? siptcpchanges.patch ? siptcpchanges2.patch ? keys/dh1024.pem ? keys/dh512.pem ? keys/servercert.pem ? keys/serverkey.pem ? keys/trustcerts.pem Index: channels/chan_sip.c =================================================================== RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.784 diff -u -r1.784 chan_sip.c --- channels/chan_sip.c 20 Jul 2005 17:05:18 -0000 1.784 +++ channels/chan_sip.c 27 Aug 2005 06:29:21 -0000 @@ -1,11709 +1,12376 @@ -/* - * Asterisk -- A telephony toolkit for Linux. - * - * Implementation of Session Initiation Protocol - * - * Copyright (C) 2004 - 2005, Digium, Inc. - * - * Mark Spencer - * - * This program is free software, distributed under the terms of - * the GNU General Public License - */ - - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "asterisk.h" - -ASTERISK_FILE_VERSION(__FILE__, "$Revision: 1.784 $") - -#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/lock.h" -#include "asterisk/sched.h" -#include "asterisk/io.h" -#include "asterisk/rtp.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/acl.h" -#include "asterisk/srv.h" -#include "asterisk/astdb.h" -#include "asterisk/causes.h" -#include "asterisk/utils.h" -#include "asterisk/file.h" -#include "asterisk/astobj.h" -#include "asterisk/dnsmgr.h" -#include "asterisk/devicestate.h" -#ifdef OSP_SUPPORT -#include "asterisk/astosp.h" -#endif - -#ifndef DEFAULT_USERAGENT -#define DEFAULT_USERAGENT "Asterisk PBX" -#endif - -#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 SIPDUMPER -#define DEFAULT_DEFAULT_EXPIRY 120 -#define DEFAULT_MAX_EXPIRY 3600 -#define DEFAULT_REGISTRATION_TIMEOUT 20 -#define DEFAULT_REGATTEMPTS_MAX 10 - -/* 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 */ - -static int max_expiry = DEFAULT_MAX_EXPIRY; -static int default_expiry = DEFAULT_DEFAULT_EXPIRY; - -#ifndef MAX -#define MAX(a,b) ((a) > (b) ? (a) : (b)) -#endif - -#define CALLERID_UNKNOWN "Unknown" - - - -#define DEFAULT_MAXMS 2000 /* Must be faster than 2 seconds by default */ -#define DEFAULT_FREQ_OK 60 * 1000 /* How often to check for the host to be up */ -#define DEFAULT_FREQ_NOTOK 10 * 1000 /* How often to check, if the host is down... */ - -#define DEFAULT_RETRANS 2000 /* How frequently to retransmit */ -#define MAX_RETRANS 5 /* Try only 5 times for retransmissions */ - - -#define DEBUG_READ 0 /* Recieved data */ -#define DEBUG_SEND 1 /* Transmit data */ - -static const char desc[] = "Session Initiation Protocol (SIP)"; -static const char channeltype[] = "SIP"; -static const char config[] = "sip.conf"; -static const char notify_config[] = "sip_notify.conf"; - -#define SIP_REGISTER 1 -#define SIP_OPTIONS 2 -#define SIP_NOTIFY 3 -#define SIP_INVITE 4 -#define SIP_ACK 5 -#define SIP_PRACK 6 -#define SIP_BYE 7 -#define SIP_REFER 8 -#define SIP_SUBSCRIBE 9 -#define SIP_MESSAGE 10 -#define SIP_UPDATE 11 -#define SIP_INFO 12 -#define SIP_CANCEL 13 -#define SIP_PUBLISH 14 -#define SIP_RESPONSE 100 - -#define RTP 1 -#define NO_RTP 0 -const struct cfsip_methods { - int id; - int need_rtp; /* when this is the 'primary' use for a pvt structure, does it need RTP? */ - char *text; -} sip_methods[] = { - { 0, RTP, "-UNKNOWN-" }, - { SIP_REGISTER, NO_RTP, "REGISTER" }, - { SIP_OPTIONS, NO_RTP, "OPTIONS" }, - { SIP_NOTIFY, NO_RTP, "NOTIFY" }, - { SIP_INVITE, RTP, "INVITE" }, - { SIP_ACK, NO_RTP, "ACK" }, - { SIP_PRACK, NO_RTP, "PRACK" }, - { SIP_BYE, NO_RTP, "BYE" }, - { SIP_REFER, NO_RTP, "REFER" }, - { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE" }, - { SIP_MESSAGE, NO_RTP, "MESSAGE" }, - { SIP_UPDATE, NO_RTP, "UPDATE" }, - { SIP_INFO, NO_RTP, "INFO" }, - { SIP_CANCEL, NO_RTP, "CANCEL" }, - { SIP_PUBLISH, NO_RTP, "PUBLISH" } -}; - -/* Structure for conversion between compressed SIP and "normal" SIP */ -static struct cfalias { - char *fullname; - char *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" }, -}; - -/* 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) - -/* List of well-known SIP options. If we get this in a require, - we should check the list and answer accordingly. */ -const struct cfsip_options { - int id; /* Bitmap ID */ - int supported; /* Supported by Asterisk ? */ - char *text; /* Text id, as in standard */ -} sip_options[] = { - /* Replaces: header for transfer */ - { SIP_OPT_REPLACES, SUPPORTED, "replaces" }, - /* RFC3262: PRACK 100% reliability */ - { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" }, - /* SIP Session Timers */ - { SIP_OPT_TIMER, NOT_SUPPORTED, "timer" }, - /* RFC3959: SIP Early session support */ - { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" }, - /* 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" }, - /* Not yet RFC, but registred with IANA */ - { SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp_anat" }, - /* RFC3329: Security agreement mechanism */ - { SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" }, - /* SIMPLE events: draft-ietf-simple-event-list-07.txt */ - { SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" }, - /* GRUU: Globally Routable User Agent URI's */ - { SIP_OPT_GRUU, NOT_SUPPORTED, "gruu" }, - /* Target-dialog: draft-ietf-sip-target-dialog-00.txt */ - { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "target-dialog" }, -}; - - -/* SIP Methods we support */ -#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY" - -/* SIP Extensions we support */ -#define SUPPORTED_EXTENSIONS "replaces" - -#define DEFAULT_SIP_PORT 5060 /* From RFC 3261 (former 2543) */ -#define SIP_MAX_PACKET 4096 /* Also from RFC 3261 (2543), should sub headers tho */ - -static char default_useragent[AST_MAX_EXTENSION] = DEFAULT_USERAGENT; - -#define DEFAULT_CONTEXT "default" -static char default_context[AST_MAX_CONTEXT] = DEFAULT_CONTEXT; - -static char default_language[MAX_LANGUAGE] = ""; - -#define DEFAULT_CALLERID "asterisk" -static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID; - -static char default_fromdomain[AST_MAX_EXTENSION] = ""; - -#define DEFAULT_NOTIFYMIME "application/simple-message-summary" -static char default_notifymime[AST_MAX_EXTENSION] = DEFAULT_NOTIFYMIME; - - -static int default_qualify = 0; /* Default Qualify= setting */ - -static struct ast_flags global_flags = {0}; /* global SIP_ flags */ -static struct ast_flags global_flags_page2 = {0}; /* more global SIP_ flags */ - -static int srvlookup = 0; /* SRV Lookup on or off. Default is off, RFC behavior is on */ - -static int pedanticsipchecking = 0; /* Extra checking ? Default off */ - -static int autocreatepeer = 0; /* Auto creation of peers at registration? Default off. */ - -static int relaxdtmf = 0; - -static int global_rtptimeout = 0; - -static int global_rtpholdtimeout = 0; - -static int global_rtpkeepalive = 0; - -static int global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT; -static int global_regattempts_max = DEFAULT_REGATTEMPTS_MAX; - -/* Object counters */ -static int suserobjs = 0; -static int ruserobjs = 0; -static int speerobjs = 0; -static int rpeerobjs = 0; -static int apeerobjs = 0; -static int regobjs = 0; - -static int global_allowguest = 1; /* allow unauthenticated users/peers to connect? */ - -#define DEFAULT_MWITIME 10 -static int global_mwitime = DEFAULT_MWITIME; /* Time between MWI checks for peers */ - -static int usecnt =0; -AST_MUTEX_DEFINE_STATIC(usecnt_lock); - - -/* Protect the interface list (of sip_pvt's) */ -AST_MUTEX_DEFINE_STATIC(iflock); - -/* 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); - -/* 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 restart_monitor(void); - -/* Codecs that we support by default: */ -static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263; -static int noncodeccapability = AST_RTP_DTMF; - -static struct in_addr __ourip; -static struct sockaddr_in outboundproxyip; -static int ourport; - -static int sipdebug = 0; -static struct sockaddr_in debugaddr; - -static int tos = 0; - -static int videosupport = 0; - -static int compactheaders = 0; /* send compact sip headers */ - -static int recordhistory = 0; /* Record SIP history. Off by default */ - -static char global_musicclass[MAX_MUSICCLASS] = ""; /* Global music on hold class */ -#define DEFAULT_REALM "asterisk" -static char global_realm[MAXHOSTNAMELEN] = DEFAULT_REALM; /* Default realm */ -static char regcontext[AST_MAX_CONTEXT] = ""; /* Context for auto-extensions */ - -/* Expire slowly */ -#define DEFAULT_EXPIRY 900 -static int expiry = DEFAULT_EXPIRY; - -static struct sched_context *sched; -static struct io_context *io; -/* The private structures of the sip channels are linked for - selecting outgoing channels */ - -#define SIP_MAX_HEADERS 64 -#define SIP_MAX_LINES 64 - -#define DEC_IN_USE 0 -#define INC_IN_USE 1 -#define DEC_OUT_USE 2 -#define INC_OUT_USE 3 - -static struct ast_codec_pref prefs; - - -/* 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 */ - char *header[SIP_MAX_HEADERS]; - int lines; /* SDP Content */ - char *line[SIP_MAX_LINES]; - char data[SIP_MAX_PACKET]; -}; - -struct sip_pkt; - -/* Parameters to the transmit_invite function */ -struct sip_invite_param { - char *distinctive_ring; - char *osptoken; - int addsipheaders; - char *vxml_url; - char *auth; - char *authheader; -}; - -struct sip_route { - struct sip_route *next; - char hop[0]; -}; - -/* sip_history: Structure for saving transactions within a SIP dialog */ -struct sip_history { - char event[80]; - struct sip_history *next; -}; - -/* sip_auth: Creadentials 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 */ -}; - -#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 */ -#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) /* Is this an outgoing call? */ -#define SIP_SELFDESTRUCT (1 << 14) -#define SIP_DYNAMIC (1 << 15) /* Is this a dynamic peer? */ -/* --- Choices for DTMF support in SIP channel */ -#define SIP_DTMF (3 << 16) /* three settings, uses two bits */ -#define SIP_DTMF_RFC2833 (0 << 16) /* RTP DTMF */ -#define SIP_DTMF_INBAND (1 << 16) /* Inband audio, only for ULAW/ALAW */ -#define SIP_DTMF_INFO (2 << 16) /* SIP Info messages */ -/* 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) -#define SIP_NAT_ROUTE (2 << 18) -#define SIP_NAT_ALWAYS (3 << 18) -/* re-INVITE related settings */ -#define SIP_REINVITE (3 << 20) /* two bits used */ -#define SIP_CAN_REINVITE (1 << 20) /* allow peers to be reinvited to send media directly p2p */ -#define SIP_REINVITE_UPDATE (2 << 20) /* use UPDATE (RFC3311) when reinviting this peer */ -/* "insecure" settings */ -#define SIP_INSECURE_PORT (1 << 22) /* don't require matching port for incoming requests */ -#define SIP_INSECURE_INVITE (1 << 23) /* don't require authentication for incoming INVITEs */ -/* Sending PROGRESS in-band settings */ -#define SIP_PROG_INBAND (3 << 24) /* three settings, uses two bits */ -#define SIP_PROG_INBAND_NEVER (0 << 24) -#define SIP_PROG_INBAND_NO (1 << 24) -#define SIP_PROG_INBAND_YES (2 << 24) -/* Open Settlement Protocol authentication */ -#define SIP_OSPAUTH (3 << 26) /* three settings, uses two bits */ -#define SIP_OSPAUTH_NO (0 << 26) -#define SIP_OSPAUTH_YES (1 << 26) -#define SIP_OSPAUTH_EXCLUSIVE (2 << 26) -/* Call states */ -#define SIP_CALL_ONHOLD (1 << 28) -#define SIP_CALL_LIMIT (1 << 29) - -/* a new page of flags for peer */ -#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0) -#define SIP_PAGE2_RTNOUPDATE (1 << 1) -#define SIP_PAGE2_RTAUTOCLEAR (1 << 2) -#define SIP_PAGE2_RTIGNOREREGEXPIRE (1 << 3) - -static int global_rtautoclear = 120; - -/* sip_pvt: PVT structures are used for each SIP conversation, ie. a call */ -static struct sip_pvt { - ast_mutex_t lock; /* Channel private lock */ - int method; /* SIP method of this packet */ - char callid[80]; /* Global CallID */ - char randdata[80]; /* Random data */ - struct ast_codec_pref prefs; /* codec prefs */ - 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 */ - unsigned int flags; /* SIP_ flags */ - unsigned int sipoptions; /* Supported SIP sipoptions on the other end */ - 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; - int callingpres; /* Calling presentation */ - int authtries; /* Times we've tried to authenticate */ - int expiry; /* How long we take to expire */ - int branch; /* One random number */ - int tag; /* Another random number */ - 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 */ - int redircodecs; /* Redirect codecs */ - struct sockaddr_in recv; /* Received as */ - struct in_addr ourip; /* Our IP */ - struct ast_channel *owner; /* Who owns us */ - char exten[AST_MAX_EXTENSION]; /* Extension where to start */ - char refer_to[AST_MAX_EXTENSION]; /* Place to store REFER-TO extension */ - char referred_by[AST_MAX_EXTENSION]; /* Place to store REFERRED-BY extension */ - char refer_contact[AST_MAX_EXTENSION]; /* Place to store Contact info from a REFER extension */ - struct sip_pvt *refer_call; /* Call we are referring */ - struct sip_route *route; /* Head of linked list of routing steps (fm Record-Route) */ - int route_persistant; /* Is this the "real" route? */ - char from[256]; /* The From: header */ - char useragent[256]; /* User agent in SIP request */ - char context[AST_MAX_CONTEXT]; /* Context for this call */ - char fromdomain[MAXHOSTNAMELEN]; /* Domain to show in the from field */ - char fromuser[AST_MAX_EXTENSION]; /* User to show in the user field */ - char fromname[AST_MAX_EXTENSION]; /* Name to show in the user field */ - char tohost[MAXHOSTNAMELEN]; /* Host we should put in the "to" field */ - char language[MAX_LANGUAGE]; /* Default language for this call */ - char musicclass[MAX_MUSICCLASS]; /* Music on Hold class */ - char rdnis[256]; /* Referring DNIS */ - char theirtag[256]; /* Their tag */ - char username[256]; /* [user] name */ - char peername[256]; /* [peer] name, not set if [user] */ - char authname[256]; /* Who we use for authentication */ - char uri[256]; /* Original requested URI */ - char okcontacturi[256]; /* URI from the 200 OK on INVITE */ - char peersecret[256]; /* Password */ - char peermd5secret[256]; - struct sip_auth *peerauth; /* Realm authentication */ - char cid_num[256]; /* Caller*ID */ - char cid_name[256]; /* Caller*ID */ - char via[256]; /* Via: header */ - char fullcontact[128]; /* The Contact: that the UA registers with us */ - char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */ - char our_contact[256]; /* Our contact header */ - char realm[MAXHOSTNAMELEN]; /* Authorization realm */ - char nonce[256]; /* Authorization nonce */ - char opaque[256]; /* Opaque nonsense */ - char qop[80]; /* Quality of Protection, since SIP wasn't complicated enough yet. */ - char domain[MAXHOSTNAMELEN]; /* Authorization domain */ - char lastmsg[256]; /* Last Message sent/received */ - int amaflags; /* AMA Flags */ - int pendinginvite; /* Any pending invite */ -#ifdef OSP_SUPPORT - int osphandle; /* OSP Handle for call */ - time_t ospstart; /* OSP Start time */ -#endif - struct sip_request initreq; /* Initial request */ - - int maxtime; /* Max time for first response */ - int maxforwards; /* keep the max-forwards info */ - int initid; /* Auto-congest ID if appropriate */ - int autokillid; /* Auto-kill ID */ - time_t lastrtprx; /* Last RTP received */ - time_t lastrtptx; /* Last RTP sent */ - int rtptimeout; /* RTP timeout time */ - int rtpholdtimeout; /* RTP timeout when on hold */ - int rtpkeepalive; /* Send RTP packets for keepalive */ - - int subscribed; /* Is this call a subscription? */ - int stateid; - int dialogver; - - struct ast_dsp *vad; /* Voice Activation Detection dsp */ - - struct sip_peer *peerpoke; /* If this calls is to poke a peer, which one */ - struct sip_registry *registry; /* If this is a REGISTER call, 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 *history; /* History of this SIP dialog */ - struct ast_variable *chanvars; /* Channel variables to set for call */ - struct sip_pvt *next; /* Next call in chain */ -} *iflist = NULL; - -#define FLAG_RESPONSE (1 << 0) -#define FLAG_FATAL (1 << 1) - -/* sip packet - read in sipsock_read, transmitted in send_request */ -struct sip_pkt { - struct sip_pkt *next; /* Next packet */ - int retrans; /* Retransmission number */ - int seqno; /* Sequence number */ - unsigned int flags; /* non-zero if this is a response packet (e.g. 200 OK) */ - struct sip_pvt *owner; /* Owner call */ - int retransid; /* Retransmission ID */ - int packetlen; /* Length of packet */ - char data[0]; -}; - -/* 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 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 musicclass[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 flags; /* SIP flags */ - unsigned int sipoptions; /* Supported SIP options */ - struct ast_flags flags_page2; /* SIP_PAGE2 flags */ - int amaflags; /* AMA flags for billing */ - int callingpres; /* Calling id presentation */ - int capability; /* Codec capability */ - int inUse; /* Number of calls in use */ - int incominglimit; /* Limit of incoming calls */ - int outUse; /* disabled */ - int outgoinglimit; /* disabled */ - struct ast_ha *ha; /* ACL setting */ - struct ast_variable *chanvars; /* Variables to set for channel created by user */ -}; - -/* Structure for SIP peer data, we place calls to peers if registred or fixed IP address (host) */ -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 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 registred 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 incominglimit; /* Limit of incoming calls */ - int outUse; /* disabled */ - int outgoinglimit; /* disabled */ - char mailbox[AST_MAX_EXTENSION]; /* Mailbox setting for MWI checks */ - char language[MAX_LANGUAGE]; /* Default language for prompts */ - char musicclass[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 flags; /* SIP flags */ - unsigned int sipoptions; /* Supported SIP options */ - struct ast_flags flags_page2; /* SIP_PAGE2 flags */ - int expire; /* When to expire this peer registration */ - int expiry; /* Duration of 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 ast_dnsmgr_entry *dnsmgr;/* DNS refresh manager for peer */ - struct sockaddr_in addr; /* IP address of peer */ - struct in_addr mask; - - /* 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_variable *chanvars; /* Variables to set for channel created by user */ - int lastmsg; -}; - -AST_MUTEX_DEFINE_STATIC(sip_reload_lock); -static int sip_reloading = 0; - -/* States for outbound registrations (with register= lines in sip.conf */ -#define REG_STATE_UNREGISTERED 0 -#define REG_STATE_REGSENT 1 -#define REG_STATE_AUTHSENT 2 -#define REG_STATE_REGISTERED 3 -#define REG_STATE_REJECTED 4 -#define REG_STATE_TIMEOUT 5 -#define REG_STATE_NOAUTH 6 -#define REG_STATE_FAILED 7 - - -/* sip_registry: Registrations with other SIP proxies */ -struct sip_registry { - ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1); - int portno; /* Optional port override */ - char username[80]; /* Who we are registering as */ - char authuser[80]; /* Who we *authenticate* as */ - char hostname[MAXHOSTNAMELEN]; /* Domain or host we register to */ - char secret[80]; /* Password or key name in []'s */ - char md5secret[80]; - char contact[256]; /* Contact extension */ - char random[80]; - 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 call" in progress */ - int regstate; /* Registration state (see above) */ - int callid_valid; /* 0 means we haven't chosen callid for this registry yet. */ - char callid[80]; /* Global CallID for this registry */ - unsigned int ocseq; /* Sequence number we got to for REGISTERs for this registry */ - struct sockaddr_in us; /* Who the server thinks we are */ - - /* Saved headers */ - char realm[MAXHOSTNAMELEN]; /* Authorization realm */ - char nonce[256]; /* Authorization nonce */ - char domain[MAXHOSTNAMELEN]; /* Authorization domain */ - char opaque[256]; /* Opaque nonsense */ - char qop[80]; /* Quality of Protection. */ - - char lastmsg[256]; /* Last Message sent/received */ -}; - -/*--- The user list: Users and friends ---*/ -static struct ast_user_list { - ASTOBJ_CONTAINER_COMPONENTS(struct sip_user); -} userl; - -/*--- The peer list: Peers and Friends ---*/ -static struct ast_peer_list { - ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer); -} peerl; - -/*--- The register list: Other SIP proxys we register with and call ---*/ -static struct ast_register_list { - ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry); - int recheck; -} regl; - - -static int __sip_do_register(struct sip_registry *r); - -static int sipsock = -1; - - -static struct sockaddr_in bindaddr; -static struct sockaddr_in externip; -static char externhost[MAXHOSTNAMELEN] = ""; -static time_t externexpire = 0; -static int externrefresh = 10; -static struct ast_ha *localaddr; - -/* The list of manual NOTIFY types we know how to send */ -struct ast_config *notify_types; - -static struct sip_auth *authl; /* Authentication list */ - - -static struct ast_frame *sip_read(struct ast_channel *ast); -static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req); -static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans); -static int transmit_response_with_unsupported(struct sip_pvt *p, char *msg, struct sip_request *req, char *unsupported); -static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_request *req, char *rand, int reliable, char *header, int stale); -static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch); -static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch); -static int transmit_invite(struct sip_pvt *p, int sipmethod, int sendsdp, struct sip_invite_param *options, int init); -static int transmit_reinvite_with_sdp(struct sip_pvt *p); -static int transmit_info_with_digit(struct sip_pvt *p, char digit); -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 sip_sipredirect(struct sip_pvt *p, const char *dest); -static struct sip_peer *temp_peer(const char *name); -static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, int sipmethod, int init); -static void free_old_route(struct sip_route *route); -static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len); -static int update_user_counter(struct sip_pvt *fup, int event); -static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int realtime); -static struct sip_user *build_user(const char *name, struct ast_variable *v, int realtime); -static int sip_do_reload(void); -static int expire_register(void *data); -static int callevents = 0; - -static struct ast_channel *sip_request(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); -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(struct ast_channel *ast, char digit); -static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */ -static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); /* Add realm authentication in list */ -static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, char *realm); /* Find authentication for a specific realm */ -static void append_date(struct sip_request *req); /* Append date to SIP packet */ - -/* Definition of this channel for channel registration */ -static const struct ast_channel_tech sip_tech = { - .type = channeltype, - .description = "Session Initiation Protocol (SIP)", - .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1), - .properties = AST_CHAN_TP_WANTSJITTER, - .requester = sip_request, - .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 = sip_senddigit, - .bridge = ast_rtp_bridge, - .send_text = sip_sendtext, -}; - -/*--- find_sip_method: Find SIP method from header */ -int find_sip_method(char *msg) -{ - int i, res = 0; - /* Strictly speaking, SIP methods are case SENSITIVE, but we don't check */ - for (i=1;(i < (sizeof(sip_methods) / sizeof(sip_methods[0]))) && !res; i++) { - if (!strcasecmp(sip_methods[i].text, msg)) - res = sip_methods[i].id; - } - return res; -} - -/* - * If there is a string in , strip everything around and return - * the content. Otherwise return the original argument. - */ -static char *get_in_brackets(char *c) -{ - char *n = strchr(c, '<'); - - if (n) { - c = n + 1; - n = strchr(c, '>'); - /* Lose the part after the > */ - if (n) - *n = '\0'; - } - return c; -} - -/*--- parse_sip_options: Parse supported header in incoming packet */ -unsigned int parse_sip_options(struct sip_pvt *pvt, char *supported) -{ - char *next = NULL; - char *sep = NULL; - char *temp = ast_strdupa(supported); - int i; - unsigned int profile = 0; - - if (!supported || ast_strlen_zero(supported) ) - return 0; - - if (option_debug > 2 && sipdebug) - ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported); - - next = temp; - while (next) { - char res=0; - if ( (sep = strchr(next, ',')) != NULL) { - *sep = '\0'; - sep++; - } - while (*next == ' ') /* Skip spaces */ - 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]))) && !res; i++) { - if (!strcasecmp(next, sip_options[i].text)) { - profile |= sip_options[i].id; - res = 1; - if (option_debug > 2 && sipdebug) - ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next); - } - } - if (!res) - if (option_debug > 2 && sipdebug) - ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next); - next = sep; - } - if (pvt) - pvt->sipoptions = profile; - - ast_log(LOG_DEBUG, "* SIP extension value: %d for call %s\n", profile, pvt->callid); - return profile; -} - -/*--- sip_debug_test_addr: See if we pass debug IP filter */ -static inline int sip_debug_test_addr(struct sockaddr_in *addr) -{ - if (sipdebug == 0) - 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; -} - -/*--- sip_debug_test_pvt: Test PVT for debugging output */ -static inline int sip_debug_test_pvt(struct sip_pvt *p) -{ - if (sipdebug == 0) - return 0; - return sip_debug_test_addr(((ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) ? &p->recv : &p->sa)); -} - - -/*--- __sip_xmit: Transmit SIP message ---*/ -static int __sip_xmit(struct sip_pvt *p, char *data, int len) -{ - int res; - char iabuf[INET_ADDRSTRLEN]; - - if (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) - res=sendto(sipsock, data, len, 0, (struct sockaddr *)&p->recv, sizeof(struct sockaddr_in)); - else - res=sendto(sipsock, data, len, 0, (struct sockaddr *)&p->sa, sizeof(struct sockaddr_in)); - if (res != len) { - ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s returned %d: %s\n", data, len, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), res, strerror(errno)); - } - return res; -} - -static void sip_destroy(struct sip_pvt *p); - -/*--- build_via: Build a Via header for a request ---*/ -static void build_via(struct sip_pvt *p, char *buf, int len) -{ - char iabuf[INET_ADDRSTRLEN]; - - /* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */ - if (ast_test_flag(p, SIP_NAT) & SIP_NAT_RFC3581) - snprintf(buf, len, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x;rport", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport, p->branch); - else /* Work around buggy UNIDEN UIP200 firmware */ - snprintf(buf, len, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport, p->branch); -} - -/*--- ast_sip_ouraddrfor: NAT fix - decide which IP address to use for ASterisk server? ---*/ -/* Only used for outbound registrations */ -static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us) -{ - /* - * Using the localaddr structure built up with localnet statements - * apply it to their address to see if we need to substitute our - * externip or can get away with our internal bindaddr - */ - struct sockaddr_in theirs; - theirs.sin_addr = *them; - if (localaddr && externip.sin_addr.s_addr && - ast_apply_ha(localaddr, &theirs)) { - char iabuf[INET_ADDRSTRLEN]; - if (externexpire && (time(NULL) >= externexpire)) { - struct ast_hostent ahp; - struct hostent *hp; - time(&externexpire); - externexpire += 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); - } - memcpy(us, &externip.sin_addr, sizeof(struct in_addr)); - ast_inet_ntoa(iabuf, sizeof(iabuf), *(struct in_addr *)&them->s_addr); - ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n", iabuf); - } - else if (bindaddr.sin_addr.s_addr) - memcpy(us, &bindaddr.sin_addr, sizeof(struct in_addr)); - else - return ast_ouraddrfor(them, us); - return 0; -} - -/*--- append_history: Append to SIP dialog history */ -/* Always returns 0 */ -static int append_history(struct sip_pvt *p, char *event, char *data) -{ - struct sip_history *hist, *prev; - char *c; - - if (!recordhistory) - return 0; - if(!(hist = malloc(sizeof(struct sip_history)))) { - ast_log(LOG_WARNING, "Can't allocate memory for history"); - return 0; - } - memset(hist, 0, sizeof(struct sip_history)); - snprintf(hist->event, sizeof(hist->event), "%-15s %s", event, data); - /* Trim up nicely */ - c = hist->event; - while(*c) { - if ((*c == '\r') || (*c == '\n')) { - *c = '\0'; - break; - } - c++; - } - /* Enqueue into history */ - prev = p->history; - if (prev) { - while(prev->next) - prev = prev->next; - prev->next = hist; - } else { - p->history = hist; - } - return 0; -} - -/*--- retrans_pkt: Retransmit SIP message if no answer ---*/ -static int retrans_pkt(void *data) -{ - struct sip_pkt *pkt=data, *prev, *cur; - int res = 0; - char iabuf[INET_ADDRSTRLEN]; - ast_mutex_lock(&pkt->owner->lock); - if (pkt->retrans < MAX_RETRANS) { - pkt->retrans++; - if (sip_debug_test_pvt(pkt->owner)) { - if (ast_test_flag(pkt->owner, SIP_NAT) & SIP_NAT_ROUTE) - ast_verbose("Retransmitting #%d (NAT) to %s:%d:\n%s\n---\n", pkt->retrans, ast_inet_ntoa(iabuf, sizeof(iabuf), pkt->owner->recv.sin_addr), ntohs(pkt->owner->recv.sin_port), pkt->data); - else - ast_verbose("Retransmitting #%d (no NAT) to %s:%d:\n%s\n---\n", pkt->retrans, ast_inet_ntoa(iabuf, sizeof(iabuf), pkt->owner->sa.sin_addr), ntohs(pkt->owner->sa.sin_port), pkt->data); - } - append_history(pkt->owner, "ReTx", pkt->data); - __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); - res = 1; - } else { - ast_log(LOG_WARNING, "Maximum retries exceeded on call %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request"); - append_history(pkt->owner, "MaxRetries", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)"); - pkt->retransid = -1; - if (ast_test_flag(pkt, FLAG_FATAL)) { - while(pkt->owner->owner && ast_mutex_trylock(&pkt->owner->owner->lock)) { - ast_mutex_unlock(&pkt->owner->lock); - usleep(1); - ast_mutex_lock(&pkt->owner->lock); - } - if (pkt->owner->owner) { - ast_set_flag(pkt->owner, SIP_ALREADYGONE); - ast_queue_hangup(pkt->owner->owner); - ast_mutex_unlock(&pkt->owner->owner->lock); - } else { - /* If no owner, destroy now */ - ast_set_flag(pkt->owner, SIP_NEEDDESTROY); - } - } - /* In any case, go ahead and remove the packet */ - prev = NULL; - cur = pkt->owner->packets; - while(cur) { - if (cur == pkt) - break; - prev = cur; - cur = cur->next; - } - 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 res; -} - -/*--- __sip_reliable_xmit: transmit packet with retransmits ---*/ -static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal) -{ - struct sip_pkt *pkt; - pkt = malloc(sizeof(struct sip_pkt) + len + 1); - if (!pkt) - return -1; - memset(pkt, 0, sizeof(struct sip_pkt)); - memcpy(pkt->data, data, len); - pkt->packetlen = len; - pkt->next = p->packets; - pkt->owner = p; - pkt->seqno = seqno; - pkt->flags = resp; - pkt->data[len] = '\0'; - if (fatal) - ast_set_flag(pkt, FLAG_FATAL); - /* Schedule retransmission */ - pkt->retransid = ast_sched_add(sched, DEFAULT_RETRANS, retrans_pkt, pkt); - pkt->next = p->packets; - p->packets = pkt; - __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); - if (!strncasecmp(pkt->data, "INVITE", 6)) { - /* Note this is a pending invite */ - p->pendinginvite = seqno; - } - return 0; -} - -/*--- __sip_autodestruct: Kill a call (called by scheduler) ---*/ -static int __sip_autodestruct(void *data) -{ - struct sip_pvt *p = data; - - p->autokillid = -1; - ast_log(LOG_DEBUG, "Auto destroying call '%s'\n", p->callid); - append_history(p, "AutoDestroy", ""); - if (p->owner) { - ast_log(LOG_WARNING, "Autodestruct on call '%s' with owner in place\n", p->callid); - ast_queue_hangup(p->owner); - } else { - sip_destroy(p); - } - return 0; -} - -/*--- sip_scheddestroy: Schedule destruction of SIP call ---*/ -static int sip_scheddestroy(struct sip_pvt *p, int ms) -{ - char tmp[80]; - if (sip_debug_test_pvt(p)) - ast_verbose("Scheduling destruction of call '%s' in %d ms\n", p->callid, ms); - if (recordhistory) { - snprintf(tmp, sizeof(tmp), "%d ms", ms); - append_history(p, "SchedDestroy", tmp); - } - if (p->autokillid > -1) - ast_sched_del(sched, p->autokillid); - p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, p); - return 0; -} - -/*--- sip_cancel_destroy: Cancel destruction of SIP call ---*/ -static int sip_cancel_destroy(struct sip_pvt *p) -{ - if (p->autokillid > -1) - ast_sched_del(sched, p->autokillid); - append_history(p, "CancelDestroy", ""); - p->autokillid = -1; - return 0; -} - -/*--- __sip_ack: Acknowledges receipt of a packet and stops retransmission ---*/ -static int __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod) -{ - struct sip_pkt *cur, *prev = NULL; - int res = -1; - int resetinvite = 0; - /* Just in case... */ - char *msg; - - msg = sip_methods[sipmethod].text; - - cur = p->packets; - while(cur) { - 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)) { - ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite); - p->pendinginvite = 0; - resetinvite = 1; - } - /* this is our baby */ - if (prev) - prev->next = cur->next; - else - p->packets = cur->next; - if (cur->retransid > -1) - ast_sched_del(sched, cur->retransid); - free(cur); - res = 0; - break; - } - prev = cur; - cur = cur->next; - } - ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found"); - return res; -} - -/* Pretend to ack all packets */ -static int __sip_pretend_ack(struct sip_pvt *p) -{ - char method[128]=""; - struct sip_pkt *cur=NULL; - char *c; - while(p->packets) { - if (cur == p->packets) { - ast_log(LOG_WARNING, "Have a packet that doesn't want to give up!\n"); - return -1; - } - cur = p->packets; - ast_copy_string(method, p->packets->data, sizeof(method)); - c = ast_skip_blanks(method); /* XXX what ? */ - *c = '\0'; - __sip_ack(p, p->packets->seqno, (ast_test_flag(p->packets, FLAG_RESPONSE)), find_sip_method(method)); - } - return 0; -} - -/*--- __sip_semi_ack: 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; - char *msg = sip_methods[sipmethod].text; - - cur = p->packets; - while(cur) { - 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)))) { - /* this is our baby */ - if (cur->retransid > -1) - ast_sched_del(sched, cur->retransid); - cur->retransid = -1; - res = 0; - break; - } - cur = cur->next; - } - ast_log(LOG_DEBUG, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found"); - return res; -} - -static void parse(struct sip_request *req); -static char *get_header(struct sip_request *req, char *name); -static void copy_request(struct sip_request *dst,struct sip_request *src); - -/*--- parse_copy: Copy SIP request, parse it */ -static void parse_copy(struct sip_request *dst, struct sip_request *src) -{ - memset(dst, 0, sizeof(*dst)); - memcpy(dst->data, src->data, sizeof(dst->data)); - dst->len = src->len; - parse(dst); -} - -/*--- send_response: Transmit response on SIP request---*/ -static int send_response(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno) -{ - int res; - char iabuf[INET_ADDRSTRLEN]; - struct sip_request tmp; - char tmpmsg[80]; - if (sip_debug_test_pvt(p)) { - if (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) - ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(iabuf, sizeof(iabuf), 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(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port), req->data); - } - if (reliable) { - if (recordhistory) { - parse_copy(&tmp, req); - snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq")); - append_history(p, "TxRespRel", tmpmsg); - } - res = __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable > 1)); - } else { - if (recordhistory) { - parse_copy(&tmp, req); - snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq")); - append_history(p, "TxResp", tmpmsg); - } - res = __sip_xmit(p, req->data, req->len); - } - if (res > 0) - return 0; - return res; -} - -/*--- send_request: Send SIP Request to the other part of the dialogue ---*/ -static int send_request(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno) -{ - int res; - char iabuf[INET_ADDRSTRLEN]; - struct sip_request tmp; - char tmpmsg[80]; - - if (sip_debug_test_pvt(p)) { - if (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) - ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(iabuf, sizeof(iabuf), 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(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port), req->data); - } - if (reliable) { - if (recordhistory) { - parse_copy(&tmp, req); - snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq")); - append_history(p, "TxReqRel", tmpmsg); - } - res = __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1)); - } else { - if (recordhistory) { - parse_copy(&tmp, req); - snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq")); - append_history(p, "TxReq", tmpmsg); - } - res = __sip_xmit(p, req->data, req->len); - } - return res; -} - -/*--- url_decode: Decode SIP URL (overwrite the string) ---*/ -static void url_decode(char *s) -{ - char *o; - unsigned int tmp; - - for (o = s; *s; s++, o++) { - if (*s == '%' && strlen(s) > 2 && sscanf(s + 1, "%2x", &tmp) == 1) { - /* have '%', two chars and correct parsing */ - *o = tmp; - s += 2; /* Will be incremented once more when we break out */ - } else /* all other cases, just copy */ - *o = *s; - } - *o = '\0'; -} - -/*--- ditch_braces: Pick out text in braces from character string ---*/ -static char *ditch_braces(char *tmp) -{ - char *c = tmp; - char *n; - char *q; - if ((q = strchr(tmp, '"')) ) { - c = q + 1; - if ((q = strchr(c, '"')) ) - c = q + 1; - else { - ast_log(LOG_WARNING, "No closing quote in '%s'\n", tmp); - c = tmp; - } - } - if ((n = strchr(c, '<')) ) { - c = n + 1; - while(*c && *c != '>') c++; - if (*c != '>') { - ast_log(LOG_WARNING, "No closing brace in '%s'\n", tmp); - } else { - *c = '\0'; - } - return n+1; - } - return c; -} - -/*--- sip_sendtext: Send SIP MESSAGE text within a call ---*/ -/* Called from PBX core text message functions */ -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 (!text || 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; -} - -/*--- realtime_update_peer: Update peer object in realtime storage ---*/ -static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, int expirey) -{ - char port[10] = ""; - char ipaddr[20] = ""; - char regseconds[20] = "0"; - - if (expirey) { /* Registration */ - time_t nowtime; - time(&nowtime); - nowtime += expirey; - snprintf(regseconds, sizeof(regseconds), "%ld", nowtime); /* Expiration time */ - ast_inet_ntoa(ipaddr, sizeof(ipaddr), sin->sin_addr); - snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port)); - } - ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr, "port", port, "regseconds", regseconds, "username", username, NULL); -} - -/*--- register_peer_exten: Automatically add peer extension to dial plan ---*/ -static void register_peer_exten(struct sip_peer *peer, int onoff) -{ - char multi[256]=""; - char *stringp, *ext; - if (!ast_strlen_zero(regcontext)) { - ast_copy_string(multi, ast_strlen_zero(peer->regexten) ? peer->name : peer->regexten, sizeof(multi)); - stringp = multi; - while((ext = strsep(&stringp, "&"))) { - if (onoff) - ast_add_extension(regcontext, 1, ext, 1, NULL, NULL, "Noop", strdup(peer->name), free, channeltype); - else - ast_context_remove_extension(regcontext, ext, 1, NULL); - } - } -} - -/*--- sip_destroy_peer: Destroy peer object from memory */ -static void sip_destroy_peer(struct sip_peer *peer) -{ - /* Delete it, it needs to disappear */ - if (peer->call) - sip_destroy(peer->call); - if (peer->chanvars) { - ast_variables_destroy(peer->chanvars); - peer->chanvars = NULL; - } - if (peer->expire > -1) - ast_sched_del(sched, peer->expire); - if (peer->pokeexpire > -1) - ast_sched_del(sched, peer->pokeexpire); - register_peer_exten(peer, 0); - ast_free_ha(peer->ha); - if (ast_test_flag(peer, SIP_SELFDESTRUCT)) - apeerobjs--; - else if (ast_test_flag(peer, SIP_REALTIME)) - rpeerobjs--; - else - speerobjs--; - clear_realm_authentication(peer->auth); - peer->auth = (struct sip_auth *) NULL; - if (peer->dnsmgr) - ast_dnsmgr_release(peer->dnsmgr); - free(peer); -} - -/*--- update_peer: Update peer data in database (if used) ---*/ -static void update_peer(struct sip_peer *p, int expiry) -{ - if (!ast_test_flag((&global_flags_page2), SIP_PAGE2_RTNOUPDATE) && - (ast_test_flag(p, SIP_REALTIME) || - ast_test_flag(&(p->flags_page2), SIP_PAGE2_RTCACHEFRIENDS))) { - realtime_update_peer(p->name, &p->addr, p->username, expiry); - } -} - - -/*--- realtime_peer: Get peer from realtime storage ---*/ -/* Checks the "sippeers" realtime family from extconfig.conf */ -static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin) -{ - struct sip_peer *peer=NULL; - struct ast_variable *var; - struct ast_variable *tmp; - char *newpeername = (char *) peername; - char iabuf[80] = ""; - - /* First check on peer name */ - if (newpeername) - var = ast_load_realtime("sippeers", "name", peername, NULL); - else if (sin) { /* Then check on IP address */ - ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr); - var = ast_load_realtime("sippeers", "ipaddr", iabuf, NULL); - } else - return NULL; - - if (!var) - return NULL; - - tmp = var; - /* If this is type=user, then skip this object. */ - while(tmp) { - 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; - } - tmp = tmp->next; - } - - if (!newpeername) { /* Did not find peer in realtime */ - ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", iabuf); - ast_variables_destroy(var); - return (struct sip_peer *) NULL; - } - - /* Peer found in realtime, now build it in memory */ - peer = build_peer(newpeername, var, !ast_test_flag((&global_flags_page2), SIP_PAGE2_RTCACHEFRIENDS)); - - if (!peer) { - ast_variables_destroy(var); - return (struct sip_peer *) NULL; - } - if (ast_test_flag((&global_flags_page2), SIP_PAGE2_RTCACHEFRIENDS)) { - /* Cache peer */ - ast_copy_flags((&peer->flags_page2),(&global_flags_page2), SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS); - if (ast_test_flag((&global_flags_page2), SIP_PAGE2_RTAUTOCLEAR)) { - if (peer->expire > -1) { - ast_sched_del(sched, peer->expire); - } - peer->expire = ast_sched_add(sched, (global_rtautoclear) * 1000, expire_register, (void *)peer); - } - ASTOBJ_CONTAINER_LINK(&peerl,peer); - } else { - ast_set_flag(peer, SIP_REALTIME); - } - ast_variables_destroy(var); - return peer; -} - -/*--- sip_addrcmp: 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, SIP_INSECURE_PORT) && - (p->addr.sin_addr.s_addr == sin->sin_addr.s_addr))); -} - -/*--- find_peer: 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) -{ - 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) { - p = realtime_peer(peer, sin); - } - - return p; -} - -/*--- sip_destroy_user: Remove user object from in-memory storage ---*/ -static void sip_destroy_user(struct sip_user *user) -{ - ast_free_ha(user->ha); - if (user->chanvars) { - ast_variables_destroy(user->chanvars); - user->chanvars = NULL; - } - if (ast_test_flag(user, SIP_REALTIME)) - ruserobjs--; - else - suserobjs--; - free(user); -} - -/*--- realtime_user: 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; - - tmp = var; - while (tmp) { - if (!strcasecmp(tmp->name, "type") && - !strcasecmp(tmp->value, "peer")) { - ast_variables_destroy(var); - return NULL; - } - tmp = tmp->next; - } - - - - user = build_user(username, var, !ast_test_flag((&global_flags_page2), SIP_PAGE2_RTCACHEFRIENDS)); - - if (!user) { /* No user found */ - ast_variables_destroy(var); - return NULL; - } - - if (ast_test_flag((&global_flags_page2), SIP_PAGE2_RTCACHEFRIENDS)) { - ast_set_flag((&user->flags_page2), SIP_PAGE2_RTCACHEFRIENDS); - suserobjs++; - ASTOBJ_CONTAINER_LINK(&userl,user); - } else { - /* Move counter from s to r... */ - suserobjs--; - ruserobjs++; - ast_set_flag(user, SIP_REALTIME); - } - ast_variables_destroy(var); - return user; -} - -/*--- find_user: 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 = NULL; - u = ASTOBJ_CONTAINER_FIND(&userl,name); - if (!u && realtime) { - u = realtime_user(name); - } - return u; -} - -/*--- create_addr_from_peer: create address structure from peer reference ---*/ -static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer) -{ - char *callhost; - - if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) && - (!peer->maxms || ((peer->lastms >= 0) && (peer->lastms <= peer->maxms)))) { - if (peer->addr.sin_addr.s_addr) { - r->sa.sin_addr = peer->addr.sin_addr; - r->sa.sin_port = peer->addr.sin_port; - } else { - r->sa.sin_addr = peer->defaddr.sin_addr; - r->sa.sin_port = peer->defaddr.sin_port; - } - memcpy(&r->recv, &r->sa, sizeof(r->recv)); - } else { - return -1; - } - - ast_copy_flags(r, peer, - SIP_PROMISCREDIR | SIP_USEREQPHONE | SIP_DTMF | SIP_NAT | SIP_REINVITE | - SIP_INSECURE_PORT | SIP_INSECURE_INVITE); - r->capability = peer->capability; - if (r->rtp) { - ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", (ast_test_flag(r, SIP_NAT) & SIP_NAT_ROUTE)); - ast_rtp_setnat(r->rtp, (ast_test_flag(r, SIP_NAT) & SIP_NAT_ROUTE)); - } - if (r->vrtp) { - ast_log(LOG_DEBUG, "Setting NAT on VRTP to %d\n", (ast_test_flag(r, SIP_NAT) & SIP_NAT_ROUTE)); - ast_rtp_setnat(r->vrtp, (ast_test_flag(r, SIP_NAT) & SIP_NAT_ROUTE)); - } - ast_copy_string(r->peername, peer->username, sizeof(r->peername)); - ast_copy_string(r->authname, peer->username, sizeof(r->authname)); - ast_copy_string(r->username, peer->username, sizeof(r->username)); - ast_copy_string(r->peersecret, peer->secret, sizeof(r->peersecret)); - ast_copy_string(r->peermd5secret, peer->md5secret, sizeof(r->peermd5secret)); - ast_copy_string(r->tohost, peer->tohost, sizeof(r->tohost)); - ast_copy_string(r->fullcontact, peer->fullcontact, sizeof(r->fullcontact)); - if (!r->initreq.headers && !ast_strlen_zero(peer->fromdomain)) { - if ((callhost = strchr(r->callid, '@'))) { - strncpy(callhost + 1, peer->fromdomain, sizeof(r->callid) - (callhost - r->callid) - 2); - } - } - if (ast_strlen_zero(r->tohost)) { - if (peer->addr.sin_addr.s_addr) - ast_inet_ntoa(r->tohost, sizeof(r->tohost), peer->addr.sin_addr); - else - ast_inet_ntoa(r->tohost, sizeof(r->tohost), peer->defaddr.sin_addr); - } - if (!ast_strlen_zero(peer->fromdomain)) - ast_copy_string(r->fromdomain, peer->fromdomain, sizeof(r->fromdomain)); - if (!ast_strlen_zero(peer->fromuser)) - ast_copy_string(r->fromuser, peer->fromuser, sizeof(r->fromuser)); - r->maxtime = peer->maxms; - r->callgroup = peer->callgroup; - r->pickupgroup = peer->pickupgroup; - if (ast_test_flag(r, SIP_DTMF) == SIP_DTMF_RFC2833) - r->noncodeccapability |= AST_RTP_DTMF; - else - r->noncodeccapability &= ~AST_RTP_DTMF; - ast_copy_string(r->context, peer->context,sizeof(r->context)); - r->rtptimeout = peer->rtptimeout; - r->rtpholdtimeout = peer->rtpholdtimeout; - r->rtpkeepalive = peer->rtpkeepalive; - - return 0; -} - -/*--- create_addr: 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 *r, char *opeer) -{ - struct hostent *hp; - struct ast_hostent ahp; - struct sip_peer *p; - int found=0; - 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'; - port++; - } - r->sa.sin_family = AF_INET; - p = find_peer(peer, NULL, 1); - - if (p) { - found++; - if (create_addr_from_peer(r, p)) - ASTOBJ_UNREF(p, sip_destroy_peer); - } - if (!p) { - if (found) - return -1; - - hostn = peer; - if (port) - portno = atoi(port); - else - portno = DEFAULT_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_copy_string(r->tohost, peer, sizeof(r->tohost)); - memcpy(&r->sa.sin_addr, hp->h_addr, sizeof(r->sa.sin_addr)); - r->sa.sin_port = htons(portno); - memcpy(&r->recv, &r->sa, sizeof(r->recv)); - return 0; - } else { - ast_log(LOG_WARNING, "No such host: %s\n", peer); - return -1; - } - } else { - ASTOBJ_UNREF(p, sip_destroy_peer); - return 0; - } -} - -/*--- auto_congest: Scheduled congestion on a call ---*/ -static int auto_congest(void *nothing) -{ - struct sip_pvt *p = nothing; - ast_mutex_lock(&p->lock); - p->initid = -1; - if (p->owner) { - if (!ast_mutex_trylock(&p->owner->lock)) { - ast_log(LOG_NOTICE, "Auto-congesting %s\n", p->owner->name); - ast_queue_control(p->owner, AST_CONTROL_CONGESTION); - ast_mutex_unlock(&p->owner->lock); - } - } - ast_mutex_unlock(&p->lock); - return 0; -} - - - - -/*--- sip_call: Initiate SIP call from PBX ---*/ -/* used from the dial() application */ -static int sip_call(struct ast_channel *ast, char *dest, int timeout) -{ - int res; - struct sip_pvt *p; -#ifdef OSP_SUPPORT - char *osphandle = NULL; -#endif - struct varshead *headp; - struct ast_var_t *current; - struct sip_invite_param options; - - memset(&options, 0, sizeof(struct sip_invite_param)); - - 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 (!options.vxml_url && !strcasecmp(ast_var_name(current),"VXML_URL")) { - options.vxml_url = ast_var_value(current); - } else if (!options.distinctive_ring && !strcasecmp(ast_var_name(current),"ALERT_INFO")) { - /* Check whether there is a ALERT_INFO variable */ - options.distinctive_ring = ast_var_value(current); - } else if (!options.addsipheaders && !strncasecmp(ast_var_name(current),"SIPADDHEADER",strlen("SIPADDHEADER"))) { - /* Check whether there is a variable with a name starting with SIPADDHEADER */ - options.addsipheaders = 1; - } - - -#ifdef OSP_SUPPORT - else if (!options.osptoken && !strcasecmp(ast_var_name(current), "OSPTOKEN")) { - options.osptoken = ast_var_value(current); - } else if (!osphandle && !strcasecmp(ast_var_name(current), "OSPHANDLE")) { - osphandle = ast_var_value(current); - } -#endif - } - - res = 0; - ast_set_flag(p, SIP_OUTGOING); -#ifdef OSP_SUPPORT - if (!options.osptoken || !osphandle || (sscanf(osphandle, "%d", &p->osphandle) != 1)) { - /* Force Disable OSP support */ - ast_log(LOG_DEBUG, "Disabling OSP support for this call. osptoken = %s, osphandle = %s\n", options.osptoken, osphandle); - options.osptoken = NULL; - osphandle = NULL; - p->osphandle = -1; - } -#endif - ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username); - res = update_user_counter(p,INC_OUT_USE); - if ( res != -1 ) { - p->callingpres = ast->cid.cid_pres; - p->jointcapability = p->capability; - transmit_invite(p, SIP_INVITE, 1, &options, 1); - if (p->maxtime) { - /* Initialize auto-congest time */ - p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p); - } - } - return res; -} - -/*--- sip_registry_destroy: Destroy registry object ---*/ -/* Objects created with the register= statement in static configuration */ -static void sip_registry_destroy(struct sip_registry *reg) -{ - /* Really delete */ - if (reg->call) { - /* Clear registry before destroying to ensure - we don't get reentered trying to grab the registry lock */ - reg->call->registry = NULL; - sip_destroy(reg->call); - } - if (reg->expire > -1) - ast_sched_del(sched, reg->expire); - if (reg->timeout > -1) - ast_sched_del(sched, reg->timeout); - regobjs--; - free(reg); - -} - -/*--- __sip_destroy: Execute destrucion of call structure, release memory---*/ -static void __sip_destroy(struct sip_pvt *p, int lockowner) -{ - struct sip_pvt *cur, *prev = NULL; - struct sip_pkt *cp; - struct sip_history *hist; - - if (sip_debug_test_pvt(p)) - ast_verbose("Destroying call '%s'\n", p->callid); - if (p->stateid > -1) - ast_extension_state_del(p->stateid, NULL); - if (p->initid > -1) - ast_sched_del(sched, p->initid); - if (p->autokillid > -1) - ast_sched_del(sched, p->autokillid); - - if (p->rtp) { - ast_rtp_destroy(p->rtp); - } - if (p->vrtp) { - ast_rtp_destroy(p->vrtp); - } - 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); - } - /* Unlink us from the owner if we have one */ - if (p->owner) { - if (lockowner) - ast_mutex_lock(&p->owner->lock); - ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name); - p->owner->tech_pvt = NULL; - if (lockowner) - ast_mutex_unlock(&p->owner->lock); - } - /* Clear history */ - while(p->history) { - hist = p->history; - p->history = p->history->next; - free(hist); - } - cur = iflist; - while(cur) { - if (cur == p) { - if (prev) - prev->next = cur->next; - else - iflist = cur->next; - break; - } - prev = cur; - cur = cur->next; - } - if (!cur) { - ast_log(LOG_WARNING, "Trying to destroy \"%s\", not found in dialog list?!?! \n", p->callid); - return; - } - if (p->initid > -1) - ast_sched_del(sched, p->initid); - while((cp = p->packets)) { - p->packets = p->packets->next; - if (cp->retransid > -1) - ast_sched_del(sched, cp->retransid); - free(cp); - } - ast_mutex_destroy(&p->lock); - if (p->chanvars) { - ast_variables_destroy(p->chanvars); - p->chanvars = NULL; - } - free(p); -} - -/*--- update_user_counter: Handle incominglimit and outgoinglimit for SIP users ---*/ -/* Note: This is going to be replaced by app_groupcount */ -/* Thought: For realtime, we should propably update storage with inuse counter... */ -static int update_user_counter(struct sip_pvt *fup, int event) -{ - char name[256] = ""; - struct sip_user *u; - struct sip_peer *p; - int *inuse, *incominglimit; - - /* 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, SIP_CALL_LIMIT)) - return 0; - - ast_copy_string(name, fup->username, sizeof(name)); - - /* Check the list of users */ - u = find_user(name, 1); - if (u) { - inuse = &u->inUse; - incominglimit = &u->incominglimit; - p = NULL; - } else { - /* Try to find peer */ - p = find_peer(fup->peername, NULL, 1); - if (p) { - inuse = &p->inUse; - incominglimit = &p->incominglimit; - ast_copy_string(name, fup->peername, sizeof(name)); - } else { - if (option_debug > 1) - ast_log(LOG_DEBUG, "%s is not a local user, no call limit\n", name); - return 0; - } - } - switch(event) { - /* incoming and outgoing affects the inUse counter */ - case DEC_OUT_USE: - case DEC_IN_USE: - if ( *inuse > 0 ) { - (*inuse)--; - } else { - *inuse = 0; - } - break; - case INC_IN_USE: - case INC_OUT_USE: - if (*incominglimit > 0 ) { - if (*inuse >= *incominglimit) { - ast_log(LOG_ERROR, "Call from %s '%s' rejected due to usage limit of %d\n", u?"user":"peer", name, *incominglimit); - /* inc inUse as well */ - if ( event == INC_OUT_USE ) { - (*inuse)++; - } - if (u) - ASTOBJ_UNREF(u,sip_destroy_user); - else - ASTOBJ_UNREF(p,sip_destroy_peer); - return -1; - } - } - (*inuse)++; - ast_log(LOG_DEBUG, "Call from %s '%s' is %d out of %d\n", u?"user":"peer", name, *inuse, *incominglimit); - break; -#ifdef DISABLED_CODE - /* we don't use these anymore */ - case DEC_OUT_USE: - if ( u->outUse > 0 ) { - u->outUse--; - } else { - u->outUse = 0; - } - break; - case INC_OUT_USE: - if ( u->outgoinglimit > 0 ) { - if ( u->outUse >= u->outgoinglimit ) { - ast_log(LOG_ERROR, "Outgoing call from user '%s' rejected due to usage limit of %d\n", u->name, u->outgoinglimit); - ast_mutex_unlock(&userl.lock); - if (u->temponly) { - destroy_user(u); - } - return -1; - } - } - u->outUse++; - break; -#endif - default: - ast_log(LOG_ERROR, "update_user_counter(%s,%d) called with no event!\n",name,event); - } - if (u) - ASTOBJ_UNREF(u,sip_destroy_user); - else - ASTOBJ_UNREF(p,sip_destroy_peer); - return 0; -} - -/*--- sip_destroy: Destroy SIP call structure ---*/ -static void sip_destroy(struct sip_pvt *p) -{ - ast_mutex_lock(&iflock); - __sip_destroy(p, 1); - ast_mutex_unlock(&iflock); -} - - -static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req, int fatal); - -/*--- hangup_sip2cause: Convert SIP hangup causes to Asterisk hangup causes ---*/ -static int hangup_sip2cause(int cause) -{ -/* Possible values taken from causes.h */ - - switch(cause) { - case 403: /* Not found */ - return AST_CAUSE_CALL_REJECTED; - case 404: /* Not found */ - return AST_CAUSE_UNALLOCATED; - case 408: /* No reaction */ - return AST_CAUSE_NO_USER_RESPONSE; - case 480: /* No answer */ - return AST_CAUSE_FAILURE; - case 483: /* Too many hops */ - return AST_CAUSE_NO_ANSWER; - case 486: /* Busy everywhere */ - return AST_CAUSE_BUSY; - case 488: /* No codecs approved */ - return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; - 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; - default: - return AST_CAUSE_NORMAL; - } - /* Never reached */ - return 0; -} - - -/*--- hangup_cause2sip: Convert Asterisk hangup causes to SIP codes ---*/ -/* 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 -*/ -static 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 */ - 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: - ast_log(LOG_DEBUG, "AST hangup cause %d (no match found in SIP)\n", cause); - return NULL; - } - - /* Never reached */ - return 0; -} - - -/*--- sip_hangup: Hangup SIP call ---*/ -/* Part of PBX interface */ -static int sip_hangup(struct ast_channel *ast) -{ - struct sip_pvt *p = ast->tech_pvt; - int needcancel = 0; - struct ast_flags locflags = {0}; - - if (option_debug) - ast_log(LOG_DEBUG, "sip_hangup(%s)\n", ast->name); - if (!p) { - ast_log(LOG_DEBUG, "Asked to hangup channel not connected\n"); - return 0; - } - ast_mutex_lock(&p->lock); -#ifdef OSP_SUPPORT - if ((p->osphandle > -1) && (ast->_state == AST_STATE_UP)) { - ast_osp_terminate(p->osphandle, AST_CAUSE_NORMAL, p->ospstart, time(NULL) - p->ospstart); - } -#endif - if (ast_test_flag(p, SIP_OUTGOING)) { - ast_log(LOG_DEBUG, "update_user_counter(%s) - decrement outUse counter\n", p->username); - update_user_counter(p, DEC_OUT_USE); - } else { - ast_log(LOG_DEBUG, "update_user_counter(%s) - decrement inUse counter\n", p->username); - update_user_counter(p, DEC_IN_USE); - } - /* Determine how to disconnect */ - if (p->owner != ast) { - ast_log(LOG_WARNING, "Huh? We aren't the owner?\n"); - ast_mutex_unlock(&p->lock); - return 0; - } - if (ast->_state != AST_STATE_UP) - needcancel = 1; - /* Disconnect */ - p = ast->tech_pvt; - if (p->vad) { - ast_dsp_free(p->vad); - } - p->owner = NULL; - ast->tech_pvt = NULL; - - ast_mutex_lock(&usecnt_lock); - usecnt--; - ast_mutex_unlock(&usecnt_lock); - ast_update_use_count(); - - ast_set_flag(&locflags, SIP_NEEDDESTROY); - /* Start the process if it's not already started */ - if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) { - if (needcancel) { - if (ast_test_flag(p, SIP_OUTGOING)) { - transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0); - /* 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. */ - ast_clear_flag(&locflags, SIP_NEEDDESTROY); - sip_scheddestroy(p, 15000); - if ( p->initid != -1 ) { - /* channel still up - reverse dec of inUse counter - only if the channel is not auto-congested */ - if (ast_test_flag(p, SIP_OUTGOING)) { - update_user_counter(p, INC_OUT_USE); - } - else { - update_user_counter(p, INC_IN_USE); - } - } - } else { - char *res; - if (ast->hangupcause && ((res = hangup_cause2sip(ast->hangupcause)))) { - transmit_response_reliable(p, res, &p->initreq, 1); - } else - transmit_response_reliable(p, "403 Forbidden", &p->initreq, 1); - } - } else { - if (!p->pendinginvite) { - /* Send a hangup */ - transmit_request_with_auth(p, SIP_BYE, 0, 1, 1); - } 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, SIP_PENDINGBYE); - ast_clear_flag(p, SIP_NEEDREINVITE); - } - } - } - ast_copy_flags(p, (&locflags), SIP_NEEDDESTROY); - ast_mutex_unlock(&p->lock); - return 0; -} - -/*--- 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,fmt; - char *codec; - struct sip_pvt *p = ast->tech_pvt; - - ast_mutex_lock(&p->lock); - if (ast->_state != AST_STATE_UP) { -#ifdef OSP_SUPPORT - time(&p->ospstart); -#endif - - codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC"); - if (codec) { - 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); - } - - ast_setstate(ast, AST_STATE_UP); - if (option_debug) - ast_log(LOG_DEBUG, "sip_answer(%s)\n", ast->name); - res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 1); - } - ast_mutex_unlock(&p->lock); - return res; -} - -/*--- sip_write: 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)) { - ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %d (read/write = %d/%d)\n", - frame->subclass, ast->nativeformats, ast->readformat, 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, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) { - transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0); - ast_set_flag(p, SIP_PROGRESS_SENT); - } - time(&p->lastrtptx); - 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, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) { - transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0); - ast_set_flag(p, SIP_PROGRESS_SENT); - } - time(&p->lastrtptx); - res = ast_rtp_write(p->vrtp, frame); - } - ast_mutex_unlock(&p->lock); - } - break; - case AST_FRAME_IMAGE: - return 0; - break; - default: - ast_log(LOG_WARNING, "Can't send %d type frames with SIP write\n", frame->frametype); - return 0; - } - - return res; -} - -/*--- 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) -{ - struct sip_pvt *p = newchan->tech_pvt; - ast_mutex_lock(&p->lock); - if (p->owner != oldchan) { - ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner); - ast_mutex_unlock(&p->lock); - return -1; - } - p->owner = newchan; - ast_mutex_unlock(&p->lock); - return 0; -} - -/*--- sip_senddigit: Send DTMF character on SIP channel */ -/* within one call, we're able to transmit in many methods simultaneously */ -static int sip_senddigit(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, SIP_DTMF)) { - case SIP_DTMF_INFO: - transmit_info_with_digit(p, digit); - break; - case SIP_DTMF_RFC2833: - if (p->rtp) - ast_rtp_senddigit(p->rtp, digit); - break; - case SIP_DTMF_INBAND: - res = -1; - break; - } - ast_mutex_unlock(&p->lock); - return res; -} - -#define DEFAULT_MAX_FORWARDS 70 - - -/*--- sip_transfer: Transfer SIP call */ -static int sip_transfer(struct ast_channel *ast, const char *dest) -{ - struct sip_pvt *p = ast->tech_pvt; - int res; - - 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; -} - -/*--- sip_indicate: Play indication to user */ -/* With SIP a lot of indications is sent as messages, letting the device play - the indication - busy signal, congestion etc */ -static int sip_indicate(struct ast_channel *ast, int condition) -{ - 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) { - if (!ast_test_flag(p, SIP_PROGRESS_SENT) || - (ast_test_flag(p, 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, SIP_RINGING); - if (ast_test_flag(p, 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(p, "486 Busy Here", &p->initreq); - ast_set_flag(p, SIP_ALREADYGONE); - ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV); - break; - } - res = -1; - break; - case AST_CONTROL_CONGESTION: - if (ast->_state != AST_STATE_UP) { - transmit_response(p, "503 Service Unavailable", &p->initreq); - ast_set_flag(p, SIP_ALREADYGONE); - ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV); - break; - } - res = -1; - break; - case AST_CONTROL_PROGRESS: - case AST_CONTROL_PROCEEDING: - if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) { - transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0); - ast_set_flag(p, SIP_PROGRESS_SENT); - break; - } - res = -1; - break; - case AST_CONTROL_HOLD: /* We are put on hold */ - /* The PBX is providing us with onhold music, but - should we clear the RTP stream with the other - end? Guess we could do that if there's no - musiconhold class defined for this channel - */ - if (sipdebug) - ast_log(LOG_DEBUG, "SIP dialog on hold: %s\n", p->callid); - res = -1; - ast_set_flag(p, SIP_CALL_ONHOLD); - break; - case AST_CONTROL_UNHOLD: /* We are back from hold */ - /* Open RTP stream if we decide to close it - */ - if (sipdebug) - ast_log(LOG_DEBUG, "SIP dialog off hold: %s\n", p->callid); - res = -1; - ast_clear_flag(p, SIP_CALL_ONHOLD); - 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; -} - - - -/*--- sip_new: Initiate a call in the SIP channel */ -/* called from sip_request (calls from the pbx ) */ -static struct ast_channel *sip_new(struct sip_pvt *i, int state, char *title) -{ - struct ast_channel *tmp; - struct ast_variable *v = NULL; - int fmt; - - ast_mutex_unlock(&i->lock); - /* Don't hold a sip pvt lock while we allocate a channel */ - tmp = ast_channel_alloc(1); - ast_mutex_lock(&i->lock); - if (!tmp) { - ast_log(LOG_WARNING, "Unable to allocate SIP channel structure\n"); - return NULL; - } - tmp->tech = &sip_tech; - /* Select our native format based on codec preference until we receive - something from another device to the contrary. */ - ast_mutex_lock(&i->lock); - if (i->jointcapability) - tmp->nativeformats = ast_codec_choose(&i->prefs, i->jointcapability, 1); - else if (i->capability) - tmp->nativeformats = ast_codec_choose(&i->prefs, i->capability, 1); - else - tmp->nativeformats = ast_codec_choose(&i->prefs, global_capability, 1); - ast_mutex_unlock(&i->lock); - fmt = ast_best_codec(tmp->nativeformats); - - if (title) - snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%04x", title, rand() & 0xffff); - else if (strchr(i->fromdomain,':')) - snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%08x", strchr(i->fromdomain,':')+1, (int)(long)(i)); - else - snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%08x", i->fromdomain, (int)(long)(i)); - - tmp->type = channeltype; - if (ast_test_flag(i, SIP_DTMF) == SIP_DTMF_INBAND) { - i->vad = ast_dsp_new(); - ast_dsp_set_features(i->vad, DSP_FEATURE_DTMF_DETECT); - if (relaxdtmf) - ast_dsp_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF); - } - tmp->fds[0] = ast_rtp_fd(i->rtp); - tmp->fds[1] = ast_rtcp_fd(i->rtp); - if (i->vrtp) { - tmp->fds[2] = ast_rtp_fd(i->vrtp); - tmp->fds[3] = ast_rtcp_fd(i->vrtp); - } - 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_copy_string(tmp->accountcode, i->accountcode, sizeof(tmp->accountcode)); - if (i->amaflags) - tmp->amaflags = i->amaflags; - if (!ast_strlen_zero(i->language)) - ast_copy_string(tmp->language, i->language, sizeof(tmp->language)); - if (!ast_strlen_zero(i->musicclass)) - ast_copy_string(tmp->musicclass, i->musicclass, sizeof(tmp->musicclass)); - i->owner = tmp; - ast_mutex_lock(&usecnt_lock); - usecnt++; - ast_mutex_unlock(&usecnt_lock); - ast_copy_string(tmp->context, i->context, sizeof(tmp->context)); - ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten)); - if (!ast_strlen_zero(i->cid_num)) - tmp->cid.cid_num = strdup(i->cid_num); - if (!ast_strlen_zero(i->cid_name)) - tmp->cid.cid_name = strdup(i->cid_name); - if (!ast_strlen_zero(i->rdnis)) - tmp->cid.cid_rdnis = strdup(i->rdnis); - if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s")) - tmp->cid.cid_dnid = 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); - } - ast_setstate(tmp, state); - if (state != AST_STATE_DOWN) { - if (ast_pbx_start(tmp)) { - ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name); - ast_hangup(tmp); - tmp = NULL; - } - } - /* 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); - - return tmp; -} - -/*--- get_sdp_by_line: Reads one line of SIP message body */ -static char* get_sdp_by_line(char* line, char *name, int nameLen) -{ - if (strncasecmp(line, name, nameLen) == 0 && line[nameLen] == '=') { - return ast_skip_blanks(line + nameLen + 1); - } - return ""; -} - -/*--- get_sdp: Gets all kind of SIP message bodies, including SDP, - but the name wrongly applies _only_ sdp */ -static char *get_sdp(struct sip_request *req, char *name) -{ - int x; - int len = strlen(name); - char *r; - - for (x=0; xlines; x++) { - r = get_sdp_by_line(req->line[x], name, len); - if (r[0] != '\0') - return r; - } - return ""; -} - - -static void sdpLineNum_iterator_init(int* iterator) -{ - *iterator = 0; -} - -static char* get_sdp_iterate(int* iterator, - struct sip_request *req, char *name) -{ - int len = strlen(name); - char *r; - - while (*iterator < req->lines) { - r = get_sdp_by_line(req->line[(*iterator)++], name, len); - if (r[0] != '\0') - return r; - } - return ""; -} - -static char *find_alias(const char *name, char *_default) -{ - int x; - for (x=0;xheaders; 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 ""; -} - -/*--- get_header: Get header from SIP request ---*/ -static char *get_header(struct sip_request *req, char *name) -{ - int start = 0; - return __get_header(req, name, &start); -} - -/*--- sip_rtp_read: Read RTP from network ---*/ -static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p) -{ - /* Retrieve audio/etc from channel. Assumes p->lock is already held. */ - struct ast_frame *f; - static struct ast_frame null_frame = { AST_FRAME_NULL, }; - 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; - default: - f = &null_frame; - } - /* Don't forward RFC2833 if we're not supposed to */ - if (f && (f->frametype == AST_FRAME_DTMF) && (ast_test_flag(p, SIP_DTMF) != SIP_DTMF_RFC2833)) - return &null_frame; - if (p->owner) { - /* We already hold the channel lock */ - if (f->frametype == AST_FRAME_VOICE) { - if (f->subclass != p->owner->nativeformats) { - ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass); - p->owner->nativeformats = f->subclass; - ast_set_read_format(p->owner, p->owner->readformat); - ast_set_write_format(p->owner, p->owner->writeformat); - } - if ((ast_test_flag(p, SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) { - f = ast_dsp_process(p->owner, p->vad, f); - if (f && (f->frametype == AST_FRAME_DTMF)) - ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass); - } - } - } - return f; -} - -/*--- sip_read: 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; - ast_mutex_lock(&p->lock); - fr = sip_rtp_read(ast, p); - time(&p->lastrtprx); - ast_mutex_unlock(&p->lock); - return fr; -} - -/*--- build_callid: Build SIP CALLID header ---*/ -static void build_callid(char *callid, int len, struct in_addr ourip, char *fromdomain) -{ - int res; - int val; - int x; - char iabuf[INET_ADDRSTRLEN]; - for (x=0; x<4; x++) { - val = rand(); - res = snprintf(callid, len, "%08x", val); - len -= res; - callid += res; - } - if (!ast_strlen_zero(fromdomain)) - snprintf(callid, len, "@%s", fromdomain); - else - /* It's not important that we really use our right IP here... */ - snprintf(callid, len, "@%s", ast_inet_ntoa(iabuf, sizeof(iabuf), ourip)); -} - -/*--- sip_alloc: Allocate SIP_PVT structure and set defaults ---*/ -static struct sip_pvt *sip_alloc(char *callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method) -{ - struct sip_pvt *p; - - p = malloc(sizeof(struct sip_pvt)); - if (!p) - return NULL; - /* Keep track of stuff */ - memset(p, 0, sizeof(struct sip_pvt)); - ast_mutex_init(&p->lock); - - p->method = intended_method; - p->initid = -1; - p->autokillid = -1; - p->stateid = -1; - p->prefs = prefs; -#ifdef OSP_SUPPORT - p->osphandle = -1; -#endif - if (sin) { - memcpy(&p->sa, sin, sizeof(p->sa)); - if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip)) - memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); - } else { - memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); - } - - p->branch = rand(); - p->tag = rand(); - /* Start with 101 instead of 1 */ - p->ocseq = 101; - - if (sip_methods[intended_method].need_rtp) { - p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr); - if (videosupport) - p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr); - if (!p->rtp) { - ast_log(LOG_WARNING, "Unable to create RTP session: %s\n", strerror(errno)); - ast_mutex_destroy(&p->lock); - if (p->chanvars) { - ast_variables_destroy(p->chanvars); - p->chanvars = NULL; - } - free(p); - return NULL; - } - ast_rtp_settos(p->rtp, tos); - if (p->vrtp) - ast_rtp_settos(p->vrtp, tos); - p->rtptimeout = global_rtptimeout; - p->rtpholdtimeout = global_rtpholdtimeout; - p->rtpkeepalive = global_rtpkeepalive; - } - - if (useglobal_nat && sin) { - /* Setup NAT structure according to global settings if we have an address */ - ast_copy_flags(p, &global_flags, SIP_NAT); - memcpy(&p->recv, sin, sizeof(p->recv)); - if (p->rtp) - ast_rtp_setnat(p->rtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); - if (p->vrtp) - ast_rtp_setnat(p->vrtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); - } - - if (p->method != SIP_REGISTER) - ast_copy_string(p->fromdomain, default_fromdomain, sizeof(p->fromdomain)); - build_via(p, p->via, sizeof(p->via)); - if (!callid) - build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); - else - ast_copy_string(p->callid, callid, sizeof(p->callid)); - ast_copy_flags(p, (&global_flags), SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_DTMF | SIP_REINVITE | SIP_PROG_INBAND | SIP_OSPAUTH); - /* Assign default music on hold class */ - strcpy(p->musicclass, global_musicclass); - p->capability = global_capability; - if (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) - p->noncodeccapability |= AST_RTP_DTMF; - strcpy(p->context, default_context); - - /* 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; -} - -/*--- find_call: 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; - char *callid; - char tmp[256] = ""; - char iabuf[INET_ADDRSTRLEN]; - char *cmd; - char *tag = "", *c; - - callid = get_header(req, "Call-ID"); - - if (pedanticsipchecking) { - /* In principle Call-ID's uniquely identify a call, however some vendors - (i.e. Pingtel) send multiple calls with the same Call-ID and different - tags in order to simplify billing. The RFC does state that we have to - compare tags in addition to the call-id, but this generate substantially - more overhead which is totally unnecessary for the vast majority of sane - SIP implementations, and thus Asterisk does not enable this behavior - by default. Short version: You'll need this option to support conferencing - on the pingtel */ - ast_copy_string(tmp, req->header[0], sizeof(tmp)); - cmd = tmp; - c = strchr(tmp, ' '); - if (c) - *c = '\0'; - if (!strcasecmp(cmd, "SIP/2.0")) - ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp)); - else - ast_copy_string(tmp, get_header(req, "From"), sizeof(tmp)); - tag = strcasestr(tmp, "tag="); - if (tag) { - tag += 4; - c = strchr(tag, ';'); - if (c) - *c = '\0'; - } - - } - - if (ast_strlen_zero(callid)) { - ast_log(LOG_WARNING, "Call missing call ID from '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr)); - return NULL; - } - ast_mutex_lock(&iflock); - p = iflist; - while(p) { - if (!strcmp(p->callid, callid) && - (!pedanticsipchecking || !tag || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) { - /* Found the call */ - ast_mutex_lock(&p->lock); - ast_mutex_unlock(&iflock); - return p; - } - p = p->next; - } - ast_mutex_unlock(&iflock); - p = sip_alloc(callid, sin, 1, intended_method); - if (p) - ast_mutex_lock(&p->lock); - return p; -} - -/*--- sip_register: Parse register=> line in sip.conf and add to registry */ -static int sip_register(char *value, int lineno) -{ - struct sip_registry *reg; - char copy[256] = ""; - char *username=NULL, *hostname=NULL, *secret=NULL, *authuser=NULL; - char *porta=NULL; - char *contact=NULL; - char *stringp=NULL; - - if (!value) - return -1; - ast_copy_string(copy, value, sizeof(copy)); - stringp=copy; - username = stringp; - hostname = strrchr(stringp, '@'); - if (hostname) { - *hostname = '\0'; - hostname++; - } - if (!username || ast_strlen_zero(username) || !hostname || 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; - } - stringp=username; - username = strsep(&stringp, ":"); - if (username) { - secret = strsep(&stringp, ":"); - if (secret) - authuser = strsep(&stringp, ":"); - } - stringp = hostname; - hostname = strsep(&stringp, "/"); - if (hostname) - contact = strsep(&stringp, "/"); - if (!contact || ast_strlen_zero(contact)) - contact = "s"; - stringp=hostname; - hostname = strsep(&stringp, ":"); - porta = strsep(&stringp, ":"); - - if (porta && !atoi(porta)) { - ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno); - return -1; - } - reg = malloc(sizeof(struct sip_registry)); - if (!reg) { - ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n"); - return -1; - } - memset(reg, 0, sizeof(struct sip_registry)); - regobjs++; - ASTOBJ_INIT(reg); - ast_copy_string(reg->contact, contact, sizeof(reg->contact)); - if (username) - ast_copy_string(reg->username, username, sizeof(reg->username)); - if (hostname) - ast_copy_string(reg->hostname, hostname, sizeof(reg->hostname)); - if (authuser) - ast_copy_string(reg->authuser, authuser, sizeof(reg->authuser)); - if (secret) - ast_copy_string(reg->secret, secret, sizeof(reg->secret)); - reg->expire = -1; - reg->timeout = -1; - reg->refresh = default_expiry; - reg->portno = porta ? atoi(porta) : 0; - reg->callid_valid = 0; - reg->ocseq = 101; - ASTOBJ_CONTAINER_LINK(®l, reg); - ASTOBJ_UNREF(reg,sip_registry_destroy); - return 0; -} - -/*--- lws2sws: 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; -} - -/*--- parse: Parse a SIP message ----*/ -static void parse(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 0 - printf("Header: %s (%d)\n", req->header[f], strlen(req->header[f])); -#endif - 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...\n"); - } else - f++; - req->header[f] = c + 1; - } else if (*c == '\r') { - /* Ignore but eliminate \r's */ - *c = 0; - } - c++; - } - /* Check for last header */ - if (!ast_strlen_zero(req->header[f])) - f++; - req->headers = f; - /* Now we process any mime content */ - f = 0; - req->line[f] = c; - while(*c) { - if (*c == '\n') { - /* We've got a new line */ - *c = 0; -#if 0 - printf("Line: %s (%d)\n", req->line[f], strlen(req->line[f])); -#endif - if (f >= SIP_MAX_LINES - 1) { - ast_log(LOG_WARNING, "Too many SDP lines...\n"); - } else - f++; - req->line[f] = c + 1; - } else if (*c == '\r') { - /* Ignore and eliminate \r's */ - *c = 0; - } - c++; - } - /* Check for last line */ - if (!ast_strlen_zero(req->line[f])) - f++; - req->lines = f; - if (*c) - ast_log(LOG_WARNING, "Odd content, extra stuff left over ('%s')\n", c); -} - -/*--- process_sdp: Process SIP SDP and activate RTP channels---*/ -static int process_sdp(struct sip_pvt *p, struct sip_request *req) -{ - char *m; - char *c; - char *a; - char host[258]; - char iabuf[INET_ADDRSTRLEN]; - int len = -1; - int portno = -1; - int vportno = -1; - int peercapability, peernoncodeccapability; - int vpeercapability=0, vpeernoncodeccapability=0; - struct sockaddr_in sin; - char *codecs; - struct hostent *hp; - struct ast_hostent ahp; - int codec; - int destiterator = 0; - int iterator; - int sendonly = 0; - int x,y; - int debug=sip_debug_test_pvt(p); - struct ast_channel *bridgepeer = NULL; - - /* Update our last rtprx when we receive an SDP, too */ - time(&p->lastrtprx); - time(&p->lastrtptx); - - /* Get codec and RTP info from SDP */ - if (strcasecmp(get_header(req, "Content-Type"), "application/sdp")) { - ast_log(LOG_NOTICE, "Content is '%s', not 'application/sdp'\n", get_header(req, "Content-Type")); - return -1; - } - m = get_sdp(req, "m"); - sdpLineNum_iterator_init(&destiterator); - 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; - } - 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, &ahp); - if (!hp) { - ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c); - return -1; - } - sdpLineNum_iterator_init(&iterator); - ast_set_flag(p, SIP_NOVIDEO); - while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') { - int found = 0; - if ((sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1) || - (sscanf(m, "audio %d/%d RTP/AVP %n", &x, &y, &len) == 2)) { - found = 1; - portno = x; - /* Scan through the RTP payload types specified in a "m=" line: */ - ast_rtp_pt_clear(p->rtp); - codecs = m + len; - while(!ast_strlen_zero(codecs)) { - 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(p->rtp, codec); - codecs = ast_skip_blanks(codecs + len); - } - } - if (p->vrtp) - ast_rtp_pt_clear(p->vrtp); /* Must be cleared in case no m=video line exists */ - - if (p->vrtp && (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) { - found = 1; - ast_clear_flag(p, SIP_NOVIDEO); - vportno = x; - /* Scan through the RTP payload types specified in a "m=" line: */ - codecs = m + len; - while(!ast_strlen_zero(codecs)) { - 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 video format %s\n", ast_getformatname(codec)); - ast_rtp_set_m_type(p->vrtp, codec); - codecs = ast_skip_blanks(codecs + len); - } - } - if (!found ) - ast_log(LOG_WARNING, "Unknown SDP media type in offer: %s\n", m); - } - if (portno == -1 && vportno == -1) { - /* No acceptable offer found in SDP */ - return -2; - } - /* Check for Media-description-level-address for audio */ - if (pedanticsipchecking) { - 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 */ - hp = ast_gethostbyname(host, &ahp); - if (!hp) { - ast_log(LOG_WARNING, "Unable to lookup host in secondary c= line, '%s'\n", c); - } - } - } - } - /* RTP addresses and ports for audio and video */ - sin.sin_family = AF_INET; - memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr)); - - /* Setup audio port number */ - sin.sin_port = htons(portno); - 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(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port)); - ast_log(LOG_DEBUG,"Peer audio RTP is at port %s:%d\n",ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port)); - } - } - /* Check for Media-description-level-address for video */ - if (pedanticsipchecking) { - 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 */ - hp = ast_gethostbyname(host, &ahp); - if (!hp) { - ast_log(LOG_WARNING, "Unable to lookup host in secondary c= line, '%s'\n", c); - } - } - } - } - /* Setup video port number */ - sin.sin_port = htons(vportno); - if (p->vrtp && sin.sin_port) { - ast_rtp_set_peer(p->vrtp, &sin); - if (debug) { - ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port)); - ast_log(LOG_DEBUG,"Peer video RTP is at port %s:%d\n",ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port)); - } - } - - /* Next, scan through each "a=rtpmap:" line, noting each - * specified RTP payload type (with corresponding MIME subtype): - */ - sdpLineNum_iterator_init(&iterator); - while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') { - char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */ - if (!strcasecmp(a, "sendonly")) { - sendonly=1; - continue; - } - if (!strcasecmp(a, "sendrecv")) { - sendonly=0; - } - if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2) continue; - if (debug) - ast_verbose("Found description format %s\n", mimeSubtype); - /* Note: should really look at the 'freq' and '#chans' params too */ - ast_rtp_set_rtpmap_type(p->rtp, codec, "audio", mimeSubtype); - if (p->vrtp) - ast_rtp_set_rtpmap_type(p->vrtp, codec, "video", mimeSubtype); - } - - /* Now gather all of the codecs that were asked for: */ - ast_rtp_get_current_formats(p->rtp, - &peercapability, &peernoncodeccapability); - if (p->vrtp) - ast_rtp_get_current_formats(p->vrtp, - &vpeercapability, &vpeernoncodeccapability); - p->jointcapability = p->capability & (peercapability | vpeercapability); - p->peercapability = (peercapability | vpeercapability); - p->noncodeccapability = noncodeccapability & peernoncodeccapability; - - if (debug) { - /* shame on whoever coded this.... */ - const unsigned slen=512; - char s1[slen], s2[slen], s3[slen], s4[slen]; - - ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n", - ast_getformatname_multiple(s1, slen, p->capability), - ast_getformatname_multiple(s2, slen, peercapability), - ast_getformatname_multiple(s3, slen, vpeercapability), - ast_getformatname_multiple(s4, slen, p->jointcapability)); - - ast_verbose("Non-codec capabilities: us - %s, peer - %s, combined - %s\n", - ast_rtp_lookup_mime_multiple(s1, slen, noncodeccapability, 0), - ast_rtp_lookup_mime_multiple(s2, slen, peernoncodeccapability, 0), - ast_rtp_lookup_mime_multiple(s3, slen, p->noncodeccapability, 0)); - } - if (!p->jointcapability) { - ast_log(LOG_NOTICE, "No compatible codecs!\n"); - return -1; - } - - if (!p->owner) /* There's no open channel owning us */ - return 0; - - if (!(p->owner->nativeformats & p->jointcapability)) { - const unsigned slen=512; - char s1[slen], s2[slen]; - ast_log(LOG_DEBUG, "Oooh, we need to change our formats since our peer supports only %s and not %s\n", - ast_getformatname_multiple(s1, slen, p->jointcapability), - ast_getformatname_multiple(s2, slen, p->owner->nativeformats)); - p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1); - ast_set_read_format(p->owner, p->owner->readformat); - ast_set_write_format(p->owner, p->owner->writeformat); - } - if ((bridgepeer=ast_bridged_channel(p->owner))) { - /* We have a bridge */ - /* Turn on/off music on hold if we are holding/unholding */ - if (sin.sin_addr.s_addr && !sendonly) { - ast_moh_stop(bridgepeer); - /* Indicate UNHOLD status to the other channel */ - ast_indicate(bridgepeer, AST_CONTROL_UNHOLD); - append_history(p, "Unhold", req->data); - if (callevents && ast_test_flag(p, SIP_CALL_ONHOLD)) { - manager_event(EVENT_FLAG_CALL, "Unhold", - "Channel: %s\r\n" - "Uniqueid: %s\r\n", - p->owner->name, - p->owner->uniqueid); - } - ast_clear_flag(p, SIP_CALL_ONHOLD); - /* Somehow, we need to check if we need to re-invite here */ - /* If this call had a external native bridge, it's broken - now and we need to start all over again. - The bridged peer, if SIP, now listens - to RTP from Asterisk instead of from - the peer - - So IF we had a native bridge before - the HOLD, we need to somehow re-invite - into a NATIVE bridge afterwards... - - */ - - } else { - /* No address for RTP, we're on hold */ - append_history(p, "Hold", req->data); - if (callevents && !ast_test_flag(p, SIP_CALL_ONHOLD)) { - manager_event(EVENT_FLAG_CALL, "Hold", - "Channel: %s\r\n" - "Uniqueid: %s\r\n", - p->owner->name, - p->owner->uniqueid); - } - ast_set_flag(p, SIP_CALL_ONHOLD); - /* Indicate HOLD status to the other channel */ - ast_indicate(bridgepeer, AST_CONTROL_HOLD); - ast_moh_start(bridgepeer, NULL); - if (sendonly) - ast_rtp_stop(p->rtp); - } - } - return 0; -} - -/*--- add_header: Add header to SIP message */ -static int add_header(struct sip_request *req, char *var, char *value) -{ - int x = 0; - char *shortname = ""; - 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 (req->len >= sizeof(req->data) - 4) { - 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) { - for (x=0;xheader[req->headers], sizeof(req->data) - req->len - 4, "%s: %s\r\n", shortname, value); - } else { - snprintf(req->header[req->headers], sizeof(req->data) - req->len - 4, "%s: %s\r\n", var, value); - } - req->len += strlen(req->header[req->headers]); - req->headers++; - return 0; -} - -/*--- add_blank_header: Add blank header to SIP message */ -static int add_blank_header(struct sip_request *req) -{ - 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 (req->len >= sizeof(req->data) - 4) { - ast_log(LOG_WARNING, "Out of space, can't add anymore\n"); - return -1; - } - req->header[req->headers] = req->data + req->len; - snprintf(req->header[req->headers], sizeof(req->data) - req->len, "\r\n"); - req->len += strlen(req->header[req->headers]); - req->headers++; - return 0; -} - -/*--- add_line: 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; -} - -/*--- copy_header: Copy one header field from one request to another */ -static int copy_header(struct sip_request *req, struct sip_request *orig, char *field) -{ - char *tmp; - 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; -} - -/*--- copy_all_header: Copy all headers from one request to another ---*/ -static int copy_all_header(struct sip_request *req, struct sip_request *orig, char *field) -{ - char *tmp; - int start = 0; - int copied = 0; - for (;;) { - tmp = __get_header(orig, field, &start); - if (!ast_strlen_zero(tmp)) { - /* Add what we're responding to */ - add_header(req, field, tmp); - copied++; - } else - break; - } - return copied ? 0 : -1; -} - -/*--- copy_via_headers: Copy SIP VIA Headers from one request to another ---*/ -static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, struct sip_request *orig, char *field) -{ - char tmp[256]="", *oh, *end; - int start = 0; - int copied = 0; - char new[256]; - char iabuf[INET_ADDRSTRLEN]; - for (;;) { - oh = __get_header(orig, field, &start); - if (!ast_strlen_zero(oh)) { - /* Strip ;rport */ - ast_copy_string(tmp, oh, sizeof(tmp)); - oh = strstr(tmp, ";rport"); - if (oh) { - end = strchr(oh + 1, ';'); - if (end) - memmove(oh, end, strlen(end) + 1); - else - *oh = '\0'; - } - if (!copied && (ast_test_flag(p, SIP_NAT) == SIP_NAT_ALWAYS)) { - /* Whoo hoo! Now we can indicate port address translation too! Just - another RFC (RFC3581). I'll leave the original comments in for - posterity. */ - snprintf(new, sizeof(new), "%s;received=%s;rport=%d", tmp, ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port)); - add_header(req, field, new); - } else { - /* Add what we're responding to */ - add_header(req, field, tmp); - } - copied++; - } else - break; - } - if (!copied) { - ast_log(LOG_NOTICE, "No header field '%s' present to copy\n", field); - return -1; - } - return 0; -} - -/*--- add_route: Add route header into request per learned route ---*/ -static void add_route(struct sip_request *req, struct sip_route *route) -{ - char r[256], *p; - int n, rem = sizeof(r); - - if (!route) return; - - p = r; - while (route) { - n = strlen(route->hop); - if ((n+3)>rem) break; - if (p != r) { - *p++ = ','; - --rem; - } - *p++ = '<'; - ast_copy_string(p, route->hop, rem); p += n; - *p++ = '>'; - rem -= (n+2); - route = route->next; - } - *p = '\0'; - add_header(req, "Route", r); -} - -/*--- set_destination: Set destination from SIP URI ---*/ -static void set_destination(struct sip_pvt *p, char *uri) -{ - char *h, *maddr, hostname[256] = ""; - char iabuf[INET_ADDRSTRLEN]; - 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 (strncmp(h, "sip:", 4) == 0) - h += 4; - else if (strncmp(h, "sips:", 5) == 0) - h += 5; - } - hn = strcspn(h, ":;>") + 1; - if (hn > sizeof(hostname)) hn = sizeof(hostname); - ast_copy_string(hostname, h, hn); - h += hn - 1; - - /* Is "port" present? if not default to DEFAULT_SIP_PORT */ - if (*h == ':') { - /* Parse port */ - ++h; - port = strtol(h, &h, 10); - } - else - port = DEFAULT_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(iabuf, sizeof(iabuf), p->sa.sin_addr), port); -} - -/*--- init_resp: Initialize SIP response, based on SIP request ---*/ -static int init_resp(struct sip_request *req, char *resp, struct sip_request *orig) -{ - /* Initialize a response */ - if (req->headers || req->len) { - ast_log(LOG_WARNING, "Request already initialized?!?\n"); - return -1; - } - req->header[req->headers] = req->data + req->len; - snprintf(req->header[req->headers], sizeof(req->data) - req->len, "SIP/2.0 %s\r\n", resp); - req->len += strlen(req->header[req->headers]); - req->headers++; - return 0; -} - -/*--- init_req: Initialize SIP request ---*/ -static int init_req(struct sip_request *req, int sipmethod, char *recip) -{ - /* Initialize a response */ - if (req->headers || req->len) { - ast_log(LOG_WARNING, "Request already initialized?!?\n"); - return -1; - } - req->header[req->headers] = req->data + req->len; - snprintf(req->header[req->headers], sizeof(req->data) - req->len, "%s %s SIP/2.0\r\n", sip_methods[sipmethod].text, recip); - req->len += strlen(req->header[req->headers]); - req->headers++; - return 0; -} - - -/*--- respprep: Prepare SIP response packet ---*/ -static int respprep(struct sip_request *resp, struct sip_pvt *p, char *msg, struct sip_request *req) -{ - char newto[256] = "", *ot; - - memset(resp, 0, sizeof(*resp)); - init_resp(resp, msg, req); - copy_via_headers(p, resp, req, "Via"); - if (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, SIP_OUTGOING)) - snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag); - else if (p->tag && !ast_test_flag(p, SIP_OUTGOING)) - snprintf(newto, sizeof(newto), "%s;tag=as%08x", ot, p->tag); - else { - ast_copy_string(newto, ot, sizeof(newto)); - newto[sizeof(newto) - 1] = '\0'; - } - ot = newto; - } - add_header(resp, "To", ot); - copy_header(resp, req, "Call-ID"); - copy_header(resp, req, "CSeq"); - add_header(resp, "User-Agent", default_useragent); - add_header(resp, "Allow", ALLOWED_METHODS); - if (p->expiry) { - /* For registration responses, we also need expiry and - contact info */ - char contact[256]; - char tmp[256]; - - snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, p->expiry); - snprintf(tmp, sizeof(tmp), "%d", p->expiry); - add_header(resp, "Expires", tmp); - add_header(resp, "Contact", contact); - } else { - add_header(resp, "Contact", p->our_contact); - } - if (p->maxforwards) { - char tmp[256]; - snprintf(tmp, sizeof(tmp), "%d", p->maxforwards); - add_header(resp, "Max-Forwards", tmp); - } - return 0; -} - -/*--- reqprep: Initialize a SIP request packet ---*/ -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]; - char *c, *n; - char *ot, *of; - - 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 (newbranch) { - p->branch ^= rand(); - build_via(p, p->via, sizeof(p->via)); - } - 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 = p->okcontacturi; - else - c = p->initreq.rlPart2; - } else if (!ast_strlen_zero(p->okcontacturi)) { - c = p->okcontacturi; /* Use for BYE, REFER or REINVITE */ - } else if (!ast_strlen_zero(p->uri)) { - c = p->uri; - } else { - /* We have no URI, use To: or From: header as URI (depending on direction) */ - c = get_header(orig, (ast_test_flag(p, SIP_OUTGOING)) ? "To" : "From"); - ast_copy_string(stripped, c, sizeof(stripped)); - c = get_in_brackets(stripped); - n = strchr(c, ';'); - if (n) - *n = '\0'; - } - 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, p->route->next); - } - - 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, SIP_OUTGOING) && !ast_strlen_zero(p->theirtag)) - snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag); - else if (!ast_test_flag(p, SIP_OUTGOING)) - snprintf(newto, sizeof(newto), "%s;tag=as%08x", ot, p->tag); - else - snprintf(newto, sizeof(newto), "%s", ot); - ot = newto; - } - - if (ast_test_flag(p, SIP_OUTGOING)) { - add_header(req, "From", of); - add_header(req, "To", ot); - } else { - add_header(req, "From", ot); - add_header(req, "To", of); - } - add_header(req, "Contact", p->our_contact); - copy_header(req, orig, "Call-ID"); - add_header(req, "CSeq", tmp); - - add_header(req, "User-Agent", default_useragent); - return 0; -} - -static int __transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req, int 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(&resp, "Content-Length", "0"); - add_blank_header(&resp); - return send_response(p, &resp, reliable, seqno); -} - -/*--- transmit_response: Transmit response, no retransmits */ -static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req) -{ - return __transmit_response(p, msg, req, 0); -} - -/*--- transmit_response_with_unsupported: Transmit response, no retransmits */ -static int transmit_response_with_unsupported(struct sip_pvt *p, char *msg, struct sip_request *req, char *unsupported) -{ - struct sip_request resp; - respprep(&resp, p, msg, req); - append_date(&resp); - add_header(&resp, "Unsupported", unsupported); - return send_response(p, &resp, 0, 0); -} - -/*--- transmit_response_reliable: Transmit response, Make sure you get a reply */ -static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req, int fatal) -{ - return __transmit_response(p, msg, req, fatal ? 2 : 1); -} - -/*--- append_date: Append date to SIP message ---*/ -static void append_date(struct sip_request *req) -{ - char tmpdat[256]; - struct tm tm; - time_t t; - - time(&t); - gmtime_r(&t, &tm); - strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T GMT", &tm); - add_header(req, "Date", tmpdat); -} - -/*--- transmit_response_with_date: Append date and content length before transmitting response ---*/ -static int transmit_response_with_date(struct sip_pvt *p, char *msg, struct sip_request *req) -{ - struct sip_request resp; - respprep(&resp, p, msg, req); - append_date(&resp); - add_header(&resp, "Content-Length", "0"); - add_blank_header(&resp); - return send_response(p, &resp, 0, 0); -} - -/*--- transmit_response_with_allow: Append Accept header, content length before transmitting response ---*/ -static int transmit_response_with_allow(struct sip_pvt *p, char *msg, struct sip_request *req, int reliable) -{ - struct sip_request resp; - respprep(&resp, p, msg, req); - add_header(&resp, "Accept", "application/sdp"); - add_header(&resp, "Content-Length", "0"); - add_blank_header(&resp); - return send_response(p, &resp, reliable, 0); -} - -/* transmit_response_with_auth: Respond with authorization request */ -static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_request *req, char *randdata, int reliable, char *header, int stale) -{ - struct sip_request resp; - char tmp[256]; - 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 realm=\"%s\", nonce=\"%s\" %s", global_realm, randdata, stale ? ", stale=true" : ""); - respprep(&resp, p, msg, req); - add_header(&resp, header, tmp); - add_header(&resp, "Content-Length", "0"); - add_blank_header(&resp); - return send_response(p, &resp, reliable, seqno); -} - -/*--- add_text: 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 */ - int len = strlen(text); - char clen[256]; - snprintf(clen, sizeof(clen), "%d", len); - add_header(req, "Content-Type", "text/plain"); - add_header(req, "Content-Length", clen); - add_line(req, text); - return 0; -} - -/*--- add_digit: 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) -{ - char tmp[256]; - int len; - char clen[256]; - snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=250\r\n", digit); - len = strlen(tmp); - snprintf(clen, sizeof(clen), "%d", len); - add_header(req, "Content-Type", "application/dtmf-relay"); - add_header(req, "Content-Length", clen); - add_line(req, tmp); - return 0; -} - -/*--- add_sdp: Add Session Description Protocol message ---*/ -static int add_sdp(struct sip_request *resp, struct sip_pvt *p) -{ - int len = 0; - int codec = 0; - int pref_codec = 0; - int alreadysent = 0; - char costr[80]; - struct sockaddr_in sin; - struct sockaddr_in vsin; - char v[256] = ""; - char s[256] = ""; - char o[256] = ""; - char c[256] = ""; - char t[256] = ""; - char m[256] = ""; - char m2[256] = ""; - char a[1024] = ""; - char a2[1024] = ""; - char iabuf[INET_ADDRSTRLEN]; - int x = 0; - int capability = 0 ; - struct sockaddr_in dest; - struct sockaddr_in vdest = { 0, }; - int debug=0; - - debug = sip_debug_test_pvt(p); - - /* XXX We break with the "recommendation" and send our IP, in order that our - peer doesn't have to ast_gethostbyname() us XXX */ - len = 0; - if (!p->rtp) { - ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n"); - return -1; - } - capability = p->capability; - - if (!p->sessionid) { - p->sessionid = getpid(); - p->sessionversion = p->sessionid; - } else - p->sessionversion++; - ast_rtp_get_us(p->rtp, &sin); - if (p->vrtp) - ast_rtp_get_us(p->vrtp, &vsin); - - if (p->redirip.sin_addr.s_addr) { - dest.sin_port = p->redirip.sin_port; - dest.sin_addr = p->redirip.sin_addr; - if (p->redircodecs) - capability = p->redircodecs; - } else { - dest.sin_addr = p->ourip; - dest.sin_port = sin.sin_port; - } - - /* Determine video destination */ - if (p->vrtp) { - if (p->vredirip.sin_addr.s_addr) { - vdest.sin_port = p->vredirip.sin_port; - vdest.sin_addr = p->vredirip.sin_addr; - } else { - vdest.sin_addr = p->ourip; - vdest.sin_port = vsin.sin_port; - } - } - if (debug){ - ast_verbose("We're at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(sin.sin_port)); - if (p->vrtp) - ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(vsin.sin_port)); - } - 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(iabuf, sizeof(iabuf), dest.sin_addr)); - snprintf(s, sizeof(s), "s=session\r\n"); - snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr)); - snprintf(t, sizeof(t), "t=0 0\r\n"); - snprintf(m, sizeof(m), "m=audio %d RTP/AVP", ntohs(dest.sin_port)); - snprintf(m2, sizeof(m2), "m=video %d RTP/AVP", ntohs(vdest.sin_port)); - /* Prefer the codec we were requested to use, first, no matter what */ - if (capability & p->prefcodec) { - if (debug) - ast_verbose("Answering/Requesting with root capability 0x%x (%s)\n", p->prefcodec, ast_getformatname(p->prefcodec)); - codec = ast_rtp_lookup_code(p->rtp, 1, p->prefcodec); - if (codec > -1) { - snprintf(costr, sizeof(costr), " %d", codec); - if (p->prefcodec <= AST_FORMAT_MAX_AUDIO) { - strncat(m, costr, sizeof(m) - strlen(m) - 1); - snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, p->prefcodec)); - ast_copy_string(a, costr, sizeof(a)); - } else { - strncat(m2, costr, sizeof(m2) - strlen(m2) - 1); - snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, p->prefcodec)); - ast_copy_string(a2, costr, sizeof(a2)); - } - } - alreadysent |= p->prefcodec; - } - /* Start by sending our preferred codecs */ - for (x = 0 ; x < 32 ; x++) { - if (!(pref_codec = ast_codec_pref_index(&p->prefs,x))) - break; - if ((capability & pref_codec) && !(alreadysent & pref_codec)) { - if (debug) - ast_verbose("Answering with preferred capability 0x%x (%s)\n", pref_codec, ast_getformatname(pref_codec)); - codec = ast_rtp_lookup_code(p->rtp, 1, pref_codec); - if (codec > -1) { - snprintf(costr, sizeof(costr), " %d", codec); - if (pref_codec <= AST_FORMAT_MAX_AUDIO) { - strncat(m, costr, sizeof(m) - strlen(m) - 1); - snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, pref_codec)); - strncat(a, costr, sizeof(a) - strlen(a) - 1); - } else { - strncat(m2, costr, sizeof(m2) - strlen(m2) - 1); - snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, pref_codec)); - strncat(a2, costr, sizeof(a2) - strlen(a) - 1); - } - } - } - alreadysent |= pref_codec; - } - - /* Now send any other common codecs, and non-codec formats: */ - for (x = 1; x <= ((videosupport && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) { - if ((capability & x) && !(alreadysent & x)) { - if (debug) - ast_verbose("Answering with capability 0x%x (%s)\n", x, ast_getformatname(x)); - codec = ast_rtp_lookup_code(p->rtp, 1, x); - if (codec > -1) { - snprintf(costr, sizeof(costr), " %d", codec); - if (x <= AST_FORMAT_MAX_AUDIO) { - strncat(m, costr, sizeof(m) - strlen(m) - 1); - snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x)); - strncat(a, costr, sizeof(a) - strlen(a) - 1); - } else { - strncat(m2, costr, sizeof(m2) - strlen(m2) - 1); - snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x)); - strncat(a2, costr, sizeof(a2) - strlen(a2) - 1); - } - } - } - } - for (x = 1; x <= AST_RTP_MAX; x <<= 1) { - if (p->noncodeccapability & x) { - if (debug) - ast_verbose("Answering with non-codec capability 0x%x (%s)\n", x, ast_rtp_lookup_mime_subtype(0, x)); - codec = ast_rtp_lookup_code(p->rtp, 0, x); - if (codec > -1) { - snprintf(costr, sizeof(costr), " %d", codec); - strncat(m, costr, sizeof(m) - strlen(m) - 1); - snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(0, x)); - strncat(a, costr, sizeof(a) - strlen(a) - 1); - if (x == AST_RTP_DTMF) { - /* Indicate we support DTMF and FLASH... */ - snprintf(costr, sizeof costr, "a=fmtp:%d 0-16\r\n", - codec); - strncat(a, costr, sizeof(a) - strlen(a) - 1); - } - } - } - } - strncat(a, "a=silenceSupp:off - - - -\r\n", sizeof(a) - strlen(a) - 1); - if (strlen(m) < sizeof(m) - 2) - strncat(m, "\r\n", sizeof(m) - strlen(m) - 1); - if (strlen(m2) < sizeof(m2) - 2) - strncat(m2, "\r\n", sizeof(m2) - strlen(m2) - 1); - if ((sizeof(m) <= strlen(m) - 2) || (sizeof(m2) <= strlen(m2) - 2) || (sizeof(a) == strlen(a)) || (sizeof(a2) == strlen(a2))) - ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n"); - len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m) + strlen(a); - if ((p->vrtp) && (!ast_test_flag(p, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */ - len += strlen(m2) + strlen(a2); - snprintf(costr, sizeof(costr), "%d", len); - add_header(resp, "Content-Type", "application/sdp"); - add_header(resp, "Content-Length", costr); - add_line(resp, v); - add_line(resp, o); - add_line(resp, s); - add_line(resp, c); - add_line(resp, t); - add_line(resp, m); - add_line(resp, a); - if ((p->vrtp) && (!ast_test_flag(p, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */ - add_line(resp, m2); - add_line(resp, a2); - } - /* Update lastrtprx when we send our SDP */ - time(&p->lastrtprx); - time(&p->lastrtptx); - return 0; -} - -/*--- copy_request: copy SIP request (mostly used to save request for responses) ---*/ -static void copy_request(struct sip_request *dst, 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; xheaders; x++) - dst->header[x] += offset; - for (x=0; xlines; x++) - dst->line[x] += offset; -} - -/*--- transmit_response_with_sdp: Used for 200 OK and 183 early media ---*/ -static int transmit_response_with_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); - ast_rtp_offered_from_local(p->rtp, 0); - add_sdp(&resp, p); - return send_response(p, &resp, retrans, seqno); -} - -/*--- determine_firstline_parts: parse first line of incoming SIP request */ -static int determine_firstline_parts( struct sip_request *req ) -{ - char *e, *cmd; - int len; - - cmd = ast_skip_blanks(req->header[0]); - if (!*cmd) - return -1; - req->rlPart1 = cmd; - e = ast_skip_nonblanks(cmd); - /* Get the command */ - if (*e) - *e++ = '\0'; - e = ast_skip_blanks(e); - if ( !*e ) - return -1; - - if ( !strcasecmp(cmd, "SIP/2.0") ) { - /* We have a response */ - req->rlPart2 = e; - len = strlen( req->rlPart2 ); - if ( len < 2 ) { - return -1; - } - ast_trim_blanks(e); - } else { - /* We have a request */ - if ( *e == '<' ) { - e++; - if ( !*e ) { - return -1; - } - } - req->rlPart2 = e; /* URI */ - if ( ( e= strrchr( req->rlPart2, 'S' ) ) == NULL ) { - return -1; - } - /* XXX maybe trim_blanks() ? */ - while( isspace( *(--e) ) ) {} - if ( *e == '>' ) { - *e = '\0'; - } else { - *(++e)= '\0'; - } - } - return 1; -} - -/*--- transmit_reinvite_with_sdp: Transmit reinvite with SDP :-) ---*/ -/* 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; - if (ast_test_flag(p, SIP_REINVITE_UPDATE)) - reqprep(&req, p, SIP_UPDATE, 0, 1); - else - reqprep(&req, p, SIP_INVITE, 0, 1); - - add_header(&req, "Allow", ALLOWED_METHODS); - ast_rtp_offered_from_local(p->rtp, 1); - add_sdp(&req, p); - /* Use this as the basis */ - copy_request(&p->initreq, &req); - parse(&p->initreq); - if (sip_debug_test_pvt(p)) - ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); - determine_firstline_parts(&p->initreq); - p->lastinvite = p->ocseq; - ast_set_flag(p, SIP_OUTGOING); - return send_request(p, &req, 1, p->ocseq); -} - -/*--- extract_uri: Check Contact: URI of SIP message ---*/ -static void extract_uri(struct sip_pvt *p, struct sip_request *req) -{ - char stripped[256]=""; - char *c, *n; - ast_copy_string(stripped, get_header(req, "Contact"), sizeof(stripped)); - c = get_in_brackets(stripped); - n = strchr(c, ';'); - if (n) - *n = '\0'; - if (c && !ast_strlen_zero(c)) - ast_copy_string(p->uri, c, sizeof(p->uri)); -} - -/*--- build_contact: Build contact header - the contact header we send out ---*/ -static void build_contact(struct sip_pvt *p) -{ - char iabuf[INET_ADDRSTRLEN]; - - /* Construct Contact: header */ - if (ourport != DEFAULT_SIP_PORT) - snprintf(p->our_contact, sizeof(p->our_contact), "", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport); - else - snprintf(p->our_contact, sizeof(p->our_contact), "", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip)); -} - -/*--- initreqprep: Initiate SIP request to peer/user ---*/ -static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, char *vxml_url) -{ - char invite[256]=""; - char from[256]; - char to[256]; - char tmp[80]; - char iabuf[INET_ADDRSTRLEN]; - char *l = default_callerid, *n=NULL; - int x; - char urioptions[256]=""; - - if (ast_test_flag(p, SIP_USEREQPHONE)) { - char onlydigits = 1; - x=0; - - /* 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 (p->username && p->username[0] == '+') - x=1; - - for (; xusername); x++) { - if (!strchr(AST_DIGIT_ANYNUM, p->username[x])) { - onlydigits = 0; - break; - } - } - - /* If we have only digits, add ;user=phone to the uri */ - if (onlydigits) - strcpy(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 (!l || (!ast_isphonenumber(l) && default_callerid[0])) - l = default_callerid; - /* if user want's his callerid restricted */ - if ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) { - l = CALLERID_UNKNOWN; - n = l; - } - if (!n || 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_copy_string(p->fromuser, l, sizeof(p->fromuser)); - - /* Allow user to be overridden */ - if (!ast_strlen_zero(p->fromname)) - n = p->fromname; - else /* Save for any further attempts */ - ast_copy_string(p->fromname, n, sizeof(p->fromname)); - - if ((ourport != DEFAULT_SIP_PORT) && ast_strlen_zero(p->fromdomain)) - snprintf(from, sizeof(from), "\"%s\" ;tag=as%08x", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, ourport, p->tag); - else - snprintf(from, sizeof(from), "\"%s\" ;tag=as%08x", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, p->tag); - - /* If we're calling a registred SIP peer, use the fullcontact to dial to the peer */ - if (!ast_strlen_zero(p->fullcontact)) { - /* If we have full contact, trust it */ - ast_copy_string(invite, p->fullcontact, sizeof(invite)); - /* Otherwise, use the username while waiting for registration */ - } else if (!ast_strlen_zero(p->username)) { - if (ntohs(p->sa.sin_port) != DEFAULT_SIP_PORT) { - snprintf(invite, sizeof(invite), "sip:%s@%s:%d%s",p->username, p->tohost, ntohs(p->sa.sin_port), urioptions); - } else { - snprintf(invite, sizeof(invite), "sip:%s@%s%s",p->username, p->tohost, urioptions); - } - } else if (ntohs(p->sa.sin_port) != DEFAULT_SIP_PORT) { - snprintf(invite, sizeof(invite), "sip:%s:%d%s", p->tohost, ntohs(p->sa.sin_port), urioptions); - } else { - snprintf(invite, sizeof(invite), "sip:%s%s", p->tohost, urioptions); - } - ast_copy_string(p->uri, invite, sizeof(p->uri)); - /* If there is a VXML URL append it to the SIP URL */ - if (vxml_url) - { - snprintf(to, sizeof(to), "<%s>;%s", invite, vxml_url); - } else { - snprintf(to, sizeof(to), "<%s>", invite); - } - memset(req, 0, sizeof(struct sip_request)); - init_req(req, sipmethod, invite); - 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 */ - add_header(req, "From", from); - ast_copy_string(p->exten, l, sizeof(p->exten)); - build_contact(p); - add_header(req, "To", to); - add_header(req, "Contact", p->our_contact); - add_header(req, "Call-ID", p->callid); - add_header(req, "CSeq", tmp); - add_header(req, "User-Agent", default_useragent); -} - -/*--- transmit_invite: Build REFER/INVITE/OPTIONS message and transmit it ---*/ -static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, struct sip_invite_param *options, int init) -{ - struct sip_request req; - - if (init) { - /* Bump branch even on initial requests */ - p->branch ^= rand(); - build_via(p, p->via, sizeof(p->via)); - initreqprep(&req, p, sipmethod, options ? options->vxml_url : (char *) NULL); - } else - reqprep(&req, p, sipmethod, 0, 1); - - if (options && options->auth) - add_header(&req, options->authheader, options->auth); - append_date(&req); - if (sipmethod == SIP_REFER) { /* Call transfer */ - if (!ast_strlen_zero(p->refer_to)) - add_header(&req, "Refer-To", p->refer_to); - if (!ast_strlen_zero(p->referred_by)) - add_header(&req, "Referred-By", p->referred_by); - } -#ifdef OSP_SUPPORT - if (options && options->osptoken && !ast_strlen_zero(options->osptoken)) { - ast_log(LOG_DEBUG,"Adding OSP Token: %s\n", options->osptoken); - add_header(&req, "P-OSP-Auth-Token", options->osptoken); - } else { - ast_log(LOG_DEBUG,"NOT Adding OSP Token\n"); - } -#endif - if (options && options->distinctive_ring && !ast_strlen_zero(options->distinctive_ring)) - { - add_header(&req, "Alert-Info", options->distinctive_ring); - } - add_header(&req, "Allow", ALLOWED_METHODS); - if (options && options->addsipheaders && init) { - struct ast_channel *ast; - char *header = (char *) NULL; - char *content = (char *) NULL; - char *end = (char *) NULL; - struct varshead *headp = (struct varshead *) NULL; - struct ast_var_t *current; - - ast = p->owner; /* The owner channel */ - if (ast) { - headp=&ast->varshead; - if (!headp) - ast_log(LOG_WARNING,"No Headp for the channel...ooops!\n"); - else { - AST_LIST_TRAVERSE(headp,current,entries) { - /* SIPADDHEADER: Add SIP header to outgoing call */ - if (!strncasecmp(ast_var_name(current),"SIPADDHEADER",strlen("SIPADDHEADER"))) { - header = ast_var_value(current); - /* Strip of the starting " (if it's there) */ - if (*header == '"') - header++; - if ((content = strchr(header, ':'))) { - *content = '\0'; - content++; /* Move pointer ahead */ - /* Skip white space */ - while (*content == ' ') - content++; - /* Strip the ending " (if it's there) */ - end = content + strlen(content) -1; - if (*end == '"') - *end = '\0'; - - add_header(&req, header, content); - if (sipdebug) - ast_log(LOG_DEBUG, "Adding SIP Header \"%s\" with content :%s: \n", header, content); - } - } - } - } - } - } - if (sdp) { - ast_rtp_offered_from_local(p->rtp, 1); - add_sdp(&req, p); - } else { - add_header(&req, "Content-Length", "0"); - add_blank_header(&req); - } - - if (!p->initreq.headers) { - /* Use this as the basis */ - copy_request(&p->initreq, &req); - parse(&p->initreq); - if (sip_debug_test_pvt(p)) - ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); - determine_firstline_parts(&p->initreq); - } - p->lastinvite = p->ocseq; - return send_request(p, &req, init ? 2 : 1, p->ocseq); -} - -/*--- transmit_state_notify: Used in the SUBSCRIBE notification subsystem ----*/ -static int transmit_state_notify(struct sip_pvt *p, int state, int full) -{ - char tmp[4000]; - int maxbytes = 0; - int bytes = 0; - char from[256], to[256]; - char *t, *c, *a; - char *mfrom, *mto; - struct sip_request req; - char clen[20]; - - memset(from, 0, sizeof(from)); - memset(to, 0, sizeof(to)); - ast_copy_string(from, get_header(&p->initreq, "From"), sizeof(from)); - - c = ditch_braces(from); - if (strncmp(c, "sip:", 4)) { - ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c); - return -1; - } - if ((a = strchr(c, ';'))) { - *a = '\0'; - } - mfrom = c; - - reqprep(&req, p, SIP_NOTIFY, 0, 1); - - if (p->subscribed == 1) { - ast_copy_string(to, get_header(&p->initreq, "To"), sizeof(to)); - - c = ditch_braces(to); - if (strncmp(c, "sip:", 4)) { - ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c); - return -1; - } - if ((a = strchr(c, ';'))) { - *a = '\0'; - } - mto = c; - - add_header(&req, "Event", "presence"); - add_header(&req, "Subscription-State", "active"); - add_header(&req, "Content-Type", "application/xpidf+xml"); - - if ((state==AST_EXTENSION_UNAVAILABLE) || (state==AST_EXTENSION_BUSY)) - state = 2; - else if (state==AST_EXTENSION_INUSE) - state = 1; - else - state = 0; - - t = tmp; - maxbytes = sizeof(tmp); - bytes = snprintf(t, maxbytes, "\n"); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "\n"); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "\n"); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "\n", mfrom); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "\n", p->exten); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "
\n", mto); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "\n", !state ? "open" : (state==1) ? "inuse" : "closed"); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "\n", !state ? "online" : (state==1) ? "onthephone" : "offline"); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "
\n\n\n"); - } else { - add_header(&req, "Event", "dialog"); - add_header(&req, "Content-Type", "application/dialog-info+xml"); - - t = tmp; - maxbytes = sizeof(tmp); - bytes = snprintf(t, maxbytes, "\n"); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "\n", p->dialogver++, full ? "full":"partial", mfrom); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "\n", p->exten); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "%s\n", state ? "confirmed" : "terminated"); - t += bytes; - maxbytes -= bytes; - bytes = snprintf(t, maxbytes, "\n\n"); - } - if (t > tmp + sizeof(tmp)) - ast_log(LOG_WARNING, "Buffer overflow detected!! (Please file a bug report)\n"); - - snprintf(clen, sizeof(clen), "%d", (int)strlen(tmp)); - add_header(&req, "Content-Length", clen); - add_line(&req, tmp); - - return send_request(p, &req, 1, p->ocseq); -} - -/*--- transmit_notify_with_mwi: Notify user of messages waiting in voicemail ---*/ -/* Notification only works for registred 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) -{ - struct sip_request req; - char tmp[256]; - char tmp2[256]; - char clen[20]; - initreqprep(&req, p, SIP_NOTIFY, NULL); - add_header(&req, "Event", "message-summary"); - add_header(&req, "Content-Type", default_notifymime); - - snprintf(tmp, sizeof(tmp), "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no"); - snprintf(tmp2, sizeof(tmp2), "Voice-Message: %d/%d (0/0)\r\n", newmsgs, oldmsgs); - snprintf(clen, sizeof(clen), "%d", (int)(strlen(tmp) + strlen(tmp2))); - add_header(&req, "Content-Length", clen); - add_line(&req, tmp); - add_line(&req, tmp2); - - if (!p->initreq.headers) { - /* Use this as the basis */ - copy_request(&p->initreq, &req); - parse(&p->initreq); - if (sip_debug_test_pvt(p)) - ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); - determine_firstline_parts(&p->initreq); - } - - return send_request(p, &req, 1, p->ocseq); -} - -/*--- transmit_sip_request: Transmit SIP request */ -static int transmit_sip_request(struct sip_pvt *p,struct sip_request *req) -{ - if (!p->initreq.headers) { - /* Use this as the basis */ - copy_request(&p->initreq, req); - parse(&p->initreq); - if (sip_debug_test_pvt(p)) - ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); - determine_firstline_parts(&p->initreq); - } - - return send_request(p, req, 0, p->ocseq); -} - -/*--- transmit_notify_with_sipfrag: Notify a transferring party of the status of trasnfer ---*/ -/* Apparently the draft SIP REFER structure was too simple, so it was decided that the - * status of transfers also needed to be sent via NOTIFY instead of just the 202 Accepted - * that had worked heretofore. - */ -static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq) -{ - struct sip_request req; - char tmp[256]; - char clen[20]; - 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", "terminated;reason=noresource"); - add_header(&req, "Content-Type", "message/sipfrag;version=2.0"); - - ast_copy_string(tmp, "SIP/2.0 200 OK", sizeof(tmp)); - snprintf(clen, sizeof(clen), "%d", (int)(strlen(tmp))); - add_header(&req, "Content-Length", clen); - add_line(&req, tmp); - - if (!p->initreq.headers) { - /* Use this as the basis */ - copy_request(&p->initreq, &req); - parse(&p->initreq); - if (sip_debug_test_pvt(p)) - ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); - determine_firstline_parts(&p->initreq); - } - - return send_request(p, &req, 1, p->ocseq); -} - -static char *regstate2str(int 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"; - } -} - -static int transmit_register(struct sip_registry *r, int sipmethod, char *auth, char *authheader); - -/*--- sip_reregister: Update registration with SIP Proxy---*/ -static int sip_reregister(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; - - /* 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; -} - -/*--- __sip_do_register: 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; -} - -/*--- sip_reg_timeout: Registration timeout, register again */ -static int sip_reg_timeout(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; - if (p->registry) - ASTOBJ_UNREF(p->registry, sip_registry_destroy); - r->call = NULL; - ast_set_flag(p, SIP_NEEDDESTROY); - /* Pretend to ACK anything just in case */ - /* OEJ: Ack what??? */ - __sip_pretend_ack(p); - } - /* 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", "Channel: SIP\r\nUser: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate)); - ASTOBJ_UNREF(r,sip_registry_destroy); - return 0; -} - -/*--- transmit_register: Transmit register to SIP proxy or UA ---*/ -static int transmit_register(struct sip_registry *r, int sipmethod, char *auth, char *authheader) -{ - struct sip_request req; - char from[256]; - char to[256]; - char tmp[80]; - char via[80]; - char addr[80]; - struct sip_pvt *p; - - /* 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))) { - 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; - p->tag = rand(); /* create a new local tag for every register attempt */ - p->theirtag[0]='\0'; /* forget their old tag, so we don't match tags when getting response */ - } - } else { - /* Build callid for registration if we haven't registred before */ - if (!r->callid_valid) { - build_callid(r->callid, sizeof(r->callid), __ourip, default_fromdomain); - r->callid_valid = 1; - } - /* Allocate SIP packet for registration */ - p=sip_alloc( r->callid, NULL, 0, SIP_REGISTER); - if (!p) { - ast_log(LOG_WARNING, "Unable to allocate registration call\n"); - return 0; - } - /* 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_sched_del(sched, r->timeout); - r->timeout = ast_sched_add(sched, global_reg_timeout*1000, sip_reg_timeout, r); - ast_log(LOG_WARNING, "Still have a registration timeout for %s@%s (create_addr() error), %d\n", r->username, r->hostname, r->timeout); - } else { - r->timeout = ast_sched_add(sched, global_reg_timeout*1000, sip_reg_timeout, r); - 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); - } - r->regattempts++; - return 0; - } - /* Copy back Call-ID in case create_addr changed it */ - ast_copy_string(r->callid, p->callid, sizeof(r->callid)); - if (r->portno) - p->sa.sin_port = htons(r->portno); - ast_set_flag(p, 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_copy_string(p->peersecret, r->secret, sizeof(p->peersecret)); - if (!ast_strlen_zero(r->md5secret)) - ast_copy_string(p->peermd5secret, r->md5secret, sizeof(p->peermd5secret)); - /* User name in this realm - - if authuser is set, use that, otherwise use username */ - if (!ast_strlen_zero(r->authuser)) { - ast_copy_string(p->peername, r->authuser, sizeof(p->peername)); - ast_copy_string(p->authname, r->authuser, sizeof(p->authname)); - } else { - if (!ast_strlen_zero(r->username)) { - ast_copy_string(p->peername, r->username, sizeof(p->peername)); - ast_copy_string(p->authname, r->username, sizeof(p->authname)); - ast_copy_string(p->fromuser, r->username, sizeof(p->fromuser)); - } - } - if (!ast_strlen_zero(r->username)) - ast_copy_string(p->username, r->username, sizeof(p->username)); - /* Save extension in packet */ - ast_copy_string(p->exten, r->contact, sizeof(p->exten)); - - /* - 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)) - memcpy(&p->ourip, &bindaddr.sin_addr, sizeof(p->ourip)); - build_contact(p); - } - - /* set up a timeout */ - if (auth == NULL) { - if (r->timeout > -1) { - ast_log(LOG_WARNING, "Still have a registration timeout, %d\n", r->timeout); - ast_sched_del(sched, r->timeout); - } - r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, r); - ast_log(LOG_DEBUG, "Scheduled a registration timeout for %s : %d\n", r->hostname, r->timeout); - } - - if (strchr(r->username, '@')) { - snprintf(from, sizeof(from), ";tag=as%08x", r->username, p->tag); - if (!ast_strlen_zero(p->theirtag)) - snprintf(to, sizeof(to), ";tag=%s", r->username, p->theirtag); - else - snprintf(to, sizeof(to), "", r->username); - } else { - snprintf(from, sizeof(from), ";tag=as%08x", r->username, p->tohost, p->tag); - if (!ast_strlen_zero(p->theirtag)) - snprintf(to, sizeof(to), ";tag=%s", r->username, p->tohost, p->theirtag); - else - snprintf(to, sizeof(to), "", r->username, p->tohost); - } - - /* Fromdomain is what we are registering to, regardless of actual - host name from SRV */ - if (p->fromdomain && !ast_strlen_zero(p->fromdomain)) - snprintf(addr, sizeof(addr), "sip:%s", p->fromdomain); - else - snprintf(addr, sizeof(addr), "sip:%s", r->hostname); - ast_copy_string(p->uri, addr, sizeof(p->uri)); - - p->branch ^= rand(); - - memset(&req, 0, sizeof(req)); - 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, via, sizeof(via)); - add_header(&req, "Via", via); - add_header(&req, "From", from); - add_header(&req, "To", to); - add_header(&req, "Call-ID", p->callid); - add_header(&req, "CSeq", tmp); - add_header(&req, "User-Agent", default_useragent); - - - 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_copy_string(p->realm, r->realm, sizeof(p->realm)); - ast_copy_string(p->nonce, r->nonce, sizeof(p->nonce)); - ast_copy_string(p->domain, r->domain, sizeof(p->domain)); - ast_copy_string(p->opaque, r->opaque, sizeof(p->opaque)); - ast_copy_string(p->qop, r->qop, sizeof(p->qop)); - - memset(digest,0,sizeof(digest)); - build_reply_digest(p, sipmethod, digest, sizeof(digest)); - add_header(&req, "Authorization", digest); - - } - - 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(&req, "Content-Length", "0"); - add_blank_header(&req); - copy_request(&p->initreq, &req); - parse(&p->initreq); - if (sip_debug_test_pvt(p)) { - ast_verbose("REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines); - } - determine_firstline_parts(&p->initreq); - 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, 2, p->ocseq); -} - -/*--- transmit_message_with_text: 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, 1, p->ocseq); -} - -/*--- transmit_refer: Transmit SIP REFER message ---*/ -static int transmit_refer(struct sip_pvt *p, const char *dest) -{ - struct sip_request req; - char from[256]; - char *of, *c; - char referto[256]; - char tmp[80]; - - if (ast_test_flag(p, SIP_OUTGOING)) - of = get_header(&p->initreq, "To"); - else - of = get_header(&p->initreq, "From"); - ast_copy_string(from, of, sizeof(from)); - of = ditch_braces(from); - ast_copy_string(p->from,of,sizeof(p->from)); - if (strncmp(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'; - c++; - } - if (c) { - snprintf(referto, sizeof(referto), "", dest, c); - } else { - snprintf(referto, sizeof(referto), "", dest); - } - - ast_copy_string(tmp, get_header(&p->initreq, "Max-Forwards"), sizeof(tmp)); - if (strlen(tmp) && atoi(tmp)) { - p->maxforwards = atoi(tmp) - 1; - } else { - p->maxforwards = DEFAULT_MAX_FORWARDS - 1; - } - if (p->maxforwards > -1) { - /* save in case we get 407 challenge */ - ast_copy_string(p->refer_to, referto, sizeof(p->refer_to)); - ast_copy_string(p->referred_by, p->our_contact, sizeof(p->referred_by)); - - reqprep(&req, p, SIP_REFER, 0, 1); - add_header(&req, "Refer-To", referto); - if (!ast_strlen_zero(p->our_contact)) - add_header(&req, "Referred-By", p->our_contact); - add_blank_header(&req); - return send_request(p, &req, 1, p->ocseq); - } else { - return -1; - } -} - -/*--- transmit_info_with_digit: Send SIP INFO dtmf message, see Cisco documentation on cisco.co -m ---*/ -static int transmit_info_with_digit(struct sip_pvt *p, char digit) -{ - struct sip_request req; - reqprep(&req, p, SIP_INFO, 0, 1); - add_digit(&req, digit); - return send_request(p, &req, 1, p->ocseq); -} - -/*--- transmit_request: transmit generic SIP request ---*/ -static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, int reliable, int newbranch) -{ - struct sip_request resp; - reqprep(&resp, p, sipmethod, seqno, newbranch); - add_header(&resp, "Content-Length", "0"); - add_blank_header(&resp); - return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq); -} - -/*--- transmit_request_with_auth: Transmit SIP request, auth added ---*/ -static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, int reliable, int newbranch) -{ - struct sip_request resp; - - reqprep(&resp, p, sipmethod, seqno, newbranch); - if (*p->realm) - { - char digest[1024]; - - memset(digest, 0, sizeof(digest)); - build_reply_digest(p, sipmethod, digest, sizeof(digest)); - add_header(&resp, "Proxy-Authorization", digest); - } - - add_header(&resp, "Content-Length", "0"); - add_blank_header(&resp); - return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq); -} - -/*--- expire_register: Expire registration of SIP peer ---*/ -static int expire_register(void *data) -{ - struct sip_peer *peer = data; - - memset(&peer->addr, 0, sizeof(peer->addr)); - ast_db_del("SIP/Registry", peer->name); - manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name); - register_peer_exten(peer, 0); - peer->expire = -1; - ast_device_state_changed("SIP/%s", peer->name); - if (ast_test_flag(peer, SIP_SELFDESTRUCT) || ast_test_flag((&peer->flags_page2), SIP_PAGE2_RTAUTOCLEAR)) { - peer = ASTOBJ_CONTAINER_UNLINK(&peerl, peer); - ASTOBJ_UNREF(peer, sip_destroy_peer); - } - - return 0; -} - -static int sip_poke_peer(struct sip_peer *peer); - -static int sip_poke_peer_s(void *data) -{ - struct sip_peer *peer = data; - peer->pokeexpire = -1; - sip_poke_peer(peer); - return 0; -} - -/*--- reg_source_db: Get registration details from Asterisk DB ---*/ -static void reg_source_db(struct sip_peer *peer) -{ - char data[256]; - char iabuf[INET_ADDRSTRLEN]; - struct in_addr in; - int expiry; - int port; - char *scan, *addr, *port_str, *expiry_str, *username, *contact; - - 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_verbose > 2) - ast_verbose(VERBOSE_PREFIX_3 "SIP Seeding peer from astdb: '%s' at %s@%s:%d for %d\n", - peer->name, peer->username, ast_inet_ntoa(iabuf, sizeof(iabuf), 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 (peer->pokeexpire > -1) - ast_sched_del(sched, peer->pokeexpire); - peer->pokeexpire = ast_sched_add(sched, rand() % 5000 + 1, sip_poke_peer_s, peer); - } else - sip_poke_peer(peer); - if (peer->expire > -1) - ast_sched_del(sched, peer->expire); - peer->expire = ast_sched_add(sched, (expiry + 10) * 1000, expire_register, peer); - register_peer_exten(peer, 1); -} - -/*--- parse_ok_contact: Parse contact header for 200 OK on INVITE ---*/ -static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req) -{ - char contact[250]= ""; - char *c, *n, *pt; - int port; - struct hostent *hp; - struct ast_hostent ahp; - struct sockaddr_in oldsin; - - /* 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_copy_string(pvt->fullcontact, c, sizeof(pvt->fullcontact)); - - /* Save URI for later ACKs, BYE or RE-invites */ - ast_copy_string(pvt->okcontacturi, c, sizeof(pvt->okcontacturi)); - - /* Make sure it's a SIP URL */ - if (strncasecmp(c, "sip:", 4)) { - ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", c); - } else - c += 4; - - /* Ditch arguments */ - n = strchr(c, ';'); - if (n) - *n = '\0'; - - /* Grab host */ - n = strchr(c, '@'); - if (!n) { - n = c; - c = NULL; - } else { - *n = '\0'; - n++; - } - pt = strchr(n, ':'); - if (pt) { - *pt = '\0'; - pt++; - port = atoi(pt); - } else - port = DEFAULT_SIP_PORT; - - memcpy(&oldsin, &pvt->sa, sizeof(oldsin)); - - if (!(ast_test_flag(pvt, SIP_NAT) & SIP_NAT_ROUTE)) { - /* 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(n, &ahp); - if (!hp) { - ast_log(LOG_WARNING, "Invalid host '%s'\n", n); - return -1; - } - pvt->sa.sin_family = AF_INET; - memcpy(&pvt->sa.sin_addr, hp->h_addr, sizeof(pvt->sa.sin_addr)); - pvt->sa.sin_port = htons(port); - } else { - /* Don't trust the contact field. Just use what they came to us - with. */ - memcpy(&pvt->sa, &pvt->recv, sizeof(pvt->sa)); - } - return 0; -} - - -/*--- parse_contact: Parse contact header and save registration ---*/ -static int parse_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req) -{ - char contact[80]= ""; - char data[256]; - char iabuf[INET_ADDRSTRLEN]; - char *expires = get_header(req, "Expires"); - int expiry = atoi(expires); - char *c, *n, *pt; - int port; - char *useragent; - struct hostent *hp; - struct ast_hostent ahp; - struct sockaddr_in oldsin; - - if (ast_strlen_zero(expires)) { /* No expires header */ - expires = strstr(get_header(req, "Contact"), "expires="); - if (expires) { - if (sscanf(expires + 8, "%d;", &expiry) != 1) - expiry = default_expiry; - } else { - /* Nothing has been specified */ - expiry = default_expiry; - } - } - /* Look for brackets */ - ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact)); - c = get_in_brackets(contact); - - if (!strcasecmp(c, "*") || !expiry) { /* Unregister this peer */ - /* This means remove all registrations and return OK */ - memset(&p->addr, 0, sizeof(p->addr)); - if (p->expire > -1) - ast_sched_del(sched, p->expire); - p->expire = -1; - ast_db_del("SIP/Registry", p->name); - register_peer_exten(p, 0); - p->fullcontact[0] = '\0'; - p->useragent[0] = '\0'; - p->sipoptions = 0; - p->lastms = 0; - - if (option_verbose > 2) - ast_verbose(VERBOSE_PREFIX_3 "Unregistered SIP '%s'\n", p->name); - manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unregistered\r\n", p->name); - return 0; - } - ast_copy_string(p->fullcontact, c, sizeof(p->fullcontact)); - /* For the 200 OK, we should use the received contact */ - snprintf(pvt->our_contact, sizeof(pvt->our_contact) - 1, "<%s>", c); - /* Make sure it's a SIP URL */ - if (strncasecmp(c, "sip:", 4)) { - ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", c); - } else - c += 4; - /* Ditch q */ - n = strchr(c, ';'); - if (n) { - *n = '\0'; - } - /* Grab host */ - n = strchr(c, '@'); - if (!n) { - n = c; - c = NULL; - } else { - *n = '\0'; - n++; - } - pt = strchr(n, ':'); - if (pt) { - *pt = '\0'; - pt++; - port = atoi(pt); - } else - port = DEFAULT_SIP_PORT; - memcpy(&oldsin, &p->addr, sizeof(oldsin)); - if (!(ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)) { - /* 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); - return -1; - } - p->addr.sin_family = AF_INET; - memcpy(&p->addr.sin_addr, hp->h_addr, sizeof(p->addr.sin_addr)); - p->addr.sin_port = htons(port); - } else { - /* Don't trust the contact field. Just use what they came to us - with */ - memcpy(&p->addr, &pvt->recv, sizeof(p->addr)); - } - - if (c) /* Overwrite the default username from config at registration */ - ast_copy_string(p->username, c, sizeof(p->username)); - else - p->username[0] = '\0'; - - if (p->expire > -1) - ast_sched_del(sched, p->expire); - if ((expiry < 1) || (expiry > max_expiry)) - expiry = max_expiry; - if (!ast_test_flag(p, SIP_REALTIME)) - p->expire = ast_sched_add(sched, (expiry + 10) * 1000, expire_register, p); - else - p->expire = -1; - pvt->expiry = expiry; - snprintf(data, sizeof(data), "%s:%d:%d:%s:%s", ast_inet_ntoa(iabuf, sizeof(iabuf), p->addr.sin_addr), ntohs(p->addr.sin_port), expiry, p->username, p->fullcontact); - if (!(ast_test_flag(p, SIP_REALTIME) && ast_test_flag((&p->flags_page2), SIP_PAGE2_RTCACHEFRIENDS))) - ast_db_put("SIP/Registry", p->name, data); - manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Registered\r\n", p->name); - if (inaddrcmp(&p->addr, &oldsin)) { - sip_poke_peer(p); - if (option_verbose > 2) - ast_verbose(VERBOSE_PREFIX_3 "Registered SIP '%s' at %s port %d expires %d\n", p->name, ast_inet_ntoa(iabuf, sizeof(iabuf), p->addr.sin_addr), ntohs(p->addr.sin_port), expiry); - register_peer_exten(p, 1); - } - - /* Save SIP options profile */ - p->sipoptions = pvt->sipoptions; - - /* Save User agent */ - useragent = get_header(req, "User-Agent"); - if (useragent && strcasecmp(useragent, p->useragent)) { - ast_copy_string(p->useragent, useragent, sizeof(p->useragent)); - if (option_verbose > 3) { - ast_verbose(VERBOSE_PREFIX_3 "Saved useragent \"%s\" for peer %s\n",p->useragent,p->name); - } - } - return 0; -} - -/*--- free_old_route: 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; - } -} - -/*--- list_route: List all routes - mostly for debugging ---*/ -static void list_route(struct sip_route *route) -{ - if (!route) { - ast_verbose("list_route: no route\n"); - return; - } - while (route) { - ast_verbose("list_route: hop: <%s>\n", route->hop); - route = route->next; - } -} - -/*--- build_route: 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; - char *rr, *contact, *c; - - /* Once a persistant route is set, don't fool with it */ - if (p->route && p->route_persistant) { - 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; - } - - p->route_persistant = backwards; - - /* We build up head, then assign it to p->route when we're done */ - 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 (;;) { - /* Each route entry */ - /* Find < */ - rr = strchr(rr, '<'); - if (!rr) break; /* No more hops */ - ++rr; - len = strcspn(rr, ">") + 1; - /* Make a struct route */ - thishop = malloc(sizeof(*thishop) + len); - if (thishop) { - ast_copy_string(thishop->hop, rr, len); - 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; - } - } - rr += len; - } - } - /* 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)) { - 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; - } - thishop = malloc(sizeof(*thishop) + len); - if (thishop) { - 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); -} - -/*--- check_auth: Check user authorization from peer definition ---*/ -/* Some actions, like REGISTER and INVITEs from peers require - authentication (if peer have secret set) */ -static int check_auth(struct sip_pvt *p, struct sip_request *req, char *randdata, int randlen, char *username, char *secret, char *md5secret, int sipmethod, char *uri, int reliable, int ignore) -{ - int res = -1; - char *response = "407 Proxy Authentication Required"; - char *reqheader = "Proxy-Authorization"; - char *respheader = "Proxy-Authenticate"; - char *authtoken; -#ifdef OSP_SUPPORT - char tmp[80]; - char *osptoken; - unsigned int osptimelimit; -#endif - /* Always OK if no secret */ - if (ast_strlen_zero(secret) && ast_strlen_zero(md5secret) -#ifdef OSP_SUPPORT - && !ast_test_flag(p, SIP_OSPAUTH) - && global_allowguest != 2 -#endif - ) - return 0; - if (sipmethod == SIP_REGISTER) { - /* 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"; - } -#ifdef OSP_SUPPORT - else if (ast_test_flag(p, SIP_OSPAUTH)) { - ast_log(LOG_DEBUG, "Checking OSP Authentication!\n"); - osptoken = get_header(req, "P-OSP-Auth-Token"); - /* Check for token existence */ - if (ast_strlen_zero(osptoken)) - return -1; - /* Validate token */ - if (ast_osp_validate(NULL, osptoken, &p->osphandle, &osptimelimit, p->cid_num, p->sa.sin_addr, p->exten) < 1) - return -1; - - snprintf(tmp, sizeof(tmp), "%d", p->osphandle); - pbx_builtin_setvar_helper(p->owner, "_OSPHANDLE", tmp); - - - /* If ospauth is 'exclusive' don't require further authentication */ - if ((ast_test_flag(p, SIP_OSPAUTH) == SIP_OSPAUTH_EXCLUSIVE) || - (ast_strlen_zero(secret) && ast_strlen_zero(md5secret))) - return 0; - } -#endif - authtoken = get_header(req, reqheader); - if (ignore && !ast_strlen_zero(randdata) && ast_strlen_zero(authtoken)) { - /* This is a retransmitted invite/register/etc, don't reconstruct authentication - information */ - if (!ast_strlen_zero(randdata)) { - if (!reliable) { - /* Resend message if this was NOT a reliable delivery. Otherwise the - retransmission should get it */ - transmit_response_with_auth(p, response, req, randdata, reliable, respheader, 0); - /* Schedule auto destroy in 15 seconds */ - sip_scheddestroy(p, 15000); - } - res = 1; - } - } else if (ast_strlen_zero(randdata) || ast_strlen_zero(authtoken)) { - snprintf(randdata, randlen, "%08x", rand()); - transmit_response_with_auth(p, response, req, randdata, reliable, respheader, 0); - /* Schedule auto destroy in 15 seconds */ - sip_scheddestroy(p, 15000); - res = 1; - } else { - /* 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. */ - char a1[256]; - char a2[256]; - char a1_hash[256]; - char a2_hash[256]; - char resp[256]; - char resp_hash[256]=""; - char tmp[256] = ""; - char *c; - char *z; - char *ua_hash =""; - char *resp_uri =""; - char *nonce = ""; - char *digestusername = ""; - int wrongnonce = 0; - char *usednonce = randdata; - - /* Find their response among the mess that we'r sent for comparison */ - ast_copy_string(tmp, authtoken, sizeof(tmp)); - c = tmp; - - while(c) { - c = ast_skip_blanks(c); - if (!*c) - break; - if (!strncasecmp(c, "response=", strlen("response="))) { - c+= strlen("response="); - if ((*c == '\"')) { - ua_hash=++c; - if ((c = strchr(c,'\"'))) - *c = '\0'; - - } else { - ua_hash=c; - if ((c = strchr(c,','))) - *c = '\0'; - } - - } else if (!strncasecmp(c, "uri=", strlen("uri="))) { - c+= strlen("uri="); - if ((*c == '\"')) { - resp_uri=++c; - if ((c = strchr(c,'\"'))) - *c = '\0'; - } else { - resp_uri=c; - if ((c = strchr(c,','))) - *c = '\0'; - } - - } else if (!strncasecmp(c, "username=", strlen("username="))) { - c+= strlen("username="); - if ((*c == '\"')) { - digestusername=++c; - if((c = strchr(c,'\"'))) - *c = '\0'; - } else { - digestusername=c; - if((c = strchr(c,','))) - *c = '\0'; - } - } else if (!strncasecmp(c, "nonce=", strlen("nonce="))) { - c+= strlen("nonce="); - if ((*c == '\"')) { - nonce=++c; - if ((c = strchr(c,'\"'))) - *c = '\0'; - } else { - nonce=c; - if ((c = strchr(c,','))) - *c = '\0'; - } - - } else - if ((z = strchr(c,' ')) || (z = strchr(c,','))) c=z; - if (c) - c++; - } - /* Verify that digest username matches the username we auth as */ - if (strcmp(username, digestusername)) { - /* Oops, we're trying something here */ - return -2; - } - - /* Verify nonce from request matches our nonce. If not, send 401 with new nonce */ - if (strncasecmp(randdata, nonce, randlen)) { - wrongnonce = 1; - usednonce = nonce; - } - - snprintf(a1, sizeof(a1), "%s:%s:%s", username, global_realm, secret); - - if (!ast_strlen_zero(resp_uri)) - snprintf(a2, sizeof(a2), "%s:%s", sip_methods[sipmethod].text, resp_uri); - else - snprintf(a2, sizeof(a2), "%s:%s", sip_methods[sipmethod].text, uri); - - if (!ast_strlen_zero(md5secret)) - snprintf(a1_hash, sizeof(a1_hash), "%s", md5secret); - else - ast_md5_hash(a1_hash, a1); - - ast_md5_hash(a2_hash, a2); - - snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, usednonce, a2_hash); - ast_md5_hash(resp_hash, resp); - - if (wrongnonce) { - ast_log(LOG_NOTICE, "stale nonce received from '%s'\n", get_header(req, "To")); - - snprintf(randdata, randlen, "%08x", rand()); - if (ua_hash && !strncasecmp(ua_hash, resp_hash, strlen(resp_hash))) { - /* We got working auth token, based on stale nonce . */ - transmit_response_with_auth(p, response, req, randdata, reliable, respheader, 1); - } else { - /* Everything was wrong, so give the device one more try */ - transmit_response_with_auth(p, response, req, randdata, reliable, respheader, 0); - } - - /* Schedule auto destroy in 15 seconds */ - sip_scheddestroy(p, 15000); - return 1; - } - /* resp_hash now has the expected response, compare the two */ - if (ua_hash && !strncasecmp(ua_hash, resp_hash, strlen(resp_hash))) { - /* Auth is OK */ - res = 0; - } - } - /* Failure */ - return res; -} - -/*--- cb_extensionstate: Part of thte SUBSCRIBE support subsystem ---*/ -static int cb_extensionstate(char *context, char* exten, int state, void *data) -{ - struct sip_pvt *p = data; - if (state == -1) { - sip_scheddestroy(p, 15000); - p->stateid = -1; - return 0; - } - - transmit_state_notify(p, state, 1); - - if (option_debug > 1) - ast_verbose(VERBOSE_PREFIX_1 "Extension Changed %s new state %d for Notify User %s\n", exten, state, p->username); - return 0; -} - -/*--- register_verify: Verify registration of user */ -static int register_verify(struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req, char *uri, int ignore) -{ - int res = -1; - struct sip_peer *peer; - char tmp[256] = ""; - char iabuf[INET_ADDRSTRLEN]; - char *name, *c; - char *t; - /* Terminate URI */ - t = uri; - while(*t && (*t > 32) && (*t != ';')) - t++; - *t = '\0'; - - ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp)); - c = ditch_braces(tmp); - /* Ditch ;user=phone */ - name = strchr(c, ';'); - if (name) - *name = '\0'; - - if (!strncmp(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(iabuf, sizeof(iabuf), sin->sin_addr)); - } - /* Strip off the domain name */ - c = strchr(name, '@'); - if (c) - *c = '\0'; - ast_copy_string(p->exten, name, sizeof(p->exten)); - build_contact(p); - peer = find_peer(name, NULL, 1); - if (!(peer && ast_apply_ha(peer->ha, sin))) { - if (peer) - ASTOBJ_UNREF(peer,sip_destroy_peer); - } - if (peer) { - if (!ast_test_flag(peer, SIP_DYNAMIC)) { - ast_log(LOG_NOTICE, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name); - } else { - ast_copy_flags(p, peer, SIP_NAT); - transmit_response(p, "100 Trying", req); - if (!(res = check_auth(p, req, p->randdata, sizeof(p->randdata), peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri, 0, ignore))) { - sip_cancel_destroy(p); - if (parse_contact(p, peer, req)) { - ast_log(LOG_WARNING, "Failed to parse contact info\n"); - } else { - update_peer(peer, p->expiry); - /* Say OK and ask subsystem to retransmit msg counter */ - transmit_response_with_date(p, "200 OK", req); - peer->lastmsgssent = -1; - res = 0; - } - } - } - } - if (!peer && autocreatepeer) { - /* Create peer if we have autocreate mode enabled */ - peer = temp_peer(name); - if (peer) { - ASTOBJ_CONTAINER_LINK(&peerl, peer); - peer->lastmsgssent = -1; - sip_cancel_destroy(p); - if (parse_contact(p, peer, req)) { - ast_log(LOG_WARNING, "Failed to parse contact info\n"); - } else { - /* Say OK and ask subsystem to retransmit msg counter */ - manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Registered\r\n", peer->name); - transmit_response_with_date(p, "200 OK", req); - peer->lastmsgssent = -1; - res = 0; - } - } - } - if (!res) { - ast_device_state_changed("SIP/%s", peer->name); - } - if (res < 0) { - switch (res) { - case -1: - /* Wrong password in authentication. Go away, don't try again until you fixed it */ - transmit_response(p, "403 Forbidden", &p->initreq); - break; - case -2: - /* 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; - } - } - if (peer) - ASTOBJ_UNREF(peer,sip_destroy_peer); - return res; -} - -/*--- get_rdnis: 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 = ditch_braces(tmp); - if (strncmp(c, "sip:", 4)) { - ast_log(LOG_WARNING, "Huh? Not an RDNIS SIP header (%s)?\n", c); - return -1; - } - c += 4; - if ((a = strchr(c, '@')) || (a = strchr(c, ';'))) { - *a = '\0'; - } - if (sip_debug_test_pvt(p)) - ast_verbose("RDNIS is %s\n", c); - ast_copy_string(p->rdnis, c, sizeof(p->rdnis)); - - return 0; -} - -/*--- get_destination: Find out who the call is for --*/ -static int get_destination(struct sip_pvt *p, struct sip_request *oreq) -{ - char tmp[256] = "", *c, *a; - char tmpf[256]= "", *fr; - struct sip_request *req; - - req = oreq; - if (!req) - req = &p->initreq; - if (req->rlPart2) - ast_copy_string(tmp, req->rlPart2, sizeof(tmp)); - c = ditch_braces(tmp); - - ast_copy_string(tmpf, get_header(req, "From"), sizeof(tmpf)); - fr = ditch_braces(tmpf); - - if (strncmp(c, "sip:", 4)) { - ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c); - return -1; - } - c += 4; - if (!ast_strlen_zero(fr)) { - if (strncmp(fr, "sip:", 4)) { - ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", fr); - return -1; - } - fr += 4; - } else - fr = NULL; - if ((a = strchr(c, '@'))) { - *a = '\0'; - a++; - ast_copy_string(p->domain, a, sizeof(p->domain)); - } - if ((a = strchr(c, ';'))) { - *a = '\0'; - } - if (fr) { - if ((a = strchr(fr, ';'))) - *a = '\0'; - if ((a = strchr(fr, '@'))) { - *a = '\0'; - ast_copy_string(p->fromdomain, a + 1, sizeof(p->fromdomain)); - } else - ast_copy_string(p->fromdomain, fr, sizeof(p->fromdomain)); - } - if (pedanticsipchecking) - url_decode(c); - if (sip_debug_test_pvt(p)) - ast_verbose("Looking for %s in %s\n", c, p->context); - if (ast_exists_extension(NULL, p->context, c, 1, fr) || - !strcmp(c, ast_pickup_ext())) { - if (!oreq) - ast_copy_string(p->exten, c, sizeof(p->exten)); - return 0; - } - - if (ast_canmatch_extension(NULL, p->context, c, 1, fr) || - !strncmp(c, ast_pickup_ext(),strlen(c))) { - return 1; - } - - return -1; -} - -/*--- get_sip_pvt_byid_locked: Lock interface lock and find matching pvt lock ---*/ -static struct sip_pvt *get_sip_pvt_byid_locked(char *callid) -{ - struct sip_pvt *sip_pvt_ptr = NULL; - - /* Search interfaces and find the match */ - ast_mutex_lock(&iflock); - sip_pvt_ptr = iflist; - while(sip_pvt_ptr) { - if (!strcmp(sip_pvt_ptr->callid, callid)) { - /* Go ahead and lock it (and its owner) before returning */ - ast_mutex_lock(&sip_pvt_ptr->lock); - if (sip_pvt_ptr->owner) { - while(ast_mutex_trylock(&sip_pvt_ptr->owner->lock)) { - ast_mutex_unlock(&sip_pvt_ptr->lock); - usleep(1); - ast_mutex_lock(&sip_pvt_ptr->lock); - if (!sip_pvt_ptr->owner) - break; - } - } - break; - } - sip_pvt_ptr = sip_pvt_ptr->next; - } - ast_mutex_unlock(&iflock); - return sip_pvt_ptr; -} - -/*--- get_refer_info: Call transfer support (the REFER method) ---*/ -static int get_refer_info(struct sip_pvt *sip_pvt, struct sip_request *outgoing_req) -{ - - char *p_refer_to = NULL, *p_referred_by = NULL, *h_refer_to = NULL, *h_referred_by = NULL, *h_contact = NULL; - char *replace_callid = "", *refer_to = NULL, *referred_by = NULL, *ptr = NULL; - struct sip_request *req = NULL; - struct sip_pvt *sip_pvt_ptr = NULL; - struct ast_channel *chan = NULL, *peer = NULL; - - req = outgoing_req; - - if (!req) { - req = &sip_pvt->initreq; - } - - if (!( (p_refer_to = get_header(req, "Refer-To")) && (h_refer_to = ast_strdupa(p_refer_to)) )) { - ast_log(LOG_WARNING, "No Refer-To Header That's illegal\n"); - return -1; - } - - refer_to = ditch_braces(h_refer_to); - - if (!( (p_referred_by = get_header(req, "Referred-By")) && (h_referred_by = ast_strdupa(p_referred_by)) )) { - ast_log(LOG_WARNING, "No Referrred-By Header That's not illegal\n"); - return -1; - } else { - referred_by = ditch_braces(h_referred_by); - } - h_contact = get_header(req, "Contact"); - - if (strncmp(refer_to, "sip:", 4)) { - ast_log(LOG_WARNING, "Refer-to: Huh? Not a SIP header (%s)?\n", refer_to); - return -1; - } - - if (strncmp(referred_by, "sip:", 4)) { - ast_log(LOG_WARNING, "Referred-by: Huh? Not a SIP header (%s) Ignoring?\n", referred_by); - referred_by = NULL; - } - - if (refer_to) - refer_to += 4; - - if (referred_by) - referred_by += 4; - - if ((ptr = strchr(refer_to, '?'))) { - /* Search for arguments */ - *ptr = '\0'; - ptr++; - if (!strncasecmp(ptr, "REPLACES=", 9)) { - char *p; - replace_callid = ast_strdupa(ptr + 9); - /* someday soon to support invite/replaces properly! - replaces_header = ast_strdupa(replace_callid); - -anthm - */ - url_decode(replace_callid); - if ((ptr = strchr(replace_callid, '%'))) - *ptr = '\0'; - if ((ptr = strchr(replace_callid, ';'))) - *ptr = '\0'; - /* Skip leading whitespace XXX memmove behaviour with overlaps ? */ - p = ast_skip_blanks(replace_callid); - if (p != replace_callid) - memmove(replace_callid, p, strlen(p)); - } - } - - if ((ptr = strchr(refer_to, '@'))) /* Skip domain (should be saved in SIPDOMAIN) */ - *ptr = '\0'; - if ((ptr = strchr(refer_to, ';'))) - *ptr = '\0'; - - if (referred_by) { - if ((ptr = strchr(referred_by, '@'))) - *ptr = '\0'; - if ((ptr = strchr(referred_by, ';'))) - *ptr = '\0'; - } - - if (sip_debug_test_pvt(sip_pvt)) { - ast_verbose("Transfer to %s in %s\n", refer_to, sip_pvt->context); - if (referred_by) - ast_verbose("Transfer from %s in %s\n", referred_by, sip_pvt->context); - } - if (!ast_strlen_zero(replace_callid)) { - /* This is a supervised transfer */ - ast_log(LOG_DEBUG,"Assigning Replace-Call-ID Info %s to REPLACE_CALL_ID\n",replace_callid); - - ast_copy_string(sip_pvt->refer_to, "", sizeof(sip_pvt->refer_to)); - ast_copy_string(sip_pvt->referred_by, "", sizeof(sip_pvt->referred_by)); - ast_copy_string(sip_pvt->refer_contact, "", sizeof(sip_pvt->refer_contact)); - sip_pvt->refer_call = NULL; - if ((sip_pvt_ptr = get_sip_pvt_byid_locked(replace_callid))) { - sip_pvt->refer_call = sip_pvt_ptr; - if (sip_pvt->refer_call == sip_pvt) { - ast_log(LOG_NOTICE, "Supervised transfer attempted to transfer into same call id (%s == %s)!\n", replace_callid, sip_pvt->callid); - sip_pvt->refer_call = NULL; - } else - return 0; - } else { - ast_log(LOG_NOTICE, "Supervised transfer requested, but unable to find callid '%s'. Both legs must reside on Asterisk box to transfer at this time.\n", replace_callid); - /* XXX The refer_to could contain a call on an entirely different machine, requiring an - INVITE with a replaces header -anthm XXX */ - /* The only way to find out is to use the dialplan - oej */ - } - } else if (ast_exists_extension(NULL, sip_pvt->context, refer_to, 1, NULL) || !strcmp(refer_to, ast_parking_ext())) { - /* This is an unsupervised transfer (blind transfer) */ - - ast_log(LOG_DEBUG,"Unsupervised transfer to (Refer-To): %s\n", refer_to); - if (referred_by) - ast_log(LOG_DEBUG,"Transferred by (Referred-by: ) %s \n", referred_by); - ast_log(LOG_DEBUG,"Transfer Contact Info %s (REFER_CONTACT)\n", h_contact); - ast_copy_string(sip_pvt->refer_to, refer_to, sizeof(sip_pvt->refer_to)); - if (referred_by) - ast_copy_string(sip_pvt->referred_by, referred_by, sizeof(sip_pvt->referred_by)); - if (h_contact) { - ast_copy_string(sip_pvt->refer_contact, h_contact, sizeof(sip_pvt->refer_contact)); - } - sip_pvt->refer_call = NULL; - if ((chan = sip_pvt->owner) && (peer = ast_bridged_channel(sip_pvt->owner))) { - pbx_builtin_setvar_helper(chan, "BLINDTRANSFER", peer->name); - pbx_builtin_setvar_helper(peer, "BLINDTRANSFER", chan->name); - } - return 0; - } else if (ast_canmatch_extension(NULL, sip_pvt->context, refer_to, 1, NULL)) { - return 1; - } - - return -1; -} - -/*--- get_also_info: Call transfer support (old way, depreciated)--*/ -static int get_also_info(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, "Also"), sizeof(tmp)); - - c = ditch_braces(tmp); - - - if (strncmp(c, "sip:", 4)) { - ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c); - return -1; - } - c += 4; - if ((a = strchr(c, '@'))) - *a = '\0'; - if ((a = strchr(c, ';'))) - *a = '\0'; - - if (sip_debug_test_pvt(p)) { - ast_verbose("Looking for %s in %s\n", c, p->context); - } - if (ast_exists_extension(NULL, p->context, c, 1, NULL)) { - /* This is an unsupervised transfer */ - ast_log(LOG_DEBUG,"Assigning Extension %s to REFER-TO\n", c); - ast_copy_string(p->refer_to, c, sizeof(p->refer_to)); - ast_copy_string(p->referred_by, "", sizeof(p->referred_by)); - ast_copy_string(p->refer_contact, "", sizeof(p->refer_contact)); - p->refer_call = NULL; - return 0; - } else if (ast_canmatch_extension(NULL, p->context, c, 1, NULL)) { - return 1; - } - - return -1; -} - -/*--- check_via: check Via: headers ---*/ -static int check_via(struct sip_pvt *p, struct sip_request *req) -{ - char via[256] = ""; - char iabuf[INET_ADDRSTRLEN]; - char *c, *pt; - struct hostent *hp; - struct ast_hostent ahp; - - memset(via, 0, sizeof(via)); - ast_copy_string(via, get_header(req, "Via"), sizeof(via)); - 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 -1; - } - 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 -1; - } - 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) : DEFAULT_SIP_PORT); - c = strstr(via, ";rport"); - if (c && (c[6] != '=')) /* rport query, not answer */ - ast_set_flag(p, SIP_NAT_ROUTE); - if (sip_debug_test_pvt(p)) { - c = (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) ? "NAT" : "non-NAT"; - ast_verbose("Sending to %s : %d (%s)\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port), c); - } - } - return 0; -} - -/*--- get_calleridname: Get caller id name from SIP headers ---*/ -static char *get_calleridname(char *input, char *output, size_t outputsize) -{ - char *end = strchr(input,'<'); - char *tmp = strchr(input,'\"'); - int bytes = 0; - int maxbytes = outputsize - 1; - - if (!end || (end == input)) return NULL; - /* move away from "<" */ - end--; - /* we found "name" */ - if (tmp && tmp < end) { - 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 { - /* we didn't find "name" */ - /* 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; -} - -/*--- get_rpid_num: 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(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; -} - - -/*--- check_user: Check if matching user or peer is defined ---*/ -static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, int reliable, struct sockaddr_in *sin, int ignore, char *mailbox, int mailboxlen) -{ - struct sip_user *user; - struct sip_peer *peer; - char *of, from[256] = "", *c; - char *rpid,rpid_num[50]; - char iabuf[INET_ADDRSTRLEN]; - int res = 0; - char *t; - char calleridname[50]; - int debug=sip_debug_test_addr(sin); - struct ast_variable *tmpvar = NULL, *v = NULL; - - /* Terminate URI */ - t = uri; - while(*t && (*t > 32) && (*t != ';')) - t++; - *t = '\0'; - of = get_header(req, "From"); - ast_copy_string(from, of, sizeof(from)); - memset(calleridname,0,sizeof(calleridname)); - get_calleridname(from, calleridname, sizeof(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 = ditch_braces(from); - if (ast_strlen_zero(p->exten)) { - t = uri; - if (!strncmp(t, "sip:", 4)) - t+= 4; - ast_copy_string(p->exten, t, sizeof(p->exten)); - t = strchr(p->exten, '@'); - if (t) - *t = '\0'; - if (ast_strlen_zero(p->our_contact)) - build_contact(p); - } - if (strncmp(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, '@'))) { - *c = '\0'; - if ((c = strchr(of, ':'))) - *c = '\0'; - ast_copy_string(p->cid_num, of, sizeof(p->cid_num)); - ast_shrink_phone_number(p->cid_num); - } - if (*calleridname) - ast_copy_string(p->cid_name, calleridname, sizeof(p->cid_name)); - if (ast_strlen_zero(of)) - return 0; - user = find_user(of, 1); - /* Find user based on user name in the from header */ - if (!mailbox && user && ast_apply_ha(user->ha, sin)) { - ast_copy_flags(p, user, SIP_TRUSTRPID | SIP_USECLIENTCODE | SIP_NAT | SIP_PROG_INBAND | SIP_OSPAUTH); - /* 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; - /* replace callerid if rpid found, and not restricted */ - if (!ast_strlen_zero(rpid_num) && ast_test_flag(p, SIP_TRUSTRPID)) { - if (*calleridname) - ast_copy_string(p->cid_name, calleridname, sizeof(p->cid_name)); - ast_copy_string(p->cid_num, rpid_num, sizeof(p->cid_num)); - ast_shrink_phone_number(p->cid_num); - } - - if (p->rtp) { - ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); - ast_rtp_setnat(p->rtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); - } - if (p->vrtp) { - ast_log(LOG_DEBUG, "Setting NAT on VRTP to %d\n", (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); - ast_rtp_setnat(p->vrtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); - } - if (!(res = check_auth(p, req, p->randdata, sizeof(p->randdata), user->name, user->secret, user->md5secret, sipmethod, uri, reliable, ignore))) { - sip_cancel_destroy(p); - ast_copy_flags(p, user, SIP_PROMISCREDIR | SIP_DTMF | SIP_REINVITE); - /* Copy SIP extensions profile from INVITE */ - if (p->sipoptions) - user->sipoptions = p->sipoptions; - - /* If we have a call limit, set flag */ - if (user->incominglimit) - ast_set_flag(p, SIP_CALL_LIMIT); - if (!ast_strlen_zero(user->context)) - ast_copy_string(p->context, user->context, sizeof(p->context)); - if (!ast_strlen_zero(user->cid_num) && !ast_strlen_zero(p->cid_num)) { - ast_copy_string(p->cid_num, user->cid_num, sizeof(p->cid_num)); - ast_shrink_phone_number(p->cid_num); - } - if (!ast_strlen_zero(user->cid_name) && !ast_strlen_zero(p->cid_num)) - ast_copy_string(p->cid_name, user->cid_name, sizeof(p->cid_name)); - ast_copy_string(p->username, user->name, sizeof(p->username)); - ast_copy_string(p->peersecret, user->secret, sizeof(p->peersecret)); - ast_copy_string(p->peermd5secret, user->md5secret, sizeof(p->peermd5secret)); - ast_copy_string(p->accountcode, user->accountcode, sizeof(p->accountcode)); - ast_copy_string(p->language, user->language, sizeof(p->language)); - ast_copy_string(p->musicclass, user->musicclass, sizeof(p->musicclass)); - p->amaflags = user->amaflags; - p->callgroup = user->callgroup; - p->pickupgroup = user->pickupgroup; - p->callingpres = user->callingpres; - p->capability = user->capability; - p->jointcapability = user->capability; - if (p->peercapability) - p->jointcapability &= p->peercapability; - if (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) - p->noncodeccapability |= AST_RTP_DTMF; - else - p->noncodeccapability &= ~AST_RTP_DTMF; - } - if (user && debug) - ast_verbose("Found user '%s'\n", user->name); - } else { - if (user) { - if (!mailbox && 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 */ - /* Look for peer based on the IP address we received data from */ - /* If peer is registred 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); - if (peer) { - if (debug) - ast_verbose("Found peer '%s'\n", peer->name); - /* Take the peer */ - ast_copy_flags(p, peer, SIP_TRUSTRPID | SIP_USECLIENTCODE | SIP_NAT | SIP_PROG_INBAND | SIP_OSPAUTH); - - /* 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, SIP_TRUSTRPID)) { - if (*calleridname) - ast_copy_string(p->cid_name, calleridname, sizeof(p->cid_name)); - ast_copy_string(p->cid_num, rpid_num, sizeof(p->cid_num)); - ast_shrink_phone_number(p->cid_num); - } - if (p->rtp) { - ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); - ast_rtp_setnat(p->rtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); - } - if (p->vrtp) { - ast_log(LOG_DEBUG, "Setting NAT on VRTP to %d\n", (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); - ast_rtp_setnat(p->vrtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); - } - ast_copy_string(p->peersecret, peer->secret, sizeof(p->peersecret)); - p->peersecret[sizeof(p->peersecret)-1] = '\0'; - ast_copy_string(p->peermd5secret, peer->md5secret, sizeof(p->peermd5secret)); - p->peermd5secret[sizeof(p->peermd5secret)-1] = '\0'; - p->callingpres = peer->callingpres; - if (ast_test_flag(peer, SIP_INSECURE_INVITE)) { - /* Pretend there is no required authentication */ - p->peersecret[0] = '\0'; - p->peermd5secret[0] = '\0'; - } - if (!(res = check_auth(p, req, p->randdata, sizeof(p->randdata), peer->name, p->peersecret, p->peermd5secret, sipmethod, uri, reliable, ignore))) { - ast_copy_flags(p, peer, SIP_PROMISCREDIR | SIP_DTMF | SIP_REINVITE); - /* If we have a call limit, set flag */ - if (peer->incominglimit) - ast_set_flag(p, SIP_CALL_LIMIT); - ast_copy_string(p->peername, peer->name, sizeof(p->peername)); - ast_copy_string(p->authname, peer->name, sizeof(p->authname)); - /* 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 (mailbox) - snprintf(mailbox, mailboxlen, ",%s,", peer->mailbox); - if (!ast_strlen_zero(peer->username)) { - ast_copy_string(p->username, peer->username, sizeof(p->username)); - /* Use the default username for authentication on outbound calls */ - ast_copy_string(p->authname, peer->username, sizeof(p->authname)); - } - if (!ast_strlen_zero(peer->cid_num) && !ast_strlen_zero(p->cid_num)) { - ast_copy_string(p->cid_num, peer->cid_num, sizeof(p->cid_num)); - ast_shrink_phone_number(p->cid_num); - } - if (!ast_strlen_zero(peer->cid_name) && !ast_strlen_zero(p->cid_name)) - ast_copy_string(p->cid_name, peer->cid_name, sizeof(p->cid_name)); - ast_copy_string(p->fullcontact, peer->fullcontact, sizeof(p->fullcontact)); - if (!ast_strlen_zero(peer->context)) - ast_copy_string(p->context, peer->context, sizeof(p->context)); - ast_copy_string(p->peersecret, peer->secret, sizeof(p->peersecret)); - ast_copy_string(p->peermd5secret, peer->md5secret, sizeof(p->peermd5secret)); - ast_copy_string(p->language, peer->language, sizeof(p->language)); - ast_copy_string(p->accountcode, peer->accountcode, sizeof(p->accountcode)); - p->amaflags = peer->amaflags; - p->callgroup = peer->callgroup; - p->pickupgroup = peer->pickupgroup; - p->capability = peer->capability; - p->jointcapability = peer->capability; - if (p->peercapability) - p->jointcapability &= p->peercapability; - if (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) - p->noncodeccapability |= AST_RTP_DTMF; - else - p->noncodeccapability &= ~AST_RTP_DTMF; - } - ASTOBJ_UNREF(peer,sip_destroy_peer); - } else { - if (debug) - ast_verbose("Found no matching peer or user for '%s:%d'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port)); - - /* do we allow guests? */ - if (!global_allowguest) - res = -1; /* we don't want any guests, authentication will fail */ -#ifdef OSP_SUPPORT - else if (global_allowguest == 2) { - ast_copy_flags(p, &global_flags, SIP_OSPAUTH); - res = check_auth(p, req, p->randdata, sizeof(p->randdata), "", "", "", sipmethod, uri, reliable, ignore); - } -#endif - } - - } - - if (user) - ASTOBJ_UNREF(user,sip_destroy_user); - return res; -} - -/*--- check_user: Find user ---*/ -static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, int reliable, struct sockaddr_in *sin, int ignore) -{ - return check_user_full(p, req, sipmethod, uri, reliable, sin, ignore, NULL, 0); -} - -/*--- get_msg_text: 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;xlines;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; -} - - -/*--- receive_message: Receive SIP MESSAGE method messages ---*/ -/* we handle messages within current calls currently */ -static void receive_message(struct sip_pvt *p, struct sip_request *req) -{ - char buf[1024]; - struct ast_frame f; - - if (get_msg_text(buf, sizeof(buf), req)) { - ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid); - 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); - } -} - -/*--- sip_show_inuse: CLI Command to show calls within limits set by - incominglimit ---*/ -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 = 0; - - if (argc < 3) - return RESULT_SHOWUSAGE; - - if (argc == 4 && !strcmp(argv[3],"all")) - showall = 1; - - ast_cli(fd, FORMAT, "* User name", "In use", "Limit"); - ASTOBJ_CONTAINER_TRAVERSE(&userl, 1, do { - ASTOBJ_RDLOCK(iterator); - if (iterator->incominglimit) - snprintf(ilimits, sizeof(ilimits), "%d", iterator->incominglimit); - else - ast_copy_string(ilimits, "N/A", sizeof(ilimits)); - /* Code disabled ---------------------------- - if (iterator->outgoinglimit) - snprintf(olimits, sizeof(olimits), "%d", iterator->outgoinglimit); - else - ast_copy_string(olimits, "N/A", sizeof(olimits)); - snprintf(oused, sizeof(oused), "%d", iterator->outUse); - ---------------------------------------------*/ - snprintf(iused, sizeof(iused), "%d", iterator->inUse); - if (showall || iterator->incominglimit) - 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->incominglimit) - snprintf(ilimits, sizeof(ilimits), "%d", iterator->incominglimit); - else - ast_copy_string(ilimits, "N/A", sizeof(ilimits)); - /* Code disabled ---------------------------- - if (iterator->outgoinglimit) - snprintf(olimits, sizeof(olimits), "%d", iterator->outgoinglimit); - else - ast_copy_string(olimits, "N/A", sizeof(olimits)); - snprintf(oused, sizeof(oused), "%d", iterator->outUse); - ---------------------------------------------*/ - snprintf(iused, sizeof(iused), "%d", iterator->inUse); - if (showall || iterator->incominglimit) - ast_cli(fd, FORMAT2, iterator->name, iused, ilimits); - ASTOBJ_UNLOCK(iterator); - } while (0) ); - - return RESULT_SUCCESS; -#undef FORMAT -#undef FORMAT2 -} - -/*--- nat2str: 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"; - } -} - -/*--- sip_show_users: CLI Command 'SIP Show Users' ---*/ -static int sip_show_users(int fd, int argc, char *argv[]) -{ - regex_t regexbuf; - int havepattern = 0; - -#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 = 1; - } 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, 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: Action ID for this transaction. Will be returned.\n"; - -static int _sip_show_peers(int fd, int *total, struct mansession *s, struct message *m, int argc, char *argv[]); - -/*--- manager_sip_show_peers: Show SIP peers in the manager API ---*/ -/* Inspired from chan_iax2 */ -static int manager_sip_show_peers( struct mansession *s, struct message *m ) -{ - char *id = astman_get_header(m,"ActionID"); - char *a[] = { "sip", "show", "peers" }; - char idtext[256] = ""; - int total = 0; - - if (id && !ast_strlen_zero(id)) - snprintf(idtext,256,"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(s->fd, &total, s, m, 3, a); - /* Send final confirmation */ - ast_mutex_lock(&s->lock); - ast_cli(s->fd, - "Event: PeerlistComplete\r\n" - "ListItems: %d\r\n" - "%s" - "\r\n", total, idtext); - ast_mutex_unlock(&s->lock); - return 0; -} - -/*--- sip_show_peers: CLI Show Peers command */ -static int sip_show_peers(int fd, int argc, char *argv[]) -{ - return _sip_show_peers(fd, NULL, NULL, NULL, argc, argv); -} - -/*--- _sip_show_peers: Execute sip show peers command */ -static int _sip_show_peers(int fd, int *total, struct mansession *s, struct message *m, int argc, char *argv[]) -{ - regex_t regexbuf; - int havepattern = 0; - -#define FORMAT2 "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-15.15s %-8s %-10s\n" -#define FORMAT "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-15.15s %-8d %-10s\n" - - char name[256] = ""; - char iabuf[INET_ADDRSTRLEN]; - int total_peers = 0; - int peers_online = 0; - int peers_offline = 0; - char *id; - char idtext[256] = ""; - - if (s) { /* Manager - get ActionID */ - id = astman_get_header(m,"ActionID"); - if (id && !ast_strlen_zero(id)) - snprintf(idtext,256,"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 = 1; - } 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", "Mask", "Port", "Status"); - } - - ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do { - char nm[20] = ""; - char status[20] = ""; - char srch[2000]; - - ASTOBJ_RDLOCK(iterator); - - if (havepattern && regexec(®exbuf, iterator->name, 0, NULL, 0)) { - ASTOBJ_UNLOCK(iterator); - continue; - } - - ast_inet_ntoa(nm, sizeof(nm), iterator->mask); - 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)); - if (iterator->maxms) { - if (iterator->lastms < 0) { - ast_copy_string(status, "UNREACHABLE", sizeof(status)); - peers_offline++; - } else if (iterator->lastms > iterator->maxms) { - snprintf(status, sizeof(status), "LAGGED (%d ms)", iterator->lastms); - peers_online++; - } else if (iterator->lastms) { - snprintf(status, sizeof(status), "OK (%d ms)", iterator->lastms); - peers_online++; - } else { - /* Checking if port is 0 */ - if ( ntohs(iterator->addr.sin_port) == 0 ) { - peers_offline++; - } else { - peers_online++; - } - ast_copy_string(status, "UNKNOWN", sizeof(status)); - } - } else { - ast_copy_string(status, "Unmonitored", sizeof(status)); - /* Checking if port is 0 */ - if ( ntohs(iterator->addr.sin_port) == 0 ) { - peers_offline++; - } else { - peers_online++; - } - } - - snprintf(srch, sizeof(srch), FORMAT, name, - iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), iterator->addr.sin_addr) : "(Unspecified)", - ast_test_flag(iterator, SIP_DYNAMIC) ? " D " : " ", /* Dynamic or not? */ - (ast_test_flag(iterator, SIP_NAT) & SIP_NAT_ROUTE) ? " N " : " ", /* NAT=yes? */ - iterator->ha ? " A " : " ", /* permit/deny */ - nm, ntohs(iterator->addr.sin_port), status); - - if (!s) {/* Normal CLI list */ - ast_cli(fd, FORMAT, name, - iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), iterator->addr.sin_addr) : "(Unspecified)", - ast_test_flag(iterator, SIP_DYNAMIC) ? " D " : " ", /* Dynamic or not? */ - (ast_test_flag(iterator, SIP_NAT) & SIP_NAT_ROUTE) ? " N " : " ", /* NAT=yes? */ - iterator->ha ? " A " : " ", /* permit/deny */ - nm, - ntohs(iterator->addr.sin_port), status); - } else { /* Manager format */ - /* The names here need to be the same as other channels */ - ast_mutex_lock(&s->lock); - ast_cli(fd, - "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" - "ACL: %s\r\n" - "Status: %s\r\n\r\n", - idtext, - iterator->name, - iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), iterator->addr.sin_addr) : "-none-", - ntohs(iterator->addr.sin_port), - ast_test_flag(iterator, SIP_DYNAMIC) ? "yes" : "no", /* Dynamic or not? */ - (ast_test_flag(iterator, SIP_NAT) & SIP_NAT_ROUTE) ? "yes" : "no", /* NAT=yes? */ - iterator->ha ? "yes" : "no", /* permit/deny */ - status); - - ast_mutex_unlock(&s->lock); - } - - ASTOBJ_UNLOCK(iterator); - - total_peers++; - } while(0) ); - - if (!s) { - ast_cli(fd,"%d sip peers [%d online , %d offline]\n",total_peers,peers_online,peers_offline); - } - - if (havepattern) - regfree(®exbuf); - - if (total) - *total = total_peers; - - - return RESULT_SUCCESS; -#undef FORMAT -#undef FORMAT2 -} - -/*--- sip_show_objects: List all allocated SIP Objects ---*/ -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; -} -/*--- print_group: Print call group and pickup group ---*/ -static void print_group(int fd, unsigned int group) -{ - char buf[256]; - ast_cli(fd, "%s\n", ast_print_group(buf, sizeof(buf), group) ); -} - -/*--- dtmfmode2str: 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"; - } - return ""; -} - -/*--- insecure2str: 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"; -} - -/*--- sip_prune_realtime: 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 = 0; - int prunepeer = 0; - int multi = 0; - 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 = 1; - pruneuser = prunepeer = 1; - } else { - pruneuser = prunepeer = 1; - 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 = 1; - name = argv[4]; - pruneuser = prunepeer = 1; - } else if (!strcasecmp(argv[3], "user")) { - pruneuser = 1; - if (!strcasecmp(argv[4], "all")) - multi = 1; - else - name = argv[4]; - } else if (!strcasecmp(argv[3], "peer")) { - prunepeer = 1; - if (!strcasecmp(argv[4], "all")) - multi = 1; - 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 = 1; - name = argv[5]; - } else if (!strcasecmp(argv[3], "peer")) { - prunepeer = 1; - 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_page2), 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_page2), 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_page2), 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_page2), 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; -} - -static char mandescr_show_peer[] = -"Description: Show one SIP peer with details on current status.\n" -" The XML format is under development, feedback welcome! /oej\n" -"Variables: \n" -" Peer: The peer name you want to check.\n" -" ActionID: Optional action ID for this AMI transaction.\n"; - -static int _sip_show_peer(int type, int fd, struct mansession *s, struct message *m, int argc, char *argv[]); - -/*--- manager_sip_show_peer: Show SIP peers in the manager API ---*/ -static int manager_sip_show_peer( struct mansession *s, struct message *m ) -{ - char *id = astman_get_header(m,"ActionID"); - char *a[4]; - char *peer; - int ret; - - peer = astman_get_header(m,"Peer"); - if (!peer || ast_strlen_zero(peer)) { - astman_send_error(s, m, "Peer: missing.\n"); - return 0; - } - ast_mutex_lock(&s->lock); - a[0] = "sip"; - a[1] = "show"; - a[2] = "peer"; - a[3] = peer; - - if (id && !ast_strlen_zero(id)) - ast_cli(s->fd, "ActionID: %s\r\n",id); - ret = _sip_show_peer(1, s->fd, s, m, 4, a ); - ast_cli( s->fd, "\r\n\r\n" ); - ast_mutex_unlock(&s->lock); - return ret; -} - - - -/*--- sip_show_peer: 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, argv); -} - -static int _sip_show_peer(int type, int fd, struct mansession *s, struct message *m, int argc, char *argv[]) -{ - char status[30] = ""; - char cbuf[256]; - char iabuf[INET_ADDRSTRLEN]; - 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 = 0; - - if (argc < 4) - return RESULT_SHOWUSAGE; - - load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? 1 : 0; - peer = find_peer(argv[3], NULL, load_realtime); - if (s) { /* Manager */ - if (peer) - ast_cli(s->fd, "Response: Success\r\n"); - else { - snprintf (cbuf, sizeof(cbuf), "Peer %s not found.\n", 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); - ast_cli(fd, " Secret : %s\n", ast_strlen_zero(peer->secret)?"":""); - ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(peer->md5secret)?"":""); - auth = peer->auth; - while(auth) { - 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)?"":(!ast_strlen_zero(auth->md5secret)?"" : "")); - auth = auth->next; - } - ast_cli(fd, " Context : %s\n", peer->context); - 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, " 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); - ast_cli(fd, " Pickupgroup : "); - print_group(fd, peer->pickupgroup); - ast_cli(fd, " Mailbox : %s\n", peer->mailbox); - ast_cli(fd, " LastMsgsSent : %d\n", peer->lastmsgssent); - ast_cli(fd, " Inc. limit : %d\n", peer->incominglimit); - ast_cli(fd, " Outg. limit : %d\n", peer->outgoinglimit); - ast_cli(fd, " Dynamic : %s\n", (ast_test_flag(peer, SIP_DYNAMIC)?"Yes":"No")); - ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "")); - ast_cli(fd, " Expire : %d\n", peer->expire); - ast_cli(fd, " Expiry : %d\n", peer->expiry); - ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(peer, SIP_INSECURE_PORT), ast_test_flag(peer, SIP_INSECURE_INVITE))); - ast_cli(fd, " Nat : %s\n", nat2str(ast_test_flag(peer, SIP_NAT))); - ast_cli(fd, " ACL : %s\n", (peer->ha?"Yes":"No")); - ast_cli(fd, " CanReinvite : %s\n", (ast_test_flag(peer, SIP_CAN_REINVITE)?"Yes":"No")); - ast_cli(fd, " PromiscRedir : %s\n", (ast_test_flag(peer, SIP_PROMISCREDIR)?"Yes":"No")); - ast_cli(fd, " User=Phone : %s\n", (ast_test_flag(peer, SIP_USEREQPHONE)?"Yes":"No")); - - /* - is enumerated */ - ast_cli(fd, " DTMFmode : %s\n", dtmfmode2str(ast_test_flag(peer, 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(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port)); - ast_cli(fd, " Defaddr->IP : %s Port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port)); - ast_cli(fd, " Def. Username: %s\n", peer->username); - ast_cli(fd, " SIP Options : "); - if (peer->sipoptions) { - for (x=0 ; (x < (sizeof(sip_options) / sizeof(sip_options[0]))); x++) { - if (peer->sipoptions & sip_options[x].id) - ast_cli(fd, "%s ", sip_options[x].text); - } - } 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 : ("); - pref = &peer->prefs; - for(x = 0; x < 32 ; x++) { - codec = ast_codec_pref_index(pref,x); - if (!codec) - break; - ast_cli(fd, "%s", ast_getformatname(codec)); - if (x < 31 && ast_codec_pref_index(pref,x+1)) - ast_cli(fd, "|"); - } - - if (!x) - ast_cli(fd, "none"); - ast_cli(fd, ")\n"); - - ast_cli(fd, " Status : "); - if (peer->lastms < 0) - ast_copy_string(status, "UNREACHABLE", sizeof(status)); - else if (peer->lastms > peer->maxms) - snprintf(status, sizeof(status), "LAGGED (%d ms)", peer->lastms); - else if (peer->lastms) - snprintf(status, sizeof(status), "OK (%d ms)", peer->lastms); - else - ast_copy_string(status, "UNKNOWN", 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 *actionid = astman_get_header(m,"ActionID"); - - ast_cli(fd, "Channeltype: SIP\r\n"); - if (actionid) - ast_cli(fd, "ActionID: %s\r\n", actionid); - ast_cli(fd, "ObjectName: %s\r\n", peer->name); - ast_cli(fd, "ChanObjectType: peer\r\n"); - ast_cli(fd, "SecretExist: %s\r\n", ast_strlen_zero(peer->secret)?"N":"Y"); - ast_cli(fd, "MD5SecretExist: %s\r\n", ast_strlen_zero(peer->md5secret)?"N":"Y"); - ast_cli(fd, "Context: %s\r\n", peer->context); - ast_cli(fd, "Language: %s\r\n", peer->language); - if (!ast_strlen_zero(peer->accountcode)) - ast_cli(fd, "Accountcode: %s\r\n", peer->accountcode); - ast_cli(fd, "AMAflags: %s\r\n", ast_cdr_flags2str(peer->amaflags)); - ast_cli(fd, "CID-CallingPres: %s\r\n", ast_describe_caller_presentation(peer->callingpres)); - if (!ast_strlen_zero(peer->fromuser)) - ast_cli(fd, "SIP-FromUser: %s\r\n", peer->fromuser); - if (!ast_strlen_zero(peer->fromdomain)) - ast_cli(fd, "SIP-FromDomain: %s\r\n", peer->fromdomain); - ast_cli(fd, "Callgroup: "); - print_group(fd, peer->callgroup); - ast_cli(fd, "Pickupgroup: "); - print_group(fd, peer->pickupgroup); - ast_cli(fd, "VoiceMailbox: %s\r\n", peer->mailbox); - ast_cli(fd, "LastMsgsSent: %d\r\n", peer->lastmsgssent); - ast_cli(fd, "Incominglimit: %d\r\n", peer->incominglimit); - ast_cli(fd, "Outgoinglimit: %d\r\n", peer->outgoinglimit); - ast_cli(fd, "Dynamic: %s\r\n", (ast_test_flag(peer, SIP_DYNAMIC)?"Y":"N")); - ast_cli(fd, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "")); - ast_cli(fd, "RegExpire: %ld seconds\r\n", ast_sched_when(sched,peer->expire)); - ast_cli(fd, "RegExpiry: %d\r\n", peer->expiry); - ast_cli(fd, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(peer, SIP_INSECURE_PORT), ast_test_flag(peer, SIP_INSECURE_INVITE))); - ast_cli(fd, "SIP-NatSupport: %s\r\n", nat2str(ast_test_flag(peer, SIP_NAT))); - ast_cli(fd, "ACL: %s\r\n", (peer->ha?"Y":"N")); - ast_cli(fd, "SIP-CanReinvite: %s\r\n", (ast_test_flag(peer, SIP_CAN_REINVITE)?"Y":"N")); - ast_cli(fd, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(peer, SIP_PROMISCREDIR)?"Y":"N")); - ast_cli(fd, "SIP-UserPhone: %s\r\n", (ast_test_flag(peer, SIP_USEREQPHONE)?"Y":"N")); - - /* - is enumerated */ - ast_cli(fd, "SIP-DTMFmode %s\r\n", dtmfmode2str(ast_test_flag(peer, SIP_DTMF))); - ast_cli(fd, "SIPLastMsg: %d\r\n", peer->lastmsg); - ast_cli(fd, "ToHost: %s\r\n", peer->tohost); - ast_cli(fd, "Address-IP: %s\r\nAddress-Port: %d\r\n", peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "", ntohs(peer->addr.sin_port)); - ast_cli(fd, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port)); - ast_cli(fd, "Default-Username: %s\r\n", peer->username); - ast_cli(fd, "Codecs: "); - ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability); - ast_cli(fd, "%s\r\n", codec_buf); - ast_cli(fd, "CodecOrder: "); - pref = &peer->prefs; - for(x = 0; x < 32 ; x++) { - codec = ast_codec_pref_index(pref,x); - if (!codec) - break; - ast_cli(fd, "%s", ast_getformatname(codec)); - if (x < 31 && ast_codec_pref_index(pref,x+1)) - ast_cli(fd, ","); - } - - ast_cli(fd, "\r\n"); - ast_cli(fd, "Status: "); - if (peer->lastms < 0) - ast_copy_string(status, "UNREACHABLE", sizeof(status)); - else if (peer->lastms > peer->maxms) - snprintf(status, sizeof(status), "LAGGED (%d ms)", peer->lastms); - else if (peer->lastms) - snprintf(status, sizeof(status), "OK (%d ms)", peer->lastms); - else - ast_copy_string(status, "UNKNOWN", sizeof(status)); - ast_cli(fd, "%s\r\n",status); - ast_cli(fd, "SIP-Useragent: %s\r\n", peer->useragent); - ast_cli(fd, "Reg-Contact : %s\r\n", peer->fullcontact); - if (peer->chanvars) { - for (v = peer->chanvars ; v ; v = v->next) { - ast_cli(fd, "ChanVariable:\n"); - ast_cli(fd, " %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; -} - -/*--- sip_show_user: 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_codec_pref *pref; - struct ast_variable *v; - int x = 0, codec = 0, load_realtime = 0; - - if (argc < 4) - return RESULT_SHOWUSAGE; - - /* Load from realtime storage? */ - load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? 1 : 0; - - 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)?"":""); - ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(user->md5secret)?"":""); - 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, " CallingPres : %s\n", ast_describe_caller_presentation(user->callingpres)); - ast_cli(fd, " Inc. limit : %d\n", user->incominglimit); - ast_cli(fd, " Outg. limit : %d\n", user->outgoinglimit); - ast_cli(fd, " Callgroup : "); - print_group(fd, user->callgroup); - ast_cli(fd, " Pickupgroup : "); - print_group(fd, user->pickupgroup); - ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), user->cid_name, user->cid_num, "")); - ast_cli(fd, " ACL : %s\n", (user->ha?"Yes":"No")); - ast_cli(fd, " Codec Order : ("); - pref = &user->prefs; - for(x = 0; x < 32 ; x++) { - codec = ast_codec_pref_index(pref,x); - if (!codec) - break; - ast_cli(fd, "%s", ast_getformatname(codec)); - if (x < 31 && ast_codec_pref_index(pref,x+1)) - ast_cli(fd, "|"); - } - - if (!x) - ast_cli(fd, "none"); - ast_cli(fd, ")\n"); - - 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; -} - -/*--- sip_show_registry: 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\n" -#define FORMAT "%-30.30s %-12.12s %8d %-20.20s\n" - char host[80]; - - if (argc != 3) - return RESULT_SHOWUSAGE; - ast_cli(fd, FORMAT2, "Host", "Username", "Refresh", "State"); - ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do { - ASTOBJ_RDLOCK(iterator); - snprintf(host, sizeof(host), "%s:%d", iterator->hostname, iterator->portno ? iterator->portno : DEFAULT_SIP_PORT); - ast_cli(fd, FORMAT, host, iterator->username, iterator->refresh, regstate2str(iterator->regstate)); - ASTOBJ_UNLOCK(iterator); - } while(0)); - return RESULT_SUCCESS; -#undef FORMAT -#undef FORMAT2 -} - -/* Forward declaration */ -static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions); - -/*--- sip_show_channels: Show active SIP channels ---*/ -static int sip_show_channels(int fd, int argc, char *argv[]) -{ - return __sip_show_channels(fd, argc, argv, 0); -} - -/*--- sip_show_subscriptions: Show active SIP subscriptions ---*/ -static int sip_show_subscriptions(int fd, int argc, char *argv[]) -{ - return __sip_show_channels(fd, argc, argv, 1); -} - -static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions) -{ -#define FORMAT3 "%-15.15s %-10.10s %-21.21s %-15.15s\n" -#define FORMAT2 "%-15.15s %-10.10s %-11.11s %-11.11s %s %s\n" -#define FORMAT "%-15.15s %-10.10s %-11.11s %5.5d/%5.5d %-6.6s%s %s\n" - struct sip_pvt *cur; - char iabuf[INET_ADDRSTRLEN]; - int numchans = 0; - 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", "Last Msg"); - else - ast_cli(fd, FORMAT3, "Peer", "User", "Call ID", "URI"); - while (cur) { - if (!cur->subscribed && !subscriptions) { - ast_cli(fd, FORMAT, ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr), - ast_strlen_zero(cur->username) ? ( ast_strlen_zero(cur->cid_num) ? "(None)" : cur->cid_num ) : cur->username, - cur->callid, - cur->ocseq, cur->icseq, - ast_getformatname(cur->owner ? cur->owner->nativeformats : 0), - ast_test_flag(cur, SIP_NEEDDESTROY) ? "(d)" : "", - cur->lastmsg ); - numchans++; - } - if (cur->subscribed && subscriptions) { - ast_cli(fd, FORMAT3, ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr), - ast_strlen_zero(cur->username) ? ( ast_strlen_zero(cur->cid_num) ? "(None)" : cur->cid_num ) : cur->username, - cur->callid, cur->uri); - - } - cur = cur->next; - } - ast_mutex_unlock(&iflock); - if (!subscriptions) - ast_cli(fd, "%d active SIP channel(s)\n", numchans); - else - ast_cli(fd, "%d active SIP subscriptions(s)\n", numchans); - return RESULT_SUCCESS; -#undef FORMAT -#undef FORMAT2 -#undef FORMAT3 -} - -/*--- complete_sipch: Support routine for 'sip show channel' CLI ---*/ -static char *complete_sipch(char *line, char *word, int pos, int state) -{ - int which=0; - struct sip_pvt *cur; - char *c = NULL; - - ast_mutex_lock(&iflock); - cur = iflist; - while(cur) { - if (!strncasecmp(word, cur->callid, strlen(word))) { - if (++which > state) { - c = strdup(cur->callid); - break; - } - } - cur = cur->next; - } - ast_mutex_unlock(&iflock); - return c; -} - -/*--- complete_sip_peer: Do completion on peer name ---*/ -static char *complete_sip_peer(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)) { - if (flags2 && !ast_test_flag((&iterator->flags_page2), flags2)) - continue; - if (++which > state) { - result = strdup(iterator->name); - } - } - } while(0) ); - return result; -} - -/*--- complete_sip_show_peer: Support routine for 'sip show peer' CLI ---*/ -static char *complete_sip_show_peer(char *line, char *word, int pos, int state) -{ - if (pos == 3) - return complete_sip_peer(word, state, 0); - - return NULL; -} - -/*--- complete_sip_debug_peer: Support routine for 'sip debug peer' CLI ---*/ -static char *complete_sip_debug_peer(char *line, char *word, int pos, int state) -{ - if (pos == 3) - return complete_sip_peer(word, state, 0); - - return NULL; -} - -/*--- complete_sip_user: Do completion on user name ---*/ -static char *complete_sip_user(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_page2), flags2)) - continue; - if (++which > state) { - result = strdup(iterator->name); - } - } - } while(0) ); - return result; -} - -/*--- complete_sip_show_user: Support routine for 'sip show user' CLI ---*/ -static char *complete_sip_show_user(char *line, char *word, int pos, int state) -{ - if (pos == 3) - return complete_sip_user(word, state, 0); - - return NULL; -} - -/*--- complete_sipnotify: Support routine for 'sip notify' CLI ---*/ -static char *complete_sipnotify(char *line, char *word, int pos, int state) -{ - char *c = NULL; - - if (pos == 2) { - int which = 0; - char *cat; - - /* do completion for notify type */ - - if (!notify_types) - return NULL; - - cat = ast_category_browse(notify_types, NULL); - while(cat) { - if (!strncasecmp(word, cat, strlen(word))) { - if (++which > state) { - c = strdup(cat); - break; - } - } - cat = ast_category_browse(notify_types, cat); - } - return c; - } - - if (pos > 2) - return complete_sip_peer(word, state, 0); - - return NULL; -} - -/*--- complete_sip_prune_realtime_peer: Support routine for 'sip prune realtime peer' CLI ---*/ -static char *complete_sip_prune_realtime_peer(char *line, char *word, int pos, int state) -{ - if (pos == 4) - return complete_sip_peer(word, state, SIP_PAGE2_RTCACHEFRIENDS); - return NULL; -} - -/*--- complete_sip_prune_realtime_user: Support routine for 'sip prune realtime user' CLI ---*/ -static char *complete_sip_prune_realtime_user(char *line, char *word, int pos, int state) -{ - if (pos == 4) - return complete_sip_user(word, state, SIP_PAGE2_RTCACHEFRIENDS); - - return NULL; -} - -/*--- sip_show_channel: Show details of one call ---*/ -static int sip_show_channel(int fd, int argc, char *argv[]) -{ - struct sip_pvt *cur; - char iabuf[INET_ADDRSTRLEN]; - size_t len; - int found = 0; - - if (argc != 4) - return RESULT_SHOWUSAGE; - len = strlen(argv[3]); - ast_mutex_lock(&iflock); - cur = iflist; - while(cur) { - if (!strncasecmp(cur->callid, argv[3],len)) { - ast_cli(fd,"\n"); - if (cur->subscribed) - ast_cli(fd, " * Subscription\n"); - else - ast_cli(fd, " * SIP Call\n"); - ast_cli(fd, " Direction: %s\n", ast_test_flag(cur, SIP_OUTGOING)?"Outgoing":"Incoming"); - ast_cli(fd, " Call-ID: %s\n", cur->callid); - ast_cli(fd, " Our Codec Capability: %d\n", cur->capability); - ast_cli(fd, " Non-Codec Capability: %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(cur->owner ? cur->owner->nativeformats : 0) ); - ast_cli(fd, " Theoretical Address: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr), ntohs(cur->sa.sin_port)); - ast_cli(fd, " Received Address: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->recv.sin_addr), ntohs(cur->recv.sin_port)); - ast_cli(fd, " NAT Support: %s\n", nat2str(ast_test_flag(cur, SIP_NAT))); - ast_cli(fd, " Our Tag: %08d\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, SIP_NEEDDESTROY)); - ast_cli(fd, " Last Message: %s\n", cur->lastmsg); - ast_cli(fd, " Promiscuous Redir: %s\n", ast_test_flag(cur, 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, 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++; - } - cur = cur->next; - } - ast_mutex_unlock(&iflock); - if (!found) - ast_cli(fd, "No such SIP Call ID starting with '%s'\n", argv[3]); - return RESULT_SUCCESS; -} - -/*--- sip_show_channel: Show details of one call ---*/ -static int sip_show_history(int fd, int argc, char *argv[]) -{ - struct sip_pvt *cur; - struct sip_history *hist; - size_t len; - int x; - 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); - cur = iflist; - while(cur) { - if (!strncasecmp(cur->callid, argv[3],len)) { - ast_cli(fd,"\n"); - if (cur->subscribed) - ast_cli(fd, " * Subscription\n"); - else - ast_cli(fd, " * SIP Call\n"); - x = 0; - hist = cur->history; - while(hist) { - x++; - ast_cli(fd, "%d. %s\n", x, hist->event); - hist = hist->next; - } - if (!x) - ast_cli(fd, "Call '%s' has no history\n", cur->callid); - found++; - } - cur = cur->next; - } - ast_mutex_unlock(&iflock); - if (!found) - ast_cli(fd, "No such SIP Call ID starting with '%s'\n", argv[3]); - return RESULT_SUCCESS; -} - - -/*--- receive_info: Receive SIP INFO Message ---*/ -/* Doesn't read the duration of the DTMF signal */ -static void receive_info(struct sip_pvt *p, struct sip_request *req) -{ - char buf[1024] = ""; - unsigned int event; - char resp = 0; - struct ast_frame f; - char *c; - - /* Need to check the media/type */ - if (!strcasecmp(get_header(req, "Content-Type"), "application/dtmf-relay") || - !strcasecmp(get_header(req, "Content-Type"), "application/vnd.nortelnetworks.digits")) { - - /* Try getting the "signal=" part */ - if (ast_strlen_zero(c = get_sdp(req, "Signal")) && ast_strlen_zero(c = get_sdp(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 (p->owner) { /* PBX call */ - if (!ast_strlen_zero(buf)) { - if (sipdebug) - ast_verbose("* DTMF received: '%c'\n", buf[0]); - 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 < 10) { - resp = '0' + event; - } else if (event < 11) { - resp = '*'; - } else if (event < 12) { - resp = '#'; - } else if (event < 16) { - resp = 'A' + (event - 12); - } - /* Build DTMF frame and deliver to PBX for transmission to other call leg*/ - memset(&f, 0, sizeof(f)); - f.frametype = AST_FRAME_DTMF; - f.subclass = resp; - f.offset = 0; - f.data = NULL; - f.datalen = 0; - ast_queue_frame(p->owner, &f); - } - transmit_response(p, "200 OK", req); - return; - } else { - transmit_response(p, "481 Call leg/transaction does not exist", req); - ast_set_flag(p, SIP_NEEDDESTROY); - } - return; - } else if ((c = get_header(req, "X-ClientCode"))) { - /* Client code (from SNOM phone) */ - if (ast_test_flag(p, 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; - } - /* 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; -} - -/*--- sip_do_debug: Enable SIP Debugging in CLI ---*/ -static int sip_do_debug_ip(int fd, int argc, char *argv[]) -{ - struct hostent *hp; - struct ast_hostent ahp; - char iabuf[INET_ADDRSTRLEN]; - int port = 0; - char *p, *arg; - - if (argc != 4) - return RESULT_SHOWUSAGE; - arg = argv[3]; - p = strstr(arg, ":"); - if (p) { - *p = '\0'; - 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(iabuf, sizeof(iabuf), debugaddr.sin_addr)); - else - ast_cli(fd, "SIP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), debugaddr.sin_addr), port); - sipdebug = 1; - return RESULT_SUCCESS; -} - -static int sip_do_debug_peer(int fd, int argc, char *argv[]) -{ - struct sip_peer *peer; - char iabuf[INET_ADDRSTRLEN]; - if (argc != 4) - return RESULT_SHOWUSAGE; - peer = find_peer(argv[3], NULL, 1); - if (peer) { - if (peer->addr.sin_addr.s_addr) { - debugaddr.sin_family = AF_INET; - memcpy(&debugaddr.sin_addr, &peer->addr.sin_addr, sizeof(debugaddr.sin_addr)); - debugaddr.sin_port = peer->addr.sin_port; - ast_cli(fd, "SIP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), debugaddr.sin_addr), ntohs(debugaddr.sin_port)); - sipdebug = 1; - } else - ast_cli(fd, "Unable to get IP address of peer '%s'\n", argv[3]); - ASTOBJ_UNREF(peer,sip_destroy_peer); - } else - ast_cli(fd, "No such peer '%s'\n", argv[3]); - return RESULT_SUCCESS; -} - -/*--- sip_do_debug: Turn on SIP debugging (CLI command) */ -static int sip_do_debug(int fd, int argc, char *argv[]) -{ - int oldsipdebug = sipdebug; - if (argc != 2) { - if (argc != 4) - return RESULT_SHOWUSAGE; - else if (strncmp(argv[2], "ip\0", 3) == 0) - return sip_do_debug_ip(fd, argc, argv); - else if (strncmp(argv[2], "peer\0", 5) == 0) - return sip_do_debug_peer(fd, argc, argv); - else return RESULT_SHOWUSAGE; - } - sipdebug = 1; - memset(&debugaddr, 0, sizeof(debugaddr)); - if (oldsipdebug) - ast_cli(fd, "SIP Debugging re-enabled\n"); - else - ast_cli(fd, "SIP Debugging enabled\n"); - return RESULT_SUCCESS; -} - -/*--- sip_notify: 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; - - p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY); - if (!p) { - ast_log(LOG_WARNING, "Unable to build sip pvt data for notify\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, NULL); - - for (var = varlist; var; var = var->next) - add_header(&req, var->name, var->value); - - add_blank_header(&req); - /* Recalculate our side, and recalculate Call ID */ - if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip)) - memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); - build_via(p, p->via, sizeof(p->via)); - build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); - ast_cli(fd, "Sending NOTIFY of type '%s' to '%s'\n", argv[2], argv[i]); - transmit_sip_request(p, &req); - sip_scheddestroy(p, 15000); - } - - return RESULT_SUCCESS; -} -/*--- sip_do_history: Enable SIP History logging (CLI) ---*/ -static int sip_do_history(int fd, int argc, char *argv[]) -{ - if (argc != 2) { - return RESULT_SHOWUSAGE; - } - recordhistory = 1; - ast_cli(fd, "SIP History Recording Enabled (use 'sip show history')\n"); - return RESULT_SUCCESS; -} - -/*--- sip_no_history: Disable SIP History logging (CLI) ---*/ -static int sip_no_history(int fd, int argc, char *argv[]) -{ - if (argc != 3) { - return RESULT_SHOWUSAGE; - } - recordhistory = 0; - ast_cli(fd, "SIP History Recording Disabled\n"); - return RESULT_SUCCESS; -} - -/*--- sip_no_debug: Disable SIP Debugging in CLI ---*/ -static int sip_no_debug(int fd, int argc, char *argv[]) - -{ - if (argc != 3) - return RESULT_SHOWUSAGE; - sipdebug = 0; - ast_cli(fd, "SIP Debugging Disabled\n"); - return RESULT_SUCCESS; -} - -static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len); - -/*--- do_register_auth: 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 (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); -} - -/*--- do_proxy_auth: 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]; - struct sip_invite_param options; - - memset(&options, 0, sizeof(struct sip_invite_param)); - p->authtries++; - 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 */ - options.auth = digest; - options.authheader = respheader; - return transmit_invite(p, sipmethod, sipmethod == SIP_INVITE, &options, init); -} - -/*--- reply_digest: reply to authentication for outbound registrations ---*/ -/* 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; - - /* table of recognised keywords, and places where they should be copied */ - const struct x { - const char *key; - char *dst; - int dstlen; - } *i, keys[] = { - { "realm=", p->realm, sizeof(p->realm) }, - { "nonce=", p->nonce, sizeof(p->nonce) }, - { "opaque=", p->opaque, sizeof(p->opaque) }, - { "qop=", p->qop, sizeof(p->qop) }, - { "domain=", p->domain, sizeof(p->domain) }, - { NULL, 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 "); - for (i = keys; i->key != NULL; i++) - i->dst[0] = '\0'; /* init all to empty strings */ - 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_copy_string(i->dst, src, i->dstlen); - break; - } - if (i->key == NULL) /* not found, try ',' */ - strsep(&c, ","); - } - - /* Save auth data for following registrations */ - if (p->registry) { - struct sip_registry *r = p->registry; - - ast_copy_string(r->realm, p->realm, sizeof(r->realm)); - ast_copy_string(r->nonce, p->nonce, sizeof(r->nonce)); - ast_copy_string(r->domain, p->domain, sizeof(r->domain)); - ast_copy_string(r->opaque, p->opaque, sizeof(r->opaque)); - ast_copy_string(r->qop, p->qop, sizeof(r->qop)); - } - build_reply_digest(p, sipmethod, digest, digest_len); - return 0; -} - -/*--- build_reply_digest: Build reply digest ---*/ -/* 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 cnonce[80]; - char iabuf[INET_ADDRSTRLEN]; - char *username; - char *secret; - char *md5secret; - struct sip_auth *auth = (struct sip_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(iabuf, sizeof(iabuf), p->sa.sin_addr)); - - snprintf(cnonce, sizeof(cnonce), "%08x", rand()); - - /* Check if we have separate auth credentials */ - if ((auth = find_realm_authentication(authl, p->realm))) { - username = auth->username; - secret = auth->secret; - md5secret = auth->md5secret; - ast_log(LOG_NOTICE,"Using realm %s authentication for this call\n", p->realm); - } else { - /* No authentication, use peer or register= config */ - username = p->authname; - secret = p->peersecret; - md5secret = p->peermd5secret; - } - - - /* 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); - /* XXX We hard code the nonce-number to 1... What are the odds? Are we seriously going to keep - track of every nonce we've seen? Also we hard code to "auth"... XXX */ - if (!ast_strlen_zero(p->qop)) - snprintf(resp,sizeof(resp),"%s:%s:%s:%s:%s:%s",a1_hash,p->nonce, "00000001", cnonce, "auth", a2_hash); - else - snprintf(resp,sizeof(resp),"%s:%s:%s",a1_hash,p->nonce,a2_hash); - ast_md5_hash(resp_hash,resp); - /* 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\", opaque=\"%s\", qop=\"%s\", cnonce=\"%s\", nc=%s", username, p->realm, uri, p->nonce, resp_hash, p->opaque, "auth", cnonce, "00000001"); - else - snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\", opaque=\"%s\"", username, p->realm, uri, p->nonce, resp_hash, p->opaque); - - return 0; -} - - - -static char notify_usage[] = -"Usage: sip notify [...]\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 ]\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 [load]\n" -" Lists 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 \n" -" Provides detailed status on a given SIP channel.\n"; - -static char show_history_usage[] = -"Usage: sip show history \n" -" Provides detailed dialog history on a given SIP channel.\n"; - -static char show_peers_usage[] = -"Usage: sip show peers [like ]\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 [load]\n" -" Lists 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] [|all|like ]\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 debug\n" -" Enables dumping of SIP packets for debugging purposes\n\n" -" sip debug ip \n" -" Enables dumping of SIP packets to and from host.\n\n" -" sip debug peer \n" -" Enables dumping of SIP packets to and from host.\n" -" Require peer to be registered.\n"; - -static char no_debug_usage[] = -"Usage: sip no debug\n" -" Disables dumping of SIP packets for debugging purposes\n"; - -static char no_history_usage[] = -"Usage: sip no history\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" -" Shows active SIP subscriptions for extension states\n"; - -static char show_objects_usage[] = -"Usage: sip show objects\n" -" Shows status of known SIP objects\n"; - - -/*--- func_header_read: Read SIP header (dialplan function) */ -static char *func_header_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) -{ - struct sip_pvt *p; - char *content; - - if (!data) { - ast_log(LOG_WARNING, "This function requires a header name.\n"); - return NULL; - } - - ast_mutex_lock(&chan->lock); - if (chan->type != channeltype) { - ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n"); - ast_mutex_unlock(&chan->lock); - return NULL; - } - - p = chan->tech_pvt; - content = get_header(&p->initreq, data); - - if (ast_strlen_zero(content)) { - ast_mutex_unlock(&chan->lock); - return NULL; - } - - ast_copy_string(buf, content, len); - ast_mutex_unlock(&chan->lock); - - return buf; -} - - -static struct ast_custom_function sip_header_function = { - .name = "SIP_HEADER", - .synopsis = "Gets or sets the specified SIP header", - .syntax = "SIP_HEADER()", - .read = func_header_read, -}; - -/*--- function_sippeer: ${SIPPEER()} Dialplan function - reads peer data */ -static char *function_sippeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) -{ - char *ret = NULL; - struct sip_peer *peer; - char *peername, *colname; - char iabuf[INET_ADDRSTRLEN]; - - if (!(peername = ast_strdupa(data))) { - ast_log(LOG_ERROR, "Memory Error!\n"); - return ret; - } - - if ((colname = strchr(peername, ':'))) { - *colname = '\0'; - colname++; - } else { - colname = "ip"; - } - if (!(peer = find_peer(peername, NULL, 1))) - return ret; - - if (!strcasecmp(colname, "ip")) { - ast_copy_string(buf, peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "", 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, SIP_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, *ptr; - int index = 0, codec = 0; - - codecnum = strchr(colname, '['); - *codecnum = '\0'; - codecnum++; - if ((ptr = strchr(codecnum, ']'))) { - *ptr = '\0'; - } - index = atoi(codecnum); - if((codec = ast_codec_pref_index(&peer->prefs, index))) { - ast_copy_string(buf, ast_getformatname(codec), len); - } - } - ret = buf; - - ASTOBJ_UNREF(peer, sip_destroy_peer); - - return ret; -} - -struct ast_custom_function sippeer_function = { - .name = "SIPPEER", - .synopsis = "Gets SIP peer information", - .syntax = "SIPPEER([: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" - "- codec[x] Preferred codec index number 'x' (beginning with zero).\n" - "\n" -}; - -/*--- parse_moved_contact: Parse 302 Moved temporalily response */ -static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req) -{ - char tmp[256] = ""; - char *s, *e; - ast_copy_string(tmp, get_header(req, "Contact"), sizeof(tmp)); - s = ditch_braces(tmp); - e = strchr(s, ';'); - if (e) - *e = '\0'; - if (ast_test_flag(p, SIP_PROMISCREDIR)) { - if (!strncasecmp(s, "sip:", 4)) - s += 4; - e = strchr(s, '/'); - if (e) - *e = '\0'; - ast_log(LOG_DEBUG, "Found promiscuous redirection to 'SIP/%s'\n", s); - if (p->owner) - snprintf(p->owner->call_forward, sizeof(p->owner->call_forward), "SIP/%s", s); - } else { - e = strchr(tmp, '@'); - if (e) - *e = '\0'; - e = strchr(tmp, '/'); - if (e) - *e = '\0'; - if (!strncasecmp(s, "sip:", 4)) - s += 4; - ast_log(LOG_DEBUG, "Found 302 Redirect to extension '%s'\n", s); - if (p->owner) - ast_copy_string(p->owner->call_forward, s, sizeof(p->owner->call_forward)); - } -} - -/*--- check_pendings: Check pending actions on SIP call ---*/ -static void check_pendings(struct sip_pvt *p) -{ - /* Go ahead and send bye at this point */ - if (ast_test_flag(p, SIP_PENDINGBYE)) { - transmit_request_with_auth(p, SIP_BYE, 0, 1, 1); - ast_set_flag(p, SIP_NEEDDESTROY); - ast_clear_flag(p, SIP_NEEDREINVITE); - } else if (ast_test_flag(p, SIP_NEEDREINVITE)) { - 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, SIP_NEEDREINVITE); - } -} - -/*--- handle_response_register: 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 > 1) || 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, 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); - p->registry->regattempts = global_regattempts_max+1; - ast_sched_del(sched, r->timeout); - ast_set_flag(p, 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); - p->registry->regattempts = global_regattempts_max+1; - ast_set_flag(p, SIP_NEEDDESTROY); - r->call = NULL; - ast_sched_del(sched, r->timeout); - break; - case 407: /* Proxy auth */ - if ((p->authtries > 1) || 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, SIP_NEEDDESTROY); - } - 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); - p->registry->regattempts = global_regattempts_max+1; - ast_set_flag(p, 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 that isn't a register\n"); - ast_set_flag(p, SIP_NEEDDESTROY); - return 0; - } - - r->regstate=REG_STATE_REGISTERED; - manager_event(EVENT_FLAG_SYSTEM, "Registry", "Channel: SIP\r\nDomain: %s\r\nStatus: %s\r\n", r->hostname, regstate2str(r->regstate)); - r->regattempts = 0; - ast_log(LOG_DEBUG, "Registration successful\n"); - if (r->timeout > -1) { - ast_log(LOG_DEBUG, "Cancelling timeout %d\n", r->timeout); - ast_sched_del(sched, r->timeout); - } - r->timeout=-1; - r->call = NULL; - p->registry = NULL; - /* Let this one hang around until we have all the responses */ - sip_scheddestroy(p, 32000); - /* ast_set_flag(p, SIP_NEEDDESTROY); */ - - /* set us up for re-registering */ - /* figure out how long we got registered for */ - if (r->expire > -1) - ast_sched_del(sched, r->expire); - /* according to section 6.13 of RFC, contact headers override - expires headers, so check those first */ - expires = 0; - if (!ast_strlen_zero(get_header(req, "Contact"))) { - char *contact = NULL; - 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 = strstr(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 ms)\n", r->hostname, expires, expires_ms); - - r->refresh= (int) expires_ms / 1000; - - /* Schedule re-registration before we expire */ - r->expire=ast_sched_add(sched, expires_ms, sip_reregister, r); - ASTOBJ_UNREF(r, sip_registry_destroy); - } - return 1; -} - -/*--- handle_response_peerpoke: Handle qualification responses (OPTIONS) */ -static int handle_response_peerpoke(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno, int sipmethod) -{ - struct sip_peer *peer; - int pingtime; - struct timeval tv; - if (resp != 100) { - int statechanged = 0; - int newstate = 0; - peer = p->peerpoke; - gettimeofday(&tv, NULL); - pingtime = ast_tvdiff_ms(tv, peer->ps); - if (pingtime < 1) - pingtime = 1; - if ((peer->lastms < 0) || (peer->lastms > peer->maxms)) { - if (pingtime <= peer->maxms) { - ast_log(LOG_NOTICE, "Peer '%s' is now REACHABLE! (%dms / %dms)\n", peer->name, pingtime, peer->maxms); - statechanged = 1; - newstate = 1; - } - } else if ((peer->lastms > 0) && (peer->lastms <= peer->maxms)) { - if (pingtime > peer->maxms) { - ast_log(LOG_NOTICE, "Peer '%s' is now TOO LAGGED! (%dms / %dms)\n", peer->name, pingtime, peer->maxms); - statechanged = 1; - newstate = 2; - } - } - if (!peer->lastms) - statechanged = 1; - peer->lastms = pingtime; - peer->call = NULL; - if (statechanged) { - ast_device_state_changed("SIP/%s", peer->name); - if (newstate == 2) { - manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Lagged\r\nTime: %d\r\n", peer->name, pingtime); - } else { - manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Reachable\r\nTime: %d\r\n", peer->name, pingtime); - } - } - - if (peer->pokeexpire > -1) - ast_sched_del(sched, peer->pokeexpire); - if (sipmethod == SIP_INVITE) /* Does this really happen? */ - transmit_request(p, SIP_ACK, seqno, 0, 0); - ast_set_flag(p, SIP_NEEDDESTROY); - - /* Try again eventually */ - if ((peer->lastms < 0) || (peer->lastms > peer->maxms)) - peer->pokeexpire = ast_sched_add(sched, DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer); - else - peer->pokeexpire = ast_sched_add(sched, DEFAULT_FREQ_OK, sip_poke_peer_s, peer); - } - return 1; -} - -/*--- handle_response: Handle SIP response in dialogue ---*/ -static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno) -{ - char *to; - char *msg, *c; - struct ast_channel *owner; - char iabuf[INET_ADDRSTRLEN]; - int sipmethod; - int res = 1; - - c = get_header(req, "Cseq"); - msg = strchr(c, ' '); /* Find method */ - if (!msg) - msg = ""; - else - msg++; - owner = p->owner; - - if (owner) - owner->hangupcause = hangup_sip2cause(resp); - - sipmethod = find_sip_method(msg); - - /* 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); - - /* Get their tag if we haven't already */ - if (ast_strlen_zero(p->theirtag)) { - to = get_header(req, "To"); - to = strcasestr(to, "tag="); - if (to) { - to += 4; - ast_copy_string(p->theirtag, to, sizeof(p->theirtag)); - to = strchr(p->theirtag, ';'); - if (to) - *to = '\0'; - } - } - if (p->peerpoke) { - /* 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" */ - - res = handle_response_peerpoke(p, resp, rest, req, ignore, seqno, sipmethod); - } else if (ast_test_flag(p, SIP_OUTGOING)) { - /* Acknowledge sequence number */ - if (p->initid > -1) { - /* Don't auto congest anymore since we've gotten something useful back */ - ast_sched_del(sched, p->initid); - p->initid = -1; - } - switch(resp) { - case 100: /* 100 Trying */ - if (sipmethod == SIP_INVITE) { - sip_cancel_destroy(p); - } - break; - case 183: /* 183 Session Progress */ - if (sipmethod == SIP_INVITE) { - sip_cancel_destroy(p); - if (!ast_strlen_zero(get_header(req, "Content-Type"))) - process_sdp(p, req); - if (p->owner) { - /* Queue a progress frame */ - ast_queue_control(p->owner, AST_CONTROL_PROGRESS); - } - } - break; - case 180: /* 180 Ringing */ - if (sipmethod == SIP_INVITE) { - sip_cancel_destroy(p); - if (p->owner) { - ast_queue_control(p->owner, AST_CONTROL_RINGING); - if (p->owner->_state != AST_STATE_UP) - ast_setstate(p->owner, AST_STATE_RINGING); - } - } - break; - case 200: /* 200 OK */ - if (sipmethod == SIP_NOTIFY) { - /* They got the notify, this is the end */ - if (p->owner) { - ast_log(LOG_WARNING, "Notify answer on an owned channel?\n"); - ast_queue_hangup(p->owner); - } else { - if (!p->subscribed) { - ast_set_flag(p, SIP_NEEDDESTROY); - } - } - } else if (sipmethod == SIP_INVITE) { - /* 200 OK on invite - someone's answering our call */ - sip_cancel_destroy(p); - if (!ast_strlen_zero(get_header(req, "Content-Type"))) - process_sdp(p, req); - - /* Parse contact header for continued conversation */ - /* When we get 200 OK, we now which device (and IP) to contact for this call */ - /* This is important when we have a SIP proxy between us and the phone */ - parse_ok_contact(p, req); - /* Save Record-Route for any later requests we make on this dialogue */ - build_route(p, req, 1); - if (p->owner) { - if (p->owner->_state != AST_STATE_UP) { -#ifdef OSP_SUPPORT - time(&p->ospstart); -#endif - ast_queue_control(p->owner, AST_CONTROL_ANSWER); - } else { - struct ast_frame af = { AST_FRAME_NULL, }; - ast_queue_frame(p->owner, &af); - } - } else /* It's possible we're getting an ACK after we've tried to disconnect - by sending CANCEL */ - ast_set_flag(p, SIP_PENDINGBYE); - p->authtries = 0; - /* If I understand this right, the branch is different for a non-200 ACK only */ - transmit_request(p, SIP_ACK, seqno, 0, 1); - check_pendings(p); - } else if (sipmethod == SIP_REGISTER) { - res = handle_response_register(p, resp, rest, req, ignore, seqno); - } - break; - case 401: /* Not www-authorized on SIP method */ - if (sipmethod == SIP_INVITE) { - /* First we ACK */ - transmit_request(p, SIP_ACK, seqno, 0, 0); - /* Then we AUTH */ - p->theirtag[0]='\0'; /* forget their old tag, so we don't match tags when getting response */ - if ((p->authtries > 1) || do_proxy_auth(p, req, "WWW-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, SIP_NEEDDESTROY); - } - } else if (p->registry && sipmethod == SIP_REGISTER) { - res = handle_response_register(p, resp, rest, req, ignore, seqno); - } 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, SIP_NEEDDESTROY); - } - break; - case 403: /* Forbidden - we failed authentication */ - if (sipmethod == SIP_INVITE) { - /* First we ACK */ - transmit_request(p, SIP_ACK, seqno, 0, 0); - ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for INVITE to '%s'\n", get_header(&p->initreq, "From")); - if (owner) - ast_queue_control(p->owner, AST_CONTROL_CONGESTION); - ast_set_flag(p, SIP_NEEDDESTROY); - } else if (p->registry && sipmethod == SIP_REGISTER) { - res = handle_response_register(p, resp, rest, req, ignore, seqno); - } else { - ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for %s\n", msg); - } - break; - case 404: /* Not found */ - if (p->registry && sipmethod == SIP_REGISTER) { - res = handle_response_register(p, resp, rest, req, ignore, seqno); - } else if (owner) - ast_queue_control(p->owner, AST_CONTROL_CONGESTION); - break; - case 407: /* Proxy auth required */ - if (sipmethod == SIP_INVITE) { - /* First we ACK */ - transmit_request(p, SIP_ACK, seqno, 0, 0); - /* Then we AUTH */ - /* But only if the packet wasn't marked as ignore in handle_request */ - if (!ignore){ - p->theirtag[0]='\0'; /* forget their old tag, so we don't match tags when getting response */ - if ((p->authtries > 1) || do_proxy_auth(p, req, "Proxy-Authenticate", "Proxy-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, SIP_NEEDDESTROY); - } - } - } else if (sipmethod == SIP_BYE || sipmethod == SIP_REFER) { - 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(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port)); - ast_set_flag(p, SIP_NEEDDESTROY); - if ((p->authtries > 1) || 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, SIP_NEEDDESTROY); - } - } else if (p->registry && sipmethod == SIP_REGISTER) { - res = handle_response_register(p, resp, rest, req, ignore, seqno); - } else - ast_set_flag(p, SIP_NEEDDESTROY); - - break; - case 501: /* Not Implemented */ - if (sipmethod == SIP_INVITE) { - if (p->owner) - ast_queue_control(p->owner, AST_CONTROL_CONGESTION); - } else - ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), msg); - break; - default: - if ((resp >= 300) && (resp < 700)) { - 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(iabuf, sizeof(iabuf), p->sa.sin_addr)); - ast_set_flag(p, SIP_ALREADYGONE); - if (p->rtp) { - /* Immediately stop RTP */ - ast_rtp_stop(p->rtp); - } - if (p->vrtp) { - /* Immediately stop VRTP */ - ast_rtp_stop(p->vrtp); - } - /* 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); - if (p->owner) - ast_queue_control(p->owner, AST_CONTROL_BUSY); - break; - case 487: - /* channel now destroyed - dec the inUse counter */ - if (ast_test_flag(p, SIP_OUTGOING)) { - update_user_counter(p, DEC_OUT_USE); - } - else { - update_user_counter(p, DEC_IN_USE); - } - break; - case 482: /* 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... */ - ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n"); - if (p->owner) - snprintf(p->owner->call_forward, sizeof(p->owner->call_forward), "Local/%s@%s", p->username, p->context); - /* 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 480: /* Temporarily Unavailable */ - case 404: /* Not Found */ - case 410: /* Gone */ - case 400: /* Bad Request */ - case 500: /* Server error */ - case 503: /* Service Unavailable */ - if (owner) - ast_queue_control(p->owner, AST_CONTROL_CONGESTION); - break; - default: - /* Send hangup */ - if (owner) - ast_queue_hangup(p->owner); - break; - } - /* ACK on invite */ - if (sipmethod == SIP_INVITE) - transmit_request(p, SIP_ACK, seqno, 0, 0); - ast_set_flag(p, SIP_ALREADYGONE); - if (!p->owner) - ast_set_flag(p, SIP_NEEDDESTROY); - } else if ((resp >= 100) && (resp < 200)) { - if (sipmethod == SIP_INVITE) { - sip_cancel_destroy(p); - if (!ast_strlen_zero(get_header(req, "Content-Type"))) - 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(iabuf, sizeof(iabuf), p->sa.sin_addr)); - } - } else { - /* Not outgoing - what is it? Unsolicited replies? */ - /* When do we get here? ---------??????????------------*/ - /* INCOMING Calls */ - if (option_debug > 2) { - ast_verbose("!!!!!!!---------------************* Why are we here with this packet???? %s\n", msg); - } - if (sip_debug_test_pvt(p)) - ast_verbose("Response message is %s\n", msg); - switch(resp) { - case 200: - /* Change branch since this is a 200 response */ - if (sipmethod == SIP_INVITE) - transmit_request(p, SIP_ACK, seqno, 0, 1); - break; - case 407: - if (sipmethod == SIP_BYE || sipmethod == SIP_REFER) { - 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(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port)); - if ((p->authtries > 1) || 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, SIP_NEEDDESTROY); - } - } - break; - } - } -} - -struct sip_dual { - struct ast_channel *chan1; - struct ast_channel *chan2; - struct sip_request req; -}; - -/*--- sip_park_thread: Park SIP call support function */ -static void *sip_park_thread(void *stuff) -{ - struct ast_channel *chan1, *chan2; - struct sip_dual *d; - struct sip_request req; - int ext; - int res; - d = stuff; - chan1 = d->chan1; - chan2 = d->chan2; - copy_request(&req, &d->req); - free(d); - ast_mutex_lock(&chan1->lock); - ast_do_masquerade(chan1); - ast_mutex_unlock(&chan1->lock); - res = ast_park_call(chan1, chan2, 0, &ext); - /* Then hangup */ - ast_hangup(chan2); - ast_log(LOG_DEBUG, "Parked on extension '%d'\n", ext); - return NULL; -} - -/*--- sip_park: Park a call ---*/ -static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req) -{ - struct sip_dual *d; - struct ast_channel *chan1m, *chan2m; - pthread_t th; - chan1m = ast_channel_alloc(0); - chan2m = ast_channel_alloc(0); - if ((!chan2m) || (!chan1m)) { - if (chan1m) - ast_hangup(chan1m); - if (chan2m) - ast_hangup(chan2m); - return -1; - } - snprintf(chan1m->name, sizeof(chan1m->name), "Parking/%s", chan1->name); - /* Make formats okay */ - chan1m->readformat = chan1->readformat; - chan1m->writeformat = chan1->writeformat; - ast_channel_masquerade(chan1m, chan1); - /* Setup the extensions and such */ - ast_copy_string(chan1m->context, chan1->context, sizeof(chan1m->context)); - ast_copy_string(chan1m->exten, chan1->exten, sizeof(chan1m->exten)); - chan1m->priority = chan1->priority; - - /* We make a clone of the peer channel too, so we can play - back the announcement */ - snprintf(chan2m->name, sizeof (chan2m->name), "SIPPeer/%s",chan2->name); - /* Make formats okay */ - chan2m->readformat = chan2->readformat; - chan2m->writeformat = chan2->writeformat; - ast_channel_masquerade(chan2m, chan2); - /* Setup the extensions and such */ - ast_copy_string(chan2m->context, chan2->context, sizeof(chan2m->context)); - ast_copy_string(chan2m->exten, chan2->exten, sizeof(chan2m->exten)); - chan2m->priority = chan2->priority; - ast_mutex_lock(&chan2m->lock); - if (ast_do_masquerade(chan2m)) { - ast_log(LOG_WARNING, "Masquerade failed :(\n"); - ast_mutex_unlock(&chan2m->lock); - ast_hangup(chan2m); - return -1; - } - ast_mutex_unlock(&chan2m->lock); - d = malloc(sizeof(struct sip_dual)); - if (d) { - memset(d, 0, sizeof(*d)); - /* Save original request for followup */ - copy_request(&d->req, req); - d->chan1 = chan1m; - d->chan2 = chan2m; - if (!ast_pthread_create(&th, NULL, sip_park_thread, d)) - return 0; - free(d); - } - return -1; -} - -/*--- ast_quiet_chan: Turn off generator data */ -static void ast_quiet_chan(struct ast_channel *chan) -{ - if (chan && chan->_state == AST_STATE_UP) { - if (chan->generatordata) - ast_deactivate_generator(chan); - } -} - -/*--- attempt_transfer: Attempt transfer of SIP call ---*/ -static int attempt_transfer(struct sip_pvt *p1, struct sip_pvt *p2) -{ - int res = 0; - struct ast_channel - *chana = NULL, - *chanb = NULL, - *bridgea = NULL, - *bridgeb = NULL, - *peera = NULL, - *peerb = NULL, - *peerc = NULL, - *peerd = NULL; - - if (!p1->owner || !p2->owner) { - ast_log(LOG_WARNING, "Transfer attempted without dual ownership?\n"); - return -1; - } - chana = p1->owner; - chanb = p2->owner; - bridgea = ast_bridged_channel(chana); - bridgeb = ast_bridged_channel(chanb); - - if (bridgea) { - peera = chana; - peerb = chanb; - peerc = bridgea; - peerd = bridgeb; - } else if (bridgeb) { - peera = chanb; - peerb = chana; - peerc = bridgeb; - peerd = bridgea; - } - - if (peera && peerb && peerc && (peerb != peerc)) { - ast_quiet_chan(peera); - ast_quiet_chan(peerb); - ast_quiet_chan(peerc); - ast_quiet_chan(peerd); - - if (peera->cdr && peerb->cdr) { - peerb->cdr = ast_cdr_append(peerb->cdr, peera->cdr); - } else if (peera->cdr) { - peerb->cdr = peera->cdr; - } - peera->cdr = NULL; - - if (peerb->cdr && peerc->cdr) { - peerb->cdr = ast_cdr_append(peerb->cdr, peerc->cdr); - } else if (peerc->cdr) { - peerb->cdr = peerc->cdr; - } - peerc->cdr = NULL; - - if (ast_channel_masquerade(peerb, peerc)) { - ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", peerb->name, peerc->name); - res = -1; - } - return res; - } else { - ast_log(LOG_NOTICE, "Transfer attempted with no appropriate bridged calls to transfer\n"); - if (chana) - ast_softhangup_nolock(chana, AST_SOFTHANGUP_DEV); - if (chanb) - ast_softhangup_nolock(chanb, AST_SOFTHANGUP_DEV); - return -1; - } - return 0; -} - -/*--- handle_request_options: Handle incoming OPTIONS request */ -static int handle_request_options(struct sip_pvt *p, struct sip_request *req, int debug) -{ - int res; - - res = get_destination(p, req); - build_contact(p); - /* XXX Should we authenticate OPTIONS? XXX */ - if (ast_strlen_zero(p->context)) - strcpy(p->context, default_context); - if (res < 0) - transmit_response_with_allow(p, "404 Not Found", req, 0); - else if (res > 0) - transmit_response_with_allow(p, "484 Address Incomplete", 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. */ - if (!p->lastinvite) - ast_set_flag(p, SIP_NEEDDESTROY); - - return res; -} - -/*--- handle_request_invite: Handle incoming INVITE request */ -static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin, int *recount, char *e) -{ - int res = 1; - struct ast_channel *c=NULL; - int gotdest; - struct ast_frame af = { AST_FRAME_NULL, }; - char *supported; - char *required; - unsigned int required_profile = 0; - - /* Find out what they support */ - if (!p->sipoptions) { - supported = get_header(req, "Supported"); - if (supported) - parse_sip_options(p, supported); - } - required = get_header(req, "Required"); - if (required && !ast_strlen_zero(required)) { - required_profile = parse_sip_options(NULL, required); - if (required_profile) { /* They require something */ - /* At this point we support no extensions, so fail */ - transmit_response_with_unsupported(p, "420 Bad extension", req, required); - ast_set_flag(p, SIP_NEEDDESTROY); - return -1; - - } - } - - /* Check if this is a loop */ - /* This happens since we do not properly support SIP domain - handling yet... -oej */ - if (ast_test_flag(p, 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 */ - transmit_response(p, "482 Loop Detected", req); - /* We do NOT destroy p here, so that our response will be accepted */ - return 0; - } - if (!ignore) { - /* Use this as the basis */ - if (debug) - ast_verbose("Using INVITE request as basis request - %s\n", p->callid); - sip_cancel_destroy(p); - /* This call is no longer outgoing if it ever was */ - ast_clear_flag(p, SIP_OUTGOING); - /* This also counts as a pending invite */ - p->pendinginvite = seqno; - copy_request(&p->initreq, req); - check_via(p, req); - if (p->owner) { - /* Handle SDP here if we already have an owner */ - if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) { - if (process_sdp(p, req)) { - transmit_response(p, "488 Not acceptable here", req); - ast_set_flag(p, SIP_NEEDDESTROY); - return -1; - } - } else { - p->jointcapability = p->capability; - ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n"); - } - } - } else if (debug) - ast_verbose("Ignoring this request\n"); - if (!p->lastinvite && !ignore && !p->owner) { - /* Handle authentication if this is our first invite */ - res = check_user(p, req, SIP_INVITE, e, 1, sin, ignore); - if (res) { - if (res < 0) { - ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From")); - if (ignore) - transmit_response(p, "403 Forbidden", req); - else - transmit_response_reliable(p, "403 Forbidden", req, 1); - ast_set_flag(p, SIP_NEEDDESTROY); - } - return 0; - } - /* Process the SDP portion */ - if (!ast_strlen_zero(get_header(req, "Content-Type"))) { - if (process_sdp(p, req)) { - transmit_response(p, "488 Not acceptable here", req); - ast_set_flag(p, SIP_NEEDDESTROY); - return -1; - } - } else { - p->jointcapability = p->capability; - ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n"); - } - /* Queue NULL frame to prod ast_rtp_bridge if appropriate */ - if (p->owner) - ast_queue_frame(p->owner, &af); - /* Initialize the context if it hasn't been already */ - if (ast_strlen_zero(p->context)) - strcpy(p->context, default_context); - /* Check number of concurrent calls -vs- incoming limit HERE */ - ast_log(LOG_DEBUG, "Checking SIP call limits for device %s\n", p->username); - res = update_user_counter(p, INC_IN_USE); - if (res) { - if (res < 0) { - ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username); - if (ignore) - transmit_response(p, "480 Temporarily Unavailable (Call limit)", req); - else - transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req, 1); - ast_set_flag(p, SIP_NEEDDESTROY); - } - return 0; - } - /* Get destination right away */ - gotdest = get_destination(p, NULL); - get_rdnis(p, NULL); - extract_uri(p, req); - build_contact(p); - - if (gotdest) { - if (gotdest < 0) { - if (ignore) - transmit_response(p, "404 Not Found", req); - else - transmit_response_reliable(p, "404 Not Found", req, 1); - update_user_counter(p,DEC_IN_USE); - } else { - if (ignore) - transmit_response(p, "484 Address Incomplete", req); - else - transmit_response_reliable(p, "484 Address Incomplete", req, 1); - update_user_counter(p,DEC_IN_USE); - } - ast_set_flag(p, SIP_NEEDDESTROY); - } else { - /* If no extension was specified, use the s one */ - if (ast_strlen_zero(p->exten)) - ast_copy_string(p->exten, "s", sizeof(p->exten)); - /* Initialize tag */ - p->tag = rand(); - /* First invitation */ - c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username ); - *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_mutex_lock(&c->lock); - } - } - - } else { - if (option_debug > 1 && sipdebug) - ast_log(LOG_DEBUG, "Got a SIP re-invite for call %s\n", p->callid); - c = p->owner; - } - if (!ignore && p) - p->lastinvite = seqno; - if (c) { - switch(c->_state) { - case AST_STATE_DOWN: - transmit_response(p, "100 Trying", req); - ast_setstate(c, AST_STATE_RING); - if (strcmp(p->exten, ast_pickup_ext())) { - if (ast_pbx_start(c)) { - ast_log(LOG_WARNING, "Failed to start PBX :(\n"); - /* 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); - if (ignore) - transmit_response(p, "503 Unavailable", req); - else - transmit_response_reliable(p, "503 Unavailable", req, 1); - c = NULL; - } - } else { - ast_mutex_unlock(&c->lock); - if (ast_pickup_call(c)) { - ast_log(LOG_NOTICE, "Nothing to pick up\n"); - if (ignore) - transmit_response(p, "503 Unavailable", req); - else - transmit_response_reliable(p, "503 Unavailable", req, 1); - ast_set_flag(p, SIP_ALREADYGONE); - /* Unlock locks so ast_hangup can do its magic */ - ast_mutex_unlock(&p->lock); - ast_hangup(c); - ast_mutex_lock(&p->lock); - c = NULL; - } else { - ast_mutex_unlock(&p->lock); - ast_setstate(c, AST_STATE_DOWN); - ast_hangup(c); - ast_mutex_lock(&p->lock); - c = NULL; - } - } - break; - case AST_STATE_RING: - transmit_response(p, "100 Trying", req); - break; - case AST_STATE_RINGING: - transmit_response(p, "180 Ringing", req); - break; - case AST_STATE_UP: - transmit_response_with_sdp(p, "200 OK", req, 1); - 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); - } - } else { - if (p && !ast_test_flag(p, SIP_NEEDDESTROY)) { - if (!p->jointcapability) { - if (ignore) - transmit_response(p, "488 Not Acceptable Here (codec error)", req); - else - transmit_response_reliable(p, "488 Not Acceptable Here (codec error)", req, 1); - ast_set_flag(p, SIP_NEEDDESTROY); - } else { - ast_log(LOG_NOTICE, "Unable to create/find channel\n"); - if (ignore) - transmit_response(p, "503 Unavailable", req); - else - transmit_response_reliable(p, "503 Unavailable", req, 1); - ast_set_flag(p, SIP_NEEDDESTROY); - } - } - } - return res; -} - -/*--- handle_request_refer: Handle incoming REFER request ---*/ -static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, int *nounlock) -{ - struct ast_channel *c=NULL; - int res; - struct ast_channel *transfer_to; - - if (option_debug > 2) - ast_log(LOG_DEBUG, "SIP call transfer received for call %s (REFER)!\n", p->callid); - if (ast_strlen_zero(p->context)) - strcpy(p->context, default_context); - res = get_refer_info(p, req); - if (res < 0) - transmit_response_with_allow(p, "404 Not Found", req, 1); - else if (res > 0) - transmit_response_with_allow(p, "484 Address Incomplete", req, 1); - else { - int nobye = 0; - if (!ignore) { - if (p->refer_call) { - ast_log(LOG_DEBUG,"202 Accepted (supervised)\n"); - attempt_transfer(p, p->refer_call); - if (p->refer_call->owner) - ast_mutex_unlock(&p->refer_call->owner->lock); - ast_mutex_unlock(&p->refer_call->lock); - p->refer_call = NULL; - ast_set_flag(p, SIP_GOTREFER); - } else { - ast_log(LOG_DEBUG,"202 Accepted (blind)\n"); - c = p->owner; - if (c) { - transfer_to = ast_bridged_channel(c); - if (transfer_to) { - ast_log(LOG_DEBUG, "Got SIP blind transfer, applying to '%s'\n", transfer_to->name); - ast_moh_stop(transfer_to); - if (!strcmp(p->refer_to, ast_parking_ext())) { - /* Must release c's lock now, because it will not longer - be accessible after the transfer! */ - *nounlock = 1; - ast_mutex_unlock(&c->lock); - sip_park(transfer_to, c, req); - nobye = 1; - } else { - /* Must release c's lock now, because it will not longer - be accessible after the transfer! */ - *nounlock = 1; - ast_mutex_unlock(&c->lock); - ast_async_goto(transfer_to,p->context, p->refer_to,1); - } - } else { - ast_log(LOG_DEBUG, "Got SIP blind transfer but nothing to transfer to.\n"); - ast_queue_hangup(p->owner); - } - } - ast_set_flag(p, SIP_GOTREFER); - } - transmit_response(p, "202 Accepted", req); - transmit_notify_with_sipfrag(p, seqno); - /* Always increment on a BYE */ - if (!nobye) { - transmit_request_with_auth(p, SIP_BYE, 0, 1, 1); - ast_set_flag(p, SIP_ALREADYGONE); - } - } - } - return res; -} -/*--- handle_request_cancel: Handle incoming CANCEL request ---*/ -static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req, int debug, int ignore) -{ - - check_via(p, req); - ast_set_flag(p, SIP_ALREADYGONE); - if (p->rtp) { - /* Immediately stop RTP */ - ast_rtp_stop(p->rtp); - } - if (p->vrtp) { - /* Immediately stop VRTP */ - ast_rtp_stop(p->vrtp); - } - if (p->owner) - ast_queue_hangup(p->owner); - else - ast_set_flag(p, SIP_NEEDDESTROY); - if (p->initreq.len > 0) { - if (!ignore) - transmit_response_reliable(p, "487 Request Terminated", &p->initreq, 1); - transmit_response(p, "200 OK", req); - return 1; - } else { - transmit_response(p, "481 Call Leg Does Not Exist", req); - return 0; - } -} - -/*--- handle_request_bye: Handle incoming BYE request ---*/ -static int handle_request_bye(struct sip_pvt *p, struct sip_request *req, int debug) -{ - struct ast_channel *c=NULL; - int res; - struct ast_channel *bridged_to; - char iabuf[INET_ADDRSTRLEN]; - - copy_request(&p->initreq, req); - check_via(p, req); - ast_set_flag(p, SIP_ALREADYGONE); - if (p->rtp) { - /* Immediately stop RTP */ - ast_rtp_stop(p->rtp); - } - if (p->vrtp) { - /* Immediately stop VRTP */ - ast_rtp_stop(p->vrtp); - } - 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(iabuf, sizeof(iabuf), p->recv.sin_addr)); - if (ast_strlen_zero(p->context)) - strcpy(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_moh_stop(bridged_to); - ast_async_goto(bridged_to, p->context, p->refer_to,1); - } else - ast_queue_hangup(p->owner); - } - } else { - ast_log(LOG_WARNING, "Invalid transfer information from '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr)); - ast_queue_hangup(p->owner); - } - } else if (p->owner) - ast_queue_hangup(p->owner); - else - ast_set_flag(p, SIP_NEEDDESTROY); - transmit_response(p, "200 OK", req); - - return 1; -} - -/*--- handle_request_message: Handle incoming MESSAGE request ---*/ -static int handle_request_message(struct sip_pvt *p, struct sip_request *req, int debug, int ignore) -{ - if (p->lastinvite) { - if (!ignore) { - if (debug) - ast_verbose("Receiving message!\n"); - receive_message(p, req); - } - transmit_response(p, "200 OK", req); - } else { - transmit_response(p, "405 Method Not Allowed", req); - ast_set_flag(p, SIP_NEEDDESTROY); - } - return 1; -} -/*--- handle_request_subscribe: Handle incoming SUBSCRIBE request ---*/ -static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, struct sockaddr_in *sin, int seqno, char *e) -{ - int gotdest; - int res = 0; - struct ast_channel *c=NULL; - - if (!ignore) { - /* Use this as the basis */ - if (debug) - ast_verbose("Using latest SUBSCRIBE request as basis request\n"); - /* This call is no longer outgoing if it ever was */ - ast_clear_flag(p, SIP_OUTGOING); - copy_request(&p->initreq, req); - check_via(p, req); - } else if (debug) - ast_verbose("Ignoring this SUBSCRIBE request\n"); - - if (!p->lastinvite) { - char mailbox[256]=""; - int found = 0; - - /* Handle authentication if this is our first subscribe */ - res = check_user_full(p, req, SIP_SUBSCRIBE, e, 0, sin, ignore, mailbox, sizeof(mailbox)); - if (res) { - if (res < 0) { - ast_log(LOG_NOTICE, "Failed to authenticate user %s for SUBSCRIBE\n", get_header(req, "From")); - ast_set_flag(p, SIP_NEEDDESTROY); - } - return 0; - } - /* Initialize the context if it hasn't been already */ - if (ast_strlen_zero(p->context)) - strcpy(p->context, default_context); - /* Get destination right away */ - gotdest = get_destination(p, NULL); - build_contact(p); - if (gotdest) { - if (gotdest < 0) - transmit_response(p, "404 Not Found", req); - else - transmit_response(p, "484 Address Incomplete", req); - ast_set_flag(p, SIP_NEEDDESTROY); - } else { - /* Initialize tag */ - p->tag = rand(); - if (!strcmp(get_header(req, "Accept"), "application/dialog-info+xml")) - p->subscribed = 2; - else if (!strcmp(get_header(req, "Accept"), "application/simple-message-summary")) { - /* Looks like they actually want a mailbox */ - - /* At this point, we should check if they subscribe to a mailbox that - has the same extension as the peer or the mailbox id. If we configure - the context to be the same as a SIP domain, we could check mailbox - context as well. To be able to securely accept subscribes on mailbox - IDs, not extensions, we need to check the digest auth user to make - sure that the user has access to the mailbox. - - Since we do not act on this subscribe anyway, we might as well - accept any authenticated peer with a mailbox definition in their - config section. - - */ - if (!ast_strlen_zero(mailbox)) { - found++; - } - - if (found){ - transmit_response(p, "200 OK", req); - ast_set_flag(p, SIP_NEEDDESTROY); - } else { - transmit_response(p, "403 Forbidden", req); - ast_set_flag(p, SIP_NEEDDESTROY); - } - return 0; - } else - p->subscribed = 1; - if (p->subscribed) - p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, p); - } - } else - c = p->owner; - - if (!ignore && p) - p->lastinvite = seqno; - if (p && !ast_test_flag(p, SIP_NEEDDESTROY)) { - if (!(p->expiry = atoi(get_header(req, "Expires")))) { - transmit_response(p, "200 OK", req); - ast_set_flag(p, SIP_NEEDDESTROY); - return 0; - } - /* The next line can be removed if the SNOM200 Expires bug is fixed */ - if (p->subscribed == 1) { - if (p->expiry>max_expiry) - p->expiry = max_expiry; - } - /* Go ahead and free RTP port */ - if (p->rtp) { - if (p->owner) { - p->owner->fds[0] = -1; - p->owner->fds[1] = -1; - } - ast_rtp_destroy(p->rtp); - p->rtp = NULL; - } - if (p->vrtp) { - if (p->owner) { - p->owner->fds[2] = -1; - p->owner->fds[3] = -1; - } - ast_rtp_destroy(p->vrtp); - p->vrtp = NULL; - } - transmit_response(p, "200 OK", req); - sip_scheddestroy(p, (p->expiry+10)*1000); - transmit_state_notify(p, ast_extension_state(NULL, p->context, p->exten),1); - } - return 1; -} - -/*--- handle_request_register: Handle incoming REGISTER request ---*/ -static int handle_request_register(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, struct sockaddr_in *sin, char *e) -{ - int res = 0; - char iabuf[INET_ADDRSTRLEN]; - - /* Use this as the basis */ - if (debug) - ast_verbose("Using latest request as basis request\n"); - copy_request(&p->initreq, req); - check_via(p, req); - if ((res = register_verify(p, sin, req, e, ignore)) < 0) - ast_log(LOG_NOTICE, "Registration from '%s' failed for '%s'\n", get_header(req, "To"), ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr)); - if (res < 1) { - /* Go ahead and free RTP port */ - if (p->rtp) { - if (p->owner) { - p->owner->fds[0] = -1; - p->owner->fds[1] = -1; - } - ast_rtp_destroy(p->rtp); - p->rtp = NULL; - } - if (p->vrtp) { - if (p->owner) { - p->owner->fds[2] = -1; - p->owner->fds[3] = -1; - } - ast_rtp_destroy(p->vrtp); - p->vrtp = NULL; - } - /* Destroy the session, but keep us around for just a bit in case they don't - get our 200 OK */ - sip_scheddestroy(p, 15*1000); - } - return res; -} -/*--- handle_request: Handle SIP requests (methods) ---*/ -/* this is where all incoming requests go first */ -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 */ - struct sip_request resp; - char *cmd; - char *cseq; - char *from; - char *useragent; - int seqno; - int len; - int ignore=0; - int respid; - int res = 0; - char iabuf[INET_ADDRSTRLEN]; - int debug = sip_debug_test_pvt(p); - char *e; - - /* Clear out potential response */ - memset(&resp, 0, sizeof(resp)); - - /* 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)) - return -1; - if (sscanf(cseq, "%d%n", &seqno, &len) != 1) { - ast_log(LOG_DEBUG, "No seqno in '%s'\n", cmd); - return -1; - } - /* Get the command XXX */ - - cmd = req->rlPart1; - e = req->rlPart2; - - /* Save useragent of the client */ - useragent = get_header(req, "User-Agent"); - ast_copy_string(p->useragent, useragent, sizeof(p->useragent)); - - /* Find out SIP method for incoming request */ - if (!strcasecmp(cmd, "SIP/2.0")) { /* Response to our request */ - p->method = SIP_RESPONSE; - /* Response to our request -- Do some sanity checks */ - if (!p->initreq.headers) { - 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, SIP_NEEDDESTROY); - return 0; - } else if (p->ocseq && (p->ocseq < seqno)) { - 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)) { - /* ignore means "don't do anything with it" but still have to - respond appropriately */ - ignore=1; - } - - extract_uri(p, req); - e = ast_skip_blanks(e); - if (sscanf(e, "%d %n", &respid, &len) != 1) { - ast_log(LOG_WARNING, "Invalid response: '%s'\n", e); - } else { - handle_response(p, respid, e + len, req,ignore, seqno); - } - return 0; - } - /* XXX what if not SIP/2.0 ? */ - /* New SIP request coming in - (could be new request in existing SIP dialog as well...) - */ - p->method = find_sip_method(cmd); /* Find out which SIP method they are using */ - if (option_debug > 2) - 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)) { - ast_log(LOG_DEBUG, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq); - return -1; - } else if (p->icseq && (p->icseq == seqno) && (strcasecmp(cmd, "CANCEL") || ast_test_flag(p, 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=1; - } - - 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)) { - from = get_header(req, "From"); - from = strcasestr(from, "tag="); - if (from) { - from += 4; - ast_copy_string(p->theirtag, from, sizeof(p->theirtag)); - from = strchr(p->theirtag, ';'); - if (from) - *from = '\0'; - } - } - snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd); - - /* Handle various incoming SIP methods in requests */ - switch (p->method) { - case SIP_OPTIONS: - res = handle_request_options(p, req, debug); - break; - case SIP_INVITE: - res = handle_request_invite(p, req, debug, ignore, seqno, sin, recount, e); - break; - case SIP_REFER: - res = handle_request_refer(p, req, debug, ignore, seqno, nounlock); - break; - case SIP_CANCEL: - res = handle_request_cancel(p, req, debug, ignore); - break; - case SIP_BYE: - res = handle_request_bye(p, req, debug); - break; - case SIP_MESSAGE: - res = handle_request_message(p, req, debug, ignore); - break; - case SIP_SUBSCRIBE: - res = handle_request_subscribe(p, req, debug, ignore, sin, seqno, e); - break; - case SIP_REGISTER: - res = handle_request_register(p, req, debug, ignore, sin, e); - break; - case SIP_INFO: - if (!ignore) { - if (debug) - ast_verbose("Receiving DTMF!\n"); - receive_info(p, req); - } else { /* if ignoring, transmit response */ - transmit_response(p, "200 OK", req); - } - break; - case SIP_NOTIFY: - /* XXX we get NOTIFY's from some servers. WHY?? Maybe we should - look into this someday XXX */ - transmit_response(p, "200 OK", req); - if (!p->lastinvite) - ast_set_flag(p, SIP_NEEDDESTROY); - break; - case SIP_ACK: - /* Make sure we don't ignore this */ - if (seqno == p->pendinginvite) { - p->pendinginvite = 0; - __sip_ack(p, seqno, FLAG_RESPONSE, -1); - if (!ast_strlen_zero(get_header(req, "Content-Type"))) { - if (process_sdp(p, req)) - return -1; - } - check_pendings(p); - } - if (!p->lastinvite && ast_strlen_zero(p->randdata)) - ast_set_flag(p, 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(iabuf, sizeof(iabuf), 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, SIP_NEEDDESTROY); - break; - } - return res; -} - -/*--- sipsock_read: Read data from SIP socket ---*/ -/* 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; - int nounlock; - int recount = 0; - int debug; - char iabuf[INET_ADDRSTRLEN]; - - len = sizeof(sin); - 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; - } - req.data[res] = '\0'; - req.len = res; - debug = sip_debug_test_addr(&sin); - if (pedanticsipchecking) - req.len = lws2sws(req.data, req.len); - if (debug) - ast_verbose("\n<-- SIP read from %s:%d: \n%s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), req.data); - parse(&req); - if (debug) { - ast_verbose("--- (%d headers %d lines)", req.headers, req.lines); - if (req.headers + req.lines == 0) - ast_verbose(" Nat keepalive "); - ast_verbose("---\n"); - } - - if (req.headers < 2) { - /* Must have at least two headers */ - return 1; - } - - /* Determine the request URI for sip, sips or tel URIs */ - if (determine_firstline_parts(&req) < 0) - return 1; - - /* Process request, with netlock held */ -retrylock: - ast_mutex_lock(&netlock); - p = find_call(&req, &sin, find_sip_method(req.rlPart1)); - if (p) { - /* Go ahead and lock the owner if it has one -- we may need it */ - if (p->owner && ast_mutex_trylock(&p->owner->lock)) { - ast_log(LOG_DEBUG, "Failed to grab lock, trying again...\n"); - ast_mutex_unlock(&p->lock); - ast_mutex_unlock(&netlock); - /* Sleep infintismly short amount of time */ - usleep(1); - goto retrylock; - } - memcpy(&p->recv, &sin, sizeof(p->recv)); - if (recordhistory) { - char tmp[80] = ""; - /* This is a response, note what it was for */ - snprintf(tmp, sizeof(tmp), "%s / %s", req.data, get_header(&req, "CSeq")); - append_history(p, "Rx", tmp); - } - nounlock = 0; - handle_request(p, &req, &sin, &recount, &nounlock); - if (p->owner && !nounlock) - ast_mutex_unlock(&p->owner->lock); - ast_mutex_unlock(&p->lock); - } - ast_mutex_unlock(&netlock); - if (recount) - ast_update_use_count(); - - return 1; -} - -/*--- sip_send_mwi_to_peer: Send message waiting indication ---*/ -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; - - /* Check for messages */ - ast_app_messagecount(peer->mailbox, &newmsgs, &oldmsgs); - - time(&peer->lastmsgcheck); - - /* Return now if it's the same thing we told them last time */ - if (((newmsgs << 8) | (oldmsgs)) == peer->lastmsgssent) { - return 0; - } - - p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY); - if (!p) { - ast_log(LOG_WARNING, "Unable to build sip pvt data for MWI\n"); - return -1; - } - peer->lastmsgssent = ((newmsgs << 8) | (oldmsgs)); - 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)) - memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); - build_via(p, p->via, sizeof(p->via)); - build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); - /* Send MWI */ - ast_set_flag(p, SIP_OUTGOING); - transmit_notify_with_mwi(p, newmsgs, oldmsgs); - sip_scheddestroy(p, 15000); - return 0; -} - -/*--- do_monitor: The SIP monitoring thread ---*/ -static void *do_monitor(void *data) -{ - int res; - struct sip_pvt *sip; - struct sip_peer *peer = NULL; - time_t t; - int fastrestart =0; - int lastpeernum = -1; - int curpeernum; - int reloading; - - /* Add an I/O event to our UDP socket */ - if (sipsock > -1) - ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL); - - /* This thread monitors all the frame relay interfaces which are not yet in use - (and thus do not have a separate thread) indefinitely */ - /* 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 = 0; - ast_mutex_unlock(&sip_reload_lock); - if (reloading) { - if (option_verbose > 0) - ast_verbose(VERBOSE_PREFIX_1 "Reloading SIP\n"); - sip_do_reload(); - } - /* Check for interfaces needing to be killed */ - ast_mutex_lock(&iflock); -restartsearch: - time(&t); - sip = iflist; - while(sip) { - ast_mutex_lock(&sip->lock); - if (sip->rtp && sip->owner && (sip->owner->_state == AST_STATE_UP) && !sip->redirip.sin_addr.s_addr) { - if (sip->lastrtptx && sip->rtpkeepalive && t > sip->lastrtptx + sip->rtpkeepalive) { - /* Need to send an empty RTP packet */ - time(&sip->lastrtptx); - ast_rtp_sendcng(sip->rtp, 0); - } - if (sip->lastrtprx && (sip->rtptimeout || sip->rtpholdtimeout) && t > sip->lastrtprx + sip->rtptimeout) { - /* 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 || - (sip->rtpholdtimeout && - (t > sip->lastrtprx + sip->rtpholdtimeout))) { - /* Needs a hangup */ - if (sip->rtptimeout) { - while(sip->owner && ast_mutex_trylock(&sip->owner->lock)) { - ast_mutex_unlock(&sip->lock); - usleep(1); - ast_mutex_lock(&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(sip->owner, AST_SOFTHANGUP_DEV); - ast_mutex_unlock(&sip->owner->lock); - } - } - } - } - } - if (ast_test_flag(sip, SIP_NEEDDESTROY) && !sip->packets && !sip->owner) { - ast_mutex_unlock(&sip->lock); - __sip_destroy(sip, 1); - goto restartsearch; - } - ast_mutex_unlock(&sip->lock); - sip = sip->next; - } - ast_mutex_unlock(&iflock); - /* Don't let anybody kill us right away. Nobody should lock the interface list - and wait for the monitor list, but the other way around is okay. */ - ast_mutex_lock(&monlock); - /* Lock the network interface */ - ast_mutex_lock(&netlock); - /* Okay, now that we know what to do, release the network lock */ - ast_mutex_unlock(&netlock); - /* And from now on, we're okay to be killed, so release the monitor lock as well */ - ast_mutex_unlock(&monlock); - 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); - ast_mutex_lock(&monlock); - if (res >= 0) - ast_sched_runq(sched); - - /* needs work to send mwi to realtime peers */ - time(&t); - fastrestart = 0; - curpeernum = 0; - peer = NULL; - ASTOBJ_CONTAINER_TRAVERSE(&peerl, !peer, do { - if ((curpeernum > lastpeernum) && !ast_strlen_zero(iterator->mailbox) && ((t - iterator->lastmsgcheck) > global_mwitime)) { - fastrestart = 1; - lastpeernum = curpeernum; - peer = ASTOBJ_REF(iterator); - }; - curpeernum++; - } while (0) - ); - 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; - -} - -/*--- restart_monitor: Start the channel monitor thread ---*/ -static int restart_monitor(void) -{ - pthread_attr_t attr; - /* If we're supposed to be stopped -- stay stopped */ - if (monitor_thread == AST_PTHREADT_STOP) - return 0; - if (ast_mutex_lock(&monlock)) { - ast_log(LOG_WARNING, "Unable to lock monitor\n"); - return -1; - } - 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 { - pthread_attr_init(&attr); - pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED); - /* Start a new monitor */ - if (ast_pthread_create(&monitor_thread, &attr, 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; -} - -/*--- sip_poke_noanswer: No answer to Qualify poke ---*/ -static int sip_poke_noanswer(void *data) -{ - struct sip_peer *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); - /* Try again quickly */ - peer->pokeexpire = ast_sched_add(sched, DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer); - return 0; -} - -/*--- sip_poke_peer: Check availability of peer, also keep NAT open ---*/ -/* This is done with the interval in qualify= option in sip.conf */ -/* Default is 2 seconds */ -static int sip_poke_peer(struct sip_peer *peer) -{ - struct sip_pvt *p; - 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 (peer->pokeexpire > -1) - ast_sched_del(sched, peer->pokeexpire); - peer->lastms = 0; - peer->pokeexpire = -1; - peer->call = NULL; - return 0; - } - if (peer->call > 0) { - ast_log(LOG_NOTICE, "Still have a call...\n"); - sip_destroy(peer->call); - } - p = peer->call = sip_alloc(NULL, NULL, 0, SIP_OPTIONS); - if (!peer->call) { - ast_log(LOG_WARNING, "Unable to allocate call for poking peer '%s'\n", peer->name); - return -1; - } - memcpy(&p->sa, &peer->addr, sizeof(p->sa)); - memcpy(&p->recv, &peer->addr, sizeof(p->sa)); - - /* Send options to peer's fullcontact */ - if (!ast_strlen_zero(peer->fullcontact)) { - ast_copy_string (p->fullcontact, peer->fullcontact, sizeof(p->fullcontact)); - } - - if (!ast_strlen_zero(peer->tohost)) - ast_copy_string(p->tohost, peer->tohost, sizeof(p->tohost)); - else - ast_inet_ntoa(p->tohost, sizeof(p->tohost), peer->addr.sin_addr); - - /* Recalculate our side, and recalculate Call ID */ - if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip)) - memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); - build_via(p, p->via, sizeof(p->via)); - build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); - - if (peer->pokeexpire > -1) - ast_sched_del(sched, peer->pokeexpire); - p->peerpoke = peer; - ast_set_flag(p, SIP_OUTGOING); -#ifdef VOCAL_DATA_HACK - ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username)); - transmit_invite(p, SIP_INVITE, 0, NULL, 1); -#else - transmit_invite(p, SIP_OPTIONS, 0, NULL, 1); -#endif - gettimeofday(&peer->ps, NULL); - peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2, sip_poke_noanswer, peer); - - return 0; -} - -/*--- sip_devicestate: Part of PBX channel interface ---*/ -static int sip_devicestate(void *data) -{ - char *ext, *host; - char tmp[256] = ""; - char *dest = data; - - struct hostent *hp; - struct ast_hostent ahp; - struct sip_peer *p; - int found = 0; - - int res = AST_DEVICE_INVALID; - - ast_copy_string(tmp, dest, sizeof(tmp)); - host = strchr(tmp, '@'); - if (host) { - *host = '\0'; - host++; - ext = tmp; - } else { - host = tmp; - ext = NULL; - } - - p = find_peer(host, NULL, 1); - if (p) { - found++; - res = AST_DEVICE_UNAVAILABLE; - if ((p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) && - (!p->maxms || ((p->lastms > -1) && (p->lastms <= p->maxms)))) { - /* peer found and valid */ - res = AST_DEVICE_UNKNOWN; - } - } - if (!p && !found) { - hp = ast_gethostbyname(host, &ahp); - if (hp) - res = AST_DEVICE_UNKNOWN; - } - - if (p) - ASTOBJ_UNREF(p,sip_destroy_peer); - return res; -} - -/*--- sip_request: PBX interface function -build SIP pvt structure ---*/ -/* SIP calls initiated by the PBX arrive here */ -static struct ast_channel *sip_request(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; - format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1); - if (!format) { - 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)); - return NULL; - } - p = sip_alloc(NULL, NULL, 0, SIP_INVITE); - if (!p) { - ast_log(LOG_WARNING, "Unable to build sip pvt data for '%s'\n", (char *)data); - return NULL; - } - - ast_copy_string(tmp, dest, sizeof(tmp)); - host = strchr(tmp, '@'); - if (host) { - *host = '\0'; - host++; - ext = tmp; - } else { - ext = strchr(tmp, '/'); - if (ext) { - *ext++ = '\0'; - host = tmp; - } - else { - host = tmp; - ext = NULL; - } - } - - /* Assign a default capability */ - p->capability = global_capability; - - if (create_addr(p, host)) { - *cause = AST_CAUSE_UNREGISTERED; - sip_destroy(p); - return NULL; - } - if (ast_strlen_zero(p->peername) && ext) - ast_copy_string(p->peername, ext, sizeof(p->peername)); - /* Recalculate our side, and recalculate Call ID */ - if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip)) - memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); - build_via(p, p->via, sizeof(p->via)); - build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); - - /* We have an extension to call, don't use the full contact here */ - /* This to enable dialling registred peers with extension dialling, - like SIP/peername/extension - SIP/peername will still use the full contact */ - if (ext) { - ast_copy_string(p->username, ext, sizeof(p->username)); - p->fullcontact[0] = 0; - } -#if 0 - printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "", host); -#endif - p->prefcodec = format; - 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; -} - -/*--- handle_common_options: Handle flag-type options common to users and peers ---*/ -static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v) -{ - int res = 0; - - if (!strcasecmp(v->name, "trustrpid")) { - ast_set_flag(mask, SIP_TRUSTRPID); - ast_set2_flag(flags, ast_true(v->value), SIP_TRUSTRPID); - res = 1; - } else if (!strcasecmp(v->name, "useclientcode")) { - ast_set_flag(mask, SIP_USECLIENTCODE); - ast_set2_flag(flags, ast_true(v->value), SIP_USECLIENTCODE); - res = 1; - } else if (!strcasecmp(v->name, "dtmfmode")) { - ast_set_flag(mask, SIP_DTMF); - ast_clear_flag(flags, SIP_DTMF); - if (!strcasecmp(v->value, "inband")) - ast_set_flag(flags, SIP_DTMF_INBAND); - else if (!strcasecmp(v->value, "rfc2833")) - ast_set_flag(flags, SIP_DTMF_RFC2833); - else if (!strcasecmp(v->value, "info")) - ast_set_flag(flags, SIP_DTMF_INFO); - else { - ast_log(LOG_WARNING, "Unknown dtmf mode '%s' on line %d, using rfc2833\n", v->value, v->lineno); - ast_set_flag(flags, SIP_DTMF_RFC2833); - } - } else if (!strcasecmp(v->name, "nat")) { - ast_set_flag(mask, SIP_NAT); - ast_clear_flag(flags, SIP_NAT); - if (!strcasecmp(v->value, "never")) - ast_set_flag(flags, SIP_NAT_NEVER); - else if (!strcasecmp(v->value, "route")) - ast_set_flag(flags, SIP_NAT_ROUTE); - else if (ast_true(v->value)) - ast_set_flag(flags, SIP_NAT_ALWAYS); - else - ast_set_flag(flags, SIP_NAT_RFC3581); - } else if (!strcasecmp(v->name, "canreinvite")) { - ast_set_flag(mask, SIP_REINVITE); - ast_clear_flag(flags, SIP_REINVITE); - if (!strcasecmp(v->value, "update")) - ast_set_flag(flags, SIP_REINVITE_UPDATE | SIP_CAN_REINVITE); - else - ast_set2_flag(flags, ast_true(v->value), SIP_CAN_REINVITE); - } else if (!strcasecmp(v->name, "insecure")) { - ast_set_flag(mask, SIP_INSECURE_PORT | SIP_INSECURE_INVITE); - ast_clear_flag(flags, SIP_INSECURE_PORT | SIP_INSECURE_INVITE); - if (!strcasecmp(v->value, "very")) - ast_set_flag(flags, SIP_INSECURE_PORT | SIP_INSECURE_INVITE); - else if (ast_true(v->value)) - ast_set_flag(flags, SIP_INSECURE_PORT); - else if (!ast_false(v->value)) { - char buf[64]; - char *word, *next; - - ast_copy_string(buf, v->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", v->value, v->lineno); - } - } - } else if (!strcasecmp(v->name, "progressinband")) { - ast_set_flag(mask, SIP_PROG_INBAND); - ast_clear_flag(flags, SIP_PROG_INBAND); - if (strcasecmp(v->value, "never")) - ast_set_flag(flags, SIP_PROG_INBAND_NO); - else if (ast_true(v->value)) - ast_set_flag(flags, SIP_PROG_INBAND_YES); - } else if (!strcasecmp(v->name, "allowguest")) { -#ifdef OSP_SUPPORT - if (!strcasecmp(v->value, "osp")) - global_allowguest = 2; - else -#endif - if (ast_true(v->value)) - global_allowguest = 1; - else - global_allowguest = 0; -#ifdef OSP_SUPPORT - } else if (!strcasecmp(v->name, "ospauth")) { - ast_set_flag(mask, SIP_OSPAUTH); - ast_clear_flag(flags, SIP_OSPAUTH); - if (!strcasecmp(v->value, "exclusive")) - ast_set_flag(flags, SIP_OSPAUTH_EXCLUSIVE); - else - ast_set2_flag(flags, ast_true(v->value), SIP_OSPAUTH_YES); -#endif - } else if (!strcasecmp(v->name, "promiscredir")) { - ast_set_flag(mask, SIP_PROMISCREDIR); - ast_set2_flag(flags, ast_true(v->value), SIP_PROMISCREDIR); - res = 1; - } - - return res; -} - -/*--- add_realm_authentication: 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 *auth; - struct sip_auth *b = NULL, *a = authlist; - - if (!configuration || ast_strlen_zero(configuration)) - return authlist; - - 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'; - realm++; - } - if (!username || ast_strlen_zero(username) || !realm || 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,"#"); - } - } - auth = malloc(sizeof(struct sip_auth)); - if (auth) { - memset(auth, 0, sizeof(struct sip_auth)); - 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)); - } else { - ast_log(LOG_ERROR, "Allocation of auth structure failed, Out of memory\n"); - return authlist; - } - - /* Add authentication to authl */ - if (!authlist) { /* No existing list */ - return auth; - } - while(a) { - b = a; - a = a->next; - } - b->next = auth; /* Add structure add end of list */ - - if (option_verbose > 2) - ast_verbose("Added authentication for realm %s\n", realm); - - return authlist; - -} - -/*--- clear_realm_authentication: 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; -} - -/*--- find_realm_authentication: Find authentication for a specific realm ---*/ -static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, char *realm) -{ - struct sip_auth *a = authlist; /* First entry in auth list */ - - while (a) { - if (!strcasecmp(a->realm, realm)){ - break; - } - a = a->next; - } - - return a; -} - -/*--- build_user: Initiate a SIP user structure from sip.conf ---*/ -static struct sip_user *build_user(const char *name, struct ast_variable *v, 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 = {(0)}; - struct ast_flags mask = {(0)}; - - - user = (struct sip_user *)malloc(sizeof(struct sip_user)); - if (!user) { - return NULL; - } - memset(user, 0, sizeof(struct sip_user)); - suserobjs++; - ASTOBJ_INIT(user); - ast_copy_string(user->name, name, sizeof(user->name)); - oldha = user->ha; - user->ha = NULL; - /* set the usage flag to a sane staring value*/ - user->inUse = 0; - user->outUse = 0; - ast_copy_flags(user, &global_flags, - SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_USECLIENTCODE | SIP_DTMF | SIP_NAT | - SIP_REINVITE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE | SIP_PROG_INBAND | SIP_OSPAUTH); - user->capability = global_capability; - user->prefs = prefs; - /* set default context */ - strcpy(user->context, default_context); - strcpy(user->language, default_language); - strcpy(user->musicclass, global_musicclass); - while(v) { - if (handle_common_options(&userflags, &mask, v)) { - v = v->next; - continue; - } - - if (!strcasecmp(v->name, "context")) { - ast_copy_string(user->context, v->value, sizeof(user->context)); - } else if (!strcasecmp(v->name, "setvar")) { - varname = ast_strdupa(v->value); - if (varname && (varval = strchr(varname,'='))) { - *varval = '\0'; - varval++; - 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, "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, "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, "musiconhold")) { - ast_copy_string(user->musicclass, v->value, sizeof(user->musicclass)); - } else if (!strcasecmp(v->name, "accountcode")) { - ast_copy_string(user->accountcode, v->value, sizeof(user->accountcode)); - } else if (!strcasecmp(v->name, "incominglimit")) { - user->incominglimit = atoi(v->value); - if (user->incominglimit < 0) - user->incominglimit = 0; - } else if (!strcasecmp(v->name, "outgoinglimit")) { - user->outgoinglimit = atoi(v->value); - if (user->outgoinglimit < 0) - user->outgoinglimit = 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, "callingpres")) { - user->callingpres = ast_parse_caller_presentation(v->value); - if (user->callingpres == -1) - user->callingpres = atoi(v->value); - } - /*else if (strcasecmp(v->name,"type")) - * ast_log(LOG_WARNING, "Ignoring %s\n", v->name); - */ - v = v->next; - } - ast_copy_flags(user, &userflags, mask.flags); - ast_free_ha(oldha); - return user; -} - -/*--- temp_peer: Create temporary peer (used in autocreatepeer mode) ---*/ -static struct sip_peer *temp_peer(const char *name) -{ - struct sip_peer *peer; - - peer = malloc(sizeof(*peer)); - if (!peer) - return NULL; - - memset(peer, 0, sizeof(*peer)); - apeerobjs++; - ASTOBJ_INIT(peer); - - peer->expire = -1; - peer->pokeexpire = -1; - ast_copy_string(peer->name, name, sizeof(peer->name)); - ast_copy_flags(peer, &global_flags, - SIP_PROMISCREDIR | SIP_USEREQPHONE | SIP_TRUSTRPID | SIP_USECLIENTCODE | - SIP_DTMF | SIP_NAT | SIP_REINVITE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE | - SIP_PROG_INBAND | SIP_OSPAUTH); - strcpy(peer->context, default_context); - strcpy(peer->language, default_language); - strcpy(peer->musicclass, global_musicclass); - peer->addr.sin_port = htons(DEFAULT_SIP_PORT); - peer->addr.sin_family = AF_INET; - peer->expiry = expiry; - peer->capability = global_capability; - peer->rtptimeout = global_rtptimeout; - peer->rtpholdtimeout = global_rtpholdtimeout; - peer->rtpkeepalive = global_rtpkeepalive; - ast_set_flag(peer, SIP_SELFDESTRUCT); - ast_set_flag(peer, SIP_DYNAMIC); - peer->prefs = prefs; - reg_source_db(peer); - - return peer; -} - -/*--- build_peer: Build peer from config file ---*/ -static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int realtime) -{ - struct sip_peer *peer = NULL; - struct ast_ha *oldha = NULL; - int maskfound=0; - int obproxyfound=0; - int found=0; - int format=0; /* Ama flags */ - time_t regseconds; - char *varname = NULL, *varval = NULL; - struct ast_variable *tmpvar = NULL; - struct ast_flags peerflags = {(0)}; - struct ast_flags mask = {(0)}; - - - if (!realtime) - /* Note we do NOT use find_peer here, to avoid realtime recursion */ - peer = ASTOBJ_CONTAINER_FIND_UNLINK(&peerl, name); - - if (peer) { - /* Already in the list, remove it and it will be added back (or FREE'd) */ - found++; - } else { - peer = malloc(sizeof(struct sip_peer)); - if (peer) { - memset(peer, 0, sizeof(struct sip_peer)); - if (realtime) - rpeerobjs++; - else - speerobjs++; - ASTOBJ_INIT(peer); - peer->expire = -1; - peer->pokeexpire = -1; - } else { - ast_log(LOG_WARNING, "Can't allocate SIP peer memory\n"); - } - } - /* Note that our peer HAS had its reference count incrased */ - if (!peer) - return NULL; - - peer->lastmsgssent = -1; - if (!found) { - if (name) - ast_copy_string(peer->name, name, sizeof(peer->name)); - peer->addr.sin_port = htons(DEFAULT_SIP_PORT); - peer->addr.sin_family = AF_INET; - peer->defaddr.sin_family = AF_INET; - peer->expiry = expiry; - } - /* If we have channel variables, remove them (reload) */ - if (peer->chanvars) { - ast_variables_destroy(peer->chanvars); - peer->chanvars = NULL; - } - strcpy(peer->context, default_context); - strcpy(peer->language, default_language); - strcpy(peer->musicclass, global_musicclass); - ast_copy_flags(peer, &global_flags, SIP_USEREQPHONE); - 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->rtpkeepalive = global_rtpkeepalive; - peer->maxms = default_qualify; - peer->prefs = prefs; - oldha = peer->ha; - peer->ha = NULL; - peer->addr.sin_family = AF_INET; - ast_copy_flags(peer, &global_flags, - SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_USECLIENTCODE | - SIP_DTMF | SIP_REINVITE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE | - SIP_PROG_INBAND | SIP_OSPAUTH); - peer->capability = global_capability; - peer->rtptimeout = global_rtptimeout; - peer->rtpholdtimeout = global_rtpholdtimeout; - while(v) { - if (handle_common_options(&peerflags, &mask, v)) { - v = v->next; - continue; - } - - if (realtime && !strcasecmp(v->name, "regseconds")) { - if (sscanf(v->value, "%li", ®seconds) != 1) - regseconds = 0; - } 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 (!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, "context")) { - ast_copy_string(peer->context, v->value, sizeof(peer->context)); - } 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, 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 */ - ast_set_flag(peer, SIP_DYNAMIC); - if (!found) { - /* Initialize stuff iff we're not found, otherwise - we keep going with what we had */ - 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; - } - } - } - } else { - /* Non-dynamic. Make sure we become that way if we're not */ - if (peer->expire > -1) - ast_sched_del(sched, peer->expire); - peer->expire = -1; - ast_clear_flag(peer, SIP_DYNAMIC); - if (!obproxyfound || !strcasecmp(v->name, "outboundproxy")) { - if (ast_get_ip_or_srv(&peer->addr, v->value, "_sip._udp")) { - 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 (!maskfound) - inet_aton("255.255.255.255", &peer->mask); - } 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, "mask")) { - maskfound++; - inet_aton(v->value, &peer->mask); - } else if (!strcasecmp(v->name, "port")) { - if (!realtime && ast_test_flag(peer, SIP_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, "incominglimit")) { - peer->incominglimit = atoi(v->value); - if (peer->incominglimit < 0) - peer->incominglimit = 0; - } else if (!strcasecmp(v->name, "outgoinglimit")) { - peer->outgoinglimit = atoi(v->value); - if (peer->outgoinglimit < 0) - peer->outgoinglimit = 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, "musiconhold")) { - ast_copy_string(peer->musicclass, v->value, sizeof(peer->musicclass)); - } else if (!strcasecmp(v->name, "mailbox")) { - ast_copy_string(peer->mailbox, v->value, sizeof(peer->mailbox)); - } else if (!strcasecmp(v->name, "callgroup")) { - peer->callgroup = ast_get_group(v->value); - } 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, "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 (varname && (varval = strchr(varname,'='))) { - *varval = '\0'; - varval++; - 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_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; - } - } - /* else if (strcasecmp(v->name,"type")) - * ast_log(LOG_WARNING, "Ignoring %s\n", v->name); - */ - v=v->next; - } - if (realtime && !ast_test_flag((&global_flags_page2), SIP_PAGE2_RTIGNOREREGEXPIRE) && ast_test_flag(peer, SIP_DYNAMIC)) { - time_t nowtime; - - time(&nowtime); - if ((nowtime - regseconds) > 0) { - memset(&peer->addr, 0, sizeof(peer->addr)); - if (option_debug) - ast_log(LOG_DEBUG, "Bah, we're expired (%ld/%ld/%ld)!\n", nowtime - regseconds, regseconds, nowtime); - } - } - ast_copy_flags(peer, &peerflags, mask.flags); - if (!found && ast_test_flag(peer, SIP_DYNAMIC) && !ast_test_flag(peer, SIP_REALTIME)) - reg_source_db(peer); - ASTOBJ_UNMARK(peer); - ast_free_ha(oldha); - return peer; -} - -/*--- reload_config: Re-read SIP.conf config file ---*/ -/* 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(void) -{ - struct ast_config *cfg; - struct ast_variable *v; - struct sip_peer *peer; - struct sip_user *user; - struct ast_hostent ahp; - char *cat; - char *utype; - struct hostent *hp; - int format; - int oldport = ntohs(bindaddr.sin_port); - char iabuf[INET_ADDRSTRLEN]; - struct ast_flags dummy; - - 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; - } - - /* Reset IP addresses */ - memset(&bindaddr, 0, sizeof(bindaddr)); - memset(&localaddr, 0, sizeof(localaddr)); - memset(&externip, 0, sizeof(externip)); - memset(&prefs, 0 , sizeof(prefs)); - - /* Initialize some reasonable defaults at SIP reload */ - ast_copy_string(default_context, DEFAULT_CONTEXT, sizeof(default_context)); - default_language[0] = '\0'; - default_fromdomain[0] = '\0'; - default_qualify = 0; - externhost[0] = '\0'; - externexpire = 0; - externrefresh = 10; - sipdebug = 0; - ast_copy_string(default_useragent, DEFAULT_USERAGENT, sizeof(default_useragent)); - ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime)); - ast_copy_string(global_realm, DEFAULT_REALM, sizeof(global_realm)); - ast_copy_string(global_musicclass, "default", sizeof(global_musicclass)); - ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid)); - memset(&outboundproxyip, 0, sizeof(outboundproxyip)); - outboundproxyip.sin_port = htons(DEFAULT_SIP_PORT); - outboundproxyip.sin_family = AF_INET; /* Type of address: IPv4 */ - videosupport = 0; - compactheaders = 0; - relaxdtmf = 0; - callevents = 0; - ourport = DEFAULT_SIP_PORT; - global_rtptimeout = 0; - global_rtpholdtimeout = 0; - global_rtpkeepalive = 0; - pedanticsipchecking = 0; - global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT; - global_regattempts_max = DEFAULT_REGATTEMPTS_MAX; - ast_clear_flag(&global_flags, AST_FLAGS_ALL); - ast_set_flag(&global_flags, SIP_DTMF_RFC2833); - ast_set_flag(&global_flags, SIP_NAT_RFC3581); - ast_set_flag(&global_flags, SIP_CAN_REINVITE); - global_mwitime = DEFAULT_MWITIME; - srvlookup = 0; - autocreatepeer = 0; - regcontext[0] = '\0'; - tos = 0; - expiry = DEFAULT_EXPIRY; - global_allowguest = 1; - - /* Read the [general] config section of sip.conf (or from realtime config) */ - v = ast_variable_browse(cfg, "general"); - while(v) { - if (handle_common_options(&global_flags, &dummy, v)) { - v = v->next; - 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, "realm")) { - ast_copy_string(global_realm, v->value, sizeof(global_realm)); - } else if (!strcasecmp(v->name, "useragent")) { - ast_copy_string(default_useragent, v->value, sizeof(default_useragent)); - ast_log(LOG_DEBUG, "Setting User Agent Name to %s\n", - default_useragent); - } else if (!strcasecmp(v->name, "rtcachefriends")) { - ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_RTCACHEFRIENDS); - } else if (!strcasecmp(v->name, "rtnoupdate")) { - ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_RTNOUPDATE); - } else if (!strcasecmp(v->name, "rtignoreregexpire")) { - ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_RTIGNOREREGEXPIRE); - } 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_page2), i || ast_true(v->value), SIP_PAGE2_RTAUTOCLEAR); - } else if (!strcasecmp(v->name, "usereqphone")) { - ast_set2_flag((&global_flags), ast_true(v->value), SIP_USEREQPHONE); - } else if (!strcasecmp(v->name, "relaxdtmf")) { - 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, "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, "videosupport")) { - videosupport = ast_true(v->value); - } 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 (!strcasecmp(v->name, "musicclass") || !strcasecmp(v->name, "musiconhold")) { - ast_copy_string(global_musicclass, v->value, sizeof(global_musicclass)); - } 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(regcontext, v->value, sizeof(regcontext)); - /* Create context if it doesn't exist already */ - if (!ast_context_find(regcontext)) - ast_context_create(NULL, regcontext, channeltype); - } 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, "_sip._udp") < 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, "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")){ - sipdebug = 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)); - time(&externexpire); - } 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(&prefs, &global_capability, v->value, 1); - } else if (!strcasecmp(v->name, "disallow")) { - ast_parse_allow_disallow(&prefs, &global_capability, v->value, 0); - } else if (!strcasecmp(v->name, "register")) { - sip_register(v->value, v->lineno); - } else if (!strcasecmp(v->name, "recordhistory")) { - recordhistory = ast_true(v->value); - } else if (!strcasecmp(v->name, "tos")) { - if (sscanf(v->value, "%d", &format) == 1) - tos = format & 0xff; - else if (!strcasecmp(v->value, "lowdelay")) - tos = IPTOS_LOWDELAY; - else if (!strcasecmp(v->value, "throughput")) - tos = IPTOS_THROUGHPUT; - else if (!strcasecmp(v->value, "reliability")) - tos = IPTOS_RELIABILITY; - else if (!strcasecmp(v->value, "mincost")) - tos = IPTOS_MINCOST; - else if (!strcasecmp(v->value, "none")) - tos = 0; - else - ast_log(LOG_WARNING, "Invalid tos value at line %d, should be 'lowdelay', 'throughput', 'reliability', 'mincost', or 'none'\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")) { - callevents = ast_true(v->value); - } - /* else if (strcasecmp(v->name,"type")) - * ast_log(LOG_WARNING, "Ignoring %s\n", v->name); - */ - v = v->next; - } - - /* Build list of authentication to various SIP realms, i.e. service providers */ - v = ast_variable_browse(cfg, "authentication"); - while(v) { - /* Format for authentication is auth = username:password@realm */ - if (!strcasecmp(v->name, "auth")) { - authl = add_realm_authentication(authl, v->value, v->lineno); - } - v = v->next; - } - - /* Load peers, users and friends */ - cat = ast_category_browse(cfg, NULL); - while(cat) { - if (strcasecmp(cat, "general") && strcasecmp(cat, "authentication")) { - utype = ast_variable_retrieve(cfg, cat, "type"); - if (utype) { - if (!strcasecmp(utype, "user") || !strcasecmp(utype, "friend")) { - user = build_user(cat, ast_variable_browse(cfg, cat), 0); - if (user) { - ASTOBJ_CONTAINER_LINK(&userl,user); - ASTOBJ_UNREF(user, sip_destroy_user); - } - } - if (!strcasecmp(utype, "peer") || !strcasecmp(utype, "friend")) { - peer = build_peer(cat, ast_variable_browse(cfg, cat), 0); - if (peer) { - ASTOBJ_CONTAINER_LINK(&peerl,peer); - ASTOBJ_UNREF(peer, sip_destroy_peer); - } - } else if (strcasecmp(utype, "user")) { - ast_log(LOG_WARNING, "Unknown type '%s' for '%s' in %s\n", utype, cat, "sip.conf"); - } - } else - ast_log(LOG_WARNING, "Section '%s' lacks type\n", cat); - } - cat = ast_category_browse(cfg, cat); - } - if (ast_find_ourip(&__ourip, bindaddr)) { - ast_log(LOG_WARNING, "Unable to get own IP address, SIP disabled\n"); - return 0; - } - if (!ntohs(bindaddr.sin_port)) - bindaddr.sin_port = ntohs(DEFAULT_SIP_PORT); - bindaddr.sin_family = AF_INET; - ast_mutex_lock(&netlock); - if ((sipsock > -1) && (ntohs(bindaddr.sin_port) != oldport)) { - 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)); - } 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); - - if (bind(sipsock, (struct sockaddr *)&bindaddr, sizeof(bindaddr)) < 0) { - ast_log(LOG_WARNING, "Failed to bind to %s:%d: %s\n", - ast_inet_ntoa(iabuf, sizeof(iabuf), 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(iabuf, sizeof(iabuf), bindaddr.sin_addr), ntohs(bindaddr.sin_port)); - ast_verbose(VERBOSE_PREFIX_2 "Using TOS bits %d\n", tos); - } - if (setsockopt(sipsock, IPPROTO_IP, IP_TOS, &tos, sizeof(tos))) - ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos); - } - } - } - ast_mutex_unlock(&netlock); - - /* 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); - - return 0; -} - -/*--- sip_get_rtp_peer: Returns null if we can't reinvite (part of RTP interface) */ -static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan) -{ - struct sip_pvt *p; - struct ast_rtp *rtp = NULL; - p = chan->tech_pvt; - if (!p) - return NULL; - ast_mutex_lock(&p->lock); - if (p->rtp && ast_test_flag(p, SIP_CAN_REINVITE)) - rtp = p->rtp; - ast_mutex_unlock(&p->lock); - return rtp; -} - -/*--- sip_get_vrtp_peer: Returns null if we can't reinvite video (part of RTP interface) */ -static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan) -{ - struct sip_pvt *p; - struct ast_rtp *rtp = NULL; - p = chan->tech_pvt; - if (!p) - return NULL; - - ast_mutex_lock(&p->lock); - if (p->vrtp && ast_test_flag(p, SIP_CAN_REINVITE)) - rtp = p->vrtp; - ast_mutex_unlock(&p->lock); - return rtp; -} - -/*--- sip_set_rtp_peer: Set the data needed to RE-INVITE this call - so that the peers media go between them, outside of Asterisk. ---*/ -static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs) -{ - struct sip_pvt *p; - p = chan->tech_pvt; - if (!p) - return -1; - - ast_mutex_lock(&p->lock); - if (rtp) - ast_rtp_get_peer(rtp, &p->redirip); - else - memset(&p->redirip, 0, sizeof(p->redirip)); - if (vrtp) - ast_rtp_get_peer(vrtp, &p->vredirip); - else - memset(&p->vredirip, 0, sizeof(p->vredirip)); - p->redircodecs = codecs; - if (!ast_test_flag(p, SIP_GOTREFER)) { - if (!p->pendinginvite) - transmit_reinvite_with_sdp(p); - else if (!ast_test_flag(p, SIP_PENDINGBYE)) { - ast_log(LOG_DEBUG, "Deferring reinvite on '%s'\n", p->callid); - ast_set_flag(p, SIP_NEEDREINVITE); - } - } - /* Reset lastrtprx timer */ - time(&p->lastrtprx); - time(&p->lastrtptx); - 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-Accuntcode:\". Use this with care.\n" -"Adding the wrong headers may jeopardize the SIP dialog.\n" -"Always returns 0\n"; - -static char *app_sipgetheader = "SIPGetHeader"; -static char *synopsis_sipgetheader = "Get a SIP header from an incoming call"; - -static char *descrip_sipgetheader = "" -" SIPGetHeader(var=headername): \n" -"Sets a channel variable to the content of a SIP header\n" -"Skips to priority+101 if header does not exist\n" -"Otherwise returns 0\n"; - -/*--- sip_dtmfmode: change the DTMFmode for a 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_mutex_lock(&chan->lock); - if (chan->type != channeltype) { - ast_log(LOG_WARNING, "Call this application only on SIP incoming calls\n"); - ast_mutex_unlock(&chan->lock); - return 0; - } - p = chan->tech_pvt; - if (!p) { - ast_mutex_unlock(&chan->lock); - return 0; - } - ast_mutex_lock(&p->lock); - if (!strcasecmp(mode,"info")) { - ast_clear_flag(p, SIP_DTMF); - ast_set_flag(p, SIP_DTMF_INFO); - } else if (!strcasecmp(mode,"rfc2833")) { - ast_clear_flag(p, SIP_DTMF); - ast_set_flag(p, SIP_DTMF_RFC2833); - } else if (!strcasecmp(mode,"inband")) { - ast_clear_flag(p, SIP_DTMF); - ast_set_flag(p, SIP_DTMF_INBAND); - } else - ast_log(LOG_WARNING, "I don't know about this dtmf mode: %s\n",mode); - if (ast_test_flag(p, 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_mutex_unlock(&chan->lock); - return 0; -} - -/*--- sip_addheader: Add a SIP header ---*/ -static int sip_addheader(struct ast_channel *chan, void *data) -{ - int arglen; - int no = 0; - int ok = 0; - char *content = (char *) NULL; - char varbuf[128]; - - arglen = strlen(data); - if (!arglen) { - ast_log(LOG_WARNING, "This application requires the argument: Header\n"); - return 0; - } - ast_mutex_lock(&chan->lock); - if (chan->type != channeltype) { - ast_log(LOG_WARNING, "Call this application only on incoming SIP calls\n"); - ast_mutex_unlock(&chan->lock); - return 0; - } - - /* Check for headers */ - while (!ok && no <= 50) { - no++; - snprintf(varbuf, sizeof(varbuf), "_SIPADDHEADER%.2d", no); - content = pbx_builtin_getvar_helper(chan, varbuf); - - if (!content) - ok = 1; - } - if (ok) { - pbx_builtin_setvar_helper (chan, varbuf, data); - if (sipdebug) - ast_log(LOG_DEBUG,"SIP Header added \"%s\" as %s\n", (char *) data, varbuf); - } else { - ast_log(LOG_WARNING, "Too many SIP headers added, max 50\n"); - } - ast_mutex_unlock(&chan->lock); - return 0; -} - -/*--- sip_getheader: Get a SIP header (dialplan app) ---*/ -static int sip_getheader(struct ast_channel *chan, void *data) -{ - static int dep_warning = 0; - struct sip_pvt *p; - char *argv, *varname = NULL, *header = NULL, *content; - - if (!dep_warning) { - ast_log(LOG_WARNING, "SIPGetHeader is deprecated, use the SIP_HEADER function instead.\n"); - dep_warning = 1; - } - - argv = ast_strdupa(data); - if (!argv) { - ast_log(LOG_DEBUG, "Memory allocation failed\n"); - return 0; - } - - if (strchr (argv, '=') ) { /* Pick out argumenet */ - varname = strsep (&argv, "="); - header = strsep (&argv, "\0"); - } - - if (!varname || !header) { - ast_log(LOG_DEBUG, "SipGetHeader: Ignoring command, Syntax error in argument\n"); - return 0; - } - - ast_mutex_lock(&chan->lock); - if (chan->type != channeltype) { - ast_log(LOG_WARNING, "Call this application only on incoming SIP calls\n"); - ast_mutex_unlock(&chan->lock); - return 0; - } - - p = chan->tech_pvt; - content = get_header(&p->initreq, header); /* Get the header */ - if (!ast_strlen_zero(content)) { - pbx_builtin_setvar_helper(chan, varname, content); - } else { - ast_log(LOG_WARNING,"SIP Header %s not found for channel variable %s\n", header, varname); - ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101); - } - - ast_mutex_unlock(&chan->lock); - return 0; -} - -/*--- sip_sipredirect: Transfer call before connect with a 302 redirect ---*/ -/* Called by the transfer() dialplan application through the sip_transfer() */ -/* pbx interface function if the call is in ringing state */ -/* coded by Martin Pycko (m78pl@yahoo.com) */ -static int sip_sipredirect(struct sip_pvt *p, const char *dest) -{ - char *cdest; - char *extension, *host, *port; - char tmp[80]; - - cdest = ast_strdupa(dest); - if (!cdest) { - ast_log(LOG_ERROR, "Problem allocating the memory\n"); - return 0; - } - extension = strsep(&cdest, "@"); - host = strsep(&cdest, ":"); - port = strsep(&cdest, ":"); - if (!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 (!strlen(tmp)) { - ast_log(LOG_ERROR, "Cannot retrieve the 'To' header from the original SIP request!\n"); - return 0; - } - if ((localtmp = strstr(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 (!strlen(lhost)) { - ast_log(LOG_ERROR, "Can't find the host address\n"); - return 0; - } - host = ast_strdupa(lhost); - if (!host) { - ast_log(LOG_ERROR, "Problem allocating the memory\n"); - return 0; - } - if (!ast_strlen_zero(lport)) { - port = ast_strdupa(lport); - if (!port) { - ast_log(LOG_ERROR, "Problem allocating the memory\n"); - return 0; - } - } - } - } - - /* make sure the forwarding won't be forever */ - ast_copy_string(tmp, get_header(&p->initreq, "Max-Forwards"), sizeof(tmp)); - if (strlen(tmp) && atoi(tmp)) { - /* we found Max-Forwards in the original SIP request */ - p->maxforwards = atoi(tmp) - 1; - } else { - /* just send our 302 Moved Temporarily */ - p->maxforwards = DEFAULT_MAX_FORWARDS - 1; - } - if (p->maxforwards > -1) { - snprintf(p->our_contact, sizeof(p->our_contact), "Transfer ", extension, host, port ? ":" : "", port ? port : ""); - transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq, 1); - } else { - transmit_response(p, "483 Too Many Hops", &p->initreq); - } - /* this is all that we want to send to that SIP device */ - ast_set_flag(p, SIP_ALREADYGONE); - - /* hangup here */ - return -1; -} - -/*--- sip_get_codec: 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->peercapability; -} - -/*--- sip_rtp: Interface structure with callbacks used to connect to rtp module --*/ -static struct ast_rtp_protocol sip_rtp = { - type: channeltype, - 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, -}; - -/*--- sip_poke_all_peers: Send a poke to all known peers */ -static void sip_poke_all_peers(void) -{ - ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do { - ASTOBJ_WRLOCK(iterator); - sip_poke_peer(iterator); - ASTOBJ_UNLOCK(iterator); - } while (0) - ); -} - -/*--- sip_send_all_registers: Send all known registrations */ -static void sip_send_all_registers(void) -{ - int ms; - - ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do { - ASTOBJ_WRLOCK(iterator); - if (iterator->expire > -1) - ast_sched_del(sched, iterator->expire); - ms = (rand() >> 12) & 0x1fff; - iterator->expire = ast_sched_add(sched, ms, sip_reregister, iterator); - ASTOBJ_UNLOCK(iterator); - } while (0) - ); -} - -/*--- sip_do_reload: Reload module */ -static int sip_do_reload(void) -{ - clear_realm_authentication(authl); - authl = NULL; - - ASTOBJ_CONTAINER_DESTROYALL(&userl, sip_destroy_user); - ASTOBJ_CONTAINER_DESTROYALL(®l, sip_registry_destroy); - ASTOBJ_CONTAINER_MARKALL(&peerl); - reload_config(); - /* Prune peers who still are supposed to be deleted */ - ASTOBJ_CONTAINER_PRUNE_MARKED(&peerl, sip_destroy_peer); - - sip_poke_all_peers(); - sip_send_all_registers(); - - return 0; -} - -/*--- sip_reload: 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 = 1; - ast_mutex_unlock(&sip_reload_lock); - restart_monitor(); - - return 0; -} - -/*--- reload: Part of Asterisk module interface ---*/ -int reload(void) -{ - return sip_reload(0, 0, NULL); -} - -// static struct ast_cli_entry cli_sip_reload = -static struct ast_cli_entry my_clis[] = { - { { "sip", "notify", NULL }, sip_notify, "Send a notify packet to a SIP peer", notify_usage, complete_sipnotify }, - { { "sip", "show", "objects", NULL }, sip_show_objects, "Show all SIP object allocations", show_objects_usage }, - { { "sip", "show", "users", NULL }, sip_show_users, "Show defined SIP users", show_users_usage }, - { { "sip", "show", "user", NULL }, sip_show_user, "Show details on specific SIP user", show_user_usage, complete_sip_show_user }, - { { "sip", "show", "subscriptions", NULL }, sip_show_subscriptions, "Show active SIP subscriptions", show_subscriptions_usage}, - { { "sip", "show", "channels", NULL }, sip_show_channels, "Show active SIP channels", show_channels_usage}, - { { "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", "debug", NULL }, sip_do_debug, "Enable SIP debugging", debug_usage }, - { { "sip", "debug", "ip", NULL }, sip_do_debug, "Enable SIP debugging on IP", debug_usage }, - { { "sip", "debug", "peer", NULL }, sip_do_debug, "Enable SIP debugging on Peername", debug_usage, complete_sip_debug_peer }, - { { "sip", "show", "peer", NULL }, sip_show_peer, "Show details on specific SIP peer", show_peer_usage, complete_sip_show_peer }, - { { "sip", "show", "peers", NULL }, sip_show_peers, "Show defined SIP peers", show_peers_usage }, - { { "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", "show", "inuse", NULL }, sip_show_inuse, "List all inuse/limits", show_inuse_usage }, - { { "sip", "show", "registry", NULL }, sip_show_registry, "Show SIP registration status", show_reg_usage }, - { { "sip", "history", NULL }, sip_do_history, "Enable SIP history", history_usage }, - { { "sip", "no", "history", NULL }, sip_no_history, "Disable SIP history", no_history_usage }, - { { "sip", "no", "debug", NULL }, sip_no_debug, "Disable SIP debugging", no_debug_usage }, - { { "sip", "reload", NULL }, sip_reload, "Reload SIP configuration", sip_reload_usage }, -}; - -/*--- load_module: PBX load module - initialization ---*/ -int load_module() -{ - ASTOBJ_CONTAINER_INIT(&userl); - ASTOBJ_CONTAINER_INIT(&peerl); - ASTOBJ_CONTAINER_INIT(®l); - sched = sched_context_create(); - if (!sched) { - ast_log(LOG_WARNING, "Unable to create schedule context\n"); - } - io = io_context_create(); - if (!io) { - ast_log(LOG_WARNING, "Unable to create I/O context\n"); - } - /* Make sure we can register our sip channel type */ - if (ast_channel_register(&sip_tech)) { - ast_log(LOG_ERROR, "Unable to register channel type %s\n", channeltype); - return -1; - } - - if (reload_config()) - return -1; - - ast_cli_register_multiple(my_clis, sizeof(my_clis)/ sizeof(my_clis[0])); - - ast_rtp_proto_register(&sip_rtp); - - ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode); - ast_register_application(app_sipaddheader, sip_addheader, synopsis_sipaddheader, descrip_sipaddheader); - ast_register_application(app_sipgetheader, sip_getheader, synopsis_sipgetheader, descrip_sipgetheader); - - 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); - - ast_custom_function_register(&sip_header_function); - ast_custom_function_register(&sippeer_function); - - sip_poke_all_peers(); - sip_send_all_registers(); - - /* And start the monitor for the first time */ - restart_monitor(); - - return 0; -} - -int unload_module() -{ - struct sip_pvt *p, *pl; - - /* First, take us out of the channel type list */ - ast_channel_unregister(&sip_tech); - - ast_custom_function_unregister(&sippeer_function); - ast_custom_function_unregister(&sip_header_function); - - ast_unregister_application(app_dtmfmode); - ast_unregister_application(app_sipaddheader); - ast_unregister_application(app_sipgetheader); - - ast_cli_unregister_multiple(my_clis, sizeof(my_clis)/ sizeof(my_clis[0])); - - ast_rtp_proto_unregister(&sip_rtp); - - ast_manager_unregister("SIPpeers"); - ast_manager_unregister("SIPshowpeer"); - - if (!ast_mutex_lock(&iflock)) { - /* Hangup all interfaces if they have an owner */ - p = iflist; - while (p) { - if (p->owner) - ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD); - p = p->next; - } - iflist = NULL; - ast_mutex_unlock(&iflock); - } else { - ast_log(LOG_WARNING, "Unable to lock the interface list\n"); - return -1; - } - - if (!ast_mutex_lock(&monlock)) { - if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP)) { - pthread_cancel(monitor_thread); - pthread_kill(monitor_thread, SIGURG); - pthread_join(monitor_thread, NULL); - } - monitor_thread = AST_PTHREADT_STOP; - ast_mutex_unlock(&monlock); - } else { - ast_log(LOG_WARNING, "Unable to lock the monitor\n"); - return -1; - } - - if (!ast_mutex_lock(&iflock)) { - /* Destroy all the interfaces and free their memory */ - p = iflist; - while (p) { - pl = p; - p = p->next; - /* Free associated memory */ - ast_mutex_destroy(&pl->lock); - if (pl->chanvars) { - ast_variables_destroy(pl->chanvars); - pl->chanvars = NULL; - } - free(pl); - } - iflist = NULL; - ast_mutex_unlock(&iflock); - } else { - ast_log(LOG_WARNING, "Unable to lock the interface list\n"); - return -1; - } - - /* 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); - close(sipsock); - - return 0; -} - -int usecount() -{ - return usecnt; -} - -char *key() -{ - return ASTERISK_GPL_KEY; -} - -char *description() -{ - return (char *) desc; -} - - +/* + * Asterisk -- A telephony toolkit for Linux. + * + * Implementation of Session Initiation Protocol + * + * Copyright (C) 2004 - 2005, Digium, Inc. + * + * Mark Spencer + * + * This program is free software, distributed under the terms of + * the GNU General Public License + */ + + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision: 1.784 $") + +#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/lock.h" +#include "asterisk/sched.h" +#include "asterisk/io.h" +#include "asterisk/rtp.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/acl.h" +#include "asterisk/srv.h" +#include "asterisk/astdb.h" +#include "asterisk/causes.h" +#include "asterisk/utils.h" +#include "asterisk/file.h" +#include "asterisk/astobj.h" +#include "asterisk/dnsmgr.h" +#include "asterisk/devicestate.h" +#ifdef OSP_SUPPORT +#include "asterisk/astosp.h" +#endif + +#define SIP_TCP_SUPPORT /* this will enable SIP over TCP/TLS */ + +#ifdef SIP_TCP_SUPPORT +#define OPENSSL_NO_KRB5 /* to prevent compile error */ +#include +#include +#include +#include +#include +#endif + +#ifndef DEFAULT_USERAGENT +#define DEFAULT_USERAGENT "Asterisk PBX" +#endif + +#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 SIPDUMPER +#define DEFAULT_DEFAULT_EXPIRY 120 +#define DEFAULT_MAX_EXPIRY 3600 +#define DEFAULT_REGISTRATION_TIMEOUT 20 +#define DEFAULT_REGATTEMPTS_MAX 10 + +/* 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 */ + +static int max_expiry = DEFAULT_MAX_EXPIRY; +static int default_expiry = DEFAULT_DEFAULT_EXPIRY; + +#ifndef MAX +#define MAX(a,b) ((a) > (b) ? (a) : (b)) +#endif + +#define CALLERID_UNKNOWN "Unknown" + + + +#define DEFAULT_MAXMS 2000 /* Must be faster than 2 seconds by default */ +#define DEFAULT_FREQ_OK 60 * 1000 /* How often to check for the host to be up */ +#define DEFAULT_FREQ_NOTOK 10 * 1000 /* How often to check, if the host is down... */ + +#define DEFAULT_RETRANS 2000 /* How frequently to retransmit */ +#define MAX_RETRANS 5 /* Try only 5 times for retransmissions */ + + +#define DEBUG_READ 0 /* Recieved data */ +#define DEBUG_SEND 1 /* Transmit data */ + +static const char desc[] = "Session Initiation Protocol (SIP)"; +static const char channeltype[] = "SIP"; +static const char config[] = "sip.conf"; +static const char notify_config[] = "sip_notify.conf"; + +#define SIP_REGISTER 1 +#define SIP_OPTIONS 2 +#define SIP_NOTIFY 3 +#define SIP_INVITE 4 +#define SIP_ACK 5 +#define SIP_PRACK 6 +#define SIP_BYE 7 +#define SIP_REFER 8 +#define SIP_SUBSCRIBE 9 +#define SIP_MESSAGE 10 +#define SIP_UPDATE 11 +#define SIP_INFO 12 +#define SIP_CANCEL 13 +#define SIP_PUBLISH 14 +#define SIP_RESPONSE 100 + +#define RTP 1 +#define NO_RTP 0 +const struct cfsip_methods { + int id; + int need_rtp; /* when this is the 'primary' use for a pvt structure, does it need RTP? */ + char *text; +} sip_methods[] = { + { 0, RTP, "-UNKNOWN-" }, + { SIP_REGISTER, NO_RTP, "REGISTER" }, + { SIP_OPTIONS, NO_RTP, "OPTIONS" }, + { SIP_NOTIFY, NO_RTP, "NOTIFY" }, + { SIP_INVITE, RTP, "INVITE" }, + { SIP_ACK, NO_RTP, "ACK" }, + { SIP_PRACK, NO_RTP, "PRACK" }, + { SIP_BYE, NO_RTP, "BYE" }, + { SIP_REFER, NO_RTP, "REFER" }, + { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE" }, + { SIP_MESSAGE, NO_RTP, "MESSAGE" }, + { SIP_UPDATE, NO_RTP, "UPDATE" }, + { SIP_INFO, NO_RTP, "INFO" }, + { SIP_CANCEL, NO_RTP, "CANCEL" }, + { SIP_PUBLISH, NO_RTP, "PUBLISH" } +}; + +/* Structure for conversion between compressed SIP and "normal" SIP */ +static struct cfalias { + char *fullname; + char *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" }, +}; + +/* 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) + +/* List of well-known SIP options. If we get this in a require, + we should check the list and answer accordingly. */ +const struct cfsip_options { + int id; /* Bitmap ID */ + int supported; /* Supported by Asterisk ? */ + char *text; /* Text id, as in standard */ +} sip_options[] = { + /* Replaces: header for transfer */ + { SIP_OPT_REPLACES, SUPPORTED, "replaces" }, + /* RFC3262: PRACK 100% reliability */ + { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" }, + /* SIP Session Timers */ + { SIP_OPT_TIMER, NOT_SUPPORTED, "timer" }, + /* RFC3959: SIP Early session support */ + { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" }, + /* 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" }, + /* Not yet RFC, but registred with IANA */ + { SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp_anat" }, + /* RFC3329: Security agreement mechanism */ + { SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" }, + /* SIMPLE events: draft-ietf-simple-event-list-07.txt */ + { SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" }, + /* GRUU: Globally Routable User Agent URI's */ + { SIP_OPT_GRUU, NOT_SUPPORTED, "gruu" }, + /* Target-dialog: draft-ietf-sip-target-dialog-00.txt */ + { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "target-dialog" }, +}; + + +/* SIP Methods we support */ +#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY" + +/* SIP Extensions we support */ +#define SUPPORTED_EXTENSIONS "replaces" + +#define DEFAULT_SIP_PORT 5060 /* From RFC 3261 (former 2543) */ +#define SIP_MAX_PACKET 4096 /* Also from RFC 3261 (2543), should sub headers tho */ + +static char default_useragent[AST_MAX_EXTENSION] = DEFAULT_USERAGENT; + +#define DEFAULT_CONTEXT "default" +static char default_context[AST_MAX_CONTEXT] = DEFAULT_CONTEXT; + +static char default_language[MAX_LANGUAGE] = ""; + +#define DEFAULT_CALLERID "asterisk" +static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID; + +static char default_fromdomain[AST_MAX_EXTENSION] = ""; + +#define DEFAULT_NOTIFYMIME "application/simple-message-summary" +static char default_notifymime[AST_MAX_EXTENSION] = DEFAULT_NOTIFYMIME; + +#ifdef SIP_TCP_SUPPORT +#define MAX_PATH_LEN 100 +#define DEFAULT_SIP_TLS_PORT 5061 /* From RFC 3261 */ +#define DEFAULT_PASSWORD "asterisk" +#define DEFAULT_ENTROPY "/dev/urandom" +#define DEFAULT_TRUSTCERTS "/var/lib/asterisk/keys/trustcerts.pem" +#define DEFAULT_TRUSTCERTSDIR "/var/lib/asterisk/keys/trustdir" +#define DEFAULT_SERVERCERT "/var/lib/asterisk/keys/servercert.pem" +#define DEFAULT_SERVEREKEY "/var/lib/asterisk/keys/serverkey.pem" +#define DEFAULT_DH512 "/var/lib/asterisk/keys/dh512.pem" +#define DEFAULT_DH1024 "/var/lib/asterisk/keys/dh1024.pem" +#define CIPHER_LIST "ALL" + +static char trustcerts_file[MAX_PATH_LEN] = DEFAULT_TRUSTCERTS; +static char servercert_file[MAX_PATH_LEN] = DEFAULT_SERVERCERT; +static char serverkey_file[MAX_PATH_LEN] = DEFAULT_SERVEREKEY; +static char serverkey_password[MAX_PATH_LEN] = DEFAULT_PASSWORD; +static char dh512param_file[MAX_PATH_LEN] = DEFAULT_DH512; +static char dh1024param_file[MAX_PATH_LEN] = DEFAULT_DH1024; +#endif + +static int default_qualify = 0; /* Default Qualify= setting */ + +static struct ast_flags global_flags = {0}; /* global SIP_ flags */ +static struct ast_flags global_flags_page2 = {0}; /* more global SIP_ flags */ + +static int srvlookup = 0; /* SRV Lookup on or off. Default is off, RFC behavior is on */ + +static int pedanticsipchecking = 0; /* Extra checking ? Default off */ + +static int autocreatepeer = 0; /* Auto creation of peers at registration? Default off. */ + +static int relaxdtmf = 0; + +static int global_rtptimeout = 0; + +static int global_rtpholdtimeout = 0; + +static int global_rtpkeepalive = 0; + +static int global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT; +static int global_regattempts_max = DEFAULT_REGATTEMPTS_MAX; + +/* Object counters */ +static int suserobjs = 0; +static int ruserobjs = 0; +static int speerobjs = 0; +static int rpeerobjs = 0; +static int apeerobjs = 0; +static int regobjs = 0; + +static int global_allowguest = 1; /* allow unauthenticated users/peers to connect? */ + +#define DEFAULT_MWITIME 10 +static int global_mwitime = DEFAULT_MWITIME; /* Time between MWI checks for peers */ + +static int usecnt =0; +AST_MUTEX_DEFINE_STATIC(usecnt_lock); + + +/* Protect the interface list (of sip_pvt's) */ +AST_MUTEX_DEFINE_STATIC(iflock); + +/* 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); + +/* 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 restart_monitor(void); + +/* Codecs that we support by default: */ +static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263; +static int noncodeccapability = AST_RTP_DTMF; + +static struct in_addr __ourip; +static struct sockaddr_in outboundproxyip; +static int ourport; + +static int sipdebug = 0; +static struct sockaddr_in debugaddr; + +static int tos = 0; + +static int videosupport = 0; + +static int compactheaders = 0; /* send compact sip headers */ + +static int recordhistory = 0; /* Record SIP history. Off by default */ + +static char global_musicclass[MAX_MUSICCLASS] = ""; /* Global music on hold class */ +#define DEFAULT_REALM "asterisk" +static char global_realm[MAXHOSTNAMELEN] = DEFAULT_REALM; /* Default realm */ +static char regcontext[AST_MAX_CONTEXT] = ""; /* Context for auto-extensions */ + +/* Expire slowly */ +#define DEFAULT_EXPIRY 900 +static int expiry = DEFAULT_EXPIRY; + +static struct sched_context *sched; +static struct io_context *io; +/* The private structures of the sip channels are linked for + selecting outgoing channels */ + +#define SIP_MAX_HEADERS 64 +#define SIP_MAX_LINES 64 + +#define DEC_IN_USE 0 +#define INC_IN_USE 1 +#define DEC_OUT_USE 2 +#define INC_OUT_USE 3 + +static struct ast_codec_pref prefs; + + +/* 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 */ + char *header[SIP_MAX_HEADERS]; + int lines; /* SDP Content */ + char *line[SIP_MAX_LINES]; + char data[SIP_MAX_PACKET]; +}; + +struct sip_pkt; + +/* Parameters to the transmit_invite function */ +struct sip_invite_param { + char *distinctive_ring; + char *osptoken; + int addsipheaders; + char *vxml_url; + char *auth; + char *authheader; +}; + +struct sip_route { + struct sip_route *next; + char hop[0]; +}; + +/* sip_history: Structure for saving transactions within a SIP dialog */ +struct sip_history { + char event[80]; + struct sip_history *next; +}; + +/* sip_auth: Creadentials 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 */ +}; + +#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 */ +#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) /* Is this an outgoing call? */ +#define SIP_SELFDESTRUCT (1 << 14) +#define SIP_DYNAMIC (1 << 15) /* Is this a dynamic peer? */ +/* --- Choices for DTMF support in SIP channel */ +#define SIP_DTMF (3 << 16) /* three settings, uses two bits */ +#define SIP_DTMF_RFC2833 (0 << 16) /* RTP DTMF */ +#define SIP_DTMF_INBAND (1 << 16) /* Inband audio, only for ULAW/ALAW */ +#define SIP_DTMF_INFO (2 << 16) /* SIP Info messages */ +/* 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) +#define SIP_NAT_ROUTE (2 << 18) +#define SIP_NAT_ALWAYS (3 << 18) +/* re-INVITE related settings */ +#define SIP_REINVITE (3 << 20) /* two bits used */ +#define SIP_CAN_REINVITE (1 << 20) /* allow peers to be reinvited to send media directly p2p */ +#define SIP_REINVITE_UPDATE (2 << 20) /* use UPDATE (RFC3311) when reinviting this peer */ +/* "insecure" settings */ +#define SIP_INSECURE_PORT (1 << 22) /* don't require matching port for incoming requests */ +#define SIP_INSECURE_INVITE (1 << 23) /* don't require authentication for incoming INVITEs */ +/* Sending PROGRESS in-band settings */ +#define SIP_PROG_INBAND (3 << 24) /* three settings, uses two bits */ +#define SIP_PROG_INBAND_NEVER (0 << 24) +#define SIP_PROG_INBAND_NO (1 << 24) +#define SIP_PROG_INBAND_YES (2 << 24) +/* Open Settlement Protocol authentication */ +#define SIP_OSPAUTH (3 << 26) /* three settings, uses two bits */ +#define SIP_OSPAUTH_NO (0 << 26) +#define SIP_OSPAUTH_YES (1 << 26) +#define SIP_OSPAUTH_EXCLUSIVE (2 << 26) +/* Call states */ +#define SIP_CALL_ONHOLD (1 << 28) +#define SIP_CALL_LIMIT (1 << 29) + +/* a new page of flags for peer */ +#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0) +#define SIP_PAGE2_RTNOUPDATE (1 << 1) +#define SIP_PAGE2_RTAUTOCLEAR (1 << 2) +#define SIP_PAGE2_RTIGNOREREGEXPIRE (1 << 3) + +static int global_rtautoclear = 120; + +/* sip_pvt: PVT structures are used for each SIP conversation, ie. a call */ +static struct sip_pvt { + ast_mutex_t lock; /* Channel private lock */ + int method; /* SIP method of this packet */ + char callid[80]; /* Global CallID */ + char randdata[80]; /* Random data */ + struct ast_codec_pref prefs; /* codec prefs */ + 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 */ + unsigned int flags; /* SIP_ flags */ + unsigned int sipoptions; /* Supported SIP sipoptions on the other end */ + 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; + int callingpres; /* Calling presentation */ + int authtries; /* Times we've tried to authenticate */ + int expiry; /* How long we take to expire */ + int branch; /* One random number */ + int tag; /* Another random number */ + 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 */ + int redircodecs; /* Redirect codecs */ + struct sockaddr_in recv; /* Received as */ + struct in_addr ourip; /* Our IP */ +#ifdef SIP_TCP_SUPPORT + SSL *ssl; /* SSL object for TLS connection */ + int sockfd; /* socket fd used by this SIP channel*/ + char transport[4]; /* transport protocol, UDP, TCP or TLS */ +#endif + struct ast_channel *owner; /* Who owns us */ + char exten[AST_MAX_EXTENSION]; /* Extension where to start */ + char refer_to[AST_MAX_EXTENSION]; /* Place to store REFER-TO extension */ + char referred_by[AST_MAX_EXTENSION]; /* Place to store REFERRED-BY extension */ + char refer_contact[AST_MAX_EXTENSION]; /* Place to store Contact info from a REFER extension */ + struct sip_pvt *refer_call; /* Call we are referring */ + struct sip_route *route; /* Head of linked list of routing steps (fm Record-Route) */ + int route_persistant; /* Is this the "real" route? */ + char from[256]; /* The From: header */ + char useragent[256]; /* User agent in SIP request */ + char context[AST_MAX_CONTEXT]; /* Context for this call */ + char fromdomain[MAXHOSTNAMELEN]; /* Domain to show in the from field */ + char fromuser[AST_MAX_EXTENSION]; /* User to show in the user field */ + char fromname[AST_MAX_EXTENSION]; /* Name to show in the user field */ + char tohost[MAXHOSTNAMELEN]; /* Host we should put in the "to" field */ + char language[MAX_LANGUAGE]; /* Default language for this call */ + char musicclass[MAX_MUSICCLASS]; /* Music on Hold class */ + char rdnis[256]; /* Referring DNIS */ + char theirtag[256]; /* Their tag */ + char username[256]; /* [user] name */ + char peername[256]; /* [peer] name, not set if [user] */ + char authname[256]; /* Who we use for authentication */ + char uri[256]; /* Original requested URI */ + char okcontacturi[256]; /* URI from the 200 OK on INVITE */ + char peersecret[256]; /* Password */ + char peermd5secret[256]; + struct sip_auth *peerauth; /* Realm authentication */ + char cid_num[256]; /* Caller*ID */ + char cid_name[256]; /* Caller*ID */ + char via[256]; /* Via: header */ + char fullcontact[128]; /* The Contact: that the UA registers with us */ + char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */ + char our_contact[256]; /* Our contact header */ + char realm[MAXHOSTNAMELEN]; /* Authorization realm */ + char nonce[256]; /* Authorization nonce */ + char opaque[256]; /* Opaque nonsense */ + char qop[80]; /* Quality of Protection, since SIP wasn't complicated enough yet. */ + char domain[MAXHOSTNAMELEN]; /* Authorization domain */ + char lastmsg[256]; /* Last Message sent/received */ + int amaflags; /* AMA Flags */ + int pendinginvite; /* Any pending invite */ +#ifdef OSP_SUPPORT + int osphandle; /* OSP Handle for call */ + time_t ospstart; /* OSP Start time */ +#endif + struct sip_request initreq; /* Initial request */ + + int maxtime; /* Max time for first response */ + int maxforwards; /* keep the max-forwards info */ + int initid; /* Auto-congest ID if appropriate */ + int autokillid; /* Auto-kill ID */ + time_t lastrtprx; /* Last RTP received */ + time_t lastrtptx; /* Last RTP sent */ + int rtptimeout; /* RTP timeout time */ + int rtpholdtimeout; /* RTP timeout when on hold */ + int rtpkeepalive; /* Send RTP packets for keepalive */ + + int subscribed; /* Is this call a subscription? */ + int stateid; + int dialogver; + + struct ast_dsp *vad; /* Voice Activation Detection dsp */ + + struct sip_peer *peerpoke; /* If this calls is to poke a peer, which one */ + struct sip_registry *registry; /* If this is a REGISTER call, 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 *history; /* History of this SIP dialog */ + struct ast_variable *chanvars; /* Channel variables to set for call */ + struct sip_pvt *next; /* Next call in chain */ +} *iflist = NULL; + +#define FLAG_RESPONSE (1 << 0) +#define FLAG_FATAL (1 << 1) + +/* sip packet - read in sipsock_read, transmitted in send_request */ +struct sip_pkt { + struct sip_pkt *next; /* Next packet */ + int retrans; /* Retransmission number */ + int seqno; /* Sequence number */ + unsigned int flags; /* non-zero if this is a response packet (e.g. 200 OK) */ + struct sip_pvt *owner; /* Owner call */ + int retransid; /* Retransmission ID */ + int packetlen; /* Length of packet */ + char data[0]; +}; + +/* 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 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 musicclass[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 flags; /* SIP flags */ + unsigned int sipoptions; /* Supported SIP options */ + struct ast_flags flags_page2; /* SIP_PAGE2 flags */ + int amaflags; /* AMA flags for billing */ + int callingpres; /* Calling id presentation */ + int capability; /* Codec capability */ + int inUse; /* Number of calls in use */ + int incominglimit; /* Limit of incoming calls */ + int outUse; /* disabled */ + int outgoinglimit; /* disabled */ + struct ast_ha *ha; /* ACL setting */ + struct ast_variable *chanvars; /* Variables to set for channel created by user */ +}; + +/* Structure for SIP peer data, we place calls to peers if registred or fixed IP address (host) */ +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 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 registred 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 incominglimit; /* Limit of incoming calls */ + int outUse; /* disabled */ + int outgoinglimit; /* disabled */ + char mailbox[AST_MAX_EXTENSION]; /* Mailbox setting for MWI checks */ + char language[MAX_LANGUAGE]; /* Default language for prompts */ + char musicclass[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 flags; /* SIP flags */ + unsigned int sipoptions; /* Supported SIP options */ + struct ast_flags flags_page2; /* SIP_PAGE2 flags */ + int expire; /* When to expire this peer registration */ + int expiry; /* Duration of 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 ast_dnsmgr_entry *dnsmgr;/* DNS refresh manager for peer */ + struct sockaddr_in addr; /* IP address of peer */ + struct in_addr mask; +#ifdef SIP_TCP_SUPPORT + SSL *ssl; /* SSL object for TLS connection */ + int tcpsockfd; /* TCP connection socket is saved to here */ + char transport[4]; /* transport protocol, UDP or TCP */ +#endif + + /* 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_variable *chanvars; /* Variables to set for channel created by user */ + int lastmsg; +}; + +AST_MUTEX_DEFINE_STATIC(sip_reload_lock); +static int sip_reloading = 0; + +/* States for outbound registrations (with register= lines in sip.conf */ +#define REG_STATE_UNREGISTERED 0 +#define REG_STATE_REGSENT 1 +#define REG_STATE_AUTHSENT 2 +#define REG_STATE_REGISTERED 3 +#define REG_STATE_REJECTED 4 +#define REG_STATE_TIMEOUT 5 +#define REG_STATE_NOAUTH 6 +#define REG_STATE_FAILED 7 + + +/* sip_registry: Registrations with other SIP proxies */ +struct sip_registry { + ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1); + int portno; /* Optional port override */ + char username[80]; /* Who we are registering as */ + char authuser[80]; /* Who we *authenticate* as */ + char hostname[MAXHOSTNAMELEN]; /* Domain or host we register to */ + char secret[80]; /* Password or key name in []'s */ + char md5secret[80]; + char contact[256]; /* Contact extension */ + char random[80]; + 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 call" in progress */ + int regstate; /* Registration state (see above) */ + int callid_valid; /* 0 means we haven't chosen callid for this registry yet. */ + char callid[80]; /* Global CallID for this registry */ + unsigned int ocseq; /* Sequence number we got to for REGISTERs for this registry */ + struct sockaddr_in us; /* Who the server thinks we are */ + + /* Saved headers */ + char realm[MAXHOSTNAMELEN]; /* Authorization realm */ + char nonce[256]; /* Authorization nonce */ + char domain[MAXHOSTNAMELEN]; /* Authorization domain */ + char opaque[256]; /* Opaque nonsense */ + char qop[80]; /* Quality of Protection. */ + + char lastmsg[256]; /* Last Message sent/received */ +}; + +/*--- The user list: Users and friends ---*/ +static struct ast_user_list { + ASTOBJ_CONTAINER_COMPONENTS(struct sip_user); +} userl; + +/*--- The peer list: Peers and Friends ---*/ +static struct ast_peer_list { + ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer); +} peerl; + +/*--- The register list: Other SIP proxys we register with and call ---*/ +static struct ast_register_list { + ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry); + int recheck; +} regl; + + +static int __sip_do_register(struct sip_registry *r); + +static int sipsock = -1; + +#ifdef SIP_TCP_SUPPORT +static int siptcpsock = -1; /* TCP listening socket */ +static int siptlssock = -1; /* TLS listening socket */ +static SSL_CTX *tlsctx = NULL; /* SSL Context for TLS */ +static struct sockaddr_in tlsbindaddr; /* TLS bind address */ +#endif + + + +static struct sockaddr_in bindaddr; +static struct sockaddr_in externip; +static char externhost[MAXHOSTNAMELEN] = ""; +static time_t externexpire = 0; +static int externrefresh = 10; +static struct ast_ha *localaddr; + +/* The list of manual NOTIFY types we know how to send */ +struct ast_config *notify_types; + +static struct sip_auth *authl; /* Authentication list */ + + +static struct ast_frame *sip_read(struct ast_channel *ast); +static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req); +static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans); +static int transmit_response_with_unsupported(struct sip_pvt *p, char *msg, struct sip_request *req, char *unsupported); +static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_request *req, char *rand, int reliable, char *header, int stale); +static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch); +static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch); +static int transmit_invite(struct sip_pvt *p, int sipmethod, int sendsdp, struct sip_invite_param *options, int init); +static int transmit_reinvite_with_sdp(struct sip_pvt *p); +static int transmit_info_with_digit(struct sip_pvt *p, char digit); +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 sip_sipredirect(struct sip_pvt *p, const char *dest); +static struct sip_peer *temp_peer(const char *name); +static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, int sipmethod, int init); +static void free_old_route(struct sip_route *route); +static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len); +static int update_user_counter(struct sip_pvt *fup, int event); +static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int realtime); +static struct sip_user *build_user(const char *name, struct ast_variable *v, int realtime); +static int sip_do_reload(void); +static int expire_register(void *data); +static int callevents = 0; +#ifdef SIP_TCP_SUPPORT +static int sipsock_read(int *id, int fd, short events, void *ignore); +static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime); +#endif + +static struct ast_channel *sip_request(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); +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(struct ast_channel *ast, char digit); +static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */ +static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); /* Add realm authentication in list */ +static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, char *realm); /* Find authentication for a specific realm */ +static void append_date(struct sip_request *req); /* Append date to SIP packet */ + +/* Definition of this channel for channel registration */ +static const struct ast_channel_tech sip_tech = { + .type = channeltype, + .description = "Session Initiation Protocol (SIP)", + .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1), + .properties = AST_CHAN_TP_WANTSJITTER, + .requester = sip_request, + .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 = sip_senddigit, + .bridge = ast_rtp_bridge, + .send_text = sip_sendtext, +}; + +/*--- find_sip_method: Find SIP method from header */ +int find_sip_method(char *msg) +{ + int i, res = 0; + /* Strictly speaking, SIP methods are case SENSITIVE, but we don't check */ + for (i=1;(i < (sizeof(sip_methods) / sizeof(sip_methods[0]))) && !res; i++) { + if (!strcasecmp(sip_methods[i].text, msg)) + res = sip_methods[i].id; + } + return res; +} + +/* + * If there is a string in , strip everything around and return + * the content. Otherwise return the original argument. + */ +static char *get_in_brackets(char *c) +{ + char *n = strchr(c, '<'); + + if (n) { + c = n + 1; + n = strchr(c, '>'); + /* Lose the part after the > */ + if (n) + *n = '\0'; + } + return c; +} + +/*--- parse_sip_options: Parse supported header in incoming packet */ +unsigned int parse_sip_options(struct sip_pvt *pvt, char *supported) +{ + char *next = NULL; + char *sep = NULL; + char *temp = ast_strdupa(supported); + int i; + unsigned int profile = 0; + + if (!supported || ast_strlen_zero(supported) ) + return 0; + + if (option_debug > 2 && sipdebug) + ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported); + + next = temp; + while (next) { + char res=0; + if ( (sep = strchr(next, ',')) != NULL) { + *sep = '\0'; + sep++; + } + while (*next == ' ') /* Skip spaces */ + 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]))) && !res; i++) { + if (!strcasecmp(next, sip_options[i].text)) { + profile |= sip_options[i].id; + res = 1; + if (option_debug > 2 && sipdebug) + ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next); + } + } + if (!res) + if (option_debug > 2 && sipdebug) + ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next); + next = sep; + } + if (pvt) + pvt->sipoptions = profile; + + ast_log(LOG_DEBUG, "* SIP extension value: %d for call %s\n", profile, pvt->callid); + return profile; +} + +/*--- sip_debug_test_addr: See if we pass debug IP filter */ +static inline int sip_debug_test_addr(struct sockaddr_in *addr) +{ + if (sipdebug == 0) + 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; +} + +/*--- sip_debug_test_pvt: Test PVT for debugging output */ +static inline int sip_debug_test_pvt(struct sip_pvt *p) +{ + if (sipdebug == 0) + return 0; + return sip_debug_test_addr(((ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) ? &p->recv : &p->sa)); +} + + +/*--- __sip_xmit: Transmit SIP message ---*/ +static int __sip_xmit(struct sip_pvt *p, char *data, int len) +{ + int res = 0; + char iabuf[INET_ADDRSTRLEN]; + +#ifdef SIP_TCP_SUPPORT + struct sip_peer *peer = NULL; + + if ((p->sockfd > -1) && (p->sockfd != sipsock)) { /* This is TCP */ + /* ast_verbose("TCP_Write: fd %d\n", p->sockfd); */ + if (p->ssl) { /* TLS write */ + while (len > res) { + res += SSL_write(p->ssl, data + res, len - res); + switch (SSL_get_error(p->ssl, res)) { + case SSL_ERROR_NONE: + break; + default: + ast_log(LOG_ERROR, "SSL write error\n"); + SSL_clear(p->ssl); + SSL_free(p->ssl); + p->ssl = NULL; + p->sockfd = -1; + peer = find_peer(p->peername, NULL, 1); + if (peer && peer->ssl) { + peer->ssl = NULL; + peer->tcpsockfd = -1; + } + break; + } + } + } else /* TCP write */ + res = write(p->sockfd, data, len); + } else { +#endif + if (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) + res=sendto(sipsock, data, len, 0, (struct sockaddr *)&p->recv, sizeof(struct sockaddr_in)); + else + res=sendto(sipsock, data, len, 0, (struct sockaddr *)&p->sa, sizeof(struct sockaddr_in)); +#ifdef SIP_TCP_SUPPORT + } +#endif + if (res != len) { + ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s returned %d: %s\n", data, len, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), res, strerror(errno)); + } + return res; +} + +static void sip_destroy(struct sip_pvt *p); + +/*--- build_via: Build a Via header for a request ---*/ +static void build_via(struct sip_pvt *p, char *buf, int len) +{ + char iabuf[INET_ADDRSTRLEN]; + + /* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */ +#ifdef SIP_TCP_SUPPORT + if (ast_test_flag(p, SIP_NAT) & SIP_NAT_RFC3581) + snprintf(p->via, sizeof(p->via), "SIP/2.0/%s %s:%d;branch=z9hG4bK%08x;rport", p->transport, ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport, p->branch); + else + snprintf(p->via, sizeof(p->via), "SIP/2.0/%s %s:%d;branch=z9hG4bK%08x", p->transport, ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport, p->branch); +#else + if (ast_test_flag(p, SIP_NAT) & SIP_NAT_RFC3581) + snprintf(buf, len, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x;rport", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport, p->branch); + else /* Work around buggy UNIDEN UIP200 firmware */ + snprintf(buf, len, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport, p->branch); +#endif +} + +/*--- ast_sip_ouraddrfor: NAT fix - decide which IP address to use for ASterisk server? ---*/ +/* Only used for outbound registrations */ +static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us) +{ + /* + * Using the localaddr structure built up with localnet statements + * apply it to their address to see if we need to substitute our + * externip or can get away with our internal bindaddr + */ + struct sockaddr_in theirs; + theirs.sin_addr = *them; + if (localaddr && externip.sin_addr.s_addr && + ast_apply_ha(localaddr, &theirs)) { + char iabuf[INET_ADDRSTRLEN]; + if (externexpire && (time(NULL) >= externexpire)) { + struct ast_hostent ahp; + struct hostent *hp; + time(&externexpire); + externexpire += 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); + } + memcpy(us, &externip.sin_addr, sizeof(struct in_addr)); + ast_inet_ntoa(iabuf, sizeof(iabuf), *(struct in_addr *)&them->s_addr); + ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n", iabuf); + } + else if (bindaddr.sin_addr.s_addr) + memcpy(us, &bindaddr.sin_addr, sizeof(struct in_addr)); + else + return ast_ouraddrfor(them, us); + return 0; +} + +/*--- append_history: Append to SIP dialog history */ +/* Always returns 0 */ +static int append_history(struct sip_pvt *p, char *event, char *data) +{ + struct sip_history *hist, *prev; + char *c; + + if (!recordhistory) + return 0; + if(!(hist = malloc(sizeof(struct sip_history)))) { + ast_log(LOG_WARNING, "Can't allocate memory for history"); + return 0; + } + memset(hist, 0, sizeof(struct sip_history)); + snprintf(hist->event, sizeof(hist->event), "%-15s %s", event, data); + /* Trim up nicely */ + c = hist->event; + while(*c) { + if ((*c == '\r') || (*c == '\n')) { + *c = '\0'; + break; + } + c++; + } + /* Enqueue into history */ + prev = p->history; + if (prev) { + while(prev->next) + prev = prev->next; + prev->next = hist; + } else { + p->history = hist; + } + return 0; +} + +/*--- retrans_pkt: Retransmit SIP message if no answer ---*/ +static int retrans_pkt(void *data) +{ + struct sip_pkt *pkt=data, *prev, *cur; + int res = 0; + char iabuf[INET_ADDRSTRLEN]; + ast_mutex_lock(&pkt->owner->lock); + if (pkt->retrans < MAX_RETRANS) { + pkt->retrans++; + if (sip_debug_test_pvt(pkt->owner)) { + if (ast_test_flag(pkt->owner, SIP_NAT) & SIP_NAT_ROUTE) + ast_verbose("Retransmitting #%d (NAT) to %s:%d:\n%s\n---\n", pkt->retrans, ast_inet_ntoa(iabuf, sizeof(iabuf), pkt->owner->recv.sin_addr), ntohs(pkt->owner->recv.sin_port), pkt->data); + else + ast_verbose("Retransmitting #%d (no NAT) to %s:%d:\n%s\n---\n", pkt->retrans, ast_inet_ntoa(iabuf, sizeof(iabuf), pkt->owner->sa.sin_addr), ntohs(pkt->owner->sa.sin_port), pkt->data); + } + append_history(pkt->owner, "ReTx", pkt->data); + __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); + res = 1; + } else { + ast_log(LOG_WARNING, "Maximum retries exceeded on call %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request"); + append_history(pkt->owner, "MaxRetries", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)"); + pkt->retransid = -1; + if (ast_test_flag(pkt, FLAG_FATAL)) { + while(pkt->owner->owner && ast_mutex_trylock(&pkt->owner->owner->lock)) { + ast_mutex_unlock(&pkt->owner->lock); + usleep(1); + ast_mutex_lock(&pkt->owner->lock); + } + if (pkt->owner->owner) { + ast_set_flag(pkt->owner, SIP_ALREADYGONE); + ast_queue_hangup(pkt->owner->owner); + ast_mutex_unlock(&pkt->owner->owner->lock); + } else { + /* If no owner, destroy now */ + ast_set_flag(pkt->owner, SIP_NEEDDESTROY); + } + } + /* In any case, go ahead and remove the packet */ + prev = NULL; + cur = pkt->owner->packets; + while(cur) { + if (cur == pkt) + break; + prev = cur; + cur = cur->next; + } + 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 res; +} + +/*--- __sip_reliable_xmit: transmit packet with retransmits ---*/ +static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal) +{ + struct sip_pkt *pkt; + pkt = malloc(sizeof(struct sip_pkt) + len + 1); + if (!pkt) + return -1; + memset(pkt, 0, sizeof(struct sip_pkt)); + memcpy(pkt->data, data, len); + pkt->packetlen = len; + pkt->next = p->packets; + pkt->owner = p; + pkt->seqno = seqno; + pkt->flags = resp; + pkt->data[len] = '\0'; + if (fatal) + ast_set_flag(pkt, FLAG_FATAL); + /* Schedule retransmission */ + pkt->retransid = ast_sched_add(sched, DEFAULT_RETRANS, retrans_pkt, pkt); + pkt->next = p->packets; + p->packets = pkt; + __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); + if (!strncasecmp(pkt->data, "INVITE", 6)) { + /* Note this is a pending invite */ + p->pendinginvite = seqno; + } + return 0; +} + +/*--- __sip_autodestruct: Kill a call (called by scheduler) ---*/ +static int __sip_autodestruct(void *data) +{ + struct sip_pvt *p = data; + + p->autokillid = -1; + ast_log(LOG_DEBUG, "Auto destroying call '%s'\n", p->callid); + append_history(p, "AutoDestroy", ""); + if (p->owner) { + ast_log(LOG_WARNING, "Autodestruct on call '%s' with owner in place\n", p->callid); + ast_queue_hangup(p->owner); + } else { + sip_destroy(p); + } + return 0; +} + +/*--- sip_scheddestroy: Schedule destruction of SIP call ---*/ +static int sip_scheddestroy(struct sip_pvt *p, int ms) +{ + char tmp[80]; + if (sip_debug_test_pvt(p)) + ast_verbose("Scheduling destruction of call '%s' in %d ms\n", p->callid, ms); + if (recordhistory) { + snprintf(tmp, sizeof(tmp), "%d ms", ms); + append_history(p, "SchedDestroy", tmp); + } + if (p->autokillid > -1) + ast_sched_del(sched, p->autokillid); + p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, p); + return 0; +} + +/*--- sip_cancel_destroy: Cancel destruction of SIP call ---*/ +static int sip_cancel_destroy(struct sip_pvt *p) +{ + if (p->autokillid > -1) + ast_sched_del(sched, p->autokillid); + append_history(p, "CancelDestroy", ""); + p->autokillid = -1; + return 0; +} + +/*--- __sip_ack: Acknowledges receipt of a packet and stops retransmission ---*/ +static int __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod) +{ + struct sip_pkt *cur, *prev = NULL; + int res = -1; + int resetinvite = 0; + /* Just in case... */ + char *msg; + + msg = sip_methods[sipmethod].text; + + cur = p->packets; + while(cur) { + 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)) { + ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite); + p->pendinginvite = 0; + resetinvite = 1; + } + /* this is our baby */ + if (prev) + prev->next = cur->next; + else + p->packets = cur->next; + if (cur->retransid > -1) + ast_sched_del(sched, cur->retransid); + free(cur); + res = 0; + break; + } + prev = cur; + cur = cur->next; + } + ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found"); + return res; +} + +/* Pretend to ack all packets */ +static int __sip_pretend_ack(struct sip_pvt *p) +{ + char method[128]=""; + struct sip_pkt *cur=NULL; + char *c; + while(p->packets) { + if (cur == p->packets) { + ast_log(LOG_WARNING, "Have a packet that doesn't want to give up!\n"); + return -1; + } + cur = p->packets; + ast_copy_string(method, p->packets->data, sizeof(method)); + c = ast_skip_blanks(method); /* XXX what ? */ + *c = '\0'; + __sip_ack(p, p->packets->seqno, (ast_test_flag(p->packets, FLAG_RESPONSE)), find_sip_method(method)); + } + return 0; +} + +/*--- __sip_semi_ack: 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; + char *msg = sip_methods[sipmethod].text; + + cur = p->packets; + while(cur) { + 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)))) { + /* this is our baby */ + if (cur->retransid > -1) + ast_sched_del(sched, cur->retransid); + cur->retransid = -1; + res = 0; + break; + } + cur = cur->next; + } + ast_log(LOG_DEBUG, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found"); + return res; +} + +static void parse(struct sip_request *req); +static char *get_header(struct sip_request *req, char *name); +static void copy_request(struct sip_request *dst,struct sip_request *src); + +/*--- parse_copy: Copy SIP request, parse it */ +static void parse_copy(struct sip_request *dst, struct sip_request *src) +{ + memset(dst, 0, sizeof(*dst)); + memcpy(dst->data, src->data, sizeof(dst->data)); + dst->len = src->len; + parse(dst); +} + +#ifdef SIP_TCP_SUPPORT +static int tcptls_connect(struct sip_pvt *p) +{ + int fd = -1; + struct sockaddr_in myaddr; + char iabuf[INET_ADDRSTRLEN]; + + BIO *bio; + SSL *ssl; + + if (!strncasecmp(p->transport, "UDP", 3)) { + return sipsock; + } + + /* Do we need to handle NAT here? */ + + if ( (fd = socket(AF_INET, SOCK_STREAM, 0)) < 0) { + ast_log(LOG_WARNING, "TCP can't create socket : %s\n", strerror(errno)); + return -1; + } + + /* bind local protocol address to socket */ + bzero(&myaddr, sizeof(struct sockaddr_in)); + myaddr.sin_family = AF_INET; + memcpy(&myaddr.sin_addr, &p->ourip, sizeof(p->ourip)); + myaddr.sin_port = htons(0); /* any port is OK? */ + + if (bind(fd, (struct sockaddr *)&myaddr, sizeof(struct sockaddr_in)) < 0) { + ast_log(LOG_WARNING, "TCP failed to bind : %s\n", strerror(errno)); + return -1; + } + + /* start 3-way hand shake with the peer */ + if (connect(fd, (struct sockaddr *) &p->sa, sizeof(struct sockaddr_in)) < 0) { + ast_log(LOG_WARNING, "TCP can't connect to %s:%d, error %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port), strerror(errno)); + return -1; + } + + if (sip_debug_test_pvt(p)) + ast_verbose(VERBOSE_PREFIX_2 "Successfuly TCP connected fd %d to %s:%d\n", fd, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port)); + + p->sockfd = fd; + + /* If TLS, do TLS handshake */ + if ((fd > -1) && !strncasecmp(p->transport, "TLS", sizeof("TLS"))) { + /* Initiate TLS handshake */ + bio = BIO_new_socket(fd, BIO_CLOSE); + ssl = SSL_new(tlsctx); + SSL_set_bio(ssl, bio, bio); + + if(SSL_connect(ssl) <= 0) { + ast_log(LOG_ERROR, "SSL_connect error"); + SSL_clear(ssl); + SSL_free(ssl); + close(fd); + p->ssl = NULL; + p->sockfd = -1; + return -1; + } + ast_log(LOG_DEBUG, "New TLS connection is opened\n"); + p->ssl = ssl; + } + + return fd; +} +#endif + + +/*--- send_response: Transmit response on SIP request---*/ +static int send_response(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno) +{ + int res; + char iabuf[INET_ADDRSTRLEN]; + struct sip_request tmp; + char tmpmsg[80]; + + if (sip_debug_test_pvt(p)) { + if (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) + ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(iabuf, sizeof(iabuf), 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(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port), req->data); + } +#ifdef SIP_TCP_SUPPORT + /* if transport is TCP and not opened connection, connect now */ + if (strncasecmp(p->transport, "UDP", 3)) { + /* make TCP connection only when not connected and no NAT/firewall */ + if ((p->sockfd < 0) && !(ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)) { + p->sockfd = tcptls_connect(p); + if (p->sockfd < 0) { + ast_log(LOG_WARNING, "Failed to make TCP connection to %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port)); + return -1; + } + ast_io_add(io, p->sockfd, sipsock_read, AST_IO_IN, NULL); + } else if (p->sockfd > 0) { + /* check TCP connection status, still alive or disconnected? + if disconnected & no-NAT, clean up and reconnect */ + + } else if (p->sockfd < 0) { + if (sip_debug_test_pvt(p)) + ast_log(LOG_WARNING, "peer is NATed, but TCP socket is closed\n"); + return -1; + } + } + + if (reliable && (p->sockfd == sipsock)) { /* only UDP needs reliable transmit */ +#else + if (reliable) { +#endif + if (recordhistory) { + parse_copy(&tmp, req); + snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq")); + append_history(p, "TxRespRel", tmpmsg); + } + res = __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable > 1)); + } else { + if (recordhistory) { + parse_copy(&tmp, req); + snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq")); + append_history(p, "TxResp", tmpmsg); + } + res = __sip_xmit(p, req->data, req->len); + } + if (res > 0) + return 0; + return res; +} + +/*--- send_request: Send SIP Request to the other part of the dialogue ---*/ +static int send_request(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno) +{ + int res; + char iabuf[INET_ADDRSTRLEN]; + struct sip_request tmp; + char tmpmsg[80]; + + if (sip_debug_test_pvt(p)) { + if (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) + ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(iabuf, sizeof(iabuf), 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(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port), req->data); + } +#ifdef SIP_TCP_SUPPORT + /* if transport is TCP and not opened connection, connect now */ + if (strncasecmp(p->transport, "UDP", 3)) { + /* make TCP connection only when not connected and no NAT/firewall */ + if ((p->sockfd < 0) && !(ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)) { + p->sockfd = tcptls_connect(p); + if (p->sockfd < 0) { + ast_log(LOG_WARNING, "Failed to make TCP connection to %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port)); + return -1; + } + ast_io_add(io, p->sockfd, sipsock_read, AST_IO_IN, NULL); + } else if (p->sockfd < 0) { + if (sip_debug_test_pvt(p)) + ast_log(LOG_WARNING, "peer is NATed, but TCP socket is closed\n"); + return -1; + } + } + + if (reliable && p->sockfd == sipsock) { /* Only UDP needs this */ +#else + if (reliable) { +#endif + if (recordhistory) { + parse_copy(&tmp, req); + snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq")); + append_history(p, "TxReqRel", tmpmsg); + } + res = __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1)); + } else { + if (recordhistory) { + parse_copy(&tmp, req); + snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq")); + append_history(p, "TxReq", tmpmsg); + } + res = __sip_xmit(p, req->data, req->len); + } + return res; +} + +/*--- url_decode: Decode SIP URL (overwrite the string) ---*/ +static void url_decode(char *s) +{ + char *o; + unsigned int tmp; + + for (o = s; *s; s++, o++) { + if (*s == '%' && strlen(s) > 2 && sscanf(s + 1, "%2x", &tmp) == 1) { + /* have '%', two chars and correct parsing */ + *o = tmp; + s += 2; /* Will be incremented once more when we break out */ + } else /* all other cases, just copy */ + *o = *s; + } + *o = '\0'; +} + +/*--- ditch_braces: Pick out text in braces from character string ---*/ +static char *ditch_braces(char *tmp) +{ + char *c = tmp; + char *n; + char *q; + if ((q = strchr(tmp, '"')) ) { + c = q + 1; + if ((q = strchr(c, '"')) ) + c = q + 1; + else { + ast_log(LOG_WARNING, "No closing quote in '%s'\n", tmp); + c = tmp; + } + } + if ((n = strchr(c, '<')) ) { + c = n + 1; + while(*c && *c != '>') c++; + if (*c != '>') { + ast_log(LOG_WARNING, "No closing brace in '%s'\n", tmp); + } else { + *c = '\0'; + } + return n+1; + } + return c; +} + +/*--- sip_sendtext: Send SIP MESSAGE text within a call ---*/ +/* Called from PBX core text message functions */ +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 (!text || 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; +} + +/*--- realtime_update_peer: Update peer object in realtime storage ---*/ +static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, int expirey) +{ + char port[10] = ""; + char ipaddr[20] = ""; + char regseconds[20] = "0"; + + if (expirey) { /* Registration */ + time_t nowtime; + time(&nowtime); + nowtime += expirey; + snprintf(regseconds, sizeof(regseconds), "%ld", nowtime); /* Expiration time */ + ast_inet_ntoa(ipaddr, sizeof(ipaddr), sin->sin_addr); + snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port)); + } + ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr, "port", port, "regseconds", regseconds, "username", username, NULL); +} + +/*--- register_peer_exten: Automatically add peer extension to dial plan ---*/ +static void register_peer_exten(struct sip_peer *peer, int onoff) +{ + char multi[256]=""; + char *stringp, *ext; + if (!ast_strlen_zero(regcontext)) { + ast_copy_string(multi, ast_strlen_zero(peer->regexten) ? peer->name : peer->regexten, sizeof(multi)); + stringp = multi; + while((ext = strsep(&stringp, "&"))) { + if (onoff) + ast_add_extension(regcontext, 1, ext, 1, NULL, NULL, "Noop", strdup(peer->name), free, channeltype); + else + ast_context_remove_extension(regcontext, ext, 1, NULL); + } + } +} + +/*--- sip_destroy_peer: Destroy peer object from memory */ +static void sip_destroy_peer(struct sip_peer *peer) +{ + /* Delete it, it needs to disappear */ + if (peer->call) + sip_destroy(peer->call); + if (peer->chanvars) { + ast_variables_destroy(peer->chanvars); + peer->chanvars = NULL; + } + if (peer->expire > -1) + ast_sched_del(sched, peer->expire); + if (peer->pokeexpire > -1) + ast_sched_del(sched, peer->pokeexpire); + register_peer_exten(peer, 0); + ast_free_ha(peer->ha); + if (ast_test_flag(peer, SIP_SELFDESTRUCT)) + apeerobjs--; + else if (ast_test_flag(peer, SIP_REALTIME)) + rpeerobjs--; + else + speerobjs--; + clear_realm_authentication(peer->auth); + peer->auth = (struct sip_auth *) NULL; + if (peer->dnsmgr) + ast_dnsmgr_release(peer->dnsmgr); +#ifdef SIP_TCP_SUPPORT + if (peer->ssl) { + SSL_clear(peer->ssl); + SSL_free(peer->ssl); + } +#endif + free(peer); +} + +/*--- update_peer: Update peer data in database (if used) ---*/ +static void update_peer(struct sip_peer *p, int expiry) +{ + if (!ast_test_flag((&global_flags_page2), SIP_PAGE2_RTNOUPDATE) && + (ast_test_flag(p, SIP_REALTIME) || + ast_test_flag(&(p->flags_page2), SIP_PAGE2_RTCACHEFRIENDS))) { + realtime_update_peer(p->name, &p->addr, p->username, expiry); + } +} + + +/*--- realtime_peer: Get peer from realtime storage ---*/ +/* Checks the "sippeers" realtime family from extconfig.conf */ +static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin) +{ + struct sip_peer *peer=NULL; + struct ast_variable *var; + struct ast_variable *tmp; + char *newpeername = (char *) peername; + char iabuf[80] = ""; + + /* First check on peer name */ + if (newpeername) + var = ast_load_realtime("sippeers", "name", peername, NULL); + else if (sin) { /* Then check on IP address */ + ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr); + var = ast_load_realtime("sippeers", "ipaddr", iabuf, NULL); + } else + return NULL; + + if (!var) + return NULL; + + tmp = var; + /* If this is type=user, then skip this object. */ + while(tmp) { + 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; + } + tmp = tmp->next; + } + + if (!newpeername) { /* Did not find peer in realtime */ + ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", iabuf); + ast_variables_destroy(var); + return (struct sip_peer *) NULL; + } + + /* Peer found in realtime, now build it in memory */ + peer = build_peer(newpeername, var, !ast_test_flag((&global_flags_page2), SIP_PAGE2_RTCACHEFRIENDS)); + + if (!peer) { + ast_variables_destroy(var); + return (struct sip_peer *) NULL; + } + if (ast_test_flag((&global_flags_page2), SIP_PAGE2_RTCACHEFRIENDS)) { + /* Cache peer */ + ast_copy_flags((&peer->flags_page2),(&global_flags_page2), SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS); + if (ast_test_flag((&global_flags_page2), SIP_PAGE2_RTAUTOCLEAR)) { + if (peer->expire > -1) { + ast_sched_del(sched, peer->expire); + } + peer->expire = ast_sched_add(sched, (global_rtautoclear) * 1000, expire_register, (void *)peer); + } + ASTOBJ_CONTAINER_LINK(&peerl,peer); + } else { + ast_set_flag(peer, SIP_REALTIME); + } + ast_variables_destroy(var); + return peer; +} + +/*--- sip_addrcmp: 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, SIP_INSECURE_PORT) && + (p->addr.sin_addr.s_addr == sin->sin_addr.s_addr))); +} + +/*--- find_peer: 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) +{ + 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) { + p = realtime_peer(peer, sin); + } + + return p; +} + +/*--- sip_destroy_user: Remove user object from in-memory storage ---*/ +static void sip_destroy_user(struct sip_user *user) +{ + ast_free_ha(user->ha); + if (user->chanvars) { + ast_variables_destroy(user->chanvars); + user->chanvars = NULL; + } + if (ast_test_flag(user, SIP_REALTIME)) + ruserobjs--; + else + suserobjs--; + free(user); +} + +/*--- realtime_user: 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; + + tmp = var; + while (tmp) { + if (!strcasecmp(tmp->name, "type") && + !strcasecmp(tmp->value, "peer")) { + ast_variables_destroy(var); + return NULL; + } + tmp = tmp->next; + } + + + + user = build_user(username, var, !ast_test_flag((&global_flags_page2), SIP_PAGE2_RTCACHEFRIENDS)); + + if (!user) { /* No user found */ + ast_variables_destroy(var); + return NULL; + } + + if (ast_test_flag((&global_flags_page2), SIP_PAGE2_RTCACHEFRIENDS)) { + ast_set_flag((&user->flags_page2), SIP_PAGE2_RTCACHEFRIENDS); + suserobjs++; + ASTOBJ_CONTAINER_LINK(&userl,user); + } else { + /* Move counter from s to r... */ + suserobjs--; + ruserobjs++; + ast_set_flag(user, SIP_REALTIME); + } + ast_variables_destroy(var); + return user; +} + +/*--- find_user: 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 = NULL; + u = ASTOBJ_CONTAINER_FIND(&userl,name); + if (!u && realtime) { + u = realtime_user(name); + } + return u; +} + +/*--- create_addr_from_peer: create address structure from peer reference ---*/ +static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer) +{ + char *callhost; + + if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) && + (!peer->maxms || ((peer->lastms >= 0) && (peer->lastms <= peer->maxms)))) { + if (peer->addr.sin_addr.s_addr) { + r->sa.sin_addr = peer->addr.sin_addr; + r->sa.sin_port = peer->addr.sin_port; + } else { + r->sa.sin_addr = peer->defaddr.sin_addr; + r->sa.sin_port = peer->defaddr.sin_port; + } + memcpy(&r->recv, &r->sa, sizeof(r->recv)); + } else { + return -1; + } + + ast_copy_flags(r, peer, + SIP_PROMISCREDIR | SIP_USEREQPHONE | SIP_DTMF | SIP_NAT | SIP_REINVITE | + SIP_INSECURE_PORT | SIP_INSECURE_INVITE); + r->capability = peer->capability; + if (r->rtp) { + ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", (ast_test_flag(r, SIP_NAT) & SIP_NAT_ROUTE)); + ast_rtp_setnat(r->rtp, (ast_test_flag(r, SIP_NAT) & SIP_NAT_ROUTE)); + } + if (r->vrtp) { + ast_log(LOG_DEBUG, "Setting NAT on VRTP to %d\n", (ast_test_flag(r, SIP_NAT) & SIP_NAT_ROUTE)); + ast_rtp_setnat(r->vrtp, (ast_test_flag(r, SIP_NAT) & SIP_NAT_ROUTE)); + } + ast_copy_string(r->peername, peer->username, sizeof(r->peername)); + ast_copy_string(r->authname, peer->username, sizeof(r->authname)); + ast_copy_string(r->username, peer->username, sizeof(r->username)); + ast_copy_string(r->peersecret, peer->secret, sizeof(r->peersecret)); + ast_copy_string(r->peermd5secret, peer->md5secret, sizeof(r->peermd5secret)); + ast_copy_string(r->tohost, peer->tohost, sizeof(r->tohost)); + ast_copy_string(r->fullcontact, peer->fullcontact, sizeof(r->fullcontact)); +#ifdef SIP_TCP_SUPPORT + r->sockfd = peer->tcpsockfd; + ast_copy_string(r->transport, peer->transport, sizeof(r->transport)); +#endif + + if (!r->initreq.headers && !ast_strlen_zero(peer->fromdomain)) { + if ((callhost = strchr(r->callid, '@'))) { + strncpy(callhost + 1, peer->fromdomain, sizeof(r->callid) - (callhost - r->callid) - 2); + } + } + if (ast_strlen_zero(r->tohost)) { + if (peer->addr.sin_addr.s_addr) + ast_inet_ntoa(r->tohost, sizeof(r->tohost), peer->addr.sin_addr); + else + ast_inet_ntoa(r->tohost, sizeof(r->tohost), peer->defaddr.sin_addr); + } + if (!ast_strlen_zero(peer->fromdomain)) + ast_copy_string(r->fromdomain, peer->fromdomain, sizeof(r->fromdomain)); + if (!ast_strlen_zero(peer->fromuser)) + ast_copy_string(r->fromuser, peer->fromuser, sizeof(r->fromuser)); + r->maxtime = peer->maxms; + r->callgroup = peer->callgroup; + r->pickupgroup = peer->pickupgroup; + if (ast_test_flag(r, SIP_DTMF) == SIP_DTMF_RFC2833) + r->noncodeccapability |= AST_RTP_DTMF; + else + r->noncodeccapability &= ~AST_RTP_DTMF; + ast_copy_string(r->context, peer->context,sizeof(r->context)); + r->rtptimeout = peer->rtptimeout; + r->rtpholdtimeout = peer->rtpholdtimeout; + r->rtpkeepalive = peer->rtpkeepalive; + + return 0; +} + +/*--- create_addr: 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 *r, char *opeer) +{ + struct hostent *hp; + struct ast_hostent ahp; + struct sip_peer *p; + int found=0; + 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'; + port++; + } + r->sa.sin_family = AF_INET; + p = find_peer(peer, NULL, 1); + + if (p) { + found++; + if (create_addr_from_peer(r, p)) + ASTOBJ_UNREF(p, sip_destroy_peer); + } + if (!p) { + if (found) + return -1; + + hostn = peer; + if (port) + portno = atoi(port); + else + portno = DEFAULT_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_copy_string(r->tohost, peer, sizeof(r->tohost)); + memcpy(&r->sa.sin_addr, hp->h_addr, sizeof(r->sa.sin_addr)); + r->sa.sin_port = htons(portno); + memcpy(&r->recv, &r->sa, sizeof(r->recv)); + return 0; + } else { + ast_log(LOG_WARNING, "No such host: %s\n", peer); + return -1; + } + } else { + ASTOBJ_UNREF(p, sip_destroy_peer); + return 0; + } +} + +/*--- auto_congest: Scheduled congestion on a call ---*/ +static int auto_congest(void *nothing) +{ + struct sip_pvt *p = nothing; + ast_mutex_lock(&p->lock); + p->initid = -1; + if (p->owner) { + if (!ast_mutex_trylock(&p->owner->lock)) { + ast_log(LOG_NOTICE, "Auto-congesting %s\n", p->owner->name); + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_mutex_unlock(&p->owner->lock); + } + } + ast_mutex_unlock(&p->lock); + return 0; +} + + + + +/*--- sip_call: Initiate SIP call from PBX ---*/ +/* used from the dial() application */ +static int sip_call(struct ast_channel *ast, char *dest, int timeout) +{ + int res; + struct sip_pvt *p; +#ifdef OSP_SUPPORT + char *osphandle = NULL; +#endif + struct varshead *headp; + struct ast_var_t *current; + struct sip_invite_param options; + + memset(&options, 0, sizeof(struct sip_invite_param)); + + 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 (!options.vxml_url && !strcasecmp(ast_var_name(current),"VXML_URL")) { + options.vxml_url = ast_var_value(current); + } else if (!options.distinctive_ring && !strcasecmp(ast_var_name(current),"ALERT_INFO")) { + /* Check whether there is a ALERT_INFO variable */ + options.distinctive_ring = ast_var_value(current); + } else if (!options.addsipheaders && !strncasecmp(ast_var_name(current),"SIPADDHEADER",strlen("SIPADDHEADER"))) { + /* Check whether there is a variable with a name starting with SIPADDHEADER */ + options.addsipheaders = 1; + } + + +#ifdef OSP_SUPPORT + else if (!options.osptoken && !strcasecmp(ast_var_name(current), "OSPTOKEN")) { + options.osptoken = ast_var_value(current); + } else if (!osphandle && !strcasecmp(ast_var_name(current), "OSPHANDLE")) { + osphandle = ast_var_value(current); + } +#endif + } + + res = 0; + ast_set_flag(p, SIP_OUTGOING); +#ifdef OSP_SUPPORT + if (!options.osptoken || !osphandle || (sscanf(osphandle, "%d", &p->osphandle) != 1)) { + /* Force Disable OSP support */ + ast_log(LOG_DEBUG, "Disabling OSP support for this call. osptoken = %s, osphandle = %s\n", options.osptoken, osphandle); + options.osptoken = NULL; + osphandle = NULL; + p->osphandle = -1; + } +#endif + ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username); + res = update_user_counter(p,INC_OUT_USE); + if ( res != -1 ) { + p->callingpres = ast->cid.cid_pres; + p->jointcapability = p->capability; + transmit_invite(p, SIP_INVITE, 1, &options, 1); + if (p->maxtime) { + /* Initialize auto-congest time */ + p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p); + } + } + return res; +} + +/*--- sip_registry_destroy: Destroy registry object ---*/ +/* Objects created with the register= statement in static configuration */ +static void sip_registry_destroy(struct sip_registry *reg) +{ + /* Really delete */ + if (reg->call) { + /* Clear registry before destroying to ensure + we don't get reentered trying to grab the registry lock */ + reg->call->registry = NULL; + sip_destroy(reg->call); + } + if (reg->expire > -1) + ast_sched_del(sched, reg->expire); + if (reg->timeout > -1) + ast_sched_del(sched, reg->timeout); + regobjs--; + free(reg); + +} + +/*--- __sip_destroy: Execute destrucion of call structure, release memory---*/ +static void __sip_destroy(struct sip_pvt *p, int lockowner) +{ + struct sip_pvt *cur, *prev = NULL; + struct sip_pkt *cp; + struct sip_history *hist; + + if (sip_debug_test_pvt(p)) + ast_verbose("Destroying call '%s'\n", p->callid); + if (p->stateid > -1) + ast_extension_state_del(p->stateid, NULL); + if (p->initid > -1) + ast_sched_del(sched, p->initid); + if (p->autokillid > -1) + ast_sched_del(sched, p->autokillid); + + if (p->rtp) { + ast_rtp_destroy(p->rtp); + } + if (p->vrtp) { + ast_rtp_destroy(p->vrtp); + } + if (p->route) { + free_old_route(p->route); + p->route = NULL; + } +/*#ifdef SIP_TCP_SUPPORT + if ((p->sockfd > -1) && (p->sockfd != sipsock)) { + close(p->sockfd); + p->sockfd = -1; + } +#endif*/ + if (p->registry) { + if (p->registry->call == p) + p->registry->call = NULL; + ASTOBJ_UNREF(p->registry,sip_registry_destroy); + } + /* Unlink us from the owner if we have one */ + if (p->owner) { + if (lockowner) + ast_mutex_lock(&p->owner->lock); + ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name); + p->owner->tech_pvt = NULL; + if (lockowner) + ast_mutex_unlock(&p->owner->lock); + } + /* Clear history */ + while(p->history) { + hist = p->history; + p->history = p->history->next; + free(hist); + } + cur = iflist; + while(cur) { + if (cur == p) { + if (prev) + prev->next = cur->next; + else + iflist = cur->next; + break; + } + prev = cur; + cur = cur->next; + } + if (!cur) { + ast_log(LOG_WARNING, "Trying to destroy \"%s\", not found in dialog list?!?! \n", p->callid); + return; + } + if (p->initid > -1) + ast_sched_del(sched, p->initid); + while((cp = p->packets)) { + p->packets = p->packets->next; + if (cp->retransid > -1) + ast_sched_del(sched, cp->retransid); + free(cp); + } + ast_mutex_destroy(&p->lock); + if (p->chanvars) { + ast_variables_destroy(p->chanvars); + p->chanvars = NULL; + } + free(p); +} + +/*--- update_user_counter: Handle incominglimit and outgoinglimit for SIP users ---*/ +/* Note: This is going to be replaced by app_groupcount */ +/* Thought: For realtime, we should propably update storage with inuse counter... */ +static int update_user_counter(struct sip_pvt *fup, int event) +{ + char name[256] = ""; + struct sip_user *u; + struct sip_peer *p; + int *inuse, *incominglimit; + + /* 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, SIP_CALL_LIMIT)) + return 0; + + ast_copy_string(name, fup->username, sizeof(name)); + + /* Check the list of users */ + u = find_user(name, 1); + if (u) { + inuse = &u->inUse; + incominglimit = &u->incominglimit; + p = NULL; + } else { + /* Try to find peer */ + p = find_peer(fup->peername, NULL, 1); + if (p) { + inuse = &p->inUse; + incominglimit = &p->incominglimit; + ast_copy_string(name, fup->peername, sizeof(name)); + } else { + if (option_debug > 1) + ast_log(LOG_DEBUG, "%s is not a local user, no call limit\n", name); + return 0; + } + } + switch(event) { + /* incoming and outgoing affects the inUse counter */ + case DEC_OUT_USE: + case DEC_IN_USE: + if ( *inuse > 0 ) { + (*inuse)--; + } else { + *inuse = 0; + } + break; + case INC_IN_USE: + case INC_OUT_USE: + if (*incominglimit > 0 ) { + if (*inuse >= *incominglimit) { + ast_log(LOG_ERROR, "Call from %s '%s' rejected due to usage limit of %d\n", u?"user":"peer", name, *incominglimit); + /* inc inUse as well */ + if ( event == INC_OUT_USE ) { + (*inuse)++; + } + if (u) + ASTOBJ_UNREF(u,sip_destroy_user); + else + ASTOBJ_UNREF(p,sip_destroy_peer); + return -1; + } + } + (*inuse)++; + ast_log(LOG_DEBUG, "Call from %s '%s' is %d out of %d\n", u?"user":"peer", name, *inuse, *incominglimit); + break; +#ifdef DISABLED_CODE + /* we don't use these anymore */ + case DEC_OUT_USE: + if ( u->outUse > 0 ) { + u->outUse--; + } else { + u->outUse = 0; + } + break; + case INC_OUT_USE: + if ( u->outgoinglimit > 0 ) { + if ( u->outUse >= u->outgoinglimit ) { + ast_log(LOG_ERROR, "Outgoing call from user '%s' rejected due to usage limit of %d\n", u->name, u->outgoinglimit); + ast_mutex_unlock(&userl.lock); + if (u->temponly) { + destroy_user(u); + } + return -1; + } + } + u->outUse++; + break; +#endif + default: + ast_log(LOG_ERROR, "update_user_counter(%s,%d) called with no event!\n",name,event); + } + if (u) + ASTOBJ_UNREF(u,sip_destroy_user); + else + ASTOBJ_UNREF(p,sip_destroy_peer); + return 0; +} + +/*--- sip_destroy: Destroy SIP call structure ---*/ +static void sip_destroy(struct sip_pvt *p) +{ + ast_mutex_lock(&iflock); + __sip_destroy(p, 1); + ast_mutex_unlock(&iflock); +} + + +static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req, int fatal); + +/*--- hangup_sip2cause: Convert SIP hangup causes to Asterisk hangup causes ---*/ +static int hangup_sip2cause(int cause) +{ +/* Possible values taken from causes.h */ + + switch(cause) { + case 403: /* Not found */ + return AST_CAUSE_CALL_REJECTED; + case 404: /* Not found */ + return AST_CAUSE_UNALLOCATED; + case 408: /* No reaction */ + return AST_CAUSE_NO_USER_RESPONSE; + case 480: /* No answer */ + return AST_CAUSE_FAILURE; + case 483: /* Too many hops */ + return AST_CAUSE_NO_ANSWER; + case 486: /* Busy everywhere */ + return AST_CAUSE_BUSY; + case 488: /* No codecs approved */ + return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; + 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; + default: + return AST_CAUSE_NORMAL; + } + /* Never reached */ + return 0; +} + + +/*--- hangup_cause2sip: Convert Asterisk hangup causes to SIP codes ---*/ +/* 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 +*/ +static 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 */ + 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: + ast_log(LOG_DEBUG, "AST hangup cause %d (no match found in SIP)\n", cause); + return NULL; + } + + /* Never reached */ + return 0; +} + + +/*--- sip_hangup: Hangup SIP call ---*/ +/* Part of PBX interface */ +static int sip_hangup(struct ast_channel *ast) +{ + struct sip_pvt *p = ast->tech_pvt; + int needcancel = 0; + struct ast_flags locflags = {0}; + + if (option_debug) + ast_log(LOG_DEBUG, "sip_hangup(%s)\n", ast->name); + if (!p) { + ast_log(LOG_DEBUG, "Asked to hangup channel not connected\n"); + return 0; + } + ast_mutex_lock(&p->lock); +#ifdef OSP_SUPPORT + if ((p->osphandle > -1) && (ast->_state == AST_STATE_UP)) { + ast_osp_terminate(p->osphandle, AST_CAUSE_NORMAL, p->ospstart, time(NULL) - p->ospstart); + } +#endif + if (ast_test_flag(p, SIP_OUTGOING)) { + ast_log(LOG_DEBUG, "update_user_counter(%s) - decrement outUse counter\n", p->username); + update_user_counter(p, DEC_OUT_USE); + } else { + ast_log(LOG_DEBUG, "update_user_counter(%s) - decrement inUse counter\n", p->username); + update_user_counter(p, DEC_IN_USE); + } + /* Determine how to disconnect */ + if (p->owner != ast) { + ast_log(LOG_WARNING, "Huh? We aren't the owner?\n"); + ast_mutex_unlock(&p->lock); + return 0; + } + if (ast->_state != AST_STATE_UP) + needcancel = 1; + /* Disconnect */ + p = ast->tech_pvt; + if (p->vad) { + ast_dsp_free(p->vad); + } + p->owner = NULL; + ast->tech_pvt = NULL; + + ast_mutex_lock(&usecnt_lock); + usecnt--; + ast_mutex_unlock(&usecnt_lock); + ast_update_use_count(); + + ast_set_flag(&locflags, SIP_NEEDDESTROY); + /* Start the process if it's not already started */ + if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) { + if (needcancel) { + if (ast_test_flag(p, SIP_OUTGOING)) { + transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0); + /* 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. */ + ast_clear_flag(&locflags, SIP_NEEDDESTROY); + sip_scheddestroy(p, 15000); + if ( p->initid != -1 ) { + /* channel still up - reverse dec of inUse counter + only if the channel is not auto-congested */ + if (ast_test_flag(p, SIP_OUTGOING)) { + update_user_counter(p, INC_OUT_USE); + } + else { + update_user_counter(p, INC_IN_USE); + } + } + } else { + char *res; + if (ast->hangupcause && ((res = hangup_cause2sip(ast->hangupcause)))) { + transmit_response_reliable(p, res, &p->initreq, 1); + } else + transmit_response_reliable(p, "403 Forbidden", &p->initreq, 1); + } + } else { + if (!p->pendinginvite) { + /* Send a hangup */ + transmit_request_with_auth(p, SIP_BYE, 0, 1, 1); + } 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, SIP_PENDINGBYE); + ast_clear_flag(p, SIP_NEEDREINVITE); + } + } + } + ast_copy_flags(p, (&locflags), SIP_NEEDDESTROY); + ast_mutex_unlock(&p->lock); + return 0; +} + +/*--- 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,fmt; + char *codec; + struct sip_pvt *p = ast->tech_pvt; + + ast_mutex_lock(&p->lock); + if (ast->_state != AST_STATE_UP) { +#ifdef OSP_SUPPORT + time(&p->ospstart); +#endif + + codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC"); + if (codec) { + 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); + } + + ast_setstate(ast, AST_STATE_UP); + if (option_debug) + ast_log(LOG_DEBUG, "sip_answer(%s)\n", ast->name); + res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 1); + } + ast_mutex_unlock(&p->lock); + return res; +} + +/*--- sip_write: 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)) { + ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %d (read/write = %d/%d)\n", + frame->subclass, ast->nativeformats, ast->readformat, 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, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) { + transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0); + ast_set_flag(p, SIP_PROGRESS_SENT); + } + time(&p->lastrtptx); + 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, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) { + transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0); + ast_set_flag(p, SIP_PROGRESS_SENT); + } + time(&p->lastrtptx); + res = ast_rtp_write(p->vrtp, frame); + } + ast_mutex_unlock(&p->lock); + } + break; + case AST_FRAME_IMAGE: + return 0; + break; + default: + ast_log(LOG_WARNING, "Can't send %d type frames with SIP write\n", frame->frametype); + return 0; + } + + return res; +} + +/*--- 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) +{ + struct sip_pvt *p = newchan->tech_pvt; + ast_mutex_lock(&p->lock); + if (p->owner != oldchan) { + ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner); + ast_mutex_unlock(&p->lock); + return -1; + } + p->owner = newchan; + ast_mutex_unlock(&p->lock); + return 0; +} + +/*--- sip_senddigit: Send DTMF character on SIP channel */ +/* within one call, we're able to transmit in many methods simultaneously */ +static int sip_senddigit(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, SIP_DTMF)) { + case SIP_DTMF_INFO: + transmit_info_with_digit(p, digit); + break; + case SIP_DTMF_RFC2833: + if (p->rtp) + ast_rtp_senddigit(p->rtp, digit); + break; + case SIP_DTMF_INBAND: + res = -1; + break; + } + ast_mutex_unlock(&p->lock); + return res; +} + +#define DEFAULT_MAX_FORWARDS 70 + + +/*--- sip_transfer: Transfer SIP call */ +static int sip_transfer(struct ast_channel *ast, const char *dest) +{ + struct sip_pvt *p = ast->tech_pvt; + int res; + + 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; +} + +/*--- sip_indicate: Play indication to user */ +/* With SIP a lot of indications is sent as messages, letting the device play + the indication - busy signal, congestion etc */ +static int sip_indicate(struct ast_channel *ast, int condition) +{ + 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) { + if (!ast_test_flag(p, SIP_PROGRESS_SENT) || + (ast_test_flag(p, 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, SIP_RINGING); + if (ast_test_flag(p, 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(p, "486 Busy Here", &p->initreq); + ast_set_flag(p, SIP_ALREADYGONE); + ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV); + break; + } + res = -1; + break; + case AST_CONTROL_CONGESTION: + if (ast->_state != AST_STATE_UP) { + transmit_response(p, "503 Service Unavailable", &p->initreq); + ast_set_flag(p, SIP_ALREADYGONE); + ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV); + break; + } + res = -1; + break; + case AST_CONTROL_PROGRESS: + case AST_CONTROL_PROCEEDING: + if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) { + transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0); + ast_set_flag(p, SIP_PROGRESS_SENT); + break; + } + res = -1; + break; + case AST_CONTROL_HOLD: /* We are put on hold */ + /* The PBX is providing us with onhold music, but + should we clear the RTP stream with the other + end? Guess we could do that if there's no + musiconhold class defined for this channel + */ + if (sipdebug) + ast_log(LOG_DEBUG, "SIP dialog on hold: %s\n", p->callid); + res = -1; + ast_set_flag(p, SIP_CALL_ONHOLD); + break; + case AST_CONTROL_UNHOLD: /* We are back from hold */ + /* Open RTP stream if we decide to close it + */ + if (sipdebug) + ast_log(LOG_DEBUG, "SIP dialog off hold: %s\n", p->callid); + res = -1; + ast_clear_flag(p, SIP_CALL_ONHOLD); + 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; +} + + + +/*--- sip_new: Initiate a call in the SIP channel */ +/* called from sip_request (calls from the pbx ) */ +static struct ast_channel *sip_new(struct sip_pvt *i, int state, char *title) +{ + struct ast_channel *tmp; + struct ast_variable *v = NULL; + int fmt; + + ast_mutex_unlock(&i->lock); + /* Don't hold a sip pvt lock while we allocate a channel */ + tmp = ast_channel_alloc(1); + ast_mutex_lock(&i->lock); + if (!tmp) { + ast_log(LOG_WARNING, "Unable to allocate SIP channel structure\n"); + return NULL; + } + tmp->tech = &sip_tech; + /* Select our native format based on codec preference until we receive + something from another device to the contrary. */ + ast_mutex_lock(&i->lock); + if (i->jointcapability) + tmp->nativeformats = ast_codec_choose(&i->prefs, i->jointcapability, 1); + else if (i->capability) + tmp->nativeformats = ast_codec_choose(&i->prefs, i->capability, 1); + else + tmp->nativeformats = ast_codec_choose(&i->prefs, global_capability, 1); + ast_mutex_unlock(&i->lock); + fmt = ast_best_codec(tmp->nativeformats); + + if (title) + snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%04x", title, rand() & 0xffff); + else if (strchr(i->fromdomain,':')) + snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%08x", strchr(i->fromdomain,':')+1, (int)(long)(i)); + else + snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%08x", i->fromdomain, (int)(long)(i)); + + tmp->type = channeltype; + if (ast_test_flag(i, SIP_DTMF) == SIP_DTMF_INBAND) { + i->vad = ast_dsp_new(); + ast_dsp_set_features(i->vad, DSP_FEATURE_DTMF_DETECT); + if (relaxdtmf) + ast_dsp_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF); + } + tmp->fds[0] = ast_rtp_fd(i->rtp); + tmp->fds[1] = ast_rtcp_fd(i->rtp); + if (i->vrtp) { + tmp->fds[2] = ast_rtp_fd(i->vrtp); + tmp->fds[3] = ast_rtcp_fd(i->vrtp); + } + 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_copy_string(tmp->accountcode, i->accountcode, sizeof(tmp->accountcode)); + if (i->amaflags) + tmp->amaflags = i->amaflags; + if (!ast_strlen_zero(i->language)) + ast_copy_string(tmp->language, i->language, sizeof(tmp->language)); + if (!ast_strlen_zero(i->musicclass)) + ast_copy_string(tmp->musicclass, i->musicclass, sizeof(tmp->musicclass)); + i->owner = tmp; + ast_mutex_lock(&usecnt_lock); + usecnt++; + ast_mutex_unlock(&usecnt_lock); + ast_copy_string(tmp->context, i->context, sizeof(tmp->context)); + ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten)); + if (!ast_strlen_zero(i->cid_num)) + tmp->cid.cid_num = strdup(i->cid_num); + if (!ast_strlen_zero(i->cid_name)) + tmp->cid.cid_name = strdup(i->cid_name); + if (!ast_strlen_zero(i->rdnis)) + tmp->cid.cid_rdnis = strdup(i->rdnis); + if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s")) + tmp->cid.cid_dnid = 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); + } + ast_setstate(tmp, state); + if (state != AST_STATE_DOWN) { + if (ast_pbx_start(tmp)) { + ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name); + ast_hangup(tmp); + tmp = NULL; + } + } + /* 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); + + return tmp; +} + +/*--- get_sdp_by_line: Reads one line of SIP message body */ +static char* get_sdp_by_line(char* line, char *name, int nameLen) +{ + if (strncasecmp(line, name, nameLen) == 0 && line[nameLen] == '=') { + return ast_skip_blanks(line + nameLen + 1); + } + return ""; +} + +/*--- get_sdp: Gets all kind of SIP message bodies, including SDP, + but the name wrongly applies _only_ sdp */ +static char *get_sdp(struct sip_request *req, char *name) +{ + int x; + int len = strlen(name); + char *r; + + for (x=0; xlines; x++) { + r = get_sdp_by_line(req->line[x], name, len); + if (r[0] != '\0') + return r; + } + return ""; +} + + +static void sdpLineNum_iterator_init(int* iterator) +{ + *iterator = 0; +} + +static char* get_sdp_iterate(int* iterator, + struct sip_request *req, char *name) +{ + int len = strlen(name); + char *r; + + while (*iterator < req->lines) { + r = get_sdp_by_line(req->line[(*iterator)++], name, len); + if (r[0] != '\0') + return r; + } + return ""; +} + +static char *find_alias(const char *name, char *_default) +{ + int x; + for (x=0;xheaders; 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 ""; +} + +/*--- get_header: Get header from SIP request ---*/ +static char *get_header(struct sip_request *req, char *name) +{ + int start = 0; + return __get_header(req, name, &start); +} + +/*--- sip_rtp_read: Read RTP from network ---*/ +static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p) +{ + /* Retrieve audio/etc from channel. Assumes p->lock is already held. */ + struct ast_frame *f; + static struct ast_frame null_frame = { AST_FRAME_NULL, }; + 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; + default: + f = &null_frame; + } + /* Don't forward RFC2833 if we're not supposed to */ + if (f && (f->frametype == AST_FRAME_DTMF) && (ast_test_flag(p, SIP_DTMF) != SIP_DTMF_RFC2833)) + return &null_frame; + if (p->owner) { + /* We already hold the channel lock */ + if (f->frametype == AST_FRAME_VOICE) { + if (f->subclass != p->owner->nativeformats) { + ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass); + p->owner->nativeformats = f->subclass; + ast_set_read_format(p->owner, p->owner->readformat); + ast_set_write_format(p->owner, p->owner->writeformat); + } + if ((ast_test_flag(p, SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) { + f = ast_dsp_process(p->owner, p->vad, f); + if (f && (f->frametype == AST_FRAME_DTMF)) + ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass); + } + } + } + return f; +} + +/*--- sip_read: 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; + ast_mutex_lock(&p->lock); + fr = sip_rtp_read(ast, p); + time(&p->lastrtprx); + ast_mutex_unlock(&p->lock); + return fr; +} + +/*--- build_callid: Build SIP CALLID header ---*/ +static void build_callid(char *callid, int len, struct in_addr ourip, char *fromdomain) +{ + int res; + int val; + int x; + char iabuf[INET_ADDRSTRLEN]; + for (x=0; x<4; x++) { + val = rand(); + res = snprintf(callid, len, "%08x", val); + len -= res; + callid += res; + } + if (!ast_strlen_zero(fromdomain)) + snprintf(callid, len, "@%s", fromdomain); + else + /* It's not important that we really use our right IP here... */ + snprintf(callid, len, "@%s", ast_inet_ntoa(iabuf, sizeof(iabuf), ourip)); +} + +/*--- sip_alloc: Allocate SIP_PVT structure and set defaults ---*/ +static struct sip_pvt *sip_alloc(char *callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method) +{ + struct sip_pvt *p; + + p = malloc(sizeof(struct sip_pvt)); + if (!p) + return NULL; + /* Keep track of stuff */ + memset(p, 0, sizeof(struct sip_pvt)); + ast_mutex_init(&p->lock); + + p->method = intended_method; + p->initid = -1; + p->autokillid = -1; + p->stateid = -1; + p->prefs = prefs; +#ifdef OSP_SUPPORT + p->osphandle = -1; +#endif + if (sin) { + memcpy(&p->sa, sin, sizeof(p->sa)); + if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip)) + memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); + } else { + memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); + } +#ifdef SIP_TCP_SUPPORT + p->ssl = NULL; + p->sockfd = -1; + ast_copy_string(p->transport, "UDP", sizeof(p->transport)); /* default transport protocol */ +#endif + + p->branch = rand(); + p->tag = rand(); + /* Start with 101 instead of 1 */ + p->ocseq = 101; + + if (sip_methods[intended_method].need_rtp) { + p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr); + if (videosupport) + p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr); + if (!p->rtp) { + ast_log(LOG_WARNING, "Unable to create RTP session: %s\n", strerror(errno)); + ast_mutex_destroy(&p->lock); + if (p->chanvars) { + ast_variables_destroy(p->chanvars); + p->chanvars = NULL; + } + free(p); + return NULL; + } + ast_rtp_settos(p->rtp, tos); + if (p->vrtp) + ast_rtp_settos(p->vrtp, tos); + p->rtptimeout = global_rtptimeout; + p->rtpholdtimeout = global_rtpholdtimeout; + p->rtpkeepalive = global_rtpkeepalive; + } + + if (useglobal_nat && sin) { + /* Setup NAT structure according to global settings if we have an address */ + ast_copy_flags(p, &global_flags, SIP_NAT); + memcpy(&p->recv, sin, sizeof(p->recv)); + if (p->rtp) + ast_rtp_setnat(p->rtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); + if (p->vrtp) + ast_rtp_setnat(p->vrtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); + } + + if (p->method != SIP_REGISTER) + ast_copy_string(p->fromdomain, default_fromdomain, sizeof(p->fromdomain)); + build_via(p, p->via, sizeof(p->via)); + if (!callid) + build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); + else + ast_copy_string(p->callid, callid, sizeof(p->callid)); + ast_copy_flags(p, (&global_flags), SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_DTMF | SIP_REINVITE | SIP_PROG_INBAND | SIP_OSPAUTH); + /* Assign default music on hold class */ + strcpy(p->musicclass, global_musicclass); + p->capability = global_capability; + if (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) + p->noncodeccapability |= AST_RTP_DTMF; + strcpy(p->context, default_context); + + /* 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; +} + +/*--- find_call: 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; + char *callid; + char tmp[256] = ""; + char iabuf[INET_ADDRSTRLEN]; + char *cmd; + char *tag = "", *c; + + callid = get_header(req, "Call-ID"); + + if (pedanticsipchecking) { + /* In principle Call-ID's uniquely identify a call, however some vendors + (i.e. Pingtel) send multiple calls with the same Call-ID and different + tags in order to simplify billing. The RFC does state that we have to + compare tags in addition to the call-id, but this generate substantially + more overhead which is totally unnecessary for the vast majority of sane + SIP implementations, and thus Asterisk does not enable this behavior + by default. Short version: You'll need this option to support conferencing + on the pingtel */ + ast_copy_string(tmp, req->header[0], sizeof(tmp)); + cmd = tmp; + c = strchr(tmp, ' '); + if (c) + *c = '\0'; + if (!strcasecmp(cmd, "SIP/2.0")) + ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp)); + else + ast_copy_string(tmp, get_header(req, "From"), sizeof(tmp)); + tag = strcasestr(tmp, "tag="); + if (tag) { + tag += 4; + c = strchr(tag, ';'); + if (c) + *c = '\0'; + } + + } + + if (ast_strlen_zero(callid)) { + ast_log(LOG_WARNING, "Call missing call ID from '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr)); + return NULL; + } + ast_mutex_lock(&iflock); + p = iflist; + while(p) { + if (!strcmp(p->callid, callid) && + (!pedanticsipchecking || !tag || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) { + /* Found the call */ + ast_mutex_lock(&p->lock); + ast_mutex_unlock(&iflock); + return p; + } + p = p->next; + } + ast_mutex_unlock(&iflock); + p = sip_alloc(callid, sin, 1, intended_method); + if (p) + ast_mutex_lock(&p->lock); + return p; +} + +/*--- sip_register: Parse register=> line in sip.conf and add to registry */ +static int sip_register(char *value, int lineno) +{ + struct sip_registry *reg; + char copy[256] = ""; + char *username=NULL, *hostname=NULL, *secret=NULL, *authuser=NULL; + char *porta=NULL; + char *contact=NULL; + char *stringp=NULL; + + if (!value) + return -1; + ast_copy_string(copy, value, sizeof(copy)); + stringp=copy; + username = stringp; + hostname = strrchr(stringp, '@'); + if (hostname) { + *hostname = '\0'; + hostname++; + } + if (!username || ast_strlen_zero(username) || !hostname || 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; + } + stringp=username; + username = strsep(&stringp, ":"); + if (username) { + secret = strsep(&stringp, ":"); + if (secret) + authuser = strsep(&stringp, ":"); + } + stringp = hostname; + hostname = strsep(&stringp, "/"); + if (hostname) + contact = strsep(&stringp, "/"); + if (!contact || ast_strlen_zero(contact)) + contact = "s"; + stringp=hostname; + hostname = strsep(&stringp, ":"); + porta = strsep(&stringp, ":"); + + if (porta && !atoi(porta)) { + ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno); + return -1; + } + reg = malloc(sizeof(struct sip_registry)); + if (!reg) { + ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n"); + return -1; + } + memset(reg, 0, sizeof(struct sip_registry)); + regobjs++; + ASTOBJ_INIT(reg); + ast_copy_string(reg->contact, contact, sizeof(reg->contact)); + if (username) + ast_copy_string(reg->username, username, sizeof(reg->username)); + if (hostname) + ast_copy_string(reg->hostname, hostname, sizeof(reg->hostname)); + if (authuser) + ast_copy_string(reg->authuser, authuser, sizeof(reg->authuser)); + if (secret) + ast_copy_string(reg->secret, secret, sizeof(reg->secret)); + reg->expire = -1; + reg->timeout = -1; + reg->refresh = default_expiry; + reg->portno = porta ? atoi(porta) : 0; + reg->callid_valid = 0; + reg->ocseq = 101; + ASTOBJ_CONTAINER_LINK(®l, reg); + ASTOBJ_UNREF(reg,sip_registry_destroy); + return 0; +} + +/*--- lws2sws: 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; +} + +/*--- parse: Parse a SIP message ----*/ +static void parse(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 0 + printf("Header: %s (%d)\n", req->header[f], strlen(req->header[f])); +#endif + 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...\n"); + } else + f++; + req->header[f] = c + 1; + } else if (*c == '\r') { + /* Ignore but eliminate \r's */ + *c = 0; + } + c++; + } + /* Check for last header */ + if (!ast_strlen_zero(req->header[f])) + f++; + req->headers = f; + /* Now we process any mime content */ + f = 0; + req->line[f] = c; + while(*c) { + if (*c == '\n') { + /* We've got a new line */ + *c = 0; +#if 0 + printf("Line: %s (%d)\n", req->line[f], strlen(req->line[f])); +#endif + if (f >= SIP_MAX_LINES - 1) { + ast_log(LOG_WARNING, "Too many SDP lines...\n"); + } else + f++; + req->line[f] = c + 1; + } else if (*c == '\r') { + /* Ignore and eliminate \r's */ + *c = 0; + } + c++; + } + /* Check for last line */ + if (!ast_strlen_zero(req->line[f])) + f++; + req->lines = f; + if (*c) + ast_log(LOG_WARNING, "Odd content, extra stuff left over ('%s')\n", c); +} + +/*--- process_sdp: Process SIP SDP and activate RTP channels---*/ +static int process_sdp(struct sip_pvt *p, struct sip_request *req) +{ + char *m; + char *c; + char *a; + char host[258]; + char iabuf[INET_ADDRSTRLEN]; + int len = -1; + int portno = -1; + int vportno = -1; + int peercapability, peernoncodeccapability; + int vpeercapability=0, vpeernoncodeccapability=0; + struct sockaddr_in sin; + char *codecs; + struct hostent *hp; + struct ast_hostent ahp; + int codec; + int destiterator = 0; + int iterator; + int sendonly = 0; + int x,y; + int debug=sip_debug_test_pvt(p); + struct ast_channel *bridgepeer = NULL; + + /* Update our last rtprx when we receive an SDP, too */ + time(&p->lastrtprx); + time(&p->lastrtptx); + + /* Get codec and RTP info from SDP */ + if (strcasecmp(get_header(req, "Content-Type"), "application/sdp")) { + ast_log(LOG_NOTICE, "Content is '%s', not 'application/sdp'\n", get_header(req, "Content-Type")); + return -1; + } + m = get_sdp(req, "m"); + sdpLineNum_iterator_init(&destiterator); + 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; + } + 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, &ahp); + if (!hp) { + ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c); + return -1; + } + sdpLineNum_iterator_init(&iterator); + ast_set_flag(p, SIP_NOVIDEO); + while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') { + int found = 0; + if ((sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1) || + (sscanf(m, "audio %d/%d RTP/AVP %n", &x, &y, &len) == 2)) { + found = 1; + portno = x; + /* Scan through the RTP payload types specified in a "m=" line: */ + ast_rtp_pt_clear(p->rtp); + codecs = m + len; + while(!ast_strlen_zero(codecs)) { + 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(p->rtp, codec); + codecs = ast_skip_blanks(codecs + len); + } + } + if (p->vrtp) + ast_rtp_pt_clear(p->vrtp); /* Must be cleared in case no m=video line exists */ + + if (p->vrtp && (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) { + found = 1; + ast_clear_flag(p, SIP_NOVIDEO); + vportno = x; + /* Scan through the RTP payload types specified in a "m=" line: */ + codecs = m + len; + while(!ast_strlen_zero(codecs)) { + 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 video format %s\n", ast_getformatname(codec)); + ast_rtp_set_m_type(p->vrtp, codec); + codecs = ast_skip_blanks(codecs + len); + } + } + if (!found ) + ast_log(LOG_WARNING, "Unknown SDP media type in offer: %s\n", m); + } + if (portno == -1 && vportno == -1) { + /* No acceptable offer found in SDP */ + return -2; + } + /* Check for Media-description-level-address for audio */ + if (pedanticsipchecking) { + 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 */ + hp = ast_gethostbyname(host, &ahp); + if (!hp) { + ast_log(LOG_WARNING, "Unable to lookup host in secondary c= line, '%s'\n", c); + } + } + } + } + /* RTP addresses and ports for audio and video */ + sin.sin_family = AF_INET; + memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr)); + + /* Setup audio port number */ + sin.sin_port = htons(portno); + 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(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port)); + ast_log(LOG_DEBUG,"Peer audio RTP is at port %s:%d\n",ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port)); + } + } + /* Check for Media-description-level-address for video */ + if (pedanticsipchecking) { + 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 */ + hp = ast_gethostbyname(host, &ahp); + if (!hp) { + ast_log(LOG_WARNING, "Unable to lookup host in secondary c= line, '%s'\n", c); + } + } + } + } + /* Setup video port number */ + sin.sin_port = htons(vportno); + if (p->vrtp && sin.sin_port) { + ast_rtp_set_peer(p->vrtp, &sin); + if (debug) { + ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port)); + ast_log(LOG_DEBUG,"Peer video RTP is at port %s:%d\n",ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port)); + } + } + + /* Next, scan through each "a=rtpmap:" line, noting each + * specified RTP payload type (with corresponding MIME subtype): + */ + sdpLineNum_iterator_init(&iterator); + while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') { + char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */ + if (!strcasecmp(a, "sendonly")) { + sendonly=1; + continue; + } + if (!strcasecmp(a, "sendrecv")) { + sendonly=0; + } + if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2) continue; + if (debug) + ast_verbose("Found description format %s\n", mimeSubtype); + /* Note: should really look at the 'freq' and '#chans' params too */ + ast_rtp_set_rtpmap_type(p->rtp, codec, "audio", mimeSubtype); + if (p->vrtp) + ast_rtp_set_rtpmap_type(p->vrtp, codec, "video", mimeSubtype); + } + + /* Now gather all of the codecs that were asked for: */ + ast_rtp_get_current_formats(p->rtp, + &peercapability, &peernoncodeccapability); + if (p->vrtp) + ast_rtp_get_current_formats(p->vrtp, + &vpeercapability, &vpeernoncodeccapability); + p->jointcapability = p->capability & (peercapability | vpeercapability); + p->peercapability = (peercapability | vpeercapability); + p->noncodeccapability = noncodeccapability & peernoncodeccapability; + + if (debug) { + /* shame on whoever coded this.... */ + const unsigned slen=512; + char s1[slen], s2[slen], s3[slen], s4[slen]; + + ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n", + ast_getformatname_multiple(s1, slen, p->capability), + ast_getformatname_multiple(s2, slen, peercapability), + ast_getformatname_multiple(s3, slen, vpeercapability), + ast_getformatname_multiple(s4, slen, p->jointcapability)); + + ast_verbose("Non-codec capabilities: us - %s, peer - %s, combined - %s\n", + ast_rtp_lookup_mime_multiple(s1, slen, noncodeccapability, 0), + ast_rtp_lookup_mime_multiple(s2, slen, peernoncodeccapability, 0), + ast_rtp_lookup_mime_multiple(s3, slen, p->noncodeccapability, 0)); + } + if (!p->jointcapability) { + ast_log(LOG_NOTICE, "No compatible codecs!\n"); + return -1; + } + + if (!p->owner) /* There's no open channel owning us */ + return 0; + + if (!(p->owner->nativeformats & p->jointcapability)) { + const unsigned slen=512; + char s1[slen], s2[slen]; + ast_log(LOG_DEBUG, "Oooh, we need to change our formats since our peer supports only %s and not %s\n", + ast_getformatname_multiple(s1, slen, p->jointcapability), + ast_getformatname_multiple(s2, slen, p->owner->nativeformats)); + p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1); + ast_set_read_format(p->owner, p->owner->readformat); + ast_set_write_format(p->owner, p->owner->writeformat); + } + if ((bridgepeer=ast_bridged_channel(p->owner))) { + /* We have a bridge */ + /* Turn on/off music on hold if we are holding/unholding */ + if (sin.sin_addr.s_addr && !sendonly) { + ast_moh_stop(bridgepeer); + /* Indicate UNHOLD status to the other channel */ + ast_indicate(bridgepeer, AST_CONTROL_UNHOLD); + append_history(p, "Unhold", req->data); + if (callevents && ast_test_flag(p, SIP_CALL_ONHOLD)) { + manager_event(EVENT_FLAG_CALL, "Unhold", + "Channel: %s\r\n" + "Uniqueid: %s\r\n", + p->owner->name, + p->owner->uniqueid); + } + ast_clear_flag(p, SIP_CALL_ONHOLD); + /* Somehow, we need to check if we need to re-invite here */ + /* If this call had a external native bridge, it's broken + now and we need to start all over again. + The bridged peer, if SIP, now listens + to RTP from Asterisk instead of from + the peer + + So IF we had a native bridge before + the HOLD, we need to somehow re-invite + into a NATIVE bridge afterwards... + + */ + + } else { + /* No address for RTP, we're on hold */ + append_history(p, "Hold", req->data); + if (callevents && !ast_test_flag(p, SIP_CALL_ONHOLD)) { + manager_event(EVENT_FLAG_CALL, "Hold", + "Channel: %s\r\n" + "Uniqueid: %s\r\n", + p->owner->name, + p->owner->uniqueid); + } + ast_set_flag(p, SIP_CALL_ONHOLD); + /* Indicate HOLD status to the other channel */ + ast_indicate(bridgepeer, AST_CONTROL_HOLD); + ast_moh_start(bridgepeer, NULL); + if (sendonly) + ast_rtp_stop(p->rtp); + } + } + return 0; +} + +/*--- add_header: Add header to SIP message */ +static int add_header(struct sip_request *req, char *var, char *value) +{ + int x = 0; + char *shortname = ""; + 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 (req->len >= sizeof(req->data) - 4) { + 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) { + for (x=0;xheader[req->headers], sizeof(req->data) - req->len - 4, "%s: %s\r\n", shortname, value); + } else { + snprintf(req->header[req->headers], sizeof(req->data) - req->len - 4, "%s: %s\r\n", var, value); + } + req->len += strlen(req->header[req->headers]); + req->headers++; + return 0; +} + +/*--- add_blank_header: Add blank header to SIP message */ +static int add_blank_header(struct sip_request *req) +{ + 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 (req->len >= sizeof(req->data) - 4) { + ast_log(LOG_WARNING, "Out of space, can't add anymore\n"); + return -1; + } + req->header[req->headers] = req->data + req->len; + snprintf(req->header[req->headers], sizeof(req->data) - req->len, "\r\n"); + req->len += strlen(req->header[req->headers]); + req->headers++; + return 0; +} + +/*--- add_line: 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; +} + +/*--- copy_header: Copy one header field from one request to another */ +static int copy_header(struct sip_request *req, struct sip_request *orig, char *field) +{ + char *tmp; + 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; +} + +/*--- copy_all_header: Copy all headers from one request to another ---*/ +static int copy_all_header(struct sip_request *req, struct sip_request *orig, char *field) +{ + char *tmp; + int start = 0; + int copied = 0; + for (;;) { + tmp = __get_header(orig, field, &start); + if (!ast_strlen_zero(tmp)) { + /* Add what we're responding to */ + add_header(req, field, tmp); + copied++; + } else + break; + } + return copied ? 0 : -1; +} + +/*--- copy_via_headers: Copy SIP VIA Headers from one request to another ---*/ +static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, struct sip_request *orig, char *field) +{ + char tmp[256]="", *oh, *end; + int start = 0; + int copied = 0; + char new[256]; + char iabuf[INET_ADDRSTRLEN]; + for (;;) { + oh = __get_header(orig, field, &start); + if (!ast_strlen_zero(oh)) { + /* Strip ;rport */ + ast_copy_string(tmp, oh, sizeof(tmp)); + oh = strstr(tmp, ";rport"); + if (oh) { + end = strchr(oh + 1, ';'); + if (end) + memmove(oh, end, strlen(end) + 1); + else + *oh = '\0'; + } + if (!copied && (ast_test_flag(p, SIP_NAT) == SIP_NAT_ALWAYS)) { + /* Whoo hoo! Now we can indicate port address translation too! Just + another RFC (RFC3581). I'll leave the original comments in for + posterity. */ + snprintf(new, sizeof(new), "%s;received=%s;rport=%d", tmp, ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port)); + add_header(req, field, new); + } else { + /* Add what we're responding to */ + add_header(req, field, tmp); + } + copied++; + } else + break; + } + if (!copied) { + ast_log(LOG_NOTICE, "No header field '%s' present to copy\n", field); + return -1; + } + return 0; +} + +/*--- add_route: Add route header into request per learned route ---*/ +static void add_route(struct sip_request *req, struct sip_route *route) +{ + char r[256], *p; + int n, rem = sizeof(r); + + if (!route) return; + + p = r; + while (route) { + n = strlen(route->hop); + if ((n+3)>rem) break; + if (p != r) { + *p++ = ','; + --rem; + } + *p++ = '<'; + ast_copy_string(p, route->hop, rem); p += n; + *p++ = '>'; + rem -= (n+2); + route = route->next; + } + *p = '\0'; + add_header(req, "Route", r); +} + +/*--- set_destination: Set destination from SIP URI ---*/ +static void set_destination(struct sip_pvt *p, char *uri) +{ + char *h, *maddr, hostname[256] = ""; + char iabuf[INET_ADDRSTRLEN]; + 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 (strncmp(h, "sip:", 4) == 0) + h += 4; + else if (strncmp(h, "sips:", 5) == 0) + h += 5; + } + hn = strcspn(h, ":;>") + 1; + if (hn > sizeof(hostname)) hn = sizeof(hostname); + ast_copy_string(hostname, h, hn); + h += hn - 1; + + /* Is "port" present? if not default to DEFAULT_SIP_PORT */ + if (*h == ':') { + /* Parse port */ + ++h; + port = strtol(h, &h, 10); + } + else + port = DEFAULT_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(iabuf, sizeof(iabuf), p->sa.sin_addr), port); +} + +/*--- init_resp: Initialize SIP response, based on SIP request ---*/ +static int init_resp(struct sip_request *req, char *resp, struct sip_request *orig) +{ + /* Initialize a response */ + if (req->headers || req->len) { + ast_log(LOG_WARNING, "Request already initialized?!?\n"); + return -1; + } + req->header[req->headers] = req->data + req->len; + snprintf(req->header[req->headers], sizeof(req->data) - req->len, "SIP/2.0 %s\r\n", resp); + req->len += strlen(req->header[req->headers]); + req->headers++; + return 0; +} + +/*--- init_req: Initialize SIP request ---*/ +static int init_req(struct sip_request *req, int sipmethod, char *recip) +{ + /* Initialize a response */ + if (req->headers || req->len) { + ast_log(LOG_WARNING, "Request already initialized?!?\n"); + return -1; + } + req->header[req->headers] = req->data + req->len; + snprintf(req->header[req->headers], sizeof(req->data) - req->len, "%s %s SIP/2.0\r\n", sip_methods[sipmethod].text, recip); + req->len += strlen(req->header[req->headers]); + req->headers++; + return 0; +} + + +/*--- respprep: Prepare SIP response packet ---*/ +static int respprep(struct sip_request *resp, struct sip_pvt *p, char *msg, struct sip_request *req) +{ + char newto[256] = "", *ot; + + memset(resp, 0, sizeof(*resp)); + init_resp(resp, msg, req); + copy_via_headers(p, resp, req, "Via"); + if (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, SIP_OUTGOING)) + snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag); + else if (p->tag && !ast_test_flag(p, SIP_OUTGOING)) + snprintf(newto, sizeof(newto), "%s;tag=as%08x", ot, p->tag); + else { + ast_copy_string(newto, ot, sizeof(newto)); + newto[sizeof(newto) - 1] = '\0'; + } + ot = newto; + } + add_header(resp, "To", ot); + copy_header(resp, req, "Call-ID"); + copy_header(resp, req, "CSeq"); + add_header(resp, "User-Agent", default_useragent); + add_header(resp, "Allow", ALLOWED_METHODS); + if (p->expiry) { + /* For registration responses, we also need expiry and + contact info */ + char contact[256]; + char tmp[256]; + + snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, p->expiry); + snprintf(tmp, sizeof(tmp), "%d", p->expiry); + add_header(resp, "Expires", tmp); + add_header(resp, "Contact", contact); + } else { + add_header(resp, "Contact", p->our_contact); + } + if (p->maxforwards) { + char tmp[256]; + snprintf(tmp, sizeof(tmp), "%d", p->maxforwards); + add_header(resp, "Max-Forwards", tmp); + } + return 0; +} + +/*--- reqprep: Initialize a SIP request packet ---*/ +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]; + char *c, *n; + char *ot, *of; + + 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 (newbranch) { + p->branch ^= rand(); + build_via(p, p->via, sizeof(p->via)); + } + 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 = p->okcontacturi; + else + c = p->initreq.rlPart2; + } else if (!ast_strlen_zero(p->okcontacturi)) { + c = p->okcontacturi; /* Use for BYE, REFER or REINVITE */ + } else if (!ast_strlen_zero(p->uri)) { + c = p->uri; + } else { + /* We have no URI, use To: or From: header as URI (depending on direction) */ + c = get_header(orig, (ast_test_flag(p, SIP_OUTGOING)) ? "To" : "From"); + ast_copy_string(stripped, c, sizeof(stripped)); + c = get_in_brackets(stripped); + n = strchr(c, ';'); + if (n) + *n = '\0'; + } + 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, p->route->next); + } + + 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, SIP_OUTGOING) && !ast_strlen_zero(p->theirtag)) + snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag); + else if (!ast_test_flag(p, SIP_OUTGOING)) + snprintf(newto, sizeof(newto), "%s;tag=as%08x", ot, p->tag); + else + snprintf(newto, sizeof(newto), "%s", ot); + ot = newto; + } + + if (ast_test_flag(p, SIP_OUTGOING)) { + add_header(req, "From", of); + add_header(req, "To", ot); + } else { + add_header(req, "From", ot); + add_header(req, "To", of); + } + add_header(req, "Contact", p->our_contact); + copy_header(req, orig, "Call-ID"); + add_header(req, "CSeq", tmp); + + add_header(req, "User-Agent", default_useragent); + return 0; +} + +static int __transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req, int 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(&resp, "Content-Length", "0"); + add_blank_header(&resp); + return send_response(p, &resp, reliable, seqno); +} + +/*--- transmit_response: Transmit response, no retransmits */ +static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req) +{ + return __transmit_response(p, msg, req, 0); +} + +/*--- transmit_response_with_unsupported: Transmit response, no retransmits */ +static int transmit_response_with_unsupported(struct sip_pvt *p, char *msg, struct sip_request *req, char *unsupported) +{ + struct sip_request resp; + respprep(&resp, p, msg, req); + append_date(&resp); + add_header(&resp, "Unsupported", unsupported); + return send_response(p, &resp, 0, 0); +} + +/*--- transmit_response_reliable: Transmit response, Make sure you get a reply */ +static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req, int fatal) +{ + return __transmit_response(p, msg, req, fatal ? 2 : 1); +} + +/*--- append_date: Append date to SIP message ---*/ +static void append_date(struct sip_request *req) +{ + char tmpdat[256]; + struct tm tm; + time_t t; + + time(&t); + gmtime_r(&t, &tm); + strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T GMT", &tm); + add_header(req, "Date", tmpdat); +} + +/*--- transmit_response_with_date: Append date and content length before transmitting response ---*/ +static int transmit_response_with_date(struct sip_pvt *p, char *msg, struct sip_request *req) +{ + struct sip_request resp; + respprep(&resp, p, msg, req); + append_date(&resp); + add_header(&resp, "Content-Length", "0"); + add_blank_header(&resp); + return send_response(p, &resp, 0, 0); +} + +/*--- transmit_response_with_allow: Append Accept header, content length before transmitting response ---*/ +static int transmit_response_with_allow(struct sip_pvt *p, char *msg, struct sip_request *req, int reliable) +{ + struct sip_request resp; + respprep(&resp, p, msg, req); + add_header(&resp, "Accept", "application/sdp"); + add_header(&resp, "Content-Length", "0"); + add_blank_header(&resp); + return send_response(p, &resp, reliable, 0); +} + +/* transmit_response_with_auth: Respond with authorization request */ +static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_request *req, char *randdata, int reliable, char *header, int stale) +{ + struct sip_request resp; + char tmp[256]; + 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 realm=\"%s\", nonce=\"%s\" %s", global_realm, randdata, stale ? ", stale=true" : ""); + respprep(&resp, p, msg, req); + add_header(&resp, header, tmp); + add_header(&resp, "Content-Length", "0"); + add_blank_header(&resp); + return send_response(p, &resp, reliable, seqno); +} + +/*--- add_text: 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 */ + int len = strlen(text); + char clen[256]; + snprintf(clen, sizeof(clen), "%d", len); + add_header(req, "Content-Type", "text/plain"); + add_header(req, "Content-Length", clen); + add_line(req, text); + return 0; +} + +/*--- add_digit: 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) +{ + char tmp[256]; + int len; + char clen[256]; + snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=250\r\n", digit); + len = strlen(tmp); + snprintf(clen, sizeof(clen), "%d", len); + add_header(req, "Content-Type", "application/dtmf-relay"); + add_header(req, "Content-Length", clen); + add_line(req, tmp); + return 0; +} + +/*--- add_sdp: Add Session Description Protocol message ---*/ +static int add_sdp(struct sip_request *resp, struct sip_pvt *p) +{ + int len = 0; + int codec = 0; + int pref_codec = 0; + int alreadysent = 0; + char costr[80]; + struct sockaddr_in sin; + struct sockaddr_in vsin; + char v[256] = ""; + char s[256] = ""; + char o[256] = ""; + char c[256] = ""; + char t[256] = ""; + char m[256] = ""; + char m2[256] = ""; + char a[1024] = ""; + char a2[1024] = ""; + char iabuf[INET_ADDRSTRLEN]; + int x = 0; + int capability = 0 ; + struct sockaddr_in dest; + struct sockaddr_in vdest = { 0, }; + int debug=0; + + debug = sip_debug_test_pvt(p); + + /* XXX We break with the "recommendation" and send our IP, in order that our + peer doesn't have to ast_gethostbyname() us XXX */ + len = 0; + if (!p->rtp) { + ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n"); + return -1; + } + capability = p->capability; + + if (!p->sessionid) { + p->sessionid = getpid(); + p->sessionversion = p->sessionid; + } else + p->sessionversion++; + ast_rtp_get_us(p->rtp, &sin); + if (p->vrtp) + ast_rtp_get_us(p->vrtp, &vsin); + + if (p->redirip.sin_addr.s_addr) { + dest.sin_port = p->redirip.sin_port; + dest.sin_addr = p->redirip.sin_addr; + if (p->redircodecs) + capability = p->redircodecs; + } else { + dest.sin_addr = p->ourip; + dest.sin_port = sin.sin_port; + } + + /* Determine video destination */ + if (p->vrtp) { + if (p->vredirip.sin_addr.s_addr) { + vdest.sin_port = p->vredirip.sin_port; + vdest.sin_addr = p->vredirip.sin_addr; + } else { + vdest.sin_addr = p->ourip; + vdest.sin_port = vsin.sin_port; + } + } + if (debug){ + ast_verbose("We're at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(sin.sin_port)); + if (p->vrtp) + ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(vsin.sin_port)); + } + 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(iabuf, sizeof(iabuf), dest.sin_addr)); + snprintf(s, sizeof(s), "s=session\r\n"); + snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr)); + snprintf(t, sizeof(t), "t=0 0\r\n"); + snprintf(m, sizeof(m), "m=audio %d RTP/AVP", ntohs(dest.sin_port)); + snprintf(m2, sizeof(m2), "m=video %d RTP/AVP", ntohs(vdest.sin_port)); + /* Prefer the codec we were requested to use, first, no matter what */ + if (capability & p->prefcodec) { + if (debug) + ast_verbose("Answering/Requesting with root capability 0x%x (%s)\n", p->prefcodec, ast_getformatname(p->prefcodec)); + codec = ast_rtp_lookup_code(p->rtp, 1, p->prefcodec); + if (codec > -1) { + snprintf(costr, sizeof(costr), " %d", codec); + if (p->prefcodec <= AST_FORMAT_MAX_AUDIO) { + strncat(m, costr, sizeof(m) - strlen(m) - 1); + snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, p->prefcodec)); + ast_copy_string(a, costr, sizeof(a)); + } else { + strncat(m2, costr, sizeof(m2) - strlen(m2) - 1); + snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, p->prefcodec)); + ast_copy_string(a2, costr, sizeof(a2)); + } + } + alreadysent |= p->prefcodec; + } + /* Start by sending our preferred codecs */ + for (x = 0 ; x < 32 ; x++) { + if (!(pref_codec = ast_codec_pref_index(&p->prefs,x))) + break; + if ((capability & pref_codec) && !(alreadysent & pref_codec)) { + if (debug) + ast_verbose("Answering with preferred capability 0x%x (%s)\n", pref_codec, ast_getformatname(pref_codec)); + codec = ast_rtp_lookup_code(p->rtp, 1, pref_codec); + if (codec > -1) { + snprintf(costr, sizeof(costr), " %d", codec); + if (pref_codec <= AST_FORMAT_MAX_AUDIO) { + strncat(m, costr, sizeof(m) - strlen(m) - 1); + snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, pref_codec)); + strncat(a, costr, sizeof(a) - strlen(a) - 1); + } else { + strncat(m2, costr, sizeof(m2) - strlen(m2) - 1); + snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, pref_codec)); + strncat(a2, costr, sizeof(a2) - strlen(a) - 1); + } + } + } + alreadysent |= pref_codec; + } + + /* Now send any other common codecs, and non-codec formats: */ + for (x = 1; x <= ((videosupport && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) { + if ((capability & x) && !(alreadysent & x)) { + if (debug) + ast_verbose("Answering with capability 0x%x (%s)\n", x, ast_getformatname(x)); + codec = ast_rtp_lookup_code(p->rtp, 1, x); + if (codec > -1) { + snprintf(costr, sizeof(costr), " %d", codec); + if (x <= AST_FORMAT_MAX_AUDIO) { + strncat(m, costr, sizeof(m) - strlen(m) - 1); + snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x)); + strncat(a, costr, sizeof(a) - strlen(a) - 1); + } else { + strncat(m2, costr, sizeof(m2) - strlen(m2) - 1); + snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x)); + strncat(a2, costr, sizeof(a2) - strlen(a2) - 1); + } + } + } + } + for (x = 1; x <= AST_RTP_MAX; x <<= 1) { + if (p->noncodeccapability & x) { + if (debug) + ast_verbose("Answering with non-codec capability 0x%x (%s)\n", x, ast_rtp_lookup_mime_subtype(0, x)); + codec = ast_rtp_lookup_code(p->rtp, 0, x); + if (codec > -1) { + snprintf(costr, sizeof(costr), " %d", codec); + strncat(m, costr, sizeof(m) - strlen(m) - 1); + snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(0, x)); + strncat(a, costr, sizeof(a) - strlen(a) - 1); + if (x == AST_RTP_DTMF) { + /* Indicate we support DTMF and FLASH... */ + snprintf(costr, sizeof costr, "a=fmtp:%d 0-16\r\n", + codec); + strncat(a, costr, sizeof(a) - strlen(a) - 1); + } + } + } + } + strncat(a, "a=silenceSupp:off - - - -\r\n", sizeof(a) - strlen(a) - 1); + if (strlen(m) < sizeof(m) - 2) + strncat(m, "\r\n", sizeof(m) - strlen(m) - 1); + if (strlen(m2) < sizeof(m2) - 2) + strncat(m2, "\r\n", sizeof(m2) - strlen(m2) - 1); + if ((sizeof(m) <= strlen(m) - 2) || (sizeof(m2) <= strlen(m2) - 2) || (sizeof(a) == strlen(a)) || (sizeof(a2) == strlen(a2))) + ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n"); + len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m) + strlen(a); + if ((p->vrtp) && (!ast_test_flag(p, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */ + len += strlen(m2) + strlen(a2); + snprintf(costr, sizeof(costr), "%d", len); + add_header(resp, "Content-Type", "application/sdp"); + add_header(resp, "Content-Length", costr); + add_line(resp, v); + add_line(resp, o); + add_line(resp, s); + add_line(resp, c); + add_line(resp, t); + add_line(resp, m); + add_line(resp, a); + if ((p->vrtp) && (!ast_test_flag(p, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */ + add_line(resp, m2); + add_line(resp, a2); + } + /* Update lastrtprx when we send our SDP */ + time(&p->lastrtprx); + time(&p->lastrtptx); + return 0; +} + +/*--- copy_request: copy SIP request (mostly used to save request for responses) ---*/ +static void copy_request(struct sip_request *dst, 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; xheaders; x++) + dst->header[x] += offset; + for (x=0; xlines; x++) + dst->line[x] += offset; +} + +/*--- transmit_response_with_sdp: Used for 200 OK and 183 early media ---*/ +static int transmit_response_with_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); + ast_rtp_offered_from_local(p->rtp, 0); + add_sdp(&resp, p); + return send_response(p, &resp, retrans, seqno); +} + +/*--- determine_firstline_parts: parse first line of incoming SIP request */ +static int determine_firstline_parts( struct sip_request *req ) +{ + char *e, *cmd; + int len; + + cmd = ast_skip_blanks(req->header[0]); + if (!*cmd) + return -1; + req->rlPart1 = cmd; + e = ast_skip_nonblanks(cmd); + /* Get the command */ + if (*e) + *e++ = '\0'; + e = ast_skip_blanks(e); + if ( !*e ) + return -1; + + if ( !strcasecmp(cmd, "SIP/2.0") ) { + /* We have a response */ + req->rlPart2 = e; + len = strlen( req->rlPart2 ); + if ( len < 2 ) { + return -1; + } + ast_trim_blanks(e); + } else { + /* We have a request */ + if ( *e == '<' ) { + e++; + if ( !*e ) { + return -1; + } + } + req->rlPart2 = e; /* URI */ + if ( ( e= strrchr( req->rlPart2, 'S' ) ) == NULL ) { + return -1; + } + /* XXX maybe trim_blanks() ? */ + while( isspace( *(--e) ) ) {} + if ( *e == '>' ) { + *e = '\0'; + } else { + *(++e)= '\0'; + } + } + return 1; +} + +/*--- transmit_reinvite_with_sdp: Transmit reinvite with SDP :-) ---*/ +/* 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; + if (ast_test_flag(p, SIP_REINVITE_UPDATE)) + reqprep(&req, p, SIP_UPDATE, 0, 1); + else + reqprep(&req, p, SIP_INVITE, 0, 1); + + add_header(&req, "Allow", ALLOWED_METHODS); + ast_rtp_offered_from_local(p->rtp, 1); + add_sdp(&req, p); + /* Use this as the basis */ + copy_request(&p->initreq, &req); + parse(&p->initreq); + if (sip_debug_test_pvt(p)) + ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); + determine_firstline_parts(&p->initreq); + p->lastinvite = p->ocseq; + ast_set_flag(p, SIP_OUTGOING); + return send_request(p, &req, 1, p->ocseq); +} + +/*--- extract_uri: Check Contact: URI of SIP message ---*/ +static void extract_uri(struct sip_pvt *p, struct sip_request *req) +{ + char stripped[256]=""; + char *c, *n; + ast_copy_string(stripped, get_header(req, "Contact"), sizeof(stripped)); + c = get_in_brackets(stripped); + n = strchr(c, ';'); + if (n) + *n = '\0'; + if (c && !ast_strlen_zero(c)) + ast_copy_string(p->uri, c, sizeof(p->uri)); +} + +/*--- build_contact: Build contact header - the contact header we send out ---*/ +static void build_contact(struct sip_pvt *p) +{ + char iabuf[INET_ADDRSTRLEN]; + + /* Construct Contact: header */ + if (ourport != DEFAULT_SIP_PORT) + snprintf(p->our_contact, sizeof(p->our_contact), "", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport); + else + snprintf(p->our_contact, sizeof(p->our_contact), "", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip)); +} + +/*--- initreqprep: Initiate SIP request to peer/user ---*/ +static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, char *vxml_url) +{ + char invite[256]=""; + char from[256]; + char to[256]; + char tmp[80]; + char iabuf[INET_ADDRSTRLEN]; + char *l = default_callerid, *n=NULL; + int x; + char urioptions[256]=""; + + if (ast_test_flag(p, SIP_USEREQPHONE)) { + char onlydigits = 1; + x=0; + + /* 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 (p->username && p->username[0] == '+') + x=1; + + for (; xusername); x++) { + if (!strchr(AST_DIGIT_ANYNUM, p->username[x])) { + onlydigits = 0; + break; + } + } + + /* If we have only digits, add ;user=phone to the uri */ + if (onlydigits) + strcpy(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 (!l || (!ast_isphonenumber(l) && default_callerid[0])) + l = default_callerid; + /* if user want's his callerid restricted */ + if ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) { + l = CALLERID_UNKNOWN; + n = l; + } + if (!n || 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_copy_string(p->fromuser, l, sizeof(p->fromuser)); + + /* Allow user to be overridden */ + if (!ast_strlen_zero(p->fromname)) + n = p->fromname; + else /* Save for any further attempts */ + ast_copy_string(p->fromname, n, sizeof(p->fromname)); + + if ((ourport != DEFAULT_SIP_PORT) && ast_strlen_zero(p->fromdomain)) + snprintf(from, sizeof(from), "\"%s\" ;tag=as%08x", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, ourport, p->tag); + else + snprintf(from, sizeof(from), "\"%s\" ;tag=as%08x", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, p->tag); + + /* If we're calling a registred SIP peer, use the fullcontact to dial to the peer */ + if (!ast_strlen_zero(p->fullcontact)) { + /* If we have full contact, trust it */ +#ifdef SIP_TCP_SUPPORT + if (!strchr(p->fullcontact, '@')) + snprintf(invite, sizeof(invite), "sip:%s@%s:%d",p->username, p->tohost, ntohs(p->sa.sin_port)); + else +#endif + ast_copy_string(invite, p->fullcontact, sizeof(invite)); + /* Otherwise, use the username while waiting for registration */ + } else if (!ast_strlen_zero(p->username)) { + if (ntohs(p->sa.sin_port) != DEFAULT_SIP_PORT) { + snprintf(invite, sizeof(invite), "sip:%s@%s:%d%s",p->username, p->tohost, ntohs(p->sa.sin_port), urioptions); + } else { + snprintf(invite, sizeof(invite), "sip:%s@%s%s",p->username, p->tohost, urioptions); + } + } else if (ntohs(p->sa.sin_port) != DEFAULT_SIP_PORT) { + snprintf(invite, sizeof(invite), "sip:%s:%d%s", p->tohost, ntohs(p->sa.sin_port), urioptions); + } else { + snprintf(invite, sizeof(invite), "sip:%s%s", p->tohost, urioptions); + } + ast_copy_string(p->uri, invite, sizeof(p->uri)); + /* If there is a VXML URL append it to the SIP URL */ + if (vxml_url) + { + snprintf(to, sizeof(to), "<%s>;%s", invite, vxml_url); + } else { + snprintf(to, sizeof(to), "<%s>", invite); + } + memset(req, 0, sizeof(struct sip_request)); + init_req(req, sipmethod, invite); + 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 */ + add_header(req, "From", from); + ast_copy_string(p->exten, l, sizeof(p->exten)); + build_contact(p); + add_header(req, "To", to); + add_header(req, "Contact", p->our_contact); + add_header(req, "Call-ID", p->callid); + add_header(req, "CSeq", tmp); + add_header(req, "User-Agent", default_useragent); +} + +/*--- transmit_invite: Build REFER/INVITE/OPTIONS message and transmit it ---*/ +static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, struct sip_invite_param *options, int init) +{ + struct sip_request req; + + if (init) { + /* Bump branch even on initial requests */ + p->branch ^= rand(); + build_via(p, p->via, sizeof(p->via)); + initreqprep(&req, p, sipmethod, options ? options->vxml_url : (char *) NULL); + } else + reqprep(&req, p, sipmethod, 0, 1); + + if (options && options->auth) + add_header(&req, options->authheader, options->auth); + append_date(&req); + if (sipmethod == SIP_REFER) { /* Call transfer */ + if (!ast_strlen_zero(p->refer_to)) + add_header(&req, "Refer-To", p->refer_to); + if (!ast_strlen_zero(p->referred_by)) + add_header(&req, "Referred-By", p->referred_by); + } +#ifdef OSP_SUPPORT + if (options && options->osptoken && !ast_strlen_zero(options->osptoken)) { + ast_log(LOG_DEBUG,"Adding OSP Token: %s\n", options->osptoken); + add_header(&req, "P-OSP-Auth-Token", options->osptoken); + } else { + ast_log(LOG_DEBUG,"NOT Adding OSP Token\n"); + } +#endif + if (options && options->distinctive_ring && !ast_strlen_zero(options->distinctive_ring)) + { + add_header(&req, "Alert-Info", options->distinctive_ring); + } + add_header(&req, "Allow", ALLOWED_METHODS); + if (options && options->addsipheaders && init) { + struct ast_channel *ast; + char *header = (char *) NULL; + char *content = (char *) NULL; + char *end = (char *) NULL; + struct varshead *headp = (struct varshead *) NULL; + struct ast_var_t *current; + + ast = p->owner; /* The owner channel */ + if (ast) { + headp=&ast->varshead; + if (!headp) + ast_log(LOG_WARNING,"No Headp for the channel...ooops!\n"); + else { + AST_LIST_TRAVERSE(headp,current,entries) { + /* SIPADDHEADER: Add SIP header to outgoing call */ + if (!strncasecmp(ast_var_name(current),"SIPADDHEADER",strlen("SIPADDHEADER"))) { + header = ast_var_value(current); + /* Strip of the starting " (if it's there) */ + if (*header == '"') + header++; + if ((content = strchr(header, ':'))) { + *content = '\0'; + content++; /* Move pointer ahead */ + /* Skip white space */ + while (*content == ' ') + content++; + /* Strip the ending " (if it's there) */ + end = content + strlen(content) -1; + if (*end == '"') + *end = '\0'; + + add_header(&req, header, content); + if (sipdebug) + ast_log(LOG_DEBUG, "Adding SIP Header \"%s\" with content :%s: \n", header, content); + } + } + } + } + } + } + if (sdp) { + ast_rtp_offered_from_local(p->rtp, 1); + add_sdp(&req, p); + } else { + add_header(&req, "Content-Length", "0"); + add_blank_header(&req); + } + + if (!p->initreq.headers) { + /* Use this as the basis */ + copy_request(&p->initreq, &req); + parse(&p->initreq); + if (sip_debug_test_pvt(p)) + ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); + determine_firstline_parts(&p->initreq); + } + p->lastinvite = p->ocseq; + return send_request(p, &req, init ? 2 : 1, p->ocseq); +} + +/*--- transmit_state_notify: Used in the SUBSCRIBE notification subsystem ----*/ +static int transmit_state_notify(struct sip_pvt *p, int state, int full) +{ + char tmp[4000]; + int maxbytes = 0; + int bytes = 0; + char from[256], to[256]; + char *t, *c, *a; + char *mfrom, *mto; + struct sip_request req; + char clen[20]; + + memset(from, 0, sizeof(from)); + memset(to, 0, sizeof(to)); + ast_copy_string(from, get_header(&p->initreq, "From"), sizeof(from)); + + c = ditch_braces(from); + if (strncmp(c, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c); + return -1; + } + if ((a = strchr(c, ';'))) { + *a = '\0'; + } + mfrom = c; + + reqprep(&req, p, SIP_NOTIFY, 0, 1); + + if (p->subscribed == 1) { + ast_copy_string(to, get_header(&p->initreq, "To"), sizeof(to)); + + c = ditch_braces(to); + if (strncmp(c, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c); + return -1; + } + if ((a = strchr(c, ';'))) { + *a = '\0'; + } + mto = c; + + add_header(&req, "Event", "presence"); + add_header(&req, "Subscription-State", "active"); + add_header(&req, "Content-Type", "application/xpidf+xml"); + + if ((state==AST_EXTENSION_UNAVAILABLE) || (state==AST_EXTENSION_BUSY)) + state = 2; + else if (state==AST_EXTENSION_INUSE) + state = 1; + else + state = 0; + + t = tmp; + maxbytes = sizeof(tmp); + bytes = snprintf(t, maxbytes, "\n"); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "\n"); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "\n"); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "\n", mfrom); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "\n", p->exten); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "
\n", mto); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "\n", !state ? "open" : (state==1) ? "inuse" : "closed"); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "\n", !state ? "online" : (state==1) ? "onthephone" : "offline"); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "
\n\n\n"); + } else { + add_header(&req, "Event", "dialog"); + add_header(&req, "Content-Type", "application/dialog-info+xml"); + + t = tmp; + maxbytes = sizeof(tmp); + bytes = snprintf(t, maxbytes, "\n"); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "\n", p->dialogver++, full ? "full":"partial", mfrom); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "\n", p->exten); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "%s\n", state ? "confirmed" : "terminated"); + t += bytes; + maxbytes -= bytes; + bytes = snprintf(t, maxbytes, "\n\n"); + } + if (t > tmp + sizeof(tmp)) + ast_log(LOG_WARNING, "Buffer overflow detected!! (Please file a bug report)\n"); + + snprintf(clen, sizeof(clen), "%d", (int)strlen(tmp)); + add_header(&req, "Content-Length", clen); + add_line(&req, tmp); + + return send_request(p, &req, 1, p->ocseq); +} + +/*--- transmit_notify_with_mwi: Notify user of messages waiting in voicemail ---*/ +/* Notification only works for registred 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) +{ + struct sip_request req; + char tmp[256]; + char tmp2[256]; + char clen[20]; + initreqprep(&req, p, SIP_NOTIFY, NULL); + add_header(&req, "Event", "message-summary"); + add_header(&req, "Content-Type", default_notifymime); + + snprintf(tmp, sizeof(tmp), "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no"); + snprintf(tmp2, sizeof(tmp2), "Voice-Message: %d/%d (0/0)\r\n", newmsgs, oldmsgs); + snprintf(clen, sizeof(clen), "%d", (int)(strlen(tmp) + strlen(tmp2))); + add_header(&req, "Content-Length", clen); + add_line(&req, tmp); + add_line(&req, tmp2); + + if (!p->initreq.headers) { + /* Use this as the basis */ + copy_request(&p->initreq, &req); + parse(&p->initreq); + if (sip_debug_test_pvt(p)) + ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); + determine_firstline_parts(&p->initreq); + } + + return send_request(p, &req, 1, p->ocseq); +} + +/*--- transmit_sip_request: Transmit SIP request */ +static int transmit_sip_request(struct sip_pvt *p,struct sip_request *req) +{ + if (!p->initreq.headers) { + /* Use this as the basis */ + copy_request(&p->initreq, req); + parse(&p->initreq); + if (sip_debug_test_pvt(p)) + ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); + determine_firstline_parts(&p->initreq); + } + + return send_request(p, req, 0, p->ocseq); +} + +/*--- transmit_notify_with_sipfrag: Notify a transferring party of the status of trasnfer ---*/ +/* Apparently the draft SIP REFER structure was too simple, so it was decided that the + * status of transfers also needed to be sent via NOTIFY instead of just the 202 Accepted + * that had worked heretofore. + */ +static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq) +{ + struct sip_request req; + char tmp[256]; + char clen[20]; + 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", "terminated;reason=noresource"); + add_header(&req, "Content-Type", "message/sipfrag;version=2.0"); + + ast_copy_string(tmp, "SIP/2.0 200 OK", sizeof(tmp)); + snprintf(clen, sizeof(clen), "%d", (int)(strlen(tmp))); + add_header(&req, "Content-Length", clen); + add_line(&req, tmp); + + if (!p->initreq.headers) { + /* Use this as the basis */ + copy_request(&p->initreq, &req); + parse(&p->initreq); + if (sip_debug_test_pvt(p)) + ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); + determine_firstline_parts(&p->initreq); + } + + return send_request(p, &req, 1, p->ocseq); +} + +static char *regstate2str(int 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"; + } +} + +static int transmit_register(struct sip_registry *r, int sipmethod, char *auth, char *authheader); + +/*--- sip_reregister: Update registration with SIP Proxy---*/ +static int sip_reregister(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; + + /* 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; +} + +/*--- __sip_do_register: 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; +} + +/*--- sip_reg_timeout: Registration timeout, register again */ +static int sip_reg_timeout(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; + if (p->registry) + ASTOBJ_UNREF(p->registry, sip_registry_destroy); + r->call = NULL; + ast_set_flag(p, SIP_NEEDDESTROY); + /* Pretend to ACK anything just in case */ + /* OEJ: Ack what??? */ + __sip_pretend_ack(p); + } + /* 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", "Channel: SIP\r\nUser: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate)); + ASTOBJ_UNREF(r,sip_registry_destroy); + return 0; +} + +/*--- transmit_register: Transmit register to SIP proxy or UA ---*/ +static int transmit_register(struct sip_registry *r, int sipmethod, char *auth, char *authheader) +{ + struct sip_request req; + char from[256]; + char to[256]; + char tmp[80]; + char via[80]; + char addr[80]; + struct sip_pvt *p; + + /* 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))) { + 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; + p->tag = rand(); /* create a new local tag for every register attempt */ + p->theirtag[0]='\0'; /* forget their old tag, so we don't match tags when getting response */ + } + } else { + /* Build callid for registration if we haven't registred before */ + if (!r->callid_valid) { + build_callid(r->callid, sizeof(r->callid), __ourip, default_fromdomain); + r->callid_valid = 1; + } + /* Allocate SIP packet for registration */ + p=sip_alloc( r->callid, NULL, 0, SIP_REGISTER); + if (!p) { + ast_log(LOG_WARNING, "Unable to allocate registration call\n"); + return 0; + } + /* 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_sched_del(sched, r->timeout); + r->timeout = ast_sched_add(sched, global_reg_timeout*1000, sip_reg_timeout, r); + ast_log(LOG_WARNING, "Still have a registration timeout for %s@%s (create_addr() error), %d\n", r->username, r->hostname, r->timeout); + } else { + r->timeout = ast_sched_add(sched, global_reg_timeout*1000, sip_reg_timeout, r); + 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); + } + r->regattempts++; + return 0; + } + /* Copy back Call-ID in case create_addr changed it */ + ast_copy_string(r->callid, p->callid, sizeof(r->callid)); + if (r->portno) + p->sa.sin_port = htons(r->portno); + ast_set_flag(p, 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_copy_string(p->peersecret, r->secret, sizeof(p->peersecret)); + if (!ast_strlen_zero(r->md5secret)) + ast_copy_string(p->peermd5secret, r->md5secret, sizeof(p->peermd5secret)); + /* User name in this realm + - if authuser is set, use that, otherwise use username */ + if (!ast_strlen_zero(r->authuser)) { + ast_copy_string(p->peername, r->authuser, sizeof(p->peername)); + ast_copy_string(p->authname, r->authuser, sizeof(p->authname)); + } else { + if (!ast_strlen_zero(r->username)) { + ast_copy_string(p->peername, r->username, sizeof(p->peername)); + ast_copy_string(p->authname, r->username, sizeof(p->authname)); + ast_copy_string(p->fromuser, r->username, sizeof(p->fromuser)); + } + } + if (!ast_strlen_zero(r->username)) + ast_copy_string(p->username, r->username, sizeof(p->username)); + /* Save extension in packet */ + ast_copy_string(p->exten, r->contact, sizeof(p->exten)); + + /* + 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)) + memcpy(&p->ourip, &bindaddr.sin_addr, sizeof(p->ourip)); + build_contact(p); + } + + /* set up a timeout */ + if (auth == NULL) { + if (r->timeout > -1) { + ast_log(LOG_WARNING, "Still have a registration timeout, %d\n", r->timeout); + ast_sched_del(sched, r->timeout); + } + r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, r); + ast_log(LOG_DEBUG, "Scheduled a registration timeout for %s : %d\n", r->hostname, r->timeout); + } + + if (strchr(r->username, '@')) { + snprintf(from, sizeof(from), ";tag=as%08x", r->username, p->tag); + if (!ast_strlen_zero(p->theirtag)) + snprintf(to, sizeof(to), ";tag=%s", r->username, p->theirtag); + else + snprintf(to, sizeof(to), "", r->username); + } else { + snprintf(from, sizeof(from), ";tag=as%08x", r->username, p->tohost, p->tag); + if (!ast_strlen_zero(p->theirtag)) + snprintf(to, sizeof(to), ";tag=%s", r->username, p->tohost, p->theirtag); + else + snprintf(to, sizeof(to), "", r->username, p->tohost); + } + + /* Fromdomain is what we are registering to, regardless of actual + host name from SRV */ + if (p->fromdomain && !ast_strlen_zero(p->fromdomain)) + snprintf(addr, sizeof(addr), "sip:%s", p->fromdomain); + else + snprintf(addr, sizeof(addr), "sip:%s", r->hostname); + ast_copy_string(p->uri, addr, sizeof(p->uri)); + + p->branch ^= rand(); + + memset(&req, 0, sizeof(req)); + 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, via, sizeof(via)); + add_header(&req, "Via", via); + add_header(&req, "From", from); + add_header(&req, "To", to); + add_header(&req, "Call-ID", p->callid); + add_header(&req, "CSeq", tmp); + add_header(&req, "User-Agent", default_useragent); + + + 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_copy_string(p->realm, r->realm, sizeof(p->realm)); + ast_copy_string(p->nonce, r->nonce, sizeof(p->nonce)); + ast_copy_string(p->domain, r->domain, sizeof(p->domain)); + ast_copy_string(p->opaque, r->opaque, sizeof(p->opaque)); + ast_copy_string(p->qop, r->qop, sizeof(p->qop)); + + memset(digest,0,sizeof(digest)); + build_reply_digest(p, sipmethod, digest, sizeof(digest)); + add_header(&req, "Authorization", digest); + + } + + 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(&req, "Content-Length", "0"); + add_blank_header(&req); + copy_request(&p->initreq, &req); + parse(&p->initreq); + if (sip_debug_test_pvt(p)) { + ast_verbose("REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines); + } + determine_firstline_parts(&p->initreq); + 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, 2, p->ocseq); +} + +/*--- transmit_message_with_text: 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, 1, p->ocseq); +} + +/*--- transmit_refer: Transmit SIP REFER message ---*/ +static int transmit_refer(struct sip_pvt *p, const char *dest) +{ + struct sip_request req; + char from[256]; + char *of, *c; + char referto[256]; + char tmp[80]; + + if (ast_test_flag(p, SIP_OUTGOING)) + of = get_header(&p->initreq, "To"); + else + of = get_header(&p->initreq, "From"); + ast_copy_string(from, of, sizeof(from)); + of = ditch_braces(from); + ast_copy_string(p->from,of,sizeof(p->from)); + if (strncmp(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'; + c++; + } + if (c) { + snprintf(referto, sizeof(referto), "", dest, c); + } else { + snprintf(referto, sizeof(referto), "", dest); + } + + ast_copy_string(tmp, get_header(&p->initreq, "Max-Forwards"), sizeof(tmp)); + if (strlen(tmp) && atoi(tmp)) { + p->maxforwards = atoi(tmp) - 1; + } else { + p->maxforwards = DEFAULT_MAX_FORWARDS - 1; + } + if (p->maxforwards > -1) { + /* save in case we get 407 challenge */ + ast_copy_string(p->refer_to, referto, sizeof(p->refer_to)); + ast_copy_string(p->referred_by, p->our_contact, sizeof(p->referred_by)); + + reqprep(&req, p, SIP_REFER, 0, 1); + add_header(&req, "Refer-To", referto); + if (!ast_strlen_zero(p->our_contact)) + add_header(&req, "Referred-By", p->our_contact); + add_blank_header(&req); + return send_request(p, &req, 1, p->ocseq); + } else { + return -1; + } +} + +/*--- transmit_info_with_digit: Send SIP INFO dtmf message, see Cisco documentation on cisco.co +m ---*/ +static int transmit_info_with_digit(struct sip_pvt *p, char digit) +{ + struct sip_request req; + reqprep(&req, p, SIP_INFO, 0, 1); + add_digit(&req, digit); + return send_request(p, &req, 1, p->ocseq); +} + +/*--- transmit_request: transmit generic SIP request ---*/ +static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, int reliable, int newbranch) +{ + struct sip_request resp; + reqprep(&resp, p, sipmethod, seqno, newbranch); + add_header(&resp, "Content-Length", "0"); + add_blank_header(&resp); + return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq); +} + +/*--- transmit_request_with_auth: Transmit SIP request, auth added ---*/ +static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, int reliable, int newbranch) +{ + struct sip_request resp; + + reqprep(&resp, p, sipmethod, seqno, newbranch); + if (*p->realm) + { + char digest[1024]; + + memset(digest, 0, sizeof(digest)); + build_reply_digest(p, sipmethod, digest, sizeof(digest)); + add_header(&resp, "Proxy-Authorization", digest); + } + + add_header(&resp, "Content-Length", "0"); + add_blank_header(&resp); + return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq); +} + +/*--- expire_register: Expire registration of SIP peer ---*/ +static int expire_register(void *data) +{ + struct sip_peer *peer = data; + + memset(&peer->addr, 0, sizeof(peer->addr)); + ast_db_del("SIP/Registry", peer->name); + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name); + register_peer_exten(peer, 0); + peer->expire = -1; + ast_device_state_changed("SIP/%s", peer->name); + if (ast_test_flag(peer, SIP_SELFDESTRUCT) || ast_test_flag((&peer->flags_page2), SIP_PAGE2_RTAUTOCLEAR)) { + peer = ASTOBJ_CONTAINER_UNLINK(&peerl, peer); + ASTOBJ_UNREF(peer, sip_destroy_peer); + } + + return 0; +} + +static int sip_poke_peer(struct sip_peer *peer); + +static int sip_poke_peer_s(void *data) +{ + struct sip_peer *peer = data; + peer->pokeexpire = -1; + sip_poke_peer(peer); + return 0; +} + +/*--- reg_source_db: Get registration details from Asterisk DB ---*/ +static void reg_source_db(struct sip_peer *peer) +{ + char data[256]; + char iabuf[INET_ADDRSTRLEN]; + struct in_addr in; + int expiry; + int port; + char *scan, *addr, *port_str, *expiry_str, *username, *contact; + + 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_verbose > 2) + ast_verbose(VERBOSE_PREFIX_3 "SIP Seeding peer from astdb: '%s' at %s@%s:%d for %d\n", + peer->name, peer->username, ast_inet_ntoa(iabuf, sizeof(iabuf), 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 (peer->pokeexpire > -1) + ast_sched_del(sched, peer->pokeexpire); + peer->pokeexpire = ast_sched_add(sched, rand() % 5000 + 1, sip_poke_peer_s, peer); + } else + sip_poke_peer(peer); + if (peer->expire > -1) + ast_sched_del(sched, peer->expire); + peer->expire = ast_sched_add(sched, (expiry + 10) * 1000, expire_register, peer); + register_peer_exten(peer, 1); +} + +/*--- parse_ok_contact: Parse contact header for 200 OK on INVITE ---*/ +static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req) +{ + char contact[250]= ""; + char *c, *n, *pt; + int port; + struct hostent *hp; + struct ast_hostent ahp; + struct sockaddr_in oldsin; + + /* 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_copy_string(pvt->fullcontact, c, sizeof(pvt->fullcontact)); + + /* Save URI for later ACKs, BYE or RE-invites */ + ast_copy_string(pvt->okcontacturi, c, sizeof(pvt->okcontacturi)); + + /* Make sure it's a SIP URL */ + if (strncasecmp(c, "sip:", 4)) { + ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", c); + } else + c += 4; + + /* Ditch arguments */ + n = strchr(c, ';'); + if (n) + *n = '\0'; + + /* Grab host */ + n = strchr(c, '@'); + if (!n) { + n = c; + c = NULL; + } else { + *n = '\0'; + n++; + } + pt = strchr(n, ':'); + if (pt) { + *pt = '\0'; + pt++; + port = atoi(pt); + } else + port = DEFAULT_SIP_PORT; + + memcpy(&oldsin, &pvt->sa, sizeof(oldsin)); + + if (!(ast_test_flag(pvt, SIP_NAT) & SIP_NAT_ROUTE)) { + /* 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(n, &ahp); + if (!hp) { + ast_log(LOG_WARNING, "Invalid host '%s'\n", n); + return -1; + } + pvt->sa.sin_family = AF_INET; + memcpy(&pvt->sa.sin_addr, hp->h_addr, sizeof(pvt->sa.sin_addr)); + pvt->sa.sin_port = htons(port); + } else { + /* Don't trust the contact field. Just use what they came to us + with. */ + memcpy(&pvt->sa, &pvt->recv, sizeof(pvt->sa)); + } + return 0; +} + + +/*--- parse_contact: Parse contact header and save registration ---*/ +static int parse_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req) +{ + char contact[80]= ""; + char data[256]; + char iabuf[INET_ADDRSTRLEN]; + char *expires = get_header(req, "Expires"); + int expiry = atoi(expires); + char *c, *n, *pt; + int port; + char *useragent; + struct hostent *hp; + struct ast_hostent ahp; + struct sockaddr_in oldsin; +#ifdef SIP_TCP_SUPPORT + char *t = NULL, *q = NULL; +#endif + + if (ast_strlen_zero(expires)) { /* No expires header */ + expires = strstr(get_header(req, "Contact"), "expires="); + if (expires) { + if (sscanf(expires + 8, "%d;", &expiry) != 1) + expiry = default_expiry; + } else { + /* Nothing has been specified */ + expiry = default_expiry; + } + } + /* Look for brackets */ + ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact)); + c = get_in_brackets(contact); + + if (!strcasecmp(c, "*") || !expiry) { /* Unregister this peer */ + /* This means remove all registrations and return OK */ + memset(&p->addr, 0, sizeof(p->addr)); + if (p->expire > -1) + ast_sched_del(sched, p->expire); + p->expire = -1; + ast_db_del("SIP/Registry", p->name); + register_peer_exten(p, 0); + p->fullcontact[0] = '\0'; + p->useragent[0] = '\0'; + p->sipoptions = 0; + p->lastms = 0; + +#ifdef SIP_TCP_SUPPORT + if (pvt->ssl) { + SSL_clear(pvt->ssl); + SSL_free(pvt->ssl); + } + p->ssl = NULL; + pvt->ssl = NULL; +#endif + if (option_verbose > 2) + ast_verbose(VERBOSE_PREFIX_3 "Unregistered SIP '%s'\n", p->name); + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unregistered\r\n", p->name); + return 0; + } +#ifdef SIP_TCP_SUPPORT + /* we should remove transport, q parameter if exist */ + n = strchr(c, ';'); + if (n) + *n = '\0'; +#endif + ast_copy_string(p->fullcontact, c, sizeof(p->fullcontact)); + /* For the 200 OK, we should use the received contact */ + snprintf(pvt->our_contact, sizeof(pvt->our_contact) - 1, "<%s>", c); + /* Make sure it's a SIP URL */ + if (strncasecmp(c, "sip:", 4)) { + ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", c); + } else + c += 4; +#ifdef SIP_TCP_SUPPORT + /* transport and q parameter */ + while (n) { + n ++; + if (!strncasecmp(n, "transport=", 10)) { + t = strchr(n, '='); + t ++; + } else if (!strncasecmp(n, "q=", 2)) { + q = strchr(n, '='); + q ++; + } + n = strchr(n, ';'); + if (n) { + *n = '\0'; + } + } + if (option_verbose > 2 && t) + ast_verbose(VERBOSE_PREFIX_3 "Contact header: transport %s\n", t); +#else + /* Ditch q */ + n = strchr(c, ';'); + if (n) { + *n = '\0'; + } +#endif + /* Grab host */ + n = strchr(c, '@'); + if (!n) { + n = c; + c = NULL; + } else { + *n = '\0'; + n++; + } + pt = strchr(n, ':'); + if (pt) { + *pt = '\0'; + pt++; + port = atoi(pt); + } else + port = DEFAULT_SIP_PORT; + memcpy(&oldsin, &p->addr, sizeof(oldsin)); + if (!(ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)) { + /* 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); + return -1; + } + p->addr.sin_family = AF_INET; + memcpy(&p->addr.sin_addr, hp->h_addr, sizeof(p->addr.sin_addr)); + p->addr.sin_port = htons(port); + } else { + /* Don't trust the contact field. Just use what they came to us + with */ + memcpy(&p->addr, &pvt->recv, sizeof(p->addr)); + } +#ifdef SIP_TCP_SUPPORT + /* Check a peer has old stalled TCP or TLS connection */ + if (p->tcpsockfd > 0) { + close(p->tcpsockfd); + p->tcpsockfd = -1; + } + if (p->ssl) { + SSL_clear(p->ssl); + SSL_free(p->ssl); + p->ssl = NULL; + } + if (t) { + if(!strncasecmp(t, "tcp", strlen("tcp"))) { + ast_copy_string(p->transport, "TCP", sizeof(p->transport)); + p->tcpsockfd = pvt->sockfd; + if (pvt->ssl) + p->ssl = pvt->ssl; + } else if (!strncasecmp(t, "tls", strlen("tls"))) { + ast_copy_string(p->transport, "TLS", sizeof(p->transport)); + p->tcpsockfd = pvt->sockfd; + if (pvt->ssl) + p->ssl = pvt->ssl; + } + } + +#endif + + if (c) /* Overwrite the default username from config at registration */ + ast_copy_string(p->username, c, sizeof(p->username)); + else +#ifdef SIP_TCP_SUPPORT + ast_copy_string(p->username, p->name, sizeof(p->username)); +#else + p->username[0] = '\0'; +#endif + + if (p->expire > -1) + ast_sched_del(sched, p->expire); + if ((expiry < 1) || (expiry > max_expiry)) + expiry = max_expiry; + if (!ast_test_flag(p, SIP_REALTIME)) + p->expire = ast_sched_add(sched, (expiry + 10) * 1000, expire_register, p); + else + p->expire = -1; + pvt->expiry = expiry; + snprintf(data, sizeof(data), "%s:%d:%d:%s:%s", ast_inet_ntoa(iabuf, sizeof(iabuf), p->addr.sin_addr), ntohs(p->addr.sin_port), expiry, p->username, p->fullcontact); + if (!(ast_test_flag(p, SIP_REALTIME) && ast_test_flag((&p->flags_page2), SIP_PAGE2_RTCACHEFRIENDS))) + ast_db_put("SIP/Registry", p->name, data); + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Registered\r\n", p->name); + if (inaddrcmp(&p->addr, &oldsin)) { + sip_poke_peer(p); + if (option_verbose > 2) + ast_verbose(VERBOSE_PREFIX_3 "Registered SIP '%s' at %s port %d expires %d\n", p->name, ast_inet_ntoa(iabuf, sizeof(iabuf), p->addr.sin_addr), ntohs(p->addr.sin_port), expiry); + register_peer_exten(p, 1); + } + + /* Save SIP options profile */ + p->sipoptions = pvt->sipoptions; + + /* Save User agent */ + useragent = get_header(req, "User-Agent"); + if (useragent && strcasecmp(useragent, p->useragent)) { + ast_copy_string(p->useragent, useragent, sizeof(p->useragent)); + if (option_verbose > 3) { + ast_verbose(VERBOSE_PREFIX_3 "Saved useragent \"%s\" for peer %s\n",p->useragent,p->name); + } + } + return 0; +} + +/*--- free_old_route: 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; + } +} + +/*--- list_route: List all routes - mostly for debugging ---*/ +static void list_route(struct sip_route *route) +{ + if (!route) { + ast_verbose("list_route: no route\n"); + return; + } + while (route) { + ast_verbose("list_route: hop: <%s>\n", route->hop); + route = route->next; + } +} + +/*--- build_route: 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; + char *rr, *contact, *c; + + /* Once a persistant route is set, don't fool with it */ + if (p->route && p->route_persistant) { + 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; + } + + p->route_persistant = backwards; + + /* We build up head, then assign it to p->route when we're done */ + 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 (;;) { + /* Each route entry */ + /* Find < */ + rr = strchr(rr, '<'); + if (!rr) break; /* No more hops */ + ++rr; + len = strcspn(rr, ">") + 1; + /* Make a struct route */ + thishop = malloc(sizeof(*thishop) + len); + if (thishop) { + ast_copy_string(thishop->hop, rr, len); + 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; + } + } + rr += len; + } + } + /* 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)) { + 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; + } + thishop = malloc(sizeof(*thishop) + len); + if (thishop) { + 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); +} + +/*--- check_auth: Check user authorization from peer definition ---*/ +/* Some actions, like REGISTER and INVITEs from peers require + authentication (if peer have secret set) */ +static int check_auth(struct sip_pvt *p, struct sip_request *req, char *randdata, int randlen, char *username, char *secret, char *md5secret, int sipmethod, char *uri, int reliable, int ignore) +{ + int res = -1; + char *response = "407 Proxy Authentication Required"; + char *reqheader = "Proxy-Authorization"; + char *respheader = "Proxy-Authenticate"; + char *authtoken; +#ifdef OSP_SUPPORT + char tmp[80]; + char *osptoken; + unsigned int osptimelimit; +#endif + /* Always OK if no secret */ + if (ast_strlen_zero(secret) && ast_strlen_zero(md5secret) +#ifdef OSP_SUPPORT + && !ast_test_flag(p, SIP_OSPAUTH) + && global_allowguest != 2 +#endif + ) + return 0; + if (sipmethod == SIP_REGISTER) { + /* 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"; + } +#ifdef OSP_SUPPORT + else if (ast_test_flag(p, SIP_OSPAUTH)) { + ast_log(LOG_DEBUG, "Checking OSP Authentication!\n"); + osptoken = get_header(req, "P-OSP-Auth-Token"); + /* Check for token existence */ + if (ast_strlen_zero(osptoken)) + return -1; + /* Validate token */ + if (ast_osp_validate(NULL, osptoken, &p->osphandle, &osptimelimit, p->cid_num, p->sa.sin_addr, p->exten) < 1) + return -1; + + snprintf(tmp, sizeof(tmp), "%d", p->osphandle); + pbx_builtin_setvar_helper(p->owner, "_OSPHANDLE", tmp); + + + /* If ospauth is 'exclusive' don't require further authentication */ + if ((ast_test_flag(p, SIP_OSPAUTH) == SIP_OSPAUTH_EXCLUSIVE) || + (ast_strlen_zero(secret) && ast_strlen_zero(md5secret))) + return 0; + } +#endif + authtoken = get_header(req, reqheader); + if (ignore && !ast_strlen_zero(randdata) && ast_strlen_zero(authtoken)) { + /* This is a retransmitted invite/register/etc, don't reconstruct authentication + information */ + if (!ast_strlen_zero(randdata)) { + if (!reliable) { + /* Resend message if this was NOT a reliable delivery. Otherwise the + retransmission should get it */ + transmit_response_with_auth(p, response, req, randdata, reliable, respheader, 0); + /* Schedule auto destroy in 15 seconds */ + sip_scheddestroy(p, 15000); + } + res = 1; + } + } else if (ast_strlen_zero(randdata) || ast_strlen_zero(authtoken)) { + snprintf(randdata, randlen, "%08x", rand()); + transmit_response_with_auth(p, response, req, randdata, reliable, respheader, 0); + /* Schedule auto destroy in 15 seconds */ + sip_scheddestroy(p, 15000); + res = 1; + } else { + /* 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. */ + char a1[256]; + char a2[256]; + char a1_hash[256]; + char a2_hash[256]; + char resp[256]; + char resp_hash[256]=""; + char tmp[256] = ""; + char *c; + char *z; + char *ua_hash =""; + char *resp_uri =""; + char *nonce = ""; + char *digestusername = ""; + int wrongnonce = 0; + char *usednonce = randdata; + + /* Find their response among the mess that we'r sent for comparison */ + ast_copy_string(tmp, authtoken, sizeof(tmp)); + c = tmp; + + while(c) { + c = ast_skip_blanks(c); + if (!*c) + break; + if (!strncasecmp(c, "response=", strlen("response="))) { + c+= strlen("response="); + if ((*c == '\"')) { + ua_hash=++c; + if ((c = strchr(c,'\"'))) + *c = '\0'; + + } else { + ua_hash=c; + if ((c = strchr(c,','))) + *c = '\0'; + } + + } else if (!strncasecmp(c, "uri=", strlen("uri="))) { + c+= strlen("uri="); + if ((*c == '\"')) { + resp_uri=++c; + if ((c = strchr(c,'\"'))) + *c = '\0'; + } else { + resp_uri=c; + if ((c = strchr(c,','))) + *c = '\0'; + } + + } else if (!strncasecmp(c, "username=", strlen("username="))) { + c+= strlen("username="); + if ((*c == '\"')) { + digestusername=++c; + if((c = strchr(c,'\"'))) + *c = '\0'; + } else { + digestusername=c; + if((c = strchr(c,','))) + *c = '\0'; + } + } else if (!strncasecmp(c, "nonce=", strlen("nonce="))) { + c+= strlen("nonce="); + if ((*c == '\"')) { + nonce=++c; + if ((c = strchr(c,'\"'))) + *c = '\0'; + } else { + nonce=c; + if ((c = strchr(c,','))) + *c = '\0'; + } + + } else + if ((z = strchr(c,' ')) || (z = strchr(c,','))) c=z; + if (c) + c++; + } + /* Verify that digest username matches the username we auth as */ + if (strcmp(username, digestusername)) { + /* Oops, we're trying something here */ + return -2; + } + + /* Verify nonce from request matches our nonce. If not, send 401 with new nonce */ + if (strncasecmp(randdata, nonce, randlen)) { + wrongnonce = 1; + usednonce = nonce; + } + + snprintf(a1, sizeof(a1), "%s:%s:%s", username, global_realm, secret); + + if (!ast_strlen_zero(resp_uri)) + snprintf(a2, sizeof(a2), "%s:%s", sip_methods[sipmethod].text, resp_uri); + else + snprintf(a2, sizeof(a2), "%s:%s", sip_methods[sipmethod].text, uri); + + if (!ast_strlen_zero(md5secret)) + snprintf(a1_hash, sizeof(a1_hash), "%s", md5secret); + else + ast_md5_hash(a1_hash, a1); + + ast_md5_hash(a2_hash, a2); + + snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, usednonce, a2_hash); + ast_md5_hash(resp_hash, resp); + + if (wrongnonce) { + ast_log(LOG_NOTICE, "stale nonce received from '%s'\n", get_header(req, "To")); + + snprintf(randdata, randlen, "%08x", rand()); + if (ua_hash && !strncasecmp(ua_hash, resp_hash, strlen(resp_hash))) { + /* We got working auth token, based on stale nonce . */ + transmit_response_with_auth(p, response, req, randdata, reliable, respheader, 1); + } else { + /* Everything was wrong, so give the device one more try */ + transmit_response_with_auth(p, response, req, randdata, reliable, respheader, 0); + } + + /* Schedule auto destroy in 15 seconds */ + sip_scheddestroy(p, 15000); + return 1; + } + /* resp_hash now has the expected response, compare the two */ + if (ua_hash && !strncasecmp(ua_hash, resp_hash, strlen(resp_hash))) { + /* Auth is OK */ + res = 0; + } + } + /* Failure */ + return res; +} + +/*--- cb_extensionstate: Part of thte SUBSCRIBE support subsystem ---*/ +static int cb_extensionstate(char *context, char* exten, int state, void *data) +{ + struct sip_pvt *p = data; + if (state == -1) { + sip_scheddestroy(p, 15000); + p->stateid = -1; + return 0; + } + + transmit_state_notify(p, state, 1); + + if (option_debug > 1) + ast_verbose(VERBOSE_PREFIX_1 "Extension Changed %s new state %d for Notify User %s\n", exten, state, p->username); + return 0; +} + +/*--- register_verify: Verify registration of user */ +static int register_verify(struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req, char *uri, int ignore) +{ + int res = -1; + struct sip_peer *peer; + char tmp[256] = ""; + char iabuf[INET_ADDRSTRLEN]; + char *name, *c; + char *t; + /* Terminate URI */ + t = uri; + while(*t && (*t > 32) && (*t != ';')) + t++; + *t = '\0'; + + ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp)); + c = ditch_braces(tmp); + /* Ditch ;user=phone */ + name = strchr(c, ';'); + if (name) + *name = '\0'; + + if (!strncmp(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(iabuf, sizeof(iabuf), sin->sin_addr)); + } + /* Strip off the domain name */ + c = strchr(name, '@'); + if (c) + *c = '\0'; + ast_copy_string(p->exten, name, sizeof(p->exten)); + build_contact(p); + peer = find_peer(name, NULL, 1); + if (!(peer && ast_apply_ha(peer->ha, sin))) { + if (peer) + ASTOBJ_UNREF(peer,sip_destroy_peer); + } + if (peer) { + if (!ast_test_flag(peer, SIP_DYNAMIC)) { + ast_log(LOG_NOTICE, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name); + } else { + ast_copy_flags(p, peer, SIP_NAT); + transmit_response(p, "100 Trying", req); + if (!(res = check_auth(p, req, p->randdata, sizeof(p->randdata), peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri, 0, ignore))) { + sip_cancel_destroy(p); + if (parse_contact(p, peer, req)) { + ast_log(LOG_WARNING, "Failed to parse contact info\n"); + } else { + update_peer(peer, p->expiry); + /* Say OK and ask subsystem to retransmit msg counter */ + transmit_response_with_date(p, "200 OK", req); + peer->lastmsgssent = -1; + res = 0; + } + } + } + } + if (!peer && autocreatepeer) { + /* Create peer if we have autocreate mode enabled */ + peer = temp_peer(name); + if (peer) { + ASTOBJ_CONTAINER_LINK(&peerl, peer); + peer->lastmsgssent = -1; + sip_cancel_destroy(p); + if (parse_contact(p, peer, req)) { + ast_log(LOG_WARNING, "Failed to parse contact info\n"); + } else { + /* Say OK and ask subsystem to retransmit msg counter */ + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Registered\r\n", peer->name); + transmit_response_with_date(p, "200 OK", req); + peer->lastmsgssent = -1; + res = 0; + } + } + } + if (!res) { + ast_device_state_changed("SIP/%s", peer->name); + } + if (res < 0) { + switch (res) { + case -1: + /* Wrong password in authentication. Go away, don't try again until you fixed it */ + transmit_response(p, "403 Forbidden", &p->initreq); + break; + case -2: + /* 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; + } + } + if (peer) + ASTOBJ_UNREF(peer,sip_destroy_peer); + return res; +} + +/*--- get_rdnis: 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 = ditch_braces(tmp); + if (strncmp(c, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not an RDNIS SIP header (%s)?\n", c); + return -1; + } + c += 4; + if ((a = strchr(c, '@')) || (a = strchr(c, ';'))) { + *a = '\0'; + } + if (sip_debug_test_pvt(p)) + ast_verbose("RDNIS is %s\n", c); + ast_copy_string(p->rdnis, c, sizeof(p->rdnis)); + + return 0; +} + +/*--- get_destination: Find out who the call is for --*/ +static int get_destination(struct sip_pvt *p, struct sip_request *oreq) +{ + char tmp[256] = "", *c, *a; + char tmpf[256]= "", *fr; + struct sip_request *req; + + req = oreq; + if (!req) + req = &p->initreq; + if (req->rlPart2) + ast_copy_string(tmp, req->rlPart2, sizeof(tmp)); + c = ditch_braces(tmp); + + ast_copy_string(tmpf, get_header(req, "From"), sizeof(tmpf)); + fr = ditch_braces(tmpf); + + if (strncmp(c, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c); + return -1; + } + c += 4; + if (!ast_strlen_zero(fr)) { + if (strncmp(fr, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", fr); + return -1; + } + fr += 4; + } else + fr = NULL; + if ((a = strchr(c, '@'))) { + *a = '\0'; + a++; + ast_copy_string(p->domain, a, sizeof(p->domain)); + } + if ((a = strchr(c, ';'))) { + *a = '\0'; + } + if (fr) { + if ((a = strchr(fr, ';'))) + *a = '\0'; + if ((a = strchr(fr, '@'))) { + *a = '\0'; + ast_copy_string(p->fromdomain, a + 1, sizeof(p->fromdomain)); + } else + ast_copy_string(p->fromdomain, fr, sizeof(p->fromdomain)); + } + if (pedanticsipchecking) + url_decode(c); + if (sip_debug_test_pvt(p)) + ast_verbose("Looking for %s in %s\n", c, p->context); + if (ast_exists_extension(NULL, p->context, c, 1, fr) || + !strcmp(c, ast_pickup_ext())) { + if (!oreq) + ast_copy_string(p->exten, c, sizeof(p->exten)); + return 0; + } + + if (ast_canmatch_extension(NULL, p->context, c, 1, fr) || + !strncmp(c, ast_pickup_ext(),strlen(c))) { + return 1; + } + + return -1; +} + +/*--- get_sip_pvt_byid_locked: Lock interface lock and find matching pvt lock ---*/ +static struct sip_pvt *get_sip_pvt_byid_locked(char *callid) +{ + struct sip_pvt *sip_pvt_ptr = NULL; + + /* Search interfaces and find the match */ + ast_mutex_lock(&iflock); + sip_pvt_ptr = iflist; + while(sip_pvt_ptr) { + if (!strcmp(sip_pvt_ptr->callid, callid)) { + /* Go ahead and lock it (and its owner) before returning */ + ast_mutex_lock(&sip_pvt_ptr->lock); + if (sip_pvt_ptr->owner) { + while(ast_mutex_trylock(&sip_pvt_ptr->owner->lock)) { + ast_mutex_unlock(&sip_pvt_ptr->lock); + usleep(1); + ast_mutex_lock(&sip_pvt_ptr->lock); + if (!sip_pvt_ptr->owner) + break; + } + } + break; + } + sip_pvt_ptr = sip_pvt_ptr->next; + } + ast_mutex_unlock(&iflock); + return sip_pvt_ptr; +} + +/*--- get_refer_info: Call transfer support (the REFER method) ---*/ +static int get_refer_info(struct sip_pvt *sip_pvt, struct sip_request *outgoing_req) +{ + + char *p_refer_to = NULL, *p_referred_by = NULL, *h_refer_to = NULL, *h_referred_by = NULL, *h_contact = NULL; + char *replace_callid = "", *refer_to = NULL, *referred_by = NULL, *ptr = NULL; + struct sip_request *req = NULL; + struct sip_pvt *sip_pvt_ptr = NULL; + struct ast_channel *chan = NULL, *peer = NULL; + + req = outgoing_req; + + if (!req) { + req = &sip_pvt->initreq; + } + + if (!( (p_refer_to = get_header(req, "Refer-To")) && (h_refer_to = ast_strdupa(p_refer_to)) )) { + ast_log(LOG_WARNING, "No Refer-To Header That's illegal\n"); + return -1; + } + + refer_to = ditch_braces(h_refer_to); + + if (!( (p_referred_by = get_header(req, "Referred-By")) && (h_referred_by = ast_strdupa(p_referred_by)) )) { + ast_log(LOG_WARNING, "No Referrred-By Header That's not illegal\n"); + return -1; + } else { + referred_by = ditch_braces(h_referred_by); + } + h_contact = get_header(req, "Contact"); + + if (strncmp(refer_to, "sip:", 4)) { + ast_log(LOG_WARNING, "Refer-to: Huh? Not a SIP header (%s)?\n", refer_to); + return -1; + } + + if (strncmp(referred_by, "sip:", 4)) { + ast_log(LOG_WARNING, "Referred-by: Huh? Not a SIP header (%s) Ignoring?\n", referred_by); + referred_by = NULL; + } + + if (refer_to) + refer_to += 4; + + if (referred_by) + referred_by += 4; + + if ((ptr = strchr(refer_to, '?'))) { + /* Search for arguments */ + *ptr = '\0'; + ptr++; + if (!strncasecmp(ptr, "REPLACES=", 9)) { + char *p; + replace_callid = ast_strdupa(ptr + 9); + /* someday soon to support invite/replaces properly! + replaces_header = ast_strdupa(replace_callid); + -anthm + */ + url_decode(replace_callid); + if ((ptr = strchr(replace_callid, '%'))) + *ptr = '\0'; + if ((ptr = strchr(replace_callid, ';'))) + *ptr = '\0'; + /* Skip leading whitespace XXX memmove behaviour with overlaps ? */ + p = ast_skip_blanks(replace_callid); + if (p != replace_callid) + memmove(replace_callid, p, strlen(p)); + } + } + + if ((ptr = strchr(refer_to, '@'))) /* Skip domain (should be saved in SIPDOMAIN) */ + *ptr = '\0'; + if ((ptr = strchr(refer_to, ';'))) + *ptr = '\0'; + + if (referred_by) { + if ((ptr = strchr(referred_by, '@'))) + *ptr = '\0'; + if ((ptr = strchr(referred_by, ';'))) + *ptr = '\0'; + } + + if (sip_debug_test_pvt(sip_pvt)) { + ast_verbose("Transfer to %s in %s\n", refer_to, sip_pvt->context); + if (referred_by) + ast_verbose("Transfer from %s in %s\n", referred_by, sip_pvt->context); + } + if (!ast_strlen_zero(replace_callid)) { + /* This is a supervised transfer */ + ast_log(LOG_DEBUG,"Assigning Replace-Call-ID Info %s to REPLACE_CALL_ID\n",replace_callid); + + ast_copy_string(sip_pvt->refer_to, "", sizeof(sip_pvt->refer_to)); + ast_copy_string(sip_pvt->referred_by, "", sizeof(sip_pvt->referred_by)); + ast_copy_string(sip_pvt->refer_contact, "", sizeof(sip_pvt->refer_contact)); + sip_pvt->refer_call = NULL; + if ((sip_pvt_ptr = get_sip_pvt_byid_locked(replace_callid))) { + sip_pvt->refer_call = sip_pvt_ptr; + if (sip_pvt->refer_call == sip_pvt) { + ast_log(LOG_NOTICE, "Supervised transfer attempted to transfer into same call id (%s == %s)!\n", replace_callid, sip_pvt->callid); + sip_pvt->refer_call = NULL; + } else + return 0; + } else { + ast_log(LOG_NOTICE, "Supervised transfer requested, but unable to find callid '%s'. Both legs must reside on Asterisk box to transfer at this time.\n", replace_callid); + /* XXX The refer_to could contain a call on an entirely different machine, requiring an + INVITE with a replaces header -anthm XXX */ + /* The only way to find out is to use the dialplan - oej */ + } + } else if (ast_exists_extension(NULL, sip_pvt->context, refer_to, 1, NULL) || !strcmp(refer_to, ast_parking_ext())) { + /* This is an unsupervised transfer (blind transfer) */ + + ast_log(LOG_DEBUG,"Unsupervised transfer to (Refer-To): %s\n", refer_to); + if (referred_by) + ast_log(LOG_DEBUG,"Transferred by (Referred-by: ) %s \n", referred_by); + ast_log(LOG_DEBUG,"Transfer Contact Info %s (REFER_CONTACT)\n", h_contact); + ast_copy_string(sip_pvt->refer_to, refer_to, sizeof(sip_pvt->refer_to)); + if (referred_by) + ast_copy_string(sip_pvt->referred_by, referred_by, sizeof(sip_pvt->referred_by)); + if (h_contact) { + ast_copy_string(sip_pvt->refer_contact, h_contact, sizeof(sip_pvt->refer_contact)); + } + sip_pvt->refer_call = NULL; + if ((chan = sip_pvt->owner) && (peer = ast_bridged_channel(sip_pvt->owner))) { + pbx_builtin_setvar_helper(chan, "BLINDTRANSFER", peer->name); + pbx_builtin_setvar_helper(peer, "BLINDTRANSFER", chan->name); + } + return 0; + } else if (ast_canmatch_extension(NULL, sip_pvt->context, refer_to, 1, NULL)) { + return 1; + } + + return -1; +} + +/*--- get_also_info: Call transfer support (old way, depreciated)--*/ +static int get_also_info(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, "Also"), sizeof(tmp)); + + c = ditch_braces(tmp); + + + if (strncmp(c, "sip:", 4)) { + ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c); + return -1; + } + c += 4; + if ((a = strchr(c, '@'))) + *a = '\0'; + if ((a = strchr(c, ';'))) + *a = '\0'; + + if (sip_debug_test_pvt(p)) { + ast_verbose("Looking for %s in %s\n", c, p->context); + } + if (ast_exists_extension(NULL, p->context, c, 1, NULL)) { + /* This is an unsupervised transfer */ + ast_log(LOG_DEBUG,"Assigning Extension %s to REFER-TO\n", c); + ast_copy_string(p->refer_to, c, sizeof(p->refer_to)); + ast_copy_string(p->referred_by, "", sizeof(p->referred_by)); + ast_copy_string(p->refer_contact, "", sizeof(p->refer_contact)); + p->refer_call = NULL; + return 0; + } else if (ast_canmatch_extension(NULL, p->context, c, 1, NULL)) { + return 1; + } + + return -1; +} + +/*--- check_via: check Via: headers ---*/ +static int check_via(struct sip_pvt *p, struct sip_request *req) +{ + char via[256] = ""; + char iabuf[INET_ADDRSTRLEN]; + char *c, *pt; + struct hostent *hp; + struct ast_hostent ahp; + + memset(via, 0, sizeof(via)); + ast_copy_string(via, get_header(req, "Via"), sizeof(via)); + c = strchr(via, ';'); + if (c) + *c = '\0'; + c = strchr(via, ' '); + if (c) { + *c = '\0'; + c = ast_skip_blanks(c+1); +#ifdef SIP_TCP_SUPPORT + if (strcasecmp(via, "SIP/2.0/UDP") && strcasecmp(via, "SIP/2.0/TCP") && strcasecmp(via, "SIP/2.0/TLS")) { +#else + if (strcasecmp(via, "SIP/2.0/UDP")) { +#endif + ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via); + return -1; + } + 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 -1; + } + 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) : DEFAULT_SIP_PORT); + c = strstr(via, ";rport"); + if (c && (c[6] != '=')) /* rport query, not answer */ + ast_set_flag(p, SIP_NAT_ROUTE); + if (sip_debug_test_pvt(p)) { + c = (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) ? "NAT" : "non-NAT"; + ast_verbose("Sending to %s : %d (%s)\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port), c); + } + } + return 0; +} + +/*--- get_calleridname: Get caller id name from SIP headers ---*/ +static char *get_calleridname(char *input, char *output, size_t outputsize) +{ + char *end = strchr(input,'<'); + char *tmp = strchr(input,'\"'); + int bytes = 0; + int maxbytes = outputsize - 1; + + if (!end || (end == input)) return NULL; + /* move away from "<" */ + end--; + /* we found "name" */ + if (tmp && tmp < end) { + 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 { + /* we didn't find "name" */ + /* 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; +} + +/*--- get_rpid_num: 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(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; +} + + +/*--- check_user: Check if matching user or peer is defined ---*/ +static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, int reliable, struct sockaddr_in *sin, int ignore, char *mailbox, int mailboxlen) +{ + struct sip_user *user; + struct sip_peer *peer; + char *of, from[256] = "", *c; + char *rpid,rpid_num[50]; + char iabuf[INET_ADDRSTRLEN]; + int res = 0; + char *t; + char calleridname[50]; + int debug=sip_debug_test_addr(sin); + struct ast_variable *tmpvar = NULL, *v = NULL; + + /* Terminate URI */ + t = uri; + while(*t && (*t > 32) && (*t != ';')) + t++; + *t = '\0'; + of = get_header(req, "From"); + ast_copy_string(from, of, sizeof(from)); + memset(calleridname,0,sizeof(calleridname)); + get_calleridname(from, calleridname, sizeof(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 = ditch_braces(from); + if (ast_strlen_zero(p->exten)) { + t = uri; + if (!strncmp(t, "sip:", 4)) + t+= 4; + ast_copy_string(p->exten, t, sizeof(p->exten)); + t = strchr(p->exten, '@'); + if (t) + *t = '\0'; + if (ast_strlen_zero(p->our_contact)) + build_contact(p); + } + if (strncmp(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, '@'))) { + *c = '\0'; + if ((c = strchr(of, ':'))) + *c = '\0'; + ast_copy_string(p->cid_num, of, sizeof(p->cid_num)); + ast_shrink_phone_number(p->cid_num); + } + if (*calleridname) + ast_copy_string(p->cid_name, calleridname, sizeof(p->cid_name)); + if (ast_strlen_zero(of)) + return 0; + user = find_user(of, 1); + /* Find user based on user name in the from header */ + if (!mailbox && user && ast_apply_ha(user->ha, sin)) { + ast_copy_flags(p, user, SIP_TRUSTRPID | SIP_USECLIENTCODE | SIP_NAT | SIP_PROG_INBAND | SIP_OSPAUTH); + /* 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; + /* replace callerid if rpid found, and not restricted */ + if (!ast_strlen_zero(rpid_num) && ast_test_flag(p, SIP_TRUSTRPID)) { + if (*calleridname) + ast_copy_string(p->cid_name, calleridname, sizeof(p->cid_name)); + ast_copy_string(p->cid_num, rpid_num, sizeof(p->cid_num)); + ast_shrink_phone_number(p->cid_num); + } + + if (p->rtp) { + ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); + ast_rtp_setnat(p->rtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); + } + if (p->vrtp) { + ast_log(LOG_DEBUG, "Setting NAT on VRTP to %d\n", (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); + ast_rtp_setnat(p->vrtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); + } + if (!(res = check_auth(p, req, p->randdata, sizeof(p->randdata), user->name, user->secret, user->md5secret, sipmethod, uri, reliable, ignore))) { + sip_cancel_destroy(p); + ast_copy_flags(p, user, SIP_PROMISCREDIR | SIP_DTMF | SIP_REINVITE); + /* Copy SIP extensions profile from INVITE */ + if (p->sipoptions) + user->sipoptions = p->sipoptions; + + /* If we have a call limit, set flag */ + if (user->incominglimit) + ast_set_flag(p, SIP_CALL_LIMIT); + if (!ast_strlen_zero(user->context)) + ast_copy_string(p->context, user->context, sizeof(p->context)); + if (!ast_strlen_zero(user->cid_num) && !ast_strlen_zero(p->cid_num)) { + ast_copy_string(p->cid_num, user->cid_num, sizeof(p->cid_num)); + ast_shrink_phone_number(p->cid_num); + } + if (!ast_strlen_zero(user->cid_name) && !ast_strlen_zero(p->cid_num)) + ast_copy_string(p->cid_name, user->cid_name, sizeof(p->cid_name)); + ast_copy_string(p->username, user->name, sizeof(p->username)); + ast_copy_string(p->peersecret, user->secret, sizeof(p->peersecret)); + ast_copy_string(p->peermd5secret, user->md5secret, sizeof(p->peermd5secret)); + ast_copy_string(p->accountcode, user->accountcode, sizeof(p->accountcode)); + ast_copy_string(p->language, user->language, sizeof(p->language)); + ast_copy_string(p->musicclass, user->musicclass, sizeof(p->musicclass)); + p->amaflags = user->amaflags; + p->callgroup = user->callgroup; + p->pickupgroup = user->pickupgroup; + p->callingpres = user->callingpres; + p->capability = user->capability; + p->jointcapability = user->capability; + if (p->peercapability) + p->jointcapability &= p->peercapability; + if (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) + p->noncodeccapability |= AST_RTP_DTMF; + else + p->noncodeccapability &= ~AST_RTP_DTMF; + } + if (user && debug) + ast_verbose("Found user '%s'\n", user->name); + } else { + if (user) { + if (!mailbox && 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 */ + /* Look for peer based on the IP address we received data from */ + /* If peer is registred 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); + if (peer) { + if (debug) + ast_verbose("Found peer '%s'\n", peer->name); + /* Take the peer */ + ast_copy_flags(p, peer, SIP_TRUSTRPID | SIP_USECLIENTCODE | SIP_NAT | SIP_PROG_INBAND | SIP_OSPAUTH); + + /* 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, SIP_TRUSTRPID)) { + if (*calleridname) + ast_copy_string(p->cid_name, calleridname, sizeof(p->cid_name)); + ast_copy_string(p->cid_num, rpid_num, sizeof(p->cid_num)); + ast_shrink_phone_number(p->cid_num); + } + if (p->rtp) { + ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); + ast_rtp_setnat(p->rtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); + } + if (p->vrtp) { + ast_log(LOG_DEBUG, "Setting NAT on VRTP to %d\n", (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); + ast_rtp_setnat(p->vrtp, (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)); + } + ast_copy_string(p->peersecret, peer->secret, sizeof(p->peersecret)); + p->peersecret[sizeof(p->peersecret)-1] = '\0'; + ast_copy_string(p->peermd5secret, peer->md5secret, sizeof(p->peermd5secret)); + p->peermd5secret[sizeof(p->peermd5secret)-1] = '\0'; + p->callingpres = peer->callingpres; + if (ast_test_flag(peer, SIP_INSECURE_INVITE)) { + /* Pretend there is no required authentication */ + p->peersecret[0] = '\0'; + p->peermd5secret[0] = '\0'; + } + if (!(res = check_auth(p, req, p->randdata, sizeof(p->randdata), peer->name, p->peersecret, p->peermd5secret, sipmethod, uri, reliable, ignore))) { + ast_copy_flags(p, peer, SIP_PROMISCREDIR | SIP_DTMF | SIP_REINVITE); + /* If we have a call limit, set flag */ + if (peer->incominglimit) + ast_set_flag(p, SIP_CALL_LIMIT); + ast_copy_string(p->peername, peer->name, sizeof(p->peername)); + ast_copy_string(p->authname, peer->name, sizeof(p->authname)); + /* 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 (mailbox) + snprintf(mailbox, mailboxlen, ",%s,", peer->mailbox); + if (!ast_strlen_zero(peer->username)) { + ast_copy_string(p->username, peer->username, sizeof(p->username)); + /* Use the default username for authentication on outbound calls */ + ast_copy_string(p->authname, peer->username, sizeof(p->authname)); + } + if (!ast_strlen_zero(peer->cid_num) && !ast_strlen_zero(p->cid_num)) { + ast_copy_string(p->cid_num, peer->cid_num, sizeof(p->cid_num)); + ast_shrink_phone_number(p->cid_num); + } + if (!ast_strlen_zero(peer->cid_name) && !ast_strlen_zero(p->cid_name)) + ast_copy_string(p->cid_name, peer->cid_name, sizeof(p->cid_name)); + ast_copy_string(p->fullcontact, peer->fullcontact, sizeof(p->fullcontact)); + if (!ast_strlen_zero(peer->context)) + ast_copy_string(p->context, peer->context, sizeof(p->context)); + ast_copy_string(p->peersecret, peer->secret, sizeof(p->peersecret)); + ast_copy_string(p->peermd5secret, peer->md5secret, sizeof(p->peermd5secret)); + ast_copy_string(p->language, peer->language, sizeof(p->language)); + ast_copy_string(p->accountcode, peer->accountcode, sizeof(p->accountcode)); + p->amaflags = peer->amaflags; + p->callgroup = peer->callgroup; + p->pickupgroup = peer->pickupgroup; + p->capability = peer->capability; + p->jointcapability = peer->capability; + if (p->peercapability) + p->jointcapability &= p->peercapability; + if (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) + p->noncodeccapability |= AST_RTP_DTMF; + else + p->noncodeccapability &= ~AST_RTP_DTMF; + } + ASTOBJ_UNREF(peer,sip_destroy_peer); + } else { + if (debug) + ast_verbose("Found no matching peer or user for '%s:%d'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port)); + + /* do we allow guests? */ + if (!global_allowguest) + res = -1; /* we don't want any guests, authentication will fail */ +#ifdef OSP_SUPPORT + else if (global_allowguest == 2) { + ast_copy_flags(p, &global_flags, SIP_OSPAUTH); + res = check_auth(p, req, p->randdata, sizeof(p->randdata), "", "", "", sipmethod, uri, reliable, ignore); + } +#endif + } + + } + + if (user) + ASTOBJ_UNREF(user,sip_destroy_user); + return res; +} + +/*--- check_user: Find user ---*/ +static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, int reliable, struct sockaddr_in *sin, int ignore) +{ + return check_user_full(p, req, sipmethod, uri, reliable, sin, ignore, NULL, 0); +} + +/*--- get_msg_text: 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;xlines;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; +} + + +/*--- receive_message: Receive SIP MESSAGE method messages ---*/ +/* we handle messages within current calls currently */ +static void receive_message(struct sip_pvt *p, struct sip_request *req) +{ + char buf[1024]; + struct ast_frame f; + + if (get_msg_text(buf, sizeof(buf), req)) { + ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid); + 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); + } +} + +/*--- sip_show_inuse: CLI Command to show calls within limits set by + incominglimit ---*/ +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 = 0; + + if (argc < 3) + return RESULT_SHOWUSAGE; + + if (argc == 4 && !strcmp(argv[3],"all")) + showall = 1; + + ast_cli(fd, FORMAT, "* User name", "In use", "Limit"); + ASTOBJ_CONTAINER_TRAVERSE(&userl, 1, do { + ASTOBJ_RDLOCK(iterator); + if (iterator->incominglimit) + snprintf(ilimits, sizeof(ilimits), "%d", iterator->incominglimit); + else + ast_copy_string(ilimits, "N/A", sizeof(ilimits)); + /* Code disabled ---------------------------- + if (iterator->outgoinglimit) + snprintf(olimits, sizeof(olimits), "%d", iterator->outgoinglimit); + else + ast_copy_string(olimits, "N/A", sizeof(olimits)); + snprintf(oused, sizeof(oused), "%d", iterator->outUse); + ---------------------------------------------*/ + snprintf(iused, sizeof(iused), "%d", iterator->inUse); + if (showall || iterator->incominglimit) + 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->incominglimit) + snprintf(ilimits, sizeof(ilimits), "%d", iterator->incominglimit); + else + ast_copy_string(ilimits, "N/A", sizeof(ilimits)); + /* Code disabled ---------------------------- + if (iterator->outgoinglimit) + snprintf(olimits, sizeof(olimits), "%d", iterator->outgoinglimit); + else + ast_copy_string(olimits, "N/A", sizeof(olimits)); + snprintf(oused, sizeof(oused), "%d", iterator->outUse); + ---------------------------------------------*/ + snprintf(iused, sizeof(iused), "%d", iterator->inUse); + if (showall || iterator->incominglimit) + ast_cli(fd, FORMAT2, iterator->name, iused, ilimits); + ASTOBJ_UNLOCK(iterator); + } while (0) ); + + return RESULT_SUCCESS; +#undef FORMAT +#undef FORMAT2 +} + +/*--- nat2str: 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"; + } +} + +/*--- sip_show_users: CLI Command 'SIP Show Users' ---*/ +static int sip_show_users(int fd, int argc, char *argv[]) +{ + regex_t regexbuf; + int havepattern = 0; + +#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 = 1; + } 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, 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: Action ID for this transaction. Will be returned.\n"; + +static int _sip_show_peers(int fd, int *total, struct mansession *s, struct message *m, int argc, char *argv[]); + +/*--- manager_sip_show_peers: Show SIP peers in the manager API ---*/ +/* Inspired from chan_iax2 */ +static int manager_sip_show_peers( struct mansession *s, struct message *m ) +{ + char *id = astman_get_header(m,"ActionID"); + char *a[] = { "sip", "show", "peers" }; + char idtext[256] = ""; + int total = 0; + + if (id && !ast_strlen_zero(id)) + snprintf(idtext,256,"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(s->fd, &total, s, m, 3, a); + /* Send final confirmation */ + ast_mutex_lock(&s->lock); + ast_cli(s->fd, + "Event: PeerlistComplete\r\n" + "ListItems: %d\r\n" + "%s" + "\r\n", total, idtext); + ast_mutex_unlock(&s->lock); + return 0; +} + +/*--- sip_show_peers: CLI Show Peers command */ +static int sip_show_peers(int fd, int argc, char *argv[]) +{ + return _sip_show_peers(fd, NULL, NULL, NULL, argc, argv); +} + +/*--- _sip_show_peers: Execute sip show peers command */ +static int _sip_show_peers(int fd, int *total, struct mansession *s, struct message *m, int argc, char *argv[]) +{ + regex_t regexbuf; + int havepattern = 0; + +#define FORMAT2 "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-15.15s %-8s %-10s\n" +#define FORMAT "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-15.15s %-8d %-10s\n" + + char name[256] = ""; + char iabuf[INET_ADDRSTRLEN]; + int total_peers = 0; + int peers_online = 0; + int peers_offline = 0; + char *id; + char idtext[256] = ""; + + if (s) { /* Manager - get ActionID */ + id = astman_get_header(m,"ActionID"); + if (id && !ast_strlen_zero(id)) + snprintf(idtext,256,"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 = 1; + } 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", "Mask", "Port", "Status"); + } + + ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do { + char nm[20] = ""; + char status[20] = ""; + char srch[2000]; + + ASTOBJ_RDLOCK(iterator); + + if (havepattern && regexec(®exbuf, iterator->name, 0, NULL, 0)) { + ASTOBJ_UNLOCK(iterator); + continue; + } + + ast_inet_ntoa(nm, sizeof(nm), iterator->mask); + 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)); + if (iterator->maxms) { + if (iterator->lastms < 0) { + ast_copy_string(status, "UNREACHABLE", sizeof(status)); + peers_offline++; + } else if (iterator->lastms > iterator->maxms) { + snprintf(status, sizeof(status), "LAGGED (%d ms)", iterator->lastms); + peers_online++; + } else if (iterator->lastms) { + snprintf(status, sizeof(status), "OK (%d ms)", iterator->lastms); + peers_online++; + } else { + /* Checking if port is 0 */ + if ( ntohs(iterator->addr.sin_port) == 0 ) { + peers_offline++; + } else { + peers_online++; + } + ast_copy_string(status, "UNKNOWN", sizeof(status)); + } + } else { + ast_copy_string(status, "Unmonitored", sizeof(status)); + /* Checking if port is 0 */ + if ( ntohs(iterator->addr.sin_port) == 0 ) { + peers_offline++; + } else { + peers_online++; + } + } + + snprintf(srch, sizeof(srch), FORMAT, name, + iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), iterator->addr.sin_addr) : "(Unspecified)", + ast_test_flag(iterator, SIP_DYNAMIC) ? " D " : " ", /* Dynamic or not? */ + (ast_test_flag(iterator, SIP_NAT) & SIP_NAT_ROUTE) ? " N " : " ", /* NAT=yes? */ + iterator->ha ? " A " : " ", /* permit/deny */ + nm, ntohs(iterator->addr.sin_port), status); + + if (!s) {/* Normal CLI list */ + ast_cli(fd, FORMAT, name, + iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), iterator->addr.sin_addr) : "(Unspecified)", + ast_test_flag(iterator, SIP_DYNAMIC) ? " D " : " ", /* Dynamic or not? */ + (ast_test_flag(iterator, SIP_NAT) & SIP_NAT_ROUTE) ? " N " : " ", /* NAT=yes? */ + iterator->ha ? " A " : " ", /* permit/deny */ + nm, + ntohs(iterator->addr.sin_port), status); + } else { /* Manager format */ + /* The names here need to be the same as other channels */ + ast_mutex_lock(&s->lock); + ast_cli(fd, + "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" + "ACL: %s\r\n" + "Status: %s\r\n\r\n", + idtext, + iterator->name, + iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), iterator->addr.sin_addr) : "-none-", + ntohs(iterator->addr.sin_port), + ast_test_flag(iterator, SIP_DYNAMIC) ? "yes" : "no", /* Dynamic or not? */ + (ast_test_flag(iterator, SIP_NAT) & SIP_NAT_ROUTE) ? "yes" : "no", /* NAT=yes? */ + iterator->ha ? "yes" : "no", /* permit/deny */ + status); + + ast_mutex_unlock(&s->lock); + } + + ASTOBJ_UNLOCK(iterator); + + total_peers++; + } while(0) ); + + if (!s) { + ast_cli(fd,"%d sip peers [%d online , %d offline]\n",total_peers,peers_online,peers_offline); + } + + if (havepattern) + regfree(®exbuf); + + if (total) + *total = total_peers; + + + return RESULT_SUCCESS; +#undef FORMAT +#undef FORMAT2 +} + +/*--- sip_show_objects: List all allocated SIP Objects ---*/ +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; +} +/*--- print_group: Print call group and pickup group ---*/ +static void print_group(int fd, unsigned int group) +{ + char buf[256]; + ast_cli(fd, "%s\n", ast_print_group(buf, sizeof(buf), group) ); +} + +/*--- dtmfmode2str: 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"; + } + return ""; +} + +/*--- insecure2str: 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"; +} + +/*--- sip_prune_realtime: 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 = 0; + int prunepeer = 0; + int multi = 0; + 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 = 1; + pruneuser = prunepeer = 1; + } else { + pruneuser = prunepeer = 1; + 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 = 1; + name = argv[4]; + pruneuser = prunepeer = 1; + } else if (!strcasecmp(argv[3], "user")) { + pruneuser = 1; + if (!strcasecmp(argv[4], "all")) + multi = 1; + else + name = argv[4]; + } else if (!strcasecmp(argv[3], "peer")) { + prunepeer = 1; + if (!strcasecmp(argv[4], "all")) + multi = 1; + 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 = 1; + name = argv[5]; + } else if (!strcasecmp(argv[3], "peer")) { + prunepeer = 1; + 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_page2), 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_page2), 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_page2), 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_page2), 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; +} + +static char mandescr_show_peer[] = +"Description: Show one SIP peer with details on current status.\n" +" The XML format is under development, feedback welcome! /oej\n" +"Variables: \n" +" Peer: The peer name you want to check.\n" +" ActionID: Optional action ID for this AMI transaction.\n"; + +static int _sip_show_peer(int type, int fd, struct mansession *s, struct message *m, int argc, char *argv[]); + +/*--- manager_sip_show_peer: Show SIP peers in the manager API ---*/ +static int manager_sip_show_peer( struct mansession *s, struct message *m ) +{ + char *id = astman_get_header(m,"ActionID"); + char *a[4]; + char *peer; + int ret; + + peer = astman_get_header(m,"Peer"); + if (!peer || ast_strlen_zero(peer)) { + astman_send_error(s, m, "Peer: missing.\n"); + return 0; + } + ast_mutex_lock(&s->lock); + a[0] = "sip"; + a[1] = "show"; + a[2] = "peer"; + a[3] = peer; + + if (id && !ast_strlen_zero(id)) + ast_cli(s->fd, "ActionID: %s\r\n",id); + ret = _sip_show_peer(1, s->fd, s, m, 4, a ); + ast_cli( s->fd, "\r\n\r\n" ); + ast_mutex_unlock(&s->lock); + return ret; +} + + + +/*--- sip_show_peer: 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, argv); +} + +static int _sip_show_peer(int type, int fd, struct mansession *s, struct message *m, int argc, char *argv[]) +{ + char status[30] = ""; + char cbuf[256]; + char iabuf[INET_ADDRSTRLEN]; + 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 = 0; + + if (argc < 4) + return RESULT_SHOWUSAGE; + + load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? 1 : 0; + peer = find_peer(argv[3], NULL, load_realtime); + if (s) { /* Manager */ + if (peer) + ast_cli(s->fd, "Response: Success\r\n"); + else { + snprintf (cbuf, sizeof(cbuf), "Peer %s not found.\n", 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); + ast_cli(fd, " Secret : %s\n", ast_strlen_zero(peer->secret)?"":""); + ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(peer->md5secret)?"":""); + auth = peer->auth; + while(auth) { + 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)?"":(!ast_strlen_zero(auth->md5secret)?"" : "")); + auth = auth->next; + } + ast_cli(fd, " Context : %s\n", peer->context); + 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, " 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); + ast_cli(fd, " Pickupgroup : "); + print_group(fd, peer->pickupgroup); + ast_cli(fd, " Mailbox : %s\n", peer->mailbox); + ast_cli(fd, " LastMsgsSent : %d\n", peer->lastmsgssent); + ast_cli(fd, " Inc. limit : %d\n", peer->incominglimit); + ast_cli(fd, " Outg. limit : %d\n", peer->outgoinglimit); + ast_cli(fd, " Dynamic : %s\n", (ast_test_flag(peer, SIP_DYNAMIC)?"Yes":"No")); + ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "")); + ast_cli(fd, " Expire : %d\n", peer->expire); + ast_cli(fd, " Expiry : %d\n", peer->expiry); + ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(peer, SIP_INSECURE_PORT), ast_test_flag(peer, SIP_INSECURE_INVITE))); + ast_cli(fd, " Nat : %s\n", nat2str(ast_test_flag(peer, SIP_NAT))); + ast_cli(fd, " ACL : %s\n", (peer->ha?"Yes":"No")); + ast_cli(fd, " CanReinvite : %s\n", (ast_test_flag(peer, SIP_CAN_REINVITE)?"Yes":"No")); + ast_cli(fd, " PromiscRedir : %s\n", (ast_test_flag(peer, SIP_PROMISCREDIR)?"Yes":"No")); + ast_cli(fd, " User=Phone : %s\n", (ast_test_flag(peer, SIP_USEREQPHONE)?"Yes":"No")); + + /* - is enumerated */ + ast_cli(fd, " DTMFmode : %s\n", dtmfmode2str(ast_test_flag(peer, 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(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port)); + ast_cli(fd, " Defaddr->IP : %s Port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port)); +#ifdef SIP_TCP_SUPPORT + ast_cli(fd, " TCP fd : %d\n", peer->tcpsockfd); + ast_cli(fd, " Transport : %s\n", peer->transport); +#endif + ast_cli(fd, " Def. Username: %s\n", peer->username); + ast_cli(fd, " SIP Options : "); + if (peer->sipoptions) { + for (x=0 ; (x < (sizeof(sip_options) / sizeof(sip_options[0]))); x++) { + if (peer->sipoptions & sip_options[x].id) + ast_cli(fd, "%s ", sip_options[x].text); + } + } 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 : ("); + pref = &peer->prefs; + for(x = 0; x < 32 ; x++) { + codec = ast_codec_pref_index(pref,x); + if (!codec) + break; + ast_cli(fd, "%s", ast_getformatname(codec)); + if (x < 31 && ast_codec_pref_index(pref,x+1)) + ast_cli(fd, "|"); + } + + if (!x) + ast_cli(fd, "none"); + ast_cli(fd, ")\n"); + + ast_cli(fd, " Status : "); + if (peer->lastms < 0) + ast_copy_string(status, "UNREACHABLE", sizeof(status)); + else if (peer->lastms > peer->maxms) + snprintf(status, sizeof(status), "LAGGED (%d ms)", peer->lastms); + else if (peer->lastms) + snprintf(status, sizeof(status), "OK (%d ms)", peer->lastms); + else + ast_copy_string(status, "UNKNOWN", 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 *actionid = astman_get_header(m,"ActionID"); + + ast_cli(fd, "Channeltype: SIP\r\n"); + if (actionid) + ast_cli(fd, "ActionID: %s\r\n", actionid); + ast_cli(fd, "ObjectName: %s\r\n", peer->name); + ast_cli(fd, "ChanObjectType: peer\r\n"); + ast_cli(fd, "SecretExist: %s\r\n", ast_strlen_zero(peer->secret)?"N":"Y"); + ast_cli(fd, "MD5SecretExist: %s\r\n", ast_strlen_zero(peer->md5secret)?"N":"Y"); + ast_cli(fd, "Context: %s\r\n", peer->context); + ast_cli(fd, "Language: %s\r\n", peer->language); + if (!ast_strlen_zero(peer->accountcode)) + ast_cli(fd, "Accountcode: %s\r\n", peer->accountcode); + ast_cli(fd, "AMAflags: %s\r\n", ast_cdr_flags2str(peer->amaflags)); + ast_cli(fd, "CID-CallingPres: %s\r\n", ast_describe_caller_presentation(peer->callingpres)); + if (!ast_strlen_zero(peer->fromuser)) + ast_cli(fd, "SIP-FromUser: %s\r\n", peer->fromuser); + if (!ast_strlen_zero(peer->fromdomain)) + ast_cli(fd, "SIP-FromDomain: %s\r\n", peer->fromdomain); + ast_cli(fd, "Callgroup: "); + print_group(fd, peer->callgroup); + ast_cli(fd, "Pickupgroup: "); + print_group(fd, peer->pickupgroup); + ast_cli(fd, "VoiceMailbox: %s\r\n", peer->mailbox); + ast_cli(fd, "LastMsgsSent: %d\r\n", peer->lastmsgssent); + ast_cli(fd, "Incominglimit: %d\r\n", peer->incominglimit); + ast_cli(fd, "Outgoinglimit: %d\r\n", peer->outgoinglimit); + ast_cli(fd, "Dynamic: %s\r\n", (ast_test_flag(peer, SIP_DYNAMIC)?"Y":"N")); + ast_cli(fd, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "")); + ast_cli(fd, "RegExpire: %ld seconds\r\n", ast_sched_when(sched,peer->expire)); + ast_cli(fd, "RegExpiry: %d\r\n", peer->expiry); + ast_cli(fd, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(peer, SIP_INSECURE_PORT), ast_test_flag(peer, SIP_INSECURE_INVITE))); + ast_cli(fd, "SIP-NatSupport: %s\r\n", nat2str(ast_test_flag(peer, SIP_NAT))); + ast_cli(fd, "ACL: %s\r\n", (peer->ha?"Y":"N")); + ast_cli(fd, "SIP-CanReinvite: %s\r\n", (ast_test_flag(peer, SIP_CAN_REINVITE)?"Y":"N")); + ast_cli(fd, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(peer, SIP_PROMISCREDIR)?"Y":"N")); + ast_cli(fd, "SIP-UserPhone: %s\r\n", (ast_test_flag(peer, SIP_USEREQPHONE)?"Y":"N")); + + /* - is enumerated */ + ast_cli(fd, "SIP-DTMFmode %s\r\n", dtmfmode2str(ast_test_flag(peer, SIP_DTMF))); + ast_cli(fd, "SIPLastMsg: %d\r\n", peer->lastmsg); + ast_cli(fd, "ToHost: %s\r\n", peer->tohost); + ast_cli(fd, "Address-IP: %s\r\nAddress-Port: %d\r\n", peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "", ntohs(peer->addr.sin_port)); + ast_cli(fd, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port)); + ast_cli(fd, "Default-Username: %s\r\n", peer->username); + ast_cli(fd, "Codecs: "); + ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability); + ast_cli(fd, "%s\r\n", codec_buf); + ast_cli(fd, "CodecOrder: "); + pref = &peer->prefs; + for(x = 0; x < 32 ; x++) { + codec = ast_codec_pref_index(pref,x); + if (!codec) + break; + ast_cli(fd, "%s", ast_getformatname(codec)); + if (x < 31 && ast_codec_pref_index(pref,x+1)) + ast_cli(fd, ","); + } + + ast_cli(fd, "\r\n"); + ast_cli(fd, "Status: "); + if (peer->lastms < 0) + ast_copy_string(status, "UNREACHABLE", sizeof(status)); + else if (peer->lastms > peer->maxms) + snprintf(status, sizeof(status), "LAGGED (%d ms)", peer->lastms); + else if (peer->lastms) + snprintf(status, sizeof(status), "OK (%d ms)", peer->lastms); + else + ast_copy_string(status, "UNKNOWN", sizeof(status)); + ast_cli(fd, "%s\r\n",status); + ast_cli(fd, "SIP-Useragent: %s\r\n", peer->useragent); + ast_cli(fd, "Reg-Contact : %s\r\n", peer->fullcontact); + if (peer->chanvars) { + for (v = peer->chanvars ; v ; v = v->next) { + ast_cli(fd, "ChanVariable:\n"); + ast_cli(fd, " %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; +} + +/*--- sip_show_user: 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_codec_pref *pref; + struct ast_variable *v; + int x = 0, codec = 0, load_realtime = 0; + + if (argc < 4) + return RESULT_SHOWUSAGE; + + /* Load from realtime storage? */ + load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? 1 : 0; + + 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)?"":""); + ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(user->md5secret)?"":""); + 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, " CallingPres : %s\n", ast_describe_caller_presentation(user->callingpres)); + ast_cli(fd, " Inc. limit : %d\n", user->incominglimit); + ast_cli(fd, " Outg. limit : %d\n", user->outgoinglimit); + ast_cli(fd, " Callgroup : "); + print_group(fd, user->callgroup); + ast_cli(fd, " Pickupgroup : "); + print_group(fd, user->pickupgroup); + ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), user->cid_name, user->cid_num, "")); + ast_cli(fd, " ACL : %s\n", (user->ha?"Yes":"No")); + ast_cli(fd, " Codec Order : ("); + pref = &user->prefs; + for(x = 0; x < 32 ; x++) { + codec = ast_codec_pref_index(pref,x); + if (!codec) + break; + ast_cli(fd, "%s", ast_getformatname(codec)); + if (x < 31 && ast_codec_pref_index(pref,x+1)) + ast_cli(fd, "|"); + } + + if (!x) + ast_cli(fd, "none"); + ast_cli(fd, ")\n"); + + 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; +} + +/*--- sip_show_registry: 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\n" +#define FORMAT "%-30.30s %-12.12s %8d %-20.20s\n" + char host[80]; + + if (argc != 3) + return RESULT_SHOWUSAGE; + ast_cli(fd, FORMAT2, "Host", "Username", "Refresh", "State"); + ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do { + ASTOBJ_RDLOCK(iterator); + snprintf(host, sizeof(host), "%s:%d", iterator->hostname, iterator->portno ? iterator->portno : DEFAULT_SIP_PORT); + ast_cli(fd, FORMAT, host, iterator->username, iterator->refresh, regstate2str(iterator->regstate)); + ASTOBJ_UNLOCK(iterator); + } while(0)); + return RESULT_SUCCESS; +#undef FORMAT +#undef FORMAT2 +} + +/* Forward declaration */ +static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions); + +/*--- sip_show_channels: Show active SIP channels ---*/ +static int sip_show_channels(int fd, int argc, char *argv[]) +{ + return __sip_show_channels(fd, argc, argv, 0); +} + +/*--- sip_show_subscriptions: Show active SIP subscriptions ---*/ +static int sip_show_subscriptions(int fd, int argc, char *argv[]) +{ + return __sip_show_channels(fd, argc, argv, 1); +} + +static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions) +{ +#define FORMAT3 "%-15.15s %-10.10s %-21.21s %-15.15s\n" +#define FORMAT2 "%-15.15s %-10.10s %-11.11s %-11.11s %s %s\n" +#define FORMAT "%-15.15s %-10.10s %-11.11s %5.5d/%5.5d %-6.6s%s %s\n" + struct sip_pvt *cur; + char iabuf[INET_ADDRSTRLEN]; + int numchans = 0; + 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", "Last Msg"); + else + ast_cli(fd, FORMAT3, "Peer", "User", "Call ID", "URI"); + while (cur) { + if (!cur->subscribed && !subscriptions) { + ast_cli(fd, FORMAT, ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr), + ast_strlen_zero(cur->username) ? ( ast_strlen_zero(cur->cid_num) ? "(None)" : cur->cid_num ) : cur->username, + cur->callid, + cur->ocseq, cur->icseq, + ast_getformatname(cur->owner ? cur->owner->nativeformats : 0), + ast_test_flag(cur, SIP_NEEDDESTROY) ? "(d)" : "", + cur->lastmsg ); + numchans++; + } + if (cur->subscribed && subscriptions) { + ast_cli(fd, FORMAT3, ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr), + ast_strlen_zero(cur->username) ? ( ast_strlen_zero(cur->cid_num) ? "(None)" : cur->cid_num ) : cur->username, + cur->callid, cur->uri); + + } + cur = cur->next; + } + ast_mutex_unlock(&iflock); + if (!subscriptions) + ast_cli(fd, "%d active SIP channel(s)\n", numchans); + else + ast_cli(fd, "%d active SIP subscriptions(s)\n", numchans); + return RESULT_SUCCESS; +#undef FORMAT +#undef FORMAT2 +#undef FORMAT3 +} + +/*--- complete_sipch: Support routine for 'sip show channel' CLI ---*/ +static char *complete_sipch(char *line, char *word, int pos, int state) +{ + int which=0; + struct sip_pvt *cur; + char *c = NULL; + + ast_mutex_lock(&iflock); + cur = iflist; + while(cur) { + if (!strncasecmp(word, cur->callid, strlen(word))) { + if (++which > state) { + c = strdup(cur->callid); + break; + } + } + cur = cur->next; + } + ast_mutex_unlock(&iflock); + return c; +} + +/*--- complete_sip_peer: Do completion on peer name ---*/ +static char *complete_sip_peer(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)) { + if (flags2 && !ast_test_flag((&iterator->flags_page2), flags2)) + continue; + if (++which > state) { + result = strdup(iterator->name); + } + } + } while(0) ); + return result; +} + +/*--- complete_sip_show_peer: Support routine for 'sip show peer' CLI ---*/ +static char *complete_sip_show_peer(char *line, char *word, int pos, int state) +{ + if (pos == 3) + return complete_sip_peer(word, state, 0); + + return NULL; +} + +/*--- complete_sip_debug_peer: Support routine for 'sip debug peer' CLI ---*/ +static char *complete_sip_debug_peer(char *line, char *word, int pos, int state) +{ + if (pos == 3) + return complete_sip_peer(word, state, 0); + + return NULL; +} + +/*--- complete_sip_user: Do completion on user name ---*/ +static char *complete_sip_user(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_page2), flags2)) + continue; + if (++which > state) { + result = strdup(iterator->name); + } + } + } while(0) ); + return result; +} + +/*--- complete_sip_show_user: Support routine for 'sip show user' CLI ---*/ +static char *complete_sip_show_user(char *line, char *word, int pos, int state) +{ + if (pos == 3) + return complete_sip_user(word, state, 0); + + return NULL; +} + +/*--- complete_sipnotify: Support routine for 'sip notify' CLI ---*/ +static char *complete_sipnotify(char *line, char *word, int pos, int state) +{ + char *c = NULL; + + if (pos == 2) { + int which = 0; + char *cat; + + /* do completion for notify type */ + + if (!notify_types) + return NULL; + + cat = ast_category_browse(notify_types, NULL); + while(cat) { + if (!strncasecmp(word, cat, strlen(word))) { + if (++which > state) { + c = strdup(cat); + break; + } + } + cat = ast_category_browse(notify_types, cat); + } + return c; + } + + if (pos > 2) + return complete_sip_peer(word, state, 0); + + return NULL; +} + +/*--- complete_sip_prune_realtime_peer: Support routine for 'sip prune realtime peer' CLI ---*/ +static char *complete_sip_prune_realtime_peer(char *line, char *word, int pos, int state) +{ + if (pos == 4) + return complete_sip_peer(word, state, SIP_PAGE2_RTCACHEFRIENDS); + return NULL; +} + +/*--- complete_sip_prune_realtime_user: Support routine for 'sip prune realtime user' CLI ---*/ +static char *complete_sip_prune_realtime_user(char *line, char *word, int pos, int state) +{ + if (pos == 4) + return complete_sip_user(word, state, SIP_PAGE2_RTCACHEFRIENDS); + + return NULL; +} + +/*--- sip_show_channel: Show details of one call ---*/ +static int sip_show_channel(int fd, int argc, char *argv[]) +{ + struct sip_pvt *cur; + char iabuf[INET_ADDRSTRLEN]; + size_t len; + int found = 0; + + if (argc != 4) + return RESULT_SHOWUSAGE; + len = strlen(argv[3]); + ast_mutex_lock(&iflock); + cur = iflist; + while(cur) { + if (!strncasecmp(cur->callid, argv[3],len)) { + ast_cli(fd,"\n"); + if (cur->subscribed) + ast_cli(fd, " * Subscription\n"); + else + ast_cli(fd, " * SIP Call\n"); + ast_cli(fd, " Direction: %s\n", ast_test_flag(cur, SIP_OUTGOING)?"Outgoing":"Incoming"); + ast_cli(fd, " Call-ID: %s\n", cur->callid); + ast_cli(fd, " Our Codec Capability: %d\n", cur->capability); + ast_cli(fd, " Non-Codec Capability: %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(cur->owner ? cur->owner->nativeformats : 0) ); + ast_cli(fd, " Theoretical Address: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr), ntohs(cur->sa.sin_port)); + ast_cli(fd, " Received Address: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->recv.sin_addr), ntohs(cur->recv.sin_port)); + ast_cli(fd, " NAT Support: %s\n", nat2str(ast_test_flag(cur, SIP_NAT))); + ast_cli(fd, " Our Tag: %08d\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, SIP_NEEDDESTROY)); + ast_cli(fd, " Last Message: %s\n", cur->lastmsg); + ast_cli(fd, " Promiscuous Redir: %s\n", ast_test_flag(cur, 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, 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++; + } + cur = cur->next; + } + ast_mutex_unlock(&iflock); + if (!found) + ast_cli(fd, "No such SIP Call ID starting with '%s'\n", argv[3]); + return RESULT_SUCCESS; +} + +/*--- sip_show_channel: Show details of one call ---*/ +static int sip_show_history(int fd, int argc, char *argv[]) +{ + struct sip_pvt *cur; + struct sip_history *hist; + size_t len; + int x; + 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); + cur = iflist; + while(cur) { + if (!strncasecmp(cur->callid, argv[3],len)) { + ast_cli(fd,"\n"); + if (cur->subscribed) + ast_cli(fd, " * Subscription\n"); + else + ast_cli(fd, " * SIP Call\n"); + x = 0; + hist = cur->history; + while(hist) { + x++; + ast_cli(fd, "%d. %s\n", x, hist->event); + hist = hist->next; + } + if (!x) + ast_cli(fd, "Call '%s' has no history\n", cur->callid); + found++; + } + cur = cur->next; + } + ast_mutex_unlock(&iflock); + if (!found) + ast_cli(fd, "No such SIP Call ID starting with '%s'\n", argv[3]); + return RESULT_SUCCESS; +} + + +/*--- receive_info: Receive SIP INFO Message ---*/ +/* Doesn't read the duration of the DTMF signal */ +static void receive_info(struct sip_pvt *p, struct sip_request *req) +{ + char buf[1024] = ""; + unsigned int event; + char resp = 0; + struct ast_frame f; + char *c; + + /* Need to check the media/type */ + if (!strcasecmp(get_header(req, "Content-Type"), "application/dtmf-relay") || + !strcasecmp(get_header(req, "Content-Type"), "application/vnd.nortelnetworks.digits")) { + + /* Try getting the "signal=" part */ + if (ast_strlen_zero(c = get_sdp(req, "Signal")) && ast_strlen_zero(c = get_sdp(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 (p->owner) { /* PBX call */ + if (!ast_strlen_zero(buf)) { + if (sipdebug) + ast_verbose("* DTMF received: '%c'\n", buf[0]); + 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 < 10) { + resp = '0' + event; + } else if (event < 11) { + resp = '*'; + } else if (event < 12) { + resp = '#'; + } else if (event < 16) { + resp = 'A' + (event - 12); + } + /* Build DTMF frame and deliver to PBX for transmission to other call leg*/ + memset(&f, 0, sizeof(f)); + f.frametype = AST_FRAME_DTMF; + f.subclass = resp; + f.offset = 0; + f.data = NULL; + f.datalen = 0; + ast_queue_frame(p->owner, &f); + } + transmit_response(p, "200 OK", req); + return; + } else { + transmit_response(p, "481 Call leg/transaction does not exist", req); + ast_set_flag(p, SIP_NEEDDESTROY); + } + return; + } else if ((c = get_header(req, "X-ClientCode"))) { + /* Client code (from SNOM phone) */ + if (ast_test_flag(p, 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; + } + /* 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; +} + +/*--- sip_do_debug: Enable SIP Debugging in CLI ---*/ +static int sip_do_debug_ip(int fd, int argc, char *argv[]) +{ + struct hostent *hp; + struct ast_hostent ahp; + char iabuf[INET_ADDRSTRLEN]; + int port = 0; + char *p, *arg; + + if (argc != 4) + return RESULT_SHOWUSAGE; + arg = argv[3]; + p = strstr(arg, ":"); + if (p) { + *p = '\0'; + 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(iabuf, sizeof(iabuf), debugaddr.sin_addr)); + else + ast_cli(fd, "SIP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), debugaddr.sin_addr), port); + sipdebug = 1; + return RESULT_SUCCESS; +} + +static int sip_do_debug_peer(int fd, int argc, char *argv[]) +{ + struct sip_peer *peer; + char iabuf[INET_ADDRSTRLEN]; + if (argc != 4) + return RESULT_SHOWUSAGE; + peer = find_peer(argv[3], NULL, 1); + if (peer) { + if (peer->addr.sin_addr.s_addr) { + debugaddr.sin_family = AF_INET; + memcpy(&debugaddr.sin_addr, &peer->addr.sin_addr, sizeof(debugaddr.sin_addr)); + debugaddr.sin_port = peer->addr.sin_port; + ast_cli(fd, "SIP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), debugaddr.sin_addr), ntohs(debugaddr.sin_port)); + sipdebug = 1; + } else + ast_cli(fd, "Unable to get IP address of peer '%s'\n", argv[3]); + ASTOBJ_UNREF(peer,sip_destroy_peer); + } else + ast_cli(fd, "No such peer '%s'\n", argv[3]); + return RESULT_SUCCESS; +} + +/*--- sip_do_debug: Turn on SIP debugging (CLI command) */ +static int sip_do_debug(int fd, int argc, char *argv[]) +{ + int oldsipdebug = sipdebug; + if (argc != 2) { + if (argc != 4) + return RESULT_SHOWUSAGE; + else if (strncmp(argv[2], "ip\0", 3) == 0) + return sip_do_debug_ip(fd, argc, argv); + else if (strncmp(argv[2], "peer\0", 5) == 0) + return sip_do_debug_peer(fd, argc, argv); + else return RESULT_SHOWUSAGE; + } + sipdebug = 1; + memset(&debugaddr, 0, sizeof(debugaddr)); + if (oldsipdebug) + ast_cli(fd, "SIP Debugging re-enabled\n"); + else + ast_cli(fd, "SIP Debugging enabled\n"); + return RESULT_SUCCESS; +} + +/*--- sip_notify: 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; + + p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY); + if (!p) { + ast_log(LOG_WARNING, "Unable to build sip pvt data for notify\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, NULL); + + for (var = varlist; var; var = var->next) + add_header(&req, var->name, var->value); + + add_blank_header(&req); + /* Recalculate our side, and recalculate Call ID */ + if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip)) + memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); + build_via(p, p->via, sizeof(p->via)); + build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); + ast_cli(fd, "Sending NOTIFY of type '%s' to '%s'\n", argv[2], argv[i]); + transmit_sip_request(p, &req); + sip_scheddestroy(p, 15000); + } + + return RESULT_SUCCESS; +} +/*--- sip_do_history: Enable SIP History logging (CLI) ---*/ +static int sip_do_history(int fd, int argc, char *argv[]) +{ + if (argc != 2) { + return RESULT_SHOWUSAGE; + } + recordhistory = 1; + ast_cli(fd, "SIP History Recording Enabled (use 'sip show history')\n"); + return RESULT_SUCCESS; +} + +/*--- sip_no_history: Disable SIP History logging (CLI) ---*/ +static int sip_no_history(int fd, int argc, char *argv[]) +{ + if (argc != 3) { + return RESULT_SHOWUSAGE; + } + recordhistory = 0; + ast_cli(fd, "SIP History Recording Disabled\n"); + return RESULT_SUCCESS; +} + +/*--- sip_no_debug: Disable SIP Debugging in CLI ---*/ +static int sip_no_debug(int fd, int argc, char *argv[]) + +{ + if (argc != 3) + return RESULT_SHOWUSAGE; + sipdebug = 0; + ast_cli(fd, "SIP Debugging Disabled\n"); + return RESULT_SUCCESS; +} + +static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len); + +/*--- do_register_auth: 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 (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); +} + +/*--- do_proxy_auth: 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]; + struct sip_invite_param options; + + memset(&options, 0, sizeof(struct sip_invite_param)); + p->authtries++; + 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 */ + options.auth = digest; + options.authheader = respheader; + return transmit_invite(p, sipmethod, sipmethod == SIP_INVITE, &options, init); +} + +/*--- reply_digest: reply to authentication for outbound registrations ---*/ +/* 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; + + /* table of recognised keywords, and places where they should be copied */ + const struct x { + const char *key; + char *dst; + int dstlen; + } *i, keys[] = { + { "realm=", p->realm, sizeof(p->realm) }, + { "nonce=", p->nonce, sizeof(p->nonce) }, + { "opaque=", p->opaque, sizeof(p->opaque) }, + { "qop=", p->qop, sizeof(p->qop) }, + { "domain=", p->domain, sizeof(p->domain) }, + { NULL, 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 "); + for (i = keys; i->key != NULL; i++) + i->dst[0] = '\0'; /* init all to empty strings */ + 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_copy_string(i->dst, src, i->dstlen); + break; + } + if (i->key == NULL) /* not found, try ',' */ + strsep(&c, ","); + } + + /* Save auth data for following registrations */ + if (p->registry) { + struct sip_registry *r = p->registry; + + ast_copy_string(r->realm, p->realm, sizeof(r->realm)); + ast_copy_string(r->nonce, p->nonce, sizeof(r->nonce)); + ast_copy_string(r->domain, p->domain, sizeof(r->domain)); + ast_copy_string(r->opaque, p->opaque, sizeof(r->opaque)); + ast_copy_string(r->qop, p->qop, sizeof(r->qop)); + } + build_reply_digest(p, sipmethod, digest, digest_len); + return 0; +} + +/*--- build_reply_digest: Build reply digest ---*/ +/* 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 cnonce[80]; + char iabuf[INET_ADDRSTRLEN]; + char *username; + char *secret; + char *md5secret; + struct sip_auth *auth = (struct sip_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(iabuf, sizeof(iabuf), p->sa.sin_addr)); + + snprintf(cnonce, sizeof(cnonce), "%08x", rand()); + + /* Check if we have separate auth credentials */ + if ((auth = find_realm_authentication(authl, p->realm))) { + username = auth->username; + secret = auth->secret; + md5secret = auth->md5secret; + ast_log(LOG_NOTICE,"Using realm %s authentication for this call\n", p->realm); + } else { + /* No authentication, use peer or register= config */ + username = p->authname; + secret = p->peersecret; + md5secret = p->peermd5secret; + } + + + /* 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); + /* XXX We hard code the nonce-number to 1... What are the odds? Are we seriously going to keep + track of every nonce we've seen? Also we hard code to "auth"... XXX */ + if (!ast_strlen_zero(p->qop)) + snprintf(resp,sizeof(resp),"%s:%s:%s:%s:%s:%s",a1_hash,p->nonce, "00000001", cnonce, "auth", a2_hash); + else + snprintf(resp,sizeof(resp),"%s:%s:%s",a1_hash,p->nonce,a2_hash); + ast_md5_hash(resp_hash,resp); + /* 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\", opaque=\"%s\", qop=\"%s\", cnonce=\"%s\", nc=%s", username, p->realm, uri, p->nonce, resp_hash, p->opaque, "auth", cnonce, "00000001"); + else + snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\", opaque=\"%s\"", username, p->realm, uri, p->nonce, resp_hash, p->opaque); + + return 0; +} + + + +static char notify_usage[] = +"Usage: sip notify [...]\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 ]\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 [load]\n" +" Lists 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 \n" +" Provides detailed status on a given SIP channel.\n"; + +static char show_history_usage[] = +"Usage: sip show history \n" +" Provides detailed dialog history on a given SIP channel.\n"; + +static char show_peers_usage[] = +"Usage: sip show peers [like ]\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 [load]\n" +" Lists 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] [|all|like ]\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 debug\n" +" Enables dumping of SIP packets for debugging purposes\n\n" +" sip debug ip \n" +" Enables dumping of SIP packets to and from host.\n\n" +" sip debug peer \n" +" Enables dumping of SIP packets to and from host.\n" +" Require peer to be registered.\n"; + +static char no_debug_usage[] = +"Usage: sip no debug\n" +" Disables dumping of SIP packets for debugging purposes\n"; + +static char no_history_usage[] = +"Usage: sip no history\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" +" Shows active SIP subscriptions for extension states\n"; + +static char show_objects_usage[] = +"Usage: sip show objects\n" +" Shows status of known SIP objects\n"; + + +/*--- func_header_read: Read SIP header (dialplan function) */ +static char *func_header_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) +{ + struct sip_pvt *p; + char *content; + + if (!data) { + ast_log(LOG_WARNING, "This function requires a header name.\n"); + return NULL; + } + + ast_mutex_lock(&chan->lock); + if (chan->type != channeltype) { + ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n"); + ast_mutex_unlock(&chan->lock); + return NULL; + } + + p = chan->tech_pvt; + content = get_header(&p->initreq, data); + + if (ast_strlen_zero(content)) { + ast_mutex_unlock(&chan->lock); + return NULL; + } + + ast_copy_string(buf, content, len); + ast_mutex_unlock(&chan->lock); + + return buf; +} + + +static struct ast_custom_function sip_header_function = { + .name = "SIP_HEADER", + .synopsis = "Gets or sets the specified SIP header", + .syntax = "SIP_HEADER()", + .read = func_header_read, +}; + +/*--- function_sippeer: ${SIPPEER()} Dialplan function - reads peer data */ +static char *function_sippeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) +{ + char *ret = NULL; + struct sip_peer *peer; + char *peername, *colname; + char iabuf[INET_ADDRSTRLEN]; + + if (!(peername = ast_strdupa(data))) { + ast_log(LOG_ERROR, "Memory Error!\n"); + return ret; + } + + if ((colname = strchr(peername, ':'))) { + *colname = '\0'; + colname++; + } else { + colname = "ip"; + } + if (!(peer = find_peer(peername, NULL, 1))) + return ret; + + if (!strcasecmp(colname, "ip")) { + ast_copy_string(buf, peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "", 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, SIP_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, *ptr; + int index = 0, codec = 0; + + codecnum = strchr(colname, '['); + *codecnum = '\0'; + codecnum++; + if ((ptr = strchr(codecnum, ']'))) { + *ptr = '\0'; + } + index = atoi(codecnum); + if((codec = ast_codec_pref_index(&peer->prefs, index))) { + ast_copy_string(buf, ast_getformatname(codec), len); + } + } + ret = buf; + + ASTOBJ_UNREF(peer, sip_destroy_peer); + + return ret; +} + +struct ast_custom_function sippeer_function = { + .name = "SIPPEER", + .synopsis = "Gets SIP peer information", + .syntax = "SIPPEER([: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" + "- codec[x] Preferred codec index number 'x' (beginning with zero).\n" + "\n" +}; + +/*--- parse_moved_contact: Parse 302 Moved temporalily response */ +static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req) +{ + char tmp[256] = ""; + char *s, *e; + ast_copy_string(tmp, get_header(req, "Contact"), sizeof(tmp)); + s = ditch_braces(tmp); + e = strchr(s, ';'); + if (e) + *e = '\0'; + if (ast_test_flag(p, SIP_PROMISCREDIR)) { + if (!strncasecmp(s, "sip:", 4)) + s += 4; + e = strchr(s, '/'); + if (e) + *e = '\0'; + ast_log(LOG_DEBUG, "Found promiscuous redirection to 'SIP/%s'\n", s); + if (p->owner) + snprintf(p->owner->call_forward, sizeof(p->owner->call_forward), "SIP/%s", s); + } else { + e = strchr(tmp, '@'); + if (e) + *e = '\0'; + e = strchr(tmp, '/'); + if (e) + *e = '\0'; + if (!strncasecmp(s, "sip:", 4)) + s += 4; + ast_log(LOG_DEBUG, "Found 302 Redirect to extension '%s'\n", s); + if (p->owner) + ast_copy_string(p->owner->call_forward, s, sizeof(p->owner->call_forward)); + } +} + +/*--- check_pendings: Check pending actions on SIP call ---*/ +static void check_pendings(struct sip_pvt *p) +{ + /* Go ahead and send bye at this point */ + if (ast_test_flag(p, SIP_PENDINGBYE)) { + transmit_request_with_auth(p, SIP_BYE, 0, 1, 1); + ast_set_flag(p, SIP_NEEDDESTROY); + ast_clear_flag(p, SIP_NEEDREINVITE); + } else if (ast_test_flag(p, SIP_NEEDREINVITE)) { + 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, SIP_NEEDREINVITE); + } +} + +/*--- handle_response_register: 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 > 1) || 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, 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); + p->registry->regattempts = global_regattempts_max+1; + ast_sched_del(sched, r->timeout); + ast_set_flag(p, 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); + p->registry->regattempts = global_regattempts_max+1; + ast_set_flag(p, SIP_NEEDDESTROY); + r->call = NULL; + ast_sched_del(sched, r->timeout); + break; + case 407: /* Proxy auth */ + if ((p->authtries > 1) || 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, SIP_NEEDDESTROY); + } + 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); + p->registry->regattempts = global_regattempts_max+1; + ast_set_flag(p, 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 that isn't a register\n"); + ast_set_flag(p, SIP_NEEDDESTROY); + return 0; + } + + r->regstate=REG_STATE_REGISTERED; + manager_event(EVENT_FLAG_SYSTEM, "Registry", "Channel: SIP\r\nDomain: %s\r\nStatus: %s\r\n", r->hostname, regstate2str(r->regstate)); + r->regattempts = 0; + ast_log(LOG_DEBUG, "Registration successful\n"); + if (r->timeout > -1) { + ast_log(LOG_DEBUG, "Cancelling timeout %d\n", r->timeout); + ast_sched_del(sched, r->timeout); + } + r->timeout=-1; + r->call = NULL; + p->registry = NULL; + /* Let this one hang around until we have all the responses */ + sip_scheddestroy(p, 32000); + /* ast_set_flag(p, SIP_NEEDDESTROY); */ + + /* set us up for re-registering */ + /* figure out how long we got registered for */ + if (r->expire > -1) + ast_sched_del(sched, r->expire); + /* according to section 6.13 of RFC, contact headers override + expires headers, so check those first */ + expires = 0; + if (!ast_strlen_zero(get_header(req, "Contact"))) { + char *contact = NULL; + 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 = strstr(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 ms)\n", r->hostname, expires, expires_ms); + + r->refresh= (int) expires_ms / 1000; + + /* Schedule re-registration before we expire */ + r->expire=ast_sched_add(sched, expires_ms, sip_reregister, r); + ASTOBJ_UNREF(r, sip_registry_destroy); + } + return 1; +} + +/*--- handle_response_peerpoke: Handle qualification responses (OPTIONS) */ +static int handle_response_peerpoke(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno, int sipmethod) +{ + struct sip_peer *peer; + int pingtime; + struct timeval tv; + if (resp != 100) { + int statechanged = 0; + int newstate = 0; + peer = p->peerpoke; + gettimeofday(&tv, NULL); + pingtime = ast_tvdiff_ms(tv, peer->ps); + if (pingtime < 1) + pingtime = 1; + if ((peer->lastms < 0) || (peer->lastms > peer->maxms)) { + if (pingtime <= peer->maxms) { + ast_log(LOG_NOTICE, "Peer '%s' is now REACHABLE! (%dms / %dms)\n", peer->name, pingtime, peer->maxms); + statechanged = 1; + newstate = 1; + } + } else if ((peer->lastms > 0) && (peer->lastms <= peer->maxms)) { + if (pingtime > peer->maxms) { + ast_log(LOG_NOTICE, "Peer '%s' is now TOO LAGGED! (%dms / %dms)\n", peer->name, pingtime, peer->maxms); + statechanged = 1; + newstate = 2; + } + } + if (!peer->lastms) + statechanged = 1; + peer->lastms = pingtime; + peer->call = NULL; + if (statechanged) { + ast_device_state_changed("SIP/%s", peer->name); + if (newstate == 2) { + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Lagged\r\nTime: %d\r\n", peer->name, pingtime); + } else { + manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Reachable\r\nTime: %d\r\n", peer->name, pingtime); + } + } + + if (peer->pokeexpire > -1) + ast_sched_del(sched, peer->pokeexpire); + if (sipmethod == SIP_INVITE) /* Does this really happen? */ + transmit_request(p, SIP_ACK, seqno, 0, 0); + ast_set_flag(p, SIP_NEEDDESTROY); + + /* Try again eventually */ + if ((peer->lastms < 0) || (peer->lastms > peer->maxms)) + peer->pokeexpire = ast_sched_add(sched, DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer); + else + peer->pokeexpire = ast_sched_add(sched, DEFAULT_FREQ_OK, sip_poke_peer_s, peer); + } + return 1; +} + +/*--- handle_response: Handle SIP response in dialogue ---*/ +static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno) +{ + char *to; + char *msg, *c; + struct ast_channel *owner; + char iabuf[INET_ADDRSTRLEN]; + int sipmethod; + int res = 1; + + c = get_header(req, "Cseq"); + msg = strchr(c, ' '); /* Find method */ + if (!msg) + msg = ""; + else + msg++; + owner = p->owner; + + if (owner) + owner->hangupcause = hangup_sip2cause(resp); + + sipmethod = find_sip_method(msg); + + /* 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); + + /* Get their tag if we haven't already */ + if (ast_strlen_zero(p->theirtag)) { + to = get_header(req, "To"); + to = strcasestr(to, "tag="); + if (to) { + to += 4; + ast_copy_string(p->theirtag, to, sizeof(p->theirtag)); + to = strchr(p->theirtag, ';'); + if (to) + *to = '\0'; + } + } + if (p->peerpoke) { + /* 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" */ + + res = handle_response_peerpoke(p, resp, rest, req, ignore, seqno, sipmethod); + } else if (ast_test_flag(p, SIP_OUTGOING)) { + /* Acknowledge sequence number */ + if (p->initid > -1) { + /* Don't auto congest anymore since we've gotten something useful back */ + ast_sched_del(sched, p->initid); + p->initid = -1; + } + switch(resp) { + case 100: /* 100 Trying */ + if (sipmethod == SIP_INVITE) { + sip_cancel_destroy(p); + } + break; + case 183: /* 183 Session Progress */ + if (sipmethod == SIP_INVITE) { + sip_cancel_destroy(p); + if (!ast_strlen_zero(get_header(req, "Content-Type"))) + process_sdp(p, req); + if (p->owner) { + /* Queue a progress frame */ + ast_queue_control(p->owner, AST_CONTROL_PROGRESS); + } + } + break; + case 180: /* 180 Ringing */ + if (sipmethod == SIP_INVITE) { + sip_cancel_destroy(p); + if (p->owner) { + ast_queue_control(p->owner, AST_CONTROL_RINGING); + if (p->owner->_state != AST_STATE_UP) + ast_setstate(p->owner, AST_STATE_RINGING); + } + } + break; + case 200: /* 200 OK */ + if (sipmethod == SIP_NOTIFY) { + /* They got the notify, this is the end */ + if (p->owner) { + ast_log(LOG_WARNING, "Notify answer on an owned channel?\n"); + ast_queue_hangup(p->owner); + } else { + if (!p->subscribed) { + ast_set_flag(p, SIP_NEEDDESTROY); + } + } + } else if (sipmethod == SIP_INVITE) { + /* 200 OK on invite - someone's answering our call */ + sip_cancel_destroy(p); + if (!ast_strlen_zero(get_header(req, "Content-Type"))) + process_sdp(p, req); + + /* Parse contact header for continued conversation */ + /* When we get 200 OK, we now which device (and IP) to contact for this call */ + /* This is important when we have a SIP proxy between us and the phone */ + parse_ok_contact(p, req); + /* Save Record-Route for any later requests we make on this dialogue */ + build_route(p, req, 1); + if (p->owner) { + if (p->owner->_state != AST_STATE_UP) { +#ifdef OSP_SUPPORT + time(&p->ospstart); +#endif + ast_queue_control(p->owner, AST_CONTROL_ANSWER); + } else { + struct ast_frame af = { AST_FRAME_NULL, }; + ast_queue_frame(p->owner, &af); + } + } else /* It's possible we're getting an ACK after we've tried to disconnect + by sending CANCEL */ + ast_set_flag(p, SIP_PENDINGBYE); + p->authtries = 0; + /* If I understand this right, the branch is different for a non-200 ACK only */ + transmit_request(p, SIP_ACK, seqno, 0, 1); + check_pendings(p); + } else if (sipmethod == SIP_REGISTER) { + res = handle_response_register(p, resp, rest, req, ignore, seqno); + } + break; + case 401: /* Not www-authorized on SIP method */ + if (sipmethod == SIP_INVITE) { + /* First we ACK */ + transmit_request(p, SIP_ACK, seqno, 0, 0); + /* Then we AUTH */ + p->theirtag[0]='\0'; /* forget their old tag, so we don't match tags when getting response */ + if ((p->authtries > 1) || do_proxy_auth(p, req, "WWW-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, SIP_NEEDDESTROY); + } + } else if (p->registry && sipmethod == SIP_REGISTER) { + res = handle_response_register(p, resp, rest, req, ignore, seqno); + } 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, SIP_NEEDDESTROY); + } + break; + case 403: /* Forbidden - we failed authentication */ + if (sipmethod == SIP_INVITE) { + /* First we ACK */ + transmit_request(p, SIP_ACK, seqno, 0, 0); + ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for INVITE to '%s'\n", get_header(&p->initreq, "From")); + if (owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_set_flag(p, SIP_NEEDDESTROY); + } else if (p->registry && sipmethod == SIP_REGISTER) { + res = handle_response_register(p, resp, rest, req, ignore, seqno); + } else { + ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for %s\n", msg); + } + break; + case 404: /* Not found */ + if (p->registry && sipmethod == SIP_REGISTER) { + res = handle_response_register(p, resp, rest, req, ignore, seqno); + } else if (owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + break; + case 407: /* Proxy auth required */ + if (sipmethod == SIP_INVITE) { + /* First we ACK */ + transmit_request(p, SIP_ACK, seqno, 0, 0); + /* Then we AUTH */ + /* But only if the packet wasn't marked as ignore in handle_request */ + if (!ignore){ + p->theirtag[0]='\0'; /* forget their old tag, so we don't match tags when getting response */ + if ((p->authtries > 1) || do_proxy_auth(p, req, "Proxy-Authenticate", "Proxy-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, SIP_NEEDDESTROY); + } + } + } else if (sipmethod == SIP_BYE || sipmethod == SIP_REFER) { + 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(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port)); + ast_set_flag(p, SIP_NEEDDESTROY); + if ((p->authtries > 1) || 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, SIP_NEEDDESTROY); + } + } else if (p->registry && sipmethod == SIP_REGISTER) { + res = handle_response_register(p, resp, rest, req, ignore, seqno); + } else + ast_set_flag(p, SIP_NEEDDESTROY); + + break; + case 501: /* Not Implemented */ + if (sipmethod == SIP_INVITE) { + if (p->owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + } else + ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), msg); + break; + default: + if ((resp >= 300) && (resp < 700)) { + 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(iabuf, sizeof(iabuf), p->sa.sin_addr)); + ast_set_flag(p, SIP_ALREADYGONE); + if (p->rtp) { + /* Immediately stop RTP */ + ast_rtp_stop(p->rtp); + } + if (p->vrtp) { + /* Immediately stop VRTP */ + ast_rtp_stop(p->vrtp); + } + /* 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); + if (p->owner) + ast_queue_control(p->owner, AST_CONTROL_BUSY); + break; + case 487: + /* channel now destroyed - dec the inUse counter */ + if (ast_test_flag(p, SIP_OUTGOING)) { + update_user_counter(p, DEC_OUT_USE); + } + else { + update_user_counter(p, DEC_IN_USE); + } + break; + case 482: /* 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... */ + ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n"); + if (p->owner) + snprintf(p->owner->call_forward, sizeof(p->owner->call_forward), "Local/%s@%s", p->username, p->context); + /* 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 480: /* Temporarily Unavailable */ + case 404: /* Not Found */ + case 410: /* Gone */ + case 400: /* Bad Request */ + case 500: /* Server error */ + case 503: /* Service Unavailable */ + if (owner) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + break; + default: + /* Send hangup */ + if (owner) + ast_queue_hangup(p->owner); + break; + } + /* ACK on invite */ + if (sipmethod == SIP_INVITE) + transmit_request(p, SIP_ACK, seqno, 0, 0); + ast_set_flag(p, SIP_ALREADYGONE); + if (!p->owner) + ast_set_flag(p, SIP_NEEDDESTROY); + } else if ((resp >= 100) && (resp < 200)) { + if (sipmethod == SIP_INVITE) { + sip_cancel_destroy(p); + if (!ast_strlen_zero(get_header(req, "Content-Type"))) + 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(iabuf, sizeof(iabuf), p->sa.sin_addr)); + } + } else { + /* Not outgoing - what is it? Unsolicited replies? */ + /* When do we get here? ---------??????????------------*/ + /* INCOMING Calls */ + if (option_debug > 2) { + ast_verbose("!!!!!!!---------------************* Why are we here with this packet???? %s\n", msg); + } + if (sip_debug_test_pvt(p)) + ast_verbose("Response message is %s\n", msg); + switch(resp) { + case 200: + /* Change branch since this is a 200 response */ + if (sipmethod == SIP_INVITE) + transmit_request(p, SIP_ACK, seqno, 0, 1); + break; + case 407: + if (sipmethod == SIP_BYE || sipmethod == SIP_REFER) { + 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(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port)); + if ((p->authtries > 1) || 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, SIP_NEEDDESTROY); + } + } + break; + } + } +} + +struct sip_dual { + struct ast_channel *chan1; + struct ast_channel *chan2; + struct sip_request req; +}; + +/*--- sip_park_thread: Park SIP call support function */ +static void *sip_park_thread(void *stuff) +{ + struct ast_channel *chan1, *chan2; + struct sip_dual *d; + struct sip_request req; + int ext; + int res; + d = stuff; + chan1 = d->chan1; + chan2 = d->chan2; + copy_request(&req, &d->req); + free(d); + ast_mutex_lock(&chan1->lock); + ast_do_masquerade(chan1); + ast_mutex_unlock(&chan1->lock); + res = ast_park_call(chan1, chan2, 0, &ext); + /* Then hangup */ + ast_hangup(chan2); + ast_log(LOG_DEBUG, "Parked on extension '%d'\n", ext); + return NULL; +} + +/*--- sip_park: Park a call ---*/ +static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req) +{ + struct sip_dual *d; + struct ast_channel *chan1m, *chan2m; + pthread_t th; + chan1m = ast_channel_alloc(0); + chan2m = ast_channel_alloc(0); + if ((!chan2m) || (!chan1m)) { + if (chan1m) + ast_hangup(chan1m); + if (chan2m) + ast_hangup(chan2m); + return -1; + } + snprintf(chan1m->name, sizeof(chan1m->name), "Parking/%s", chan1->name); + /* Make formats okay */ + chan1m->readformat = chan1->readformat; + chan1m->writeformat = chan1->writeformat; + ast_channel_masquerade(chan1m, chan1); + /* Setup the extensions and such */ + ast_copy_string(chan1m->context, chan1->context, sizeof(chan1m->context)); + ast_copy_string(chan1m->exten, chan1->exten, sizeof(chan1m->exten)); + chan1m->priority = chan1->priority; + + /* We make a clone of the peer channel too, so we can play + back the announcement */ + snprintf(chan2m->name, sizeof (chan2m->name), "SIPPeer/%s",chan2->name); + /* Make formats okay */ + chan2m->readformat = chan2->readformat; + chan2m->writeformat = chan2->writeformat; + ast_channel_masquerade(chan2m, chan2); + /* Setup the extensions and such */ + ast_copy_string(chan2m->context, chan2->context, sizeof(chan2m->context)); + ast_copy_string(chan2m->exten, chan2->exten, sizeof(chan2m->exten)); + chan2m->priority = chan2->priority; + ast_mutex_lock(&chan2m->lock); + if (ast_do_masquerade(chan2m)) { + ast_log(LOG_WARNING, "Masquerade failed :(\n"); + ast_mutex_unlock(&chan2m->lock); + ast_hangup(chan2m); + return -1; + } + ast_mutex_unlock(&chan2m->lock); + d = malloc(sizeof(struct sip_dual)); + if (d) { + memset(d, 0, sizeof(*d)); + /* Save original request for followup */ + copy_request(&d->req, req); + d->chan1 = chan1m; + d->chan2 = chan2m; + if (!ast_pthread_create(&th, NULL, sip_park_thread, d)) + return 0; + free(d); + } + return -1; +} + +/*--- ast_quiet_chan: Turn off generator data */ +static void ast_quiet_chan(struct ast_channel *chan) +{ + if (chan && chan->_state == AST_STATE_UP) { + if (chan->generatordata) + ast_deactivate_generator(chan); + } +} + +/*--- attempt_transfer: Attempt transfer of SIP call ---*/ +static int attempt_transfer(struct sip_pvt *p1, struct sip_pvt *p2) +{ + int res = 0; + struct ast_channel + *chana = NULL, + *chanb = NULL, + *bridgea = NULL, + *bridgeb = NULL, + *peera = NULL, + *peerb = NULL, + *peerc = NULL, + *peerd = NULL; + + if (!p1->owner || !p2->owner) { + ast_log(LOG_WARNING, "Transfer attempted without dual ownership?\n"); + return -1; + } + chana = p1->owner; + chanb = p2->owner; + bridgea = ast_bridged_channel(chana); + bridgeb = ast_bridged_channel(chanb); + + if (bridgea) { + peera = chana; + peerb = chanb; + peerc = bridgea; + peerd = bridgeb; + } else if (bridgeb) { + peera = chanb; + peerb = chana; + peerc = bridgeb; + peerd = bridgea; + } + + if (peera && peerb && peerc && (peerb != peerc)) { + ast_quiet_chan(peera); + ast_quiet_chan(peerb); + ast_quiet_chan(peerc); + ast_quiet_chan(peerd); + + if (peera->cdr && peerb->cdr) { + peerb->cdr = ast_cdr_append(peerb->cdr, peera->cdr); + } else if (peera->cdr) { + peerb->cdr = peera->cdr; + } + peera->cdr = NULL; + + if (peerb->cdr && peerc->cdr) { + peerb->cdr = ast_cdr_append(peerb->cdr, peerc->cdr); + } else if (peerc->cdr) { + peerb->cdr = peerc->cdr; + } + peerc->cdr = NULL; + + if (ast_channel_masquerade(peerb, peerc)) { + ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", peerb->name, peerc->name); + res = -1; + } + return res; + } else { + ast_log(LOG_NOTICE, "Transfer attempted with no appropriate bridged calls to transfer\n"); + if (chana) + ast_softhangup_nolock(chana, AST_SOFTHANGUP_DEV); + if (chanb) + ast_softhangup_nolock(chanb, AST_SOFTHANGUP_DEV); + return -1; + } + return 0; +} + +/*--- handle_request_options: Handle incoming OPTIONS request */ +static int handle_request_options(struct sip_pvt *p, struct sip_request *req, int debug) +{ + int res; + + res = get_destination(p, req); + build_contact(p); + /* XXX Should we authenticate OPTIONS? XXX */ + if (ast_strlen_zero(p->context)) + strcpy(p->context, default_context); + if (res < 0) + transmit_response_with_allow(p, "404 Not Found", req, 0); + else if (res > 0) + transmit_response_with_allow(p, "484 Address Incomplete", 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. */ + if (!p->lastinvite) + ast_set_flag(p, SIP_NEEDDESTROY); + + return res; +} + +/*--- handle_request_invite: Handle incoming INVITE request */ +static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin, int *recount, char *e) +{ + int res = 1; + struct ast_channel *c=NULL; + int gotdest; + struct ast_frame af = { AST_FRAME_NULL, }; + char *supported; + char *required; + unsigned int required_profile = 0; + + /* Find out what they support */ + if (!p->sipoptions) { + supported = get_header(req, "Supported"); + if (supported) + parse_sip_options(p, supported); + } + required = get_header(req, "Required"); + if (required && !ast_strlen_zero(required)) { + required_profile = parse_sip_options(NULL, required); + if (required_profile) { /* They require something */ + /* At this point we support no extensions, so fail */ + transmit_response_with_unsupported(p, "420 Bad extension", req, required); + ast_set_flag(p, SIP_NEEDDESTROY); + return -1; + + } + } + + /* Check if this is a loop */ + /* This happens since we do not properly support SIP domain + handling yet... -oej */ + if (ast_test_flag(p, 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 */ + transmit_response(p, "482 Loop Detected", req); + /* We do NOT destroy p here, so that our response will be accepted */ + return 0; + } + if (!ignore) { + /* Use this as the basis */ + if (debug) + ast_verbose("Using INVITE request as basis request - %s\n", p->callid); + sip_cancel_destroy(p); + /* This call is no longer outgoing if it ever was */ + ast_clear_flag(p, SIP_OUTGOING); + /* This also counts as a pending invite */ + p->pendinginvite = seqno; + copy_request(&p->initreq, req); + check_via(p, req); + if (p->owner) { + /* Handle SDP here if we already have an owner */ + if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) { + if (process_sdp(p, req)) { + transmit_response(p, "488 Not acceptable here", req); + ast_set_flag(p, SIP_NEEDDESTROY); + return -1; + } + } else { + p->jointcapability = p->capability; + ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n"); + } + } + } else if (debug) + ast_verbose("Ignoring this request\n"); + if (!p->lastinvite && !ignore && !p->owner) { + /* Handle authentication if this is our first invite */ + res = check_user(p, req, SIP_INVITE, e, 1, sin, ignore); + if (res) { + if (res < 0) { + ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From")); + if (ignore) + transmit_response(p, "403 Forbidden", req); + else + transmit_response_reliable(p, "403 Forbidden", req, 1); + ast_set_flag(p, SIP_NEEDDESTROY); + } + return 0; + } + /* Process the SDP portion */ + if (!ast_strlen_zero(get_header(req, "Content-Type"))) { + if (process_sdp(p, req)) { + transmit_response(p, "488 Not acceptable here", req); + ast_set_flag(p, SIP_NEEDDESTROY); + return -1; + } + } else { + p->jointcapability = p->capability; + ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n"); + } + /* Queue NULL frame to prod ast_rtp_bridge if appropriate */ + if (p->owner) + ast_queue_frame(p->owner, &af); + /* Initialize the context if it hasn't been already */ + if (ast_strlen_zero(p->context)) + strcpy(p->context, default_context); + /* Check number of concurrent calls -vs- incoming limit HERE */ + ast_log(LOG_DEBUG, "Checking SIP call limits for device %s\n", p->username); + res = update_user_counter(p, INC_IN_USE); + if (res) { + if (res < 0) { + ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username); + if (ignore) + transmit_response(p, "480 Temporarily Unavailable (Call limit)", req); + else + transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req, 1); + ast_set_flag(p, SIP_NEEDDESTROY); + } + return 0; + } + /* Get destination right away */ + gotdest = get_destination(p, NULL); + get_rdnis(p, NULL); + extract_uri(p, req); + build_contact(p); + + if (gotdest) { + if (gotdest < 0) { + if (ignore) + transmit_response(p, "404 Not Found", req); + else + transmit_response_reliable(p, "404 Not Found", req, 1); + update_user_counter(p,DEC_IN_USE); + } else { + if (ignore) + transmit_response(p, "484 Address Incomplete", req); + else + transmit_response_reliable(p, "484 Address Incomplete", req, 1); + update_user_counter(p,DEC_IN_USE); + } + ast_set_flag(p, SIP_NEEDDESTROY); + } else { + /* If no extension was specified, use the s one */ + if (ast_strlen_zero(p->exten)) + ast_copy_string(p->exten, "s", sizeof(p->exten)); + /* Initialize tag */ + p->tag = rand(); + /* First invitation */ + c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username ); + *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_mutex_lock(&c->lock); + } + } + + } else { + if (option_debug > 1 && sipdebug) + ast_log(LOG_DEBUG, "Got a SIP re-invite for call %s\n", p->callid); + c = p->owner; + } + if (!ignore && p) + p->lastinvite = seqno; + if (c) { + switch(c->_state) { + case AST_STATE_DOWN: + transmit_response(p, "100 Trying", req); + ast_setstate(c, AST_STATE_RING); + if (strcmp(p->exten, ast_pickup_ext())) { + if (ast_pbx_start(c)) { + ast_log(LOG_WARNING, "Failed to start PBX :(\n"); + /* 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); + if (ignore) + transmit_response(p, "503 Unavailable", req); + else + transmit_response_reliable(p, "503 Unavailable", req, 1); + c = NULL; + } + } else { + ast_mutex_unlock(&c->lock); + if (ast_pickup_call(c)) { + ast_log(LOG_NOTICE, "Nothing to pick up\n"); + if (ignore) + transmit_response(p, "503 Unavailable", req); + else + transmit_response_reliable(p, "503 Unavailable", req, 1); + ast_set_flag(p, SIP_ALREADYGONE); + /* Unlock locks so ast_hangup can do its magic */ + ast_mutex_unlock(&p->lock); + ast_hangup(c); + ast_mutex_lock(&p->lock); + c = NULL; + } else { + ast_mutex_unlock(&p->lock); + ast_setstate(c, AST_STATE_DOWN); + ast_hangup(c); + ast_mutex_lock(&p->lock); + c = NULL; + } + } + break; + case AST_STATE_RING: + transmit_response(p, "100 Trying", req); + break; + case AST_STATE_RINGING: + transmit_response(p, "180 Ringing", req); + break; + case AST_STATE_UP: + transmit_response_with_sdp(p, "200 OK", req, 1); + 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); + } + } else { + if (p && !ast_test_flag(p, SIP_NEEDDESTROY)) { + if (!p->jointcapability) { + if (ignore) + transmit_response(p, "488 Not Acceptable Here (codec error)", req); + else + transmit_response_reliable(p, "488 Not Acceptable Here (codec error)", req, 1); + ast_set_flag(p, SIP_NEEDDESTROY); + } else { + ast_log(LOG_NOTICE, "Unable to create/find channel\n"); + if (ignore) + transmit_response(p, "503 Unavailable", req); + else + transmit_response_reliable(p, "503 Unavailable", req, 1); + ast_set_flag(p, SIP_NEEDDESTROY); + } + } + } + return res; +} + +/*--- handle_request_refer: Handle incoming REFER request ---*/ +static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, int *nounlock) +{ + struct ast_channel *c=NULL; + int res; + struct ast_channel *transfer_to; + + if (option_debug > 2) + ast_log(LOG_DEBUG, "SIP call transfer received for call %s (REFER)!\n", p->callid); + if (ast_strlen_zero(p->context)) + strcpy(p->context, default_context); + res = get_refer_info(p, req); + if (res < 0) + transmit_response_with_allow(p, "404 Not Found", req, 1); + else if (res > 0) + transmit_response_with_allow(p, "484 Address Incomplete", req, 1); + else { + int nobye = 0; + if (!ignore) { + if (p->refer_call) { + ast_log(LOG_DEBUG,"202 Accepted (supervised)\n"); + attempt_transfer(p, p->refer_call); + if (p->refer_call->owner) + ast_mutex_unlock(&p->refer_call->owner->lock); + ast_mutex_unlock(&p->refer_call->lock); + p->refer_call = NULL; + ast_set_flag(p, SIP_GOTREFER); + } else { + ast_log(LOG_DEBUG,"202 Accepted (blind)\n"); + c = p->owner; + if (c) { + transfer_to = ast_bridged_channel(c); + if (transfer_to) { + ast_log(LOG_DEBUG, "Got SIP blind transfer, applying to '%s'\n", transfer_to->name); + ast_moh_stop(transfer_to); + if (!strcmp(p->refer_to, ast_parking_ext())) { + /* Must release c's lock now, because it will not longer + be accessible after the transfer! */ + *nounlock = 1; + ast_mutex_unlock(&c->lock); + sip_park(transfer_to, c, req); + nobye = 1; + } else { + /* Must release c's lock now, because it will not longer + be accessible after the transfer! */ + *nounlock = 1; + ast_mutex_unlock(&c->lock); + ast_async_goto(transfer_to,p->context, p->refer_to,1); + } + } else { + ast_log(LOG_DEBUG, "Got SIP blind transfer but nothing to transfer to.\n"); + ast_queue_hangup(p->owner); + } + } + ast_set_flag(p, SIP_GOTREFER); + } + transmit_response(p, "202 Accepted", req); + transmit_notify_with_sipfrag(p, seqno); + /* Always increment on a BYE */ + if (!nobye) { + transmit_request_with_auth(p, SIP_BYE, 0, 1, 1); + ast_set_flag(p, SIP_ALREADYGONE); + } + } + } + return res; +} +/*--- handle_request_cancel: Handle incoming CANCEL request ---*/ +static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req, int debug, int ignore) +{ + + check_via(p, req); + ast_set_flag(p, SIP_ALREADYGONE); + if (p->rtp) { + /* Immediately stop RTP */ + ast_rtp_stop(p->rtp); + } + if (p->vrtp) { + /* Immediately stop VRTP */ + ast_rtp_stop(p->vrtp); + } + if (p->owner) + ast_queue_hangup(p->owner); + else + ast_set_flag(p, SIP_NEEDDESTROY); + if (p->initreq.len > 0) { + if (!ignore) + transmit_response_reliable(p, "487 Request Terminated", &p->initreq, 1); + transmit_response(p, "200 OK", req); + return 1; + } else { + transmit_response(p, "481 Call Leg Does Not Exist", req); + return 0; + } +} + +/*--- handle_request_bye: Handle incoming BYE request ---*/ +static int handle_request_bye(struct sip_pvt *p, struct sip_request *req, int debug) +{ + struct ast_channel *c=NULL; + int res; + struct ast_channel *bridged_to; + char iabuf[INET_ADDRSTRLEN]; + + copy_request(&p->initreq, req); + check_via(p, req); + ast_set_flag(p, SIP_ALREADYGONE); + if (p->rtp) { + /* Immediately stop RTP */ + ast_rtp_stop(p->rtp); + } + if (p->vrtp) { + /* Immediately stop VRTP */ + ast_rtp_stop(p->vrtp); + } + 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(iabuf, sizeof(iabuf), p->recv.sin_addr)); + if (ast_strlen_zero(p->context)) + strcpy(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_moh_stop(bridged_to); + ast_async_goto(bridged_to, p->context, p->refer_to,1); + } else + ast_queue_hangup(p->owner); + } + } else { + ast_log(LOG_WARNING, "Invalid transfer information from '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr)); + ast_queue_hangup(p->owner); + } + } else if (p->owner) + ast_queue_hangup(p->owner); + else + ast_set_flag(p, SIP_NEEDDESTROY); + transmit_response(p, "200 OK", req); + + return 1; +} + +/*--- handle_request_message: Handle incoming MESSAGE request ---*/ +static int handle_request_message(struct sip_pvt *p, struct sip_request *req, int debug, int ignore) +{ + if (p->lastinvite) { + if (!ignore) { + if (debug) + ast_verbose("Receiving message!\n"); + receive_message(p, req); + } + transmit_response(p, "200 OK", req); + } else { + transmit_response(p, "405 Method Not Allowed", req); + ast_set_flag(p, SIP_NEEDDESTROY); + } + return 1; +} +/*--- handle_request_subscribe: Handle incoming SUBSCRIBE request ---*/ +static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, struct sockaddr_in *sin, int seqno, char *e) +{ + int gotdest; + int res = 0; + struct ast_channel *c=NULL; + + if (!ignore) { + /* Use this as the basis */ + if (debug) + ast_verbose("Using latest SUBSCRIBE request as basis request\n"); + /* This call is no longer outgoing if it ever was */ + ast_clear_flag(p, SIP_OUTGOING); + copy_request(&p->initreq, req); + check_via(p, req); + } else if (debug) + ast_verbose("Ignoring this SUBSCRIBE request\n"); + + if (!p->lastinvite) { + char mailbox[256]=""; + int found = 0; + + /* Handle authentication if this is our first subscribe */ + res = check_user_full(p, req, SIP_SUBSCRIBE, e, 0, sin, ignore, mailbox, sizeof(mailbox)); + if (res) { + if (res < 0) { + ast_log(LOG_NOTICE, "Failed to authenticate user %s for SUBSCRIBE\n", get_header(req, "From")); + ast_set_flag(p, SIP_NEEDDESTROY); + } + return 0; + } + /* Initialize the context if it hasn't been already */ + if (ast_strlen_zero(p->context)) + strcpy(p->context, default_context); + /* Get destination right away */ + gotdest = get_destination(p, NULL); + build_contact(p); + if (gotdest) { + if (gotdest < 0) + transmit_response(p, "404 Not Found", req); + else + transmit_response(p, "484 Address Incomplete", req); + ast_set_flag(p, SIP_NEEDDESTROY); + } else { + /* Initialize tag */ + p->tag = rand(); + if (!strcmp(get_header(req, "Accept"), "application/dialog-info+xml")) + p->subscribed = 2; + else if (!strcmp(get_header(req, "Accept"), "application/simple-message-summary")) { + /* Looks like they actually want a mailbox */ + + /* At this point, we should check if they subscribe to a mailbox that + has the same extension as the peer or the mailbox id. If we configure + the context to be the same as a SIP domain, we could check mailbox + context as well. To be able to securely accept subscribes on mailbox + IDs, not extensions, we need to check the digest auth user to make + sure that the user has access to the mailbox. + + Since we do not act on this subscribe anyway, we might as well + accept any authenticated peer with a mailbox definition in their + config section. + + */ + if (!ast_strlen_zero(mailbox)) { + found++; + } + + if (found){ + transmit_response(p, "200 OK", req); + ast_set_flag(p, SIP_NEEDDESTROY); + } else { + transmit_response(p, "403 Forbidden", req); + ast_set_flag(p, SIP_NEEDDESTROY); + } + return 0; + } else + p->subscribed = 1; + if (p->subscribed) + p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, p); + } + } else + c = p->owner; + + if (!ignore && p) + p->lastinvite = seqno; + if (p && !ast_test_flag(p, SIP_NEEDDESTROY)) { + if (!(p->expiry = atoi(get_header(req, "Expires")))) { + transmit_response(p, "200 OK", req); + ast_set_flag(p, SIP_NEEDDESTROY); + return 0; + } + /* The next line can be removed if the SNOM200 Expires bug is fixed */ + if (p->subscribed == 1) { + if (p->expiry>max_expiry) + p->expiry = max_expiry; + } + /* Go ahead and free RTP port */ + if (p->rtp) { + if (p->owner) { + p->owner->fds[0] = -1; + p->owner->fds[1] = -1; + } + ast_rtp_destroy(p->rtp); + p->rtp = NULL; + } + if (p->vrtp) { + if (p->owner) { + p->owner->fds[2] = -1; + p->owner->fds[3] = -1; + } + ast_rtp_destroy(p->vrtp); + p->vrtp = NULL; + } + transmit_response(p, "200 OK", req); + sip_scheddestroy(p, (p->expiry+10)*1000); + transmit_state_notify(p, ast_extension_state(NULL, p->context, p->exten),1); + } + return 1; +} + +/*--- handle_request_register: Handle incoming REGISTER request ---*/ +static int handle_request_register(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, struct sockaddr_in *sin, char *e) +{ + int res = 0; + char iabuf[INET_ADDRSTRLEN]; + + /* Use this as the basis */ + if (debug) + ast_verbose("Using latest request as basis request\n"); + copy_request(&p->initreq, req); + check_via(p, req); + if ((res = register_verify(p, sin, req, e, ignore)) < 0) + ast_log(LOG_NOTICE, "Registration from '%s' failed for '%s'\n", get_header(req, "To"), ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr)); + if (res < 1) { + /* Go ahead and free RTP port */ + if (p->rtp) { + if (p->owner) { + p->owner->fds[0] = -1; + p->owner->fds[1] = -1; + } + ast_rtp_destroy(p->rtp); + p->rtp = NULL; + } + if (p->vrtp) { + if (p->owner) { + p->owner->fds[2] = -1; + p->owner->fds[3] = -1; + } + ast_rtp_destroy(p->vrtp); + p->vrtp = NULL; + } + /* Destroy the session, but keep us around for just a bit in case they don't + get our 200 OK */ + sip_scheddestroy(p, 15*1000); + } + return res; +} +/*--- handle_request: Handle SIP requests (methods) ---*/ +/* this is where all incoming requests go first */ +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 */ + struct sip_request resp; + char *cmd; + char *cseq; + char *from; + char *useragent; + int seqno; + int len; + int ignore=0; + int respid; + int res = 0; + char iabuf[INET_ADDRSTRLEN]; + int debug = sip_debug_test_pvt(p); + char *e; + + /* Clear out potential response */ + memset(&resp, 0, sizeof(resp)); + + /* 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)) + return -1; + if (sscanf(cseq, "%d%n", &seqno, &len) != 1) { + ast_log(LOG_DEBUG, "No seqno in '%s'\n", cmd); + return -1; + } + /* Get the command XXX */ + + cmd = req->rlPart1; + e = req->rlPart2; + + /* Save useragent of the client */ + useragent = get_header(req, "User-Agent"); + ast_copy_string(p->useragent, useragent, sizeof(p->useragent)); + + /* Find out SIP method for incoming request */ + if (!strcasecmp(cmd, "SIP/2.0")) { /* Response to our request */ + p->method = SIP_RESPONSE; + /* Response to our request -- Do some sanity checks */ + if (!p->initreq.headers) { + 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, SIP_NEEDDESTROY); + return 0; + } else if (p->ocseq && (p->ocseq < seqno)) { + 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)) { + /* ignore means "don't do anything with it" but still have to + respond appropriately */ + ignore=1; + } + + extract_uri(p, req); + e = ast_skip_blanks(e); + if (sscanf(e, "%d %n", &respid, &len) != 1) { + ast_log(LOG_WARNING, "Invalid response: '%s'\n", e); + } else { + handle_response(p, respid, e + len, req,ignore, seqno); + } + return 0; + } + /* XXX what if not SIP/2.0 ? */ + /* New SIP request coming in + (could be new request in existing SIP dialog as well...) + */ + p->method = find_sip_method(cmd); /* Find out which SIP method they are using */ + if (option_debug > 2) + 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)) { + ast_log(LOG_DEBUG, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq); + return -1; + } else if (p->icseq && (p->icseq == seqno) && (strcasecmp(cmd, "CANCEL") || ast_test_flag(p, 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=1; + } + + 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)) { + from = get_header(req, "From"); + from = strcasestr(from, "tag="); + if (from) { + from += 4; + ast_copy_string(p->theirtag, from, sizeof(p->theirtag)); + from = strchr(p->theirtag, ';'); + if (from) + *from = '\0'; + } + } + snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd); + + /* Handle various incoming SIP methods in requests */ + switch (p->method) { + case SIP_OPTIONS: + res = handle_request_options(p, req, debug); + break; + case SIP_INVITE: + res = handle_request_invite(p, req, debug, ignore, seqno, sin, recount, e); + break; + case SIP_REFER: + res = handle_request_refer(p, req, debug, ignore, seqno, nounlock); + break; + case SIP_CANCEL: + res = handle_request_cancel(p, req, debug, ignore); + break; + case SIP_BYE: + res = handle_request_bye(p, req, debug); + break; + case SIP_MESSAGE: + res = handle_request_message(p, req, debug, ignore); + break; + case SIP_SUBSCRIBE: + res = handle_request_subscribe(p, req, debug, ignore, sin, seqno, e); + break; + case SIP_REGISTER: + res = handle_request_register(p, req, debug, ignore, sin, e); + break; + case SIP_INFO: + if (!ignore) { + if (debug) + ast_verbose("Receiving DTMF!\n"); + receive_info(p, req); + } else { /* if ignoring, transmit response */ + transmit_response(p, "200 OK", req); + } + break; + case SIP_NOTIFY: + /* XXX we get NOTIFY's from some servers. WHY?? Maybe we should + look into this someday XXX */ + transmit_response(p, "200 OK", req); + if (!p->lastinvite) + ast_set_flag(p, SIP_NEEDDESTROY); + break; + case SIP_ACK: + /* Make sure we don't ignore this */ + if (seqno == p->pendinginvite) { + p->pendinginvite = 0; + __sip_ack(p, seqno, FLAG_RESPONSE, -1); + if (!ast_strlen_zero(get_header(req, "Content-Type"))) { + if (process_sdp(p, req)) + return -1; + } + check_pendings(p); + } + if (!p->lastinvite && ast_strlen_zero(p->randdata)) + ast_set_flag(p, 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(iabuf, sizeof(iabuf), 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, SIP_NEEDDESTROY); + break; + } + return res; +} + +/*--- sipsock_read: Read data from SIP socket ---*/ +/* 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; + int nounlock; + int recount = 0; + int debug; + char iabuf[INET_ADDRSTRLEN]; +#ifdef SIP_TCP_SUPPORT + SSL *ssl = (SSL *)ignore; +#endif + + len = sizeof(sin); + memset(&req, 0, sizeof(req)); + +#ifdef SIP_TCP_SUPPORT + if (fd != sipsock) { /* It is TCP socket */ + res = getpeername(fd, (struct sockaddr *)&sin, &len); + if (res < 0) + ast_log(LOG_WARNING, "SIP TCP getpeername error: %s\n", strerror(errno)); + + if (ssl){ /* This must be TLS connection socket */ + /* ToDo: TLS read needs check the whole SIP message is read or not + by looking at double CRLF and Contect-Length header */ + + res = SSL_read(ssl, req.data, sizeof(req.data) - 1); + + switch(SSL_get_error(ssl, res)) { + case SSL_ERROR_NONE: + break; + case SSL_ERROR_ZERO_RETURN: /* The peer closed the connection */ + ast_log(LOG_NOTICE, "sipsock_read: The peer closed TLS Connection\n"); + return 0; + break; + case SSL_ERROR_SYSCALL: + ast_log(LOG_ERROR, "sipsock_read: TLS SSL_ERROR_SYSCALL\n"); + return 0; + break; + default: + ast_log(LOG_ERROR, "sipsock_read: TLS read error %d, %s\n", SSL_get_error(ssl, res), strerror(errno)); + return 0; + break; + } + } else { /* This must be TCP connection socket */ + /* ToDo: TCP read needs check the whole SIP message is read or not + by looking at double CRLF and Contect-Length header */ + + res = read(fd, req.data, sizeof(req.data) - 1); + if (res == 0) { /* The client closed the TCP connection? */ + ast_log(LOG_NOTICE, "SIP TCP connection closed : fd %d\n", fd); + close(fd); + return 0; /* Remove it */ + } + else if (res < 0) { + ast_log(LOG_ERROR, "SIP TCP read error: %s\n", strerror(errno)); + close(fd); + return 0; /* Remove it */ + } + } + } else +#endif + 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; + } + req.data[res] = '\0'; + req.len = res; + debug = sip_debug_test_addr(&sin); + if (pedanticsipchecking) + req.len = lws2sws(req.data, req.len); + if (debug) +#ifdef SIP_TCP_SUPPORT + ast_verbose("\n<-- Sip read from %s:%d:%s \n%s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), fd == sipsock ? "UDP" : "TCP", req.data); +#else + ast_verbose("\n<-- SIP read from %s:%d: \n%s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), req.data); +#endif + parse(&req); + if (debug) { + ast_verbose("--- (%d headers %d lines)", req.headers, req.lines); + if (req.headers + req.lines == 0) + ast_verbose(" Nat keepalive "); + ast_verbose("---\n"); + } + + if (req.headers < 2) { + /* Must have at least two headers */ + return 1; + } + + /* Determine the request URI for sip, sips or tel URIs */ + if (determine_firstline_parts(&req) < 0) + return 1; + + /* Process request, with netlock held */ +retrylock: + ast_mutex_lock(&netlock); + p = find_call(&req, &sin, find_sip_method(req.rlPart1)); + if (p) { + /* Go ahead and lock the owner if it has one -- we may need it */ + if (p->owner && ast_mutex_trylock(&p->owner->lock)) { + ast_log(LOG_DEBUG, "Failed to grab lock, trying again...\n"); + ast_mutex_unlock(&p->lock); + ast_mutex_unlock(&netlock); + /* Sleep infintismly short amount of time */ + usleep(1); + goto retrylock; + } + memcpy(&p->recv, &sin, sizeof(p->recv)); + if (recordhistory) { + char tmp[80] = ""; + /* This is a response, note what it was for */ + snprintf(tmp, sizeof(tmp), "%s / %s", req.data, get_header(&req, "CSeq")); + append_history(p, "Rx", tmp); + } +#ifdef SIP_TCP_SUPPORT + p->sockfd = fd; /* Save socket fd to send a response for TCP */ + if (fd != sipsock) { + if (ssl) { + p->ssl = ssl; /* Only TLS will have non-null ssl */ + ast_copy_string(p->transport, "TLS", sizeof(p->transport)); + } else + ast_copy_string(p->transport, "TCP", sizeof(p->transport)); + } +#endif + nounlock = 0; + handle_request(p, &req, &sin, &recount, &nounlock); + if (p->owner && !nounlock) + ast_mutex_unlock(&p->owner->lock); + ast_mutex_unlock(&p->lock); + } + ast_mutex_unlock(&netlock); + if (recount) + ast_update_use_count(); + + return 1; +} + +/*--- sip_send_mwi_to_peer: Send message waiting indication ---*/ +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; + + /* Check for messages */ + ast_app_messagecount(peer->mailbox, &newmsgs, &oldmsgs); + + time(&peer->lastmsgcheck); + + /* Return now if it's the same thing we told them last time */ + if (((newmsgs << 8) | (oldmsgs)) == peer->lastmsgssent) { + return 0; + } + + p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY); + if (!p) { + ast_log(LOG_WARNING, "Unable to build sip pvt data for MWI\n"); + return -1; + } + peer->lastmsgssent = ((newmsgs << 8) | (oldmsgs)); + 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)) + memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); + build_via(p, p->via, sizeof(p->via)); + build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); + /* Send MWI */ + ast_set_flag(p, SIP_OUTGOING); + transmit_notify_with_mwi(p, newmsgs, oldmsgs); + sip_scheddestroy(p, 15000); + return 0; +} + +#ifdef SIP_TCP_SUPPORT +static int siptcp_accept(int *id, int fd, short events, void *ignore) +{ + int tcpconnfd = -1; + struct sockaddr_in cliaddr; + socklen_t sa_len; + char iabuf[INET_ADDRSTRLEN]; + + sa_len = sizeof(cliaddr); + if ((tcpconnfd = accept(siptcpsock, (struct sockaddr *)&cliaddr, &sa_len)) < 0) { + ast_log(LOG_WARNING, "Failed to accept SIP TCP connection from TCP listening sock %d : %s\n", + siptcpsock, strerror(errno)); + return 1; + } + + if (sipdebug) + ast_verbose(VERBOSE_PREFIX_2 "Accepted TCP connection fd %d from %s:%d\n", + tcpconnfd, ast_inet_ntoa(iabuf, sizeof(iabuf), cliaddr.sin_addr), ntohs(cliaddr.sin_port)); + ast_io_add(io, tcpconnfd, sipsock_read, AST_IO_IN, NULL); + return 1; +} + +static int siptls_accept(int *id, int fd, short events, void *ignore) +{ + int ret, tlsconnfd = -1; + struct sockaddr_in cliaddr; + socklen_t sa_len; + char iabuf[INET_ADDRSTRLEN]; + BIO *bio = NULL; + SSL *ssl = NULL; + + sa_len = sizeof(cliaddr); + if ((tlsconnfd = accept(siptlssock, (struct sockaddr *)&cliaddr, &sa_len)) < 0) { + ast_log(LOG_WARNING, "Failed to accept SIP TLS connection from TLS listening sock %d : %s\n", + siptlssock, strerror(errno)); + return 1; + } + + /* Initiate TLS handshake */ + bio = BIO_new_socket(tlsconnfd, BIO_CLOSE); + if (!(ssl = SSL_new(tlsctx))) { + ast_log(LOG_ERROR, "SSL_new error : %s\n", ERR_reason_error_string(ERR_get_error())); + return 2; + } + SSL_set_bio(ssl, bio, bio); + + if((ret = SSL_accept(ssl) <= 0)) { + ast_log(LOG_ERROR, "SSL accept error : %s\n", ERR_reason_error_string(ERR_get_error())); + SSL_clear(ssl); + SSL_free(ssl); + return 3; + } + + /* ToDo : Client authentication code */ + + if (sipdebug) + ast_verbose(VERBOSE_PREFIX_2 "Accepted TLS connection fd %d from %s:%d\n", + tlsconnfd, ast_inet_ntoa(iabuf, sizeof(iabuf), cliaddr.sin_addr), ntohs(cliaddr.sin_port)); + ast_io_add(io, tlsconnfd, sipsock_read, AST_IO_IN, ssl); + return 1; +} +#endif + +/*--- do_monitor: The SIP monitoring thread ---*/ +static void *do_monitor(void *data) +{ + int res; + struct sip_pvt *sip; + struct sip_peer *peer = NULL; + time_t t; + int fastrestart =0; + int lastpeernum = -1; + int curpeernum; + int reloading; + + /* Add an I/O event to our UDP socket */ + if (sipsock > -1) + ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL); + +#ifdef SIP_TCP_SUPPORT + if (siptcpsock > -1) + ast_io_add(io, siptcpsock, siptcp_accept, AST_IO_IN, NULL); + if (siptlssock > -1) + ast_io_add(io, siptlssock, siptls_accept, AST_IO_IN, NULL); +#endif + + /* This thread monitors all the frame relay interfaces which are not yet in use + (and thus do not have a separate thread) indefinitely */ + /* 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 = 0; + ast_mutex_unlock(&sip_reload_lock); + if (reloading) { + if (option_verbose > 0) + ast_verbose(VERBOSE_PREFIX_1 "Reloading SIP\n"); + sip_do_reload(); + } + /* Check for interfaces needing to be killed */ + ast_mutex_lock(&iflock); +restartsearch: + time(&t); + sip = iflist; + while(sip) { + ast_mutex_lock(&sip->lock); + if (sip->rtp && sip->owner && (sip->owner->_state == AST_STATE_UP) && !sip->redirip.sin_addr.s_addr) { + if (sip->lastrtptx && sip->rtpkeepalive && t > sip->lastrtptx + sip->rtpkeepalive) { + /* Need to send an empty RTP packet */ + time(&sip->lastrtptx); + ast_rtp_sendcng(sip->rtp, 0); + } + if (sip->lastrtprx && (sip->rtptimeout || sip->rtpholdtimeout) && t > sip->lastrtprx + sip->rtptimeout) { + /* 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 || + (sip->rtpholdtimeout && + (t > sip->lastrtprx + sip->rtpholdtimeout))) { + /* Needs a hangup */ + if (sip->rtptimeout) { + while(sip->owner && ast_mutex_trylock(&sip->owner->lock)) { + ast_mutex_unlock(&sip->lock); + usleep(1); + ast_mutex_lock(&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(sip->owner, AST_SOFTHANGUP_DEV); + ast_mutex_unlock(&sip->owner->lock); + } + } + } + } + } + if (ast_test_flag(sip, SIP_NEEDDESTROY) && !sip->packets && !sip->owner) { + ast_mutex_unlock(&sip->lock); + __sip_destroy(sip, 1); + goto restartsearch; + } + ast_mutex_unlock(&sip->lock); + sip = sip->next; + } + ast_mutex_unlock(&iflock); + /* Don't let anybody kill us right away. Nobody should lock the interface list + and wait for the monitor list, but the other way around is okay. */ + ast_mutex_lock(&monlock); + /* Lock the network interface */ + ast_mutex_lock(&netlock); + /* Okay, now that we know what to do, release the network lock */ + ast_mutex_unlock(&netlock); + /* And from now on, we're okay to be killed, so release the monitor lock as well */ + ast_mutex_unlock(&monlock); + 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); + ast_mutex_lock(&monlock); + if (res >= 0) + ast_sched_runq(sched); + + /* needs work to send mwi to realtime peers */ + time(&t); + fastrestart = 0; + curpeernum = 0; + peer = NULL; + ASTOBJ_CONTAINER_TRAVERSE(&peerl, !peer, do { + if ((curpeernum > lastpeernum) && !ast_strlen_zero(iterator->mailbox) && ((t - iterator->lastmsgcheck) > global_mwitime)) { + fastrestart = 1; + lastpeernum = curpeernum; + peer = ASTOBJ_REF(iterator); + }; + curpeernum++; + } while (0) + ); + 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; + +} + +/*--- restart_monitor: Start the channel monitor thread ---*/ +static int restart_monitor(void) +{ + pthread_attr_t attr; + /* If we're supposed to be stopped -- stay stopped */ + if (monitor_thread == AST_PTHREADT_STOP) + return 0; + if (ast_mutex_lock(&monlock)) { + ast_log(LOG_WARNING, "Unable to lock monitor\n"); + return -1; + } + 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 { + pthread_attr_init(&attr); + pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED); + /* Start a new monitor */ + if (ast_pthread_create(&monitor_thread, &attr, 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; +} + +/*--- sip_poke_noanswer: No answer to Qualify poke ---*/ +static int sip_poke_noanswer(void *data) +{ + struct sip_peer *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); + /* Try again quickly */ + peer->pokeexpire = ast_sched_add(sched, DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer); + return 0; +} + +/*--- sip_poke_peer: Check availability of peer, also keep NAT open ---*/ +/* This is done with the interval in qualify= option in sip.conf */ +/* Default is 2 seconds */ +static int sip_poke_peer(struct sip_peer *peer) +{ + struct sip_pvt *p; + 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 (peer->pokeexpire > -1) + ast_sched_del(sched, peer->pokeexpire); + peer->lastms = 0; + peer->pokeexpire = -1; + peer->call = NULL; + return 0; + } + if (peer->call > 0) { + ast_log(LOG_NOTICE, "Still have a call...\n"); + sip_destroy(peer->call); + } + p = peer->call = sip_alloc(NULL, NULL, 0, SIP_OPTIONS); + if (!peer->call) { + ast_log(LOG_WARNING, "Unable to allocate call for poking peer '%s'\n", peer->name); + return -1; + } + memcpy(&p->sa, &peer->addr, sizeof(p->sa)); + memcpy(&p->recv, &peer->addr, sizeof(p->sa)); + + /* Send options to peer's fullcontact */ + if (!ast_strlen_zero(peer->fullcontact)) { + ast_copy_string (p->fullcontact, peer->fullcontact, sizeof(p->fullcontact)); + } + + if (!ast_strlen_zero(peer->tohost)) + ast_copy_string(p->tohost, peer->tohost, sizeof(p->tohost)); + else + ast_inet_ntoa(p->tohost, sizeof(p->tohost), peer->addr.sin_addr); +#ifdef SIP_TCP_SUPPORT + p->sockfd = peer->tcpsockfd; + ast_copy_string(p->transport, peer->transport, sizeof(p->transport)); + if (option_verbose > 2) + ast_verbose("poking %s: transport %s, sockfd %d \n", peer->username, p->transport, p->sockfd); +#endif + + /* Recalculate our side, and recalculate Call ID */ + if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip)) + memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); + build_via(p, p->via, sizeof(p->via)); + build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); + + if (peer->pokeexpire > -1) + ast_sched_del(sched, peer->pokeexpire); + p->peerpoke = peer; + ast_set_flag(p, SIP_OUTGOING); +#ifdef VOCAL_DATA_HACK + ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username)); + transmit_invite(p, SIP_INVITE, 0, NULL, 1); +#else + transmit_invite(p, SIP_OPTIONS, 0, NULL, 1); +#endif + gettimeofday(&peer->ps, NULL); + peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2, sip_poke_noanswer, peer); + + return 0; +} + +/*--- sip_devicestate: Part of PBX channel interface ---*/ +static int sip_devicestate(void *data) +{ + char *ext, *host; + char tmp[256] = ""; + char *dest = data; + + struct hostent *hp; + struct ast_hostent ahp; + struct sip_peer *p; + int found = 0; + + int res = AST_DEVICE_INVALID; + + ast_copy_string(tmp, dest, sizeof(tmp)); + host = strchr(tmp, '@'); + if (host) { + *host = '\0'; + host++; + ext = tmp; + } else { + host = tmp; + ext = NULL; + } + + p = find_peer(host, NULL, 1); + if (p) { + found++; + res = AST_DEVICE_UNAVAILABLE; + if ((p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) && + (!p->maxms || ((p->lastms > -1) && (p->lastms <= p->maxms)))) { + /* peer found and valid */ + res = AST_DEVICE_UNKNOWN; + } + } + if (!p && !found) { + hp = ast_gethostbyname(host, &ahp); + if (hp) + res = AST_DEVICE_UNKNOWN; + } + + if (p) + ASTOBJ_UNREF(p,sip_destroy_peer); + return res; +} + +/*--- sip_request: PBX interface function -build SIP pvt structure ---*/ +/* SIP calls initiated by the PBX arrive here */ +static struct ast_channel *sip_request(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; + format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1); + if (!format) { + 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)); + return NULL; + } + p = sip_alloc(NULL, NULL, 0, SIP_INVITE); + if (!p) { + ast_log(LOG_WARNING, "Unable to build sip pvt data for '%s'\n", (char *)data); + return NULL; + } + + ast_copy_string(tmp, dest, sizeof(tmp)); + host = strchr(tmp, '@'); + if (host) { + *host = '\0'; + host++; + ext = tmp; + } else { + ext = strchr(tmp, '/'); + if (ext) { + *ext++ = '\0'; + host = tmp; + } + else { + host = tmp; + ext = NULL; + } + } + + /* Assign a default capability */ + p->capability = global_capability; + + if (create_addr(p, host)) { + *cause = AST_CAUSE_UNREGISTERED; + sip_destroy(p); + return NULL; + } + if (ast_strlen_zero(p->peername) && ext) + ast_copy_string(p->peername, ext, sizeof(p->peername)); + /* Recalculate our side, and recalculate Call ID */ + if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip)) + memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); + build_via(p, p->via, sizeof(p->via)); + build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); + + /* We have an extension to call, don't use the full contact here */ + /* This to enable dialling registred peers with extension dialling, + like SIP/peername/extension + SIP/peername will still use the full contact */ + if (ext) { + ast_copy_string(p->username, ext, sizeof(p->username)); + p->fullcontact[0] = 0; + } +#if 0 + printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "", host); +#endif + p->prefcodec = format; + 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; +} + +/*--- handle_common_options: Handle flag-type options common to users and peers ---*/ +static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v) +{ + int res = 0; + + if (!strcasecmp(v->name, "trustrpid")) { + ast_set_flag(mask, SIP_TRUSTRPID); + ast_set2_flag(flags, ast_true(v->value), SIP_TRUSTRPID); + res = 1; + } else if (!strcasecmp(v->name, "useclientcode")) { + ast_set_flag(mask, SIP_USECLIENTCODE); + ast_set2_flag(flags, ast_true(v->value), SIP_USECLIENTCODE); + res = 1; + } else if (!strcasecmp(v->name, "dtmfmode")) { + ast_set_flag(mask, SIP_DTMF); + ast_clear_flag(flags, SIP_DTMF); + if (!strcasecmp(v->value, "inband")) + ast_set_flag(flags, SIP_DTMF_INBAND); + else if (!strcasecmp(v->value, "rfc2833")) + ast_set_flag(flags, SIP_DTMF_RFC2833); + else if (!strcasecmp(v->value, "info")) + ast_set_flag(flags, SIP_DTMF_INFO); + else { + ast_log(LOG_WARNING, "Unknown dtmf mode '%s' on line %d, using rfc2833\n", v->value, v->lineno); + ast_set_flag(flags, SIP_DTMF_RFC2833); + } + } else if (!strcasecmp(v->name, "nat")) { + ast_set_flag(mask, SIP_NAT); + ast_clear_flag(flags, SIP_NAT); + if (!strcasecmp(v->value, "never")) + ast_set_flag(flags, SIP_NAT_NEVER); + else if (!strcasecmp(v->value, "route")) + ast_set_flag(flags, SIP_NAT_ROUTE); + else if (ast_true(v->value)) + ast_set_flag(flags, SIP_NAT_ALWAYS); + else + ast_set_flag(flags, SIP_NAT_RFC3581); + } else if (!strcasecmp(v->name, "canreinvite")) { + ast_set_flag(mask, SIP_REINVITE); + ast_clear_flag(flags, SIP_REINVITE); + if (!strcasecmp(v->value, "update")) + ast_set_flag(flags, SIP_REINVITE_UPDATE | SIP_CAN_REINVITE); + else + ast_set2_flag(flags, ast_true(v->value), SIP_CAN_REINVITE); + } else if (!strcasecmp(v->name, "insecure")) { + ast_set_flag(mask, SIP_INSECURE_PORT | SIP_INSECURE_INVITE); + ast_clear_flag(flags, SIP_INSECURE_PORT | SIP_INSECURE_INVITE); + if (!strcasecmp(v->value, "very")) + ast_set_flag(flags, SIP_INSECURE_PORT | SIP_INSECURE_INVITE); + else if (ast_true(v->value)) + ast_set_flag(flags, SIP_INSECURE_PORT); + else if (!ast_false(v->value)) { + char buf[64]; + char *word, *next; + + ast_copy_string(buf, v->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", v->value, v->lineno); + } + } + } else if (!strcasecmp(v->name, "progressinband")) { + ast_set_flag(mask, SIP_PROG_INBAND); + ast_clear_flag(flags, SIP_PROG_INBAND); + if (strcasecmp(v->value, "never")) + ast_set_flag(flags, SIP_PROG_INBAND_NO); + else if (ast_true(v->value)) + ast_set_flag(flags, SIP_PROG_INBAND_YES); + } else if (!strcasecmp(v->name, "allowguest")) { +#ifdef OSP_SUPPORT + if (!strcasecmp(v->value, "osp")) + global_allowguest = 2; + else +#endif + if (ast_true(v->value)) + global_allowguest = 1; + else + global_allowguest = 0; +#ifdef OSP_SUPPORT + } else if (!strcasecmp(v->name, "ospauth")) { + ast_set_flag(mask, SIP_OSPAUTH); + ast_clear_flag(flags, SIP_OSPAUTH); + if (!strcasecmp(v->value, "exclusive")) + ast_set_flag(flags, SIP_OSPAUTH_EXCLUSIVE); + else + ast_set2_flag(flags, ast_true(v->value), SIP_OSPAUTH_YES); +#endif + } else if (!strcasecmp(v->name, "promiscredir")) { + ast_set_flag(mask, SIP_PROMISCREDIR); + ast_set2_flag(flags, ast_true(v->value), SIP_PROMISCREDIR); + res = 1; + } + + return res; +} + +/*--- add_realm_authentication: 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 *auth; + struct sip_auth *b = NULL, *a = authlist; + + if (!configuration || ast_strlen_zero(configuration)) + return authlist; + + 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'; + realm++; + } + if (!username || ast_strlen_zero(username) || !realm || 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,"#"); + } + } + auth = malloc(sizeof(struct sip_auth)); + if (auth) { + memset(auth, 0, sizeof(struct sip_auth)); + 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)); + } else { + ast_log(LOG_ERROR, "Allocation of auth structure failed, Out of memory\n"); + return authlist; + } + + /* Add authentication to authl */ + if (!authlist) { /* No existing list */ + return auth; + } + while(a) { + b = a; + a = a->next; + } + b->next = auth; /* Add structure add end of list */ + + if (option_verbose > 2) + ast_verbose("Added authentication for realm %s\n", realm); + + return authlist; + +} + +/*--- clear_realm_authentication: 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; +} + +/*--- find_realm_authentication: Find authentication for a specific realm ---*/ +static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, char *realm) +{ + struct sip_auth *a = authlist; /* First entry in auth list */ + + while (a) { + if (!strcasecmp(a->realm, realm)){ + break; + } + a = a->next; + } + + return a; +} + +/*--- build_user: Initiate a SIP user structure from sip.conf ---*/ +static struct sip_user *build_user(const char *name, struct ast_variable *v, 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 = {(0)}; + struct ast_flags mask = {(0)}; + + + user = (struct sip_user *)malloc(sizeof(struct sip_user)); + if (!user) { + return NULL; + } + memset(user, 0, sizeof(struct sip_user)); + suserobjs++; + ASTOBJ_INIT(user); + ast_copy_string(user->name, name, sizeof(user->name)); + oldha = user->ha; + user->ha = NULL; + /* set the usage flag to a sane staring value*/ + user->inUse = 0; + user->outUse = 0; + ast_copy_flags(user, &global_flags, + SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_USECLIENTCODE | SIP_DTMF | SIP_NAT | + SIP_REINVITE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE | SIP_PROG_INBAND | SIP_OSPAUTH); + user->capability = global_capability; + user->prefs = prefs; + /* set default context */ + strcpy(user->context, default_context); + strcpy(user->language, default_language); + strcpy(user->musicclass, global_musicclass); + while(v) { + if (handle_common_options(&userflags, &mask, v)) { + v = v->next; + continue; + } + + if (!strcasecmp(v->name, "context")) { + ast_copy_string(user->context, v->value, sizeof(user->context)); + } else if (!strcasecmp(v->name, "setvar")) { + varname = ast_strdupa(v->value); + if (varname && (varval = strchr(varname,'='))) { + *varval = '\0'; + varval++; + 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, "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, "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, "musiconhold")) { + ast_copy_string(user->musicclass, v->value, sizeof(user->musicclass)); + } else if (!strcasecmp(v->name, "accountcode")) { + ast_copy_string(user->accountcode, v->value, sizeof(user->accountcode)); + } else if (!strcasecmp(v->name, "incominglimit")) { + user->incominglimit = atoi(v->value); + if (user->incominglimit < 0) + user->incominglimit = 0; + } else if (!strcasecmp(v->name, "outgoinglimit")) { + user->outgoinglimit = atoi(v->value); + if (user->outgoinglimit < 0) + user->outgoinglimit = 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, "callingpres")) { + user->callingpres = ast_parse_caller_presentation(v->value); + if (user->callingpres == -1) + user->callingpres = atoi(v->value); + } + /*else if (strcasecmp(v->name,"type")) + * ast_log(LOG_WARNING, "Ignoring %s\n", v->name); + */ + v = v->next; + } + ast_copy_flags(user, &userflags, mask.flags); + ast_free_ha(oldha); + return user; +} + +/*--- temp_peer: Create temporary peer (used in autocreatepeer mode) ---*/ +static struct sip_peer *temp_peer(const char *name) +{ + struct sip_peer *peer; + + peer = malloc(sizeof(*peer)); + if (!peer) + return NULL; + + memset(peer, 0, sizeof(*peer)); + apeerobjs++; + ASTOBJ_INIT(peer); + + peer->expire = -1; + peer->pokeexpire = -1; + ast_copy_string(peer->name, name, sizeof(peer->name)); + ast_copy_flags(peer, &global_flags, + SIP_PROMISCREDIR | SIP_USEREQPHONE | SIP_TRUSTRPID | SIP_USECLIENTCODE | + SIP_DTMF | SIP_NAT | SIP_REINVITE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE | + SIP_PROG_INBAND | SIP_OSPAUTH); + strcpy(peer->context, default_context); + strcpy(peer->language, default_language); + strcpy(peer->musicclass, global_musicclass); + peer->addr.sin_port = htons(DEFAULT_SIP_PORT); + peer->addr.sin_family = AF_INET; + peer->expiry = expiry; + peer->capability = global_capability; + peer->rtptimeout = global_rtptimeout; + peer->rtpholdtimeout = global_rtpholdtimeout; + peer->rtpkeepalive = global_rtpkeepalive; + ast_set_flag(peer, SIP_SELFDESTRUCT); + ast_set_flag(peer, SIP_DYNAMIC); + peer->prefs = prefs; + reg_source_db(peer); + + return peer; +} + +/*--- build_peer: Build peer from config file ---*/ +static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int realtime) +{ + struct sip_peer *peer = NULL; + struct ast_ha *oldha = NULL; + int maskfound=0; + int obproxyfound=0; + int found=0; + int format=0; /* Ama flags */ + time_t regseconds; + char *varname = NULL, *varval = NULL; + struct ast_variable *tmpvar = NULL; + struct ast_flags peerflags = {(0)}; + struct ast_flags mask = {(0)}; + + + if (!realtime) + /* Note we do NOT use find_peer here, to avoid realtime recursion */ + peer = ASTOBJ_CONTAINER_FIND_UNLINK(&peerl, name); + + if (peer) { + /* Already in the list, remove it and it will be added back (or FREE'd) */ + found++; + } else { + peer = malloc(sizeof(struct sip_peer)); + if (peer) { + memset(peer, 0, sizeof(struct sip_peer)); + if (realtime) + rpeerobjs++; + else + speerobjs++; + ASTOBJ_INIT(peer); + peer->expire = -1; + peer->pokeexpire = -1; + } else { + ast_log(LOG_WARNING, "Can't allocate SIP peer memory\n"); + } + } + /* Note that our peer HAS had its reference count incrased */ + if (!peer) + return NULL; + + peer->lastmsgssent = -1; + if (!found) { + if (name) + ast_copy_string(peer->name, name, sizeof(peer->name)); + peer->addr.sin_port = htons(DEFAULT_SIP_PORT); + peer->addr.sin_family = AF_INET; + peer->defaddr.sin_family = AF_INET; + peer->expiry = expiry; + } + /* If we have channel variables, remove them (reload) */ + if (peer->chanvars) { + ast_variables_destroy(peer->chanvars); + peer->chanvars = NULL; + } + strcpy(peer->context, default_context); + strcpy(peer->language, default_language); + strcpy(peer->musicclass, global_musicclass); + ast_copy_flags(peer, &global_flags, SIP_USEREQPHONE); + 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->rtpkeepalive = global_rtpkeepalive; + peer->maxms = default_qualify; + peer->prefs = prefs; + oldha = peer->ha; + peer->ha = NULL; + peer->addr.sin_family = AF_INET; +#ifdef SIP_TCP_SUPPORT + peer->tcpsockfd = -1; + ast_copy_string(peer->transport, "UDP", sizeof(peer->transport)); +#endif + ast_copy_flags(peer, &global_flags, + SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_USECLIENTCODE | + SIP_DTMF | SIP_REINVITE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE | + SIP_PROG_INBAND | SIP_OSPAUTH); + peer->capability = global_capability; + peer->rtptimeout = global_rtptimeout; + peer->rtpholdtimeout = global_rtpholdtimeout; + while(v) { + if (handle_common_options(&peerflags, &mask, v)) { + v = v->next; + continue; + } + + if (realtime && !strcasecmp(v->name, "regseconds")) { + if (sscanf(v->value, "%li", ®seconds) != 1) + regseconds = 0; + } 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 (!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, "context")) { + ast_copy_string(peer->context, v->value, sizeof(peer->context)); + } 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, 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 */ + ast_set_flag(peer, SIP_DYNAMIC); + if (!found) { + /* Initialize stuff iff we're not found, otherwise + we keep going with what we had */ + 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; + } + } + } + } else { + /* Non-dynamic. Make sure we become that way if we're not */ + if (peer->expire > -1) + ast_sched_del(sched, peer->expire); + peer->expire = -1; + ast_clear_flag(peer, SIP_DYNAMIC); + if (!obproxyfound || !strcasecmp(v->name, "outboundproxy")) { + if (ast_get_ip_or_srv(&peer->addr, v->value, "_sip._udp")) { + 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 (!maskfound) + inet_aton("255.255.255.255", &peer->mask); + } 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, "mask")) { + maskfound++; + inet_aton(v->value, &peer->mask); + } else if (!strcasecmp(v->name, "port")) { + if (!realtime && ast_test_flag(peer, SIP_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, "incominglimit")) { + peer->incominglimit = atoi(v->value); + if (peer->incominglimit < 0) + peer->incominglimit = 0; + } else if (!strcasecmp(v->name, "outgoinglimit")) { + peer->outgoinglimit = atoi(v->value); + if (peer->outgoinglimit < 0) + peer->outgoinglimit = 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, "musiconhold")) { + ast_copy_string(peer->musicclass, v->value, sizeof(peer->musicclass)); + } else if (!strcasecmp(v->name, "mailbox")) { + ast_copy_string(peer->mailbox, v->value, sizeof(peer->mailbox)); + } else if (!strcasecmp(v->name, "callgroup")) { + peer->callgroup = ast_get_group(v->value); + } 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, "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 (varname && (varval = strchr(varname,'='))) { + *varval = '\0'; + varval++; + 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_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; + } + } + /* else if (strcasecmp(v->name,"type")) + * ast_log(LOG_WARNING, "Ignoring %s\n", v->name); + */ + v=v->next; + } + if (realtime && !ast_test_flag((&global_flags_page2), SIP_PAGE2_RTIGNOREREGEXPIRE) && ast_test_flag(peer, SIP_DYNAMIC)) { + time_t nowtime; + + time(&nowtime); + if ((nowtime - regseconds) > 0) { + memset(&peer->addr, 0, sizeof(peer->addr)); + if (option_debug) + ast_log(LOG_DEBUG, "Bah, we're expired (%ld/%ld/%ld)!\n", nowtime - regseconds, regseconds, nowtime); + } + } + ast_copy_flags(peer, &peerflags, mask.flags); + if (!found && ast_test_flag(peer, SIP_DYNAMIC) && !ast_test_flag(peer, SIP_REALTIME)) + reg_source_db(peer); + ASTOBJ_UNMARK(peer); + ast_free_ha(oldha); + return peer; +} + +/*--- reload_config: Re-read SIP.conf config file ---*/ +/* 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(void) +{ + struct ast_config *cfg; + struct ast_variable *v; + struct sip_peer *peer; + struct sip_user *user; + struct ast_hostent ahp; + char *cat; + char *utype; + struct hostent *hp; + int format; + int oldport = ntohs(bindaddr.sin_port); +#ifdef SIP_TCP_SUPPORT + int oldtlsport = ntohs(tlsbindaddr.sin_port); +#endif + char iabuf[INET_ADDRSTRLEN]; + struct ast_flags dummy; + + 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; + } + + /* Reset IP addresses */ + memset(&bindaddr, 0, sizeof(bindaddr)); + memset(&localaddr, 0, sizeof(localaddr)); + memset(&externip, 0, sizeof(externip)); + memset(&prefs, 0 , sizeof(prefs)); +#ifdef SIP_TCP_SUPPORT + memset(&tlsbindaddr, 0, sizeof(tlsbindaddr)); +#endif + + /* Initialize some reasonable defaults at SIP reload */ + ast_copy_string(default_context, DEFAULT_CONTEXT, sizeof(default_context)); + default_language[0] = '\0'; + default_fromdomain[0] = '\0'; + default_qualify = 0; + externhost[0] = '\0'; + externexpire = 0; + externrefresh = 10; + sipdebug = 0; + ast_copy_string(default_useragent, DEFAULT_USERAGENT, sizeof(default_useragent)); + ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime)); + ast_copy_string(global_realm, DEFAULT_REALM, sizeof(global_realm)); + ast_copy_string(global_musicclass, "default", sizeof(global_musicclass)); + ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid)); + memset(&outboundproxyip, 0, sizeof(outboundproxyip)); + outboundproxyip.sin_port = htons(DEFAULT_SIP_PORT); + outboundproxyip.sin_family = AF_INET; /* Type of address: IPv4 */ + videosupport = 0; + compactheaders = 0; + relaxdtmf = 0; + callevents = 0; + ourport = DEFAULT_SIP_PORT; + global_rtptimeout = 0; + global_rtpholdtimeout = 0; + global_rtpkeepalive = 0; + pedanticsipchecking = 0; + global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT; + global_regattempts_max = DEFAULT_REGATTEMPTS_MAX; + ast_clear_flag(&global_flags, AST_FLAGS_ALL); + ast_set_flag(&global_flags, SIP_DTMF_RFC2833); + ast_set_flag(&global_flags, SIP_NAT_RFC3581); + ast_set_flag(&global_flags, SIP_CAN_REINVITE); + global_mwitime = DEFAULT_MWITIME; + srvlookup = 0; + autocreatepeer = 0; + regcontext[0] = '\0'; + tos = 0; + expiry = DEFAULT_EXPIRY; + global_allowguest = 1; + + /* Read the [general] config section of sip.conf (or from realtime config) */ + v = ast_variable_browse(cfg, "general"); + while(v) { + if (handle_common_options(&global_flags, &dummy, v)) { + v = v->next; + 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, "realm")) { + ast_copy_string(global_realm, v->value, sizeof(global_realm)); + } else if (!strcasecmp(v->name, "useragent")) { + ast_copy_string(default_useragent, v->value, sizeof(default_useragent)); + ast_log(LOG_DEBUG, "Setting User Agent Name to %s\n", + default_useragent); + } else if (!strcasecmp(v->name, "rtcachefriends")) { + ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_RTCACHEFRIENDS); + } else if (!strcasecmp(v->name, "rtnoupdate")) { + ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_RTNOUPDATE); + } else if (!strcasecmp(v->name, "rtignoreregexpire")) { + ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_RTIGNOREREGEXPIRE); + } 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_page2), i || ast_true(v->value), SIP_PAGE2_RTAUTOCLEAR); + } else if (!strcasecmp(v->name, "usereqphone")) { + ast_set2_flag((&global_flags), ast_true(v->value), SIP_USEREQPHONE); + } else if (!strcasecmp(v->name, "relaxdtmf")) { + 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, "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, "videosupport")) { + videosupport = ast_true(v->value); + } 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 (!strcasecmp(v->name, "musicclass") || !strcasecmp(v->name, "musiconhold")) { + ast_copy_string(global_musicclass, v->value, sizeof(global_musicclass)); + } 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(regcontext, v->value, sizeof(regcontext)); + /* Create context if it doesn't exist already */ + if (!ast_context_find(regcontext)) + ast_context_create(NULL, regcontext, channeltype); + } 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, "_sip._udp") < 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, "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")){ + sipdebug = 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)); + time(&externexpire); + } 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(&prefs, &global_capability, v->value, 1); + } else if (!strcasecmp(v->name, "disallow")) { + ast_parse_allow_disallow(&prefs, &global_capability, v->value, 0); + } else if (!strcasecmp(v->name, "register")) { + sip_register(v->value, v->lineno); + } else if (!strcasecmp(v->name, "recordhistory")) { + recordhistory = ast_true(v->value); + } else if (!strcasecmp(v->name, "tos")) { + if (sscanf(v->value, "%d", &format) == 1) + tos = format & 0xff; + else if (!strcasecmp(v->value, "lowdelay")) + tos = IPTOS_LOWDELAY; + else if (!strcasecmp(v->value, "throughput")) + tos = IPTOS_THROUGHPUT; + else if (!strcasecmp(v->value, "reliability")) + tos = IPTOS_RELIABILITY; + else if (!strcasecmp(v->value, "mincost")) + tos = IPTOS_MINCOST; + else if (!strcasecmp(v->value, "none")) + tos = 0; + else + ast_log(LOG_WARNING, "Invalid tos value at line %d, should be 'lowdelay', 'throughput', 'reliability', 'mincost', or 'none'\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")) { + callevents = ast_true(v->value); + } +#ifdef SIP_TCP_SUPPORT + else if (!strcasecmp(v->name, "tlsport")) { + if (sscanf(v->value, "%d", &ourport) == 1) { + tlsbindaddr.sin_port = htons(ourport); + } else { + ast_log(LOG_WARNING, "Invalid tlsport number '%s' at line %d of %s\n", v->value, v->lineno, config); + } + } else if (!strcasecmp(v->name, "trustcerts")) { + ast_copy_string(trustcerts_file, v->value, sizeof(trustcerts_file)); + } else if (!strcasecmp(v->name, "servercert")) { + ast_copy_string(servercert_file, v->value, sizeof(servercert_file)); + } else if (!strcasecmp(v->name, "serverkey")) { + ast_copy_string(serverkey_file, v->value, sizeof(serverkey_file)); + } else if (!strcasecmp(v->name, "serverkeypassword")) { + ast_copy_string(serverkey_password, v->value, sizeof(serverkey_password)); + } else if (!strcasecmp(v->name, "dh512param")) { + ast_copy_string(dh512param_file, v->value, sizeof(dh512param_file)); + } else if (!strcasecmp(v->name, "dh1024param")) { + ast_copy_string(dh1024param_file, v->value, sizeof(dh1024param_file)); + } +#endif + /* else if (strcasecmp(v->name,"type")) + * ast_log(LOG_WARNING, "Ignoring %s\n", v->name); + */ + v = v->next; + } + + /* Build list of authentication to various SIP realms, i.e. service providers */ + v = ast_variable_browse(cfg, "authentication"); + while(v) { + /* Format for authentication is auth = username:password@realm */ + if (!strcasecmp(v->name, "auth")) { + authl = add_realm_authentication(authl, v->value, v->lineno); + } + v = v->next; + } + + /* Load peers, users and friends */ + cat = ast_category_browse(cfg, NULL); + while(cat) { + if (strcasecmp(cat, "general") && strcasecmp(cat, "authentication")) { + utype = ast_variable_retrieve(cfg, cat, "type"); + if (utype) { + if (!strcasecmp(utype, "user") || !strcasecmp(utype, "friend")) { + user = build_user(cat, ast_variable_browse(cfg, cat), 0); + if (user) { + ASTOBJ_CONTAINER_LINK(&userl,user); + ASTOBJ_UNREF(user, sip_destroy_user); + } + } + if (!strcasecmp(utype, "peer") || !strcasecmp(utype, "friend")) { + peer = build_peer(cat, ast_variable_browse(cfg, cat), 0); + if (peer) { + ASTOBJ_CONTAINER_LINK(&peerl,peer); + ASTOBJ_UNREF(peer, sip_destroy_peer); + } + } else if (strcasecmp(utype, "user")) { + ast_log(LOG_WARNING, "Unknown type '%s' for '%s' in %s\n", utype, cat, "sip.conf"); + } + } else + ast_log(LOG_WARNING, "Section '%s' lacks type\n", cat); + } + cat = ast_category_browse(cfg, cat); + } + if (ast_find_ourip(&__ourip, bindaddr)) { + ast_log(LOG_WARNING, "Unable to get own IP address, SIP disabled\n"); + return 0; + } + if (!ntohs(bindaddr.sin_port)) + bindaddr.sin_port = ntohs(DEFAULT_SIP_PORT); + bindaddr.sin_family = AF_INET; + ast_mutex_lock(&netlock); + if ((sipsock > -1) && (ntohs(bindaddr.sin_port) != oldport)) { + 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)); + } 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); + + if (bind(sipsock, (struct sockaddr *)&bindaddr, sizeof(bindaddr)) < 0) { + ast_log(LOG_WARNING, "Failed to bind to %s:%d: %s\n", + ast_inet_ntoa(iabuf, sizeof(iabuf), 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(iabuf, sizeof(iabuf), bindaddr.sin_addr), ntohs(bindaddr.sin_port)); + ast_verbose(VERBOSE_PREFIX_2 "Using TOS bits %d\n", tos); + } + if (setsockopt(sipsock, IPPROTO_IP, IP_TOS, &tos, sizeof(tos))) + ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos); + } + } + } +#ifdef SIP_TCP_SUPPORT + if ((siptcpsock > -1) && (ntohs(bindaddr.sin_port) != oldport)) { + close(siptcpsock); + siptcpsock = -1; + } + /* Open a TCP listening socket */ + if (siptcpsock < 0) { + siptcpsock = socket(AF_INET, SOCK_STREAM, 0); + if (siptcpsock < 0) { + ast_log(LOG_WARNING, "Unable to create SIP TCP socket: %s\n", strerror(errno)); + } else { + const int reuseFlag = 1; + setsockopt(siptcpsock, SOL_SOCKET, SO_REUSEADDR, (const char *)&reuseFlag, sizeof reuseFlag); + + if (bind(siptcpsock, (struct sockaddr *)&bindaddr, sizeof(bindaddr)) < 0) { + ast_log(LOG_WARNING, "Failed to bind SIP TCP socket to %s:%d: %s\n", + ast_inet_ntoa(iabuf, sizeof(iabuf), bindaddr.sin_addr), ntohs(bindaddr.sin_port), strerror(errno)); + close(siptcpsock); + siptcpsock = -1; + } else { + if (setsockopt(siptcpsock, IPPROTO_IP, IP_TOS, &tos, sizeof(tos))) + ast_log(LOG_WARNING, "Unable to set TOS to %d in SIP TCP\n", tos); + if (listen(siptcpsock, 30) < 0) { + ast_log(LOG_WARNING, "Failed to listen on SIP TCP\n"); + } else { + if (option_verbose > 1) { + ast_verbose(VERBOSE_PREFIX_2 "SIP TCP Listening on %s:%d\n", + ast_inet_ntoa(iabuf, sizeof(iabuf), bindaddr.sin_addr), ntohs(bindaddr.sin_port)); + } + } + } + } + } + + /* Open a TLS listening socket */ + memcpy(&tlsbindaddr.sin_addr, &bindaddr.sin_addr, sizeof(tlsbindaddr.sin_addr)); + if (!ntohs(tlsbindaddr.sin_port)) + tlsbindaddr.sin_port = ntohs(DEFAULT_SIP_TLS_PORT); + tlsbindaddr.sin_family = AF_INET; + if ((siptlssock > -1) && (ntohs(tlsbindaddr.sin_port) != oldtlsport)) { + close(siptlssock); + siptlssock = -1; + } + if (siptlssock < 0) { + siptlssock = socket(AF_INET, SOCK_STREAM, 0); + if (siptlssock < 0) { + ast_log(LOG_WARNING, "Unable to create SIP TLS socket: %s\n", strerror(errno)); + } else { + const int reuseFlag = 1; + setsockopt(siptlssock, SOL_SOCKET, SO_REUSEADDR, (const char *)&reuseFlag, sizeof reuseFlag); + + if (bind(siptlssock, (struct sockaddr *)&tlsbindaddr, sizeof(tlsbindaddr)) < 0) { + ast_log(LOG_WARNING, "Failed to bind SIP TLS socket to %s:%d: %s\n", + ast_inet_ntoa(iabuf, sizeof(iabuf), tlsbindaddr.sin_addr), ntohs(tlsbindaddr.sin_port), strerror(errno)); + close(siptlssock); + siptlssock = -1; + } else { + if (setsockopt(siptlssock, IPPROTO_IP, IP_TOS, &tos, sizeof(tos))) + ast_log(LOG_WARNING, "Unable to set TOS to %d in SIP TLS\n", tos); + if (listen(siptlssock, 30) < 0) { + ast_log(LOG_WARNING, "Failed to listen on SIP TLS\n"); + } else { + if (option_verbose > 1) { + ast_verbose(VERBOSE_PREFIX_2 "SIP TLS Listening on %s:%d\n", + ast_inet_ntoa(iabuf, sizeof(iabuf), tlsbindaddr.sin_addr), ntohs(tlsbindaddr.sin_port)); + } + } + } + } + } +#endif + ast_mutex_unlock(&netlock); + + /* 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); + + return 0; +} + +/*--- sip_get_rtp_peer: Returns null if we can't reinvite (part of RTP interface) */ +static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan) +{ + struct sip_pvt *p; + struct ast_rtp *rtp = NULL; + p = chan->tech_pvt; + if (!p) + return NULL; + ast_mutex_lock(&p->lock); + if (p->rtp && ast_test_flag(p, SIP_CAN_REINVITE)) + rtp = p->rtp; + ast_mutex_unlock(&p->lock); + return rtp; +} + +/*--- sip_get_vrtp_peer: Returns null if we can't reinvite video (part of RTP interface) */ +static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan) +{ + struct sip_pvt *p; + struct ast_rtp *rtp = NULL; + p = chan->tech_pvt; + if (!p) + return NULL; + + ast_mutex_lock(&p->lock); + if (p->vrtp && ast_test_flag(p, SIP_CAN_REINVITE)) + rtp = p->vrtp; + ast_mutex_unlock(&p->lock); + return rtp; +} + +/*--- sip_set_rtp_peer: Set the data needed to RE-INVITE this call + so that the peers media go between them, outside of Asterisk. ---*/ +static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs) +{ + struct sip_pvt *p; + p = chan->tech_pvt; + if (!p) + return -1; + + ast_mutex_lock(&p->lock); + if (rtp) + ast_rtp_get_peer(rtp, &p->redirip); + else + memset(&p->redirip, 0, sizeof(p->redirip)); + if (vrtp) + ast_rtp_get_peer(vrtp, &p->vredirip); + else + memset(&p->vredirip, 0, sizeof(p->vredirip)); + p->redircodecs = codecs; + if (!ast_test_flag(p, SIP_GOTREFER)) { + if (!p->pendinginvite) + transmit_reinvite_with_sdp(p); + else if (!ast_test_flag(p, SIP_PENDINGBYE)) { + ast_log(LOG_DEBUG, "Deferring reinvite on '%s'\n", p->callid); + ast_set_flag(p, SIP_NEEDREINVITE); + } + } + /* Reset lastrtprx timer */ + time(&p->lastrtprx); + time(&p->lastrtptx); + 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-Accuntcode:\". Use this with care.\n" +"Adding the wrong headers may jeopardize the SIP dialog.\n" +"Always returns 0\n"; + +static char *app_sipgetheader = "SIPGetHeader"; +static char *synopsis_sipgetheader = "Get a SIP header from an incoming call"; + +static char *descrip_sipgetheader = "" +" SIPGetHeader(var=headername): \n" +"Sets a channel variable to the content of a SIP header\n" +"Skips to priority+101 if header does not exist\n" +"Otherwise returns 0\n"; + +/*--- sip_dtmfmode: change the DTMFmode for a 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_mutex_lock(&chan->lock); + if (chan->type != channeltype) { + ast_log(LOG_WARNING, "Call this application only on SIP incoming calls\n"); + ast_mutex_unlock(&chan->lock); + return 0; + } + p = chan->tech_pvt; + if (!p) { + ast_mutex_unlock(&chan->lock); + return 0; + } + ast_mutex_lock(&p->lock); + if (!strcasecmp(mode,"info")) { + ast_clear_flag(p, SIP_DTMF); + ast_set_flag(p, SIP_DTMF_INFO); + } else if (!strcasecmp(mode,"rfc2833")) { + ast_clear_flag(p, SIP_DTMF); + ast_set_flag(p, SIP_DTMF_RFC2833); + } else if (!strcasecmp(mode,"inband")) { + ast_clear_flag(p, SIP_DTMF); + ast_set_flag(p, SIP_DTMF_INBAND); + } else + ast_log(LOG_WARNING, "I don't know about this dtmf mode: %s\n",mode); + if (ast_test_flag(p, 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_mutex_unlock(&chan->lock); + return 0; +} + +/*--- sip_addheader: Add a SIP header ---*/ +static int sip_addheader(struct ast_channel *chan, void *data) +{ + int arglen; + int no = 0; + int ok = 0; + char *content = (char *) NULL; + char varbuf[128]; + + arglen = strlen(data); + if (!arglen) { + ast_log(LOG_WARNING, "This application requires the argument: Header\n"); + return 0; + } + ast_mutex_lock(&chan->lock); + if (chan->type != channeltype) { + ast_log(LOG_WARNING, "Call this application only on incoming SIP calls\n"); + ast_mutex_unlock(&chan->lock); + return 0; + } + + /* Check for headers */ + while (!ok && no <= 50) { + no++; + snprintf(varbuf, sizeof(varbuf), "_SIPADDHEADER%.2d", no); + content = pbx_builtin_getvar_helper(chan, varbuf); + + if (!content) + ok = 1; + } + if (ok) { + pbx_builtin_setvar_helper (chan, varbuf, data); + if (sipdebug) + ast_log(LOG_DEBUG,"SIP Header added \"%s\" as %s\n", (char *) data, varbuf); + } else { + ast_log(LOG_WARNING, "Too many SIP headers added, max 50\n"); + } + ast_mutex_unlock(&chan->lock); + return 0; +} + +/*--- sip_getheader: Get a SIP header (dialplan app) ---*/ +static int sip_getheader(struct ast_channel *chan, void *data) +{ + static int dep_warning = 0; + struct sip_pvt *p; + char *argv, *varname = NULL, *header = NULL, *content; + + if (!dep_warning) { + ast_log(LOG_WARNING, "SIPGetHeader is deprecated, use the SIP_HEADER function instead.\n"); + dep_warning = 1; + } + + argv = ast_strdupa(data); + if (!argv) { + ast_log(LOG_DEBUG, "Memory allocation failed\n"); + return 0; + } + + if (strchr (argv, '=') ) { /* Pick out argumenet */ + varname = strsep (&argv, "="); + header = strsep (&argv, "\0"); + } + + if (!varname || !header) { + ast_log(LOG_DEBUG, "SipGetHeader: Ignoring command, Syntax error in argument\n"); + return 0; + } + + ast_mutex_lock(&chan->lock); + if (chan->type != channeltype) { + ast_log(LOG_WARNING, "Call this application only on incoming SIP calls\n"); + ast_mutex_unlock(&chan->lock); + return 0; + } + + p = chan->tech_pvt; + content = get_header(&p->initreq, header); /* Get the header */ + if (!ast_strlen_zero(content)) { + pbx_builtin_setvar_helper(chan, varname, content); + } else { + ast_log(LOG_WARNING,"SIP Header %s not found for channel variable %s\n", header, varname); + ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101); + } + + ast_mutex_unlock(&chan->lock); + return 0; +} + +/*--- sip_sipredirect: Transfer call before connect with a 302 redirect ---*/ +/* Called by the transfer() dialplan application through the sip_transfer() */ +/* pbx interface function if the call is in ringing state */ +/* coded by Martin Pycko (m78pl@yahoo.com) */ +static int sip_sipredirect(struct sip_pvt *p, const char *dest) +{ + char *cdest; + char *extension, *host, *port; + char tmp[80]; + + cdest = ast_strdupa(dest); + if (!cdest) { + ast_log(LOG_ERROR, "Problem allocating the memory\n"); + return 0; + } + extension = strsep(&cdest, "@"); + host = strsep(&cdest, ":"); + port = strsep(&cdest, ":"); + if (!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 (!strlen(tmp)) { + ast_log(LOG_ERROR, "Cannot retrieve the 'To' header from the original SIP request!\n"); + return 0; + } + if ((localtmp = strstr(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 (!strlen(lhost)) { + ast_log(LOG_ERROR, "Can't find the host address\n"); + return 0; + } + host = ast_strdupa(lhost); + if (!host) { + ast_log(LOG_ERROR, "Problem allocating the memory\n"); + return 0; + } + if (!ast_strlen_zero(lport)) { + port = ast_strdupa(lport); + if (!port) { + ast_log(LOG_ERROR, "Problem allocating the memory\n"); + return 0; + } + } + } + } + + /* make sure the forwarding won't be forever */ + ast_copy_string(tmp, get_header(&p->initreq, "Max-Forwards"), sizeof(tmp)); + if (strlen(tmp) && atoi(tmp)) { + /* we found Max-Forwards in the original SIP request */ + p->maxforwards = atoi(tmp) - 1; + } else { + /* just send our 302 Moved Temporarily */ + p->maxforwards = DEFAULT_MAX_FORWARDS - 1; + } + if (p->maxforwards > -1) { + snprintf(p->our_contact, sizeof(p->our_contact), "Transfer ", extension, host, port ? ":" : "", port ? port : ""); + transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq, 1); + } else { + transmit_response(p, "483 Too Many Hops", &p->initreq); + } + /* this is all that we want to send to that SIP device */ + ast_set_flag(p, SIP_ALREADYGONE); + + /* hangup here */ + return -1; +} + +/*--- sip_get_codec: 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->peercapability; +} + +/*--- sip_rtp: Interface structure with callbacks used to connect to rtp module --*/ +static struct ast_rtp_protocol sip_rtp = { + type: channeltype, + 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, +}; + +/*--- sip_poke_all_peers: Send a poke to all known peers */ +static void sip_poke_all_peers(void) +{ + ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do { + ASTOBJ_WRLOCK(iterator); + sip_poke_peer(iterator); + ASTOBJ_UNLOCK(iterator); + } while (0) + ); +} + +/*--- sip_send_all_registers: Send all known registrations */ +static void sip_send_all_registers(void) +{ + int ms; + + ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do { + ASTOBJ_WRLOCK(iterator); + if (iterator->expire > -1) + ast_sched_del(sched, iterator->expire); + ms = (rand() >> 12) & 0x1fff; + iterator->expire = ast_sched_add(sched, ms, sip_reregister, iterator); + ASTOBJ_UNLOCK(iterator); + } while (0) + ); +} + +/*--- sip_do_reload: Reload module */ +static int sip_do_reload(void) +{ + clear_realm_authentication(authl); + authl = NULL; + + ASTOBJ_CONTAINER_DESTROYALL(&userl, sip_destroy_user); + ASTOBJ_CONTAINER_DESTROYALL(®l, sip_registry_destroy); + ASTOBJ_CONTAINER_MARKALL(&peerl); + reload_config(); + /* Prune peers who still are supposed to be deleted */ + ASTOBJ_CONTAINER_PRUNE_MARKED(&peerl, sip_destroy_peer); + + sip_poke_all_peers(); + sip_send_all_registers(); + + return 0; +} + +/*--- sip_reload: 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 = 1; + ast_mutex_unlock(&sip_reload_lock); + restart_monitor(); + + return 0; +} + +/*--- reload: Part of Asterisk module interface ---*/ +int reload(void) +{ + return sip_reload(0, 0, NULL); +} + +#ifdef SIP_TCP_SUPPORT + +DH *read_dhparams(char *dhfile) +{ + BIO *bio = NULL; + DH *dh = NULL; + + bio = BIO_new_file(dhfile, "r"); + if (!bio) + ast_log(LOG_ERROR, "Error opening file %s", dhfile); + dh = PEM_read_bio_DHparams(bio, NULL, NULL, NULL); + if (!dh) + ast_log(LOG_ERROR, "Error reading DH parameters from %s", dhfile); + BIO_free(bio); + return dh; +} + +DH *tmp_dh_callback(SSL *ssl, int is_export, int keylength) +{ + DH *ret; + + switch (keylength) + { + case 512: + ret = read_dhparams(dh512param_file); + break; + case 1024: + default: + ret = read_dhparams(dh1024param_file); + break; + } + return ret; +} + +int password_callback(char *buf, int num, int rwflag, void *userdata) +{ + if (num < (strlen(serverkey_password)+1)) { + ast_log(LOG_ERROR, "password buf len %d is too small\n", num); + return 0; + } + strcpy(buf, serverkey_password); + return (strlen(serverkey_password)); +} + +int verify_callback(int ok, X509_STORE_CTX *store) +{ + char data[256]; + + if (!ok) + { + X509 *cert = X509_STORE_CTX_get_current_cert(store); + int depth = X509_STORE_CTX_get_error_depth(store); + int err = X509_STORE_CTX_get_error(store); + + ast_log(LOG_ERROR, "-Error with certificate at depth: %i\n", depth); + X509_NAME_oneline(X509_get_issuer_name(cert), data, 256); + ast_log(LOG_ERROR, " issuer = %s\n", data); + X509_NAME_oneline(X509_get_subject_name(cert), data, 256); + ast_log(LOG_ERROR, " subject = %s\n", data); + ast_log(LOG_ERROR, " err %i:%s\n", err, X509_verify_cert_error_string(err)); + } + + return ok; +} + +/* initailize OpenSSL library */ +SSL_CTX *init_OpenSSL(void) +{ + SSL_CTX *ctx = NULL; + + if (!SSL_library_init()) { + ast_log(LOG_ERROR, "SSL_library_init failed\n"); + return NULL; + } + SSL_load_error_strings(); + RAND_load_file(DEFAULT_ENTROPY, 1024); + + ctx = SSL_CTX_new(TLSv1_method()); + + if (SSL_CTX_load_verify_locations(ctx, trustcerts_file, NULL) != 1) + ast_log(LOG_ERROR, "Error loading a trust certs\n"); + + SSL_CTX_set_default_passwd_cb(ctx, password_callback); + + if (SSL_CTX_set_default_verify_paths(ctx) != 1) + ast_log(LOG_ERROR, "Error to set_default_verify_path\n"); + if (SSL_CTX_use_certificate_chain_file(ctx, servercert_file) != 1) + ast_log(LOG_ERROR, "Error loading certificate from file\n"); + if (SSL_CTX_use_PrivateKey_file(ctx, serverkey_file, SSL_FILETYPE_PEM) != 1) + ast_log(LOG_ERROR, "Error loading private key from file\n"); + +/* SSL_CTX_set_verify(ctx, SSL_VERIFY_PEER, verify_callback); */ +/* SSL_CTX_set_verify_depth(ctx, 4); */ + SSL_CTX_set_options(ctx, SSL_OP_ALL | SSL_OP_SINGLE_DH_USE); + SSL_CTX_set_tmp_dh_callback(ctx, tmp_dh_callback); + if (SSL_CTX_set_cipher_list(ctx, CIPHER_LIST) != 1) + ast_log(LOG_ERROR, "Error setting cipher list (no valid ciphers)\n"); + SSL_CTX_set_mode(ctx, SSL_MODE_AUTO_RETRY); + return ctx; +} + +#endif + +// static struct ast_cli_entry cli_sip_reload = +static struct ast_cli_entry my_clis[] = { + { { "sip", "notify", NULL }, sip_notify, "Send a notify packet to a SIP peer", notify_usage, complete_sipnotify }, + { { "sip", "show", "objects", NULL }, sip_show_objects, "Show all SIP object allocations", show_objects_usage }, + { { "sip", "show", "users", NULL }, sip_show_users, "Show defined SIP users", show_users_usage }, + { { "sip", "show", "user", NULL }, sip_show_user, "Show details on specific SIP user", show_user_usage, complete_sip_show_user }, + { { "sip", "show", "subscriptions", NULL }, sip_show_subscriptions, "Show active SIP subscriptions", show_subscriptions_usage}, + { { "sip", "show", "channels", NULL }, sip_show_channels, "Show active SIP channels", show_channels_usage}, + { { "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", "debug", NULL }, sip_do_debug, "Enable SIP debugging", debug_usage }, + { { "sip", "debug", "ip", NULL }, sip_do_debug, "Enable SIP debugging on IP", debug_usage }, + { { "sip", "debug", "peer", NULL }, sip_do_debug, "Enable SIP debugging on Peername", debug_usage, complete_sip_debug_peer }, + { { "sip", "show", "peer", NULL }, sip_show_peer, "Show details on specific SIP peer", show_peer_usage, complete_sip_show_peer }, + { { "sip", "show", "peers", NULL }, sip_show_peers, "Show defined SIP peers", show_peers_usage }, + { { "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", "show", "inuse", NULL }, sip_show_inuse, "List all inuse/limits", show_inuse_usage }, + { { "sip", "show", "registry", NULL }, sip_show_registry, "Show SIP registration status", show_reg_usage }, + { { "sip", "history", NULL }, sip_do_history, "Enable SIP history", history_usage }, + { { "sip", "no", "history", NULL }, sip_no_history, "Disable SIP history", no_history_usage }, + { { "sip", "no", "debug", NULL }, sip_no_debug, "Disable SIP debugging", no_debug_usage }, + { { "sip", "reload", NULL }, sip_reload, "Reload SIP configuration", sip_reload_usage }, +}; + +/*--- load_module: PBX load module - initialization ---*/ +int load_module() +{ + ASTOBJ_CONTAINER_INIT(&userl); + ASTOBJ_CONTAINER_INIT(&peerl); + ASTOBJ_CONTAINER_INIT(®l); + sched = sched_context_create(); + if (!sched) { + ast_log(LOG_WARNING, "Unable to create schedule context\n"); + } + io = io_context_create(); + if (!io) { + ast_log(LOG_WARNING, "Unable to create I/O context\n"); + } + /* Make sure we can register our sip channel type */ + if (ast_channel_register(&sip_tech)) { + ast_log(LOG_ERROR, "Unable to register channel type %s\n", channeltype); + return -1; + } + + if (reload_config()) + return -1; + +#ifdef SIP_TCP_SUPPORT + tlsctx = init_OpenSSL(); +#endif + + ast_cli_register_multiple(my_clis, sizeof(my_clis)/ sizeof(my_clis[0])); + + ast_rtp_proto_register(&sip_rtp); + + ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode); + ast_register_application(app_sipaddheader, sip_addheader, synopsis_sipaddheader, descrip_sipaddheader); + ast_register_application(app_sipgetheader, sip_getheader, synopsis_sipgetheader, descrip_sipgetheader); + + 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); + + ast_custom_function_register(&sip_header_function); + ast_custom_function_register(&sippeer_function); + + sip_poke_all_peers(); + sip_send_all_registers(); + + /* And start the monitor for the first time */ + restart_monitor(); + + return 0; +} + +int unload_module() +{ + struct sip_pvt *p, *pl; + + /* First, take us out of the channel type list */ + ast_channel_unregister(&sip_tech); + + ast_custom_function_unregister(&sippeer_function); + ast_custom_function_unregister(&sip_header_function); + + ast_unregister_application(app_dtmfmode); + ast_unregister_application(app_sipaddheader); + ast_unregister_application(app_sipgetheader); + + ast_cli_unregister_multiple(my_clis, sizeof(my_clis)/ sizeof(my_clis[0])); + + ast_rtp_proto_unregister(&sip_rtp); + + ast_manager_unregister("SIPpeers"); + ast_manager_unregister("SIPshowpeer"); + + if (!ast_mutex_lock(&iflock)) { + /* Hangup all interfaces if they have an owner */ + p = iflist; + while (p) { + if (p->owner) + ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD); + p = p->next; + } + iflist = NULL; + ast_mutex_unlock(&iflock); + } else { + ast_log(LOG_WARNING, "Unable to lock the interface list\n"); + return -1; + } + + if (!ast_mutex_lock(&monlock)) { + if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP)) { + pthread_cancel(monitor_thread); + pthread_kill(monitor_thread, SIGURG); + pthread_join(monitor_thread, NULL); + } + monitor_thread = AST_PTHREADT_STOP; + ast_mutex_unlock(&monlock); + } else { + ast_log(LOG_WARNING, "Unable to lock the monitor\n"); + return -1; + } + + if (!ast_mutex_lock(&iflock)) { + /* Destroy all the interfaces and free their memory */ + p = iflist; + while (p) { + pl = p; + p = p->next; + /* Free associated memory */ + ast_mutex_destroy(&pl->lock); + if (pl->chanvars) { + ast_variables_destroy(pl->chanvars); + pl->chanvars = NULL; + } + free(pl); + } + iflist = NULL; + ast_mutex_unlock(&iflock); + } else { + ast_log(LOG_WARNING, "Unable to lock the interface list\n"); + return -1; + } + + /* 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); + close(sipsock); + +#ifdef SIP_TCP_SUPPORT + if (siptcpsock > -1) + close(siptcpsock); + if (siptlssock > -1) + close(siptlssock); + if (tlsctx) + SSL_CTX_free(tlsctx); /* destroy TLS CTX */ +#endif + + return 0; +} + +int usecount() +{ + return usecnt; +} + +char *key() +{ + return ASTERISK_GPL_KEY; +} + +char *description() +{ + return (char *) desc; +} + + Index: configs/sip.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v retrieving revision 1.64 diff -u -r1.64 sip.conf.sample --- configs/sip.conf.sample 9 Jun 2005 21:11:30 -0000 1.64 +++ configs/sip.conf.sample 27 Aug 2005 06:29:21 -0000 @@ -86,6 +86,17 @@ ; the moment the channel loads this configuration ; +; The configuration for TLS support +;tlsport=5061 ; TLS port (SIP standard TLS port is 5061) +;serverkeypassword=asterisk ; password for encrypted private key +;trustcerts=/var/lib/asterisk/keys/trustcerts.pem ; Trusted root CA or certificates files +;servercert=/var/lib/asterisk/keys/servercert.pem ; Asterisk server certificate file +;serverkey=/var/lib/asterisk/keys/serverkey.pem ; Asterisk server private key file +;dh512param=/var/lib/asterisk/keys/dh512.pem ; ephemeral Diffe-Hellman parameter +;dh1024param=/var/lib/asterisk/keys/dh1024.pem ; ephemeral Diffe-Hellman parameter +; + +; ; If regcontext is specified, Asterisk will dynamically ; create and destroy a NoOp priority 1 extension for a given ; peer who registers or unregisters with us. The actual extension