SIP header", + .syntax = "SIP_HEADER(<name>[,<number>])", + .desc = "Since there are several headers (such as Via) which can occur multiple\n" + "times, SIP_HEADER takes an optional second argument to specify which header with\n" + "that name to retrieve. Headers start at offset 1.\n", + .read = func_header_read, +}; + +/*! \brief Dial plan function to check if domain is local */ +static int func_check_sipdomain(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) +{ + if (ast_strlen_zero(data)) { + ast_log(LOG_WARNING, "CHECKSIPDOMAIN requires an argument - A domain name\n"); + return -1; + } + if (check_sip_domain(data, NULL, 0)) + ast_copy_string(buf, data, len); + else + buf[0] = '\0'; + return 0; +} + +static struct ast_custom_function checksipdomain_function = { + .name = "CHECKSIPDOMAIN", + .synopsis = "Checks if domain is a local domain", + .syntax = "CHECKSIPDOMAIN(<domain|IP>)", + .read = func_check_sipdomain, + .desc = "This function checks if the domain in the argument is configured\n" + "as a local SIP domain that this Asterisk server is configured to handle.\n" + "Returns the domain name if it is locally handled, otherwise an empty string.\n" + "Check the domain= configuration in sip.conf\n", +}; + +/*! \brief ${SIPPEER()} Dialplan function - reads peer data */ +static int function_sippeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) +{ + struct sip_peer *peer; + char *colname; + + if ((colname = strchr(data, ':'))) /*! \todo Will be deprecated after 1.4 */ + *colname++ = '\0'; + else if ((colname = strchr(data, '|'))) + *colname++ = '\0'; + else + colname = "ip"; + + if (!(peer = find_peer(data, NULL, 1, 0))) + return -1; + + if (!strcasecmp(colname, "ip")) { + ast_copy_string(buf, peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "", len); + } else if (!strcasecmp(colname, "status")) { + peer_status(peer, buf, len); + } else if (!strcasecmp(colname, "language")) { + ast_copy_string(buf, peer->language, len); + } else if (!strcasecmp(colname, "regexten")) { + ast_copy_string(buf, peer->regexten, len); + } else if (!strcasecmp(colname, "limit")) { + snprintf(buf, len, "%d", peer->call_limit); + } else if (!strcasecmp(colname, "curcalls")) { + snprintf(buf, len, "%d", peer->inUse); + } else if (!strcasecmp(colname, "accountcode")) { + ast_copy_string(buf, peer->accountcode, len); + } else if (!strcasecmp(colname, "useragent")) { + ast_copy_string(buf, peer->useragent, len); + } else if (!strcasecmp(colname, "mailbox")) { + ast_copy_string(buf, peer->mailbox, len); + } else if (!strcasecmp(colname, "context")) { + ast_copy_string(buf, peer->context, len); + } else if (!strcasecmp(colname, "expire")) { + snprintf(buf, len, "%d", peer->expire); + } else if (!strcasecmp(colname, "dynamic")) { + ast_copy_string(buf, (ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC) ? "yes" : "no"), len); + } else if (!strcasecmp(colname, "callerid_name")) { + ast_copy_string(buf, peer->cid_name, len); + } else if (!strcasecmp(colname, "callerid_num")) { + ast_copy_string(buf, peer->cid_num, len); + } else if (!strcasecmp(colname, "codecs")) { + ast_getformatname_multiple(buf, len -1, peer->capability); + } else if (!strncasecmp(colname, "codec[", 6)) { + char *codecnum; + int index = 0, codec = 0; + + codecnum = colname + 6; /* move past the '[' */ + codecnum = strsep(&codecnum, "]"); /* trim trailing ']' if any */ + index = atoi(codecnum); + if((codec = ast_codec_pref_index(&peer->prefs, index))) { + ast_copy_string(buf, ast_getformatname(codec), len); + } else { + buf[0] = '\0'; + } + } else { + buf[0] = '\0'; + } + + ASTOBJ_UNREF(peer, sip_destroy_peer); + + return 0; +} + +/*! \brief Structure to declare a dialplan function: SIPPEER */ +struct ast_custom_function sippeer_function = { + .name = "SIPPEER", + .synopsis = "Gets SIP peer information", + .syntax = "SIPPEER(<peername>[|item])", + .read = function_sippeer, + .desc = "Valid items are:\n" + "- ip (default) The IP address.\n" + "- mailbox The configured mailbox.\n" + "- context The configured context.\n" + "- expire The epoch time of the next expire.\n" + "- dynamic Is it dynamic? (yes/no).\n" + "- callerid_name The configured Caller ID name.\n" + "- callerid_num The configured Caller ID number.\n" + "- codecs The configured codecs.\n" + "- status Status (if qualify=yes).\n" + "- regexten Registration extension\n" + "- limit Call limit (call-limit)\n" + "- curcalls Current amount of calls \n" + " Only available if call-limit is set\n" + "- language Default language for peer\n" + "- accountcode Account code for this peer\n" + "- useragent Current user agent id for peer\n" + "- codec[x] Preferred codec index number 'x' (beginning with zero).\n" + "\n" +}; + +/*! \brief ${SIPCHANINFO()} Dialplan function - reads sip channel data */ +static int function_sipchaninfo_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) +{ + struct sip_pvt *p; + + *buf = 0; + + if (!data) { + ast_log(LOG_WARNING, "This function requires a parameter name.\n"); + return -1; + } + + ast_channel_lock(chan); + if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) { + ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n"); + ast_channel_unlock(chan); + return -1; + } + + p = chan->tech_pvt; + + /* If there is no private structure, this channel is no longer alive */ + if (!p) { + ast_channel_unlock(chan); + return -1; + } + + if (!strcasecmp(data, "peerip")) { + ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(p->sa.sin_addr) : "", len); + } else if (!strcasecmp(data, "recvip")) { + ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(p->recv.sin_addr) : "", len); + } else if (!strcasecmp(data, "from")) { + ast_copy_string(buf, p->from, len); + } else if (!strcasecmp(data, "uri")) { + ast_copy_string(buf, p->uri, len); + } else if (!strcasecmp(data, "useragent")) { + ast_copy_string(buf, p->useragent, len); + } else if (!strcasecmp(data, "peername")) { + ast_copy_string(buf, p->peername, len); + } else if (!strcasecmp(data, "t38passthrough")) { + if (p->t38.state == T38_DISABLED) + ast_copy_string(buf, "0", sizeof("0")); + else /* T38 is offered or enabled in this call */ + ast_copy_string(buf, "1", sizeof("1")); + } else { + ast_channel_unlock(chan); + return -1; + } + ast_channel_unlock(chan); + + return 0; +} + +/*! \brief Structure to declare a dialplan function: SIPCHANINFO */ +static struct ast_custom_function sipchaninfo_function = { + .name = "SIPCHANINFO", + .synopsis = "Gets the specified SIP parameter from the current channel", + .syntax = "SIPCHANINFO(item)", + .read = function_sipchaninfo_read, + .desc = "Valid items are:\n" + "- peerip The IP address of the peer.\n" + "- recvip The source IP address of the peer.\n" + "- from The URI from the From: header.\n" + "- uri The URI from the Contact: header.\n" + "- useragent The useragent.\n" + "- peername The name of the peer.\n" + "- t38passthrough 1 if T38 is offered or enabled in this channel, otherwise 0\n" +}; + +/*! \brief Parse 302 Moved temporalily response */ +static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req) +{ + char tmp[SIPBUFSIZE]; + char *s, *e, *uri, *t; + char *domain; + + ast_copy_string(tmp, get_header(req, "Contact"), sizeof(tmp)); + if ((t = strchr(tmp, ','))) + *t = '\0'; + s = get_in_brackets(tmp); + uri = ast_strdupa(s); + if (ast_test_flag(&p->flags[0], SIP_PROMISCREDIR)) { + if (!strncasecmp(s, "sip:", 4)) + s += 4; + e = strchr(s, ';'); + if (e) + *e = '\0'; + if (option_debug) + ast_log(LOG_DEBUG, "Found promiscuous redirection to 'SIP/%s'\n", s); + if (p->owner) + ast_string_field_build(p->owner, call_forward, "SIP/%s", s); + } else { + e = strchr(tmp, '@'); + if (e) { + *e++ = '\0'; + domain = e; + } else { + /* No username part */ + domain = tmp; + } + e = strchr(s, ';'); /* Strip of parameters in the username part */ + if (e) + *e = '\0'; + e = strchr(domain, ';'); /* Strip of parameters in the domain part */ + if (e) + *e = '\0'; + + if (!strncasecmp(s, "sip:", 4)) + s += 4; + if (option_debug > 1) + ast_log(LOG_DEBUG, "Received 302 Redirect to extension '%s' (domain %s)\n", s, domain); + if (p->owner) { + pbx_builtin_setvar_helper(p->owner, "SIPREDIRECTURI", uri); + pbx_builtin_setvar_helper(p->owner, "SIPDOMAIN", domain); + ast_string_field_set(p->owner, call_forward, s); + } + } +} + +/*! \brief Check pending actions on SIP call */ +static void check_pendings(struct sip_pvt *p) +{ + if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { + /* if we can't BYE, then this is really a pending CANCEL */ + if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA) + transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE); + /* Actually don't destroy us yet, wait for the 487 on our original + INVITE, but do set an autodestruct just in case we never get it. */ + else { + /* We have a pending outbound invite, don't send someting + new in-transaction */ + if (p->pendinginvite) + return; + + /* Perhaps there is an SD change INVITE outstanding */ + transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE); + } + ast_clear_flag(&p->flags[0], SIP_PENDINGBYE); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) { + /* if we can't REINVITE, hold it for later */ + if (p->pendinginvite || p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA || p->waitid > 0) { + if (option_debug) + ast_log(LOG_DEBUG, "NOT Sending pending reinvite (yet) on '%s'\n", p->callid); + } else { + if (option_debug) + ast_log(LOG_DEBUG, "Sending pending reinvite on '%s'\n", p->callid); + /* Didn't get to reinvite yet, so do it now */ + transmit_reinvite_with_sdp(p); + ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE); + } + } +} + +/*! \brief Reset the NEEDREINVITE flag after waiting when we get 491 on a Re-invite + to avoid race conditions between asterisk servers. + Called from the scheduler. +*/ +static int sip_reinvite_retry(const void *data) +{ + struct sip_pvt *p = (struct sip_pvt *) data; + + ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); + p->waitid = -1; + return 0; +} + + +/*! \brief Handle SIP response to INVITE dialogue */ +static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno) +{ + int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING); + int res = 0; + int xmitres = 0; + int reinvite = (p->owner && p->owner->_state == AST_STATE_UP); + struct ast_channel *bridgepeer = NULL; + + if (option_debug > 3) { + if (reinvite) + ast_log(LOG_DEBUG, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid); + else + ast_log(LOG_DEBUG, "SIP response %d to standard invite\n", resp); + } + + if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) { /* This call is already gone */ + if (option_debug) + ast_log(LOG_DEBUG, "Got response on call that is already terminated: %s (ignoring)\n", p->callid); + return; + } + + /* Acknowledge sequence number - This only happens on INVITE from SIP-call */ + /* Don't auto congest anymore since we've gotten something useful back */ + AST_SCHED_DEL(sched, p->initid); + + /* RFC3261 says we must treat every 1xx response (but not 100) + that we don't recognize as if it was 183. + */ + if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 182 && resp != 183) + resp = 183; + + /* Any response between 100 and 199 is PROCEEDING */ + if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING) + p->invitestate = INV_PROCEEDING; + + /* Final response, not 200 ? */ + if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA )) + p->invitestate = INV_COMPLETED; + + + switch (resp) { + case 100: /* Trying */ + case 101: /* Dialog establishment */ + if (!ast_test_flag(req, SIP_PKT_IGNORE) && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + check_pendings(p); + break; + + case 180: /* 180 Ringing */ + case 182: /* 182 Queued */ + if (!ast_test_flag(req, SIP_PKT_IGNORE) && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) { + ast_queue_control(p->owner, AST_CONTROL_RINGING); + if (p->owner->_state != AST_STATE_UP) { + ast_setstate(p->owner, AST_STATE_RINGING); + } + } + if (find_sdp(req)) { + if (p->invitestate != INV_CANCELLED) + p->invitestate = INV_EARLY_MEDIA; + res = process_sdp(p, req); + if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) { + /* Queue a progress frame only if we have SDP in 180 or 182 */ + ast_queue_control(p->owner, AST_CONTROL_PROGRESS); + } + } + check_pendings(p); + break; + + case 183: /* Session progress */ + if (!ast_test_flag(req, SIP_PKT_IGNORE) && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + /* Ignore 183 Session progress without SDP */ + if (find_sdp(req)) { + if (p->invitestate != INV_CANCELLED) + p->invitestate = INV_EARLY_MEDIA; + res = process_sdp(p, req); + if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) { + /* Queue a progress frame */ + ast_queue_control(p->owner, AST_CONTROL_PROGRESS); + } + } + check_pendings(p); + break; + + case 200: /* 200 OK on invite - someone's answering our call */ + if (!ast_test_flag(req, SIP_PKT_IGNORE) && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + p->authtries = 0; + if (find_sdp(req)) { + if ((res = process_sdp(p, req)) && !ast_test_flag(req, SIP_PKT_IGNORE)) + if (!reinvite) + /* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */ + /* For re-invites, we try to recover */ + ast_set_flag(&p->flags[0], SIP_PENDINGBYE); + } + + /* Parse contact header for continued conversation */ + /* When we get 200 OK, we know which device (and IP) to contact for this call */ + /* This is important when we have a SIP proxy between us and the phone */ + if (outgoing) { + update_call_counter(p, DEC_CALL_RINGING); + parse_ok_contact(p, req); + /* Save Record-Route for any later requests we make on this dialogue */ + if (!reinvite) + build_route(p, req, 1); + + if(set_address_from_contact(p)) { + /* Bad contact - we don't know how to reach this device */ + /* We need to ACK, but then send a bye */ + if (!p->route && !ast_test_flag(req, SIP_PKT_IGNORE)) + ast_set_flag(&p->flags[0], SIP_PENDINGBYE); + } + + } + + if (p->owner && (p->owner->_state == AST_STATE_UP) && (bridgepeer = ast_bridged_channel(p->owner))) { /* if this is a re-invite */ + struct sip_pvt *bridgepvt = NULL; + + if (!bridgepeer->tech) { + ast_log(LOG_WARNING, "Ooooh.. no tech! That's REALLY bad\n"); + break; + } + if (bridgepeer->tech == &sip_tech || bridgepeer->tech == &sip_tech_info) { + bridgepvt = (struct sip_pvt*)(bridgepeer->tech_pvt); + if (bridgepvt->udptl) { + if (p->t38.state == T38_PEER_REINVITE) { + sip_handle_t38_reinvite(bridgepeer, p, 0); + ast_rtp_set_rtptimers_onhold(p->rtp); + if (p->vrtp) + ast_rtp_set_rtptimers_onhold(p->vrtp); /* Turn off RTP timers while we send fax */ + } else if (p->t38.state == T38_DISABLED && bridgepeer && (bridgepvt->t38.state == T38_ENABLED)) { + ast_log(LOG_WARNING, "RTP re-invite after T38 session not handled yet !\n"); + /* Insted of this we should somehow re-invite the other side of the bridge to RTP */ + /* XXXX Should we really destroy this session here, without any response at all??? */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } + } else { + if (option_debug > 1) + ast_log(LOG_DEBUG, "Strange... The other side of the bridge does not have a udptl struct\n"); + ast_mutex_lock(&bridgepvt->lock); + bridgepvt->t38.state = T38_DISABLED; + ast_mutex_unlock(&bridgepvt->lock); + if (option_debug) + ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->tech->type); + p->t38.state = T38_DISABLED; + if (option_debug > 1) + ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + } + } else { + /* Other side is not a SIP channel */ + if (option_debug > 1) + ast_log(LOG_DEBUG, "Strange... The other side of the bridge is not a SIP channel\n"); + p->t38.state = T38_DISABLED; + if (option_debug > 1) + ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + } + } + if ((p->t38.state == T38_LOCAL_REINVITE) || (p->t38.state == T38_LOCAL_DIRECT)) { + /* If there was T38 reinvite and we are supposed to answer with 200 OK than this should set us to T38 negotiated mode */ + p->t38.state = T38_ENABLED; + if (option_debug) + ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + } + + if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) { + if (!reinvite) { + ast_queue_control(p->owner, AST_CONTROL_ANSWER); + } else { /* RE-invite */ + ast_queue_frame(p->owner, &ast_null_frame); + } + } else { + /* It's possible we're getting an 200 OK after we've tried to disconnect + by sending CANCEL */ + /* First send ACK, then send bye */ + if (!ast_test_flag(req, SIP_PKT_IGNORE)) + ast_set_flag(&p->flags[0], SIP_PENDINGBYE); + } + /* If I understand this right, the branch is different for a non-200 ACK only */ + p->invitestate = INV_TERMINATED; + ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE); + check_pendings(p); + break; + case 407: /* Proxy authentication */ + case 401: /* Www auth */ + /* First we ACK */ + xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE); + if (p->options) + p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH); + + /* Then we AUTH */ + ast_string_field_free(p, theirtag); /* forget their old tag, so we don't match tags when getting response */ + if (!ast_test_flag(req, SIP_PKT_IGNORE)) { + char *authenticate = (resp == 401 ? "WWW-Authenticate" : "Proxy-Authenticate"); + char *authorization = (resp == 401 ? "Authorization" : "Proxy-Authorization"); + if (p->authtries < MAX_AUTHTRIES) + p->invitestate = INV_CALLING; + if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, authenticate, authorization, SIP_INVITE, 1)) { + ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From")); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + sip_alreadygone(p); + if (p->owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + } + } + break; + + case 403: /* Forbidden */ + /* First we ACK */ + xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE); + ast_log(LOG_WARNING, "Received response: \"Forbidden\" from '%s'\n", get_header(&p->initreq, "From")); + if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + sip_alreadygone(p); + break; + + case 404: /* Not found */ + xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE); + if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE)) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + sip_alreadygone(p); + break; + + case 408: /* Request timeout */ + case 481: /* Call leg does not exist */ + /* Could be REFER caused INVITE with replaces */ + ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid); + xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE); + if (p->owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + break; + case 487: /* Cancelled transaction */ + /* We have sent CANCEL on an outbound INVITE + This transaction is already scheduled to be killed by sip_hangup(). + */ + xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE); + if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE)) { + ast_queue_hangup(p->owner); + append_history(p, "Hangup", "Got 487 on CANCEL request from us. Queued AST hangup request"); + } else if (!ast_test_flag(req, SIP_PKT_IGNORE)) { + update_call_counter(p, DEC_CALL_LIMIT); + append_history(p, "Hangup", "Got 487 on CANCEL request from us on call without owner. Killing this dialog."); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + sip_alreadygone(p); + } + break; + case 488: /* Not acceptable here */ + xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE); + if (reinvite && p->udptl) { + /* If this is a T.38 call, we should go back to + audio. If this is an audio call - something went + terribly wrong since we don't renegotiate codecs, + only IP/port . + */ + p->t38.state = T38_DISABLED; + /* Try to reset RTP timers */ + ast_rtp_set_rtptimers_onhold(p->rtp); + ast_log(LOG_ERROR, "Got error on T.38 re-invite. Bad configuration. Peer needs to have T.38 disabled.\n"); + + /*! \bug Is there any way we can go back to the audio call on both + sides here? + */ + /* While figuring that out, hangup the call */ + if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE)) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } else if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) { + /* We tried to send T.38 out in an initial INVITE and the remote side rejected it, + right now we can't fall back to audio so totally abort. + */ + p->t38.state = T38_DISABLED; + /* Try to reset RTP timers */ + ast_rtp_set_rtptimers_onhold(p->rtp); + ast_log(LOG_ERROR, "Got error on T.38 initial invite. Bailing out.\n"); + + /* The dialog is now terminated */ + if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE)) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + sip_alreadygone(p); + } else { + /* We can't set up this call, so give up */ + if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE)) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + /* If there's no dialog to end, then mark p as already gone */ + if (!reinvite) + sip_alreadygone(p); + } + break; + case 491: /* Pending */ + /* we really should have to wait a while, then retransmit + * We should support the retry-after at some point + * At this point, we treat this as a congestion if the call is not in UP state + */ + xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE); + if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE)) { + if (p->owner->_state != AST_STATE_UP) { + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } else { + /* This is a re-invite that failed. + * Reset the flag after a while + */ + int wait = 3 + ast_random() % 5; + p->waitid = ast_sched_add(sched, wait, sip_reinvite_retry, p); + if (option_debug > 2) + ast_log(LOG_DEBUG, "Reinvite race. Waiting %d secs before retry\n", wait); + } + } + break; + + case 501: /* Not implemented */ + xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE); + if (p->owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + break; + } + if (xmitres == XMIT_ERROR) + ast_log(LOG_WARNING, "Could not transmit message in dialog %s\n", p->callid); +} + +/* \brief Handle SIP response in REFER transaction + We've sent a REFER, now handle responses to it + */ +static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno) +{ + char *auth = "Proxy-Authenticate"; + char *auth2 = "Proxy-Authorization"; + + /* If no refer structure exists, then do nothing */ + if (!p->refer) + return; + + switch (resp) { + case 202: /* Transfer accepted */ + /* We need to do something here */ + /* The transferee is now sending INVITE to target */ + p->refer->status = REFER_ACCEPTED; + /* Now wait for next message */ + if (option_debug > 2) + ast_log(LOG_DEBUG, "Got 202 accepted on transfer\n"); + /* We should hang along, waiting for NOTIFY's here */ + break; + + case 401: /* Not www-authorized on SIP method */ + case 407: /* Proxy auth */ + if (ast_strlen_zero(p->authname)) { + ast_log(LOG_WARNING, "Asked to authenticate REFER to %s:%d but we have no matching peer or realm auth!\n", + ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port)); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + if (resp == 401) { + auth = "WWW-Authenticate"; + auth2 = "Authorization"; + } + if ((p->authtries > 1) || do_proxy_auth(p, req, auth, auth2, SIP_REFER, 0)) { + ast_log(LOG_NOTICE, "Failed to authenticate on REFER to '%s'\n", get_header(&p->initreq, "From")); + p->refer->status = REFER_NOAUTH; + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + break; + case 481: /* Call leg does not exist */ + + /* A transfer with Replaces did not work */ + /* OEJ: We should Set flag, cancel the REFER, go back + to original call - but right now we can't */ + ast_log(LOG_WARNING, "Remote host can't match REFER request to call '%s'. Giving up.\n", p->callid); + if (p->owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + break; + + case 500: /* Server error */ + case 501: /* Method not implemented */ + /* Return to the current call onhold */ + /* Status flag needed to be reset */ + ast_log(LOG_NOTICE, "SIP transfer to %s failed, call miserably fails. \n", p->refer->refer_to); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + p->refer->status = REFER_FAILED; + break; + case 603: /* Transfer declined */ + ast_log(LOG_NOTICE, "SIP transfer to %s declined, call miserably fails. \n", p->refer->refer_to); + p->refer->status = REFER_FAILED; + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + break; + } +} + +/*! \brief Handle responses on REGISTER to services */ +static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno) +{ + int expires, expires_ms; + struct sip_registry *r; + r=p->registry; + + switch (resp) { + case 401: /* Unauthorized */ + if ((p->authtries == MAX_AUTHTRIES) || do_register_auth(p, req, "WWW-Authenticate", "Authorization")) { + ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s@%s' (Tries %d)\n", p->registry->username, p->registry->hostname, p->authtries); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + break; + case 403: /* Forbidden */ + ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for REGISTER for '%s' to '%s'\n", p->registry->username, p->registry->hostname); + if (global_regattempts_max) + p->registry->regattempts = global_regattempts_max+1; + AST_SCHED_DEL(sched, r->timeout); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + break; + case 404: /* Not found */ + ast_log(LOG_WARNING, "Got 404 Not found on SIP register to service %s@%s, giving up\n", p->registry->username,p->registry->hostname); + if (global_regattempts_max) + p->registry->regattempts = global_regattempts_max+1; + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + r->call = NULL; + AST_SCHED_DEL(sched, r->timeout); + break; + case 407: /* Proxy auth */ + if ((p->authtries == MAX_AUTHTRIES) || do_register_auth(p, req, "Proxy-Authenticate", "Proxy-Authorization")) { + ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s' (tries '%d')\n", get_header(&p->initreq, "From"), p->authtries); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + break; + case 408: /* Request timeout */ + /* Got a timeout response, so reset the counter of failed responses */ + r->regattempts = 0; + break; + case 479: /* SER: Not able to process the URI - address is wrong in register*/ + ast_log(LOG_WARNING, "Got error 479 on register to %s@%s, giving up (check config)\n", p->registry->username,p->registry->hostname); + if (global_regattempts_max) + p->registry->regattempts = global_regattempts_max+1; + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + r->call = NULL; + AST_SCHED_DEL(sched, r->timeout); + break; + case 200: /* 200 OK */ + if (!r) { + ast_log(LOG_WARNING, "Got 200 OK on REGISTER, but there isn't a registry entry for '%s' (we probably already got the OK)\n", S_OR(p->peername, p->username)); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + return 0; + } + + r->regstate = REG_STATE_REGISTERED; + r->regtime = time(NULL); /* Reset time of last succesful registration */ + manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelDriver: SIP\r\nDomain: %s\r\nStatus: %s\r\n", r->hostname, regstate2str(r->regstate)); + r->regattempts = 0; + if (option_debug) + ast_log(LOG_DEBUG, "Registration successful\n"); + if (r->timeout > -1) { + if (option_debug) + ast_log(LOG_DEBUG, "Cancelling timeout %d\n", r->timeout); + } + AST_SCHED_DEL(sched, r->timeout); + r->call = NULL; + p->registry = NULL; + /* Let this one hang around until we have all the responses */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + /* ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); */ + + /* set us up for re-registering */ + /* figure out how long we got registered for */ + AST_SCHED_DEL(sched, r->expire); + /* according to section 6.13 of RFC, contact headers override + expires headers, so check those first */ + expires = 0; + + /* XXX todo: try to save the extra call */ + if (!ast_strlen_zero(get_header(req, "Contact"))) { + const char *contact = NULL; + const char *tmptmp = NULL; + int start = 0; + for(;;) { + contact = __get_header(req, "Contact", &start); + /* this loop ensures we get a contact header about our register request */ + if(!ast_strlen_zero(contact)) { + if( (tmptmp=strstr(contact, p->our_contact))) { + contact=tmptmp; + break; + } + } else + break; + } + tmptmp = strcasestr(contact, "expires="); + if (tmptmp) { + if (sscanf(tmptmp + 8, "%d;", &expires) != 1) + expires = 0; + } + + } + if (!expires) + expires=atoi(get_header(req, "expires")); + if (!expires) + expires=default_expiry; + + expires_ms = expires * 1000; + if (expires <= EXPIRY_GUARD_LIMIT) + expires_ms -= MAX((expires_ms * EXPIRY_GUARD_PCT),EXPIRY_GUARD_MIN); + else + expires_ms -= EXPIRY_GUARD_SECS * 1000; + if (sipdebug) + ast_log(LOG_NOTICE, "Outbound Registration: Expiry for %s is %d sec (Scheduling reregistration in %d s)\n", r->hostname, expires, expires_ms/1000); + + r->refresh= (int) expires_ms / 1000; + + /* Schedule re-registration before we expire */ + AST_SCHED_DEL(sched, r->expire); + r->expire = ast_sched_add(sched, expires_ms, sip_reregister, r); + ASTOBJ_UNREF(r, sip_registry_destroy); + } + return 1; +} + +/*! \brief Handle qualification responses (OPTIONS) */ +static void handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_request *req) +{ + struct sip_peer *peer = p->relatedpeer; + int statechanged, is_reachable, was_reachable; + int pingtime = ast_tvdiff_ms(ast_tvnow(), peer->ps); + + /* + * Compute the response time to a ping (goes in peer->lastms.) + * -1 means did not respond, 0 means unknown, + * 1..maxms is a valid response, >maxms means late response. + */ + if (pingtime < 1) /* zero = unknown, so round up to 1 */ + pingtime = 1; + + /* Now determine new state and whether it has changed. + * Use some helper variables to simplify the writing + * of the expressions. + */ + was_reachable = peer->lastms > 0 && peer->lastms <= peer->maxms; + is_reachable = pingtime <= peer->maxms; + statechanged = peer->lastms == 0 /* yes, unknown before */ + || was_reachable != is_reachable; + + peer->lastms = pingtime; + peer->call = NULL; + if (statechanged) { + const char *s = is_reachable ? "Reachable" : "Lagged"; + + ast_log(LOG_NOTICE, "Peer '%s' is now %s. (%dms / %dms)\n", + peer->name, s, pingtime, peer->maxms); + ast_device_state_changed("SIP/%s", peer->name); + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", + "Peer: SIP/%s\r\nPeerStatus: %s\r\nTime: %d\r\n", + peer->name, s, pingtime); + } + + if (!AST_SCHED_DEL(sched, peer->pokeexpire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + + /* Try again eventually */ + peer->pokeexpire = ast_sched_add(sched, + is_reachable ? DEFAULT_FREQ_OK : DEFAULT_FREQ_NOTOK, + sip_poke_peer_s, ASTOBJ_REF(peer)); + + if (peer->pokeexpire == -1) { + ASTOBJ_UNREF(peer, sip_destroy_peer); + } +} + +/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */ +static void stop_media_flows(struct sip_pvt *p) +{ + /* Immediately stop RTP, VRTP and UDPTL as applicable */ + if (p->rtp) + ast_rtp_stop(p->rtp); + if (p->vrtp) + ast_rtp_stop(p->vrtp); + if (p->udptl) + ast_udptl_stop(p->udptl); +} + +/*! \brief Handle SIP response in dialogue */ +/* XXX only called by handle_request */ +static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno) +{ + struct ast_channel *owner; + int sipmethod; + int res = 1; + const char *c = get_header(req, "Cseq"); + /* GCC 4.2 complains if I try to cast c as a char * when passing it to ast_skip_nonblanks, so make a copy of it */ + char *c_copy = ast_strdupa(c); + /* Skip the Cseq and its subsequent spaces */ + const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy)); + + if (!msg) + msg = ""; + + sipmethod = find_sip_method(msg); + + owner = p->owner; + if (owner) + owner->hangupcause = hangup_sip2cause(resp); + + /* Acknowledge whatever it is destined for */ + if ((resp >= 100) && (resp <= 199)) + __sip_semi_ack(p, seqno, 0, sipmethod); + else + __sip_ack(p, seqno, 0, sipmethod); + + /* If this is a NOTIFY for a subscription clear the flag that indicates that we have a NOTIFY pending */ + if (!p->owner && sipmethod == SIP_NOTIFY && p->pendinginvite) + p->pendinginvite = 0; + + /* Get their tag if we haven't already */ + if (ast_strlen_zero(p->theirtag) || (resp >= 200)) { + char tag[128]; + + gettag(req, "To", tag, sizeof(tag)); + ast_string_field_set(p, theirtag, tag); + } + + /* RFC 3261 Section 15 specifies that if we receive a 408 or 481 + * in response to a BYE, then we should end the current dialog + * and session. It is known that at least one phone manufacturer + * potentially will send a 404 in response to a BYE, so we'll be + * liberal in what we accept and end the dialog and session if we + * receive any of those responses to a BYE. + */ + if ((resp == 404 || resp == 408 || resp == 481) && sipmethod == SIP_BYE) { + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + return; + } + + if (p->relatedpeer && p->method == SIP_OPTIONS) { + /* We don't really care what the response is, just that it replied back. + Well, as long as it's not a 100 response... since we might + need to hang around for something more "definitive" */ + if (resp != 100) + handle_response_peerpoke(p, resp, req); + } else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) { + switch(resp) { + case 100: /* 100 Trying */ + case 101: /* 101 Dialog establishment */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + break; + case 183: /* 183 Session Progress */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + break; + case 180: /* 180 Ringing */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + break; + case 182: /* 182 Queued */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + break; + case 200: /* 200 OK */ + p->authtries = 0; /* Reset authentication counter */ + if (sipmethod == SIP_MESSAGE || sipmethod == SIP_INFO) { + /* We successfully transmitted a message + or a video update request in INFO */ + /* Nothing happens here - the message is inside a dialog */ + } else if (sipmethod == SIP_INVITE) { + handle_response_invite(p, resp, rest, req, seqno); + } else if (sipmethod == SIP_NOTIFY) { + /* They got the notify, this is the end */ + if (p->owner) { + if (!p->refer) { + ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name); + ast_queue_hangup(p->owner); + } else if (option_debug > 3) + ast_log(LOG_DEBUG, "Got OK on REFER Notify message\n"); + } else { + if (p->subscribed == NONE) + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) { + /* Ready to send the next state we have on queue */ + ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE); + cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p); + } + } + } else if (sipmethod == SIP_REGISTER) + res = handle_response_register(p, resp, rest, req, ignore, seqno); + else if (sipmethod == SIP_BYE) { /* Ok, we're ready to go */ + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + } else if (sipmethod == SIP_SUBSCRIBE) + ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + break; + case 202: /* Transfer accepted */ + if (sipmethod == SIP_REFER) + handle_response_refer(p, resp, rest, req, seqno); + break; + case 401: /* Not www-authorized on SIP method */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + else if (sipmethod == SIP_REFER) + handle_response_refer(p, resp, rest, req, seqno); + else if (p->registry && sipmethod == SIP_REGISTER) + res = handle_response_register(p, resp, rest, req, ignore, seqno); + else if (sipmethod == SIP_BYE) { + if (ast_strlen_zero(p->authname)) { + ast_log(LOG_WARNING, "Asked to authenticate %s, to %s:%d but we have no matching peer!\n", + msg, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port)); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } else if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, "WWW-Authenticate", "Authorization", sipmethod, 0)) { + ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From")); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + /* We fail to auth bye on our own call, but still needs to tear down the call. + Life, they call it. */ + } + } else { + ast_log(LOG_WARNING, "Got authentication request (401) on unknown %s to '%s'\n", sip_methods[sipmethod].text, get_header(req, "To")); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + break; + case 403: /* Forbidden - we failed authentication */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + else if (p->registry && sipmethod == SIP_REGISTER) + res = handle_response_register(p, resp, rest, req, ignore, seqno); + else { + ast_log(LOG_WARNING, "Forbidden - maybe wrong password on authentication for %s\n", msg); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + break; + case 404: /* Not found */ + if (p->registry && sipmethod == SIP_REGISTER) + res = handle_response_register(p, resp, rest, req, ignore, seqno); + else if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + else if (owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + break; + case 407: /* Proxy auth required */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + else if (sipmethod == SIP_REFER) + handle_response_refer(p, resp, rest, req, seqno); + else if (p->registry && sipmethod == SIP_REGISTER) + res = handle_response_register(p, resp, rest, req, ignore, seqno); + else if (sipmethod == SIP_BYE) { + if (ast_strlen_zero(p->authname)) { + ast_log(LOG_WARNING, "Asked to authenticate %s, to %s:%d but we have no matching peer!\n", + msg, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port)); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } else if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, "Proxy-Authenticate", "Proxy-Authorization", sipmethod, 0)) { + ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From")); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + } else /* We can't handle this, giving up in a bad way */ + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + + break; + case 408: /* Request timeout - terminate dialog */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + else if (sipmethod == SIP_REGISTER) + res = handle_response_register(p, resp, rest, req, ignore, seqno); + else if (sipmethod == SIP_BYE) { + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + if (option_debug) + ast_log(LOG_DEBUG, "Got timeout on bye. Thanks for the answer. Now, kill this call\n"); + } else { + if (owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + break; + case 481: /* Call leg does not exist */ + if (sipmethod == SIP_INVITE) { + handle_response_invite(p, resp, rest, req, seqno); + } else if (sipmethod == SIP_REFER) { + handle_response_refer(p, resp, rest, req, seqno); + } else if (sipmethod == SIP_BYE) { + /* The other side has no transaction to bye, + just assume it's all right then */ + ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid); + } else if (sipmethod == SIP_CANCEL) { + /* The other side has no transaction to cancel, + just assume it's all right then */ + ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid); + } else { + ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid); + /* Guessing that this is not an important request */ + } + break; + case 487: + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + break; + case 488: /* Not acceptable here - codec error */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + break; + case 491: /* Pending */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + else { + if (option_debug) + ast_log(LOG_DEBUG, "Got 491 on %s, unspported. Call ID %s\n", sip_methods[sipmethod].text, p->callid); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + break; + case 501: /* Not Implemented */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + else if (sipmethod == SIP_REFER) + handle_response_refer(p, resp, rest, req, seqno); + else + ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n", ast_inet_ntoa(p->sa.sin_addr), msg); + break; + case 603: /* Declined transfer */ + if (sipmethod == SIP_REFER) { + handle_response_refer(p, resp, rest, req, seqno); + break; + } + /* Fallthrough */ + default: + if ((resp >= 300) && (resp < 700)) { + /* Fatal response */ + if ((option_verbose > 2) && (resp != 487)) + ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr)); + + if (sipmethod == SIP_INVITE) + stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */ + + /* XXX Locking issues?? XXX */ + switch(resp) { + case 300: /* Multiple Choices */ + case 301: /* Moved permenantly */ + case 302: /* Moved temporarily */ + case 305: /* Use Proxy */ + parse_moved_contact(p, req); + /* Fall through */ + case 486: /* Busy here */ + case 600: /* Busy everywhere */ + case 603: /* Decline */ + if (p->owner) + ast_queue_control(p->owner, AST_CONTROL_BUSY); + break; + case 482: /* + \note SIP is incapable of performing a hairpin call, which + is yet another failure of not having a layer 2 (again, YAY + IETF for thinking ahead). So we treat this as a call + forward and hope we end up at the right place... */ + if (option_debug) + ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n"); + if (p->owner) + ast_string_field_build(p->owner, call_forward, + "Local/%s@%s", p->username, p->context); + /* Fall through */ + case 480: /* Temporarily Unavailable */ + case 404: /* Not Found */ + case 410: /* Gone */ + case 400: /* Bad Request */ + case 500: /* Server error */ + if (sipmethod == SIP_REFER) { + handle_response_refer(p, resp, rest, req, seqno); + break; + } + /* Fall through */ + case 502: /* Bad gateway */ + case 503: /* Service Unavailable */ + case 504: /* Server Timeout */ + if (owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + break; + default: + /* Send hangup */ + if (owner && sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO && sipmethod != SIP_BYE) + ast_queue_hangup(p->owner); + break; + } + /* ACK on invite */ + if (sipmethod == SIP_INVITE) + transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE); + if (sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO) + sip_alreadygone(p); + if (!p->owner) + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } else if ((resp >= 100) && (resp < 200)) { + if (sipmethod == SIP_INVITE) { + if (!ast_test_flag(req, SIP_PKT_IGNORE) && sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + if (find_sdp(req)) + process_sdp(p, req); + if (p->owner) { + /* Queue a progress frame */ + ast_queue_control(p->owner, AST_CONTROL_PROGRESS); + } + } + } else + ast_log(LOG_NOTICE, "Dont know how to handle a %d %s response from %s\n", resp, rest, p->owner ? p->owner->name : ast_inet_ntoa(p->sa.sin_addr)); + } + } else { + /* Responses to OUTGOING SIP requests on INCOMING calls + get handled here. As well as out-of-call message responses */ + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg); + + if (sipmethod == SIP_INVITE && resp == 200) { + /* Tags in early session is replaced by the tag in 200 OK, which is + the final reply to our INVITE */ + char tag[128]; + + gettag(req, "To", tag, sizeof(tag)); + ast_string_field_set(p, theirtag, tag); + } + + switch(resp) { + case 200: + if (sipmethod == SIP_INVITE) { + handle_response_invite(p, resp, rest, req, seqno); + } else if (sipmethod == SIP_CANCEL) { + if (option_debug) + ast_log(LOG_DEBUG, "Got 200 OK on CANCEL\n"); + + /* Wait for 487, then destroy */ + } else if (sipmethod == SIP_NOTIFY) { + /* They got the notify, this is the end */ + if (p->owner) { + if (p->refer) { + if (option_debug) + ast_log(LOG_DEBUG, "Got 200 OK on NOTIFY for transfer\n"); + } else + ast_log(LOG_WARNING, "Notify answer on an owned channel?\n"); + /* ast_queue_hangup(p->owner); Disabled */ + } else { + if (!p->subscribed && !p->refer) + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) { + /* Ready to send the next state we have on queue */ + ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE); + cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p); + } + } + } else if (sipmethod == SIP_BYE) + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + else if (sipmethod == SIP_MESSAGE || sipmethod == SIP_INFO) + /* We successfully transmitted a message or + a video update request in INFO */ + ; + else if (sipmethod == SIP_BYE) + /* Ok, we're ready to go */ + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + break; + case 202: /* Transfer accepted */ + if (sipmethod == SIP_REFER) + handle_response_refer(p, resp, rest, req, seqno); + break; + case 401: /* www-auth */ + case 407: + if (sipmethod == SIP_REFER) + handle_response_refer(p, resp, rest, req, seqno); + else if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + else if (sipmethod == SIP_BYE) { + char *auth, *auth2; + + auth = (resp == 407 ? "Proxy-Authenticate" : "WWW-Authenticate"); + auth2 = (resp == 407 ? "Proxy-Authorization" : "Authorization"); + if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, auth, auth2, sipmethod, 0)) { + ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From")); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + } + break; + case 481: /* Call leg does not exist */ + if (sipmethod == SIP_INVITE) { + /* Re-invite failed */ + handle_response_invite(p, resp, rest, req, seqno); + } else if (sipmethod == SIP_BYE) { + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } else if (sipdebug) { + ast_log (LOG_DEBUG, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid); + } + break; + case 501: /* Not Implemented */ + if (sipmethod == SIP_INVITE) + handle_response_invite(p, resp, rest, req, seqno); + else if (sipmethod == SIP_REFER) + handle_response_refer(p, resp, rest, req, seqno); + break; + case 603: /* Declined transfer */ + if (sipmethod == SIP_REFER) { + handle_response_refer(p, resp, rest, req, seqno); + break; + } + /* Fallthrough */ + default: /* Errors without handlers */ + if ((resp >= 100) && (resp < 200)) { + if (sipmethod == SIP_INVITE) { /* re-invite */ + if (!ast_test_flag(req, SIP_PKT_IGNORE) && sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + } + } + if ((resp >= 300) && (resp < 700)) { + if ((option_verbose > 2) && (resp != 487)) + ast_verbose(VERBOSE_PREFIX_3 "Incoming call: Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr)); + switch(resp) { + case 488: /* Not acceptable here - codec error */ + case 603: /* Decline */ + case 500: /* Server error */ + case 502: /* Bad gateway */ + case 503: /* Service Unavailable */ + case 504: /* Server timeout */ + + /* re-invite failed */ + if (sipmethod == SIP_INVITE && sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + break; + } + } + break; + } + } +} + + +/*! \brief Park SIP call support function + Starts in a new thread, then parks the call + XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the + audio can't be heard before hangup +*/ +static void *sip_park_thread(void *stuff) +{ + struct ast_channel *transferee, *transferer; /* Chan1: The transferee, Chan2: The transferer */ + struct sip_dual *d; + struct sip_request req; + int ext; + int res; + + d = stuff; + transferee = d->chan1; + transferer = d->chan2; + copy_request(&req, &d->req); + + if (!transferee || !transferer) { + ast_log(LOG_ERROR, "Missing channels for parking! Transferer %s Transferee %s\n", transferer ? "<available>" : "<missing>", transferee ? "<available>" : "<missing>" ); + return NULL; + } + if (option_debug > 3) + ast_log(LOG_DEBUG, "SIP Park: Transferer channel %s, Transferee %s\n", transferer->name, transferee->name); + + ast_channel_lock(transferee); + if (ast_do_masquerade(transferee)) { + ast_log(LOG_WARNING, "Masquerade failed.\n"); + transmit_response(transferer->tech_pvt, "503 Internal error", &req); + ast_channel_unlock(transferee); + return NULL; + } + ast_channel_unlock(transferee); + + res = ast_park_call(transferee, transferer, 0, &ext); + + +#ifdef WHEN_WE_KNOW_THAT_THE_CLIENT_SUPPORTS_MESSAGE + if (!res) { + transmit_message_with_text(transferer->tech_pvt, "Unable to park call.\n"); + } else { + /* Then tell the transferer what happened */ + sprintf(buf, "Call parked on extension '%d'", ext); + transmit_message_with_text(transferer->tech_pvt, buf); + } +#endif + + /* Any way back to the current call??? */ + /* Transmit response to the REFER request */ + transmit_response(transferer->tech_pvt, "202 Accepted", &req); + if (!res) { + /* Transfer succeeded */ + append_history(transferer->tech_pvt, "SIPpark","Parked call on %d", ext); + transmit_notify_with_sipfrag(transferer->tech_pvt, d->seqno, "200 OK", TRUE); + transferer->hangupcause = AST_CAUSE_NORMAL_CLEARING; + ast_hangup(transferer); /* This will cause a BYE */ + if (option_debug) + ast_log(LOG_DEBUG, "SIP Call parked on extension '%d'\n", ext); + } else { + transmit_notify_with_sipfrag(transferer->tech_pvt, d->seqno, "503 Service Unavailable", TRUE); + append_history(transferer->tech_pvt, "SIPpark","Parking failed\n"); + if (option_debug) + ast_log(LOG_DEBUG, "SIP Call parked failed \n"); + /* Do not hangup call */ + } + free(d); + return NULL; +} + +/*! \brief Park a call using the subsystem in res_features.c + This is executed in a separate thread +*/ +static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno) +{ + struct sip_dual *d; + struct ast_channel *transferee, *transferer; + /* Chan2m: The transferer, chan1m: The transferee */ + pthread_t th; + + transferee = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, chan1->accountcode, chan1->exten, chan1->context, chan1->amaflags, "Parking/%s", chan1->name); + transferer = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, chan2->accountcode, chan2->exten, chan2->context, chan2->amaflags, "SIPPeer/%s", chan2->name); + if ((!transferer) || (!transferee)) { + if (transferee) { + transferee->hangupcause = AST_CAUSE_SWITCH_CONGESTION; + ast_hangup(transferee); + } + if (transferer) { + transferer->hangupcause = AST_CAUSE_SWITCH_CONGESTION; + ast_hangup(transferer); + } + return -1; + } + + /* Make formats okay */ + transferee->readformat = chan1->readformat; + transferee->writeformat = chan1->writeformat; + + /* Prepare for taking over the channel */ + ast_channel_masquerade(transferee, chan1); + + /* Setup the extensions and such */ + ast_copy_string(transferee->context, chan1->context, sizeof(transferee->context)); + ast_copy_string(transferee->exten, chan1->exten, sizeof(transferee->exten)); + transferee->priority = chan1->priority; + + /* We make a clone of the peer channel too, so we can play + back the announcement */ + + /* Make formats okay */ + transferer->readformat = chan2->readformat; + transferer->writeformat = chan2->writeformat; + + /* Prepare for taking over the channel. Go ahead and grab this channel + * lock here to avoid a deadlock with callbacks into the channel driver + * that hold the channel lock and want the pvt lock. */ + while (ast_channel_trylock(chan2)) { + struct sip_pvt *pvt = chan2->tech_pvt; + DEADLOCK_AVOIDANCE(&pvt->lock); + } + ast_channel_masquerade(transferer, chan2); + ast_channel_unlock(chan2); + + /* Setup the extensions and such */ + ast_copy_string(transferer->context, chan2->context, sizeof(transferer->context)); + ast_copy_string(transferer->exten, chan2->exten, sizeof(transferer->exten)); + transferer->priority = chan2->priority; + + ast_channel_lock(transferer); + if (ast_do_masquerade(transferer)) { + ast_log(LOG_WARNING, "Masquerade failed :(\n"); + ast_channel_unlock(transferer); + transferer->hangupcause = AST_CAUSE_SWITCH_CONGESTION; + ast_hangup(transferer); + return -1; + } + ast_channel_unlock(transferer); + if (!transferer || !transferee) { + if (!transferer) { + if (option_debug) + ast_log(LOG_DEBUG, "No transferer channel, giving up parking\n"); + } + if (!transferee) { + if (option_debug) + ast_log(LOG_DEBUG, "No transferee channel, giving up parking\n"); + } + return -1; + } + if ((d = ast_calloc(1, sizeof(*d)))) { + pthread_attr_t attr; + + pthread_attr_init(&attr); + pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED); + + /* Save original request for followup */ + copy_request(&d->req, req); + d->chan1 = transferee; /* Transferee */ + d->chan2 = transferer; /* Transferer */ + d->seqno = seqno; + if (ast_pthread_create_background(&th, &attr, sip_park_thread, d) < 0) { + /* Could not start thread */ + free(d); /* We don't need it anymore. If thread is created, d will be free'd + by sip_park_thread() */ + pthread_attr_destroy(&attr); + return 0; + } + pthread_attr_destroy(&attr); + } + return -1; +} + +/*! \brief Turn off generator data + XXX Does this function belong in the SIP channel? +*/ +static void ast_quiet_chan(struct ast_channel *chan) +{ + if (chan && chan->_state == AST_STATE_UP) { + if (ast_test_flag(chan, AST_FLAG_MOH)) + ast_moh_stop(chan); + else if (chan->generatordata) + ast_deactivate_generator(chan); + } +} + +/*! \brief Attempt transfer of SIP call + This fix for attended transfers on a local PBX */ +static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target) +{ + int res = 0; + struct ast_channel *peera = NULL, + *peerb = NULL, + *peerc = NULL, + *peerd = NULL; + + + /* We will try to connect the transferee with the target and hangup + all channels to the transferer */ + if (option_debug > 3) { + ast_log(LOG_DEBUG, "Sip transfer:--------------------\n"); + if (transferer->chan1) + ast_log(LOG_DEBUG, "-- Transferer to PBX channel: %s State %s\n", transferer->chan1->name, ast_state2str(transferer->chan1->_state)); + else + ast_log(LOG_DEBUG, "-- No transferer first channel - odd??? \n"); + if (target->chan1) + ast_log(LOG_DEBUG, "-- Transferer to PBX second channel (target): %s State %s\n", target->chan1->name, ast_state2str(target->chan1->_state)); + else + ast_log(LOG_DEBUG, "-- No target first channel ---\n"); + if (transferer->chan2) + ast_log(LOG_DEBUG, "-- Bridged call to transferee: %s State %s\n", transferer->chan2->name, ast_state2str(transferer->chan2->_state)); + else + ast_log(LOG_DEBUG, "-- No bridged call to transferee\n"); + if (target->chan2) + ast_log(LOG_DEBUG, "-- Bridged call to transfer target: %s State %s\n", target->chan2 ? target->chan2->name : "<none>", target->chan2 ? ast_state2str(target->chan2->_state) : "(none)"); + else + ast_log(LOG_DEBUG, "-- No target second channel ---\n"); + ast_log(LOG_DEBUG, "-- END Sip transfer:--------------------\n"); + } + if (transferer->chan2) { /* We have a bridge on the transferer's channel */ + peera = transferer->chan1; /* Transferer - PBX -> transferee channel * the one we hangup */ + peerb = target->chan1; /* Transferer - PBX -> target channel - This will get lost in masq */ + peerc = transferer->chan2; /* Asterisk to Transferee */ + peerd = target->chan2; /* Asterisk to Target */ + if (option_debug > 2) + ast_log(LOG_DEBUG, "SIP transfer: Four channels to handle\n"); + } else if (target->chan2) { /* Transferer has no bridge (IVR), but transferee */ + peera = target->chan1; /* Transferer to PBX -> target channel */ + peerb = transferer->chan1; /* Transferer to IVR*/ + peerc = target->chan2; /* Asterisk to Target */ + peerd = transferer->chan2; /* Nothing */ + if (option_debug > 2) + ast_log(LOG_DEBUG, "SIP transfer: Three channels to handle\n"); + } + + if (peera && peerb && peerc && (peerb != peerc)) { + ast_quiet_chan(peera); /* Stop generators */ + ast_quiet_chan(peerb); + ast_quiet_chan(peerc); + if (peerd) + ast_quiet_chan(peerd); + + if (option_debug > 3) + ast_log(LOG_DEBUG, "SIP transfer: trying to masquerade %s into %s\n", peerc->name, peerb->name); + if (ast_channel_masquerade(peerb, peerc)) { + ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", peerb->name, peerc->name); + res = -1; + } else + ast_log(LOG_DEBUG, "SIP transfer: Succeeded to masquerade channels.\n"); + return res; + } else { + ast_log(LOG_NOTICE, "SIP Transfer attempted with no appropriate bridged calls to transfer\n"); + if (transferer->chan1) + ast_softhangup_nolock(transferer->chan1, AST_SOFTHANGUP_DEV); + if (target->chan1) + ast_softhangup_nolock(target->chan1, AST_SOFTHANGUP_DEV); + return -2; + } + return 0; +} + +/*! \brief Get tag from packet + * + * \return Returns the pointer to the provided tag buffer, + * or NULL if the tag was not found. + */ +static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize) +{ + const char *thetag; + + if (!tagbuf) + return NULL; + tagbuf[0] = '\0'; /* reset the buffer */ + thetag = get_header(req, header); + thetag = strcasestr(thetag, ";tag="); + if (thetag) { + thetag += 5; + ast_copy_string(tagbuf, thetag, tagbufsize); + return strsep(&tagbuf, ";"); + } + return NULL; +} + +/*! \brief Handle incoming notifications */ +static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e) +{ + /* This is mostly a skeleton for future improvements */ + /* Mostly created to return proper answers on notifications on outbound REFER's */ + int res = 0; + const char *event = get_header(req, "Event"); + char *eventid = NULL; + char *sep; + + if( (sep = strchr(event, ';')) ) { /* XXX bug here - overwriting string ? */ + *sep++ = '\0'; + eventid = sep; + } + + if (option_debug > 1 && sipdebug) + ast_log(LOG_DEBUG, "Got NOTIFY Event: %s\n", event); + + if (strcmp(event, "refer")) { + /* We don't understand this event. */ + /* Here's room to implement incoming voicemail notifications :-) */ + transmit_response(p, "489 Bad event", req); + res = -1; + } else { + /* Save nesting depth for now, since there might be other events we will + support in the future */ + + /* Handle REFER notifications */ + + char buf[1024]; + char *cmd, *code; + int respcode; + int success = TRUE; + + /* EventID for each transfer... EventID is basically the REFER cseq + + We are getting notifications on a call that we transfered + We should hangup when we are getting a 200 OK in a sipfrag + Check if we have an owner of this event */ + + /* Check the content type */ + if (strncasecmp(get_header(req, "Content-Type"), "message/sipfrag", strlen("message/sipfrag"))) { + /* We need a sipfrag */ + transmit_response(p, "400 Bad request", req); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return -1; + } + + /* Get the text of the attachment */ + if (get_msg_text(buf, sizeof(buf), req)) { + ast_log(LOG_WARNING, "Unable to retrieve attachment from NOTIFY %s\n", p->callid); + transmit_response(p, "400 Bad request", req); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return -1; + } + + /* + From the RFC... + A minimal, but complete, implementation can respond with a single + NOTIFY containing either the body: + SIP/2.0 100 Trying + + if the subscription is pending, the body: + SIP/2.0 200 OK + if the reference was successful, the body: + SIP/2.0 503 Service Unavailable + if the reference failed, or the body: + SIP/2.0 603 Declined + + if the REFER request was accepted before approval to follow the + reference could be obtained and that approval was subsequently denied + (see Section 2.4.7). + + If there are several REFERs in the same dialog, we need to + match the ID of the event header... + */ + if (option_debug > 2) + ast_log(LOG_DEBUG, "* SIP Transfer NOTIFY Attachment: \n---%s\n---\n", buf); + cmd = ast_skip_blanks(buf); + code = cmd; + /* We are at SIP/2.0 */ + while(*code && (*code > 32)) { /* Search white space */ + code++; + } + *code++ = '\0'; + code = ast_skip_blanks(code); + sep = code; + sep++; + while(*sep && (*sep > 32)) { /* Search white space */ + sep++; + } + *sep++ = '\0'; /* Response string */ + respcode = atoi(code); + switch (respcode) { + case 100: /* Trying: */ + case 101: /* dialog establishment */ + /* Don't do anything yet */ + break; + case 183: /* Ringing: */ + /* Don't do anything yet */ + break; + case 200: /* OK: The new call is up, hangup this call */ + /* Hangup the call that we are replacing */ + break; + case 301: /* Moved permenantly */ + case 302: /* Moved temporarily */ + /* Do we get the header in the packet in this case? */ + success = FALSE; + break; + case 503: /* Service Unavailable: The new call failed */ + /* Cancel transfer, continue the call */ + success = FALSE; + break; + case 603: /* Declined: Not accepted */ + /* Cancel transfer, continue the current call */ + success = FALSE; + break; + } + if (!success) { + ast_log(LOG_NOTICE, "Transfer failed. Sorry. Nothing further to do with this call\n"); + } + + /* Confirm that we received this packet */ + transmit_response(p, "200 OK", req); + }; + + if (!p->lastinvite) + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + + return res; +} + +/*! \brief Handle incoming OPTIONS request */ +static int handle_request_options(struct sip_pvt *p, struct sip_request *req) +{ + int res; + + + /* XXX Should we authenticate OPTIONS? XXX */ + + if (p->lastinvite) { + /* if this is a request in an active dialog, just confirm that the dialog exists. */ + transmit_response_with_allow(p, "200 OK", req, 0); + return 0; + } + + res = get_destination(p, req); + build_contact(p); + + if (ast_strlen_zero(p->context)) + ast_string_field_set(p, context, default_context); + + if (ast_shutting_down()) + transmit_response_with_allow(p, "503 Unavailable", req, 0); + else if (res < 0) + transmit_response_with_allow(p, "404 Not Found", req, 0); + else + transmit_response_with_allow(p, "200 OK", req, 0); + + /* Destroy if this OPTIONS was the opening request, but not if + it's in the middle of a normal call flow. */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + + return res; +} + +/*! \brief Handle the transfer part of INVITE with a replaces: header, + meaning a target pickup or an attended transfer */ +static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin) +{ + struct ast_frame *f; + int earlyreplace = 0; + int oneleggedreplace = 0; /* Call with no bridge, propably IVR or voice message */ + struct ast_channel *c = p->owner; /* Our incoming call */ + struct ast_channel *replacecall = p->refer->refer_call->owner; /* The channel we're about to take over */ + struct ast_channel *targetcall; /* The bridge to the take-over target */ + + /* Check if we're in ring state */ + if (replacecall->_state == AST_STATE_RING) + earlyreplace = 1; + + /* Check if we have a bridge */ + if (!(targetcall = ast_bridged_channel(replacecall))) { + /* We have no bridge */ + if (!earlyreplace) { + if (option_debug > 1) + ast_log(LOG_DEBUG, " Attended transfer attempted to replace call with no bridge (maybe ringing). Channel %s!\n", replacecall->name); + oneleggedreplace = 1; + } + } + if (option_debug > 3 && targetcall && targetcall->_state == AST_STATE_RINGING) + ast_log(LOG_DEBUG, "SIP transfer: Target channel is in ringing state\n"); + + if (option_debug > 3) { + if (targetcall) + ast_log(LOG_DEBUG, "SIP transfer: Invite Replace incoming channel should bridge to channel %s while hanging up channel %s\n", targetcall->name, replacecall->name); + else + ast_log(LOG_DEBUG, "SIP transfer: Invite Replace incoming channel should replace and hang up channel %s (one call leg)\n", replacecall->name); + } + + if (ignore) { + ast_log(LOG_NOTICE, "Ignoring this INVITE with replaces in a stupid way.\n"); + /* We should answer something here. If we are here, the + call we are replacing exists, so an accepted + can't harm */ + transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE); + /* Do something more clever here */ + ast_channel_unlock(c); + ast_mutex_unlock(&p->refer->refer_call->lock); + return 1; + } + if (!c) { + /* What to do if no channel ??? */ + ast_log(LOG_ERROR, "Unable to create new channel. Invite/replace failed.\n"); + transmit_response_reliable(p, "503 Service Unavailable", req); + append_history(p, "Xfer", "INVITE/Replace Failed. No new channel."); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + ast_mutex_unlock(&p->refer->refer_call->lock); + return 1; + } + append_history(p, "Xfer", "INVITE/Replace received"); + /* We have three channels to play with + channel c: New incoming call + targetcall: Call from PBX to target + p->refer->refer_call: SIP pvt dialog from transferer to pbx. + replacecall: The owner of the previous + We need to masq C into refer_call to connect to + targetcall; + If we are talking to internal audio stream, target call is null. + */ + + /* Fake call progress */ + transmit_response(p, "100 Trying", req); + ast_setstate(c, AST_STATE_RING); + + /* Masquerade the new call into the referred call to connect to target call + Targetcall is not touched by the masq */ + + /* Answer the incoming call and set channel to UP state */ + transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE); + + ast_setstate(c, AST_STATE_UP); + + /* Stop music on hold and other generators */ + ast_quiet_chan(replacecall); + ast_quiet_chan(targetcall); + if (option_debug > 3) + ast_log(LOG_DEBUG, "Invite/Replaces: preparing to masquerade %s into %s\n", c->name, replacecall->name); + /* Unlock clone, but not original (replacecall) */ + if (!oneleggedreplace) + ast_channel_unlock(c); + + /* Unlock PVT */ + ast_mutex_unlock(&p->refer->refer_call->lock); + + /* Make sure that the masq does not free our PVT for the old call */ + if (! earlyreplace && ! oneleggedreplace ) + ast_set_flag(&p->refer->refer_call->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */ + + /* Prepare the masquerade - if this does not happen, we will be gone */ + if(ast_channel_masquerade(replacecall, c)) + ast_log(LOG_ERROR, "Failed to masquerade C into Replacecall\n"); + else if (option_debug > 3) + ast_log(LOG_DEBUG, "Invite/Replaces: Going to masquerade %s into %s\n", c->name, replacecall->name); + + /* The masquerade will happen as soon as someone reads a frame from the channel */ + + /* C should now be in place of replacecall */ + /* ast_read needs to lock channel */ + ast_channel_unlock(c); + + if (earlyreplace || oneleggedreplace ) { + /* Force the masq to happen */ + if ((f = ast_read(replacecall))) { /* Force the masq to happen */ + ast_frfree(f); + f = NULL; + if (option_debug > 3) + ast_log(LOG_DEBUG, "Invite/Replace: Could successfully read frame from RING channel!\n"); + } else { + ast_log(LOG_WARNING, "Invite/Replace: Could not read frame from RING channel \n"); + } + c->hangupcause = AST_CAUSE_SWITCH_CONGESTION; + if (!oneleggedreplace) + ast_channel_unlock(replacecall); + } else { /* Bridged call, UP channel */ + if ((f = ast_read(replacecall))) { /* Force the masq to happen */ + /* Masq ok */ + ast_frfree(f); + f = NULL; + if (option_debug > 2) + ast_log(LOG_DEBUG, "Invite/Replace: Could successfully read frame from channel! Masq done.\n"); + } else { + ast_log(LOG_WARNING, "Invite/Replace: Could not read frame from channel. Transfer failed\n"); + } + ast_channel_unlock(replacecall); + } + ast_mutex_unlock(&p->refer->refer_call->lock); + + ast_setstate(c, AST_STATE_DOWN); + if (option_debug > 3) { + struct ast_channel *test; + ast_log(LOG_DEBUG, "After transfer:----------------------------\n"); + ast_log(LOG_DEBUG, " -- C: %s State %s\n", c->name, ast_state2str(c->_state)); + if (replacecall) + ast_log(LOG_DEBUG, " -- replacecall: %s State %s\n", replacecall->name, ast_state2str(replacecall->_state)); + if (p->owner) { + ast_log(LOG_DEBUG, " -- P->owner: %s State %s\n", p->owner->name, ast_state2str(p->owner->_state)); + test = ast_bridged_channel(p->owner); + if (test) + ast_log(LOG_DEBUG, " -- Call bridged to P->owner: %s State %s\n", test->name, ast_state2str(test->_state)); + else + ast_log(LOG_DEBUG, " -- No call bridged to C->owner \n"); + } else + ast_log(LOG_DEBUG, " -- No channel yet \n"); + ast_log(LOG_DEBUG, "End After transfer:----------------------------\n"); + } + + ast_channel_unlock(p->owner); /* Unlock new owner */ + if (!oneleggedreplace) + ast_mutex_unlock(&p->lock); /* Unlock SIP structure */ + + /* The call should be down with no ast_channel, so hang it up */ + c->tech_pvt = NULL; + ast_hangup(c); + return 0; +} + +/*! \brief helper routine for sip_uri_cmp + * + * This takes the parameters from two SIP URIs and determines + * if the URIs match. The rules for parameters *suck*. Here's a breakdown + * 1. If a parameter appears in both URIs, then they must have the same value + * in order for the URIs to match + * 2. If one URI has a user, maddr, ttl, or method parameter, then the other + * URI must also have that parameter and must have the same value + * in order for the URIs to match + * 3. All other headers appearing in only one URI are not considered when + * determining if URIs match + * + * \param input1 Parameters from URI 1 + * \param input2 Parameters from URI 2 + * \return Return 0 if the URIs' parameters match, 1 if they do not + */ +static int sip_uri_params_cmp(const char *input1, const char *input2) +{ + char *params1 = ast_strdupa(input1); + char *params2 = ast_strdupa(input2); + char *pos1; + char *pos2; + int maddrmatch = 0; + int ttlmatch = 0; + int usermatch = 0; + int methodmatch = 0; + + /*Quick optimization. If both params are zero-length, then + * they match + */ + if (ast_strlen_zero(params1) && ast_strlen_zero(params2)) { + return 0; + } + + pos1 = params1; + while (!ast_strlen_zero(pos1)) { + char *name1 = pos1; + char *value1 = strchr(pos1, '='); + char *semicolon1 = strchr(pos1, ';'); + int matched = 0; + if (semicolon1) { + *semicolon1++ = '\0'; + } + if (!value1) { + goto fail; + } + *value1++ = '\0'; + /* Checkpoint reached. We have the name and value parsed for param1 + * We have to duplicate params2 each time through the second loop + * or else we can't search and replace the semicolons with \0 each + * time + */ + pos2 = ast_strdupa(params2); + while (!ast_strlen_zero(pos2)) { + char *name2 = pos2; + char *value2 = strchr(pos2, '='); + char *semicolon2 = strchr(pos2, ';'); + if (semicolon2) { + *semicolon2++ = '\0'; + } + if (!value2) { + goto fail; + } + *value2++ = '\0'; + if (!strcasecmp(name1, name2)) { + if (strcasecmp(value1, value2)) { + goto fail; + } else { + matched = 1; + break; + } + } + pos2 = semicolon2; + } + /* Need to see if the parameter we're looking at is one of the 'must-match' parameters */ + if (!strcasecmp(name1, "maddr")) { + if (matched) { + maddrmatch = 1; + } else { + goto fail; + } + } else if (!strcasecmp(name1, "ttl")) { + if (matched) { + ttlmatch = 1; + } else { + goto fail; + } + } else if (!strcasecmp(name1, "user")) { + if (matched) { + usermatch = 1; + } else { + goto fail; + } + } else if (!strcasecmp(name1, "method")) { + if (matched) { + methodmatch = 1; + } else { + goto fail; + } + } + pos1 = semicolon1; + } + + /* We've made it out of that horrible O(m*n) construct and there are no + * failures yet. We're not done yet, though, because params2 could have + * an maddr, ttl, user, or method header and params1 did not. + */ + pos2 = params2; + while (!ast_strlen_zero(pos2)) { + char *name2 = pos2; + char *value2 = strchr(pos2, '='); + char *semicolon2 = strchr(pos2, ';'); + if (semicolon2) { + *semicolon2++ = '\0'; + } + if (!value2) { + goto fail; + } + *value2++ = '\0'; + if ((!strcasecmp(name2, "maddr") && !maddrmatch) || + (!strcasecmp(name2, "ttl") && !ttlmatch) || + (!strcasecmp(name2, "user") && !usermatch) || + (!strcasecmp(name2, "method") && !methodmatch)) { + goto fail; + } + } + return 0; + +fail: + return 1; +} + +/*! \brief helper routine for sip_uri_cmp + * + * This takes the "headers" from two SIP URIs and determines + * if the URIs match. The rules for headers is simple. If a header + * appears in one URI, then it must also appear in the other URI. The + * order in which the headers appear does not matter. + * + * \param input1 Headers from URI 1 + * \param input2 Headers from URI 2 + * \return Return 0 if the URIs' headers match, 1 if they do not + */ +static int sip_uri_headers_cmp(const char *input1, const char *input2) +{ + char *headers1 = ast_strdupa(input1); + char *headers2 = ast_strdupa(input2); + int zerolength1 = ast_strlen_zero(headers1); + int zerolength2 = ast_strlen_zero(headers2); + int different = 0; + char *header1; + + if ((zerolength1 && !zerolength2) || + (zerolength2 && !zerolength1)) + return 1; + + if (zerolength1 && zerolength2) + return 0; + + /* At this point, we can definitively state that both inputs are + * not zero-length. First, one more optimization. If the length + * of the headers is not equal, then we definitely have no match + */ + if (strlen(headers1) != strlen(headers2)) { + return 1; + } + + for (header1 = strsep(&headers1, "&"); header1; header1 = strsep(&headers1, "&")) { + if (!strcasestr(headers2, header1)) { + different = 1; + break; + } + } + + return different; +} + +static int sip_uri_cmp(const char *input1, const char *input2) +{ + char *uri1 = ast_strdupa(input1); + char *uri2 = ast_strdupa(input2); + char *host1; + char *host2; + char *params1; + char *params2; + char *headers1; + char *headers2; + + /* Strip off "sip:" from the URI. We know this is present + * because it was checked back in parse_request() + */ + strsep(&uri1, ":"); + strsep(&uri2, ":"); + + if ((host1 = strchr(uri1, '@'))) { + *host1++ = '\0'; + } + if ((host2 = strchr(uri2, '@'))) { + *host2++ = '\0'; + } + + /* Check for mismatched username and passwords. This is the + * only case-sensitive comparison of a SIP URI + */ + if ((host1 && !host2) || + (host2 && !host1) || + (host1 && host2 && strcmp(uri1, uri2))) { + return 1; + } + + if (!host1) + host1 = uri1; + if (!host2) + host2 = uri2; + + /* Strip off the parameters and headers so we can compare + * host and port + */ + + if ((params1 = strchr(host1, ';'))) { + *params1++ = '\0'; + } + if ((params2 = strchr(host2, ';'))) { + *params2++ = '\0'; + } + + /* Headers come after parameters, but there may be headers without + * parameters, thus the S_OR + */ + if ((headers1 = strchr(S_OR(params1, host1), '?'))) { + *headers1++ = '\0'; + } + if ((headers2 = strchr(S_OR(params2, host2), '?'))) { + *headers2++ = '\0'; + } + + /* Now the host/port are properly isolated. We can get by with a string comparison + * because the SIP URI checking rules have some interesting exceptions that make + * this possible. I will note 2 in particular + * 1. hostnames which resolve to the same IP address as well as a hostname and its + * IP address are not considered a match with SIP URI's. + * 2. If one URI specifies a port and the other does not, then the URIs do not match. + * This includes if one URI explicitly contains port 5060 and the other implies it + * by not having a port specified. + */ + + if (strcasecmp(host1, host2)) { + return 1; + } + + /* Headers have easier rules to follow, so do those first */ + if (sip_uri_headers_cmp(headers1, headers2)) { + return 1; + } + + /* And now the parameters. Ugh */ + return sip_uri_params_cmp(params1, params2); +} + + +/*! \brief Handle incoming INVITE request +\note If the INVITE has a Replaces header, it is part of an + * attended transfer. If so, we do not go through the dial + * plan but tries to find the active call and masquerade + * into it + */ +static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e, int *nounlock) +{ + int res = 1; + int gotdest; + const char *p_replaces; + char *replace_id = NULL; + const char *required; + unsigned int required_profile = 0; + struct ast_channel *c = NULL; /* New channel */ + int reinvite = 0; + + /* Find out what they support */ + if (!p->sipoptions) { + const char *supported = get_header(req, "Supported"); + if (!ast_strlen_zero(supported)) + parse_sip_options(p, supported); + } + + /* Find out what they require */ + required = get_header(req, "Require"); + if (!ast_strlen_zero(required)) { + required_profile = parse_sip_options(NULL, required); + if (required_profile && required_profile != SIP_OPT_REPLACES) { + /* At this point we only support REPLACES */ + transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, required); + ast_log(LOG_WARNING,"Received SIP INVITE with unsupported required extension: %s\n", required); + p->invitestate = INV_COMPLETED; + if (!p->lastinvite) + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return -1; + } + } + + /* Check if this is a loop */ + if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && p->owner && (p->owner->_state != AST_STATE_UP)) { + /* This is a call to ourself. Send ourselves an error code and stop + processing immediately, as SIP really has no good mechanism for + being able to call yourself */ + /* If pedantic is on, we need to check the tags. If they're different, this is + in fact a forked call through a SIP proxy somewhere. */ + int different; + if (pedanticsipchecking) + different = sip_uri_cmp(p->initreq.rlPart2, req->rlPart2); + else + different = strcmp(p->initreq.rlPart2, req->rlPart2); + if (!different) { + transmit_response(p, "482 Loop Detected", req); + p->invitestate = INV_COMPLETED; + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return 0; + } else { + /* This is a spiral. What we need to do is to just change the outgoing INVITE + * so that it now routes to the new Request URI. Since we created the INVITE ourselves + * that should be all we need to do. + */ + char *uri = ast_strdupa(req->rlPart2); + char *at = strchr(uri, '@'); + char *peerorhost; + if (option_debug > 2) { + ast_log(LOG_DEBUG, "Potential spiral detected. Original RURI was %s, new RURI is %s\n", p->initreq.rlPart2, req->rlPart2); + } + if (at) { + *at = '\0'; + } + /* Parse out "sip:" */ + if ((peerorhost = strchr(uri, ':'))) { + *peerorhost++ = '\0'; + } + ast_string_field_free(p, theirtag); + /* Treat this as if there were a call forward instead... + */ + ast_string_field_set(p->owner, call_forward, peerorhost); + ast_queue_control(p->owner, AST_CONTROL_BUSY); + return 0; + } + } + + if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->pendinginvite) { + /* We already have a pending invite. Sorry. You are on hold. */ + transmit_response_reliable(p, "491 Request Pending", req); + if (option_debug) + ast_log(LOG_DEBUG, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid); + /* Don't destroy dialog here */ + return 0; + } + + p_replaces = get_header(req, "Replaces"); + if (!ast_strlen_zero(p_replaces)) { + /* We have a replaces header */ + char *ptr; + char *fromtag = NULL; + char *totag = NULL; + char *start, *to; + int error = 0; + + if (p->owner) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid); + transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */ + /* Do not destroy existing call */ + return -1; + } + + if (sipdebug && option_debug > 2) + ast_log(LOG_DEBUG, "INVITE part of call transfer. Replaces [%s]\n", p_replaces); + /* Create a buffer we can manipulate */ + replace_id = ast_strdupa(p_replaces); + ast_uri_decode(replace_id); + + if (!p->refer && !sip_refer_allocate(p)) { + transmit_response_reliable(p, "500 Server Internal Error", req); + append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory."); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + p->invitestate = INV_COMPLETED; + return -1; + } + + /* Todo: (When we find phones that support this) + if the replaces header contains ";early-only" + we can only replace the call in early + stage, not after it's up. + + If it's not in early mode, 486 Busy. + */ + + /* Skip leading whitespace */ + replace_id = ast_skip_blanks(replace_id); + + start = replace_id; + while ( (ptr = strsep(&start, ";")) ) { + ptr = ast_skip_blanks(ptr); /* XXX maybe unnecessary ? */ + if ( (to = strcasestr(ptr, "to-tag=") ) ) + totag = to + 7; /* skip the keyword */ + else if ( (to = strcasestr(ptr, "from-tag=") ) ) { + fromtag = to + 9; /* skip the keyword */ + fromtag = strsep(&fromtag, "&"); /* trim what ? */ + } + } + + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG,"Invite/replaces: Will use Replace-Call-ID : %s Fromtag: %s Totag: %s\n", replace_id, fromtag ? fromtag : "<no from tag>", totag ? totag : "<no to tag>"); + + + /* Try to find call that we are replacing + If we have a Replaces header, we need to cancel that call if we succeed with this call + */ + if ((p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) { + ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id); + transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req); + error = 1; + } + + /* At this point, bot the pvt and the owner of the call to be replaced is locked */ + + /* The matched call is the call from the transferer to Asterisk . + We want to bridge the bridged part of the call to the + incoming invite, thus taking over the refered call */ + + if (p->refer->refer_call == p) { + ast_log(LOG_NOTICE, "INVITE with replaces into it's own call id (%s == %s)!\n", replace_id, p->callid); + p->refer->refer_call = NULL; + transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */ + error = 1; + } + + if (!error && !p->refer->refer_call->owner) { + /* Oops, someting wrong anyway, no owner, no call */ + ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existing call id (%s)!\n", replace_id); + /* Check for better return code */ + transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replace)", req); + error = 1; + } + + if (!error && p->refer->refer_call->owner->_state != AST_STATE_RINGING && p->refer->refer_call->owner->_state != AST_STATE_RING && p->refer->refer_call->owner->_state != AST_STATE_UP ) { + ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id); + transmit_response_reliable(p, "603 Declined (Replaces)", req); + error = 1; + } + + if (error) { /* Give up this dialog */ + append_history(p, "Xfer", "INVITE/Replace Failed."); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + ast_mutex_unlock(&p->lock); + if (p->refer->refer_call) { + ast_mutex_unlock(&p->refer->refer_call->lock); + if (p->refer->refer_call->owner) { + ast_channel_unlock(p->refer->refer_call->owner); + } + } + p->invitestate = INV_COMPLETED; + return -1; + } + } + + + /* Check if this is an INVITE that sets up a new dialog or + a re-invite in an existing dialog */ + + if (!ast_test_flag(req, SIP_PKT_IGNORE)) { + int newcall = (p->initreq.headers ? TRUE : FALSE); + + if (sip_cancel_destroy(p)) + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + /* This also counts as a pending invite */ + p->pendinginvite = seqno; + check_via(p, req); + + copy_request(&p->initreq, req); /* Save this INVITE as the transaction basis */ + if (!p->owner) { /* Not a re-invite */ + if (debug) + ast_verbose("Using INVITE request as basis request - %s\n", p->callid); + if (newcall) + append_history(p, "Invite", "New call: %s", p->callid); + parse_ok_contact(p, req); + } else { /* Re-invite on existing call */ + ast_clear_flag(&p->flags[0], SIP_OUTGOING); /* This is now an inbound dialog */ + /* Handle SDP here if we already have an owner */ + if (find_sdp(req)) { + if (process_sdp(p, req)) { + transmit_response_reliable(p, "488 Not acceptable here", req); + if (!p->lastinvite) + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return -1; + } + } else { + p->jointcapability = p->capability; + if (option_debug > 2) + ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n"); + /* Some devices signal they want to be put off hold by sending a re-invite + *without* an SDP, which is supposed to mean "Go back to your state" + and since they put os on remote hold, we go back to off hold */ + if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) + change_hold_state(p, req, FALSE, 0); + } + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a response, note what it was for */ + append_history(p, "ReInv", "Re-invite received"); + } + } else if (debug) + ast_verbose("Ignoring this INVITE request\n"); + + + if (!p->lastinvite && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner) { + /* This is a new invite */ + /* Handle authentication if this is our first invite */ + res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin); + if (res == AUTH_CHALLENGE_SENT) { + p->invitestate = INV_COMPLETED; /* Needs to restart in another INVITE transaction */ + return 0; + } + if (res < 0) { /* Something failed in authentication */ + if (res == AUTH_FAKE_AUTH) { + ast_log(LOG_NOTICE, "Sending fake auth rejection for user %s\n", get_header(req, "From")); + transmit_fake_auth_response(p, req, 1); + } else { + ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From")); + transmit_response_reliable(p, "403 Forbidden", req); + } + p->invitestate = INV_COMPLETED; + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + ast_string_field_free(p, theirtag); + return 0; + } + + /* We have a succesful authentication, process the SDP portion if there is one */ + if (find_sdp(req)) { + if (process_sdp(p, req)) { + /* Unacceptable codecs */ + transmit_response_reliable(p, "488 Not acceptable here", req); + p->invitestate = INV_COMPLETED; + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + if (option_debug) + ast_log(LOG_DEBUG, "No compatible codecs for this SIP call.\n"); + return -1; + } + } else { /* No SDP in invite, call control session */ + p->jointcapability = p->capability; + if (option_debug > 1) + ast_log(LOG_DEBUG, "No SDP in Invite, third party call control\n"); + } + + /* Queue NULL frame to prod ast_rtp_bridge if appropriate */ + /* This seems redundant ... see !p-owner above */ + if (p->owner) + ast_queue_frame(p->owner, &ast_null_frame); + + + /* Initialize the context if it hasn't been already */ + if (ast_strlen_zero(p->context)) + ast_string_field_set(p, context, default_context); + + + /* Check number of concurrent calls -vs- incoming limit HERE */ + if (option_debug) + ast_log(LOG_DEBUG, "Checking SIP call limits for device %s\n", p->username); + if ((res = update_call_counter(p, INC_CALL_LIMIT))) { + if (res < 0) { + ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username); + transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + p->invitestate = INV_COMPLETED; + } + return 0; + } + gotdest = get_destination(p, NULL); /* Get destination right away */ + get_rdnis(p, NULL); /* Get redirect information */ + extract_uri(p, req); /* Get the Contact URI */ + build_contact(p); /* Build our contact header */ + + if (p->rtp) { + ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833); + ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE)); + } + + if (!replace_id && gotdest) { /* No matching extension found */ + if (gotdest == 1 && ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) + transmit_response_reliable(p, "484 Address Incomplete", req); + else { + char *decoded_exten = ast_strdupa(p->exten); + + transmit_response_reliable(p, "404 Not Found", req); + ast_uri_decode(decoded_exten); + ast_log(LOG_NOTICE, "Call from '%s' to extension" + " '%s' rejected because extension not found.\n", + S_OR(p->username, p->peername), decoded_exten); + } + p->invitestate = INV_COMPLETED; + update_call_counter(p, DEC_CALL_LIMIT); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return 0; + } else { + /* If no extension was specified, use the s one */ + /* Basically for calling to IP/Host name only */ + if (ast_strlen_zero(p->exten)) + ast_string_field_set(p, exten, "s"); + /* Initialize our tag */ + + make_our_tag(p->tag, sizeof(p->tag)); + /* First invitation - create the channel */ + c = sip_new(p, AST_STATE_DOWN, S_OR(p->username, NULL)); + *recount = 1; + + /* Save Record-Route for any later requests we make on this dialogue */ + build_route(p, req, 0); + + if (c) { + /* Pre-lock the call */ + ast_channel_lock(c); + } + } + } else { + if (option_debug > 1 && sipdebug) { + if (!ast_test_flag(req, SIP_PKT_IGNORE)) + ast_log(LOG_DEBUG, "Got a SIP re-invite for call %s\n", p->callid); + else + ast_log(LOG_DEBUG, "Got a SIP re-transmit of INVITE for call %s\n", p->callid); + } + if (!ast_test_flag(req, SIP_PKT_IGNORE)) + reinvite = 1; + c = p->owner; + } + + if (!ast_test_flag(req, SIP_PKT_IGNORE) && p) + p->lastinvite = seqno; + + if (replace_id) { /* Attended transfer or call pickup - we're the target */ + /* Go and take over the target call */ + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG, "Sending this call to the invite/replcaes handler %s\n", p->callid); + return handle_invite_replaces(p, req, debug, ast_test_flag(req, SIP_PKT_IGNORE), seqno, sin); + } + + + if (c) { /* We have a call -either a new call or an old one (RE-INVITE) */ + switch(c->_state) { + case AST_STATE_DOWN: + if (option_debug > 1) + ast_log(LOG_DEBUG, "%s: New call is still down.... Trying... \n", c->name); + transmit_response(p, "100 Trying", req); + p->invitestate = INV_PROCEEDING; + ast_setstate(c, AST_STATE_RING); + if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */ + enum ast_pbx_result res; + + res = ast_pbx_start(c); + + switch(res) { + case AST_PBX_FAILED: + ast_log(LOG_WARNING, "Failed to start PBX :(\n"); + p->invitestate = INV_COMPLETED; + if (ast_test_flag(req, SIP_PKT_IGNORE)) + transmit_response(p, "503 Unavailable", req); + else + transmit_response_reliable(p, "503 Unavailable", req); + break; + case AST_PBX_CALL_LIMIT: + ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n"); + p->invitestate = INV_COMPLETED; + if (ast_test_flag(req, SIP_PKT_IGNORE)) + transmit_response(p, "480 Temporarily Unavailable", req); + else + transmit_response_reliable(p, "480 Temporarily Unavailable", req); + break; + case AST_PBX_SUCCESS: + /* nothing to do */ + break; + } + + if (res) { + + /* Unlock locks so ast_hangup can do its magic */ + ast_mutex_unlock(&c->lock); + ast_mutex_unlock(&p->lock); + ast_hangup(c); + ast_mutex_lock(&p->lock); + c = NULL; + } + } else { /* Pickup call in call group */ + ast_channel_unlock(c); + *nounlock = 1; + if (ast_pickup_call(c)) { + ast_log(LOG_NOTICE, "Nothing to pick up for %s\n", p->callid); + if (ast_test_flag(req, SIP_PKT_IGNORE)) + transmit_response(p, "503 Unavailable", req); /* OEJ - Right answer? */ + else + transmit_response_reliable(p, "503 Unavailable", req); + sip_alreadygone(p); + /* Unlock locks so ast_hangup can do its magic */ + ast_mutex_unlock(&p->lock); + c->hangupcause = AST_CAUSE_CALL_REJECTED; + } else { + ast_mutex_unlock(&p->lock); + ast_setstate(c, AST_STATE_DOWN); + c->hangupcause = AST_CAUSE_NORMAL_CLEARING; + } + p->invitestate = INV_COMPLETED; + ast_hangup(c); + ast_mutex_lock(&p->lock); + c = NULL; + } + break; + case AST_STATE_RING: + transmit_response(p, "100 Trying", req); + p->invitestate = INV_PROCEEDING; + break; + case AST_STATE_RINGING: + transmit_response(p, "180 Ringing", req); + p->invitestate = INV_PROCEEDING; + break; + case AST_STATE_UP: + if (option_debug > 1) + ast_log(LOG_DEBUG, "%s: This call is UP.... \n", c->name); + + transmit_response(p, "100 Trying", req); + + if (p->t38.state == T38_PEER_REINVITE) { + struct ast_channel *bridgepeer = NULL; + struct sip_pvt *bridgepvt = NULL; + + if ((bridgepeer = ast_bridged_channel(p->owner))) { + /* We have a bridge, and this is re-invite to switchover to T38 so we send re-invite with T38 SDP, to other side of bridge*/ + /*! XXX: we should also check here does the other side supports t38 at all !!! XXX */ + if (bridgepeer->tech == &sip_tech || bridgepeer->tech == &sip_tech_info) { + bridgepvt = (struct sip_pvt*)bridgepeer->tech_pvt; + if (bridgepvt->t38.state == T38_DISABLED) { + if (bridgepvt->udptl) { /* If everything is OK with other side's udptl struct */ + /* Send re-invite to the bridged channel */ + sip_handle_t38_reinvite(bridgepeer, p, 1); + } else { /* Something is wrong with peers udptl struct */ + ast_log(LOG_WARNING, "Strange... The other side of the bridge don't have udptl struct\n"); + ast_mutex_lock(&bridgepvt->lock); + bridgepvt->t38.state = T38_DISABLED; + ast_mutex_unlock(&bridgepvt->lock); + if (option_debug > 1) + ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->name); + if (ast_test_flag(req, SIP_PKT_IGNORE)) + transmit_response(p, "488 Not acceptable here", req); + else + transmit_response_reliable(p, "488 Not acceptable here", req); + + } + } else { + /* The other side is already setup for T.38 most likely so we need to acknowledge this too */ + ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + transmit_response_with_t38_sdp(p, "200 OK", req, XMIT_CRITICAL); + p->t38.state = T38_ENABLED; + if (option_debug) + ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + } + } else { + /* Other side is not a SIP channel */ + if (ast_test_flag(req, SIP_PKT_IGNORE)) + transmit_response(p, "488 Not acceptable here", req); + else + transmit_response_reliable(p, "488 Not acceptable here", req); + p->t38.state = T38_DISABLED; + if (option_debug > 1) + ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + + if (!p->lastinvite) /* Only destroy if this is *not* a re-invite */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } + } else { + /* we are not bridged in a call */ + ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + transmit_response_with_t38_sdp(p, "200 OK", req, XMIT_CRITICAL); + p->t38.state = T38_ENABLED; + if (option_debug) + ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + } + } else if (p->t38.state == T38_DISABLED) { /* Channel doesn't have T38 offered or enabled */ + int sendok = TRUE; + + /* If we are bridged to a channel that has T38 enabled than this is a case of RTP re-invite after T38 session */ + /* so handle it here (re-invite other party to RTP) */ + struct ast_channel *bridgepeer = NULL; + struct sip_pvt *bridgepvt = NULL; + if ((bridgepeer = ast_bridged_channel(p->owner))) { + if ((bridgepeer->tech == &sip_tech || bridgepeer->tech == &sip_tech_info) && !ast_check_hangup(bridgepeer)) { + bridgepvt = (struct sip_pvt*)bridgepeer->tech_pvt; + /* Does the bridged peer have T38 ? */ + if (bridgepvt->t38.state == T38_ENABLED) { + ast_log(LOG_WARNING, "RTP re-invite after T38 session not handled yet !\n"); + /* Insted of this we should somehow re-invite the other side of the bridge to RTP */ + if (ast_test_flag(req, SIP_PKT_IGNORE)) + transmit_response(p, "488 Not Acceptable Here (unsupported)", req); + else + transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req); + sendok = FALSE; + } + /* No bridged peer with T38 enabled*/ + } + } + /* Respond to normal re-invite */ + if (sendok) { + /* If this is not a re-invite or something to ignore - it's critical */ + ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (ast_test_flag(req, SIP_PKT_IGNORE) ? XMIT_UNRELIABLE : XMIT_CRITICAL))); + } + } + p->invitestate = INV_TERMINATED; + break; + default: + ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state); + transmit_response(p, "100 Trying", req); + break; + } + } else { + if (p && (p->autokillid == -1)) { + const char *msg; + + if (!p->jointcapability) + msg = "488 Not Acceptable Here (codec error)"; + else { + ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n"); + msg = "503 Unavailable"; + } + if (ast_test_flag(req, SIP_PKT_IGNORE)) + transmit_response(p, msg, req); + else + transmit_response_reliable(p, msg, req); + p->invitestate = INV_COMPLETED; + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } + } + return res; +} + +/*! \brief Find all call legs and bridge transferee with target + * called from handle_request_refer */ +static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno) +{ + struct sip_dual target; /* Chan 1: Call from tranferer to Asterisk */ + /* Chan 2: Call from Asterisk to target */ + int res = 0; + struct sip_pvt *targetcall_pvt; + + /* Check if the call ID of the replaces header does exist locally */ + if (!(targetcall_pvt = get_sip_pvt_byid_locked(transferer->refer->replaces_callid, transferer->refer->replaces_callid_totag, + transferer->refer->replaces_callid_fromtag))) { + if (transferer->refer->localtransfer) { + /* We did not find the refered call. Sorry, can't accept then */ + transmit_response(transferer, "202 Accepted", req); + /* Let's fake a response from someone else in order + to follow the standard */ + transmit_notify_with_sipfrag(transferer, seqno, "481 Call leg/transaction does not exist", TRUE); + append_history(transferer, "Xfer", "Refer failed"); + ast_clear_flag(&transferer->flags[0], SIP_GOTREFER); + transferer->refer->status = REFER_FAILED; + return -1; + } + /* Fall through for remote transfers that we did not find locally */ + if (option_debug > 2) + ast_log(LOG_DEBUG, "SIP attended transfer: Not our call - generating INVITE with replaces\n"); + return 0; + } + + /* Ok, we can accept this transfer */ + transmit_response(transferer, "202 Accepted", req); + append_history(transferer, "Xfer", "Refer accepted"); + if (!targetcall_pvt->owner) { /* No active channel */ + if (option_debug > 3) + ast_log(LOG_DEBUG, "SIP attended transfer: Error: No owner of target call\n"); + /* Cancel transfer */ + transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE); + append_history(transferer, "Xfer", "Refer failed"); + ast_clear_flag(&transferer->flags[0], SIP_GOTREFER); + transferer->refer->status = REFER_FAILED; + ast_mutex_unlock(&targetcall_pvt->lock); + ast_channel_unlock(current->chan1); + return -1; + } + + /* We have a channel, find the bridge */ + target.chan1 = targetcall_pvt->owner; /* Transferer to Asterisk */ + target.chan2 = ast_bridged_channel(targetcall_pvt->owner); /* Asterisk to target */ + + if (!target.chan2 || !(target.chan2->_state == AST_STATE_UP || target.chan2->_state == AST_STATE_RINGING) ) { + /* Wrong state of new channel */ + if (option_debug > 3) { + if (target.chan2) + ast_log(LOG_DEBUG, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(target.chan2->_state)); + else if (target.chan1->_state != AST_STATE_RING) + ast_log(LOG_DEBUG, "SIP attended transfer: Error: No target channel\n"); + else + ast_log(LOG_DEBUG, "SIP attended transfer: Attempting transfer in ringing state\n"); + } + } + + /* Transfer */ + if (option_debug > 3 && sipdebug) { + if (current->chan2) /* We have two bridges */ + ast_log(LOG_DEBUG, "SIP attended transfer: trying to bridge %s and %s\n", target.chan1->name, current->chan2->name); + else /* One bridge, propably transfer of IVR/voicemail etc */ + ast_log(LOG_DEBUG, "SIP attended transfer: trying to make %s take over (masq) %s\n", target.chan1->name, current->chan1->name); + } + + ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */ + + /* Perform the transfer */ + res = attempt_transfer(current, &target); + ast_mutex_unlock(&targetcall_pvt->lock); + if (res) { + /* Failed transfer */ + transmit_notify_with_sipfrag(transferer, seqno, "486 Busy Here", TRUE); + append_history(transferer, "Xfer", "Refer failed"); + transferer->refer->status = REFER_FAILED; + if (targetcall_pvt->owner) + ast_channel_unlock(targetcall_pvt->owner); + /* Right now, we have to hangup, sorry. Bridge is destroyed */ + if (res != -2) + ast_hangup(transferer->owner); + else + ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER); + } else { + /* Transfer succeeded! */ + + /* Tell transferer that we're done. */ + transmit_notify_with_sipfrag(transferer, seqno, "200 OK", TRUE); + append_history(transferer, "Xfer", "Refer succeeded"); + transferer->refer->status = REFER_200OK; + if (targetcall_pvt->owner) { + if (option_debug) + ast_log(LOG_DEBUG, "SIP attended transfer: Unlocking channel %s\n", targetcall_pvt->owner->name); + ast_channel_unlock(targetcall_pvt->owner); + } + } + return 1; +} + + +/*! \brief Handle incoming REFER request */ +/*! \page SIP_REFER SIP transfer Support (REFER) + + REFER is used for call transfer in SIP. We get a REFER + to place a new call with an INVITE somwhere and then + keep the transferor up-to-date of the transfer. If the + transfer fails, get back on line with the orginal call. + + - REFER can be sent outside or inside of a dialog. + Asterisk only accepts REFER inside of a dialog. + + - If we get a replaces header, it is an attended transfer + + \par Blind transfers + The transferor provides the transferee + with the transfer targets contact. The signalling between + transferer or transferee should not be cancelled, so the + call is recoverable if the transfer target can not be reached + by the transferee. + + In this case, Asterisk receives a TRANSFER from + the transferor, thus is the transferee. We should + try to set up a call to the contact provided + and if that fails, re-connect the current session. + If the new call is set up, we issue a hangup. + In this scenario, we are following section 5.2 + in the SIP CC Transfer draft. (Transfer without + a GRUU) + + \par Transfer with consultation hold + In this case, the transferor + talks to the transfer target before the transfer takes place. + This is implemented with SIP hold and transfer. + Note: The invite From: string could indicate a transfer. + (Section 6. Transfer with consultation hold) + The transferor places the transferee on hold, starts a call + with the transfer target to alert them to the impending + transfer, terminates the connection with the target, then + proceeds with the transfer (as in Blind transfer above) + + \par Attended transfer + The transferor places the transferee + on hold, calls the transfer target to alert them, + places the target on hold, then proceeds with the transfer + using a Replaces header field in the Refer-to header. This + will force the transfee to send an Invite to the target, + with a replaces header that instructs the target to + hangup the call between the transferor and the target. + In this case, the Refer/to: uses the AOR address. (The same + URI that the transferee used to establish the session with + the transfer target (To: ). The Require: replaces header should + be in the INVITE to avoid the wrong UA in a forked SIP proxy + scenario to answer and have no call to replace with. + + The referred-by header is *NOT* required, but if we get it, + can be copied into the INVITE to the transfer target to + inform the target about the transferor + + "Any REFER request has to be appropriately authenticated.". + + We can't destroy dialogs, since we want the call to continue. + + */ +static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, int *nounlock) +{ + struct sip_dual current; /* Chan1: Call between asterisk and transferer */ + /* Chan2: Call between asterisk and transferee */ + + int res = 0; + + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n", p->callid, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "callee" : "caller"); + + if (!p->owner) { + /* This is a REFER outside of an existing SIP dialog */ + /* We can't handle that, so decline it */ + if (option_debug > 2) + ast_log(LOG_DEBUG, "Call %s: Declined REFER, outside of dialog...\n", p->callid); + transmit_response(p, "603 Declined (No dialog)", req); + if (!ast_test_flag(req, SIP_PKT_IGNORE)) { + append_history(p, "Xfer", "Refer failed. Outside of dialog."); + sip_alreadygone(p); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + return 0; + } + + + /* Check if transfer is allowed from this device */ + if (p->allowtransfer == TRANSFER_CLOSED ) { + /* Transfer not allowed, decline */ + transmit_response(p, "603 Declined (policy)", req); + append_history(p, "Xfer", "Refer failed. Allowtransfer == closed."); + /* Do not destroy SIP session */ + return 0; + } + + if(!ignore && ast_test_flag(&p->flags[0], SIP_GOTREFER)) { + /* Already have a pending REFER */ + transmit_response(p, "491 Request pending", req); + append_history(p, "Xfer", "Refer failed. Request pending."); + return 0; + } + + /* Allocate memory for call transfer data */ + if (!p->refer && !sip_refer_allocate(p)) { + transmit_response(p, "500 Internal Server Error", req); + append_history(p, "Xfer", "Refer failed. Memory allocation error."); + return -3; + } + + res = get_refer_info(p, req); /* Extract headers */ + + p->refer->status = REFER_SENT; + + if (res != 0) { + switch (res) { + case -2: /* Syntax error */ + transmit_response(p, "400 Bad Request (Refer-to missing)", req); + append_history(p, "Xfer", "Refer failed. Refer-to missing."); + if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug) + ast_log(LOG_DEBUG, "SIP transfer to black hole can't be handled (no refer-to: )\n"); + break; + case -3: + transmit_response(p, "603 Declined (Non sip: uri)", req); + append_history(p, "Xfer", "Refer failed. Non SIP uri"); + if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug) + ast_log(LOG_DEBUG, "SIP transfer to non-SIP uri denied\n"); + break; + default: + /* Refer-to extension not found, fake a failed transfer */ + transmit_response(p, "202 Accepted", req); + append_history(p, "Xfer", "Refer failed. Bad extension."); + transmit_notify_with_sipfrag(p, seqno, "404 Not found", TRUE); + ast_clear_flag(&p->flags[0], SIP_GOTREFER); + if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug) + ast_log(LOG_DEBUG, "SIP transfer to bad extension: %s\n", p->refer->refer_to); + break; + } + return 0; + } + if (ast_strlen_zero(p->context)) + ast_string_field_set(p, context, default_context); + + /* If we do not support SIP domains, all transfers are local */ + if (allow_external_domains && check_sip_domain(p->refer->refer_to_domain, NULL, 0)) { + p->refer->localtransfer = 1; + if (sipdebug && option_debug > 2) + ast_log(LOG_DEBUG, "This SIP transfer is local : %s\n", p->refer->refer_to_domain); + } else if (AST_LIST_EMPTY(&domain_list) || check_sip_domain(p->refer->refer_to_domain, NULL, 0)) { + /* This PBX doesn't bother with SIP domains or domain is local, so this transfer is local */ + p->refer->localtransfer = 1; + } else if (sipdebug && option_debug > 2) + ast_log(LOG_DEBUG, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain); + + /* Is this a repeat of a current request? Ignore it */ + /* Don't know what else to do right now. */ + if (ignore) + return res; + + /* If this is a blind transfer, we have the following + channels to work with: + - chan1, chan2: The current call between transferer and transferee (2 channels) + - target_channel: A new call from the transferee to the target (1 channel) + We need to stay tuned to what happens in order to be able + to bring back the call to the transferer */ + + /* If this is a attended transfer, we should have all call legs within reach: + - chan1, chan2: The call between the transferer and transferee (2 channels) + - target_channel, targetcall_pvt: The call between the transferer and the target (2 channels) + We want to bridge chan2 with targetcall_pvt! + + The replaces call id in the refer message points + to the call leg between Asterisk and the transferer. + So we need to connect the target and the transferee channel + and hangup the two other channels silently + + If the target is non-local, the call ID could be on a remote + machine and we need to send an INVITE with replaces to the + target. We basically handle this as a blind transfer + and let the sip_call function catch that we need replaces + header in the INVITE. + */ + + + /* Get the transferer's channel */ + current.chan1 = p->owner; + + /* Find the other part of the bridge (2) - transferee */ + current.chan2 = ast_bridged_channel(current.chan1); + + if (sipdebug && option_debug > 2) + ast_log(LOG_DEBUG, "SIP %s transfer: Transferer channel %s, transferee channel %s\n", p->refer->attendedtransfer ? "attended" : "blind", current.chan1->name, current.chan2 ? current.chan2->name : "<none>"); + + if (!current.chan2 && !p->refer->attendedtransfer) { + /* No bridged channel, propably IVR or echo or similar... */ + /* Guess we should masquerade or something here */ + /* Until we figure it out, refuse transfer of such calls */ + if (sipdebug && option_debug > 2) + ast_log(LOG_DEBUG,"Refused SIP transfer on non-bridged channel.\n"); + p->refer->status = REFER_FAILED; + append_history(p, "Xfer", "Refer failed. Non-bridged channel."); + transmit_response(p, "603 Declined", req); + return -1; + } + + if (current.chan2) { + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG, "Got SIP transfer, applying to bridged peer '%s'\n", current.chan2->name); + + ast_queue_control(current.chan1, AST_CONTROL_UNHOLD); + } + + ast_set_flag(&p->flags[0], SIP_GOTREFER); + + /* Attended transfer: Find all call legs and bridge transferee with target*/ + if (p->refer->attendedtransfer) { + if ((res = local_attended_transfer(p, ¤t, req, seqno))) + return res; /* We're done with the transfer */ + /* Fall through for remote transfers that we did not find locally */ + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG, "SIP attended transfer: Still not our call - generating INVITE with replaces\n"); + /* Fallthrough if we can't find the call leg internally */ + } + + + /* Parking a call */ + if (p->refer->localtransfer && !strcmp(p->refer->refer_to, ast_parking_ext())) { + /* Must release c's lock now, because it will not longer be accessible after the transfer! */ + *nounlock = 1; + ast_channel_unlock(current.chan1); + copy_request(¤t.req, req); + ast_clear_flag(&p->flags[0], SIP_GOTREFER); + p->refer->status = REFER_200OK; + append_history(p, "Xfer", "REFER to call parking."); + if (sipdebug && option_debug > 3) + ast_log(LOG_DEBUG, "SIP transfer to parking: trying to park %s. Parked by %s\n", current.chan2->name, current.chan1->name); + sip_park(current.chan2, current.chan1, req, seqno); + return res; + } + + /* Blind transfers and remote attended xfers */ + transmit_response(p, "202 Accepted", req); + + if (current.chan1 && current.chan2) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "chan1->name: %s\n", current.chan1->name); + pbx_builtin_setvar_helper(current.chan1, "BLINDTRANSFER", current.chan2->name); + } + if (current.chan2) { + pbx_builtin_setvar_helper(current.chan2, "BLINDTRANSFER", current.chan1->name); + pbx_builtin_setvar_helper(current.chan2, "SIPDOMAIN", p->refer->refer_to_domain); + pbx_builtin_setvar_helper(current.chan2, "SIPTRANSFER", "yes"); + /* One for the new channel */ + pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER", "yes"); + /* Attended transfer to remote host, prepare headers for the INVITE */ + if (p->refer->referred_by) + pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REFERER", p->refer->referred_by); + } + /* Generate a Replaces string to be used in the INVITE during attended transfer */ + if (p->refer->replaces_callid && !ast_strlen_zero(p->refer->replaces_callid)) { + char tempheader[SIPBUFSIZE]; + snprintf(tempheader, sizeof(tempheader), "%s%s%s%s%s", p->refer->replaces_callid, + p->refer->replaces_callid_totag ? ";to-tag=" : "", + p->refer->replaces_callid_totag, + p->refer->replaces_callid_fromtag ? ";from-tag=" : "", + p->refer->replaces_callid_fromtag); + if (current.chan2) + pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REPLACES", tempheader); + } + /* Must release lock now, because it will not longer + be accessible after the transfer! */ + *nounlock = 1; + ast_channel_unlock(current.chan1); + + /* Connect the call */ + + /* FAKE ringing if not attended transfer */ + if (!p->refer->attendedtransfer) + transmit_notify_with_sipfrag(p, seqno, "183 Ringing", FALSE); + + /* For blind transfer, this will lead to a new call */ + /* For attended transfer to remote host, this will lead to + a new SIP call with a replaces header, if the dial plan allows it + */ + if (!current.chan2) { + /* We have no bridge, so we're talking with Asterisk somehow */ + /* We need to masquerade this call */ + /* What to do to fix this situation: + * Set up the new call in a new channel + * Let the new channel masq into this channel + Please add that code here :-) + */ + p->refer->status = REFER_FAILED; + transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable (can't handle one-legged xfers)", TRUE); + ast_clear_flag(&p->flags[0], SIP_GOTREFER); + append_history(p, "Xfer", "Refer failed (only bridged calls)."); + return -1; + } + ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */ + + /* For blind transfers, move the call to the new extensions. For attended transfers on multiple + servers - generate an INVITE with Replaces. Either way, let the dial plan decided */ + res = ast_async_goto(current.chan2, p->refer->refer_to_context, p->refer->refer_to, 1); + + if (!res) { + /* Success - we have a new channel */ + if (option_debug > 2) + ast_log(LOG_DEBUG, "%s transfer succeeded. Telling transferer.\n", p->refer->attendedtransfer? "Attended" : "Blind"); + transmit_notify_with_sipfrag(p, seqno, "200 Ok", TRUE); + if (p->refer->localtransfer) + p->refer->status = REFER_200OK; + if (p->owner) + p->owner->hangupcause = AST_CAUSE_NORMAL_CLEARING; + append_history(p, "Xfer", "Refer succeeded."); + ast_clear_flag(&p->flags[0], SIP_GOTREFER); + /* Do not hangup call, the other side do that when we say 200 OK */ + /* We could possibly implement a timer here, auto congestion */ + res = 0; + } else { + ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Don't delay hangup */ + if (option_debug > 2) + ast_log(LOG_DEBUG, "%s transfer failed. Resuming original call.\n", p->refer->attendedtransfer? "Attended" : "Blind"); + append_history(p, "Xfer", "Refer failed."); + /* Failure of some kind */ + p->refer->status = REFER_FAILED; + transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable", TRUE); + ast_clear_flag(&p->flags[0], SIP_GOTREFER); + res = -1; + } + return res; +} + +/*! \brief Handle incoming CANCEL request */ +static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req) +{ + + check_via(p, req); + sip_alreadygone(p); + + /* At this point, we could have cancelled the invite at the same time + as the other side sends a CANCEL. Our final reply with error code + might not have been received by the other side before the CANCEL + was sent, so let's just give up retransmissions and waiting for + ACK on our error code. The call is hanging up any way. */ + if (p->invitestate == INV_TERMINATED) + __sip_pretend_ack(p); + else + p->invitestate = INV_CANCELLED; + + if (p->owner && p->owner->_state == AST_STATE_UP) { + /* This call is up, cancel is ignored, we need a bye */ + transmit_response(p, "200 OK", req); + if (option_debug) + ast_log(LOG_DEBUG, "Got CANCEL on an answered call. Ignoring... \n"); + return 0; + } + + if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) + update_call_counter(p, DEC_CALL_LIMIT); + + stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */ + if (p->owner) + ast_queue_hangup(p->owner); + else + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + if (p->initreq.len > 0) { + transmit_response_reliable(p, "487 Request Terminated", &p->initreq); + transmit_response(p, "200 OK", req); + return 1; + } else { + transmit_response(p, "481 Call Leg Does Not Exist", req); + return 0; + } +} + +static int acf_channel_read(struct ast_channel *chan, char *funcname, char *preparse, char *buf, size_t buflen) +{ + struct ast_rtp_quality qos; + struct sip_pvt *p = chan->tech_pvt; + char *all = "", *parse = ast_strdupa(preparse); + AST_DECLARE_APP_ARGS(args, + AST_APP_ARG(param); + AST_APP_ARG(type); + AST_APP_ARG(field); + ); + AST_STANDARD_APP_ARGS(args, parse); + + /* Sanity check */ + if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) { + ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname); + return 0; + } + + if (strcasecmp(args.param, "rtpqos")) + return 0; + + /* Default arguments of audio,all */ + if (ast_strlen_zero(args.type)) + args.type = "audio"; + if (ast_strlen_zero(args.field)) + args.field = "all"; + + memset(buf, 0, buflen); + memset(&qos, 0, sizeof(qos)); + + if (strcasecmp(args.type, "AUDIO") == 0) { + all = ast_rtp_get_quality(p->rtp, &qos); + } else if (strcasecmp(args.type, "VIDEO") == 0) { + all = ast_rtp_get_quality(p->vrtp, &qos); + } + + if (strcasecmp(args.field, "local_ssrc") == 0) + snprintf(buf, buflen, "%u", qos.local_ssrc); + else if (strcasecmp(args.field, "local_lostpackets") == 0) + snprintf(buf, buflen, "%u", qos.local_lostpackets); + else if (strcasecmp(args.field, "local_jitter") == 0) + snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0); + else if (strcasecmp(args.field, "local_count") == 0) + snprintf(buf, buflen, "%u", qos.local_count); + else if (strcasecmp(args.field, "remote_ssrc") == 0) + snprintf(buf, buflen, "%u", qos.remote_ssrc); + else if (strcasecmp(args.field, "remote_lostpackets") == 0) + snprintf(buf, buflen, "%u", qos.remote_lostpackets); + else if (strcasecmp(args.field, "remote_jitter") == 0) + snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0); + else if (strcasecmp(args.field, "remote_count") == 0) + snprintf(buf, buflen, "%u", qos.remote_count); + else if (strcasecmp(args.field, "rtt") == 0) + snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0); + else if (strcasecmp(args.field, "all") == 0) + ast_copy_string(buf, all, buflen); + else { + ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname); + return -1; + } + return 0; +} + +/*! \brief Handle incoming BYE request */ +static int handle_request_bye(struct sip_pvt *p, struct sip_request *req) +{ + struct ast_channel *c=NULL; + int res; + struct ast_channel *bridged_to; + + /* If we have an INCOMING invite that we haven't answered, terminate that transaction */ + if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner) + transmit_response_reliable(p, "487 Request Terminated", &p->initreq); + + __sip_pretend_ack(p); + + p->invitestate = INV_TERMINATED; + + copy_request(&p->initreq, req); + check_via(p, req); + sip_alreadygone(p); + + /* Get RTCP quality before end of call */ + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) { + char *audioqos, *videoqos; + if (p->rtp) { + audioqos = ast_rtp_get_quality(p->rtp, NULL); + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) + append_history(p, "RTCPaudio", "Quality:%s", audioqos); + if (p->owner) + pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos); + } + if (p->vrtp) { + videoqos = ast_rtp_get_quality(p->vrtp, NULL); + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) + append_history(p, "RTCPvideo", "Quality:%s", videoqos); + if (p->owner) + pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos); + } + } + + stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */ + + if (!ast_strlen_zero(get_header(req, "Also"))) { + ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead\n", + ast_inet_ntoa(p->recv.sin_addr)); + if (ast_strlen_zero(p->context)) + ast_string_field_set(p, context, default_context); + res = get_also_info(p, req); + if (!res) { + c = p->owner; + if (c) { + bridged_to = ast_bridged_channel(c); + if (bridged_to) { + /* Don't actually hangup here... */ + ast_queue_control(c, AST_CONTROL_UNHOLD); + ast_async_goto(bridged_to, p->context, p->refer->refer_to,1); + } else + ast_queue_hangup(p->owner); + } + } else { + ast_log(LOG_WARNING, "Invalid transfer information from '%s'\n", ast_inet_ntoa(p->recv.sin_addr)); + if (p->owner) + ast_queue_hangup(p->owner); + } + } else if (p->owner) { + ast_queue_hangup(p->owner); + if (option_debug > 2) + ast_log(LOG_DEBUG, "Received bye, issuing owner hangup\n"); + } else { + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + if (option_debug > 2) + ast_log(LOG_DEBUG, "Received bye, no owner, selfdestruct soon.\n"); + } + ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + transmit_response(p, "200 OK", req); + + return 1; +} + +/*! \brief Handle incoming MESSAGE request */ +static int handle_request_message(struct sip_pvt *p, struct sip_request *req) +{ + if (!ast_test_flag(req, SIP_PKT_IGNORE)) { + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_verbose("Receiving message!\n"); + receive_message(p, req); + } else + transmit_response(p, "202 Accepted", req); + return 1; +} + +/*! \brief Handle incoming SUBSCRIBE request */ +static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e) +{ + int gotdest; + int res = 0; + int firststate = AST_EXTENSION_REMOVED; + struct sip_peer *authpeer = NULL; + const char *eventheader = get_header(req, "Event"); /* Get Event package name */ + const char *accept = get_header(req, "Accept"); + int resubscribe = (p->subscribed != NONE); + char *temp, *event; + + if (p->initreq.headers) { + /* We already have a dialog */ + if (p->initreq.method != SIP_SUBSCRIBE) { + /* This is a SUBSCRIBE within another SIP dialog, which we do not support */ + /* For transfers, this could happen, but since we haven't seen it happening, let us just refuse this */ + transmit_response(p, "403 Forbidden (within dialog)", req); + /* Do not destroy session, since we will break the call if we do */ + if (option_debug) + ast_log(LOG_DEBUG, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text); + return 0; + } else if (ast_test_flag(req, SIP_PKT_DEBUG)) { + if (option_debug) { + if (resubscribe) + ast_log(LOG_DEBUG, "Got a re-subscribe on existing subscription %s\n", p->callid); + else + ast_log(LOG_DEBUG, "Got a new subscription %s (possibly with auth)\n", p->callid); + } + } + } + + /* Check if we have a global disallow setting on subscriptions. + if so, we don't have to check peer/user settings after auth, which saves a lot of processing + */ + if (!global_allowsubscribe) { + transmit_response(p, "403 Forbidden (policy)", req); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + return 0; + } + + if (!ast_test_flag(req, SIP_PKT_IGNORE) && !resubscribe) { /* Set up dialog, new subscription */ + const char *to = get_header(req, "To"); + char totag[128]; + + /* Check to see if a tag was provided, if so this is actually a resubscription of a dialog we no longer know about */ + if (!ast_strlen_zero(to) && gettag(req, "To", totag, sizeof(totag))) { + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_verbose("Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.\n"); + transmit_response(p, "481 Subscription does not exist", req); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + return 0; + } + + /* Use this as the basis */ + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_verbose("Creating new subscription\n"); + + copy_request(&p->initreq, req); + check_via(p, req); + } else if (ast_test_flag(req, SIP_PKT_DEBUG) && ast_test_flag(req, SIP_PKT_IGNORE)) + ast_verbose("Ignoring this SUBSCRIBE request\n"); + + /* Find parameters to Event: header value and remove them for now */ + if (ast_strlen_zero(eventheader)) { + transmit_response(p, "489 Bad Event", req); + if (option_debug > 1) + ast_log(LOG_DEBUG, "Received SIP subscribe for unknown event package: <none>\n"); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + return 0; + } + + if ( (strchr(eventheader, ';'))) { + event = ast_strdupa(eventheader); /* Since eventheader is a const, we can't change it */ + temp = strchr(event, ';'); + *temp = '\0'; /* Remove any options for now */ + /* We might need to use them later :-) */ + } else + event = (char *) eventheader; /* XXX is this legal ? */ + + /* Handle authentication */ + res = check_user_full(p, req, SIP_SUBSCRIBE, e, 0, sin, &authpeer); + /* if an authentication response was sent, we are done here */ + if (res == AUTH_CHALLENGE_SENT) { + if (authpeer) + ASTOBJ_UNREF(authpeer, sip_destroy_peer); + return 0; + } + if (res < 0) { + if (res == AUTH_FAKE_AUTH) { + ast_log(LOG_NOTICE, "Sending fake auth rejection for user %s\n", get_header(req, "From")); + transmit_fake_auth_response(p, req, 1); + } else { + ast_log(LOG_NOTICE, "Failed to authenticate user %s for SUBSCRIBE\n", get_header(req, "From")); + transmit_response_reliable(p, "403 Forbidden", req); + } + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + if (authpeer) + ASTOBJ_UNREF(authpeer, sip_destroy_peer); + return 0; + } + + /* Check if this user/peer is allowed to subscribe at all */ + if (!ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) { + transmit_response(p, "403 Forbidden (policy)", req); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + if (authpeer) + ASTOBJ_UNREF(authpeer, sip_destroy_peer); + return 0; + } + + /* Get destination right away */ + gotdest = get_destination(p, NULL); + + /* Get full contact header - this needs to be used as a request URI in NOTIFY's */ + parse_ok_contact(p, req); + + build_contact(p); + if (gotdest) { + transmit_response(p, "404 Not Found", req); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + if (authpeer) + ASTOBJ_UNREF(authpeer, sip_destroy_peer); + return 0; + } + + /* Initialize tag for new subscriptions */ + if (ast_strlen_zero(p->tag)) + make_our_tag(p->tag, sizeof(p->tag)); + + if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */ + if (authpeer) /* No need for authpeer here */ + ASTOBJ_UNREF(authpeer, sip_destroy_peer); + + /* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */ + /* Polycom phones only handle xpidf+xml, even if they say they can + handle pidf+xml as well + */ + if (strstr(p->useragent, "Polycom")) { + p->subscribed = XPIDF_XML; + } else if (strstr(accept, "application/pidf+xml")) { + p->subscribed = PIDF_XML; /* RFC 3863 format */ + } else if (strstr(accept, "application/dialog-info+xml")) { + p->subscribed = DIALOG_INFO_XML; + /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */ + } else if (strstr(accept, "application/cpim-pidf+xml")) { + p->subscribed = CPIM_PIDF_XML; /* RFC 3863 format */ + } else if (strstr(accept, "application/xpidf+xml")) { + p->subscribed = XPIDF_XML; /* Early pre-RFC 3863 format with MSN additions (Microsoft Messenger) */ + } else if (ast_strlen_zero(accept)) { + if (p->subscribed == NONE) { /* if the subscribed field is not already set, and there is no accept header... */ + transmit_response(p, "489 Bad Event", req); + + ast_log(LOG_WARNING,"SUBSCRIBE failure: no Accept header: pvt: stateid: %d, laststate: %d, dialogver: %d, subscribecont: '%s', subscribeuri: '%s'\n", + p->stateid, p->laststate, p->dialogver, p->subscribecontext, p->subscribeuri); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + return 0; + } + /* if p->subscribed is non-zero, then accept is not obligatory; according to rfc 3265 section 3.1.3, at least. + so, we'll just let it ride, keeping the value from a previous subscription, and not abort the subscription */ + } else { + /* Can't find a format for events that we know about */ + char mybuf[200]; + snprintf(mybuf,sizeof(mybuf),"489 Bad Event (format %s)", accept); + transmit_response(p, mybuf, req); + + ast_log(LOG_WARNING,"SUBSCRIBE failure: unrecognized format: '%s' pvt: subscribed: %d, stateid: %d, laststate: %d, dialogver: %d, subscribecont: '%s', subscribeuri: '%s'\n", + accept, (int)p->subscribed, p->stateid, p->laststate, p->dialogver, p->subscribecontext, p->subscribeuri); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + return 0; + } + } else if (!strcmp(event, "message-summary")) { + if (!ast_strlen_zero(accept) && strcmp(accept, "application/simple-message-summary")) { + /* Format requested that we do not support */ + transmit_response(p, "406 Not Acceptable", req); + if (option_debug > 1) + ast_log(LOG_DEBUG, "Received SIP mailbox subscription for unknown format: %s\n", accept); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + if (authpeer) /* No need for authpeer here */ + ASTOBJ_UNREF(authpeer, sip_destroy_peer); + return 0; + } + /* Looks like they actually want a mailbox status + This version of Asterisk supports mailbox subscriptions + The subscribed URI needs to exist in the dial plan + In most devices, this is configurable to the voicemailmain extension you use + */ + if (!authpeer || ast_strlen_zero(authpeer->mailbox)) { + transmit_response(p, "404 Not found (no mailbox)", req); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", authpeer->name); + if (authpeer) /* No need for authpeer here */ + ASTOBJ_UNREF(authpeer, sip_destroy_peer); + return 0; + } + + p->subscribed = MWI_NOTIFICATION; + if (authpeer->mwipvt && authpeer->mwipvt != p) /* Destroy old PVT if this is a new one */ + /* We only allow one subscription per peer */ + sip_destroy(authpeer->mwipvt); + authpeer->mwipvt = p; /* Link from peer to pvt */ + p->relatedpeer = ASTOBJ_REF(authpeer); /* Link from pvt to peer */ + } else { /* At this point, Asterisk does not understand the specified event */ + transmit_response(p, "489 Bad Event", req); + if (option_debug > 1) + ast_log(LOG_DEBUG, "Received SIP subscribe for unknown event package: %s\n", event); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + if (authpeer) /* No need for authpeer here */ + ASTOBJ_UNREF(authpeer, sip_destroy_peer); + return 0; + } + + if (p->subscribed != MWI_NOTIFICATION && !resubscribe) { + if (p->stateid > -1) + ast_extension_state_del(p->stateid, cb_extensionstate); + p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, p); + } + + if (!ast_test_flag(req, SIP_PKT_IGNORE) && p) + p->lastinvite = seqno; + if (p && !ast_test_flag(&p->flags[0], SIP_NEEDDESTROY)) { + p->expiry = atoi(get_header(req, "Expires")); + + /* check if the requested expiry-time is within the approved limits from sip.conf */ + if (p->expiry > max_expiry) + p->expiry = max_expiry; + if (p->expiry < min_expiry && p->expiry > 0) + p->expiry = min_expiry; + + if (sipdebug || option_debug > 1) { + if (p->subscribed == MWI_NOTIFICATION && p->relatedpeer) + ast_log(LOG_DEBUG, "Adding subscription for mailbox notification - peer %s Mailbox %s\n", p->relatedpeer->name, p->relatedpeer->mailbox); + else + ast_log(LOG_DEBUG, "Adding subscription for extension %s context %s for peer %s\n", p->exten, p->context, p->username); + } + if (p->autokillid > -1 && sip_cancel_destroy(p)) /* Remove subscription expiry for renewals */ + ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); + if (p->expiry > 0) + sip_scheddestroy(p, (p->expiry + 10) * 1000); /* Set timer for destruction of call at expiration */ + + if (p->subscribed == MWI_NOTIFICATION) { + ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + transmit_response(p, "200 OK", req); + if (p->relatedpeer) { /* Send first notification */ + ASTOBJ_WRLOCK(p->relatedpeer); + sip_send_mwi_to_peer(p->relatedpeer); + ASTOBJ_UNLOCK(p->relatedpeer); + } + } else { + struct sip_pvt *p_old; + + if ((firststate = ast_extension_state(NULL, p->context, p->exten)) < 0) { + + ast_log(LOG_NOTICE, "Got SUBSCRIBE for extension %s@%s from %s, but there is no hint for that extension.\n", p->exten, p->context, ast_inet_ntoa(p->sa.sin_addr)); + transmit_response(p, "404 Not found", req); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + return 0; + } + ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); + transmit_response(p, "200 OK", req); + transmit_state_notify(p, firststate, 1, FALSE); /* Send first notification */ + append_history(p, "Subscribestatus", "%s", ast_extension_state2str(firststate)); + /* hide the 'complete' exten/context in the refer_to field for later display */ + ast_string_field_build(p, subscribeuri, "%s@%s", p->exten, p->context); + + /* remove any old subscription from this peer for the same exten/context, + as the peer has obviously forgotten about it and it's wasteful to wait + for it to expire and send NOTIFY messages to the peer only to have them + ignored (or generate errors) + */ + ast_mutex_lock(&iflock); + for (p_old = iflist; p_old; p_old = p_old->next) { + if (p_old == p) + continue; + if (p_old->initreq.method != SIP_SUBSCRIBE) + continue; + if (p_old->subscribed == NONE) + continue; + ast_mutex_lock(&p_old->lock); + if (!strcmp(p_old->username, p->username)) { + if (!strcmp(p_old->exten, p->exten) && + !strcmp(p_old->context, p->context)) { + ast_set_flag(&p_old->flags[0], SIP_NEEDDESTROY); + ast_mutex_unlock(&p_old->lock); + break; + } + } + ast_mutex_unlock(&p_old->lock); + } + ast_mutex_unlock(&iflock); + } + if (!p->expiry) + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } + return 1; +} + +/*! \brief Handle incoming REGISTER request */ +static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e) +{ + enum check_auth_result res; + + /* Use this as the basis */ + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_verbose("Using latest REGISTER request as basis request\n"); + copy_request(&p->initreq, req); + check_via(p, req); + if ((res = register_verify(p, sin, req, e)) < 0) { + const char *reason; + + switch (res) { + case AUTH_SECRET_FAILED: + reason = "Wrong password"; + break; + case AUTH_USERNAME_MISMATCH: + reason = "Username/auth name mismatch"; + break; + case AUTH_NOT_FOUND: + reason = "No matching peer found"; + break; + case AUTH_UNKNOWN_DOMAIN: + reason = "Not a local domain"; + break; + case AUTH_PEER_NOT_DYNAMIC: + reason = "Peer is not supposed to register"; + break; + case AUTH_ACL_FAILED: + reason = "Device does not match ACL"; + break; + default: + reason = "Unknown failure"; + break; + } + ast_log(LOG_NOTICE, "Registration from '%s' failed for '%s' - %s\n", + get_header(req, "To"), ast_inet_ntoa(sin->sin_addr), + reason); + append_history(p, "RegRequest", "Failed : Account %s : %s", get_header(req, "To"), reason); + } else + append_history(p, "RegRequest", "Succeeded : Account %s", get_header(req, "To")); + + if (res < 1) { + /* Destroy the session, but keep us around for just a bit in case they don't + get our 200 OK */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } + return res; +} + +/*! \brief Handle incoming SIP requests (methods) +\note This is where all incoming requests go first */ +/* called with p and p->owner locked */ +static int handle_request(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock) +{ + /* Called with p->lock held, as well as p->owner->lock if appropriate, keeping things + relatively static */ + const char *cmd; + const char *cseq; + const char *useragent; + int seqno; + int len; + int ignore = FALSE; + int respid; + int res = 0; + int debug = sip_debug_test_pvt(p); + char *e; + int error = 0; + + /* Get Method and Cseq */ + cseq = get_header(req, "Cseq"); + cmd = req->header[0]; + + /* Must have Cseq */ + if (ast_strlen_zero(cmd) || ast_strlen_zero(cseq)) { + ast_log(LOG_ERROR, "Missing Cseq. Dropping this SIP message, it's incomplete.\n"); + error = 1; + } + if (!error && sscanf(cseq, "%d%n", &seqno, &len) != 1) { + ast_log(LOG_ERROR, "No seqno in '%s'. Dropping incomplete message.\n", cmd); + error = 1; + } + if (error) { + if (!p->initreq.headers) /* New call */ + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); /* Make sure we destroy this dialog */ + return -1; + } + /* Get the command XXX */ + + cmd = req->rlPart1; + e = req->rlPart2; + + /* Save useragent of the client */ + useragent = get_header(req, "User-Agent"); + if (!ast_strlen_zero(useragent)) + ast_string_field_set(p, useragent, useragent); + + /* Find out SIP method for incoming request */ + if (req->method == SIP_RESPONSE) { /* Response to our request */ + /* Response to our request -- Do some sanity checks */ + if (!p->initreq.headers) { + if (option_debug) + ast_log(LOG_DEBUG, "That's odd... Got a response on a call we dont know about. Cseq %d Cmd %s\n", seqno, cmd); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + return 0; + } else if (p->ocseq && (p->ocseq < seqno) && (seqno != p->lastnoninvite)) { + if (option_debug) + ast_log(LOG_DEBUG, "Ignoring out of order response %d (expecting %d)\n", seqno, p->ocseq); + return -1; + } else if (p->ocseq && (p->ocseq != seqno) && (seqno != p->lastnoninvite)) { + /* ignore means "don't do anything with it" but still have to + respond appropriately */ + ignore = TRUE; + ast_set_flag(req, SIP_PKT_IGNORE); + ast_set_flag(req, SIP_PKT_IGNORE_RESP); + append_history(p, "Ignore", "Ignoring this retransmit\n"); + } else if (e) { + e = ast_skip_blanks(e); + if (sscanf(e, "%d %n", &respid, &len) != 1) { + ast_log(LOG_WARNING, "Invalid response: '%s'\n", e); + } else { + if (respid <= 0) { + ast_log(LOG_WARNING, "Invalid SIP response code: '%d'\n", respid); + return 0; + } + /* More SIP ridiculousness, we have to ignore bogus contacts in 100 etc responses */ + if ((respid == 200) || ((respid >= 300) && (respid <= 399))) + extract_uri(p, req); + handle_response(p, respid, e + len, req, ignore, seqno); + } + } + return 0; + } + + /* New SIP request coming in + (could be new request in existing SIP dialog as well...) + */ + + p->method = req->method; /* Find out which SIP method they are using */ + if (option_debug > 3) + ast_log(LOG_DEBUG, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd); + + if (p->icseq && (p->icseq > seqno) ) { + if (p->pendinginvite && seqno == p->pendinginvite && (req->method == SIP_ACK || req->method == SIP_CANCEL)) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Got CANCEL or ACK on INVITE with transactions in between.\n"); + } else { + if (option_debug) + ast_log(LOG_DEBUG, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq); + if (req->method != SIP_ACK) + transmit_response(p, "503 Server error", req); /* We must respond according to RFC 3261 sec 12.2 */ + return -1; + } + } else if (p->icseq && + p->icseq == seqno && + req->method != SIP_ACK && + (p->method != SIP_CANCEL || ast_test_flag(&p->flags[0], SIP_ALREADYGONE))) { + /* ignore means "don't do anything with it" but still have to + respond appropriately. We do this if we receive a repeat of + the last sequence number */ + ignore = 2; + ast_set_flag(req, SIP_PKT_IGNORE); + ast_set_flag(req, SIP_PKT_IGNORE_REQ); + if (option_debug > 2) + ast_log(LOG_DEBUG, "Ignoring SIP message because of retransmit (%s Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno); + } + + if (seqno >= p->icseq) + /* Next should follow monotonically (but not necessarily + incrementally -- thanks again to the genius authors of SIP -- + increasing */ + p->icseq = seqno; + + /* Find their tag if we haven't got it */ + if (ast_strlen_zero(p->theirtag)) { + char tag[128]; + + gettag(req, "From", tag, sizeof(tag)); + ast_string_field_set(p, theirtag, tag); + } + snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd); + + if (pedanticsipchecking) { + /* If this is a request packet without a from tag, it's not + correct according to RFC 3261 */ + /* Check if this a new request in a new dialog with a totag already attached to it, + RFC 3261 - section 12.2 - and we don't want to mess with recovery */ + if (!p->initreq.headers && ast_test_flag(req, SIP_PKT_WITH_TOTAG)) { + /* If this is a first request and it got a to-tag, it is not for us */ + if (!ast_test_flag(req, SIP_PKT_IGNORE) && req->method == SIP_INVITE) { + transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req); + /* Will cease to exist after ACK */ + } else if (req->method != SIP_ACK) { + transmit_response(p, "481 Call/Transaction Does Not Exist", req); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } + return res; + } + } + + if (!e && (p->method == SIP_INVITE || p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER || p->method == SIP_NOTIFY)) { + transmit_response(p, "400 Bad request", req); + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + return -1; + } + + /* Handle various incoming SIP methods in requests */ + switch (p->method) { + case SIP_OPTIONS: + res = handle_request_options(p, req); + break; + case SIP_INVITE: + res = handle_request_invite(p, req, debug, seqno, sin, recount, e, nounlock); + break; + case SIP_REFER: + res = handle_request_refer(p, req, debug, ignore, seqno, nounlock); + break; + case SIP_CANCEL: + res = handle_request_cancel(p, req); + break; + case SIP_BYE: + res = handle_request_bye(p, req); + break; + case SIP_MESSAGE: + res = handle_request_message(p, req); + break; + case SIP_SUBSCRIBE: + res = handle_request_subscribe(p, req, sin, seqno, e); + break; + case SIP_REGISTER: + res = handle_request_register(p, req, sin, e); + break; + case SIP_INFO: + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_verbose("Receiving INFO!\n"); + if (!ignore) + handle_request_info(p, req); + else /* if ignoring, transmit response */ + transmit_response(p, "200 OK", req); + break; + case SIP_NOTIFY: + res = handle_request_notify(p, req, sin, seqno, e); + break; + case SIP_ACK: + /* Make sure we don't ignore this */ + if (seqno == p->pendinginvite) { + p->invitestate = INV_TERMINATED; + p->pendinginvite = 0; + __sip_ack(p, seqno, FLAG_RESPONSE, 0); + if (find_sdp(req)) { + if (process_sdp(p, req)) + return -1; + } + check_pendings(p); + } + /* Got an ACK that we did not match. Ignore silently */ + if (!p->lastinvite && ast_strlen_zero(p->randdata)) + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + break; + default: + transmit_response_with_allow(p, "501 Method Not Implemented", req, 0); + ast_log(LOG_NOTICE, "Unknown SIP command '%s' from '%s'\n", + cmd, ast_inet_ntoa(p->sa.sin_addr)); + /* If this is some new method, and we don't have a call, destroy it now */ + if (!p->initreq.headers) + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + break; + } + return res; +} + +static void process_request_queue(struct sip_pvt *p, int *recount, int *nounlock) +{ + struct sip_request *req; + + while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) { + if (handle_request(p, req, &p->recv, recount, nounlock) == -1) { + /* Request failed */ + if (option_debug) { + ast_log(LOG_DEBUG, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>"); + } + } + ast_free(req); + } +} + +static int scheduler_process_request_queue(const void *data) +{ + struct sip_pvt *p = (struct sip_pvt *) data; + int recount = 0; + int nounlock = 0; + int lockretry; + + for (lockretry = 10; lockretry > 0; lockretry--) { + ast_mutex_lock(&p->lock); + + /* lock the owner if it has one -- we may need it */ + /* because this is deadlock-prone, we need to try and unlock if failed */ + if (!p->owner || !ast_channel_trylock(p->owner)) { + break; /* locking succeeded */ + } + + if (lockretry != 1) { + ast_mutex_unlock(&p->lock); + /* Sleep for a very short amount of time */ + usleep(1); + } + } + + if (!lockretry) { + int retry = !AST_LIST_EMPTY(&p->request_queue); + + /* we couldn't get the owner lock, which is needed to process + the queued requests, so return a non-zero value, which will + cause the scheduler to run this request again later if there + still requests to be processed + */ + ast_mutex_unlock(&p->lock); + return retry; + }; + + process_request_queue(p, &recount, &nounlock); + p->request_queue_sched_id = -1; + + if (p->owner && !nounlock) { + ast_channel_unlock(p->owner); + } + ast_mutex_unlock(&p->lock); + + if (recount) { + ast_update_use_count(); + } + + return 0; +} + +static int queue_request(struct sip_pvt *p, const struct sip_request *req) +{ + struct sip_request *newreq; + + if (!(newreq = ast_calloc(1, sizeof(*newreq)))) { + return -1; + } + + copy_request(newreq, req); + AST_LIST_INSERT_TAIL(&p->request_queue, newreq, next); + if (p->request_queue_sched_id == -1) { + p->request_queue_sched_id = ast_sched_add(sched, 10, scheduler_process_request_queue, p); + } + + return 0; +} + +/*! \brief Read data from SIP socket +\note sipsock_read locks the owner channel while we are processing the SIP message +\return 1 on error, 0 on success +\note Successful messages is connected to SIP call and forwarded to handle_request() +*/ +static int sipsock_read(int *id, int fd, short events, void *ignore) +{ + struct sip_request req; + struct sockaddr_in sin = { 0, }; + struct sip_pvt *p; + int res; + socklen_t len = sizeof(sin); + int nounlock = 0; + int recount = 0; + int lockretry; + + memset(&req, 0, sizeof(req)); + res = recvfrom(sipsock, req.data, sizeof(req.data) - 1, 0, (struct sockaddr *)&sin, &len); + if (res < 0) { +#if !defined(__FreeBSD__) + if (errno == EAGAIN) + ast_log(LOG_NOTICE, "SIP: Received packet with bad UDP checksum\n"); + else +#endif + if (errno != ECONNREFUSED) + ast_log(LOG_WARNING, "Recv error: %s\n", strerror(errno)); + return 1; + } + if (option_debug && res == sizeof(req.data) - 1) + ast_log(LOG_DEBUG, "Received packet exceeds buffer. Data is possibly lost\n"); + + req.data[res] = '\0'; + req.len = res; + if(sip_debug_test_addr(&sin)) /* Set the debug flag early on packet level */ + ast_set_flag(&req, SIP_PKT_DEBUG); + if (pedanticsipchecking) + req.len = lws2sws(req.data, req.len); /* Fix multiline headers */ + if (ast_test_flag(&req, SIP_PKT_DEBUG)) + ast_verbose("\n<--- SIP read from %s:%d --->\n%s\n<------------->\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), req.data); + + if(parse_request(&req) == -1) /* Bad packet, can't parse */ + return 1; + + req.method = find_sip_method(req.rlPart1); + + if (ast_test_flag(&req, SIP_PKT_DEBUG)) + ast_verbose("--- (%d headers %d lines)%s ---\n", req.headers, req.lines, (req.headers + req.lines == 0) ? " Nat keepalive" : ""); + + if (req.headers < 2) /* Must have at least two headers */ + return 1; + + /* Process request, with netlock held, and with usual deadlock avoidance */ + for (lockretry = 10; lockretry > 0; lockretry--) { + ast_mutex_lock(&netlock); + + /* Find the active SIP dialog or create a new one */ + p = find_call(&req, &sin, req.method); /* returns p locked */ + if (p == NULL) { + if (option_debug) + ast_log(LOG_DEBUG, "Invalid SIP message - rejected , no callid, len %d\n", req.len); + ast_mutex_unlock(&netlock); + return 1; + } + /* Go ahead and lock the owner if it has one -- we may need it */ + /* because this is deadlock-prone, we need to try and unlock if failed */ + if (!p->owner || !ast_channel_trylock(p->owner)) + break; /* locking succeeded */ + if (lockretry != 1) { + ast_mutex_unlock(&p->lock); + ast_mutex_unlock(&netlock); + /* Sleep for a very short amount of time */ + usleep(1); + } + } + p->recv = sin; + + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a request or response, note what it was for */ + append_history(p, "Rx", "%s / %s / %s", req.data, get_header(&req, "CSeq"), req.rlPart2); + + if (!lockretry) { + if (!queue_request(p, &req)) { + /* the request has been queued for later handling */ + ast_mutex_unlock(&p->lock); + ast_mutex_unlock(&netlock); + return 1; + } + + /* This is unsafe, since p->owner is not locked. */ + if (p->owner) + ast_log(LOG_ERROR, "Channel lock for %s could not be obtained, and request was unable to be queued.\n", S_OR(p->owner->name, "- no channel name ??? - ")); + ast_log(LOG_ERROR, "SIP transaction failed: %s \n", p->callid); + if (req.method != SIP_ACK) + transmit_response(p, "503 Server error", &req); /* We must respond according to RFC 3261 sec 12.2 */ + /* XXX We could add retry-after to make sure they come back */ + append_history(p, "LockFail", "Owner lock failed, transaction failed."); + ast_mutex_unlock(&p->lock); + ast_mutex_unlock(&netlock); + return 1; + } + + /* if there are queued requests on this sip_pvt, process them first, so that everything is + handled in order + */ + if (!AST_LIST_EMPTY(&p->request_queue)) { + AST_SCHED_DEL(sched, p->request_queue_sched_id); + process_request_queue(p, &recount, &nounlock); + } + + if (handle_request(p, &req, &sin, &recount, &nounlock) == -1) { + /* Request failed */ + if (option_debug) + ast_log(LOG_DEBUG, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>"); + } + + if (p->owner && !nounlock) + ast_channel_unlock(p->owner); + ast_mutex_unlock(&p->lock); + ast_mutex_unlock(&netlock); + if (recount) + ast_update_use_count(); + + return 1; +} + +/*! \brief Send message waiting indication to alert peer that they've got voicemail */ +static int sip_send_mwi_to_peer(struct sip_peer *peer) +{ + /* Called with peerl lock, but releases it */ + struct sip_pvt *p; + int newmsgs, oldmsgs; + + /* Do we have an IP address? If not, skip this peer */ + if (!peer->addr.sin_addr.s_addr && !peer->defaddr.sin_addr.s_addr) + return 0; + + /* Check for messages */ + ast_app_inboxcount(peer->mailbox, &newmsgs, &oldmsgs); + + peer->lastmsgcheck = time(NULL); + + /* Return now if it's the same thing we told them last time */ + if (((newmsgs > 0x7fff ? 0x7fff0000 : (newmsgs << 16)) | (oldmsgs > 0xffff ? 0xffff : oldmsgs)) == peer->lastmsgssent) { + return 0; + } + + + peer->lastmsgssent = ((newmsgs > 0x7fff ? 0x7fff0000 : (newmsgs << 16)) | (oldmsgs > 0xffff ? 0xffff : oldmsgs)); + + if (peer->mwipvt) { + /* Base message on subscription */ + p = peer->mwipvt; + } else { + /* Build temporary dialog for this message */ + if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY))) + return -1; + if (create_addr_from_peer(p, peer)) { + /* Maybe they're not registered, etc. */ + sip_destroy(p); + return 0; + } + /* Recalculate our side, and recalculate Call ID */ + if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip)) + p->ourip = __ourip; + build_via(p); + build_callid_pvt(p); + /* Destroy this session after 32 secs */ + sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); + } + /* Send MWI */ + ast_set_flag(&p->flags[0], SIP_OUTGOING); + transmit_notify_with_mwi(p, newmsgs, oldmsgs, peer->vmexten); + return 0; +} + +/*! \brief Check whether peer needs a new MWI notification check */ +static int does_peer_need_mwi(struct sip_peer *peer) +{ + time_t t = time(NULL); + + if (ast_test_flag(&peer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY) && + !peer->mwipvt) { /* We don't have a subscription */ + peer->lastmsgcheck = t; /* Reset timer */ + return FALSE; + } + + if (!ast_strlen_zero(peer->mailbox) && (t - peer->lastmsgcheck) > global_mwitime) + return TRUE; + + return FALSE; +} + + +/*! \brief The SIP monitoring thread +\note This thread monitors all the SIP sessions and peers that needs notification of mwi + (and thus do not have a separate thread) indefinitely +*/ +static void *do_monitor(void *data) +{ + int res; + struct sip_pvt *sip; + struct sip_peer *peer = NULL; + time_t t; + int fastrestart = FALSE; + int lastpeernum = -1; + int curpeernum; + int reloading; + + /* Add an I/O event to our SIP UDP socket */ + if (sipsock > -1) + sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL); + + /* From here on out, we die whenever asked */ + for(;;) { + /* Check for a reload request */ + ast_mutex_lock(&sip_reload_lock); + reloading = sip_reloading; + sip_reloading = FALSE; + ast_mutex_unlock(&sip_reload_lock); + if (reloading) { + if (option_verbose > 0) + ast_verbose(VERBOSE_PREFIX_1 "Reloading SIP\n"); + sip_do_reload(sip_reloadreason); + + /* Change the I/O fd of our UDP socket */ + if (sipsock > -1) { + if (sipsock_read_id) + sipsock_read_id = ast_io_change(io, sipsock_read_id, sipsock, NULL, 0, NULL); + else + sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL); + } else if (sipsock_read_id) { + ast_io_remove(io, sipsock_read_id); + sipsock_read_id = NULL; + } + } +restartsearch: + /* Check for interfaces needing to be killed */ + ast_mutex_lock(&iflock); + t = time(NULL); + /* don't scan the interface list if it hasn't been a reasonable period + of time since the last time we did it (when MWI is being sent, we can + get back to this point every millisecond or less) + */ + for (sip = iflist; !fastrestart && sip; sip = sip->next) { + /*! \note If we can't get a lock on an interface, skip it and come + * back later. Note that there is the possibility of a deadlock with + * sip_hangup otherwise, because sip_hangup is called with the channel + * locked first, and the iface lock is attempted second. + */ + if (ast_mutex_trylock(&sip->lock)) + continue; + + /* Check RTP timeouts and kill calls if we have a timeout set and do not get RTP */ + if (sip->rtp && sip->owner && + (sip->owner->_state == AST_STATE_UP) && + !sip->redirip.sin_addr.s_addr && + sip->t38.state != T38_ENABLED) { + if (sip->lastrtptx && + ast_rtp_get_rtpkeepalive(sip->rtp) && + (t > sip->lastrtptx + ast_rtp_get_rtpkeepalive(sip->rtp))) { + /* Need to send an empty RTP packet */ + sip->lastrtptx = time(NULL); + ast_rtp_sendcng(sip->rtp, 0); + } + if (sip->lastrtprx && + (ast_rtp_get_rtptimeout(sip->rtp) || ast_rtp_get_rtpholdtimeout(sip->rtp)) && + (t > sip->lastrtprx + ast_rtp_get_rtptimeout(sip->rtp))) { + /* Might be a timeout now -- see if we're on hold */ + struct sockaddr_in sin; + ast_rtp_get_peer(sip->rtp, &sin); + if (sin.sin_addr.s_addr || + (ast_rtp_get_rtpholdtimeout(sip->rtp) && + (t > sip->lastrtprx + ast_rtp_get_rtpholdtimeout(sip->rtp)))) { + /* Needs a hangup */ + if (ast_rtp_get_rtptimeout(sip->rtp)) { + while (sip->owner && ast_channel_trylock(sip->owner)) { + DEADLOCK_AVOIDANCE(&sip->lock); + } + if (sip->owner) { + ast_log(LOG_NOTICE, + "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n", + sip->owner->name, + (long) (t - sip->lastrtprx)); + /* Issue a softhangup */ + ast_softhangup_nolock(sip->owner, AST_SOFTHANGUP_DEV); + ast_channel_unlock(sip->owner); + /* forget the timeouts for this call, since a hangup + has already been requested and we don't want to + repeatedly request hangups + */ + ast_rtp_set_rtptimeout(sip->rtp, 0); + ast_rtp_set_rtpholdtimeout(sip->rtp, 0); + if (sip->vrtp) { + ast_rtp_set_rtptimeout(sip->vrtp, 0); + ast_rtp_set_rtpholdtimeout(sip->vrtp, 0); + } + } + } + } + } + } + /* If we have sessions that needs to be destroyed, do it now */ + if (ast_test_flag(&sip->flags[0], SIP_NEEDDESTROY) && !sip->packets && + !sip->owner) { + ast_mutex_unlock(&sip->lock); + __sip_destroy(sip, 1); + ast_mutex_unlock(&iflock); + usleep(1); + goto restartsearch; + } + ast_mutex_unlock(&sip->lock); + } + ast_mutex_unlock(&iflock); + + /* XXX TODO The scheduler usage in this module does not have sufficient + * synchronization being done between running the scheduler and places + * scheduling tasks. As it is written, any scheduled item may not run + * any sooner than about 1 second, regardless of whether a sooner time + * was asked for. */ + + pthread_testcancel(); + /* Wait for sched or io */ + res = ast_sched_wait(sched); + if ((res < 0) || (res > 1000)) + res = 1000; + /* If we might need to send more mailboxes, don't wait long at all.*/ + if (fastrestart) + res = 1; + res = ast_io_wait(io, res); + if (option_debug && res > 20) + ast_log(LOG_DEBUG, "chan_sip: ast_io_wait ran %d all at once\n", res); + ast_mutex_lock(&monlock); + res = ast_sched_runq(sched); + if (option_debug && res >= 20) + ast_log(LOG_DEBUG, "chan_sip: ast_sched_runq ran %d all at once\n", res); + + /* Send MWI notifications to peers - static and cached realtime peers */ + t = time(NULL); + fastrestart = FALSE; + curpeernum = 0; + peer = NULL; + /* Find next peer that needs mwi */ + ASTOBJ_CONTAINER_TRAVERSE(&peerl, !peer, do { + if ((curpeernum > lastpeernum) && does_peer_need_mwi(iterator)) { + fastrestart = TRUE; + lastpeernum = curpeernum; + peer = ASTOBJ_REF(iterator); + }; + curpeernum++; + } while (0) + ); + /* Send MWI to the peer */ + if (peer) { + ASTOBJ_WRLOCK(peer); + sip_send_mwi_to_peer(peer); + ASTOBJ_UNLOCK(peer); + ASTOBJ_UNREF(peer,sip_destroy_peer); + } else { + /* Reset where we come from */ + lastpeernum = -1; + } + ast_mutex_unlock(&monlock); + } + /* Never reached */ + return NULL; + +} + +/*! \brief Start the channel monitor thread */ +static int restart_monitor(void) +{ + /* If we're supposed to be stopped -- stay stopped */ + if (monitor_thread == AST_PTHREADT_STOP) + return 0; + ast_mutex_lock(&monlock); + if (monitor_thread == pthread_self()) { + ast_mutex_unlock(&monlock); + ast_log(LOG_WARNING, "Cannot kill myself\n"); + return -1; + } + if (monitor_thread != AST_PTHREADT_NULL) { + /* Wake up the thread */ + pthread_kill(monitor_thread, SIGURG); + } else { + /* Start a new monitor */ + if (ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL) < 0) { + ast_mutex_unlock(&monlock); + ast_log(LOG_ERROR, "Unable to start monitor thread.\n"); + return -1; + } + } + ast_mutex_unlock(&monlock); + return 0; +} + +/*! \brief React to lack of answer to Qualify poke */ +static int sip_poke_noanswer(const void *data) +{ + struct sip_peer *peer = (struct sip_peer *)data; + + peer->pokeexpire = -1; + if (peer->lastms > -1) { + ast_log(LOG_NOTICE, "Peer '%s' is now UNREACHABLE! Last qualify: %d\n", peer->name, peer->lastms); + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unreachable\r\nTime: %d\r\n", peer->name, -1); + } + if (peer->call) + sip_destroy(peer->call); + peer->call = NULL; + peer->lastms = -1; + ast_device_state_changed("SIP/%s", peer->name); + + /* This function gets called one place outside of the scheduler ... */ + if (!AST_SCHED_DEL(sched, peer->pokeexpire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + + /* There is no need to ASTOBJ_REF() here. Just let the scheduled callback + * inherit the reference that the current callback already has. */ + peer->pokeexpire = ast_sched_add(sched, DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer); + if (peer->pokeexpire == -1) { + ASTOBJ_UNREF(peer, sip_destroy_peer); + } + + return 0; +} + +/*! \brief Check availability of peer, also keep NAT open +\note This is done with the interval in qualify= configuration option + Default is 2 seconds */ +static int sip_poke_peer(struct sip_peer *peer) +{ + struct sip_pvt *p; + int xmitres = 0; + + if (!peer->maxms || !peer->addr.sin_addr.s_addr) { + /* IF we have no IP, or this isn't to be monitored, return + imeediately after clearing things out */ + if (!AST_SCHED_DEL(sched, peer->pokeexpire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + peer->lastms = 0; + peer->call = NULL; + return 0; + } + if (peer->call) { + if (sipdebug) + ast_log(LOG_NOTICE, "Still have a QUALIFY dialog active, deleting\n"); + sip_destroy(peer->call); + } + if (!(p = peer->call = sip_alloc(NULL, NULL, 0, SIP_OPTIONS))) + return -1; + + p->sa = peer->addr; + p->recv = peer->addr; + ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY); + ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY); + + /* Send OPTIONs to peer's fullcontact */ + if (!ast_strlen_zero(peer->fullcontact)) + ast_string_field_set(p, fullcontact, peer->fullcontact); + + if (!ast_strlen_zero(peer->tohost)) + ast_string_field_set(p, tohost, peer->tohost); + else + ast_string_field_set(p, tohost, ast_inet_ntoa(peer->addr.sin_addr)); + + /* Recalculate our side, and recalculate Call ID */ + if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip)) + p->ourip = __ourip; + build_via(p); + build_callid_pvt(p); + + if (!AST_SCHED_DEL(sched, peer->pokeexpire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + + p->relatedpeer = ASTOBJ_REF(peer); + ast_set_flag(&p->flags[0], SIP_OUTGOING); +#ifdef VOCAL_DATA_HACK + ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username)); + xmitres = transmit_invite(p, SIP_INVITE, 0, 2); +#else + xmitres = transmit_invite(p, SIP_OPTIONS, 0, 2); +#endif + gettimeofday(&peer->ps, NULL); + if (xmitres == XMIT_ERROR) { + sip_poke_noanswer(ASTOBJ_REF(peer)); /* Immediately unreachable, network problems */ + } else { + if (!AST_SCHED_DEL(sched, peer->pokeexpire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + peer->pokeexpire = ast_sched_add(sched, peer->maxms * 2, sip_poke_noanswer, ASTOBJ_REF(peer)); + if (peer->pokeexpire == -1) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + } + + return 0; +} + +/*! \brief Part of PBX channel interface +\note +\par Return values:--- + + If we have qualify on and the device is not reachable, regardless of registration + state we return AST_DEVICE_UNAVAILABLE + + For peers with call limit: + - not registered AST_DEVICE_UNAVAILABLE + - registered, no call AST_DEVICE_NOT_INUSE + - registered, active calls AST_DEVICE_INUSE + - registered, call limit reached AST_DEVICE_BUSY + - registered, onhold AST_DEVICE_ONHOLD + - registered, ringing AST_DEVICE_RINGING + + For peers without call limit: + - not registered AST_DEVICE_UNAVAILABLE + - registered AST_DEVICE_NOT_INUSE + - fixed IP (!dynamic) AST_DEVICE_NOT_INUSE + + Peers that does not have a known call and can't be reached by OPTIONS + - unreachable AST_DEVICE_UNAVAILABLE + + If we return AST_DEVICE_UNKNOWN, the device state engine will try to find + out a state by walking the channel list. + + The queue system (\ref app_queue.c) treats a member as "active" + if devicestate is != AST_DEVICE_UNAVAILBALE && != AST_DEVICE_INVALID + + When placing a call to the queue member, queue system sets a member to busy if + != AST_DEVICE_NOT_INUSE and != AST_DEVICE_UNKNOWN + +*/ +static int sip_devicestate(void *data) +{ + char *host; + char *tmp; + + struct hostent *hp; + struct ast_hostent ahp; + struct sip_peer *p; + + int res = AST_DEVICE_INVALID; + + /* make sure data is not null. Maybe unnecessary, but better be safe */ + host = ast_strdupa(data ? data : ""); + if ((tmp = strchr(host, '@'))) + host = tmp + 1; + + if (option_debug > 2) + ast_log(LOG_DEBUG, "Checking device state for peer %s\n", host); + + /* If find_peer asks for a realtime peer, then this breaks rtautoclear. This + * is because when a peer tries to autoexpire, the last thing it does is to + * queue up an event telling the system that the devicestate has changed + * (presumably to unavailable). If we ask for a realtime peer here, this would + * load it BACK into memory, thus defeating the point of trying to trying to + * clear dead hosts out of memory. + */ + if ((p = find_peer(host, NULL, 0, 1))) { + if (p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) { + /* we have an address for the peer */ + + /* Check status in this order + - Hold + - Ringing + - Busy (enforced only by call limit) + - Inuse (we have a call) + - Unreachable (qualify) + If we don't find any of these state, report AST_DEVICE_NOT_INUSE + for registered devices */ + + if (p->onHold) + /* First check for hold or ring states */ + res = AST_DEVICE_ONHOLD; + else if (p->inRinging) { + if (p->inRinging == p->inUse) + res = AST_DEVICE_RINGING; + else + res = AST_DEVICE_RINGINUSE; + } else if (p->call_limit && (p->inUse == p->call_limit)) + /* check call limit */ + res = AST_DEVICE_BUSY; + else if (p->call_limit && p->inUse) + /* Not busy, but we do have a call */ + res = AST_DEVICE_INUSE; + else if (p->maxms && ((p->lastms > p->maxms) || (p->lastms < 0))) + /* We don't have a call. Are we reachable at all? Requires qualify= */ + res = AST_DEVICE_UNAVAILABLE; + else /* Default reply if we're registered and have no other data */ + res = AST_DEVICE_NOT_INUSE; + } else { + /* there is no address, it's unavailable */ + res = AST_DEVICE_UNAVAILABLE; + } + ASTOBJ_UNREF(p,sip_destroy_peer); + } else { + char *port = strchr(host, ':'); + if (port) + *port = '\0'; + hp = ast_gethostbyname(host, &ahp); + if (hp) + res = AST_DEVICE_UNKNOWN; + } + + return res; +} + +/*! \brief PBX interface function -build SIP pvt structure + SIP calls initiated by the PBX arrive here */ +static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause) +{ + int oldformat; + struct sip_pvt *p; + struct ast_channel *tmpc = NULL; + char *ext, *host; + char tmp[256]; + char *dest = data; + + oldformat = format; + if (!(format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1))) { + ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", ast_getformatname(oldformat), ast_getformatname(global_capability)); + *cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find codec to connect to host */ + return NULL; + } + if (option_debug) + ast_log(LOG_DEBUG, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat)); + + if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) { + ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", (char *)data); + *cause = AST_CAUSE_SWITCH_CONGESTION; + return NULL; + } + + ast_set_flag(&p->flags[1], SIP_PAGE2_OUTGOING_CALL); + + if (!(p->options = ast_calloc(1, sizeof(*p->options)))) { + sip_destroy(p); + ast_log(LOG_ERROR, "Unable to build option SIP data structure - Out of memory\n"); + *cause = AST_CAUSE_SWITCH_CONGESTION; + return NULL; + } + + ast_copy_string(tmp, dest, sizeof(tmp)); + host = strchr(tmp, '@'); + if (host) { + *host++ = '\0'; + ext = tmp; + } else { + ext = strchr(tmp, '/'); + if (ext) + *ext++ = '\0'; + host = tmp; + } + + if (create_addr(p, host)) { + *cause = AST_CAUSE_UNREGISTERED; + if (option_debug > 2) + ast_log(LOG_DEBUG, "Cant create SIP call - target device not registred\n"); + sip_destroy(p); + return NULL; + } + if (ast_strlen_zero(p->peername) && ext) + ast_string_field_set(p, peername, ext); + /* Recalculate our side, and recalculate Call ID */ + if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip)) + p->ourip = __ourip; + build_via(p); + build_callid_pvt(p); + + /* We have an extension to call, don't use the full contact here */ + /* This to enable dialing registered peers with extension dialling, + like SIP/peername/extension + SIP/peername will still use the full contact */ + if (ext) { + ast_string_field_set(p, username, ext); + ast_string_field_free(p, fullcontact); + } +#if 0 + printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host); +#endif + p->prefcodec = oldformat; /* Format for this call */ + ast_mutex_lock(&p->lock); + tmpc = sip_new(p, AST_STATE_DOWN, host); /* Place the call */ + ast_mutex_unlock(&p->lock); + if (!tmpc) + sip_destroy(p); + ast_update_use_count(); + restart_monitor(); + return tmpc; +} + +/*! + * \brief Parse the "insecure" setting from sip.conf or from realtime. + * \param flags a pointer to an ast_flags structure + * \param value the value of the SIP insecure setting + * \param lineno linenumber in sip.conf or -1 for realtime + */ +static void set_insecure_flags(struct ast_flags *flags, const char *value, int lineno) +{ + static int dep_insecure_very = 0; + static int dep_insecure_yes = 0; + + if (ast_strlen_zero(value)) + return; + + if (!strcasecmp(value, "very")) { + ast_set_flag(flags, SIP_INSECURE_PORT | SIP_INSECURE_INVITE); + if(!dep_insecure_very) { + if(lineno != -1) + ast_log(LOG_WARNING, "insecure=very at line %d is deprecated; use insecure=port,invite instead\n", lineno); + else + ast_log(LOG_WARNING, "insecure=very is deprecated; use insecure=port,invite instead\n"); + dep_insecure_very = 1; + } + } + else if (ast_true(value)) { + ast_set_flag(flags, SIP_INSECURE_PORT); + if(!dep_insecure_yes) { + if(lineno != -1) + ast_log(LOG_WARNING, "insecure=%s at line %d is deprecated; use insecure=port instead\n", value, lineno); + else + ast_log(LOG_WARNING, "insecure=%s is deprecated; use insecure=port instead\n", value); + dep_insecure_yes = 1; + } + } + else if (!ast_false(value)) { + char buf[64]; + char *word, *next; + ast_copy_string(buf, value, sizeof(buf)); + next = buf; + while ((word = strsep(&next, ","))) { + if (!strcasecmp(word, "port")) + ast_set_flag(flags, SIP_INSECURE_PORT); + else if (!strcasecmp(word, "invite")) + ast_set_flag(flags, SIP_INSECURE_INVITE); + else + ast_log(LOG_WARNING, "Unknown insecure mode '%s' on line %d\n", value, lineno); + } + } +} + +/*! + \brief Handle flag-type options common to configuration of devices - users and peers + \param flags array of two struct ast_flags + \param mask array of two struct ast_flags + \param v linked list of config variables to process + \returns non-zero if any config options were handled, zero otherwise +*/ +static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v) +{ + int res = 1; + + if (!strcasecmp(v->name, "trustrpid")) { + ast_set_flag(&mask[0], SIP_TRUSTRPID); + ast_set2_flag(&flags[0], ast_true(v->value), SIP_TRUSTRPID); + } else if (!strcasecmp(v->name, "sendrpid")) { + ast_set_flag(&mask[0], SIP_SENDRPID); + ast_set2_flag(&flags[0], ast_true(v->value), SIP_SENDRPID); + } else if (!strcasecmp(v->name, "g726nonstandard")) { + ast_set_flag(&mask[0], SIP_G726_NONSTANDARD); + ast_set2_flag(&flags[0], ast_true(v->value), SIP_G726_NONSTANDARD); + } else if (!strcasecmp(v->name, "useclientcode")) { + ast_set_flag(&mask[0], SIP_USECLIENTCODE); + ast_set2_flag(&flags[0], ast_true(v->value), SIP_USECLIENTCODE); + } else if (!strcasecmp(v->name, "dtmfmode")) { + ast_set_flag(&mask[0], SIP_DTMF); + ast_clear_flag(&flags[0], SIP_DTMF); + if (!strcasecmp(v->value, "inband")) + ast_set_flag(&flags[0], SIP_DTMF_INBAND); + else if (!strcasecmp(v->value, "rfc2833")) + ast_set_flag(&flags[0], SIP_DTMF_RFC2833); + else if (!strcasecmp(v->value, "info")) + ast_set_flag(&flags[0], SIP_DTMF_INFO); + else if (!strcasecmp(v->value, "auto")) + ast_set_flag(&flags[0], SIP_DTMF_AUTO); + else { + ast_log(LOG_WARNING, "Unknown dtmf mode '%s' on line %d, using rfc2833\n", v->value, v->lineno); + ast_set_flag(&flags[0], SIP_DTMF_RFC2833); + } + } else if (!strcasecmp(v->name, "nat")) { + ast_set_flag(&mask[0], SIP_NAT); + ast_clear_flag(&flags[0], SIP_NAT); + if (!strcasecmp(v->value, "never")) + ast_set_flag(&flags[0], SIP_NAT_NEVER); + else if (!strcasecmp(v->value, "route")) + ast_set_flag(&flags[0], SIP_NAT_ROUTE); + else if (ast_true(v->value)) + ast_set_flag(&flags[0], SIP_NAT_ALWAYS); + else + ast_set_flag(&flags[0], SIP_NAT_RFC3581); + } else if (!strcasecmp(v->name, "canreinvite")) { + ast_set_flag(&mask[0], SIP_REINVITE); + ast_clear_flag(&flags[0], SIP_REINVITE); + if(ast_true(v->value)) { + ast_set_flag(&flags[0], SIP_CAN_REINVITE | SIP_CAN_REINVITE_NAT); + } else if (!ast_false(v->value)) { + char buf[64]; + char *word, *next = buf; + + ast_copy_string(buf, v->value, sizeof(buf)); + while ((word = strsep(&next, ","))) { + if(!strcasecmp(word, "update")) { + ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_CAN_REINVITE); + } else if(!strcasecmp(word, "nonat")) { + ast_set_flag(&flags[0], SIP_CAN_REINVITE); + ast_clear_flag(&flags[0], SIP_CAN_REINVITE_NAT); + } else { + ast_log(LOG_WARNING, "Unknown canreinvite mode '%s' on line %d\n", v->value, v->lineno); + } + } + } + } else if (!strcasecmp(v->name, "insecure")) { + ast_set_flag(&mask[0], SIP_INSECURE_PORT | SIP_INSECURE_INVITE); + ast_clear_flag(&flags[0], SIP_INSECURE_PORT | SIP_INSECURE_INVITE); + set_insecure_flags(flags, v->value, v->lineno); + } else if (!strcasecmp(v->name, "progressinband")) { + ast_set_flag(&mask[0], SIP_PROG_INBAND); + ast_clear_flag(&flags[0], SIP_PROG_INBAND); + if (ast_true(v->value)) + ast_set_flag(&flags[0], SIP_PROG_INBAND_YES); + else if (strcasecmp(v->value, "never")) + ast_set_flag(&flags[0], SIP_PROG_INBAND_NO); + } else if (!strcasecmp(v->name, "promiscredir")) { + ast_set_flag(&mask[0], SIP_PROMISCREDIR); + ast_set2_flag(&flags[0], ast_true(v->value), SIP_PROMISCREDIR); + } else if (!strcasecmp(v->name, "videosupport")) { + ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT); + ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_VIDEOSUPPORT); + } else if (!strcasecmp(v->name, "allowoverlap")) { + ast_set_flag(&mask[1], SIP_PAGE2_ALLOWOVERLAP); + ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWOVERLAP); + } else if (!strcasecmp(v->name, "allowsubscribe")) { + ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE); + ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE); + } else if (!strcasecmp(v->name, "t38pt_udptl")) { + ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_UDPTL); + ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_UDPTL); +#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS + } else if (!strcasecmp(v->name, "t38pt_rtp")) { + ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_RTP); + ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_RTP); + } else if (!strcasecmp(v->name, "t38pt_tcp")) { + ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_TCP); + ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_TCP); +#endif + } else if (!strcasecmp(v->name, "rfc2833compensate")) { + ast_set_flag(&mask[1], SIP_PAGE2_RFC2833_COMPENSATE); + ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RFC2833_COMPENSATE); + } else if (!strcasecmp(v->name, "buggymwi")) { + ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI); + ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI); + } else if (!strcasecmp(v->name, "t38pt_usertpsource")) { + ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION); + ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION); + } else + res = 0; + + return res; +} + +/*! \brief Add SIP domain to list of domains we are responsible for */ +static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context) +{ + struct domain *d; + + if (ast_strlen_zero(domain)) { + ast_log(LOG_WARNING, "Zero length domain.\n"); + return 1; + } + + if (!(d = ast_calloc(1, sizeof(*d)))) + return 0; + + ast_copy_string(d->domain, domain, sizeof(d->domain)); + + if (!ast_strlen_zero(context)) + ast_copy_string(d->context, context, sizeof(d->context)); + + d->mode = mode; + + AST_LIST_LOCK(&domain_list); + AST_LIST_INSERT_TAIL(&domain_list, d, list); + AST_LIST_UNLOCK(&domain_list); + + if (sipdebug) + ast_log(LOG_DEBUG, "Added local SIP domain '%s'\n", domain); + + return 1; +} + +/*! \brief check_sip_domain: Check if domain part of uri is local to our server */ +static int check_sip_domain(const char *domain, char *context, size_t len) +{ + struct domain *d; + int result = 0; + + AST_LIST_LOCK(&domain_list); + AST_LIST_TRAVERSE(&domain_list, d, list) { + if (strcasecmp(d->domain, domain)) + continue; + + if (len && !ast_strlen_zero(d->context)) + ast_copy_string(context, d->context, len); + + result = 1; + break; + } + AST_LIST_UNLOCK(&domain_list); + + return result; +} + +/*! \brief Clear our domain list (at reload) */ +static void clear_sip_domains(void) +{ + struct domain *d; + + AST_LIST_LOCK(&domain_list); + while ((d = AST_LIST_REMOVE_HEAD(&domain_list, list))) + free(d); + AST_LIST_UNLOCK(&domain_list); +} + + +/*! \brief Add realm authentication in list */ +static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno) +{ + char authcopy[256]; + char *username=NULL, *realm=NULL, *secret=NULL, *md5secret=NULL; + char *stringp; + struct sip_auth *a, *b, *auth; + + if (ast_strlen_zero(configuration)) + return authlist; + + if (option_debug) + ast_log(LOG_DEBUG, "Auth config :: %s\n", configuration); + + ast_copy_string(authcopy, configuration, sizeof(authcopy)); + stringp = authcopy; + + username = stringp; + realm = strrchr(stringp, '@'); + if (realm) + *realm++ = '\0'; + if (ast_strlen_zero(username) || ast_strlen_zero(realm)) { + ast_log(LOG_WARNING, "Format for authentication entry is user[:secret]@realm at line %d\n", lineno); + return authlist; + } + stringp = username; + username = strsep(&stringp, ":"); + if (username) { + secret = strsep(&stringp, ":"); + if (!secret) { + stringp = username; + md5secret = strsep(&stringp,"#"); + } + } + if (!(auth = ast_calloc(1, sizeof(*auth)))) + return authlist; + + ast_copy_string(auth->realm, realm, sizeof(auth->realm)); + ast_copy_string(auth->username, username, sizeof(auth->username)); + if (secret) + ast_copy_string(auth->secret, secret, sizeof(auth->secret)); + if (md5secret) + ast_copy_string(auth->md5secret, md5secret, sizeof(auth->md5secret)); + + /* find the end of the list */ + for (b = NULL, a = authlist; a ; b = a, a = a->next) + ; + if (b) + b->next = auth; /* Add structure add end of list */ + else + authlist = auth; + + if (option_verbose > 2) + ast_verbose("Added authentication for realm %s\n", realm); + + return authlist; + +} + +/*! \brief Clear realm authentication list (at reload) */ +static int clear_realm_authentication(struct sip_auth *authlist) +{ + struct sip_auth *a = authlist; + struct sip_auth *b; + + while (a) { + b = a; + a = a->next; + free(b); + } + + return 1; +} + +/*! \brief Find authentication for a specific realm */ +static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm) +{ + struct sip_auth *a; + + for (a = authlist; a; a = a->next) { + if (!strcasecmp(a->realm, realm)) + break; + } + + return a; +} + +/*! \brief Initiate a SIP user structure from configuration (configuration or realtime) */ +static struct sip_user *build_user(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime) +{ + struct sip_user *user; + int format; + struct ast_ha *oldha = NULL; + char *varname = NULL, *varval = NULL; + struct ast_variable *tmpvar = NULL; + struct ast_flags userflags[2] = {{(0)}}; + struct ast_flags mask[2] = {{(0)}}; + + + if (!(user = ast_calloc(1, sizeof(*user)))) + return NULL; + + suserobjs++; + ASTOBJ_INIT(user); + ast_copy_string(user->name, name, sizeof(user->name)); + oldha = user->ha; + user->ha = NULL; + ast_copy_flags(&user->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY); + ast_copy_flags(&user->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY); + user->capability = global_capability; + user->allowtransfer = global_allowtransfer; + user->maxcallbitrate = default_maxcallbitrate; + user->autoframing = global_autoframing; + user->prefs = default_prefs; + /* set default context */ + strcpy(user->context, default_context); + strcpy(user->language, default_language); + strcpy(user->mohinterpret, default_mohinterpret); + strcpy(user->mohsuggest, default_mohsuggest); + /* First we walk through the v parameters list and then the alt parameters list */ + for (; v || ((v = alt) && !(alt=NULL)); v = v->next) { + if (handle_common_options(&userflags[0], &mask[0], v)) + continue; + + if (!strcasecmp(v->name, "context")) { + ast_copy_string(user->context, v->value, sizeof(user->context)); + } else if (!strcasecmp(v->name, "subscribecontext")) { + ast_copy_string(user->subscribecontext, v->value, sizeof(user->subscribecontext)); + } else if (!strcasecmp(v->name, "setvar")) { + varname = ast_strdupa(v->value); + if ((varval = strchr(varname,'='))) { + *varval++ = '\0'; + if ((tmpvar = ast_variable_new(varname, varval))) { + tmpvar->next = user->chanvars; + user->chanvars = tmpvar; + } + } + } else if (!strcasecmp(v->name, "permit") || + !strcasecmp(v->name, "deny")) { + user->ha = ast_append_ha(v->name, v->value, user->ha); + } else if (!strcasecmp(v->name, "allowtransfer")) { + user->allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED; + } else if (!strcasecmp(v->name, "secret")) { + ast_copy_string(user->secret, v->value, sizeof(user->secret)); + } else if (!strcasecmp(v->name, "md5secret")) { + ast_copy_string(user->md5secret, v->value, sizeof(user->md5secret)); + } else if (!strcasecmp(v->name, "callerid")) { + ast_callerid_split(v->value, user->cid_name, sizeof(user->cid_name), user->cid_num, sizeof(user->cid_num)); + } else if (!strcasecmp(v->name, "fullname")) { + ast_copy_string(user->cid_name, v->value, sizeof(user->cid_name)); + } else if (!strcasecmp(v->name, "cid_number")) { + ast_copy_string(user->cid_num, v->value, sizeof(user->cid_num)); + } else if (!strcasecmp(v->name, "callgroup")) { + user->callgroup = ast_get_group(v->value); + } else if (!strcasecmp(v->name, "pickupgroup")) { + user->pickupgroup = ast_get_group(v->value); + } else if (!strcasecmp(v->name, "language")) { + ast_copy_string(user->language, v->value, sizeof(user->language)); + } else if (!strcasecmp(v->name, "mohinterpret") + || !strcasecmp(v->name, "musicclass") || !strcasecmp(v->name, "musiconhold")) { + ast_copy_string(user->mohinterpret, v->value, sizeof(user->mohinterpret)); + } else if (!strcasecmp(v->name, "mohsuggest")) { + ast_copy_string(user->mohsuggest, v->value, sizeof(user->mohsuggest)); + } else if (!strcasecmp(v->name, "accountcode")) { + ast_copy_string(user->accountcode, v->value, sizeof(user->accountcode)); + } else if (!strcasecmp(v->name, "call-limit")) { + user->call_limit = atoi(v->value); + if (user->call_limit < 0) + user->call_limit = 0; + } else if (!strcasecmp(v->name, "amaflags")) { + format = ast_cdr_amaflags2int(v->value); + if (format < 0) { + ast_log(LOG_WARNING, "Invalid AMA Flags: %s at line %d\n", v->value, v->lineno); + } else { + user->amaflags = format; + } + } else if (!strcasecmp(v->name, "allow")) { + ast_parse_allow_disallow(&user->prefs, &user->capability, v->value, 1); + } else if (!strcasecmp(v->name, "disallow")) { + ast_parse_allow_disallow(&user->prefs, &user->capability, v->value, 0); + } else if (!strcasecmp(v->name, "autoframing")) { + user->autoframing = ast_true(v->value); + } else if (!strcasecmp(v->name, "callingpres")) { + user->callingpres = ast_parse_caller_presentation(v->value); + if (user->callingpres == -1) + user->callingpres = atoi(v->value); + } else if (!strcasecmp(v->name, "maxcallbitrate")) { + user->maxcallbitrate = atoi(v->value); + if (user->maxcallbitrate < 0) + user->maxcallbitrate = default_maxcallbitrate; + } + /* We can't just report unknown options here because this may be a + * type=friend entry. All user options are valid for a peer, but not + * the other way around. */ + } + ast_copy_flags(&user->flags[0], &userflags[0], mask[0].flags); + ast_copy_flags(&user->flags[1], &userflags[1], mask[1].flags); + if (ast_test_flag(&user->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) + global_allowsubscribe = TRUE; /* No global ban any more */ + ast_free_ha(oldha); + return user; +} + +/*! \brief Set peer defaults before configuring specific configurations */ +static void set_peer_defaults(struct sip_peer *peer) +{ + if (peer->expire == 0) { + /* Don't reset expire or port time during reload + if we have an active registration + */ + peer->expire = -1; + peer->pokeexpire = -1; + peer->addr.sin_port = htons(STANDARD_SIP_PORT); + } + ast_copy_flags(&peer->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY); + ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY); + strcpy(peer->context, default_context); + strcpy(peer->subscribecontext, default_subscribecontext); + strcpy(peer->language, default_language); + strcpy(peer->mohinterpret, default_mohinterpret); + strcpy(peer->mohsuggest, default_mohsuggest); + peer->addr.sin_family = AF_INET; + peer->defaddr.sin_family = AF_INET; + peer->capability = global_capability; + peer->maxcallbitrate = default_maxcallbitrate; + peer->rtptimeout = global_rtptimeout; + peer->rtpholdtimeout = global_rtpholdtimeout; + peer->rtpkeepalive = global_rtpkeepalive; + peer->allowtransfer = global_allowtransfer; + peer->autoframing = global_autoframing; + strcpy(peer->vmexten, default_vmexten); + peer->secret[0] = '\0'; + peer->md5secret[0] = '\0'; + peer->cid_num[0] = '\0'; + peer->cid_name[0] = '\0'; + peer->fromdomain[0] = '\0'; + peer->fromuser[0] = '\0'; + peer->regexten[0] = '\0'; + peer->mailbox[0] = '\0'; + peer->callgroup = 0; + peer->pickupgroup = 0; + peer->maxms = default_qualify; + peer->prefs = default_prefs; +} + +/*! \brief Create temporary peer (used in autocreatepeer mode) */ +static struct sip_peer *temp_peer(const char *name) +{ + struct sip_peer *peer; + + if (!(peer = ast_calloc(1, sizeof(*peer)))) + return NULL; + + apeerobjs++; + ASTOBJ_INIT(peer); + set_peer_defaults(peer); + + ast_copy_string(peer->name, name, sizeof(peer->name)); + + ast_set_flag(&peer->flags[1], SIP_PAGE2_SELFDESTRUCT); + ast_set_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC); + peer->prefs = default_prefs; + reg_source_db(peer); + + return peer; +} + +/*! \brief Build peer from configuration (file or realtime static/dynamic) */ +static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime) +{ + struct sip_peer *peer = NULL; + struct ast_ha *oldha = NULL; + int obproxyfound=0; + int found=0; + int firstpass=1; + int format=0; /* Ama flags */ + time_t regseconds = 0; + char *varname = NULL, *varval = NULL; + struct ast_variable *tmpvar = NULL; + struct ast_flags peerflags[2] = {{(0)}}; + struct ast_flags mask[2] = {{(0)}}; + char fullcontact[sizeof(peer->fullcontact)] = ""; + + if (!realtime || ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) + /* Note we do NOT use find_peer here, to avoid realtime recursion */ + /* We also use a case-sensitive comparison (unlike find_peer) so + that case changes made to the peer name will be properly handled + during reload + */ + peer = ASTOBJ_CONTAINER_FIND_UNLINK_FULL(&peerl, name, name, 0, 0, strcmp); + + if (peer) { + /* Already in the list, remove it and it will be added back (or FREE'd) */ + found = 1; + if (!(peer->objflags & ASTOBJ_FLAG_MARKED)) + firstpass = 0; + } else { + if (!(peer = ast_calloc(1, sizeof(*peer)))) + return NULL; + + if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) + rpeerobjs++; + else + speerobjs++; + ASTOBJ_INIT(peer); + } + /* Note that our peer HAS had its reference count incrased */ + if (firstpass) { + peer->lastmsgssent = -1; + oldha = peer->ha; + peer->ha = NULL; + set_peer_defaults(peer); /* Set peer defaults */ + } + if (!found && name) + ast_copy_string(peer->name, name, sizeof(peer->name)); + + /* If we have channel variables, remove them (reload) */ + if (peer->chanvars) { + ast_variables_destroy(peer->chanvars); + peer->chanvars = NULL; + /* XXX should unregister ? */ + } + + /* If we have realm authentication information, remove them (reload) */ + clear_realm_authentication(peer->auth); + peer->auth = NULL; + + for (; v || ((v = alt) && !(alt=NULL)); v = v->next) { + if (handle_common_options(&peerflags[0], &mask[0], v)) + continue; + if (realtime && !strcasecmp(v->name, "regseconds")) { + ast_get_time_t(v->value, ®seconds, 0, NULL); + } else if (realtime && !strcasecmp(v->name, "ipaddr") && !ast_strlen_zero(v->value) ) { + inet_aton(v->value, &(peer->addr.sin_addr)); + } else if (realtime && !strcasecmp(v->name, "name")) + ast_copy_string(peer->name, v->value, sizeof(peer->name)); + else if (realtime && !strcasecmp(v->name, "fullcontact")) { + /* Reconstruct field, because realtime separates our value at the ';' */ + if (!ast_strlen_zero(fullcontact)) { + strncat(fullcontact, ";", sizeof(fullcontact) - strlen(fullcontact) - 1); + strncat(fullcontact, v->value, sizeof(fullcontact) - strlen(fullcontact) - 1); + } else { + ast_copy_string(fullcontact, v->value, sizeof(fullcontact)); + ast_set_flag(&peer->flags[1], SIP_PAGE2_RT_FROMCONTACT); + } + } else if (!strcasecmp(v->name, "secret")) + ast_copy_string(peer->secret, v->value, sizeof(peer->secret)); + else if (!strcasecmp(v->name, "md5secret")) + ast_copy_string(peer->md5secret, v->value, sizeof(peer->md5secret)); + else if (!strcasecmp(v->name, "auth")) + peer->auth = add_realm_authentication(peer->auth, v->value, v->lineno); + else if (!strcasecmp(v->name, "callerid")) { + ast_callerid_split(v->value, peer->cid_name, sizeof(peer->cid_name), peer->cid_num, sizeof(peer->cid_num)); + } else if (!strcasecmp(v->name, "fullname")) { + ast_copy_string(peer->cid_name, v->value, sizeof(peer->cid_name)); + } else if (!strcasecmp(v->name, "cid_number")) { + ast_copy_string(peer->cid_num, v->value, sizeof(peer->cid_num)); + } else if (!strcasecmp(v->name, "context")) { + ast_copy_string(peer->context, v->value, sizeof(peer->context)); + } else if (!strcasecmp(v->name, "subscribecontext")) { + ast_copy_string(peer->subscribecontext, v->value, sizeof(peer->subscribecontext)); + } else if (!strcasecmp(v->name, "fromdomain")) { + ast_copy_string(peer->fromdomain, v->value, sizeof(peer->fromdomain)); + } else if (!strcasecmp(v->name, "usereqphone")) { + ast_set2_flag(&peer->flags[0], ast_true(v->value), SIP_USEREQPHONE); + } else if (!strcasecmp(v->name, "fromuser")) { + ast_copy_string(peer->fromuser, v->value, sizeof(peer->fromuser)); + } else if (!strcasecmp(v->name, "host") || !strcasecmp(v->name, "outboundproxy")) { + if (!strcasecmp(v->value, "dynamic")) { + if (!strcasecmp(v->name, "outboundproxy") || obproxyfound) { + ast_log(LOG_WARNING, "You can't have a dynamic outbound proxy, you big silly head at line %d.\n", v->lineno); + } else { + /* They'll register with us */ + if (!found || !ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)) { + /* Initialize stuff if this is a new peer, or if it used to be + * non-dynamic before the reload. */ + memset(&peer->addr.sin_addr, 0, 4); + if (peer->addr.sin_port) { + /* If we've already got a port, make it the default rather than absolute */ + peer->defaddr.sin_port = peer->addr.sin_port; + peer->addr.sin_port = 0; + } + } + ast_set_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC); + } + } else { + /* Non-dynamic. Make sure we become that way if we're not */ + if (!AST_SCHED_DEL(sched, peer->expire)) { + struct sip_peer *peer_ptr = peer; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + ast_clear_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC); + if (!obproxyfound || !strcasecmp(v->name, "outboundproxy")) { + if (ast_get_ip_or_srv(&peer->addr, v->value, srvlookup ? "_sip._udp" : NULL)) { + ASTOBJ_UNREF(peer, sip_destroy_peer); + return NULL; + } + } + if (!strcasecmp(v->name, "outboundproxy")) + obproxyfound=1; + else { + ast_copy_string(peer->tohost, v->value, sizeof(peer->tohost)); + if (!peer->addr.sin_port) + peer->addr.sin_port = htons(STANDARD_SIP_PORT); + } + if (global_dynamic_exclude_static) { + global_contact_ha = ast_append_ha("deny", (char *)ast_inet_ntoa(peer->addr.sin_addr), global_contact_ha); + } + } + } else if (!strcasecmp(v->name, "defaultip")) { + if (ast_get_ip(&peer->defaddr, v->value)) { + ASTOBJ_UNREF(peer, sip_destroy_peer); + return NULL; + } + } else if (!strcasecmp(v->name, "permit") || !strcasecmp(v->name, "deny")) { + peer->ha = ast_append_ha(v->name, v->value, peer->ha); + } else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny")) { + peer->contactha = ast_append_ha(v->name + 7, v->value, peer->contactha); + } else if (!strcasecmp(v->name, "port")) { + if (!realtime && ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)) + peer->defaddr.sin_port = htons(atoi(v->value)); + else + peer->addr.sin_port = htons(atoi(v->value)); + } else if (!strcasecmp(v->name, "callingpres")) { + peer->callingpres = ast_parse_caller_presentation(v->value); + if (peer->callingpres == -1) + peer->callingpres = atoi(v->value); + } else if (!strcasecmp(v->name, "username")) { + ast_copy_string(peer->username, v->value, sizeof(peer->username)); + } else if (!strcasecmp(v->name, "language")) { + ast_copy_string(peer->language, v->value, sizeof(peer->language)); + } else if (!strcasecmp(v->name, "regexten")) { + ast_copy_string(peer->regexten, v->value, sizeof(peer->regexten)); + } else if (!strcasecmp(v->name, "call-limit") || !strcasecmp(v->name, "incominglimit")) { + peer->call_limit = atoi(v->value); + if (peer->call_limit < 0) + peer->call_limit = 0; + } else if (!strcasecmp(v->name, "amaflags")) { + format = ast_cdr_amaflags2int(v->value); + if (format < 0) { + ast_log(LOG_WARNING, "Invalid AMA Flags for peer: %s at line %d\n", v->value, v->lineno); + } else { + peer->amaflags = format; + } + } else if (!strcasecmp(v->name, "accountcode")) { + ast_copy_string(peer->accountcode, v->value, sizeof(peer->accountcode)); + } else if (!strcasecmp(v->name, "mohinterpret") + || !strcasecmp(v->name, "musicclass") || !strcasecmp(v->name, "musiconhold")) { + ast_copy_string(peer->mohinterpret, v->value, sizeof(peer->mohinterpret)); + } else if (!strcasecmp(v->name, "mohsuggest")) { + ast_copy_string(peer->mohsuggest, v->value, sizeof(peer->mohsuggest)); + } else if (!strcasecmp(v->name, "mailbox")) { + ast_copy_string(peer->mailbox, v->value, sizeof(peer->mailbox)); + } else if (!strcasecmp(v->name, "hasvoicemail")) { + /* People expect that if 'hasvoicemail' is set, that the mailbox will + * be also set, even if not explicitly specified. */ + if (ast_true(v->value) && ast_strlen_zero(peer->mailbox)) { + ast_copy_string(peer->mailbox, name, sizeof(peer->mailbox)); + } + } else if (!strcasecmp(v->name, "subscribemwi")) { + ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_SUBSCRIBEMWIONLY); + } else if (!strcasecmp(v->name, "vmexten")) { + ast_copy_string(peer->vmexten, v->value, sizeof(peer->vmexten)); + } else if (!strcasecmp(v->name, "callgroup")) { + peer->callgroup = ast_get_group(v->value); + } else if (!strcasecmp(v->name, "allowtransfer")) { + peer->allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED; + } else if (!strcasecmp(v->name, "pickupgroup")) { + peer->pickupgroup = ast_get_group(v->value); + } else if (!strcasecmp(v->name, "allow")) { + ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, 1); + } else if (!strcasecmp(v->name, "disallow")) { + ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, 0); + } else if (!strcasecmp(v->name, "autoframing")) { + peer->autoframing = ast_true(v->value); + } else if (!strcasecmp(v->name, "rtptimeout")) { + if ((sscanf(v->value, "%d", &peer->rtptimeout) != 1) || (peer->rtptimeout < 0)) { + ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno); + peer->rtptimeout = global_rtptimeout; + } + } else if (!strcasecmp(v->name, "rtpholdtimeout")) { + if ((sscanf(v->value, "%d", &peer->rtpholdtimeout) != 1) || (peer->rtpholdtimeout < 0)) { + ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno); + peer->rtpholdtimeout = global_rtpholdtimeout; + } + } else if (!strcasecmp(v->name, "rtpkeepalive")) { + if ((sscanf(v->value, "%d", &peer->rtpkeepalive) != 1) || (peer->rtpkeepalive < 0)) { + ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno); + peer->rtpkeepalive = global_rtpkeepalive; + } + } else if (!strcasecmp(v->name, "setvar")) { + /* Set peer channel variable */ + varname = ast_strdupa(v->value); + if ((varval = strchr(varname, '='))) { + *varval++ = '\0'; + if ((tmpvar = ast_variable_new(varname, varval))) { + tmpvar->next = peer->chanvars; + peer->chanvars = tmpvar; + } + } + } else if (!strcasecmp(v->name, "qualify")) { + if (!strcasecmp(v->value, "no")) { + peer->maxms = 0; + } else if (!strcasecmp(v->value, "yes")) { + peer->maxms = default_qualify ? default_qualify : DEFAULT_MAXMS; + } else if (sscanf(v->value, "%d", &peer->maxms) != 1) { + ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno); + peer->maxms = 0; + } + if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && peer->maxms > 0) { + /* This would otherwise cause a network storm, where the + * qualify response refreshes the peer from the database, + * which in turn causes another qualify to be sent, ad + * infinitum. */ + ast_log(LOG_WARNING, "Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer '%s'\n", peer->name); + peer->maxms = 0; + } + } else if (!strcasecmp(v->name, "maxcallbitrate")) { + peer->maxcallbitrate = atoi(v->value); + if (peer->maxcallbitrate < 0) + peer->maxcallbitrate = default_maxcallbitrate; + } + } + if (!ast_strlen_zero(fullcontact)) { + ast_copy_string(peer->fullcontact, fullcontact, sizeof(peer->fullcontact)); + /* We have a hostname in the fullcontact, but if we don't have an + * address listed on the entry (or if it's 'dynamic'), then we need to + * parse the entry to obtain the IP address, so a dynamic host can be + * contacted immediately after reload (as opposed to waiting for it to + * register once again). */ + __set_address_from_contact(fullcontact, &peer->addr); + } + + if (!ast_test_flag(&global_flags[1], SIP_PAGE2_IGNOREREGEXPIRE) && ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC) && realtime) { + time_t nowtime = time(NULL); + + if ((nowtime - regseconds) > 0) { + destroy_association(peer); + memset(&peer->addr, 0, sizeof(peer->addr)); + if (option_debug) + ast_log(LOG_DEBUG, "Bah, we're expired (%d/%d/%d)!\n", (int)(nowtime - regseconds), (int)regseconds, (int)nowtime); + } + } + ast_copy_flags(&peer->flags[0], &peerflags[0], mask[0].flags); + ast_copy_flags(&peer->flags[1], &peerflags[1], mask[1].flags); + if (ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) + global_allowsubscribe = TRUE; /* No global ban any more */ + if (!found && ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC) && !ast_test_flag(&peer->flags[0], SIP_REALTIME)) + reg_source_db(peer); + ASTOBJ_UNMARK(peer); + ast_free_ha(oldha); + return peer; +} + +/*! \brief Re-read SIP.conf config file +\note This function reloads all config data, except for + active peers (with registrations). They will only + change configuration data at restart, not at reload. + SIP debug and recordhistory state will not change + */ +static int reload_config(enum channelreloadreason reason) +{ + struct ast_config *cfg, *ucfg; + struct ast_variable *v; + struct sip_peer *peer; + struct sip_user *user; + struct ast_hostent ahp; + char *cat, *stringp, *context, *oldregcontext; + char newcontexts[AST_MAX_CONTEXT], oldcontexts[AST_MAX_CONTEXT]; + struct hostent *hp; + int format; + struct ast_flags dummy[2]; + int auto_sip_domains = FALSE; + struct sockaddr_in old_bindaddr = bindaddr; + int registry_count = 0, peer_count = 0, user_count = 0; + unsigned int temp_tos = 0; + struct ast_flags debugflag = {0}; + + cfg = ast_config_load(config); + + /* We *must* have a config file otherwise stop immediately */ + if (!cfg) { + ast_log(LOG_NOTICE, "Unable to load config %s\n", config); + return -1; + } + + if (option_debug > 3) + ast_log(LOG_DEBUG, "--------------- SIP reload started\n"); + + clear_realm_authentication(authl); + clear_sip_domains(); + authl = NULL; + + ast_free_ha(global_contact_ha); + global_contact_ha = NULL; + + /* First, destroy all outstanding registry calls */ + /* This is needed, since otherwise active registry entries will not be destroyed */ + ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do { + ASTOBJ_RDLOCK(iterator); + if (iterator->call) { + if (option_debug > 2) + ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", iterator->username, iterator->hostname); + /* This will also remove references to the registry */ + sip_destroy(iterator->call); + } + ASTOBJ_UNLOCK(iterator); + + } while(0)); + + /* Then, actually destroy users and registry */ + ASTOBJ_CONTAINER_DESTROYALL(&userl, sip_destroy_user); + if (option_debug > 3) + ast_log(LOG_DEBUG, "--------------- Done destroying user list\n"); + ASTOBJ_CONTAINER_DESTROYALL(®l, sip_registry_destroy); + if (option_debug > 3) + ast_log(LOG_DEBUG, "--------------- Done destroying registry list\n"); + ASTOBJ_CONTAINER_MARKALL(&peerl); + + /* Initialize copy of current global_regcontext for later use in removing stale contexts */ + ast_copy_string(oldcontexts, global_regcontext, sizeof(oldcontexts)); + oldregcontext = oldcontexts; + + /* Clear all flags before setting default values */ + /* Preserve debugging settings for console */ + ast_copy_flags(&debugflag, &global_flags[1], SIP_PAGE2_DEBUG_CONSOLE); + ast_clear_flag(&global_flags[0], AST_FLAGS_ALL); + ast_clear_flag(&global_flags[1], AST_FLAGS_ALL); + ast_copy_flags(&global_flags[1], &debugflag, SIP_PAGE2_DEBUG_CONSOLE); + + /* Reset IP addresses */ + memset(&bindaddr, 0, sizeof(bindaddr)); + ast_free_ha(localaddr); + memset(&localaddr, 0, sizeof(localaddr)); + memset(&externip, 0, sizeof(externip)); + memset(&default_prefs, 0 , sizeof(default_prefs)); + outboundproxyip.sin_port = htons(STANDARD_SIP_PORT); + outboundproxyip.sin_family = AF_INET; /* Type of address: IPv4 */ + ourport = STANDARD_SIP_PORT; + srvlookup = DEFAULT_SRVLOOKUP; + global_tos_sip = DEFAULT_TOS_SIP; + global_tos_audio = DEFAULT_TOS_AUDIO; + global_tos_video = DEFAULT_TOS_VIDEO; + externhost[0] = '\0'; /* External host name (for behind NAT DynDNS support) */ + externexpire = 0; /* Expiration for DNS re-issuing */ + externrefresh = 10; + memset(&outboundproxyip, 0, sizeof(outboundproxyip)); + + /* Reset channel settings to default before re-configuring */ + allow_external_domains = DEFAULT_ALLOW_EXT_DOM; /* Allow external invites */ + global_regcontext[0] = '\0'; + expiry = DEFAULT_EXPIRY; + global_notifyringing = DEFAULT_NOTIFYRINGING; + global_limitonpeers = FALSE; + global_directrtpsetup = FALSE; /* Experimental feature, disabled by default */ + global_notifyhold = FALSE; + global_alwaysauthreject = 0; + global_allowsubscribe = FALSE; + ast_copy_string(global_useragent, DEFAULT_USERAGENT, sizeof(global_useragent)); + ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime)); + if (ast_strlen_zero(ast_config_AST_SYSTEM_NAME)) + ast_copy_string(global_realm, DEFAULT_REALM, sizeof(global_realm)); + else + ast_copy_string(global_realm, ast_config_AST_SYSTEM_NAME, sizeof(global_realm)); + ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid)); + compactheaders = DEFAULT_COMPACTHEADERS; + global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT; + global_regattempts_max = 0; + pedanticsipchecking = DEFAULT_PEDANTIC; + global_mwitime = DEFAULT_MWITIME; + autocreatepeer = DEFAULT_AUTOCREATEPEER; + global_autoframing = 0; + global_allowguest = DEFAULT_ALLOWGUEST; + global_rtptimeout = 0; + global_rtpholdtimeout = 0; + global_rtpkeepalive = 0; + global_allowtransfer = TRANSFER_OPENFORALL; /* Merrily accept all transfers by default */ + global_rtautoclear = 120; + ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE); /* Default for peers, users: TRUE */ + ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP); /* Default for peers, users: TRUE */ + ast_set_flag(&global_flags[1], SIP_PAGE2_RTUPDATE); + + /* Initialize some reasonable defaults at SIP reload (used both for channel and as default for peers and users */ + ast_copy_string(default_context, DEFAULT_CONTEXT, sizeof(default_context)); + default_subscribecontext[0] = '\0'; + default_language[0] = '\0'; + default_fromdomain[0] = '\0'; + default_qualify = DEFAULT_QUALIFY; + default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE; + ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret)); + ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest)); + ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten)); + ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */ + ast_set_flag(&global_flags[0], SIP_NAT_RFC3581); /*!< NAT support if requested by device with rport */ + ast_set_flag(&global_flags[0], SIP_CAN_REINVITE); /*!< Allow re-invites */ + + /* Debugging settings, always default to off */ + dumphistory = FALSE; + recordhistory = FALSE; + ast_clear_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG); + + /* Misc settings for the channel */ + global_relaxdtmf = FALSE; + global_callevents = FALSE; + global_t1min = DEFAULT_T1MIN; + + global_matchexterniplocally = FALSE; + + /* Copy the default jb config over global_jbconf */ + memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf)); + + ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT); + + /* Read the [general] config section of sip.conf (or from realtime config) */ + for (v = ast_variable_browse(cfg, "general"); v; v = v->next) { + if (handle_common_options(&global_flags[0], &dummy[0], v)) + continue; + /* handle jb conf */ + if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) + continue; + + /* Create the interface list */ + if (!strcasecmp(v->name, "context")) { + ast_copy_string(default_context, v->value, sizeof(default_context)); + } else if (!strcasecmp(v->name, "subscribecontext")) { + ast_copy_string(default_subscribecontext, v->value, sizeof(default_subscribecontext)); + } else if (!strcasecmp(v->name, "allowguest")) { + global_allowguest = ast_true(v->value) ? 1 : 0; + } else if (!strcasecmp(v->name, "realm")) { + ast_copy_string(global_realm, v->value, sizeof(global_realm)); + } else if (!strcasecmp(v->name, "useragent")) { + ast_copy_string(global_useragent, v->value, sizeof(global_useragent)); + if (option_debug) + ast_log(LOG_DEBUG, "Setting SIP channel User-Agent Name to %s\n", global_useragent); + } else if (!strcasecmp(v->name, "allowtransfer")) { + global_allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED; + } else if (!strcasecmp(v->name, "rtcachefriends")) { + ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_RTCACHEFRIENDS); + } else if (!strcasecmp(v->name, "rtsavesysname")) { + ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_RTSAVE_SYSNAME); + } else if (!strcasecmp(v->name, "rtupdate")) { + ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_RTUPDATE); + } else if (!strcasecmp(v->name, "ignoreregexpire")) { + ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_IGNOREREGEXPIRE); + } else if (!strcasecmp(v->name, "t1min")) { + global_t1min = atoi(v->value); + } else if (!strcasecmp(v->name, "dynamic_exclude_static") || !strcasecmp(v->name, "dynamic_excludes_static")) { + global_dynamic_exclude_static = ast_true(v->value); + } else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny")) { + global_contact_ha = ast_append_ha(v->name + 7, v->value, global_contact_ha); + } else if (!strcasecmp(v->name, "rtautoclear")) { + int i = atoi(v->value); + if (i > 0) + global_rtautoclear = i; + else + i = 0; + ast_set2_flag(&global_flags[1], i || ast_true(v->value), SIP_PAGE2_RTAUTOCLEAR); + } else if (!strcasecmp(v->name, "usereqphone")) { + ast_set2_flag(&global_flags[0], ast_true(v->value), SIP_USEREQPHONE); + } else if (!strcasecmp(v->name, "relaxdtmf")) { + global_relaxdtmf = ast_true(v->value); + } else if (!strcasecmp(v->name, "checkmwi")) { + if ((sscanf(v->value, "%d", &global_mwitime) != 1) || (global_mwitime < 0)) { + ast_log(LOG_WARNING, "'%s' is not a valid MWI time setting at line %d. Using default (10).\n", v->value, v->lineno); + global_mwitime = DEFAULT_MWITIME; + } + } else if (!strcasecmp(v->name, "vmexten")) { + ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten)); + } else if (!strcasecmp(v->name, "rtptimeout")) { + if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) { + ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno); + global_rtptimeout = 0; + } + } else if (!strcasecmp(v->name, "rtpholdtimeout")) { + if ((sscanf(v->value, "%d", &global_rtpholdtimeout) != 1) || (global_rtpholdtimeout < 0)) { + ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno); + global_rtpholdtimeout = 0; + } + } else if (!strcasecmp(v->name, "rtpkeepalive")) { + if ((sscanf(v->value, "%d", &global_rtpkeepalive) != 1) || (global_rtpkeepalive < 0)) { + ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno); + global_rtpkeepalive = 0; + } + } else if (!strcasecmp(v->name, "compactheaders")) { + compactheaders = ast_true(v->value); + } else if (!strcasecmp(v->name, "notifymimetype")) { + ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime)); + } else if (!strncasecmp(v->name, "limitonpeer", 11)) { + global_limitonpeers = ast_true(v->value); + } else if (!strcasecmp(v->name, "directrtpsetup")) { + global_directrtpsetup = ast_true(v->value); + } else if (!strcasecmp(v->name, "notifyringing")) { + global_notifyringing = ast_true(v->value); + } else if (!strcasecmp(v->name, "notifyhold")) { + global_notifyhold = ast_true(v->value); + } else if (!strcasecmp(v->name, "alwaysauthreject")) { + global_alwaysauthreject = ast_true(v->value); + } else if (!strcasecmp(v->name, "mohinterpret") + || !strcasecmp(v->name, "musicclass") || !strcasecmp(v->name, "musiconhold")) { + ast_copy_string(default_mohinterpret, v->value, sizeof(default_mohinterpret)); + } else if (!strcasecmp(v->name, "mohsuggest")) { + ast_copy_string(default_mohsuggest, v->value, sizeof(default_mohsuggest)); + } else if (!strcasecmp(v->name, "language")) { + ast_copy_string(default_language, v->value, sizeof(default_language)); + } else if (!strcasecmp(v->name, "regcontext")) { + ast_copy_string(newcontexts, v->value, sizeof(newcontexts)); + stringp = newcontexts; + /* Let's remove any contexts that are no longer defined in regcontext */ + cleanup_stale_contexts(stringp, oldregcontext); + /* Create contexts if they don't exist already */ + while ((context = strsep(&stringp, "&"))) { + if (!ast_context_find(context)) + ast_context_create(NULL, context,"SIP"); + } + ast_copy_string(global_regcontext, v->value, sizeof(global_regcontext)); + } else if (!strcasecmp(v->name, "callerid")) { + ast_copy_string(default_callerid, v->value, sizeof(default_callerid)); + } else if (!strcasecmp(v->name, "fromdomain")) { + ast_copy_string(default_fromdomain, v->value, sizeof(default_fromdomain)); + } else if (!strcasecmp(v->name, "outboundproxy")) { + if (ast_get_ip_or_srv(&outboundproxyip, v->value, srvlookup ? "_sip._udp" : NULL) < 0) + ast_log(LOG_WARNING, "Unable to locate host '%s'\n", v->value); + } else if (!strcasecmp(v->name, "outboundproxyport")) { + /* Port needs to be after IP */ + sscanf(v->value, "%d", &format); + outboundproxyip.sin_port = htons(format); + } else if (!strcasecmp(v->name, "autocreatepeer")) { + autocreatepeer = ast_true(v->value); + } else if (!strcasecmp(v->name, "srvlookup")) { + srvlookup = ast_true(v->value); + } else if (!strcasecmp(v->name, "pedantic")) { + pedanticsipchecking = ast_true(v->value); + } else if (!strcasecmp(v->name, "maxexpirey") || !strcasecmp(v->name, "maxexpiry")) { + max_expiry = atoi(v->value); + if (max_expiry < 1) + max_expiry = DEFAULT_MAX_EXPIRY; + } else if (!strcasecmp(v->name, "minexpirey") || !strcasecmp(v->name, "minexpiry")) { + min_expiry = atoi(v->value); + if (min_expiry < 1) + min_expiry = DEFAULT_MIN_EXPIRY; + } else if (!strcasecmp(v->name, "defaultexpiry") || !strcasecmp(v->name, "defaultexpirey")) { + default_expiry = atoi(v->value); + if (default_expiry < 1) + default_expiry = DEFAULT_DEFAULT_EXPIRY; + } else if (!strcasecmp(v->name, "sipdebug")) { /* XXX maybe ast_set2_flags ? */ + if (ast_true(v->value)) + ast_set_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG); + } else if (!strcasecmp(v->name, "dumphistory")) { + dumphistory = ast_true(v->value); + } else if (!strcasecmp(v->name, "recordhistory")) { + recordhistory = ast_true(v->value); + } else if (!strcasecmp(v->name, "registertimeout")) { + global_reg_timeout = atoi(v->value); + if (global_reg_timeout < 1) + global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT; + } else if (!strcasecmp(v->name, "registerattempts")) { + global_regattempts_max = atoi(v->value); + } else if (!strcasecmp(v->name, "bindaddr")) { + if (!(hp = ast_gethostbyname(v->value, &ahp))) { + ast_log(LOG_WARNING, "Invalid address: %s\n", v->value); + } else { + memcpy(&bindaddr.sin_addr, hp->h_addr, sizeof(bindaddr.sin_addr)); + } + } else if (!strcasecmp(v->name, "localnet")) { + struct ast_ha *na; + if (!(na = ast_append_ha("d", v->value, localaddr))) + ast_log(LOG_WARNING, "Invalid localnet value: %s\n", v->value); + else + localaddr = na; + } else if (!strcasecmp(v->name, "localmask")) { + ast_log(LOG_WARNING, "Use of localmask is no long supported -- use localnet with mask syntax\n"); + } else if (!strcasecmp(v->name, "externip")) { + if (!(hp = ast_gethostbyname(v->value, &ahp))) + ast_log(LOG_WARNING, "Invalid address for externip keyword: %s\n", v->value); + else + memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr)); + externexpire = 0; + } else if (!strcasecmp(v->name, "externhost")) { + ast_copy_string(externhost, v->value, sizeof(externhost)); + if (!(hp = ast_gethostbyname(externhost, &ahp))) + ast_log(LOG_WARNING, "Invalid address for externhost keyword: %s\n", externhost); + else + memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr)); + externexpire = time(NULL); + } else if (!strcasecmp(v->name, "externrefresh")) { + if (sscanf(v->value, "%d", &externrefresh) != 1) { + ast_log(LOG_WARNING, "Invalid externrefresh value '%s', must be an integer >0 at line %d\n", v->value, v->lineno); + externrefresh = 10; + } + } else if (!strcasecmp(v->name, "allow")) { + ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, 1); + } else if (!strcasecmp(v->name, "disallow")) { + ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, 0); + } else if (!strcasecmp(v->name, "autoframing")) { + global_autoframing = ast_true(v->value); + } else if (!strcasecmp(v->name, "allowexternaldomains")) { + allow_external_domains = ast_true(v->value); + } else if (!strcasecmp(v->name, "autodomain")) { + auto_sip_domains = ast_true(v->value); + } else if (!strcasecmp(v->name, "domain")) { + char *domain = ast_strdupa(v->value); + char *context = strchr(domain, ','); + + if (context) + *context++ = '\0'; + + if (option_debug && ast_strlen_zero(context)) + ast_log(LOG_DEBUG, "No context specified at line %d for domain '%s'\n", v->lineno, domain); + if (ast_strlen_zero(domain)) + ast_log(LOG_WARNING, "Empty domain specified at line %d\n", v->lineno); + else + add_sip_domain(ast_strip(domain), SIP_DOMAIN_CONFIG, context ? ast_strip(context) : ""); + } else if (!strcasecmp(v->name, "register")) { + if (sip_register(v->value, v->lineno) == 0) + registry_count++; + } else if (!strcasecmp(v->name, "tos")) { + if (!ast_str2tos(v->value, &temp_tos)) { + global_tos_sip = temp_tos; + global_tos_audio = temp_tos; + global_tos_video = temp_tos; + ast_log(LOG_WARNING, "tos value at line %d is deprecated. See doc/ip-tos.txt for more information.\n", v->lineno); + } else + ast_log(LOG_WARNING, "Invalid tos value at line %d, See doc/ip-tos.txt for more information.\n", v->lineno); + } else if (!strcasecmp(v->name, "tos_sip")) { + if (ast_str2tos(v->value, &global_tos_sip)) + ast_log(LOG_WARNING, "Invalid tos_sip value at line %d, recommended value is 'cs3'. See doc/ip-tos.txt.\n", v->lineno); + } else if (!strcasecmp(v->name, "tos_audio")) { + if (ast_str2tos(v->value, &global_tos_audio)) + ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, recommended value is 'ef'. See doc/ip-tos.txt.\n", v->lineno); + } else if (!strcasecmp(v->name, "tos_video")) { + if (ast_str2tos(v->value, &global_tos_video)) + ast_log(LOG_WARNING, "Invalid tos_video value at line %d, recommended value is 'af41'. See doc/ip-tos.txt.\n", v->lineno); + } else if (!strcasecmp(v->name, "bindport")) { + if (sscanf(v->value, "%d", &ourport) == 1) { + bindaddr.sin_port = htons(ourport); + } else { + ast_log(LOG_WARNING, "Invalid port number '%s' at line %d of %s\n", v->value, v->lineno, config); + } + } else if (!strcasecmp(v->name, "qualify")) { + if (!strcasecmp(v->value, "no")) { + default_qualify = 0; + } else if (!strcasecmp(v->value, "yes")) { + default_qualify = DEFAULT_MAXMS; + } else if (sscanf(v->value, "%d", &default_qualify) != 1) { + ast_log(LOG_WARNING, "Qualification default should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", v->lineno); + default_qualify = 0; + } + } else if (!strcasecmp(v->name, "callevents")) { + global_callevents = ast_true(v->value); + } else if (!strcasecmp(v->name, "maxcallbitrate")) { + default_maxcallbitrate = atoi(v->value); + if (default_maxcallbitrate < 0) + default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE; + } else if (!strcasecmp(v->name, "matchexterniplocally")) { + global_matchexterniplocally = ast_true(v->value); + } + } + + if (!allow_external_domains && AST_LIST_EMPTY(&domain_list)) { + ast_log(LOG_WARNING, "To disallow external domains, you need to configure local SIP domains.\n"); + allow_external_domains = 1; + } + + /* Build list of authentication to various SIP realms, i.e. service providers */ + for (v = ast_variable_browse(cfg, "authentication"); v ; v = v->next) { + /* Format for authentication is auth = username:password@realm */ + if (!strcasecmp(v->name, "auth")) + authl = add_realm_authentication(authl, v->value, v->lineno); + } + + ucfg = ast_config_load("users.conf"); + if (ucfg) { + struct ast_variable *gen; + int genhassip, genregistersip; + const char *hassip, *registersip; + + genhassip = ast_true(ast_variable_retrieve(ucfg, "general", "hassip")); + genregistersip = ast_true(ast_variable_retrieve(ucfg, "general", "registersip")); + gen = ast_variable_browse(ucfg, "general"); + cat = ast_category_browse(ucfg, NULL); + while (cat) { + if (strcasecmp(cat, "general")) { + hassip = ast_variable_retrieve(ucfg, cat, "hassip"); + registersip = ast_variable_retrieve(ucfg, cat, "registersip"); + if (ast_true(hassip) || (!hassip && genhassip)) { + user = build_user(cat, gen, ast_variable_browse(ucfg, cat), 0); + if (user) { + ASTOBJ_CONTAINER_LINK(&userl,user); + ASTOBJ_UNREF(user, sip_destroy_user); + user_count++; + } + peer = build_peer(cat, gen, ast_variable_browse(ucfg, cat), 0); + if (peer) { + ast_device_state_changed("SIP/%s", peer->name); + ASTOBJ_CONTAINER_LINK(&peerl,peer); + ASTOBJ_UNREF(peer, sip_destroy_peer); + peer_count++; + } + } + if (ast_true(registersip) || (!registersip && genregistersip)) { + char tmp[256]; + const char *host = ast_variable_retrieve(ucfg, cat, "host"); + const char *username = ast_variable_retrieve(ucfg, cat, "username"); + const char *secret = ast_variable_retrieve(ucfg, cat, "secret"); + const char *contact = ast_variable_retrieve(ucfg, cat, "contact"); + if (!host) + host = ast_variable_retrieve(ucfg, "general", "host"); + if (!username) + username = ast_variable_retrieve(ucfg, "general", "username"); + if (!secret) + secret = ast_variable_retrieve(ucfg, "general", "secret"); + if (!contact) + contact = "s"; + if (!ast_strlen_zero(username) && !ast_strlen_zero(host)) { + if (!ast_strlen_zero(secret)) + snprintf(tmp, sizeof(tmp), "%s:%s@%s/%s", username, secret, host, contact); + else + snprintf(tmp, sizeof(tmp), "%s@%s/%s", username, host, contact); + if (sip_register(tmp, 0) == 0) + registry_count++; + } + } + } + cat = ast_category_browse(ucfg, cat); + } + ast_config_destroy(ucfg); + } + + + /* Load peers, users and friends */ + cat = NULL; + while ( (cat = ast_category_browse(cfg, cat)) ) { + const char *utype; + if (!strcasecmp(cat, "general") || !strcasecmp(cat, "authentication")) + continue; + utype = ast_variable_retrieve(cfg, cat, "type"); + if (!utype) { + ast_log(LOG_WARNING, "Section '%s' lacks type\n", cat); + continue; + } else { + int is_user = 0, is_peer = 0; + if (!strcasecmp(utype, "user")) + is_user = 1; + else if (!strcasecmp(utype, "friend")) + is_user = is_peer = 1; + else if (!strcasecmp(utype, "peer")) + is_peer = 1; + else { + ast_log(LOG_WARNING, "Unknown type '%s' for '%s' in %s\n", utype, cat, "sip.conf"); + continue; + } + if (is_user) { + user = build_user(cat, ast_variable_browse(cfg, cat), NULL, 0); + if (user) { + ASTOBJ_CONTAINER_LINK(&userl,user); + ASTOBJ_UNREF(user, sip_destroy_user); + user_count++; + } + } + if (is_peer) { + peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0); + if (peer) { + ASTOBJ_CONTAINER_LINK(&peerl,peer); + ASTOBJ_UNREF(peer, sip_destroy_peer); + peer_count++; + } + } + } + } + if (ast_find_ourip(&__ourip, bindaddr)) { + ast_log(LOG_WARNING, "Unable to get own IP address, SIP disabled\n"); + ast_config_destroy(cfg); + return 0; + } + if (!ntohs(bindaddr.sin_port)) + bindaddr.sin_port = ntohs(STANDARD_SIP_PORT); + bindaddr.sin_family = AF_INET; + ast_mutex_lock(&netlock); + if ((sipsock > -1) && (memcmp(&old_bindaddr, &bindaddr, sizeof(struct sockaddr_in)))) { + close(sipsock); + sipsock = -1; + } + if (sipsock < 0) { + sipsock = socket(AF_INET, SOCK_DGRAM, 0); + if (sipsock < 0) { + ast_log(LOG_WARNING, "Unable to create SIP socket: %s\n", strerror(errno)); + ast_config_destroy(cfg); + return -1; + } else { + /* Allow SIP clients on the same host to access us: */ + const int reuseFlag = 1; + + setsockopt(sipsock, SOL_SOCKET, SO_REUSEADDR, + (const char*)&reuseFlag, + sizeof reuseFlag); + + ast_enable_packet_fragmentation(sipsock); + + if (bind(sipsock, (struct sockaddr *)&bindaddr, sizeof(bindaddr)) < 0) { + ast_log(LOG_WARNING, "Failed to bind to %s:%d: %s\n", + ast_inet_ntoa(bindaddr.sin_addr), ntohs(bindaddr.sin_port), + strerror(errno)); + close(sipsock); + sipsock = -1; + } else { + if (option_verbose > 1) { + ast_verbose(VERBOSE_PREFIX_2 "SIP Listening on %s:%d\n", + ast_inet_ntoa(bindaddr.sin_addr), ntohs(bindaddr.sin_port)); + ast_verbose(VERBOSE_PREFIX_2 "Using SIP TOS: %s\n", ast_tos2str(global_tos_sip)); + } + if (setsockopt(sipsock, IPPROTO_IP, IP_TOS, &global_tos_sip, sizeof(global_tos_sip))) + ast_log(LOG_WARNING, "Unable to set SIP TOS to %s\n", ast_tos2str(global_tos_sip)); + } + } + } + ast_mutex_unlock(&netlock); + + /* Add default domains - host name, IP address and IP:port */ + /* Only do this if user added any sip domain with "localdomains" */ + /* In order to *not* break backwards compatibility */ + /* Some phones address us at IP only, some with additional port number */ + if (auto_sip_domains) { + char temp[MAXHOSTNAMELEN]; + + /* First our default IP address */ + if (bindaddr.sin_addr.s_addr) + add_sip_domain(ast_inet_ntoa(bindaddr.sin_addr), SIP_DOMAIN_AUTO, NULL); + else + ast_log(LOG_NOTICE, "Can't add wildcard IP address to domain list, please add IP address to domain manually.\n"); + + /* Our extern IP address, if configured */ + if (externip.sin_addr.s_addr) + add_sip_domain(ast_inet_ntoa(externip.sin_addr), SIP_DOMAIN_AUTO, NULL); + + /* Extern host name (NAT traversal support) */ + if (!ast_strlen_zero(externhost)) + add_sip_domain(externhost, SIP_DOMAIN_AUTO, NULL); + + /* Our host name */ + if (!gethostname(temp, sizeof(temp))) + add_sip_domain(temp, SIP_DOMAIN_AUTO, NULL); + } + + /* Release configuration from memory */ + ast_config_destroy(cfg); + + /* Load the list of manual NOTIFY types to support */ + if (notify_types) + ast_config_destroy(notify_types); + notify_types = ast_config_load(notify_config); + + /* Done, tell the manager */ + manager_event(EVENT_FLAG_SYSTEM, "ChannelReload", "Channel: SIP\r\nReloadReason: %s\r\nRegistry_Count: %d\r\nPeer_Count: %d\r\nUser_Count: %d\r\n", channelreloadreason2txt(reason), registry_count, peer_count, user_count); + + return 0; +} + +static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan) +{ + struct sip_pvt *p; + struct ast_udptl *udptl = NULL; + + p = chan->tech_pvt; + if (!p) + return NULL; + + ast_mutex_lock(&p->lock); + if (p->udptl && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) + udptl = p->udptl; + ast_mutex_unlock(&p->lock); + return udptl; +} + +static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl) +{ + struct sip_pvt *p; + + p = chan->tech_pvt; + if (!p) + return -1; + ast_mutex_lock(&p->lock); + if (udptl) + ast_udptl_get_peer(udptl, &p->udptlredirip); + else + memset(&p->udptlredirip, 0, sizeof(p->udptlredirip)); + if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) { + if (!p->pendinginvite) { + if (option_debug > 2) { + ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0); + } + transmit_reinvite_with_t38_sdp(p); + } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { + if (option_debug > 2) { + ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0); + } + ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); + } + } + /* Reset lastrtprx timer */ + p->lastrtprx = p->lastrtptx = time(NULL); + ast_mutex_unlock(&p->lock); + return 0; +} + +/*! \brief Handle T38 reinvite + \todo Make sure we don't destroy the call if we can't handle the re-invite. + Nothing should be changed until we have processed the SDP and know that we + can handle it. +*/ +static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite) +{ + struct sip_pvt *p; + int flag = 0; + + p = chan->tech_pvt; + if (!p || !pvt->udptl) + return -1; + + /* Setup everything on the other side like offered/responded from first side */ + ast_mutex_lock(&p->lock); + + /*! \todo check if this is not set earlier when setting up the PVT. If not + maybe it should move there. */ + p->t38.jointcapability = p->t38.peercapability = pvt->t38.jointcapability; + + ast_udptl_set_far_max_datagram(p->udptl, ast_udptl_get_local_max_datagram(pvt->udptl)); + ast_udptl_set_local_max_datagram(p->udptl, ast_udptl_get_local_max_datagram(pvt->udptl)); + ast_udptl_set_error_correction_scheme(p->udptl, ast_udptl_get_error_correction_scheme(pvt->udptl)); + + if (reinvite) { /* If we are handling sending re-invite to the other side of the bridge */ + /*! \note The SIP_CAN_REINVITE flag is for RTP media redirects, + not really T38 re-invites which are different. In this + case it's used properly, to see if we can reinvite over + NAT + */ + if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE) && ast_test_flag(&pvt->flags[0], SIP_CAN_REINVITE)) { + ast_udptl_get_peer(pvt->udptl, &p->udptlredirip); + flag =1; + } else { + memset(&p->udptlredirip, 0, sizeof(p->udptlredirip)); + } + if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) { + if (!p->pendinginvite) { + if (option_debug > 2) { + if (flag) + ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port)); + else + ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip)); + } + transmit_reinvite_with_t38_sdp(p); + } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { + if (option_debug > 2) { + if (flag) + ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port)); + else + ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip)); + } + ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); + } + } + /* Reset lastrtprx timer */ + p->lastrtprx = p->lastrtptx = time(NULL); + ast_mutex_unlock(&p->lock); + return 0; + } else { /* If we are handling sending 200 OK to the other side of the bridge */ + if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE) && ast_test_flag(&pvt->flags[0], SIP_CAN_REINVITE)) { + ast_udptl_get_peer(pvt->udptl, &p->udptlredirip); + flag = 1; + } else { + memset(&p->udptlredirip, 0, sizeof(p->udptlredirip)); + } + if (option_debug > 2) { + if (flag) + ast_log(LOG_DEBUG, "Responding 200 OK on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port)); + else + ast_log(LOG_DEBUG, "Responding 200 OK on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip)); + } + pvt->t38.state = T38_ENABLED; + p->t38.state = T38_ENABLED; + if (option_debug > 1) { + ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>"); + ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", p->t38.state, chan ? chan->name : "<none>"); + } + transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL); + p->lastrtprx = p->lastrtptx = time(NULL); + ast_mutex_unlock(&p->lock); + return 0; + } +} + + +/*! \brief Returns null if we can't reinvite audio (part of RTP interface) */ +static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp) +{ + struct sip_pvt *p = NULL; + enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL; + + if (!(p = chan->tech_pvt)) + return AST_RTP_GET_FAILED; + + ast_mutex_lock(&p->lock); + if (!(p->rtp)) { + ast_mutex_unlock(&p->lock); + return AST_RTP_GET_FAILED; + } + + *rtp = p->rtp; + + if (ast_rtp_getnat(*rtp) && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) + res = AST_RTP_TRY_PARTIAL; + else if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) + res = AST_RTP_TRY_NATIVE; + else if (ast_test_flag(&global_jbconf, AST_JB_FORCED)) + res = AST_RTP_GET_FAILED; + + ast_mutex_unlock(&p->lock); + + return res; +} + +/*! \brief Returns null if we can't reinvite video (part of RTP interface) */ +static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp) +{ + struct sip_pvt *p = NULL; + enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL; + + if (!(p = chan->tech_pvt)) + return AST_RTP_GET_FAILED; + + ast_mutex_lock(&p->lock); + if (!(p->vrtp)) { + ast_mutex_unlock(&p->lock); + return AST_RTP_GET_FAILED; + } + + *rtp = p->vrtp; + + if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) + res = AST_RTP_TRY_NATIVE; + + ast_mutex_unlock(&p->lock); + + return res; +} + +/*! \brief Set the RTP peer for this call */ +static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active) +{ + struct sip_pvt *p; + int changed = 0; + + p = chan->tech_pvt; + if (!p) + return -1; + + /* Disable early RTP bridge */ + if (chan->_state != AST_STATE_UP && !global_directrtpsetup) /* We are in early state */ + return 0; + + ast_mutex_lock(&p->lock); + if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) { + /* If we're destroyed, don't bother */ + ast_mutex_unlock(&p->lock); + return 0; + } + + /* if this peer cannot handle reinvites of the media stream to devices + that are known to be behind a NAT, then stop the process now + */ + if (nat_active && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) { + ast_mutex_unlock(&p->lock); + return 0; + } + + if (rtp) { + changed |= ast_rtp_get_peer(rtp, &p->redirip); + } else if (p->redirip.sin_addr.s_addr || ntohs(p->redirip.sin_port) != 0) { + memset(&p->redirip, 0, sizeof(p->redirip)); + changed = 1; + } + if (vrtp) { + changed |= ast_rtp_get_peer(vrtp, &p->vredirip); + } else if (p->vredirip.sin_addr.s_addr || ntohs(p->vredirip.sin_port) != 0) { + memset(&p->vredirip, 0, sizeof(p->vredirip)); + changed = 1; + } + if (codecs) { + if ((p->redircodecs != codecs)) { + p->redircodecs = codecs; + changed = 1; + } + if ((p->capability & codecs) != p->capability) { + p->jointcapability &= codecs; + p->capability &= codecs; + changed = 1; + } + } + if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER) && !ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) { + if (chan->_state != AST_STATE_UP) { /* We are in early state */ + if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) + append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal."); + if (option_debug) + ast_log(LOG_DEBUG, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); + } else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */ + if (option_debug > 2) { + ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); + } + transmit_reinvite_with_sdp(p); + } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { + if (option_debug > 2) { + ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); + } + /* We have a pending Invite. Send re-invite when we're done with the invite */ + ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); + } + } + /* Reset lastrtprx timer */ + p->lastrtprx = p->lastrtptx = time(NULL); + ast_mutex_unlock(&p->lock); + return 0; +} + +static char *synopsis_dtmfmode = "Change the dtmfmode for a SIP call"; +static char *descrip_dtmfmode = "SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call\n"; +static char *app_dtmfmode = "SIPDtmfMode"; + +static char *app_sipaddheader = "SIPAddHeader"; +static char *synopsis_sipaddheader = "Add a SIP header to the outbound call"; + +static char *descrip_sipaddheader = "" +" SIPAddHeader(Header: Content)\n" +"Adds a header to a SIP call placed with DIAL.\n" +"Remember to user the X-header if you are adding non-standard SIP\n" +"headers, like \"X-Asterisk-Accountcode:\". Use this with care.\n" +"Adding the wrong headers may jeopardize the SIP dialog.\n" +"Always returns 0\n"; + + +/*! \brief Set the DTMFmode for an outbound SIP call (application) */ +static int sip_dtmfmode(struct ast_channel *chan, void *data) +{ + struct sip_pvt *p; + char *mode; + if (data) + mode = (char *)data; + else { + ast_log(LOG_WARNING, "This application requires the argument: info, inband, rfc2833\n"); + return 0; + } + ast_channel_lock(chan); + if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) { + ast_log(LOG_WARNING, "Call this application only on SIP incoming calls\n"); + ast_channel_unlock(chan); + return 0; + } + p = chan->tech_pvt; + if (!p) { + ast_channel_unlock(chan); + return 0; + } + ast_mutex_lock(&p->lock); + if (!strcasecmp(mode,"info")) { + ast_clear_flag(&p->flags[0], SIP_DTMF); + ast_set_flag(&p->flags[0], SIP_DTMF_INFO); + p->jointnoncodeccapability &= ~AST_RTP_DTMF; + } else if (!strcasecmp(mode,"rfc2833")) { + ast_clear_flag(&p->flags[0], SIP_DTMF); + ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833); + p->jointnoncodeccapability |= AST_RTP_DTMF; + } else if (!strcasecmp(mode,"inband")) { + ast_clear_flag(&p->flags[0], SIP_DTMF); + ast_set_flag(&p->flags[0], SIP_DTMF_INBAND); + p->jointnoncodeccapability &= ~AST_RTP_DTMF; + } else + ast_log(LOG_WARNING, "I don't know about this dtmf mode: %s\n",mode); + if (p->rtp) + ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833); + if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) { + if (!p->vad) { + p->vad = ast_dsp_new(); + ast_dsp_set_features(p->vad, DSP_FEATURE_DTMF_DETECT); + } + } else { + if (p->vad) { + ast_dsp_free(p->vad); + p->vad = NULL; + } + } + ast_mutex_unlock(&p->lock); + ast_channel_unlock(chan); + return 0; +} + +/*! \brief Add a SIP header to an outbound INVITE */ +static int sip_addheader(struct ast_channel *chan, void *data) +{ + int no = 0; + int ok = FALSE; + char varbuf[30]; + char *inbuf = (char *) data; + + if (ast_strlen_zero(inbuf)) { + ast_log(LOG_WARNING, "This application requires the argument: Header\n"); + return 0; + } + ast_channel_lock(chan); + + /* Check for headers */ + while (!ok && no <= 50) { + no++; + snprintf(varbuf, sizeof(varbuf), "__SIPADDHEADER%.2d", no); + + /* Compare without the leading underscores */ + if( (pbx_builtin_getvar_helper(chan, (const char *) varbuf + 2) == (const char *) NULL) ) + ok = TRUE; + } + if (ok) { + pbx_builtin_setvar_helper (chan, varbuf, inbuf); + if (sipdebug) + ast_log(LOG_DEBUG,"SIP Header added \"%s\" as %s\n", inbuf, varbuf); + } else { + ast_log(LOG_WARNING, "Too many SIP headers added, max 50\n"); + } + ast_channel_unlock(chan); + return 0; +} + +/*! \brief Transfer call before connect with a 302 redirect +\note Called by the transfer() dialplan application through the sip_transfer() + pbx interface function if the call is in ringing state +\todo Fix this function so that we wait for reply to the REFER and + react to errors, denials or other issues the other end might have. + */ +static int sip_sipredirect(struct sip_pvt *p, const char *dest) +{ + char *cdest; + char *extension, *host, *port; + char tmp[80]; + + cdest = ast_strdupa(dest); + + extension = strsep(&cdest, "@"); + host = strsep(&cdest, ":"); + port = strsep(&cdest, ":"); + if (ast_strlen_zero(extension)) { + ast_log(LOG_ERROR, "Missing mandatory argument: extension\n"); + return 0; + } + + /* we'll issue the redirect message here */ + if (!host) { + char *localtmp; + ast_copy_string(tmp, get_header(&p->initreq, "To"), sizeof(tmp)); + if (ast_strlen_zero(tmp)) { + ast_log(LOG_ERROR, "Cannot retrieve the 'To' header from the original SIP request!\n"); + return 0; + } + if ((localtmp = strcasestr(tmp, "sip:")) && (localtmp = strchr(localtmp, '@'))) { + char lhost[80], lport[80]; + memset(lhost, 0, sizeof(lhost)); + memset(lport, 0, sizeof(lport)); + localtmp++; + /* This is okey because lhost and lport are as big as tmp */ + sscanf(localtmp, "%[^<>:; ]:%[^<>:; ]", lhost, lport); + if (ast_strlen_zero(lhost)) { + ast_log(LOG_ERROR, "Can't find the host address\n"); + return 0; + } + host = ast_strdupa(lhost); + if (!ast_strlen_zero(lport)) { + port = ast_strdupa(lport); + } + } + } + + ast_string_field_build(p, our_contact, "Transfer <sip:%s@%s%s%s>", extension, host, port ? ":" : "", port ? port : ""); + transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq); + + sip_scheddestroy(p, SIP_TRANS_TIMEOUT); /* Make sure we stop send this reply. */ + sip_alreadygone(p); + return 0; +} + +/*! \brief Return SIP UA's codec (part of the RTP interface) */ +static int sip_get_codec(struct ast_channel *chan) +{ + struct sip_pvt *p = chan->tech_pvt; + return p->jointcapability ? p->jointcapability : p->capability; +} + +/*! \brief Send a poke to all known peers + Space them out 100 ms apart + XXX We might have a cool algorithm for this or use random - any suggestions? +*/ +static void sip_poke_all_peers(void) +{ + int ms = 0; + + if (!speerobjs) /* No peers, just give up */ + return; + + ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do { + ASTOBJ_WRLOCK(iterator); + if (!AST_SCHED_DEL(sched, iterator->pokeexpire)) { + struct sip_peer *peer_ptr = iterator; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + ms += 100; + iterator->pokeexpire = ast_sched_add(sched, ms, sip_poke_peer_s, ASTOBJ_REF(iterator)); + if (iterator->pokeexpire == -1) { + struct sip_peer *peer_ptr = iterator; + ASTOBJ_UNREF(peer_ptr, sip_destroy_peer); + } + ASTOBJ_UNLOCK(iterator); + } while (0) + ); +} + +/*! \brief Send all known registrations */ +static void sip_send_all_registers(void) +{ + int ms; + int regspacing; + if (!regobjs) + return; + regspacing = default_expiry * 1000/regobjs; + if (regspacing > 100) + regspacing = 100; + ms = regspacing; + ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do { + ASTOBJ_WRLOCK(iterator); + AST_SCHED_DEL(sched, iterator->expire); + ms += regspacing; + iterator->expire = ast_sched_add(sched, ms, sip_reregister, iterator); + ASTOBJ_UNLOCK(iterator); + } while (0) + ); +} + +/*! \brief Reload module */ +static int sip_do_reload(enum channelreloadreason reason) +{ + reload_config(reason); + + /* Prune peers who still are supposed to be deleted */ + ASTOBJ_CONTAINER_PRUNE_MARKED(&peerl, sip_destroy_peer); + if (option_debug > 3) + ast_log(LOG_DEBUG, "--------------- Done destroying pruned peers\n"); + + /* Send qualify (OPTIONS) to all peers */ + sip_poke_all_peers(); + + /* Register with all services */ + sip_send_all_registers(); + + if (option_debug > 3) + ast_log(LOG_DEBUG, "--------------- SIP reload done\n"); + + return 0; +} + +/*! \brief Force reload of module from cli */ +static int sip_reload(int fd, int argc, char *argv[]) +{ + ast_mutex_lock(&sip_reload_lock); + if (sip_reloading) + ast_verbose("Previous SIP reload not yet done\n"); + else { + sip_reloading = TRUE; + if (fd) + sip_reloadreason = CHANNEL_CLI_RELOAD; + else + sip_reloadreason = CHANNEL_MODULE_RELOAD; + } + ast_mutex_unlock(&sip_reload_lock); + restart_monitor(); + + return 0; +} + +/*! \brief Part of Asterisk module interface */ +static int reload(void) +{ + return sip_reload(0, 0, NULL); +} + +static struct ast_cli_entry cli_sip_debug_deprecated = + { { "sip", "debug", NULL }, + sip_do_debug_deprecated, "Enable SIP debugging", + debug_usage }; + +static struct ast_cli_entry cli_sip_no_debug_deprecated = + { { "sip", "no", "debug", NULL }, + sip_no_debug_deprecated, "Disable SIP debugging", + debug_usage }; + +static struct ast_cli_entry cli_sip[] = { + { { "sip", "show", "channels", NULL }, + sip_show_channels, "List active SIP channels", + show_channels_usage }, + + { { "sip", "show", "domains", NULL }, + sip_show_domains, "List our local SIP domains.", + show_domains_usage }, + + { { "sip", "show", "inuse", NULL }, + sip_show_inuse, "List all inuse/limits", + show_inuse_usage }, + + { { "sip", "show", "objects", NULL }, + sip_show_objects, "List all SIP object allocations", + show_objects_usage }, + + { { "sip", "show", "peers", NULL }, + sip_show_peers, "List defined SIP peers", + show_peers_usage }, + + { { "sip", "show", "registry", NULL }, + sip_show_registry, "List SIP registration status", + show_reg_usage }, + + { { "sip", "show", "settings", NULL }, + sip_show_settings, "Show SIP global settings", + show_settings_usage }, + + { { "sip", "show", "subscriptions", NULL }, + sip_show_subscriptions, "List active SIP subscriptions", + show_subscriptions_usage }, + + { { "sip", "show", "users", NULL }, + sip_show_users, "List defined SIP users", + show_users_usage }, + + { { "sip", "notify", NULL }, + sip_notify, "Send a notify packet to a SIP peer", + notify_usage, complete_sipnotify }, + + { { "sip", "show", "channel", NULL }, + sip_show_channel, "Show detailed SIP channel info", + show_channel_usage, complete_sipch }, + + { { "sip", "show", "history", NULL }, + sip_show_history, "Show SIP dialog history", + show_history_usage, complete_sipch }, + + { { "sip", "show", "peer", NULL }, + sip_show_peer, "Show details on specific SIP peer", + show_peer_usage, complete_sip_show_peer }, + + { { "sip", "show", "user", NULL }, + sip_show_user, "Show details on specific SIP user", + show_user_usage, complete_sip_show_user }, + + { { "sip", "prune", "realtime", NULL }, + sip_prune_realtime, "Prune cached Realtime object(s)", + prune_realtime_usage }, + + { { "sip", "prune", "realtime", "peer", NULL }, + sip_prune_realtime, "Prune cached Realtime peer(s)", + prune_realtime_usage, complete_sip_prune_realtime_peer }, + + { { "sip", "prune", "realtime", "user", NULL }, + sip_prune_realtime, "Prune cached Realtime user(s)", + prune_realtime_usage, complete_sip_prune_realtime_user }, + + { { "sip", "set", "debug", NULL }, + sip_do_debug, "Enable SIP debugging", + debug_usage, NULL, &cli_sip_debug_deprecated }, + + { { "sip", "set", "debug", "ip", NULL }, + sip_do_debug, "Enable SIP debugging on IP", + debug_usage }, + + { { "sip", "set", "debug", "peer", NULL }, + sip_do_debug, "Enable SIP debugging on Peername", + debug_usage, complete_sip_debug_peer }, + + { { "sip", "set", "debug", "off", NULL }, + sip_no_debug, "Disable SIP debugging", + no_debug_usage, NULL, &cli_sip_no_debug_deprecated }, + + { { "sip", "history", NULL }, + sip_do_history, "Enable SIP history", + history_usage }, + + { { "sip", "history", "off", NULL }, + sip_no_history, "Disable SIP history", + no_history_usage }, + + { { "sip", "reload", NULL }, + sip_reload, "Reload SIP configuration", + sip_reload_usage }, +}; + +/*! \brief PBX load module - initialization */ +static int load_module(void) +{ + ASTOBJ_CONTAINER_INIT(&userl); /* User object list */ + ASTOBJ_CONTAINER_INIT(&peerl); /* Peer object list */ + ASTOBJ_CONTAINER_INIT(®l); /* Registry object list */ + + if (!(sched = sched_context_create())) { + ast_log(LOG_ERROR, "Unable to create scheduler context\n"); + return AST_MODULE_LOAD_FAILURE; + } + + if (!(io = io_context_create())) { + ast_log(LOG_ERROR, "Unable to create I/O context\n"); + sched_context_destroy(sched); + return AST_MODULE_LOAD_FAILURE; + } + + sip_reloadreason = CHANNEL_MODULE_LOAD; + + if(reload_config(sip_reloadreason)) /* Load the configuration from sip.conf */ + return AST_MODULE_LOAD_DECLINE; + + /* Make sure we can register our sip channel type */ + if (ast_channel_register(&sip_tech)) { + ast_log(LOG_ERROR, "Unable to register channel type 'SIP'\n"); + io_context_destroy(io); + sched_context_destroy(sched); + return AST_MODULE_LOAD_FAILURE; + } + + /* Register all CLI functions for SIP */ + ast_cli_register_multiple(cli_sip, sizeof(cli_sip)/ sizeof(struct ast_cli_entry)); + + /* Tell the RTP subdriver that we're here */ + ast_rtp_proto_register(&sip_rtp); + + /* Tell the UDPTL subdriver that we're here */ + ast_udptl_proto_register(&sip_udptl); + + /* Register dialplan applications */ + ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode); + ast_register_application(app_sipaddheader, sip_addheader, synopsis_sipaddheader, descrip_sipaddheader); + + /* Register dialplan functions */ + ast_custom_function_register(&sip_header_function); + ast_custom_function_register(&sippeer_function); + ast_custom_function_register(&sipchaninfo_function); + ast_custom_function_register(&checksipdomain_function); + + /* Register manager commands */ + ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM, manager_sip_show_peers, + "List SIP peers (text format)", mandescr_show_peers); + ast_manager_register2("SIPshowpeer", EVENT_FLAG_SYSTEM, manager_sip_show_peer, + "Show SIP peer (text format)", mandescr_show_peer); + + sip_poke_all_peers(); + sip_send_all_registers(); + + /* And start the monitor for the first time */ + restart_monitor(); + + return AST_MODULE_LOAD_SUCCESS; +} + +/*! \brief PBX unload module API */ +static int unload_module(void) +{ + struct sip_pvt *p, *pl; + + /* First, take us out of the channel type list */ + ast_channel_unregister(&sip_tech); + + /* Unregister dial plan functions */ + ast_custom_function_unregister(&sipchaninfo_function); + ast_custom_function_unregister(&sippeer_function); + ast_custom_function_unregister(&sip_header_function); + ast_custom_function_unregister(&checksipdomain_function); + + /* Unregister dial plan applications */ + ast_unregister_application(app_dtmfmode); + ast_unregister_application(app_sipaddheader); + + /* Unregister CLI commands */ + ast_cli_unregister_multiple(cli_sip, sizeof(cli_sip) / sizeof(struct ast_cli_entry)); + + /* Disconnect from the RTP subsystem */ + ast_rtp_proto_unregister(&sip_rtp); + + /* Disconnect from UDPTL */ + ast_udptl_proto_unregister(&sip_udptl); + + /* Unregister AMI actions */ + ast_manager_unregister("SIPpeers"); + ast_manager_unregister("SIPshowpeer"); + + ast_mutex_lock(&iflock); + /* Hangup all interfaces if they have an owner */ + for (p = iflist; p ; p = p->next) { + if (p->owner) + ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD); + } + ast_mutex_unlock(&iflock); + + ast_mutex_lock(&monlock); + if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) { + pthread_cancel(monitor_thread); + pthread_kill(monitor_thread, SIGURG); + pthread_join(monitor_thread, NULL); + } + monitor_thread = AST_PTHREADT_STOP; + ast_mutex_unlock(&monlock); + +restartdestroy: + ast_mutex_lock(&iflock); + /* Destroy all the interfaces and free their memory */ + p = iflist; + while (p) { + pl = p; + p = p->next; + if (__sip_destroy(pl, TRUE) < 0) { + /* Something is still bridged, let it react to getting a hangup */ + iflist = p; + ast_mutex_unlock(&iflock); + usleep(1); + goto restartdestroy; + } + } + iflist = NULL; + ast_mutex_unlock(&iflock); + + /* Free memory for local network address mask */ + ast_free_ha(localaddr); + + ASTOBJ_CONTAINER_DESTROYALL(&userl, sip_destroy_user); + ASTOBJ_CONTAINER_DESTROY(&userl); + ASTOBJ_CONTAINER_DESTROYALL(&peerl, sip_destroy_peer); + ASTOBJ_CONTAINER_DESTROY(&peerl); + ASTOBJ_CONTAINER_DESTROYALL(®l, sip_registry_destroy); + ASTOBJ_CONTAINER_DESTROY(®l); + + clear_realm_authentication(authl); + clear_sip_domains(); + close(sipsock); + sched_context_destroy(sched); + + return 0; +} + +AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Session Initiation Protocol (SIP)", + .load = load_module, + .unload = unload_module, + .reload = reload, + ); |