diff options
Diffstat (limited to '1.2-netsec/channels/chan_sip.c')
| -rw-r--r-- | 1.2-netsec/channels/chan_sip.c | 13481 |
1 files changed, 0 insertions, 13481 deletions
diff --git a/1.2-netsec/channels/chan_sip.c b/1.2-netsec/channels/chan_sip.c deleted file mode 100644 index 9960b7174..000000000 --- a/1.2-netsec/channels/chan_sip.c +++ /dev/null @@ -1,13481 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (C) 1999 - 2006, Digium, Inc. - * - * Mark Spencer <markster@digium.com> - * - * See http://www.asterisk.org for more information about - * the Asterisk project. Please do not directly contact - * any of the maintainers of this project for assistance; - * the project provides a web site, mailing lists and IRC - * channels for your use. - * - * This program is free software, distributed under the terms of - * the GNU General Public License Version 2. See the LICENSE file - * at the top of the source tree. - */ - -/*! - * \file - * \brief Implementation of Session Initiation Protocol - * - * Implementation of RFC 3261 - without S/MIME, TCP and TLS support - * Configuration file \link Config_sip sip.conf \endlink - * - * \todo SIP over TCP - * \todo SIP over TLS - * \todo Better support of forking - */ - - -#include <stdio.h> -#include <ctype.h> -#include <string.h> -#include <unistd.h> -#include <sys/socket.h> -#include <sys/ioctl.h> -#include <net/if.h> -#include <errno.h> -#include <stdlib.h> -#include <fcntl.h> -#include <netdb.h> -#include <signal.h> -#include <sys/signal.h> -#include <netinet/in.h> -#include <netinet/in_systm.h> -#include <arpa/inet.h> -#include <netinet/ip.h> -#include <regex.h> - -#include "asterisk.h" - -ASTERISK_FILE_VERSION(__FILE__, "$Revision$") - -#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" -#include "asterisk/linkedlists.h" - -#ifdef OSP_SUPPORT -#include "asterisk/astosp.h" -#endif - -#ifdef SIP_MIDCOM -#include "asterisk/res_netsec.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_MAX_FORWARDS "70" - -/* guard limit must be larger than guard secs */ -/* guard min must be < 1000, and should be >= 250 */ -#define EXPIRY_GUARD_SECS 15 /* How long before expiry do we reregister */ -#define EXPIRY_GUARD_LIMIT 30 /* Below here, we use EXPIRY_GUARD_PCT instead of - EXPIRY_GUARD_SECS */ -#define EXPIRY_GUARD_MIN 500 /* This is the minimum guard time applied. If - GUARD_PCT turns out to be lower than this, it - will use this time instead. - This is in milliseconds. */ -#define EXPIRY_GUARD_PCT 0.20 /* Percentage of expires timeout to use when - below EXPIRY_GUARD_LIMIT */ - -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 1000 /* How frequently to retransmit */ - /* 2 * 500 ms in RFC 3261 */ -#define MAX_RETRANS 6 /* Try only 6 times for retransmissions, a total of 7 transmissions */ -#define MAX_AUTHTRIES 3 /* Try authentication three times, then fail */ - - -#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 RTP 1 -#define NO_RTP 0 - -/* Do _NOT_ make any changes to this enum, or the array following it; - if you think you are doing the right thing, you are probably - not doing the right thing. If you think there are changes - needed, get someone else to review them first _before_ - submitting a patch. If these two lists do not match properly - bad things will happen. -*/ - -enum subscriptiontype { - NONE = 0, - TIMEOUT, - XPIDF_XML, - DIALOG_INFO_XML, - CPIM_PIDF_XML, - PIDF_XML -}; - -static const struct cfsubscription_types { - enum subscriptiontype type; - const char * const event; - const char * const mediatype; - const char * const text; -} subscription_types[] = { - { NONE, "-", "unknown", "unknown" }, - /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */ - { DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" }, - { CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */ - { PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */ - { XPIDF_XML, "presence", "application/xpidf+xml", "xpidf+xml" } /* Pre-RFC 3863 with MS additions */ -}; - -enum sipmethod { - SIP_UNKNOWN, - SIP_RESPONSE, - SIP_REGISTER, - SIP_OPTIONS, - SIP_NOTIFY, - SIP_INVITE, - SIP_ACK, - SIP_PRACK, - SIP_BYE, - SIP_REFER, - SIP_SUBSCRIBE, - SIP_MESSAGE, - SIP_UPDATE, - SIP_INFO, - SIP_CANCEL, - SIP_PUBLISH, -} sip_method_list; - -enum sip_auth_type { - PROXY_AUTH, - WWW_AUTH, -}; - -/*! XXX Note that sip_methods[i].id == i must hold or the code breaks */ -static const struct cfsip_methods { - enum sipmethod id; - int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */ - char * const text; -} sip_methods[] = { - { SIP_UNKNOWN, RTP, "-UNKNOWN-" }, - { SIP_RESPONSE, NO_RTP, "SIP/2.0" }, - { 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" } -}; - -/*! \brief Structure for conversion between compressed SIP and "normal" SIP */ -static const struct cfalias { - char * const fullname; - char * const shortname; -} aliases[] = { - { "Content-Type", "c" }, - { "Content-Encoding", "e" }, - { "From", "f" }, - { "Call-ID", "i" }, - { "Contact", "m" }, - { "Content-Length", "l" }, - { "Subject", "s" }, - { "To", "t" }, - { "Supported", "k" }, - { "Refer-To", "r" }, - { "Referred-By", "b" }, - { "Allow-Events", "u" }, - { "Event", "o" }, - { "Via", "v" }, - { "Accept-Contact", "a" }, - { "Reject-Contact", "j" }, - { "Request-Disposition", "d" }, - { "Session-Expires", "x" }, -}; - -/*! 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) - -/*! \brief List of well-known SIP options. If we get this in a require, - we should check the list and answer accordingly. */ -static const struct cfsip_options { - int id; /*!< Bitmap ID */ - int supported; /*!< Supported by Asterisk ? */ - char * const text; /*!< Text id, as in standard */ -} sip_options[] = { - /* 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" }, - /* RFC4092: Usage of the SDP ANAT Semantics in the SIP */ - { SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp-anat" }, - /* RFC3329: Security agreement mechanism */ - { SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" }, - /* SIMPLE events: 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" }, -}; - - -/*! \brief SIP Methods we support */ -#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY" - -/*! \brief SIP Extensions we support */ -#define SUPPORTED_EXTENSIONS "replaces" - -#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_subscribecontext[AST_MAX_CONTEXT]; - -#define DEFAULT_VMEXTEN "asterisk" -static char global_vmexten[AST_MAX_EXTENSION] = DEFAULT_VMEXTEN; - -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 global_notifyringing = 1; /*!< Send notifications on ringing */ - -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 = 0; - -/* 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); - -AST_MUTEX_DEFINE_STATIC(rand_lock); - -/*! \brief Protect the interface list (of sip_pvt's) */ -AST_MUTEX_DEFINE_STATIC(iflock); - -/*! \brief Protect the monitoring thread, so only one process can kill or start it, and not - when it's doing something critical. */ -AST_MUTEX_DEFINE_STATIC(netlock); - -AST_MUTEX_DEFINE_STATIC(monlock); - -/*! \brief This is the thread for the monitor which checks for input on the channels - which are not currently in use. */ -static pthread_t monitor_thread = AST_PTHREADT_NULL; - -static int restart_monitor(void); - -/*! \brief 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; - -#define SIP_DEBUG_CONFIG 1 << 0 -#define SIP_DEBUG_CONSOLE 1 << 1 -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 int dumphistory = 0; /*!< Dump history to verbose before destroying SIP dialog */ - -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 */ - -#define DEFAULT_EXPIRY 900 /*!< Expire slowly */ -static int expiry = DEFAULT_EXPIRY; - -static struct sched_context *sched; -static struct io_context *io; - -#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */ -#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */ - -#define DEC_CALL_LIMIT 0 -#define INC_CALL_LIMIT 1 - -static struct ast_codec_pref prefs; - - -/*! \brief sip_request: The data grabbed from the UDP socket */ -struct sip_request { - char *rlPart1; /*!< SIP Method Name or "SIP/2.0" protocol version */ - char *rlPart2; /*!< The Request URI or Response Status */ - int len; /*!< Length */ - int headers; /*!< # of SIP Headers */ - int method; /*!< Method of this request */ - char *header[SIP_MAX_HEADERS]; - int lines; /*!< SDP Content */ - char *line[SIP_MAX_LINES]; - char data[SIP_MAX_PACKET]; - int debug; /*!< Debug flag for this packet */ - unsigned int flags; /*!< SIP_PKT Flags for this packet */ -}; - -struct sip_pkt; - -/*! \brief Parameters to the transmit_invite function */ -struct sip_invite_param { - char *distinctive_ring; /*!< Distinctive ring header */ - char *osptoken; /*!< OSP token for this call */ - int addsipheaders; /*!< Add extra SIP headers */ - char *uri_options; /*!< URI options to add to the URI */ - char *vxml_url; /*!< VXML url for Cisco phones */ - char *auth; /*!< Authentication */ - char *authheader; /*!< Auth header */ - enum sip_auth_type auth_type; /*!< Authentication type */ -}; - -struct sip_route { - struct sip_route *next; - char hop[0]; -}; - -enum domain_mode { - SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */ - SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */ -}; - -struct domain { - char domain[MAXHOSTNAMELEN]; /*!< SIP domain we are responsible for */ - char context[AST_MAX_EXTENSION]; /*!< Incoming context for this domain */ - enum domain_mode mode; /*!< How did we find this domain? */ - AST_LIST_ENTRY(domain) list; /*!< List mechanics */ -}; - -static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */ - -int allow_external_domains; /*!< Accept calls to external SIP domains? */ - -/*! \brief sip_history: Structure for saving transactions within a SIP dialog */ -struct sip_history { - char event[80]; - struct sip_history *next; -}; - -/*! \brief 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 */ -#define SIP_DTMF_AUTO (3 << 16) /*!< AUTO switch between rfc2833 and in-band DTMF */ -/* NAT settings */ -#define SIP_NAT (3 << 18) /*!< four settings, uses two bits */ -#define SIP_NAT_NEVER (0 << 18) /*!< No nat support */ -#define SIP_NAT_RFC3581 (1 << 18) -#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) /*!< four settings, uses two bits */ -#define SIP_OSPAUTH_NO (0 << 26) -#define SIP_OSPAUTH_GATEWAY (1 << 26) -#define SIP_OSPAUTH_PROXY (2 << 26) -#define SIP_OSPAUTH_EXCLUSIVE (3 << 26) -/* Call states */ -#define SIP_CALL_ONHOLD (1 << 28) -#define SIP_CALL_LIMIT (1 << 29) -/* Remote Party-ID Support */ -#define SIP_SENDRPID (1 << 30) -/* Did this connection increment the counter of in-use calls? */ -#define SIP_INC_COUNT (1 << 31) - -#define SIP_FLAGS_TO_COPY \ - (SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \ - SIP_PROG_INBAND | SIP_OSPAUTH | SIP_USECLIENTCODE | SIP_NAT | \ - SIP_INSECURE_PORT | SIP_INSECURE_INVITE) - -/* a new page of flags for peer */ -#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0) -#define SIP_PAGE2_RTUPDATE (1 << 1) -#define SIP_PAGE2_RTAUTOCLEAR (1 << 2) -#define SIP_PAGE2_IGNOREREGEXPIRE (1 << 3) -#define SIP_PAGE2_RT_FROMCONTACT (1 << 4) - -/* SIP packet flags */ -#define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */ -#define SIP_PKT_WITH_TOTAG (1 << 1) /*!< This packet has a to-tag */ - -static int global_rtautoclear = 120; - -/*! \brief 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 */ - int timer_t1; /*!< SIP timer T1, ms rtt */ - 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 */ - char tag[11]; /*!< 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 subscribecontext[AST_MAX_CONTEXT]; /*!< Subscribecontext */ - 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 *rpid; /*!< Our RPID header */ - char *rpid_from; /*!< Our RPID From header */ - char realm[MAXHOSTNAMELEN]; /*!< Authorization realm */ - char nonce[256]; /*!< Authorization nonce */ - int noncecount; /*!< Nonce-count */ - 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 */ - unsigned int osptimelimit; /*!< OSP call duration limit */ -#endif - struct sip_request initreq; /*!< Initial request */ - - int maxtime; /*!< Max time for first response */ - 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 */ - enum subscriptiontype subscribed; /*!< Is this call a subscription? */ - int stateid; - int laststate; /*!< Last known extension state */ - int dialogver; - - struct ast_dsp *vad; /*!< Voice Activation Detection dsp */ - -#ifdef SIP_MIDCOM - void *r; -#endif - - 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 */ - struct sip_invite_param *options; /*!< Options for INVITE */ -} *iflist = NULL; - -#define FLAG_RESPONSE (1 << 0) -#define FLAG_FATAL (1 << 1) - -/*! \brief sip packet - read in sipsock_read, transmitted in send_request */ -struct sip_pkt { - struct sip_pkt *next; /*!< Next packet */ - int retrans; /*!< Retransmission number */ - int method; /*!< SIP method for this packet */ - int seqno; /*!< Sequence number */ - unsigned int flags; /*!< non-zero if this is a response packet (e.g. 200 OK) */ - struct sip_pvt *owner; /*!< Owner call */ - int retransid; /*!< Retransmission ID */ - int timer_a; /*!< SIP timer A, retransmission timer */ - int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */ - int packetlen; /*!< Length of packet */ - char data[0]; -}; - -/*! \brief Structure for SIP user data. User's place calls to us */ -struct sip_user { - /* Users who can access various contexts */ - ASTOBJ_COMPONENTS(struct sip_user); - char secret[80]; /*!< Password */ - char md5secret[80]; /*!< Password in md5 */ - char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */ - char subscribecontext[AST_MAX_CONTEXT]; /* Default context for subscriptions */ - char cid_num[80]; /*!< Caller ID num */ - char cid_name[80]; /*!< Caller ID name */ - char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */ - char language[MAX_LANGUAGE]; /*!< Default language for this user */ - char 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 call_limit; /*!< Limit of concurrent calls */ - 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 registered 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 subscribecontext[AST_MAX_CONTEXT]; /*!< Default context for subscriptions */ - char username[80]; /*!< Temporary username until registration */ - char accountcode[AST_MAX_ACCOUNT_CODE]; /*!< Account code */ - int amaflags; /*!< AMA Flags (for billing) */ - char tohost[MAXHOSTNAMELEN]; /*!< If not dynamic, IP address */ - char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */ - char fromuser[80]; /*!< From: user when calling this peer */ - char fromdomain[MAXHOSTNAMELEN]; /*!< From: domain when calling this peer */ - char fullcontact[256]; /*!< Contact registered with us (not in sip.conf) */ - char cid_num[80]; /*!< Caller ID num */ - char cid_name[80]; /*!< Caller ID name */ - int callingpres; /*!< Calling id presentation */ - int inUse; /*!< Number of calls in use */ - int call_limit; /*!< Limit of concurrent calls */ - char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/ - char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */ - char language[MAX_LANGUAGE]; /*!< Default language for prompts */ - char 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 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 */ - - /* 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 - - -/*! \brief 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 in clear text */ - char md5secret[80]; /*!< Password in md5 */ - 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. */ - int noncecount; /*!< Nonce-count */ - - char lastmsg[256]; /*!< Last Message sent/received */ -}; - -/*! \brief The user list: Users and friends ---*/ -static struct ast_user_list { - ASTOBJ_CONTAINER_COMPONENTS(struct sip_user); -} userl; - -/*! \brief The peer list: Peers and Friends ---*/ -static struct ast_peer_list { - ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer); -} peerl; - -/*! \brief The register list: Other SIP proxys we register with and 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 = { 0, }; -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 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, 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_info_with_vidupdate(struct sip_pvt *p); -static int transmit_message_with_text(struct sip_pvt *p, const char *text); -static int transmit_refer(struct sip_pvt *p, const char *dest); -static int 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_call_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_call(const char *type, int format, void *data, int *cause); -static int sip_devicestate(void *data); -static int sip_sendtext(struct ast_channel *ast, const char *text); -static int sip_call(struct ast_channel *ast, char *dest, int timeout); -static int sip_hangup(struct ast_channel *ast); -static int sip_answer(struct ast_channel *ast); -static struct ast_frame *sip_read(struct ast_channel *ast); -static int sip_write(struct ast_channel *ast, struct ast_frame *frame); -static int sip_indicate(struct ast_channel *ast, int condition); -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 int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */ -static void append_date(struct sip_request *req); /* Append date to SIP packet */ -static int determine_firstline_parts(struct sip_request *req); -static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to LOG_DEBUG at end of dialog, before destroying data */ -static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype); -static int transmit_state_notify(struct sip_pvt *p, int state, int full, int substate); -static char *gettag(struct sip_request *req, char *header, char *tagbuf, int tagbufsize); - -#ifdef SIP_MIDCOM -static void sip_rtp_get_peer_audio_helper(void *p, struct sockaddr_in *them); -static void sip_rtp_get_peer_video_helper(void *p, struct sockaddr_in *them); -static void sip_rtp_get_us_audio_helper(void *p, struct sockaddr_in *sin); -static void sip_rtp_get_us_video_helper(void *p, struct sockaddr_in *vsin); -static void sip_map_hook_struct(void *p, void *r); -static void *sip_get_hook_struct(void *p); -static int sip_get_flag_novideo(void *p); -static int sip_cmp_sa_addr(void *p, struct sockaddr_in *addr); -static void sip_get_recv_addr(void *p, struct in_addr *addr); -static char *sip_get_username(void *p); -static struct ast_channel *sip_channel_helper(void *p); -static struct ast_channel *sip_bridged_channel_helper(void *p); -static int sip_get_capability_helper(void *p); -static void sip_softhangup_helper(void *p); - -extern struct ast_sip_hook_cb *m_cb; -#endif - -/*! \brief Definition of this channel for PBX 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_call, - .devicestate = sip_devicestate, - .call = sip_call, - .hangup = sip_hangup, - .answer = sip_answer, - .read = sip_read, - .write = sip_write, - .write_video = sip_write, - .indicate = sip_indicate, - .transfer = sip_transfer, - .fixup = sip_fixup, - .send_digit = sip_senddigit, - .bridge = ast_rtp_bridge, - .send_text = sip_sendtext, -}; - -/*! - \brief Thread-safe random number generator - \return a random number - - This function uses a mutex lock to guarantee that no - two threads will receive the same random number. - */ -static force_inline int thread_safe_rand(void) -{ - int val; - - ast_mutex_lock(&rand_lock); - val = rand(); - ast_mutex_unlock(&rand_lock); - - return val; -} - -/*! \brief find_sip_method: Find SIP method from header - * Strictly speaking, SIP methods are case SENSITIVE, but we don't check - * following Jon Postel's rule: Be gentle in what you accept, strict with what you send */ -int find_sip_method(char *msg) -{ - int i, res = 0; - - if (ast_strlen_zero(msg)) - return 0; - - for (i = 1; (i < (sizeof(sip_methods) / sizeof(sip_methods[0]))) && !res; i++) { - if (!strcasecmp(sip_methods[i].text, msg)) - res = sip_methods[i].id; - } - return res; -} - -/*! \brief 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 (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; - if (option_debug) - ast_log(LOG_DEBUG, "* SIP extension value: %d for call %s\n", profile, pvt->callid); - } - return profile; -} - -/*! \brief 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; -} - -/*! \brief 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)); -} - - -/*! \brief __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:%d returned %d: %s\n", data, len, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port), res, strerror(errno)); - } - return res; -} - -static void sip_destroy(struct sip_pvt *p); - -/*! \brief 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); -} - -/*! \brief 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; -} - -/*! \brief append_history: Append to SIP dialog history */ -/* Always returns 0 */ -static int append_history(struct sip_pvt *p, const char *event, const char *data) -{ - struct sip_history *hist, *prev; - char *c; - - if (!recordhistory || !p) - 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; -} - -/*! \brief retrans_pkt: Retransmit SIP message if no answer ---*/ -static int retrans_pkt(void *data) -{ - struct sip_pkt *pkt=data, *prev, *cur = NULL; - char iabuf[INET_ADDRSTRLEN]; - int reschedule = DEFAULT_RETRANS; - - /* Lock channel */ - ast_mutex_lock(&pkt->owner->lock); - - if (pkt->retrans < MAX_RETRANS) { - char buf[80]; - - pkt->retrans++; - if (!pkt->timer_t1) { /* Re-schedule using timer_a and timer_t1 */ - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_methods[pkt->method].text, pkt->method); - } else { - int siptimer_a; - - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method); - if (!pkt->timer_a) - pkt->timer_a = 2 ; - else - pkt->timer_a = 2 * pkt->timer_a; - - /* For non-invites, a maximum of 4 secs */ - siptimer_a = pkt->timer_t1 * pkt->timer_a; /* Double each time */ - if (pkt->method != SIP_INVITE && siptimer_a > 4000) - siptimer_a = 4000; - - /* Reschedule re-transmit */ - reschedule = siptimer_a; - if (option_debug > 3) - ast_log(LOG_DEBUG, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid); - } - - if (pkt->owner && 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); - } - snprintf(buf, sizeof(buf), "ReTx %d", reschedule); - - append_history(pkt->owner, buf, pkt->data); - __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); - ast_mutex_unlock(&pkt->owner->lock); - return reschedule; - } - /* Too many retries */ - if (pkt->owner && pkt->method != SIP_OPTIONS) { - if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */ - ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request"); - } else { - if (pkt->method == SIP_OPTIONS && sipdebug) - ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid); - } - 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_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid); - ast_queue_hangup(pkt->owner->owner); - ast_mutex_unlock(&pkt->owner->owner->lock); - } else { - /* If no channel 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 0; -} - -/*! \brief __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, int sipmethod) -{ - struct sip_pkt *pkt; - int siptimer_a = DEFAULT_RETRANS; - - 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->method = sipmethod; - pkt->packetlen = len; - pkt->next = p->packets; - pkt->owner = p; - pkt->seqno = seqno; - pkt->flags = resp; - pkt->data[len] = '\0'; - pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */ - if (fatal) - ast_set_flag(pkt, FLAG_FATAL); - if (pkt->timer_t1) - siptimer_a = pkt->timer_t1 * 2; - - /* Schedule retransmission */ - pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1); - if (option_debug > 3 && sipdebug) - ast_log(LOG_DEBUG, "*** SIP TIMER: Initalizing retransmit timer on packet: Id #%d\n", pkt->retransid); - pkt->next = p->packets; - p->packets = pkt; - - __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */ - if (sipmethod == SIP_INVITE) { - /* Note this is a pending invite */ - p->pendinginvite = seqno; - } - return 0; -} - -/*! \brief __sip_autodestruct: Kill a call (called by scheduler) ---*/ -static int __sip_autodestruct(void *data) -{ - struct sip_pvt *p = data; - - - /* If this is a subscription, tell the phone that we got a timeout */ - if (p->subscribed) { - p->subscribed = TIMEOUT; - transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, 1); /* Send first notification */ - p->subscribed = NONE; - append_history(p, "Subscribestatus", "timeout"); - return 10000; /* Reschedule this destruction so that we know that it's gone */ - } - - /* This scheduled event is now considered done. */ - 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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief __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)))) { - ast_mutex_lock(&p->lock); - 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) { - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid); - ast_sched_del(sched, cur->retransid); - } - free(cur); - ast_mutex_unlock(&p->lock); - res = 0; - break; - } - prev = cur; - cur = cur->next; - } - ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %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) -{ - struct sip_pkt *cur=NULL; - - while(p->packets) { - if (cur == p->packets) { - ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text); - return -1; - } - cur = p->packets; - if (cur->method) - __sip_ack(p, p->packets->seqno, (ast_test_flag(p->packets, FLAG_RESPONSE)), cur->method); - else { /* Unknown packet type */ - char *c; - char method[128]; - 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; -} - -/*! \brief __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) { - if (option_debug > 3 && sipdebug) - ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, msg); - 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_request(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); - -/*! \brief 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_request(dst); -} - -/*! \brief 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), req->method); - } 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; -} - -/*! \brief 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), req->method); - } 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; -} - -/*! \brief get_in_brackets: Pick out text in brackets from character string ---*/ -/* returns pointer to terminated stripped string. modifies input string. */ -static char *get_in_brackets(char *tmp) -{ - char *parse; - char *first_quote; - char *first_bracket; - char *second_bracket; - char last_char; - - parse = tmp; - while (1) { - first_quote = strchr(parse, '"'); - first_bracket = strchr(parse, '<'); - if (first_quote && first_bracket && (first_quote < first_bracket)) { - last_char = '\0'; - for (parse = first_quote + 1; *parse; parse++) { - if ((*parse == '"') && (last_char != '\\')) - break; - last_char = *parse; - } - if (!*parse) { - ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp); - return tmp; - } - parse++; - continue; - } - if (first_bracket) { - second_bracket = strchr(first_bracket + 1, '>'); - if (second_bracket) { - *second_bracket = '\0'; - return first_bracket + 1; - } else { - ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp); - return tmp; - } - } - return tmp; - } -} - -/*! \brief 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 (ast_strlen_zero(text)) - return 0; - if (debug) - ast_verbose("Really sending text %s on %s\n", text, ast->name); - transmit_message_with_text(p, text); - return 0; -} - -/*! \brief realtime_update_peer: Update peer object in realtime storage ---*/ -static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey) -{ - char port[10]; - char ipaddr[20]; - char regseconds[20]; - time_t nowtime; - - time(&nowtime); - nowtime += expirey; - snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime); /* Expiration time */ - ast_inet_ntoa(ipaddr, sizeof(ipaddr), sin->sin_addr); - snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port)); - - if (fullcontact) - ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr, "port", port, "regseconds", regseconds, "username", username, "fullcontact", fullcontact, NULL); - else - ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr, "port", port, "regseconds", regseconds, "username", username, NULL); -} - -/*! \brief 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); - } - } -} - -/*! \brief 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); -} - -/*! \brief update_peer: Update peer data in database (if used) ---*/ -static void update_peer(struct sip_peer *p, int expiry) -{ - int rtcachefriends = ast_test_flag(&(p->flags_page2), SIP_PAGE2_RTCACHEFRIENDS); - if (ast_test_flag((&global_flags_page2), SIP_PAGE2_RTUPDATE) && - (ast_test_flag(p, SIP_REALTIME) || rtcachefriends)) { - realtime_update_peer(p->name, &p->addr, p->username, rtcachefriends ? p->fullcontact : NULL, expiry); - } -} - - -/*! \brief realtime_peer: Get peer from realtime storage - * Checks the "sippeers" realtime family from extconfig.conf */ -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", "host", iabuf, NULL); /* First check for fixed IP hosts */ - if (!var) - var = ast_load_realtime("sippeers", "ipaddr", iabuf, NULL); /* Then check for registred hosts */ - - } 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; -} - -/*! \brief 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))); -} - -/*! \brief 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; -} - -/*! \brief 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); -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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_family = peer->addr.sin_family; - r->sa.sin_addr = peer->addr.sin_addr; - r->sa.sin_port = peer->addr.sin_port; - } else { - r->sa.sin_family = peer->defaddr.sin_family; - 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_FLAGS_TO_COPY); - r->capability = peer->capability; - r->prefs = peer->prefs; - 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; - /* Set timer T1 to RTT for this peer (if known by qualify=) */ - if (peer->maxms && peer->lastms) - r->timer_t1 = peer->lastms; - if ((ast_test_flag(r, SIP_DTMF) == SIP_DTMF_RFC2833) || (ast_test_flag(r, SIP_DTMF) == SIP_DTMF_AUTO)) - 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; - if (peer->call_limit) - ast_set_flag(r, SIP_CALL_LIMIT); - - return 0; -} - -/*! \brief 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 *dialog, 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++; - } - dialog->sa.sin_family = AF_INET; - dialog->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */ - p = find_peer(peer, NULL, 1); - - if (p) { - found++; - if (create_addr_from_peer(dialog, 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(dialog->tohost, peer, sizeof(dialog->tohost)); - memcpy(&dialog->sa.sin_addr, hp->h_addr, sizeof(dialog->sa.sin_addr)); - dialog->sa.sin_port = htons(portno); - memcpy(&dialog->recv, &dialog->sa, sizeof(dialog->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; - } -} - -/*! \brief 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; -} - - - - -/*! \brief 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; - - - - p = ast->tech_pvt; - if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) { - ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name); - return -1; - } - - - /* Check whether there is vxml_url, distinctive ring variables */ - - headp=&ast->varshead; - AST_LIST_TRAVERSE(headp,current,entries) { - /* Check whether there is a VXML_URL variable */ - if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) { - p->options->vxml_url = ast_var_value(current); - } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) { - p->options->uri_options = ast_var_value(current); - } else if (!p->options->distinctive_ring && !strcasecmp(ast_var_name(current), "ALERT_INFO")) { - /* Check whether there is a ALERT_INFO variable */ - p->options->distinctive_ring = ast_var_value(current); - } else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) { - /* Check whether there is a variable with a name starting with SIPADDHEADER */ - p->options->addsipheaders = 1; - } - - -#ifdef OSP_SUPPORT - else if (!p->options->osptoken && !strcasecmp(ast_var_name(current), "OSPTOKEN")) { - p->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 (!p->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", p->options->osptoken, osphandle); - p->options->osptoken = NULL; - osphandle = NULL; - p->osphandle = -1; - } -#endif - ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username); - res = update_call_counter(p, INC_CALL_LIMIT); - if ( res != -1 ) { - p->callingpres = ast->cid.cid_pres; - p->jointcapability = p->capability; - transmit_invite(p, SIP_INVITE, 1, 2); - if (p->maxtime) { - /* Initialize auto-congest time */ - p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p); - } - } - return res; -} - -/*! \brief 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); - -} - -/*! \brief __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); - -#ifdef SIP_MIDCOM - if (m_cb) - m_cb->__sip_destroy_hook(p); -#endif - - if (dumphistory) - sip_dump_history(p); - - if (p->options) - free(p->options); - - 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); - } - - if (p->rpid) - free(p->rpid); - - if (p->rpid_from) - free(p->rpid_from); - - /* 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); - } - if (p->chanvars) { - ast_variables_destroy(p->chanvars); - p->chanvars = NULL; - } - ast_mutex_destroy(&p->lock); - free(p); -} - -/*! \brief update_call_counter: Handle call_limit 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_call_counter(struct sip_pvt *fup, int event) -{ - char name[256]; - int *inuse, *call_limit; - int outgoing = ast_test_flag(fup, SIP_OUTGOING); - struct sip_user *u = NULL; - struct sip_peer *p = NULL; - - if (option_debug > 2) - ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming"); - /* Test if we need to check call limits, in order to avoid - realtime lookups if we do not need it */ - if (!ast_test_flag(fup, 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; - call_limit = &u->call_limit; - p = NULL; - } else { - /* Try to find peer */ - if (!p) - p = find_peer(fup->peername, NULL, 1); - if (p) { - inuse = &p->inUse; - call_limit = &p->call_limit; - 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_CALL_LIMIT: - if ( *inuse > 0 ) { - if (ast_test_flag(fup,SIP_INC_COUNT)) - (*inuse)--; - } else { - *inuse = 0; - } - if (option_debug > 1 || sipdebug) { - ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit); - } - break; - case INC_CALL_LIMIT: - if (*call_limit > 0 ) { - if (*inuse >= *call_limit) { - ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit); - if (u) - ASTOBJ_UNREF(u,sip_destroy_user); - else - ASTOBJ_UNREF(p,sip_destroy_peer); - return -1; - } - } - (*inuse)++; - ast_set_flag(fup,SIP_INC_COUNT); - if (option_debug > 1 || sipdebug) { - ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit); - } - break; - default: - ast_log(LOG_ERROR, "update_call_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; -} - -/*! \brief 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); - -/*! \brief 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 603: /* Declined */ - 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; -} - - -/*! \brief hangup_cause2sip: Convert Asterisk hangup causes to SIP codes -\verbatim - Possible values from causes.h - AST_CAUSE_NOTDEFINED AST_CAUSE_NORMAL AST_CAUSE_BUSY - AST_CAUSE_FAILURE AST_CAUSE_CONGESTION AST_CAUSE_UNALLOCATED - - In addition to these, a lot of PRI codes is defined in causes.h - ...should we take care of them too ? - - Quote RFC 3398 - - ISUP Cause value SIP response - ---------------- ------------ - 1 unallocated number 404 Not Found - 2 no route to network 404 Not found - 3 no route to destination 404 Not found - 16 normal call clearing --- (*) - 17 user busy 486 Busy here - 18 no user responding 408 Request Timeout - 19 no answer from the user 480 Temporarily unavailable - 20 subscriber absent 480 Temporarily unavailable - 21 call rejected 403 Forbidden (+) - 22 number changed (w/o diagnostic) 410 Gone - 22 number changed (w/ diagnostic) 301 Moved Permanently - 23 redirection to new destination 410 Gone - 26 non-selected user clearing 404 Not Found (=) - 27 destination out of order 502 Bad Gateway - 28 address incomplete 484 Address incomplete - 29 facility rejected 501 Not implemented - 31 normal unspecified 480 Temporarily unavailable -\endverbatim -*/ -static 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; -} - - -/*! \brief sip_hangup: Hangup SIP call - * Part of PBX interface, called from ast_hangup */ -static int sip_hangup(struct ast_channel *ast) -{ - struct sip_pvt *p = ast->tech_pvt; - int needcancel = 0; - struct ast_flags locflags = {0}; - - if (!p) { - ast_log(LOG_DEBUG, "Asked to hangup channel not connected\n"); - return 0; - } - if (option_debug) - ast_log(LOG_DEBUG, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid); - - 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 - ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter\n", p->username); - update_call_counter(p, DEC_CALL_LIMIT); - /* Determine how to disconnect */ - if (p->owner != ast) { - ast_log(LOG_WARNING, "Huh? We aren't the owner? Can't hangup call.\n"); - ast_mutex_unlock(&p->lock); - return 0; - } - /* If the call is not UP, we need to send CANCEL instead of BYE */ - if (ast->_state != AST_STATE_UP) - needcancel = 1; - -#ifdef SIP_MIDCOM - /* For callee to shutdown, send "BYE" instead of "CANCEL" - -- this needs to be verified */ - if (m_cb && ast_test_flag(p, SIP_OUTGOING)) needcancel = 0; -#endif - - /* 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) { /* Outgoing call, not up */ - 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); - /* stop retransmitting an INVITE that has not received a response */ - __sip_pretend_ack(p); - if ( p->initid != -1 ) { - /* channel still up - reverse dec of inUse counter - only if the channel is not auto-congested */ - update_call_counter(p, INC_CALL_LIMIT); - } - } else { /* Incoming call, not up */ - char *res; - if (ast->hangupcause && ((res = hangup_cause2sip(ast->hangupcause)))) { - transmit_response_reliable(p, res, &p->initreq, 1); - } else - transmit_response_reliable(p, "603 Declined", &p->initreq, 1); - } - } else { /* Call is in UP state, send BYE */ - 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; -} - -/*! \brief sip_answer: Answer SIP call , send 200 OK on Invite - * Part of PBX interface */ -static int sip_answer(struct ast_channel *ast) -{ - int res = 0,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; -} - -/*! \brief 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; -} - -/*! \brief sip_fixup: Fix up a channel: If a channel is consumed, this is called. - Basically update any ->owner links ----*/ -static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan) -{ - 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; -} - -/*! \brief 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; -} - - - -/*! \brief 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; -} - -/*! \brief 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_PROCEEDING: - if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) { - transmit_response(p, "100 Trying", &p->initreq); - break; - } - res = -1; - break; - case AST_CONTROL_PROGRESS: - 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: /* The other part of the bridge are put on hold */ - if (sipdebug) - ast_log(LOG_DEBUG, "Bridged channel now on hold%s\n", p->callid); - res = -1; - break; - case AST_CONTROL_UNHOLD: /* The other part of the bridge are back from hold */ - if (sipdebug) - ast_log(LOG_DEBUG, "Bridged channel is back from hold, let's talk! : %s\n", p->callid); - res = -1; - break; - case AST_CONTROL_VIDUPDATE: /* Request a video frame update */ - if (p->vrtp && !ast_test_flag(p, SIP_NOVIDEO)) { - transmit_info_with_vidupdate(p); - res = 0; - } else - res = -1; - break; - case -1: - res = -1; - break; - default: - ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition); - res = -1; - break; - } - ast_mutex_unlock(&p->lock); - return res; -} - - - -/*! \brief sip_new: Initiate a call in the SIP channel */ -/* called from sip_request_call (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; -#ifdef OSP_SUPPORT - char iabuf[INET_ADDRSTRLEN]; - char peer[MAXHOSTNAMELEN]; -#endif - - 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. */ - 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); - fmt = ast_best_codec(tmp->nativeformats); - - if (title) - snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%04x", title, thread_safe_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); - } - if (i->rtp) { - 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); - } -#ifdef OSP_SUPPORT - snprintf(peer, sizeof(peer), "[%s]:%d", ast_inet_ntoa(iabuf, sizeof(iabuf), i->sa.sin_addr), ntohs(i->sa.sin_port)); - pbx_builtin_setvar_helper(tmp, "OSPPEER", peer); -#endif - 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; -} - -/*! \brief 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 ""; -} - -/*! \brief 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; x<req->lines; 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;x<sizeof(aliases) / sizeof(aliases[0]); x++) - if (!strcasecmp(aliases[x].fullname, name)) - return aliases[x].shortname; - return _default; -} - -static char *__get_header(struct sip_request *req, char *name, int *start) -{ - int pass; - - /* - * Technically you can place arbitrary whitespace both before and after the ':' in - * a header, although RFC3261 clearly says you shouldn't before, and place just - * one afterwards. If you shouldn't do it, what absolute idiot decided it was - * a good idea to say you can do it, and if you can do it, why in the hell would. - * you say you shouldn't. - * Anyways, pedanticsipchecking controls whether we allow spaces before ':', - * and we always allow spaces after that for compatibility. - */ - for (pass = 0; name && pass < 2;pass++) { - int x, len = strlen(name); - for (x=*start; x<req->headers; x++) { - if (!strncasecmp(req->header[x], name, len)) { - char *r = req->header[x] + len; /* skip name */ - if (pedanticsipchecking) - r = ast_skip_blanks(r); - - if (*r == ':') { - *start = x+1; - return ast_skip_blanks(r+1); - } - } - } - if (pass == 0) /* Try aliases */ - name = find_alias(name, NULL); - } - - /* Don't return NULL, so get_header is always a valid pointer */ - return ""; -} - -/*! \brief get_header: Get header from SIP request ---*/ -static char *get_header(struct sip_request *req, char *name) -{ - int start = 0; - return __get_header(req, name, &start); -} - -/*! \brief 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, }; - - if (!p->rtp) { - /* We have no RTP allocated for this channel */ - return &null_frame; - } - - switch(ast->fdno) { - case 0: - f = ast_rtp_read(p->rtp); /* RTP Audio */ - break; - case 1: - f = ast_rtcp_read(p->rtp); /* RTCP Control Channel */ - break; - case 2: - f = ast_rtp_read(p->vrtp); /* RTP Video */ - break; - case 3: - f = ast_rtcp_read(p->vrtp); /* RTCP Control Channel for video */ - break; - 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; -} - -/*! \brief 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; -} - -/*! \brief 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 = thread_safe_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)); -} - -static void make_our_tag(char *tagbuf, size_t len) -{ - snprintf(tagbuf, len, "as%08x", thread_safe_rand()); -} - -/*! \brief 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; - - if (!(p = calloc(1, sizeof(*p)))) - return NULL; - - ast_mutex_init(&p->lock); - - p->method = intended_method; - p->initid = -1; - p->autokillid = -1; - p->subscribed = NONE; - p->stateid = -1; - p->prefs = prefs; - if (intended_method != SIP_OPTIONS) /* Peerpoke has it's own system */ - p->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */ -#ifdef OSP_SUPPORT - p->osphandle = -1; - p->osptimelimit = 0; -#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 = thread_safe_rand(); - make_our_tag(p->tag, sizeof(p->tag)); - /* 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 || (videosupport && !p->vrtp)) { - ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n", videosupport ? "and video" : "", strerror(errno)); - ast_mutex_destroy(&p->lock); - if (p->chanvars) { - ast_variables_destroy(p->chanvars); - p->chanvars = NULL; - } - free(p); - return NULL; - } - ast_rtp_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_FLAGS_TO_COPY); - /* 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) || (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_AUTO)) - 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; -} - -/*! \brief 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 *tag = ""; - char totag[128]; - char fromtag[128]; - - callid = get_header(req, "Call-ID"); - - if (pedanticsipchecking) { - /* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy - we need more to identify a branch - so we have to check branch, from - and to tags to identify a call leg. - For Asterisk to behave correctly, you need to turn on pedanticsipchecking - in sip.conf - */ - if (gettag(req, "To", totag, sizeof(totag))) - ast_set_flag(req, SIP_PKT_WITH_TOTAG); /* Used in handle_request/response */ - gettag(req, "From", fromtag, sizeof(fromtag)); - - if (req->method == SIP_RESPONSE) - tag = totag; - else - tag = fromtag; - - - if (option_debug > 4 ) - ast_log(LOG_DEBUG, "= Looking for Call ID: %s (Checking %s) --From tag %s --To-tag %s \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag); - } - - ast_mutex_lock(&iflock); - p = iflist; - while(p) { /* In pedantic, we do not want packets with bad syntax to be connected to a PVT */ - int found = 0; - if (req->method == SIP_REGISTER) - found = (!strcmp(p->callid, callid)); - else - found = (!strcmp(p->callid, callid) && - (!pedanticsipchecking || !tag || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) ; - - if (option_debug > 4) - ast_log(LOG_DEBUG, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag); - - /* If we get a new request within an existing to-tag - check the to tag as well */ - if (pedanticsipchecking && found && req->method != SIP_RESPONSE) { /* SIP Request */ - if (p->tag[0] == '\0' && totag[0]) { - /* We have no to tag, but they have. Wrong dialog */ - found = 0; - } else if (totag[0]) { /* Both have tags, compare them */ - if (strcmp(totag, p->tag)) { - found = 0; /* This is not our packet */ - } - } - if (!found && option_debug > 4) - ast_log(LOG_DEBUG, "= Being pedantic: This is not our match on request: Call ID: %s Ourtag <null> Totag %s Method %s\n", p->callid, totag, sip_methods[req->method].text); - } - - - if (found) { - /* Found the call */ - ast_mutex_lock(&p->lock); - ast_mutex_unlock(&iflock); - return p; - } - p = p->next; - } - ast_mutex_unlock(&iflock); - p = sip_alloc(callid, sin, 1, intended_method); - if (p) - ast_mutex_lock(&p->lock); - return p; -} - -/*! \brief 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 (ast_strlen_zero(username) || ast_strlen_zero(hostname)) { - ast_log(LOG_WARNING, "Format for registration is user[:secret[:authuser]]@host[:port][/contact] at line %d\n", lineno); - return -1; - } - 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 (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; -} - -/*! \brief 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; -} - -/*! \brief parse_request: Parse a SIP message ----*/ -static void parse_request(struct sip_request *req) -{ - /* Divide fields by NULL's */ - char *c; - int f = 0; - - c = req->data; - - /* First header starts immediately */ - req->header[f] = c; - while(*c) { - if (*c == '\n') { - /* We've got a new header */ - *c = 0; - - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "Header %d: %s (%d)\n", f, req->header[f], (int) strlen(req->header[f])); - if (ast_strlen_zero(req->header[f])) { - /* Line by itself means we're now in content */ - c++; - break; - } - if (f >= SIP_MAX_HEADERS - 1) { - ast_log(LOG_WARNING, "Too many SIP headers. Ignoring.\n"); - } else - f++; - req->header[f] = c + 1; - } else if (*c == '\r') { - /* Ignore but eliminate \r's */ - *c = 0; - } - c++; - } - /* Check for last header */ - if (!ast_strlen_zero(req->header[f])) { - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "Header %d: %s (%d)\n", f, req->header[f], (int) strlen(req->header[f])); - 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 (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "Line: %s (%d)\n", req->line[f], (int) strlen(req->line[f])); - if (f >= SIP_MAX_LINES - 1) { - ast_log(LOG_WARNING, "Too many SDP lines. Ignoring.\n"); - } 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); - /* Split up the first line parts */ - determine_firstline_parts(req); -} - -/*! \brief 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; - - if (!p->rtp) { - ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n"); - return -1; - } - - /* 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/%d RTP/AVP %n", &x, &y, &len) == 2) || - (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) { - 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 RTP video format %d\n", 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 (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_AUTO) { - ast_clear_flag(p, SIP_DTMF); - if (p->noncodeccapability & AST_RTP_DTMF) { - /* XXX Would it be reasonable to drop the DSP at this point? XXX */ - ast_set_flag(p, SIP_DTMF_RFC2833); - } else { - ast_set_flag(p, SIP_DTMF_INBAND); - } - } - - 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 */ - struct ast_frame af = { AST_FRAME_NULL, }; - if (sin.sin_addr.s_addr && !sendonly) { - ast_moh_stop(bridgepeer); - - /* Activate a re-invite */ - ast_queue_frame(p->owner, &af); - } else { - /* No address for RTP, we're on hold */ - - ast_moh_start(bridgepeer, NULL); - if (sendonly) - ast_rtp_stop(p->rtp); - /* Activate a re-invite */ - ast_queue_frame(p->owner, &af); - } - } - - /* Manager Hold and Unhold events must be generated, if necessary */ - if (sin.sin_addr.s_addr && !sendonly) { - 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); - } 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); - } - - return 0; -} - -/*! \brief add_header: Add header to SIP message */ -static int add_header(struct sip_request *req, const char *var, const char *value) -{ - int x = 0; - - 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; x < (sizeof(aliases) / sizeof(aliases[0])); x++) - if (!strcasecmp(aliases[x].fullname, var)) - var = aliases[x].shortname; - } - - 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; -} - -/*! \brief add_header_contentLen: Add 'Content-Length' header to SIP message */ -static int add_header_contentLength(struct sip_request *req, int len) -{ - char clen[10]; - - snprintf(clen, sizeof(clen), "%d", len); - return add_header(req, "Content-Length", clen); -} - -/*! \brief add_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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief copy_via_headers: Copy SIP VIA Headers from the request to the response ---*/ -/* If the client indicates that it wishes to know the port we received from, - it adds ;rport without an argument to the topmost via header. We need to - add the port number (from our point of view) to that parameter. - We always add ;received=<ip address> to the topmost via header. - Received: RFC 3261, rport RFC 3581 */ -static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, struct sip_request *orig, char *field) -{ - char tmp[256], *oh, *end; - int start = 0; - int copied = 0; - char iabuf[INET_ADDRSTRLEN]; - - for (;;) { - oh = __get_header(orig, field, &start); - if (!ast_strlen_zero(oh)) { - if (!copied) { /* Only check for empty rport in topmost via header */ - char *rport; - char new[256]; - - /* Find ;rport; (empty request) */ - rport = strstr(oh, ";rport"); - if (rport && *(rport+6) == '=') - rport = NULL; /* We already have a parameter to rport */ - - if (rport && (ast_test_flag(p, SIP_NAT) == SIP_NAT_ALWAYS)) { - /* We need to add received port - rport */ - ast_copy_string(tmp, oh, sizeof(tmp)); - - rport = strstr(tmp, ";rport"); - - if (rport) { - end = strchr(rport + 1, ';'); - if (end) - memmove(rport, end, strlen(end) + 1); - else - *rport = '\0'; - } - - /* Add rport to first VIA header if requested */ - /* 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)); - } else { - /* We should *always* add a received to the topmost via */ - snprintf(new, sizeof(new), "%s;received=%s", oh, ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr)); - } - add_header(req, field, new); - } else { - /* Add the following via headers untouched */ - add_header(req, field, oh); - } - copied++; - } else - break; - } - if (!copied) { - ast_log(LOG_NOTICE, "No header field '%s' present to copy\n", field); - return -1; - } - return 0; -} - -/*! \brief 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); -} - -/*! \brief 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); -} - -/*! \brief 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->method = SIP_RESPONSE; - 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; -} - -/*! \brief 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++; - req->method = sipmethod; - return 0; -} - - -/*! \brief 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=%s", 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 (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER)) { - /* For registration responses, we also need expiry and - contact info */ - char tmp[256]; - - snprintf(tmp, sizeof(tmp), "%d", p->expiry); - add_header(resp, "Expires", tmp); - if (p->expiry) { /* Only add contact if we have an expiry time */ - char contact[256]; - snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, p->expiry); - add_header(resp, "Contact", contact); /* Not when we unregister */ - } - } else if (p->our_contact[0]) { - add_header(resp, "Contact", p->our_contact); - } - return 0; -} - -/*! \brief reqprep: Initialize a SIP request response 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; - int is_strict = 0; /* Strict routing flag */ - - memset(req, 0, sizeof(struct sip_request)); - - snprintf(p->lastmsg, sizeof(p->lastmsg), "Tx: %s", sip_methods[sipmethod].text); - - if (!seqno) { - p->ocseq++; - seqno = p->ocseq; - } - - if (newbranch) { - p->branch ^= thread_safe_rand(); - build_via(p, p->via, sizeof(p->via)); - } - - /* Check for strict or loose router */ - if (p->route && !ast_strlen_zero(p->route->hop) && strstr(p->route->hop,";lr") == NULL) - is_strict = 1; - - if (sipmethod == SIP_CANCEL) { - c = p->initreq.rlPart2; /* Use original URI */ - } else if (sipmethod == SIP_ACK) { - /* Use URI from Contact: in 200 OK (if INVITE) - (we only have the contacturi on INVITEs) */ - if (!ast_strlen_zero(p->okcontacturi)) - c = is_strict ? p->route->hop : p->okcontacturi; - else - c = p->initreq.rlPart2; - } else if (!ast_strlen_zero(p->okcontacturi)) { - c = is_strict ? p->route->hop : p->okcontacturi; /* Use for BYE or REINVITE */ - } else if (!ast_strlen_zero(p->uri)) { - c = p->uri; - } else { - /* 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); - if (is_strict) - add_route(req, p->route->next); - else - add_route(req, p->route); - } - - ot = get_header(orig, "To"); - of = get_header(orig, "From"); - - /* Add tag *unless* this is a CANCEL, in which case we need to send it exactly - as our original request, including tag (or presumably lack thereof) */ - if (!strcasestr(ot, "tag=") && sipmethod != SIP_CANCEL) { - /* Add the proper tag if we don't have it already. If they have specified - their tag, use it. Otherwise, use our own tag */ - if (ast_test_flag(p, 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=%s", 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); - add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS); - - if (p->rpid) - add_header(req, "Remote-Party-ID", p->rpid); - - return 0; -} - -/*! \brief __transmit_response: Base transmit response function */ -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_contentLength(&resp, 0); - /* If we are cancelling an incoming invite for some reason, add information - about the reason why we are doing this in clear text */ - if (p->owner && p->owner->hangupcause) { - add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause)); - } - add_blank_header(&resp); - return send_response(p, &resp, reliable, seqno); -} - -/*! \brief 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); -} - -/*! \brief 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); -} - -/*! \brief 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); -} - -/*! \brief 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); -} - -/*! \brief 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_contentLength(&resp, 0); - add_blank_header(&resp); - return send_response(p, &resp, 0, 0); -} - -/*! \brief 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_contentLength(&resp, 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_contentLength(&resp, 0); - add_blank_header(&resp); - return send_response(p, &resp, reliable, seqno); -} - -/*! \brief 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 */ - add_header(req, "Content-Type", "text/plain"); - add_header_contentLength(req, strlen(text)); - add_line(req, text); - return 0; -} - -/*! \brief 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]; - - snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=250\r\n", digit); - add_header(req, "Content-Type", "application/dtmf-relay"); - add_header_contentLength(req, strlen(tmp)); - add_line(req, tmp); - return 0; -} - -/*! \brief add_vidupdate: add XML encoded media control with update ---*/ -/* XML: The only way to turn 0 bits of information into a few hundred. */ -static int add_vidupdate(struct sip_request *req) -{ - const char *xml_is_a_huge_waste_of_space = - "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n" - " <media_control>\r\n" - " <vc_primitive>\r\n" - " <to_encoder>\r\n" - " <picture_fast_update>\r\n" - " </picture_fast_update>\r\n" - " </to_encoder>\r\n" - " </vc_primitive>\r\n" - " </media_control>\r\n"; - add_header(req, "Content-Type", "application/media_control+xml"); - add_header_contentLength(req, strlen(xml_is_a_huge_waste_of_space)); - add_line(req, xml_is_a_huge_waste_of_space); - return 0; -} - -static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate, - char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, - int debug) -{ - int rtp_code; - - if (debug) - ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec)); - if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1) - return; - - ast_build_string(m_buf, m_size, " %d", rtp_code); - ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code, - ast_rtp_lookup_mime_subtype(1, codec), - sample_rate); - if (codec == AST_FORMAT_G729A) - /* Indicate that we don't support VAD (G.729 annex B) */ - ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code); -} - -static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate, - char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, - int debug) -{ - int rtp_code; - - if (debug) - ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype(0, format)); - if ((rtp_code = ast_rtp_lookup_code(p->rtp, 0, format)) == -1) - return; - - ast_build_string(m_buf, m_size, " %d", rtp_code); - ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code, - ast_rtp_lookup_mime_subtype(0, format), - sample_rate); - if (format == AST_RTP_DTMF) - /* Indicate we support DTMF and FLASH... */ - ast_build_string(a_buf, a_size, "a=fmtp:%d 0-16\r\n", rtp_code); -} - -/*! \brief add_sdp: Add Session Description Protocol message ---*/ -static int add_sdp(struct sip_request *resp, struct sip_pvt *p) -{ - int len = 0; - int pref_codec; - int alreadysent = 0; - 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_audio[256]; - char m_video[256]; - char a_audio[1024]; - char a_video[1024]; - char *m_audio_next = m_audio; - char *m_video_next = m_video; - size_t m_audio_left = sizeof(m_audio); - size_t m_video_left = sizeof(m_video); - char *a_audio_next = a_audio; - char *a_video_next = a_video; - size_t a_audio_left = sizeof(a_audio); - size_t a_video_left = sizeof(a_video); - char iabuf[INET_ADDRSTRLEN]; - int x; - int capability; - struct sockaddr_in dest; - struct sockaddr_in vdest = { 0, }; - int debug; - - debug = sip_debug_test_pvt(p); - - len = 0; - if (!p->rtp) { - ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n"); - return -1; - } - capability = p->jointcapability; - - 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) { -#ifdef SIP_MIDCOM - if (m_cb && p->r) { - struct sockaddr_in redirip_hook; - char iabuf2[INET_ADDRSTRLEN]; - m_cb->ast_get_redirip_audio_hook(p->r, &redirip_hook); - ast_log(LOG_DEBUG, "Replacing %s:%d by %s:%d in SDP before sending to %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->redirip.sin_addr), ntohs(p->redirip.sin_port), ast_inet_ntoa(iabuf2, sizeof(iabuf2), redirip_hook.sin_addr), ntohs(redirip_hook.sin_port), p->username); - dest.sin_port = redirip_hook.sin_port; - dest.sin_addr = redirip_hook.sin_addr; - } else { - dest.sin_port = p->redirip.sin_port; - dest.sin_addr = p->redirip.sin_addr; - } -#else - dest.sin_port = p->redirip.sin_port; - dest.sin_addr = p->redirip.sin_addr; -#endif - 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) { -#ifdef SIP_MIDCOM - if (m_cb && p->r) { - struct sockaddr_in vredirip_hook; - char iabuf2[INET_ADDRSTRLEN]; - m_cb->ast_get_vredirip_video_hook(p->r, &vredirip_hook); - ast_log(LOG_DEBUG, "Replacing %s:%d by %s:%d in video SDP before sending to %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->vredirip.sin_addr), ntohs(p->vredirip.sin_port), ast_inet_ntoa(iabuf2, sizeof(iabuf2), vredirip_hook.sin_addr), ntohs(vredirip_hook.sin_port), p->username); - vdest.sin_port = vredirip_hook.sin_port; - vdest.sin_addr = vredirip_hook.sin_addr; - } else { - vdest.sin_port = p->vredirip.sin_port; - vdest.sin_addr = p->vredirip.sin_addr; - } -#else - vdest.sin_port = p->vredirip.sin_port; - vdest.sin_addr = p->vredirip.sin_addr; -#endif - } 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)); - } - - /* We break with the "recommendation" and send our IP, in order that our - peer doesn't have to ast_gethostbyname() us */ - - 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"); - - ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port)); - ast_build_string(&m_video_next, &m_video_left, "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 (p->prefcodec <= AST_FORMAT_MAX_AUDIO) - add_codec_to_sdp(p, p->prefcodec, 8000, - &m_audio_next, &m_audio_left, - &a_audio_next, &a_audio_left, - debug); - else - add_codec_to_sdp(p, p->prefcodec, 90000, - &m_video_next, &m_video_left, - &a_video_next, &a_video_left, - debug); - 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)) - continue; - - if (alreadysent & pref_codec) - continue; - - if (pref_codec <= AST_FORMAT_MAX_AUDIO) - add_codec_to_sdp(p, pref_codec, 8000, - &m_audio_next, &m_audio_left, - &a_audio_next, &a_audio_left, - debug); - else - add_codec_to_sdp(p, pref_codec, 90000, - &m_video_next, &m_video_left, - &a_video_next, &a_video_left, - debug); - 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)) - continue; - - if (alreadysent & x) - continue; - - if (x <= AST_FORMAT_MAX_AUDIO) - add_codec_to_sdp(p, x, 8000, - &m_audio_next, &m_audio_left, - &a_audio_next, &a_audio_left, - debug); - else - add_codec_to_sdp(p, x, 90000, - &m_video_next, &m_video_left, - &a_video_next, &a_video_left, - debug); - } - - for (x = 1; x <= AST_RTP_MAX; x <<= 1) { - if (!(p->noncodeccapability & x)) - continue; - - add_noncodec_to_sdp(p, x, 8000, - &m_audio_next, &m_audio_left, - &a_audio_next, &a_audio_left, - debug); - } - - ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n"); - - if ((m_audio_left < 2) || (m_video_left < 2) || (a_audio_left == 0) || (a_video_left == 0)) - ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n"); - - ast_build_string(&m_audio_next, &m_audio_left, "\r\n"); - ast_build_string(&m_video_next, &m_video_left, "\r\n"); - - len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m_audio) + strlen(a_audio); - if ((p->vrtp) && (!ast_test_flag(p, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */ - len += strlen(m_video) + strlen(a_video); - - add_header(resp, "Content-Type", "application/sdp"); - add_header_contentLength(resp, len); - add_line(resp, v); - add_line(resp, o); - add_line(resp, s); - add_line(resp, c); - add_line(resp, t); - add_line(resp, m_audio); - add_line(resp, a_audio); - if ((p->vrtp) && (!ast_test_flag(p, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */ - add_line(resp, m_video); - add_line(resp, a_video); - } - - /* Update lastrtprx when we send our SDP */ - time(&p->lastrtprx); - time(&p->lastrtptx); - - return 0; -} - -/*! \brief 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; x < src->headers; x++) - dst->header[x] += offset; - for (x=0; x < src->lines; x++) - dst->line[x] += offset; -} - -/*! \brief 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); - if (p->rtp) { - ast_rtp_offered_from_local(p->rtp, 0); - add_sdp(&resp, p); - } else { - ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid); - } -#ifdef SIP_MIDCOM - if (m_cb) { - if (!m_cb->transmit_response_with_sdp_hook(p)) { - ast_log(LOG_NOTICE, "Failed transmit_response_with_sdp_hook()\n"); - return -1; - } - } -#endif - return send_response(p, &resp, retrans, seqno); -} - -/*! \brief 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; -} - -/*! \brief 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; - -#ifdef SIP_MIDCOM - if (m_cb) { - if (!m_cb->transmit_reinvite_with_sdp_hook(p)) { - ast_log(LOG_NOTICE, "Failed transmit_reinvite_with_sdp_hook()\n"); - if (p->owner) - ast_queue_hangup(p->owner); - else - ast_set_flag(p, SIP_NEEDDESTROY); - } - } -#endif - - 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); - if (sipdebug) - add_header(&req, "X-asterisk-info", "SIP re-invite (RTP bridge)"); - ast_rtp_offered_from_local(p->rtp, 1); - add_sdp(&req, p); - /* Use this as the basis */ - copy_request(&p->initreq, &req); - parse_request(&p->initreq); - if (sip_debug_test_pvt(p)) - ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); - p->lastinvite = p->ocseq; - ast_set_flag(p, SIP_OUTGOING); - return send_request(p, &req, 1, p->ocseq); -} - -/*! \brief 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 (!ast_strlen_zero(c)) - ast_copy_string(p->uri, c, sizeof(p->uri)); -} - -/*! \brief 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 != 5060) /* Needs to be 5060, according to the RFC */ - snprintf(p->our_contact, sizeof(p->our_contact), "<sip:%s%s%s:%d>", 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), "<sip:%s%s%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip)); -} - -/*! \brief build_rpid: Build the Remote Party-ID & From using callingpres options ---*/ -static void build_rpid(struct sip_pvt *p) -{ - int send_pres_tags = 1; - const char *privacy=NULL; - const char *screen=NULL; - char buf[256]; - const char *clid = default_callerid; - const char *clin = NULL; - char iabuf[INET_ADDRSTRLEN]; - const char *fromdomain; - - if (p->rpid || p->rpid_from) - return; - - if (p->owner && p->owner->cid.cid_num) - clid = p->owner->cid.cid_num; - if (p->owner && p->owner->cid.cid_name) - clin = p->owner->cid.cid_name; - if (ast_strlen_zero(clin)) - clin = clid; - - switch (p->callingpres) { - case AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED: - privacy = "off"; - screen = "no"; - break; - case AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN: - privacy = "off"; - screen = "pass"; - break; - case AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN: - privacy = "off"; - screen = "fail"; - break; - case AST_PRES_ALLOWED_NETWORK_NUMBER: - privacy = "off"; - screen = "yes"; - break; - case AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED: - privacy = "full"; - screen = "no"; - break; - case AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN: - privacy = "full"; - screen = "pass"; - break; - case AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN: - privacy = "full"; - screen = "fail"; - break; - case AST_PRES_PROHIB_NETWORK_NUMBER: - privacy = "full"; - screen = "pass"; - break; - case AST_PRES_NUMBER_NOT_AVAILABLE: - send_pres_tags = 0; - break; - default: - ast_log(LOG_WARNING, "Unsupported callingpres (%d)\n", p->callingpres); - if ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) - privacy = "full"; - else - privacy = "off"; - screen = "no"; - break; - } - - fromdomain = ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain; - - snprintf(buf, sizeof(buf), "\"%s\" <sip:%s@%s>", clin, clid, fromdomain); - if (send_pres_tags) - snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), ";privacy=%s;screen=%s", privacy, screen); - p->rpid = strdup(buf); - - snprintf(buf, sizeof(buf), "\"%s\" <sip:%s@%s>;tag=%s", clin, - ast_strlen_zero(p->fromuser) ? clid : p->fromuser, - fromdomain, p->tag); - p->rpid_from = strdup(buf); -} - -/*! \brief initreqprep: Initiate new SIP request to peer/user ---*/ -static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod) -{ - char invite_buf[256] = ""; - char *invite = invite_buf; - size_t invite_max = sizeof(invite_buf); - char from[256]; - char to[256]; - char tmp[BUFSIZ/2]; - char tmp2[BUFSIZ/2]; - char iabuf[INET_ADDRSTRLEN]; - char *l = NULL, *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 (; x < strlen(p->username); 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 we are not sending RPID and user wants his callerid restricted */ - if (!ast_test_flag(p, SIP_SENDRPID) && ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED)) { - l = CALLERID_UNKNOWN; - n = l; - } - if (!l) - l = default_callerid; - if (ast_strlen_zero(n)) - n = l; - /* Allow user to be overridden */ - if (!ast_strlen_zero(p->fromuser)) - l = p->fromuser; - else /* Save for any further attempts */ - ast_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 (pedanticsipchecking) { - ast_uri_encode(n, tmp, sizeof(tmp), 0); - n = tmp; - ast_uri_encode(l, tmp2, sizeof(tmp2), 0); - l = tmp2; - } - - if ((ourport != 5060) && ast_strlen_zero(p->fromdomain)) /* Needs to be 5060 */ - snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", 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\" <sip:%s@%s>;tag=%s", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, p->tag); - - /* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */ - if (!ast_strlen_zero(p->fullcontact)) { - /* If we have full contact, trust it */ - ast_build_string(&invite, &invite_max, "%s", p->fullcontact); - } else { - /* Otherwise, use the username while waiting for registration */ - ast_build_string(&invite, &invite_max, "sip:"); - if (!ast_strlen_zero(p->username)) { - n = p->username; - if (pedanticsipchecking) { - ast_uri_encode(n, tmp, sizeof(tmp), 0); - n = tmp; - } - ast_build_string(&invite, &invite_max, "%s@", n); - } - ast_build_string(&invite, &invite_max, "%s", p->tohost); - if (ntohs(p->sa.sin_port) != 5060) /* Needs to be 5060 */ - ast_build_string(&invite, &invite_max, ":%d", ntohs(p->sa.sin_port)); - ast_build_string(&invite, &invite_max, "%s", urioptions); - } - - /* If custom URI options have been provided, append them */ - if (p->options && p->options->uri_options) - ast_build_string(&invite, &invite_max, ";%s", p->options->uri_options); - - ast_copy_string(p->uri, invite_buf, sizeof(p->uri)); - - /* If there is a VXML URL append it to the SIP URL */ - if (p->options && p->options->vxml_url) { - snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url); - } else { - snprintf(to, sizeof(to), "<%s>", p->uri); - } - memset(req, 0, sizeof(struct sip_request)); - init_req(req, sipmethod, p->uri); - snprintf(tmp, sizeof(tmp), "%d %s", ++p->ocseq, sip_methods[sipmethod].text); - - add_header(req, "Via", p->via); - /* SLD: FIXME?: do Route: here too? I think not cos this is the first request. - * OTOH, then we won't have anything in p->route anyway */ - /* Build Remote Party-ID and From */ - if (ast_test_flag(p, SIP_SENDRPID) && (sipmethod == SIP_INVITE)) { - build_rpid(p); - add_header(req, "From", p->rpid_from); - } else { - add_header(req, "From", from); - } - add_header(req, "To", to); - ast_copy_string(p->exten, l, sizeof(p->exten)); - build_contact(p); - 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); - add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS); - if (p->rpid) - add_header(req, "Remote-Party-ID", p->rpid); -} - -/*! \brief transmit_invite: Build REFER/INVITE/OPTIONS message and transmit it ---*/ -static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init) -{ - struct sip_request req; - - req.method = sipmethod; - if (init) { - /* Bump branch even on initial requests */ - p->branch ^= thread_safe_rand(); - build_via(p, p->via, sizeof(p->via)); - if (init > 1) - initreqprep(&req, p, sipmethod); - else - reqprep(&req, p, sipmethod, 0, 1); - } else - reqprep(&req, p, sipmethod, 0, 1); - - if (p->options && p->options->auth) - add_header(&req, p->options->authheader, p->options->auth); - append_date(&req); - if (sipmethod == SIP_REFER) { /* Call transfer */ - if (!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 ((req.method != SIP_OPTIONS) && p->options && !ast_strlen_zero(p->options->osptoken)) { - ast_log(LOG_DEBUG,"Adding OSP Token: %s\n", p->options->osptoken); - add_header(&req, "P-OSP-Auth-Token", p->options->osptoken); - } -#endif - if (p->options && !ast_strlen_zero(p->options->distinctive_ring)) - { - add_header(&req, "Alert-Info", p->options->distinctive_ring); - } - add_header(&req, "Allow", ALLOWED_METHODS); - if (p->options && p->options->addsipheaders ) { - 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) { - char *headdup; - 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); - headdup = ast_strdupa(header); - /* Strip of the starting " (if it's there) */ - if (*headdup == '"') - headdup++; - if ((content = strchr(headdup, ':'))) { - *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, headdup, content); - if (sipdebug) - ast_log(LOG_DEBUG, "Adding SIP Header \"%s\" with content :%s: \n", headdup, content); - } - } - } - } - } - } - if (sdp && p->rtp) { - ast_rtp_offered_from_local(p->rtp, 1); - add_sdp(&req, p); - } else { - add_header_contentLength(&req, 0); - add_blank_header(&req); - } - - if (!p->initreq.headers) { - /* Use this as the basis */ - copy_request(&p->initreq, &req); - parse_request(&p->initreq); - if (sip_debug_test_pvt(p)) - ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); - } - p->lastinvite = p->ocseq; - return send_request(p, &req, init ? 2 : 1, p->ocseq); -} - -/*! \brief transmit_state_notify: Used in the SUBSCRIBE notification subsystem ----*/ -static int transmit_state_notify(struct sip_pvt *p, int state, int full, int substate) -{ - char tmp[4000], from[256], to[256]; - char *t = tmp, *c, *a, *mfrom, *mto; - size_t maxbytes = sizeof(tmp); - struct sip_request req; - char hint[AST_MAX_EXTENSION]; - char *statestring = "terminated"; - const struct cfsubscription_types *subscriptiontype; - enum state { NOTIFY_OPEN, NOTIFY_INUSE, NOTIFY_CLOSED } local_state = NOTIFY_OPEN; - char *pidfstate = "--"; - char *pidfnote= "Ready"; - - memset(from, 0, sizeof(from)); - memset(to, 0, sizeof(to)); - memset(tmp, 0, sizeof(tmp)); - - switch (state) { - case (AST_EXTENSION_RINGING | AST_EXTENSION_INUSE): - if (global_notifyringing) - statestring = "early"; - else - statestring = "confirmed"; - local_state = NOTIFY_INUSE; - pidfstate = "busy"; - pidfnote = "Ringing"; - break; - case AST_EXTENSION_RINGING: - statestring = "early"; - local_state = NOTIFY_INUSE; - pidfstate = "busy"; - pidfnote = "Ringing"; - break; - case AST_EXTENSION_INUSE: - statestring = "confirmed"; - local_state = NOTIFY_INUSE; - pidfstate = "busy"; - pidfnote = "On the phone"; - break; - case AST_EXTENSION_BUSY: - statestring = "confirmed"; - local_state = NOTIFY_CLOSED; - pidfstate = "busy"; - pidfnote = "On the phone"; - break; - case AST_EXTENSION_UNAVAILABLE: - statestring = "confirmed"; - local_state = NOTIFY_CLOSED; - pidfstate = "away"; - pidfnote = "Unavailable"; - break; - case AST_EXTENSION_NOT_INUSE: - default: - /* Default setting */ - break; - } - - subscriptiontype = find_subscription_type(p->subscribed); - - /* Check which device/devices we are watching and if they are registered */ - if (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, p->context, p->exten)) { - /* If they are not registered, we will override notification and show no availability */ - if (ast_device_state(hint) == AST_DEVICE_UNAVAILABLE) { - local_state = NOTIFY_CLOSED; - pidfstate = "away"; - pidfnote = "Not online"; - } - } - - ast_copy_string(from, get_header(&p->initreq, "From"), sizeof(from)); - c = get_in_brackets(from); - if (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; - - ast_copy_string(to, get_header(&p->initreq, "To"), sizeof(to)); - c = get_in_brackets(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; - - reqprep(&req, p, SIP_NOTIFY, 0, 1); - - - add_header(&req, "Event", subscriptiontype->event); - add_header(&req, "Content-Type", subscriptiontype->mediatype); - switch(state) { - case AST_EXTENSION_DEACTIVATED: - if (p->subscribed == TIMEOUT) - add_header(&req, "Subscription-State", "terminated;reason=timeout"); - else { - add_header(&req, "Subscription-State", "terminated;reason=probation"); - add_header(&req, "Retry-After", "60"); - } - break; - case AST_EXTENSION_REMOVED: - add_header(&req, "Subscription-State", "terminated;reason=noresource"); - break; - break; - default: - if (p->expiry) - add_header(&req, "Subscription-State", "active"); - else /* Expired */ - add_header(&req, "Subscription-State", "terminated;reason=timeout"); - } - switch (p->subscribed) { - case XPIDF_XML: - case CPIM_PIDF_XML: - ast_build_string(&t, &maxbytes, "<?xml version=\"1.0\"?>\n"); - ast_build_string(&t, &maxbytes, "<!DOCTYPE presence PUBLIC \"-//IETF//DTD RFCxxxx XPIDF 1.0//EN\" \"xpidf.dtd\">\n"); - ast_build_string(&t, &maxbytes, "<presence>\n"); - ast_build_string(&t, &maxbytes, "<presentity uri=\"%s;method=SUBSCRIBE\" />\n", mfrom); - ast_build_string(&t, &maxbytes, "<atom id=\"%s\">\n", p->exten); - ast_build_string(&t, &maxbytes, "<address uri=\"%s;user=ip\" priority=\"0.800000\">\n", mto); - ast_build_string(&t, &maxbytes, "<status status=\"%s\" />\n", (local_state == NOTIFY_OPEN) ? "open" : (local_state == NOTIFY_INUSE) ? "inuse" : "closed"); - ast_build_string(&t, &maxbytes, "<msnsubstatus substatus=\"%s\" />\n", (local_state == NOTIFY_OPEN) ? "online" : (local_state == NOTIFY_INUSE) ? "onthephone" : "offline"); - ast_build_string(&t, &maxbytes, "</address>\n</atom>\n</presence>\n"); - break; - case PIDF_XML: /* Eyebeam supports this format */ - ast_build_string(&t, &maxbytes, "<?xml version=\"1.0\" encoding=\"ISO-8859-1\"?>\n"); - ast_build_string(&t, &maxbytes, "<presence xmlns=\"urn:ietf:params:xml:ns:pidf\" \nxmlns:pp=\"urn:ietf:params:xml:ns:pidf:person\"\nxmlns:es=\"urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status\"\nxmlns:ep=\"urn:ietf:params:xml:ns:pidf:rpid:rpid-person\"\nentity=\"%s\">\n", mfrom); - ast_build_string(&t, &maxbytes, "<pp:person><status>\n"); - if (pidfstate[0] != '-') - ast_build_string(&t, &maxbytes, "<ep:activities><ep:%s/></ep:activities>\n", pidfstate); - ast_build_string(&t, &maxbytes, "</status></pp:person>\n"); - ast_build_string(&t, &maxbytes, "<note>%s</note>\n", pidfnote); /* Note */ - ast_build_string(&t, &maxbytes, "<tuple id=\"%s\">\n", p->exten); /* Tuple start */ - ast_build_string(&t, &maxbytes, "<contact priority=\"1\">%s</contact>\n", mto); - if (pidfstate[0] == 'b') /* Busy? Still open ... */ - ast_build_string(&t, &maxbytes, "<status><basic>open</basic></status>\n"); - else - ast_build_string(&t, &maxbytes, "<status><basic>%s</basic></status>\n", (local_state != NOTIFY_CLOSED) ? "open" : "closed"); - ast_build_string(&t, &maxbytes, "</tuple>\n</presence>\n"); - break; - case DIALOG_INFO_XML: /* SNOM subscribes in this format */ - ast_build_string(&t, &maxbytes, "<?xml version=\"1.0\"?>\n"); - ast_build_string(&t, &maxbytes, "<dialog-info xmlns=\"urn:ietf:params:xml:ns:dialog-info\" version=\"%d\" state=\"%s\" entity=\"%s\">\n", p->dialogver++, full ? "full":"partial", mto); - if ((state & AST_EXTENSION_RINGING) && global_notifyringing) - ast_build_string(&t, &maxbytes, "<dialog id=\"%s\" direction=\"recipient\">\n", p->exten); - else - ast_build_string(&t, &maxbytes, "<dialog id=\"%s\">\n", p->exten); - ast_build_string(&t, &maxbytes, "<state>%s</state>\n", statestring); - ast_build_string(&t, &maxbytes, "</dialog>\n</dialog-info>\n"); - break; - case NONE: - default: - break; - } - - if (t > tmp + sizeof(tmp)) - ast_log(LOG_WARNING, "Buffer overflow detected!! (Please file a bug report)\n"); - - add_header_contentLength(&req, strlen(tmp)); - add_line(&req, tmp); - - return send_request(p, &req, 1, p->ocseq); -} - -/*! \brief transmit_notify_with_mwi: Notify user of messages waiting in voicemail ---*/ -/* Notification only works for registered peers with mailbox= definitions - * in sip.conf - * We use the SIP Event package message-summary - * MIME type defaults to "application/simple-message-summary"; - */ -static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten) -{ - struct sip_request req; - char tmp[500]; - char *t = tmp; - size_t maxbytes = sizeof(tmp); - char iabuf[INET_ADDRSTRLEN]; - - initreqprep(&req, p, SIP_NOTIFY); - add_header(&req, "Event", "message-summary"); - add_header(&req, "Content-Type", default_notifymime); - - ast_build_string(&t, &maxbytes, "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no"); - ast_build_string(&t, &maxbytes, "Message-Account: sip:%s@%s\r\n", !ast_strlen_zero(vmexten) ? vmexten : global_vmexten, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain); - ast_build_string(&t, &maxbytes, "Voice-Message: %d/%d (0/0)\r\n", newmsgs, oldmsgs); - - if (t > tmp + sizeof(tmp)) - ast_log(LOG_WARNING, "Buffer overflow detected!! (Please file a bug report)\n"); - - add_header_contentLength(&req, strlen(tmp)); - add_line(&req, tmp); - - if (!p->initreq.headers) { /* Use this as the basis */ - copy_request(&p->initreq, &req); - parse_request(&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); -} - -/*! \brief 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_request(&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); -} - -/*! \brief 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[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"); - - strcpy(tmp, "SIP/2.0 200 OK"); - add_header_contentLength(&req, strlen(tmp)); - add_line(&req, tmp); - - if (!p->initreq.headers) { - /* Use this as the basis */ - copy_request(&p->initreq, &req); - parse_request(&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); - -/*! \brief 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; - - if (r->call && recordhistory) { - char tmp[80]; - snprintf(tmp, sizeof(tmp), "Account: %s@%s", r->username, r->hostname); - append_history(r->call, "RegistryRenew", tmp); - } - /* Since registry's are only added/removed by the the monitor thread, this - may be overkill to reference/dereference at all here */ - if (sipdebug) - ast_log(LOG_NOTICE, " -- Re-registration for %s@%s\n", r->username, r->hostname); - - r->expire = -1; - __sip_do_register(r); - ASTOBJ_UNREF(r, sip_registry_destroy); - return 0; -} - -/*! \brief __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; -} - -/*! \brief 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 */ - __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\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate)); - ASTOBJ_UNREF(r,sip_registry_destroy); - return 0; -} - -/*! \brief transmit_register: 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; - make_our_tag(p->tag, sizeof(p->tag)); /* 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 registered 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; - } - if (recordhistory) { - char tmp[80]; - snprintf(tmp, sizeof(tmp), "Account: %s@%s", r->username, r->hostname); - append_history(p, "RegistryInit", tmp); - } - /* 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 - deleting it\n", r->timeout); - ast_sched_del(sched, r->timeout); - } - r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, r); - ast_log(LOG_DEBUG, "Scheduled a registration timeout for %s id #%d \n", r->hostname, r->timeout); - } - - if (strchr(r->username, '@')) { - snprintf(from, sizeof(from), "<sip:%s>;tag=%s", r->username, p->tag); - if (!ast_strlen_zero(p->theirtag)) - snprintf(to, sizeof(to), "<sip:%s>;tag=%s", r->username, p->theirtag); - else - snprintf(to, sizeof(to), "<sip:%s>", r->username); - } else { - snprintf(from, sizeof(from), "<sip:%s@%s>;tag=%s", r->username, p->tohost, p->tag); - if (!ast_strlen_zero(p->theirtag)) - snprintf(to, sizeof(to), "<sip:%s@%s>;tag=%s", r->username, p->tohost, p->theirtag); - else - snprintf(to, sizeof(to), "<sip:%s@%s>", r->username, p->tohost); - } - - /* Fromdomain is what we are registering to, regardless of actual - host name from SRV */ - if (!ast_strlen_zero(p->fromdomain)) - 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 ^= thread_safe_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); - add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS); - - - if (auth) /* Add auth header */ - add_header(&req, authheader, auth); - else if (!ast_strlen_zero(r->nonce)) { - char digest[1024]; - - /* We have auth data to reuse, build a digest header! */ - if (sipdebug) - ast_log(LOG_DEBUG, " >>> Re-using Auth data for %s@%s\n", r->username, r->hostname); - ast_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)); - p->noncecount = r->noncecount++; - - memset(digest,0,sizeof(digest)); - if(!build_reply_digest(p, sipmethod, digest, sizeof(digest))) - add_header(&req, "Authorization", digest); - else - ast_log(LOG_NOTICE, "No authorization available for authentication of registration to %s@%s\n", r->username, r->hostname); - - } - - snprintf(tmp, sizeof(tmp), "%d", default_expiry); - add_header(&req, "Expires", tmp); - add_header(&req, "Contact", p->our_contact); - add_header(&req, "Event", "registration"); - add_header_contentLength(&req, 0); - add_blank_header(&req); - copy_request(&p->initreq, &req); - parse_request(&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); -} - -/*! \brief 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); -} - -/*! \brief 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]; - - 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 = get_in_brackets(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), "<sip:%s@%s>", dest, c); - } else { - snprintf(referto, sizeof(referto), "<sip:%s>", dest); - } - - /* 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); -} - -/*! \brief 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); -} - -/*! \brief transmit_info_with_vidupdate: Send SIP INFO with video update request ---*/ -static int transmit_info_with_vidupdate(struct sip_pvt *p) -{ - struct sip_request req; - reqprep(&req, p, SIP_INFO, 0, 1); - add_vidupdate(&req); - return send_request(p, &req, 1, p->ocseq); -} - -/*! \brief 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_contentLength(&resp, 0); - add_blank_header(&resp); - return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq); -} - -/*! \brief 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)); - if(!build_reply_digest(p, sipmethod, digest, sizeof(digest))) { - if (p->options && p->options->auth_type == PROXY_AUTH) - add_header(&resp, "Proxy-Authorization", digest); - else if (p->options && p->options->auth_type == WWW_AUTH) - add_header(&resp, "Authorization", digest); - else /* Default, to be backwards compatible (maybe being too careful, but leaving it for now) */ - add_header(&resp, "Proxy-Authorization", digest); - } else - ast_log(LOG_WARNING, "No authentication available for call %s\n", p->callid); - } - /* If we are hanging up and know a cause for that, send it in clear text to make - debugging easier. */ - if (sipmethod == SIP_BYE) { - if (p->owner && p->owner->hangupcause) { - add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause)); - } - } - - add_header_contentLength(&resp, 0); - add_blank_header(&resp); - return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq); -} - -static void destroy_association(struct sip_peer *peer) -{ - if (!ast_test_flag((&global_flags_page2), SIP_PAGE2_IGNOREREGEXPIRE)) { - if (ast_test_flag(&(peer->flags_page2), SIP_PAGE2_RT_FROMCONTACT)) { - ast_update_realtime("sippeers", "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", "username", "", NULL); - } else { - ast_db_del("SIP/Registry", peer->name); - } - } -} - -/*! \brief 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)); - - destroy_association(peer); - - 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; -} - -/*! \brief 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_test_flag(&(peer->flags_page2), SIP_PAGE2_RT_FROMCONTACT)) - return; - if (ast_db_get("SIP/Registry", peer->name, data, sizeof(data))) - return; - - scan = data; - addr = strsep(&scan, ":"); - port_str = strsep(&scan, ":"); - expiry_str = strsep(&scan, ":"); - username = strsep(&scan, ":"); - contact = scan; /* Contact include sip: and has to be the last part of the database entry as long as we use : as a separator */ - - if (!inet_aton(addr, &in)) - return; - - if (port_str) - port = atoi(port_str); - else - return; - - if (expiry_str) - expiry = atoi(expiry_str); - else - return; - - if (username) - ast_copy_string(peer->username, username, sizeof(peer->username)); - if (contact) - ast_copy_string(peer->fullcontact, contact, sizeof(peer->fullcontact)); - - if (option_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, thread_safe_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); -} - -/*! \brief 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; -} - - -enum parse_register_result { - PARSE_REGISTER_FAILED, - PARSE_REGISTER_UPDATE, - PARSE_REGISTER_QUERY, -}; - -/*! \brief parse_register_contact: Parse contact header and save registration ---*/ -static enum parse_register_result parse_register_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 = strcasestr(get_header(req, "Contact"), ";expires="); - if (expires) { - char *ptr; - if ((ptr = strchr(expires, ';'))) - *ptr = '\0'; - if (sscanf(expires + 9, "%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)); - if (strchr(contact, '<') == NULL) { /* No <, check for ; and strip it */ - char *ptr = strchr(contact, ';'); /* This is Header options, not URI options */ - if (ptr) - *ptr = '\0'; - } - c = get_in_brackets(contact); - - /* if they did not specify Contact: or Expires:, they are querying - what we currently have stored as their contact address, so return - it - */ - if (ast_strlen_zero(c) && ast_strlen_zero(expires)) { - /* If we have an active registration, tell them when the registration is going to expire */ - if ((p->expire > -1) && !ast_strlen_zero(p->fullcontact)) { - pvt->expiry = ast_sched_when(sched, p->expire); - } - return PARSE_REGISTER_QUERY; - } else 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; - - destroy_association(p); - - 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 PARSE_REGISTER_UPDATE; - } - 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 PARSE_REGISTER_FAILED; - } - 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->flags_page2), SIP_PAGE2_RT_FROMCONTACT)) - 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 PARSE_REGISTER_UPDATE; -} - -/*! \brief 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; - } -} - -/*! \brief 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; - } -} - -/*! \brief 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; - } - } - - /* Only append the contact if we are dealing with a strict router */ - if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop,";lr") == NULL) ) { - /* 2nd append the Contact: if there is one */ - /* Can be multiple Contact headers, comma separated values - we just take the first */ - contact = get_header(req, "Contact"); - if (!ast_strlen_zero(contact)) { - 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); -} - -#ifdef OSP_SUPPORT -/*! \brief check_osptoken: Validate OSP token for user authrroization ---*/ -static int check_osptoken (struct sip_pvt *p, char *token) -{ - char tmp[80]; - - if (ast_osp_validate (NULL, token, &p->osphandle, &p->osptimelimit, p->cid_num, p->sa.sin_addr, p->exten) < 1) { - return (-1); - } else { - snprintf (tmp, sizeof (tmp), "%d", p->osphandle); - pbx_builtin_setvar_helper (p->owner, "_OSPHANDLE", tmp); - return (0); - } -} -#endif - -/*! \brief 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 *osptoken; -#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 || sipmethod == SIP_SUBSCRIBE) { - /* On a REGISTER, we have to use 401 and its family of headers instead of 407 and its family - of headers -- GO SIP! Whoo hoo! Two things that do the same thing but are used in - different circumstances! What a surprise. */ - response = "401 Unauthorized"; - reqheader = "Authorization"; - respheader = "WWW-Authenticate"; - } -#ifdef OSP_SUPPORT - else { - ast_log (LOG_DEBUG, "Checking OSP Authentication!\n"); - osptoken = get_header (req, "P-OSP-Auth-Token"); - switch (ast_test_flag (p, SIP_OSPAUTH)) { - case SIP_OSPAUTH_NO: - break; - case SIP_OSPAUTH_GATEWAY: - if (ast_strlen_zero (osptoken)) { - if (ast_strlen_zero (secret) && ast_strlen_zero (md5secret)) { - return (0); - } - } - else { - return (check_osptoken (p, osptoken)); - } - break; - case SIP_OSPAUTH_PROXY: - if (ast_strlen_zero (osptoken)) { - return (0); - } - else { - return (check_osptoken (p, osptoken)); - } - break; - case SIP_OSPAUTH_EXCLUSIVE: - if (ast_strlen_zero (osptoken)) { - return (-1); - } - else { - return (check_osptoken (p, osptoken)); - } - break; - default: - return (-1); - } - } -#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", thread_safe_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) { - - snprintf(randdata, randlen, "%08x", thread_safe_rand()); - if (ua_hash && !strncasecmp(ua_hash, resp_hash, strlen(resp_hash))) { - if (sipdebug) - ast_log(LOG_NOTICE, "stale nonce received from '%s'\n", get_header(req, "To")); - /* 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 with a new challenge */ - if (sipdebug) - ast_log(LOG_NOTICE, "Bad authentication received from '%s'\n", get_header(req, "To")); - 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; -} - -/*! \brief cb_extensionstate: Callback for the devicestate notification (SUBSCRIBE) support subsystem ---*/ -/* If you add an "hint" priority to the extension in the dial plan, - you will get notifications on device state changes */ -static int cb_extensionstate(char *context, char* exten, int state, void *data) -{ - struct sip_pvt *p = data; - - switch(state) { - case AST_EXTENSION_DEACTIVATED: /* Retry after a while */ - case AST_EXTENSION_REMOVED: /* Extension is gone */ - if (p->autokillid > -1) - sip_cancel_destroy(p); /* Remove subscription expiry for renewals */ - sip_scheddestroy(p, 15000); /* Delete subscription in 15 secs */ - ast_verbose(VERBOSE_PREFIX_2 "Extension state: Watcher for hint %s %s. Notify User %s\n", exten, state == AST_EXTENSION_DEACTIVATED ? "deactivated" : "removed", p->username); - p->stateid = -1; - p->subscribed = NONE; - append_history(p, "Subscribestatus", state == AST_EXTENSION_REMOVED ? "HintRemoved" : "Deactivated"); - break; - default: /* Tell user */ - p->laststate = state; - break; - } - transmit_state_notify(p, state, 1, 1); - - if (option_debug > 1) - ast_verbose(VERBOSE_PREFIX_1 "Extension Changed %s new state %s for Notify User %s\n", exten, ast_extension_state2str(state), p->username); - return 0; -} - -/*! \brief 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 = -3; - struct sip_peer *peer; - char tmp[256]; - char iabuf[INET_ADDRSTRLEN]; - char *name, *c; - char *t; - char *domain; - - /* Terminate URI */ - t = uri; - while(*t && (*t > 32) && (*t != ';')) - t++; - *t = '\0'; - - ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp)); - if (pedanticsipchecking) - ast_uri_decode(tmp); - - c = get_in_brackets(tmp); - /* 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 */ - if ((c = strchr(name, '@'))) { - *c++ = '\0'; - domain = c; - if ((c = strchr(domain, ':'))) /* Remove :port */ - *c = '\0'; - if (!AST_LIST_EMPTY(&domain_list)) { - if (!check_sip_domain(domain, NULL, 0)) { - transmit_response(p, "404 Not found (unknown domain)", &p->initreq); - return -3; - } - } - } - - 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_ERROR, "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); - switch (parse_register_contact(p, peer, req)) { - case PARSE_REGISTER_FAILED: - ast_log(LOG_WARNING, "Failed to parse contact info\n"); - break; - case PARSE_REGISTER_QUERY: - transmit_response_with_date(p, "200 OK", req); - peer->lastmsgssent = -1; - res = 0; - break; - case PARSE_REGISTER_UPDATE: - update_peer(peer, p->expiry); - /* Say OK and ask subsystem to retransmit msg counter */ - transmit_response_with_date(p, "200 OK", req); - peer->lastmsgssent = -1; - res = 0; - break; - } - } - } - } - if (!peer && autocreatepeer) { - /* Create peer if we have autocreate mode enabled */ - peer = temp_peer(name); - if (peer) { - ASTOBJ_CONTAINER_LINK(&peerl, peer); - peer->lastmsgssent = -1; - sip_cancel_destroy(p); - switch (parse_register_contact(p, peer, req)) { - case PARSE_REGISTER_FAILED: - ast_log(LOG_WARNING, "Failed to parse contact info\n"); - break; - case PARSE_REGISTER_QUERY: - transmit_response_with_date(p, "200 OK", req); - peer->lastmsgssent = -1; - res = 0; - break; - case PARSE_REGISTER_UPDATE: - /* Say OK and ask subsystem to retransmit msg counter */ - transmit_response_with_date(p, "200 OK", req); - manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Registered\r\n", peer->name); - peer->lastmsgssent = -1; - res = 0; - break; - } - } - } - if (!res) { - ast_device_state_changed("SIP/%s", peer->name); - } - if (res < 0) { - switch (res) { - case -1: - /* Wrong password in authentication. Go away, don't try again until you fixed it */ - transmit_response(p, "403 Forbidden (Bad auth)", &p->initreq); - break; - case -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; - case -3: - /* URI not found */ - transmit_response(p, "404 Not found", &p->initreq); - /* Set res back to -2 because we don't want to return an invalid domain message. That check already happened up above. */ - res = -2; - break; - } - if (option_debug > 1) { - ast_log(LOG_DEBUG, "SIP REGISTER attempt failed for %s : %s\n", - peer->name, - (res == -1) ? "Bad password" : ((res == -2 ) ? "Bad digest user" : "Peer not found")); - } - } - if (peer) - ASTOBJ_UNREF(peer,sip_destroy_peer); - - return res; -} - -/*! \brief 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 = get_in_brackets(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; -} - -/*! \brief get_destination: Find out who the call is for --*/ -static int get_destination(struct sip_pvt *p, struct sip_request *oreq) -{ - char tmp[256] = "", *uri, *a; - char tmpf[256], *from; - struct sip_request *req; - - req = oreq; - if (!req) - req = &p->initreq; - if (req->rlPart2) - ast_copy_string(tmp, req->rlPart2, sizeof(tmp)); - uri = get_in_brackets(tmp); - - ast_copy_string(tmpf, get_header(req, "From"), sizeof(tmpf)); - - from = get_in_brackets(tmpf); - - if (strncmp(uri, "sip:", 4)) { - ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", uri); - return -1; - } - uri += 4; - if (!ast_strlen_zero(from)) { - if (strncmp(from, "sip:", 4)) { - ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", from); - return -1; - } - from += 4; - } else - from = NULL; - - if (pedanticsipchecking) { - ast_uri_decode(uri); - ast_uri_decode(from); - } - - /* Get the target domain */ - if ((a = strchr(uri, '@'))) { - char *colon; - *a = '\0'; - a++; - colon = strchr(a, ':'); /* Remove :port */ - if (colon) - *colon = '\0'; - ast_copy_string(p->domain, a, sizeof(p->domain)); - } - /* Skip any options */ - if ((a = strchr(uri, ';'))) { - *a = '\0'; - } - - if (!AST_LIST_EMPTY(&domain_list)) { - char domain_context[AST_MAX_EXTENSION]; - - domain_context[0] = '\0'; - if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) { - if (!allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) { - ast_log(LOG_DEBUG, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain); - return -2; - } - } - /* If we have a context defined, overwrite the original context */ - if (!ast_strlen_zero(domain_context)) - ast_copy_string(p->context, domain_context, sizeof(p->context)); - } - - if (from) { - if ((a = strchr(from, ';'))) - *a = '\0'; - if ((a = strchr(from, '@'))) { - *a = '\0'; - ast_copy_string(p->fromdomain, a + 1, sizeof(p->fromdomain)); - } else - ast_copy_string(p->fromdomain, from, sizeof(p->fromdomain)); - } - if (sip_debug_test_pvt(p)) - ast_verbose("Looking for %s in %s (domain %s)\n", uri, p->context, p->domain); - - /* Return 0 if we have a matching extension */ - if (ast_exists_extension(NULL, p->context, uri, 1, from) || - !strcmp(uri, ast_pickup_ext())) { - if (!oreq) - ast_copy_string(p->exten, uri, sizeof(p->exten)); - return 0; - } - - /* Return 1 for overlap dialling support */ - if (ast_canmatch_extension(NULL, p->context, uri, 1, from) || - !strncmp(uri, ast_pickup_ext(),strlen(uri))) { - return 1; - } - - return -1; -} - -/*! \brief 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; -} - -/*! \brief 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 = get_in_brackets(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 { - if (pedanticsipchecking) { - ast_uri_decode(h_referred_by); - } - referred_by = get_in_brackets(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 - */ - ast_uri_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; -} - -/*! \brief 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 = get_in_brackets(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; -} - -/*! \brief check Via: header for hostname, port and rport request/answer */ -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; - - ast_copy_string(via, get_header(req, "Via"), sizeof(via)); - - /* Check for rport */ - c = strstr(via, ";rport"); - if (c && (c[6] != '=')) /* rport query, not answer */ - ast_set_flag(p, SIP_NAT_ROUTE); - - c = strchr(via, ';'); - if (c) - *c = '\0'; - - c = strchr(via, ' '); - if (c) { - *c = '\0'; - c = ast_skip_blanks(c+1); - if (strcasecmp(via, "SIP/2.0/UDP")) { - ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via); - return -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); - - 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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - - -/*! \brief check_user_full: Check if matching user or peer is defined ---*/ -/* Match user on From: user name and peer on IP/port */ -/* This is used on first invite (not re-invites) and subscribe requests */ -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 = NULL; - 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"); - if (pedanticsipchecking) - ast_uri_decode(of); - - ast_copy_string(from, of, sizeof(from)); - - memset(calleridname,0,sizeof(calleridname)); - get_calleridname(from, calleridname, sizeof(calleridname)); - if (calleridname[0]) - ast_copy_string(p->cid_name, calleridname, sizeof(p->cid_name)); - - rpid = get_header(req, "Remote-Party-ID"); - memset(rpid_num,0,sizeof(rpid_num)); - if (!ast_strlen_zero(rpid)) - p->callingpres = get_rpid_num(rpid,rpid_num, sizeof(rpid_num)); - - of = get_in_brackets(from); - if (ast_strlen_zero(p->exten)) { - t = 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); - } - /* save the URI part of the From header */ - 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(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 (ast_strlen_zero(of)) - return 0; - - if (!mailbox) /* If it's a mailbox SUBSCRIBE, don't check users */ - user = find_user(of, 1); - - /* Find user based on user name in the from header */ - if (user && ast_apply_ha(user->ha, sin)) { - ast_copy_flags(p, user, SIP_FLAGS_TO_COPY); - /* 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_FLAGS_TO_COPY); - /* Copy SIP extensions profile from INVITE */ - if (p->sipoptions) - user->sipoptions = p->sipoptions; - - /* If we have a call limit, set flag */ - if (user->call_limit) - ast_set_flag(p, 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->subscribecontext, user->subscribecontext, sizeof(p->subscribecontext)); - 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) || (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_AUTO)) - 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 */ - if (sipmethod == SIP_SUBSCRIBE) - /* For subscribes, match on peer name only */ - peer = find_peer(of, NULL, 1); - else - /* Look for peer based on the IP address we received data from */ - /* If peer is registered from this IP address or have this as a default - IP address, this call is from the peer - */ - peer = find_peer(NULL, &p->recv, 1); - - if (peer) { - if (debug) - ast_verbose("Found peer '%s'\n", peer->name); - /* Take the peer */ - ast_copy_flags(p, peer, SIP_FLAGS_TO_COPY); - - /* Copy SIP extensions profile to peer */ - if (p->sipoptions) - peer->sipoptions = p->sipoptions; - - /* replace callerid if rpid found, and not restricted */ - if (!ast_strlen_zero(rpid_num) && ast_test_flag(p, 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->subscribecontext, peer->subscribecontext, sizeof(p->subscribecontext)); - ast_copy_string(p->peermd5secret, peer->md5secret, sizeof(p->peermd5secret)); - p->peermd5secret[sizeof(p->peermd5secret)-1] = '\0'; - p->callingpres = peer->callingpres; - if (peer->maxms && peer->lastms) - p->timer_t1 = peer->lastms; - 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_FLAGS_TO_COPY); - /* If we have a call limit, set flag */ - if (peer->call_limit) - 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->prefs = peer->prefs; - p->jointcapability = peer->capability; - if (p->peercapability) - p->jointcapability &= p->peercapability; - if ((ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) || (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_AUTO)) - 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; -} - -/*! \brief 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); -} - -/*! \brief 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;x<req->lines;x++) { - strncat(buf, req->line[x], y); /* safe */ - y -= strlen(req->line[x]) + 1; - if (y < 0) - y = 0; - if (y != 0) - strcat(buf, "\n"); /* safe */ - } - return 0; -} - - -/*! \brief receive_message: Receive SIP MESSAGE method messages ---*/ -/* We only handle messages within current calls currently */ -/* Reference: RFC 3428 */ -static void receive_message(struct sip_pvt *p, struct sip_request *req) -{ - char buf[1024]; - struct ast_frame f; - char *content_type; - - content_type = get_header(req, "Content-Type"); - if (strcmp(content_type, "text/plain")) { /* No text/plain attachment */ - transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */ - ast_set_flag(p, SIP_NEEDDESTROY); - return; - } - - if (get_msg_text(buf, sizeof(buf), req)) { - ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid); - transmit_response(p, "202 Accepted", req); - ast_set_flag(p, SIP_NEEDDESTROY); - return; - } - - if (p->owner) { - if (sip_debug_test_pvt(p)) - ast_verbose("Message received: '%s'\n", buf); - memset(&f, 0, sizeof(f)); - f.frametype = AST_FRAME_TEXT; - f.subclass = 0; - f.offset = 0; - f.data = buf; - f.datalen = strlen(buf); - ast_queue_frame(p->owner, &f); - transmit_response(p, "202 Accepted", req); /* We respond 202 accepted, since we relay the message */ - } else { /* Message outside of a call, we do not support that */ - ast_log(LOG_WARNING,"Received message to %s from %s, dropped it...\n Content-Type:%s\n Message: %s\n", get_header(req,"To"), get_header(req,"From"), content_type, buf); - transmit_response(p, "405 Method Not Allowed", req); /* Good enough, or? */ - } - ast_set_flag(p, SIP_NEEDDESTROY); - return; -} - -/*! \brief sip_show_inuse: CLI Command to show calls within limits set by - call_limit ---*/ -static int sip_show_inuse(int fd, int argc, char *argv[]) { -#define FORMAT "%-25.25s %-15.15s %-15.15s \n" -#define FORMAT2 "%-25.25s %-15.15s %-15.15s \n" - char ilimits[40]; - char iused[40]; - int showall = 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->call_limit) - snprintf(ilimits, sizeof(ilimits), "%d", iterator->call_limit); - else - ast_copy_string(ilimits, "N/A", sizeof(ilimits)); - snprintf(iused, sizeof(iused), "%d", iterator->inUse); - if (showall || iterator->call_limit) - ast_cli(fd, FORMAT2, iterator->name, iused, ilimits); - ASTOBJ_UNLOCK(iterator); - } while (0) ); - - ast_cli(fd, FORMAT, "* Peer name", "In use", "Limit"); - - ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do { - ASTOBJ_RDLOCK(iterator); - if (iterator->call_limit) - snprintf(ilimits, sizeof(ilimits), "%d", iterator->call_limit); - else - ast_copy_string(ilimits, "N/A", sizeof(ilimits)); - snprintf(iused, sizeof(iused), "%d", iterator->inUse); - if (showall || iterator->call_limit) - ast_cli(fd, FORMAT2, iterator->name, iused, ilimits); - ASTOBJ_UNLOCK(iterator); - } while (0) ); - - return RESULT_SUCCESS; -#undef FORMAT -#undef FORMAT2 -} - -/*! \brief 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"; - } -} - -/*! \brief peer_status: Report Peer status in character string */ -/* returns 1 if peer is online, -1 if unmonitored */ -static int peer_status(struct sip_peer *peer, char *status, int statuslen) -{ - int res = 0; - if (peer->maxms) { - if (peer->lastms < 0) { - ast_copy_string(status, "UNREACHABLE", statuslen); - } else if (peer->lastms > peer->maxms) { - snprintf(status, statuslen, "LAGGED (%d ms)", peer->lastms); - res = 1; - } else if (peer->lastms) { - snprintf(status, statuslen, "OK (%d ms)", peer->lastms); - res = 1; - } else { - ast_copy_string(status, "UNKNOWN", statuslen); - } - } else { - ast_copy_string(status, "Unmonitored", statuslen); - /* Checking if port is 0 */ - res = -1; - } - return res; -} - -/*! \brief 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: <id> 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[]); - -/*! \brief 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 (!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_cli(s->fd, - "Event: PeerlistComplete\r\n" - "ListItems: %d\r\n" - "%s" - "\r\n", total, idtext); - return 0; -} - -/*! \brief 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); -} - -/*! \brief _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 %-8s %-10s\n" -#define FORMAT "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-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 (!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", "Port", "Status"); - } - - ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do { - char status[20] = ""; - char srch[2000]; - char pstatus; - - ASTOBJ_RDLOCK(iterator); - - if (havepattern && regexec(®exbuf, iterator->name, 0, NULL, 0)) { - ASTOBJ_UNLOCK(iterator); - continue; - } - - if (!ast_strlen_zero(iterator->username) && !s) - snprintf(name, sizeof(name), "%s/%s", iterator->name, iterator->username); - else - ast_copy_string(name, iterator->name, sizeof(name)); - - pstatus = peer_status(iterator, status, sizeof(status)); - if (pstatus) - peers_online++; - else { - if (pstatus == 0) - peers_offline++; - else { /* Unmonitored */ - /* 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 */ - 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 */ - - ntohs(iterator->addr.sin_port), status); - } else { /* Manager format */ - /* The names here need to be the same as other channels */ - 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); - } - - 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 -} - -/*! \brief 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; -} -/*! \brief print_group: Print call group and pickup group ---*/ -static void print_group(int fd, unsigned int group, int crlf) -{ - char buf[256]; - ast_cli(fd, crlf ? "%s\r\n" : "%s\n", ast_print_group(buf, sizeof(buf), group) ); -} - -/*! \brief 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"; - case SIP_DTMF_AUTO: - return "auto"; - } - return "<error>"; -} - -/*! \brief 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"; -} - -/*! \brief 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; -} - -/*! \brief print_codec_to_cli: Print codec list from preference to CLI/manager */ -static void print_codec_to_cli(int fd, struct ast_codec_pref *pref) -{ - int x, codec; - - for(x = 0; x < 32 ; x++) { - codec = ast_codec_pref_index(pref, x); - if (!codec) - break; - ast_cli(fd, "%s", ast_getformatname(codec)); - if (x < 31 && ast_codec_pref_index(pref, x + 1)) - ast_cli(fd, ","); - } - if (!x) - ast_cli(fd, "none"); -} - -static const char *domain_mode_to_text(const enum domain_mode mode) -{ - switch (mode) { - case SIP_DOMAIN_AUTO: - return "[Automatic]"; - case SIP_DOMAIN_CONFIG: - return "[Configured]"; - } - - return ""; -} - -/*! \brief sip_show_domains: CLI command to list local domains */ -#define FORMAT "%-40.40s %-20.20s %-16.16s\n" -static int sip_show_domains(int fd, int argc, char *argv[]) -{ - struct domain *d; - - if (AST_LIST_EMPTY(&domain_list)) { - ast_cli(fd, "SIP Domain support not enabled.\n\n"); - return RESULT_SUCCESS; - } else { - ast_cli(fd, FORMAT, "Our local SIP domains:", "Context", "Set by"); - AST_LIST_LOCK(&domain_list); - AST_LIST_TRAVERSE(&domain_list, d, list) - ast_cli(fd, FORMAT, d->domain, ast_strlen_zero(d->context) ? "(default)": d->context, - domain_mode_to_text(d->mode)); - AST_LIST_UNLOCK(&domain_list); - ast_cli(fd, "\n"); - return RESULT_SUCCESS; - } -} -#undef FORMAT - -static char mandescr_show_peer[] = -"Description: Show one SIP peer with details on current status.\n" -" The XML format is under development, feedback welcome! /oej\n" -"Variables: \n" -" Peer: <name> The peer name you want to check.\n" -" ActionID: <id> 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[]); - -/*! \brief 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 (ast_strlen_zero(peer)) { - astman_send_error(s, m, "Peer: <name> missing.\n"); - return 0; - } - a[0] = "sip"; - a[1] = "show"; - a[2] = "peer"; - a[3] = peer; - - if (!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" ); - return ret; -} - - - -/*! \brief 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)?"<Not set>":"<Set>"); - ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(peer->md5secret)?"<Not set>":"<Set>"); - 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)?"<Secret set>":(!ast_strlen_zero(auth->md5secret)?"<MD5secret set>" : "<Not set>")); - auth = auth->next; - } - ast_cli(fd, " Context : %s\n", peer->context); - ast_cli(fd, " Subscr.Cont. : %s\n", ast_strlen_zero(peer->subscribecontext)?"<Not set>":peer->subscribecontext); - 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, 0); - ast_cli(fd, " Pickupgroup : "); - print_group(fd, peer->pickupgroup, 0); - ast_cli(fd, " Mailbox : %s\n", peer->mailbox); - ast_cli(fd, " VM Extension : %s\n", peer->vmexten); - ast_cli(fd, " LastMsgsSent : %d\n", peer->lastmsgssent); - ast_cli(fd, " Call limit : %d\n", peer->call_limit); - 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, "<unspecified>")); - ast_cli(fd, " Expire : %d\n", peer->expire); - 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")); - ast_cli(fd, " Trust RPID : %s\n", (ast_test_flag(peer, SIP_TRUSTRPID) ? "Yes" : "No")); - ast_cli(fd, " Send RPID : %s\n", (ast_test_flag(peer, SIP_SENDRPID) ? "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 : ("); - print_codec_to_cli(fd, &peer->prefs); - - ast_cli(fd, ")\n"); - - ast_cli(fd, " Status : "); - peer_status(peer, status, sizeof(status)); - ast_cli(fd, "%s\n",status); - ast_cli(fd, " Useragent : %s\n", peer->useragent); - ast_cli(fd, " Reg. Contact : %s\n", peer->fullcontact); - if (peer->chanvars) { - ast_cli(fd, " Variables :\n"); - for (v = peer->chanvars ; v ; v = v->next) - ast_cli(fd, " %s = %s\n", v->name, v->value); - } - ast_cli(fd,"\n"); - ASTOBJ_UNREF(peer,sip_destroy_peer); - } else if (peer && type == 1) { /* manager listing */ - char *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, 1); - ast_cli(fd, "Pickupgroup: "); - print_group(fd, peer->pickupgroup, 1); - ast_cli(fd, "VoiceMailbox: %s\r\n", peer->mailbox); - ast_cli(fd, "LastMsgsSent: %d\r\n", peer->lastmsgssent); - ast_cli(fd, "Call limit: %d\r\n", peer->call_limit); - 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, "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: "); - peer_status(peer, status, 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; -} - -/*! \brief 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)?"<Not set>":"<Set>"); - ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(user->md5secret)?"<Not set>":"<Set>"); - ast_cli(fd, " Context : %s\n", user->context); - ast_cli(fd, " Language : %s\n", user->language); - if (!ast_strlen_zero(user->accountcode)) - ast_cli(fd, " Accountcode : %s\n", user->accountcode); - ast_cli(fd, " AMA flags : %s\n", ast_cdr_flags2str(user->amaflags)); - ast_cli(fd, " CallingPres : %s\n", ast_describe_caller_presentation(user->callingpres)); - ast_cli(fd, " Call limit : %d\n", user->call_limit); - ast_cli(fd, " Callgroup : "); - print_group(fd, user->callgroup, 0); - ast_cli(fd, " Pickupgroup : "); - print_group(fd, user->pickupgroup, 0); - ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), user->cid_name, user->cid_num, "<unspecified>")); - ast_cli(fd, " ACL : %s\n", (user->ha?"Yes":"No")); - ast_cli(fd, " Codec Order : ("); - 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; -} - -/*! \brief 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 -} - -/*! \brief sip_show_settings: List global settings for the SIP channel ---*/ -static int sip_show_settings(int fd, int argc, char *argv[]) -{ - char tmp[BUFSIZ]; - int realtimepeers = 0; - int realtimeusers = 0; - - realtimepeers = ast_check_realtime("sippeers"); - realtimeusers = ast_check_realtime("sipusers"); - - if (argc != 3) - return RESULT_SHOWUSAGE; - ast_cli(fd, "\n\nGlobal Settings:\n"); - ast_cli(fd, "----------------\n"); - ast_cli(fd, " SIP Port: %d\n", ntohs(bindaddr.sin_port)); - ast_cli(fd, " Bindaddress: %s\n", ast_inet_ntoa(tmp, sizeof(tmp), bindaddr.sin_addr)); - ast_cli(fd, " Videosupport: %s\n", videosupport ? "Yes" : "No"); - ast_cli(fd, " AutoCreatePeer: %s\n", autocreatepeer ? "Yes" : "No"); - ast_cli(fd, " Allow unknown access: %s\n", global_allowguest ? "Yes" : "No"); - ast_cli(fd, " Promsic. redir: %s\n", ast_test_flag(&global_flags, SIP_PROMISCREDIR) ? "Yes" : "No"); - ast_cli(fd, " SIP domain support: %s\n", AST_LIST_EMPTY(&domain_list) ? "No" : "Yes"); - ast_cli(fd, " Call to non-local dom.: %s\n", allow_external_domains ? "Yes" : "No"); - ast_cli(fd, " URI user is phone no: %s\n", ast_test_flag(&global_flags, SIP_USEREQPHONE) ? "Yes" : "No"); - ast_cli(fd, " Our auth realm %s\n", global_realm); - ast_cli(fd, " Realm. auth: %s\n", authl ? "Yes": "No"); - ast_cli(fd, " User Agent: %s\n", default_useragent); - ast_cli(fd, " MWI checking interval: %d secs\n", global_mwitime); - ast_cli(fd, " Reg. context: %s\n", ast_strlen_zero(regcontext) ? "(not set)" : regcontext); - ast_cli(fd, " Caller ID: %s\n", default_callerid); - ast_cli(fd, " From: Domain: %s\n", default_fromdomain); - ast_cli(fd, " Record SIP history: %s\n", recordhistory ? "On" : "Off"); - ast_cli(fd, " Call Events: %s\n", callevents ? "On" : "Off"); - ast_cli(fd, " IP ToS: 0x%x\n", tos); -#ifdef OSP_SUPPORT - ast_cli(fd, " OSP Support: Yes\n"); -#else - ast_cli(fd, " OSP Support: No\n"); -#endif - if (!realtimepeers && !realtimeusers) - ast_cli(fd, " SIP realtime: Disabled\n" ); - else - ast_cli(fd, " SIP realtime: Enabled\n" ); - - ast_cli(fd, "\nGlobal Signalling Settings:\n"); - ast_cli(fd, "---------------------------\n"); - ast_cli(fd, " Codecs: "); - print_codec_to_cli(fd, &prefs); - ast_cli(fd, "\n"); - ast_cli(fd, " Relax DTMF: %s\n", relaxdtmf ? "Yes" : "No"); - ast_cli(fd, " Compact SIP headers: %s\n", compactheaders ? "Yes" : "No"); - ast_cli(fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" ); - ast_cli(fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)"); - ast_cli(fd, " MWI NOTIFY mime type: %s\n", default_notifymime); - ast_cli(fd, " DNS SRV lookup: %s\n", srvlookup ? "Yes" : "No"); - ast_cli(fd, " Pedantic SIP support: %s\n", pedanticsipchecking ? "Yes" : "No"); - ast_cli(fd, " Reg. max duration: %d secs\n", max_expiry); - ast_cli(fd, " Reg. default duration: %d secs\n", default_expiry); - ast_cli(fd, " Outbound reg. timeout: %d secs\n", global_reg_timeout); - ast_cli(fd, " Outbound reg. attempts: %d\n", global_regattempts_max); - ast_cli(fd, " Notify ringing state: %s\n", global_notifyringing ? "Yes" : "No"); - ast_cli(fd, "\nDefault Settings:\n"); - ast_cli(fd, "-----------------\n"); - ast_cli(fd, " Context: %s\n", default_context); - ast_cli(fd, " Nat: %s\n", nat2str(ast_test_flag(&global_flags, SIP_NAT))); - ast_cli(fd, " DTMF: %s\n", dtmfmode2str(ast_test_flag(&global_flags, SIP_DTMF))); - ast_cli(fd, " Qualify: %d\n", default_qualify); - ast_cli(fd, " Use ClientCode: %s\n", ast_test_flag(&global_flags, SIP_USECLIENTCODE) ? "Yes" : "No"); - ast_cli(fd, " Progress inband: %s\n", (ast_test_flag(&global_flags, SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER) ? "Never" : (ast_test_flag(&global_flags, SIP_PROG_INBAND) == SIP_PROG_INBAND_NO) ? "No" : "Yes" ); - ast_cli(fd, " Language: %s\n", ast_strlen_zero(default_language) ? "(Defaults to English)" : default_language); - ast_cli(fd, " Musicclass: %s\n", global_musicclass); - ast_cli(fd, " Voice Mail Extension: %s\n", global_vmexten); - - - if (realtimepeers || realtimeusers) { - ast_cli(fd, "\nRealtime SIP Settings:\n"); - ast_cli(fd, "----------------------\n"); - ast_cli(fd, " Realtime Peers: %s\n", realtimepeers ? "Yes" : "No"); - ast_cli(fd, " Realtime Users: %s\n", realtimeusers ? "Yes" : "No"); - ast_cli(fd, " Cache Friends: %s\n", ast_test_flag(&global_flags_page2, SIP_PAGE2_RTCACHEFRIENDS) ? "Yes" : "No"); - ast_cli(fd, " Update: %s\n", ast_test_flag(&global_flags_page2, SIP_PAGE2_RTUPDATE) ? "Yes" : "No"); - ast_cli(fd, " Ignore Reg. Expire: %s\n", ast_test_flag(&global_flags_page2, SIP_PAGE2_IGNOREREGEXPIRE) ? "Yes" : "No"); - ast_cli(fd, " Auto Clear: %d\n", global_rtautoclear); - } - ast_cli(fd, "\n----\n"); - return RESULT_SUCCESS; -} - -/*! \brief subscription_type2str: Show subscription type in string format */ -static const char *subscription_type2str(enum subscriptiontype subtype) { - int i; - - for (i = 1; (i < (sizeof(subscription_types) / sizeof(subscription_types[0]))); i++) { - if (subscription_types[i].type == subtype) { - return subscription_types[i].text; - } - } - return subscription_types[0].text; -} - -/*! \brief find_subscription_type: Find subscription type in array */ -static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype) { - int i; - - for (i = 1; (i < (sizeof(subscription_types) / sizeof(subscription_types[0]))); i++) { - if (subscription_types[i].type == subtype) { - return &subscription_types[i]; - } - } - return &subscription_types[0]; -} - -/* Forward declaration */ -static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions); - -/*! \brief 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); -} - -/*! \brief 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 %-11.11s %-15.15s %-13.13s %-15.15s\n" -#define FORMAT2 "%-15.15s %-10.10s %-11.11s %-11.11s %-4.4s %-7.7s %-15.15s\n" -#define FORMAT "%-15.15s %-10.10s %-11.11s %5.5d/%5.5d %-4.4s %-3.3s %-3.3s %-15.15s\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", "Hold", "Last Message"); - else - ast_cli(fd, FORMAT3, "Peer", "User", "Call ID", "Extension", "Last state", "Type"); - while (cur) { - if (cur->subscribed == NONE && !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_CALL_ONHOLD) ? "Yes" : "No", - ast_test_flag(cur, SIP_NEEDDESTROY) ? "(d)" : "", - cur->lastmsg ); - numchans++; - } - if (cur->subscribed != NONE && 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->exten, ast_extension_state2str(cur->laststate), - subscription_type2str(cur->subscribed)); - numchans++; - } - cur = cur->next; - } - ast_mutex_unlock(&iflock); - if (!subscriptions) - ast_cli(fd, "%d active SIP channel%s\n", numchans, (numchans != 1) ? "s" : ""); - else - ast_cli(fd, "%d active SIP subscription%s\n", numchans, (numchans != 1) ? "s" : ""); - return RESULT_SUCCESS; -#undef FORMAT -#undef FORMAT2 -#undef FORMAT3 -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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 != NONE) - ast_cli(fd, " * Subscription (type: %s)\n", subscription_type2str(cur->subscribed)); - 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, " Audio IP: %s %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->redirip.sin_addr.s_addr ? cur->redirip.sin_addr : cur->ourip), cur->redirip.sin_addr.s_addr ? "(Outside bridge)" : "(local)" ); - ast_cli(fd, " Our Tag: %s\n", cur->tag); - ast_cli(fd, " Their Tag: %s\n", cur->theirtag); - ast_cli(fd, " SIP User agent: %s\n", cur->useragent); - if (!ast_strlen_zero(cur->username)) - ast_cli(fd, " Username: %s\n", cur->username); - if (!ast_strlen_zero(cur->peername)) - ast_cli(fd, " Peername: %s\n", cur->peername); - if (!ast_strlen_zero(cur->uri)) - ast_cli(fd, " Original uri: %s\n", cur->uri); - if (!ast_strlen_zero(cur->cid_num)) - ast_cli(fd, " Caller-ID: %s\n", cur->cid_num); - ast_cli(fd, " Need Destroy: %d\n", ast_test_flag(cur, 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; -} - -/*! \brief sip_show_history: Show history 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 != NONE) - 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; -} - -/*! \brief dump_history: Dump SIP history to debug log file at end of - lifespan for SIP dialog */ -void sip_dump_history(struct sip_pvt *dialog) -{ - int x; - struct sip_history *hist; - - if (!dialog) - return; - - ast_log(LOG_DEBUG, "\n---------- SIP HISTORY for '%s' \n", dialog->callid); - if (dialog->subscribed) - ast_log(LOG_DEBUG, " * Subscription\n"); - else - ast_log(LOG_DEBUG, " * SIP Call\n"); - x = 0; - hist = dialog->history; - while(hist) { - x++; - ast_log(LOG_DEBUG, " %d. %s\n", x, hist->event); - hist = hist->next; - } - if (!x) - ast_log(LOG_DEBUG, "Call '%s' has no history\n", dialog->callid); - ast_log(LOG_DEBUG, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid); - -} - - -/*! \brief handle_request_info: Receive SIP INFO Message ---*/ -/* Doesn't read the duration of the DTMF signal */ -static void handle_request_info(struct sip_pvt *p, struct sip_request *req) -{ - char buf[1024]; - unsigned int event; - 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) { /* not a PBX call */ - transmit_response(p, "481 Call leg/transaction does not exist", req); - ast_set_flag(p, SIP_NEEDDESTROY); - return; - } - - if (ast_strlen_zero(buf)) { - transmit_response(p, "200 OK", req); - return; - } - - if (buf[0] == '*') - event = 10; - else if (buf[0] == '#') - event = 11; - else if ((buf[0] >= 'A') && (buf[0] <= 'D')) - event = 12 + buf[0] - 'A'; - else - event = atoi(buf); - if (event == 16) { - /* send a FLASH event */ - struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH, }; - ast_queue_frame(p->owner, &f); - if (sipdebug) - ast_verbose("* DTMF-relay event received: FLASH\n"); - } else { - /* send a DTMF event */ - struct ast_frame f = { AST_FRAME_DTMF, }; - if (event < 10) { - f.subclass = '0' + event; - } else if (event < 11) { - f.subclass = '*'; - } else if (event < 12) { - f.subclass = '#'; - } else if (event < 16) { - f.subclass = 'A' + (event - 12); - } - ast_queue_frame(p->owner, &f); - if (sipdebug) - ast_verbose("* DTMF-relay event received: %c\n", f.subclass); - } - transmit_response(p, "200 OK", req); - return; - } else if (!strcasecmp(get_header(req, "Content-Type"), "application/media_control+xml")) { - /* Eh, we'll just assume it's a fast picture update for now */ - if (p->owner) - ast_queue_control(p->owner, AST_CONTROL_VIDUPDATE); - transmit_response(p, "200 OK", req); - return; - } else if ((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; -} - -/*! \brief 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 |= SIP_DEBUG_CONSOLE; - return RESULT_SUCCESS; -} - -/*! \brief sip_do_debug_peer: Turn on SIP debugging with peer mask */ -static int sip_do_debug_peer(int fd, int argc, char *argv[]) -{ - struct sip_peer *peer; - 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 |= SIP_DEBUG_CONSOLE; - } 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; -} - -/*! \brief sip_do_debug: Turn on SIP debugging (CLI command) */ -static int sip_do_debug(int fd, int argc, char *argv[]) -{ - int oldsipdebug = sipdebug & SIP_DEBUG_CONSOLE; - 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 |= SIP_DEBUG_CONSOLE; - 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; -} - -/*! \brief 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); - - 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; -} -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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 &= ~SIP_DEBUG_CONSOLE; - 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); - -/*! \brief 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 (recordhistory) { - char tmp[80]; - snprintf(tmp, sizeof(tmp), "Try: %d", p->authtries); - append_history(p, "RegistryAuth", tmp); - } - if (sip_debug_test_pvt(p) && p->registry) - ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname); - return transmit_register(p->registry, SIP_REGISTER, digest, respheader); -} - -/*! \brief 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]; - - if (!p->options) { - p->options = calloc(1, sizeof(*p->options)); - if (!p->options) { - ast_log(LOG_ERROR, "Out of memory\n"); - return -2; - } - } - - p->authtries++; - if (option_debug > 1) - ast_log(LOG_DEBUG, "Auth attempt %d on %s\n", p->authtries, sip_methods[sipmethod].text); - memset(digest, 0, sizeof(digest)); - if (reply_digest(p, req, header, sipmethod, digest, sizeof(digest) )) { - /* No way to authenticate */ - return -1; - } - /* Now we have a reply digest */ - p->options->auth = digest; - p->options->authheader = respheader; - return transmit_invite(p, sipmethod, sipmethod == SIP_INVITE, init); -} - -/*! \brief reply_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. */ -/* Returns -1 if we have no auth */ -static int reply_digest(struct sip_pvt *p, struct sip_request *req, - char *header, int sipmethod, char *digest, int digest_len) -{ - char tmp[512]; - char *c; - char oldnonce[256]; - - /* table of recognised keywords, and places where they should be copied */ - const struct x { - const char *key; - 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 */ - ast_copy_string(oldnonce, p->nonce, sizeof(oldnonce)); - while (c && *(c = ast_skip_blanks(c))) { /* lookup for keys */ - for (i = keys; i->key != NULL; i++) { - char *src, *separator; - if (strncasecmp(c, i->key, strlen(i->key)) != 0) - continue; - /* Found. Skip keyword, take text in quotes or up to the separator. */ - c += strlen(i->key); - if (*c == '\"') { - src = ++c; - separator = "\""; - } else { - src = c; - separator = ","; - } - strsep(&c, separator); /* clear separator and move ptr */ - ast_copy_string(i->dst, src, i->dstlen); - break; - } - if (i->key == NULL) /* not found, try ',' */ - strsep(&c, ","); - } - /* Reset nonce count */ - if (strcmp(p->nonce, oldnonce)) - p->noncecount = 0; - - /* Save auth data for following registrations */ - if (p->registry) { - struct sip_registry *r = p->registry; - - if (strcmp(r->nonce, p->nonce)) { - ast_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)); - r->noncecount = 0; - } - } - return build_reply_digest(p, sipmethod, digest, digest_len); -} - -/*! \brief 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 */ -/* Returns -1 if we have no auth */ -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", thread_safe_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; - if (sipdebug) - ast_log(LOG_DEBUG,"Using realm %s authentication for call %s\n", p->realm, p->callid); - } else { - /* No authentication, use peer or register= config */ - username = p->authname; - secret = p->peersecret; - md5secret = p->peermd5secret; - } - if (ast_strlen_zero(username)) /* We have no authentication */ - return -1; - - - /* Calculate SIP digest response */ - snprintf(a1,sizeof(a1),"%s:%s:%s", username, p->realm, secret); - snprintf(a2,sizeof(a2),"%s:%s", sip_methods[method].text, uri); - if (!ast_strlen_zero(md5secret)) - ast_copy_string(a1_hash, md5secret, sizeof(a1_hash)); - else - ast_md5_hash(a1_hash,a1); - ast_md5_hash(a2_hash,a2); - - p->noncecount++; - if (!ast_strlen_zero(p->qop)) - snprintf(resp,sizeof(resp),"%s:%s:%08x:%s:%s:%s", a1_hash, p->nonce, p->noncecount, cnonce, "auth", a2_hash); - else - snprintf(resp,sizeof(resp),"%s:%s:%s", a1_hash, p->nonce, a2_hash); - ast_md5_hash(resp_hash, resp); - /* 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=auth, cnonce=\"%s\", nc=%08x", username, p->realm, uri, p->nonce, resp_hash, p->opaque, cnonce, p->noncecount); - 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 show_domains_usage[] = -"Usage: sip show domains\n" -" Lists all configured SIP local domains.\n" -" Asterisk only responds to SIP messages to local domains.\n"; - -static char notify_usage[] = -"Usage: sip notify <type> <peer> [<peer>...]\n" -" Send a NOTIFY message to a SIP peer or peers\n" -" Message types are defined in sip_notify.conf\n"; - -static char show_users_usage[] = -"Usage: sip show users [like <pattern>]\n" -" Lists all known SIP users.\n" -" Optional regular expression pattern is used to filter the user list.\n"; - -static char show_user_usage[] = -"Usage: sip show user <name> [load]\n" -" 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 <channel>\n" -" Provides detailed status on a given SIP channel.\n"; - -static char show_history_usage[] = -"Usage: sip show history <channel>\n" -" Provides detailed dialog history on a given SIP channel.\n"; - -static char show_peers_usage[] = -"Usage: sip show peers [like <pattern>]\n" -" Lists all known SIP peers.\n" -" Optional regular expression pattern is used to filter the peer list.\n"; - -static char show_peer_usage[] = -"Usage: sip show peer <name> [load]\n" -" 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] [<name>|all|like <pattern>]\n" -" Prunes object(s) from the cache.\n" -" Optional regular expression pattern is used to filter the objects.\n"; - -static char show_reg_usage[] = -"Usage: sip show registry\n" -" Lists all registration requests and status.\n"; - -static char debug_usage[] = -"Usage: sip debug\n" -" Enables dumping of SIP packets for debugging purposes\n\n" -" sip debug ip <host[:PORT]>\n" -" Enables dumping of SIP packets to and from host.\n\n" -" sip debug peer <peername>\n" -" Enables dumping of SIP packets to and from host.\n" -" Require peer to be registered.\n"; - -static char no_debug_usage[] = -"Usage: sip 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"; - -static char show_settings_usage[] = -"Usage: sip show settings\n" -" Provides detailed list of the configuration of the SIP channel.\n"; - - - -/*! \brief 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; - - /* If there is no private structure, this channel is no longer alive */ - if (!p) { - ast_mutex_unlock(&chan->lock); - return NULL; - } - - 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(<name>)", - .read = func_header_read, -}; - -/*! \brief function_check_sipdomain: Dial plan function to check if domain is local */ -static char *func_check_sipdomain(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) -{ - if (ast_strlen_zero(data)) { - ast_log(LOG_WARNING, "CHECKSIPDOMAIN requires an argument - A domain name\n"); - return buf; - } - if (check_sip_domain(data, NULL, 0)) - ast_copy_string(buf, data, len); - else - buf[0] = '\0'; - return buf; -} - -static struct ast_custom_function checksipdomain_function = { - .name = "CHECKSIPDOMAIN", - .synopsis = "Checks if domain is a local domain", - .syntax = "CHECKSIPDOMAIN(<domain|IP>)", - .read = func_check_sipdomain, - .desc = "This function checks if the domain in the argument is configured\n" - "as a local SIP domain that this Asterisk server is configured to handle.\n" - "Returns the domain name if it is locally handled, otherwise an empty string.\n" - "Check the domain= configuration in sip.conf\n", -}; - - -/*! \brief 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, "status")) { - peer_status(peer, buf, sizeof(buf)); - } else if (!strcasecmp(colname, "language")) { - ast_copy_string(buf, peer->language, len); - } else if (!strcasecmp(colname, "regexten")) { - ast_copy_string(buf, peer->regexten, len); - } else if (!strcasecmp(colname, "limit")) { - snprintf(buf, len, "%d", peer->call_limit); - } else if (!strcasecmp(colname, "curcalls")) { - snprintf(buf, len, "%d", peer->inUse); - } else if (!strcasecmp(colname, "useragent")) { - ast_copy_string(buf, peer->useragent, len); - } else if (!strcasecmp(colname, "mailbox")) { - ast_copy_string(buf, peer->mailbox, len); - } else if (!strcasecmp(colname, "context")) { - ast_copy_string(buf, peer->context, len); - } else if (!strcasecmp(colname, "expire")) { - snprintf(buf, len, "%d", peer->expire); - } else if (!strcasecmp(colname, "dynamic")) { - ast_copy_string(buf, (ast_test_flag(peer, 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; -} - -/* Structure to declare a dialplan function: SIPPEER */ -struct ast_custom_function sippeer_function = { - .name = "SIPPEER", - .synopsis = "Gets SIP peer information", - .syntax = "SIPPEER(<peername>[:item])", - .read = function_sippeer, - .desc = "Valid items are:\n" - "- ip (default) The IP address.\n" - "- mailbox The configured mailbox.\n" - "- context The configured context.\n" - "- expire The epoch time of the next expire.\n" - "- dynamic Is it dynamic? (yes/no).\n" - "- callerid_name The configured Caller ID name.\n" - "- callerid_num The configured Caller ID number.\n" - "- codecs The configured codecs.\n" - "- status Status (if qualify=yes).\n" - "- regexten Registration extension\n" - "- limit Call limit (call-limit)\n" - "- curcalls Current amount of calls \n" - " Only available if call-limit is set\n" - "- language Default language for peer\n" - "- useragent Current user agent id for peer\n" - "- codec[x] Preferred codec index number 'x' (beginning with zero).\n" - "\n" -}; - -/*! \brief function_sipchaninfo_read: ${SIPCHANINFO()} Dialplan function - reads sip channel data */ -static char *function_sipchaninfo_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) -{ - struct sip_pvt *p; - char iabuf[INET_ADDRSTRLEN]; - - *buf = 0; - - if (!data) { - ast_log(LOG_WARNING, "This function requires a parameter 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; - } - -/* ast_verbose("function_sipchaninfo_read: %s\n", data); */ - p = chan->tech_pvt; - - /* If there is no private structure, this channel is no longer alive */ - if (!p) { - ast_mutex_unlock(&chan->lock); - return NULL; - } - - if (!strcasecmp(data, "peerip")) { - ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr) : "", len); - } else if (!strcasecmp(data, "recvip")) { - ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr) : "", len); - } else if (!strcasecmp(data, "from")) { - ast_copy_string(buf, p->from, len); - } else if (!strcasecmp(data, "uri")) { - ast_copy_string(buf, p->uri, len); - } else if (!strcasecmp(data, "useragent")) { - ast_copy_string(buf, p->useragent, len); - } else if (!strcasecmp(data, "peername")) { - ast_copy_string(buf, p->peername, len); - } else { - ast_mutex_unlock(&chan->lock); - return NULL; - } - ast_mutex_unlock(&chan->lock); - - return buf; -} - -/* Structure to declare a dialplan function: SIPCHANINFO */ -static struct ast_custom_function sipchaninfo_function = { - .name = "SIPCHANINFO", - .synopsis = "Gets the specified SIP parameter from the current channel", - .syntax = "SIPCHANINFO(item)", - .read = function_sipchaninfo_read, - .desc = "Valid items are:\n" - "- peerip The IP address of the peer.\n" - "- recvip The source IP address of the peer.\n" - "- from The URI from the From: header.\n" - "- uri The URI from the Contact: header.\n" - "- useragent The useragent.\n" - "- peername The name of the peer.\n" -}; - - - -/*! \brief 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 = get_in_brackets(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)); - } -} - -/*! \brief 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); - } -} - -/*! \brief handle_response_invite: Handle SIP response in dialogue ---*/ -static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno) -{ - int outgoing = ast_test_flag(p, SIP_OUTGOING); - - if (option_debug > 3) { - int reinvite = (p->owner && p->owner->_state == AST_STATE_UP); - if (reinvite) - ast_log(LOG_DEBUG, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid); - else - ast_log(LOG_DEBUG, "SIP response %d to standard invite\n", resp); - } - - if (ast_test_flag(p, SIP_ALREADYGONE)) { /* This call is already gone */ - ast_log(LOG_DEBUG, "Got response on call that is already terminated: %s (ignoring)\n", p->callid); - return; - } - - switch (resp) { - case 100: /* Trying */ - sip_cancel_destroy(p); - break; - case 180: /* 180 Ringing */ - sip_cancel_destroy(p); - if (!ignore && p->owner) { - ast_queue_control(p->owner, AST_CONTROL_RINGING); - if (p->owner->_state != AST_STATE_UP) - ast_setstate(p->owner, AST_STATE_RINGING); - } - if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) { - process_sdp(p, req); - if (!ignore && p->owner) { - /* Queue a progress frame only if we have SDP in 180 */ - ast_queue_control(p->owner, AST_CONTROL_PROGRESS); - } - } - break; - case 183: /* Session progress */ - sip_cancel_destroy(p); - if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) { - process_sdp(p, req); - } - if (!ignore && p->owner) { - /* Queue a progress frame */ - ast_queue_control(p->owner, AST_CONTROL_PROGRESS); - } - break; - case 200: /* 200 OK on invite - someone's answering our call */ - sip_cancel_destroy(p); - p->authtries = 0; - if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) { - process_sdp(p, req); -#ifdef SIP_MIDCOM - if (m_cb) { - if (!m_cb->handle_response_invite_hook(p)) { - if (p->owner) - ast_queue_hangup(p->owner); - else - ast_set_flag(p, SIP_NEEDDESTROY); - } - } -#endif - } - - /* Parse contact header for continued conversation */ - /* When we get 200 OK, we know which device (and IP) to contact for this call */ - /* This is important when we have a SIP proxy between us and the phone */ - if (outgoing) { - parse_ok_contact(p, req); - - /* Save Record-Route for any later requests we make on this dialogue */ - build_route(p, req, 1); - } - - if (!ignore && 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 { /* RE-invite */ - 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 */ - /* THIS NEEDS TO BE CHECKED: OEJ */ - if (!ignore) - ast_set_flag(p, SIP_PENDINGBYE); - } - /* 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); - break; - case 407: /* Proxy authentication */ - case 401: /* Www auth */ - /* First we ACK */ - transmit_request(p, SIP_ACK, seqno, 0, 0); - if (p->options) - p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH); - - /* Then we AUTH */ - p->theirtag[0]='\0'; /* forget their old tag, so we don't match tags when getting response */ - if (!ignore) { - char *authenticate = (resp == 401 ? "WWW-Authenticate" : "Proxy-Authenticate"); - char *authorization = (resp == 401 ? "Authorization" : "Proxy-Authorization"); - if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, authenticate, authorization, SIP_INVITE, 1)) { - ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From")); - ast_set_flag(p, SIP_NEEDDESTROY); - ast_set_flag(p, SIP_ALREADYGONE); - if (p->owner) - ast_queue_control(p->owner, AST_CONTROL_CONGESTION); - } - } - break; - case 403: /* Forbidden */ - /* 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 (!ignore && p->owner) - ast_queue_control(p->owner, AST_CONTROL_CONGESTION); - ast_set_flag(p, SIP_NEEDDESTROY); - ast_set_flag(p, SIP_ALREADYGONE); - break; - case 404: /* Not found */ - transmit_request(p, SIP_ACK, seqno, 0, 0); - if (p->owner && !ignore) - ast_queue_control(p->owner, AST_CONTROL_CONGESTION); - ast_set_flag(p, SIP_ALREADYGONE); - break; - case 481: /* Call leg does not exist */ - /* Could be REFER or INVITE */ - ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid); - transmit_request(p, SIP_ACK, seqno, 0, 0); - break; - case 491: /* Pending */ - /* we have to wait a while, then retransmit */ - /* Transmission is rescheduled, so everything should be taken care of. - We should support the retry-after at some point */ - break; - case 501: /* Not implemented */ - if (p->owner) - ast_queue_control(p->owner, AST_CONTROL_CONGESTION); - break; - } -} - -/*! \brief 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 == MAX_AUTHTRIES) || do_register_auth(p, req, "WWW-Authenticate", "Authorization")) { - ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s@%s' (Tries %d)\n", p->registry->username, p->registry->hostname, p->authtries); - ast_set_flag(p, SIP_NEEDDESTROY); - } - break; - case 403: /* Forbidden */ - ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for REGISTER for '%s' to '%s'\n", p->registry->username, p->registry->hostname); - if (global_regattempts_max) - p->registry->regattempts = global_regattempts_max+1; - ast_sched_del(sched, r->timeout); - ast_set_flag(p, SIP_NEEDDESTROY); - break; - case 404: /* Not found */ - ast_log(LOG_WARNING, "Got 404 Not found on SIP register to service %s@%s, giving up\n", p->registry->username,p->registry->hostname); - if (global_regattempts_max) - p->registry->regattempts = global_regattempts_max+1; - ast_set_flag(p, SIP_NEEDDESTROY); - r->call = NULL; - ast_sched_del(sched, r->timeout); - break; - case 407: /* Proxy auth */ - if ((p->authtries == MAX_AUTHTRIES) || do_register_auth(p, req, "Proxy-Authenticate", "Proxy-Authorization")) { - ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s' (tries '%d')\n", get_header(&p->initreq, "From"), p->authtries); - ast_set_flag(p, 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); - if (global_regattempts_max) - 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 = strcasestr(contact, "expires="); - if (tmptmp) { - if (sscanf(tmptmp + 8, "%d;", &expires) != 1) - expires = 0; - } - - } - if (!expires) - expires=atoi(get_header(req, "expires")); - if (!expires) - expires=default_expiry; - - expires_ms = expires * 1000; - if (expires <= EXPIRY_GUARD_LIMIT) - expires_ms -= MAX((expires_ms * EXPIRY_GUARD_PCT),EXPIRY_GUARD_MIN); - else - expires_ms -= EXPIRY_GUARD_SECS * 1000; - if (sipdebug) - ast_log(LOG_NOTICE, "Outbound Registration: Expiry for %s is %d sec (Scheduling reregistration in %d s)\n", r->hostname, expires, expires_ms/1000); - - r->refresh= (int) expires_ms / 1000; - - /* Schedule re-registration before we expire */ - r->expire=ast_sched_add(sched, expires_ms, sip_reregister, r); - ASTOBJ_UNREF(r, sip_registry_destroy); - } - return 1; -} - -/*! \brief 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; -} - -/*! \brief 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 *msg, *c; - struct ast_channel *owner; - char iabuf[INET_ADDRSTRLEN]; - int sipmethod; - int res = 1; - - c = get_header(req, "Cseq"); - msg = strchr(c, ' '); - if (!msg) - msg = ""; - else - msg++; - sipmethod = find_sip_method(msg); - - owner = p->owner; - if (owner) - owner->hangupcause = hangup_sip2cause(resp); - - /* Acknowledge whatever it is destined for */ - if ((resp >= 100) && (resp <= 199)) - __sip_semi_ack(p, seqno, 0, sipmethod); - else - __sip_ack(p, seqno, 0, sipmethod); - - /* Get their tag if we haven't already */ - if (ast_strlen_zero(p->theirtag) || (resp >= 200)) { - gettag(req, "To", p->theirtag, sizeof(p->theirtag)); - } - 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) - handle_response_invite(p, resp, rest, req, ignore, seqno); - break; - case 183: /* 183 Session Progress */ - if (sipmethod == SIP_INVITE) - handle_response_invite(p, resp, rest, req, ignore, seqno); - break; - case 180: /* 180 Ringing */ - if (sipmethod == SIP_INVITE) - handle_response_invite(p, resp, rest, req, ignore, seqno); - break; - case 200: /* 200 OK */ - p->authtries = 0; /* Reset authentication counter */ - if (sipmethod == SIP_MESSAGE) { - /* We successfully transmitted a message */ - ast_set_flag(p, SIP_NEEDDESTROY); - } else 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 == NONE) { - ast_set_flag(p, SIP_NEEDDESTROY); - } - } - } else if (sipmethod == SIP_INVITE) { - handle_response_invite(p, resp, rest, req, ignore, seqno); - } 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) { - handle_response_invite(p, resp, rest, req, ignore, seqno); - } 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) { - handle_response_invite(p, resp, rest, req, ignore, seqno); - } 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 (sipmethod == SIP_INVITE) { - handle_response_invite(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) { - handle_response_invite(p, resp, rest, req, ignore, seqno); - } 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 == MAX_AUTHTRIES) || do_proxy_auth(p, req, "Proxy-Authenticate", "Proxy-Authorization", sipmethod, 0)) { - ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From")); - ast_set_flag(p, SIP_NEEDDESTROY); - } - } else if (p->registry && sipmethod == SIP_REGISTER) { - res = handle_response_register(p, resp, rest, req, ignore, seqno); - } else /* We can't handle this, giving up in a bad way */ - ast_set_flag(p, SIP_NEEDDESTROY); - - break; - case 491: /* Pending */ - if (sipmethod == SIP_INVITE) { - handle_response_invite(p, resp, rest, req, ignore, seqno); - } - case 501: /* Not Implemented */ - if (sipmethod == SIP_INVITE) { - handle_response_invite(p, resp, rest, req, ignore, seqno); - } 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); - /* 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 487: - /* channel now destroyed - dec the inUse counter */ - update_call_counter(p, DEC_CALL_LIMIT); - 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 488: /* Not acceptable here - codec error */ - 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 { - /* Responses to OUTGOING SIP requests on INCOMING calls - get handled here. As well as out-of-call message responses */ - if (req->debug) - ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg); - if (resp == 200) { - /* Tags in early session is replaced by the tag in 200 OK, which is - the final reply to our INVITE */ - gettag(req, "To", p->theirtag, sizeof(p->theirtag)); - } - - switch(resp) { - case 200: - if (sipmethod == SIP_INVITE) { - handle_response_invite(p, resp, rest, req, ignore, seqno); - } else if (sipmethod == SIP_CANCEL) { - ast_log(LOG_DEBUG, "Got 200 OK on CANCEL\n"); - } else if (sipmethod == SIP_MESSAGE) - /* We successfully transmitted a message */ - ast_set_flag(p, SIP_NEEDDESTROY); - break; - case 401: /* www-auth */ - case 407: - if (sipmethod == SIP_BYE || sipmethod == SIP_REFER) { - char *auth, *auth2; - - if (resp == 407) { - auth = "Proxy-Authenticate"; - auth2 = "Proxy-Authorization"; - } else { - auth = "WWW-Authenticate"; - auth2 = "Authorization"; - } - if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, auth, auth2, sipmethod, 0)) { - ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From")); - ast_set_flag(p, SIP_NEEDDESTROY); - } - } else if (sipmethod == SIP_INVITE) { - handle_response_invite(p, resp, rest, req, ignore, seqno); - } - break; - case 481: /* Call leg does not exist */ - if (sipmethod == SIP_INVITE) { - /* Re-invite failed */ - handle_response_invite(p, resp, rest, req, ignore, seqno); - } - break; - default: /* Errors without handlers */ - if ((resp >= 100) && (resp < 200)) { - if (sipmethod == SIP_INVITE) { /* re-invite */ - sip_cancel_destroy(p); - } - } - if ((resp >= 300) && (resp < 700)) { - if ((option_verbose > 2) && (resp != 487)) - ast_verbose(VERBOSE_PREFIX_3 "Incoming call: Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr)); - switch(resp) { - case 488: /* Not acceptable here - codec error */ - case 603: /* Decline */ - case 500: /* Server error */ - case 503: /* Service Unavailable */ - - if (sipmethod == SIP_INVITE) { /* re-invite failed */ - sip_cancel_destroy(p); - } - break; - } - } - break; - } - } -} - -struct sip_dual { - struct ast_channel *chan1; - struct ast_channel *chan2; - struct sip_request req; -}; - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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); - } -} - -/*! \brief 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; -} - -/*! \brief gettag: Get tag from packet */ -static char *gettag(struct sip_request *req, char *header, char *tagbuf, int tagbufsize) -{ - - char *thetag, *sep; - - - if (!tagbuf) - return NULL; - tagbuf[0] = '\0'; /* reset the buffer */ - thetag = get_header(req, header); - thetag = strcasestr(thetag, ";tag="); - if (thetag) { - thetag += 5; - ast_copy_string(tagbuf, thetag, tagbufsize); - sep = strchr(tagbuf, ';'); - if (sep) - *sep = '\0'; - } - return thetag; -} - -/*! \brief 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; -} - -/*! \brief 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 (!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); - if (!p->lastinvite) - 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); - if (!p->lastinvite) - 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 INVITE 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); - p->theirtag[0] = '\0'; /* Forget their to-tag, we'll get a new one */ - } - 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; - } -#ifdef SIP_MIDCOM - if (m_cb) { - if (!m_cb->handle_request_invite_hook((void *)p)) { - ast_log(LOG_NOTICE, "Failed to NAT for (%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; - } - } -#endif - } 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_call_counter(p, INC_CALL_LIMIT); - 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_call_counter(p, DEC_CALL_LIMIT); - } else { - if (ignore) - transmit_response(p, "484 Address Incomplete", req); - else - transmit_response_reliable(p, "484 Address Incomplete", req, 1); - update_call_counter(p, DEC_CALL_LIMIT); - } - 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 */ - make_our_tag(p->tag, sizeof(p->tag)); - /* 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) { -#ifdef OSP_SUPPORT - ast_channel_setwhentohangup (c, p->osptimelimit); -#endif - 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())) { - enum ast_pbx_result res; - - res = ast_pbx_start(c); - - switch (res) { - case AST_PBX_FAILED: - ast_log(LOG_WARNING, "Failed to start PBX :(\n"); - if (ignore) - transmit_response(p, "503 Unavailable", req); - else - transmit_response_reliable(p, "503 Unavailable", req, 1); - break; - case AST_PBX_CALL_LIMIT: - ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n"); - if (ignore) - transmit_response(p, "480 Temporarily Unavailable", req); - else - transmit_response_reliable(p, "480 Temporarily Unavailable", req, 1); - break; - case AST_PBX_SUCCESS: - /* nothing to do */ - break; - } - - if (res) { - 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); - 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) && !ignore) { - 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; -} - -/*! \brief 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; -} -/*! \brief 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; - } -} - -/*! \brief handle_request_bye: Handle incoming BYE request ---*/ -static int handle_request_bye(struct sip_pvt *p, struct sip_request *req, int debug, int ignore) -{ - struct ast_channel *c=NULL; - int res; - struct ast_channel *bridged_to; - char iabuf[INET_ADDRSTRLEN]; - - if (p->pendinginvite && !ast_test_flag(p, SIP_OUTGOING) && !ignore) - transmit_response_reliable(p, "487 Request Terminated", &p->initreq, 1); - - 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)); - if (p->owner) - 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; -} - -/*! \brief 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 (!ignore) { - if (debug) - ast_verbose("Receiving message!\n"); - receive_message(p, req); - } else { - transmit_response(p, "202 Accepted", req); - } - return 1; -} -/*! \brief 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; - int firststate = AST_EXTENSION_REMOVED; - - if (p->initreq.headers) { - /* We already have a dialog */ - if (p->initreq.method != SIP_SUBSCRIBE) { - /* This is a SUBSCRIBE within another SIP dialog, which we do not support */ - /* For transfers, this could happen, but since we haven't seen it happening, let us just refuse this */ - transmit_response(p, "403 Forbidden (within dialog)", req); - /* Do not destroy session, since we will break the call if we do */ - ast_log(LOG_DEBUG, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text); - return 0; - } else { - if (debug) - ast_log(LOG_DEBUG, "Got a re-subscribe on existing subscription %s\n", p->callid); - } - } - if (!ignore && !p->initreq.headers) { - /* 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 && ignore) - ast_verbose("Ignoring this SUBSCRIBE request\n"); - - if (!p->lastinvite) { - char mailboxbuf[256]=""; - int found = 0; - char *mailbox = NULL; - int mailboxsize = 0; - - char *event = get_header(req, "Event"); /* Get Event package name */ - char *accept = get_header(req, "Accept"); - - if (!strcmp(event, "message-summary") && !strcmp(accept, "application/simple-message-summary")) { - mailbox = mailboxbuf; - mailboxsize = sizeof(mailboxbuf); - } - /* Handle authentication if this is our first subscribe */ - res = check_user_full(p, req, SIP_SUBSCRIBE, e, 0, sin, ignore, mailbox, mailboxsize); - 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->subscribecontext)) - ast_copy_string(p->context, p->subscribecontext, sizeof(p->context)); - else 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); /* Overlap dialing on SUBSCRIBE?? */ - ast_set_flag(p, SIP_NEEDDESTROY); - } else { - - /* Initialize tag for new subscriptions */ - if (ast_strlen_zero(p->tag)) - make_our_tag(p->tag, sizeof(p->tag)); - - if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */ - - /* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */ - if (strstr(accept, "application/pidf+xml")) { - p->subscribed = PIDF_XML; /* RFC 3863 format */ - } else if (strstr(accept, "application/dialog-info+xml")) { - p->subscribed = DIALOG_INFO_XML; - /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */ - } else if (strstr(accept, "application/cpim-pidf+xml")) { - p->subscribed = CPIM_PIDF_XML; /* RFC 3863 format */ - } else if (strstr(accept, "application/xpidf+xml")) { - p->subscribed = XPIDF_XML; /* Early pre-RFC 3863 format with MSN additions (Microsoft Messenger) */ - } else if (strstr(p->useragent, "Polycom")) { - p->subscribed = XPIDF_XML; /* Polycoms subscribe for "event: dialog" but don't include an "accept:" header */ - } else { - /* Can't find a format for events that we know about */ - transmit_response(p, "489 Bad Event", req); - ast_set_flag(p, SIP_NEEDDESTROY); - return 0; - } - } else if (!strcmp(event, "message-summary") && !strcmp(accept, "application/simple-message-summary")) { - /* Looks like they actually want a mailbox status */ - - /* 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, "404 Not found", req); - ast_set_flag(p, SIP_NEEDDESTROY); - } - return 0; - } else { /* At this point, Asterisk does not understand the specified event */ - transmit_response(p, "489 Bad Event", req); - if (option_debug > 1) - ast_log(LOG_DEBUG, "Received SIP subscribe for unknown event package: %s\n", event); - ast_set_flag(p, SIP_NEEDDESTROY); - return 0; - } - if (p->subscribed != NONE) - p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, p); - } - } - - if (!ignore && p) - p->lastinvite = seqno; - if (p && !ast_test_flag(p, SIP_NEEDDESTROY)) { - p->expiry = atoi(get_header(req, "Expires")); - - /* The next 4 lines can be removed if the SNOM Expires bug is fixed */ - if (p->subscribed == DIALOG_INFO_XML) { - if (p->expiry > max_expiry) - p->expiry = max_expiry; - } - if (sipdebug || option_debug > 1) - ast_log(LOG_DEBUG, "Adding subscription for extension %s context %s for peer %s\n", p->exten, p->context, p->username); - if (p->autokillid > -1) - sip_cancel_destroy(p); /* Remove subscription expiry for renewals */ - sip_scheddestroy(p, (p->expiry + 10) * 1000); /* Set timer for destruction of call at expiration */ - - if ((firststate = ast_extension_state(NULL, p->context, p->exten)) < 0) { - ast_log(LOG_ERROR, "Got SUBSCRIBE for extensions without hint. Please add hint to %s in context %s\n", p->exten, p->context); - transmit_response(p, "404 Not found", req); - ast_set_flag(p, SIP_NEEDDESTROY); - return 0; - } else { - struct sip_pvt *p_old; - - transmit_response(p, "200 OK", req); - transmit_state_notify(p, firststate, 1, 1); /* Send first notification */ - append_history(p, "Subscribestatus", ast_extension_state2str(firststate)); - - /* remove any old subscription from this peer for the same exten/context, - as the peer has obviously forgotten about it and it's wasteful to wait - for it to expire and send NOTIFY messages to the peer only to have them - ignored (or generate errors) - */ - ast_mutex_lock(&iflock); - for (p_old = iflist; p_old; p_old = p_old->next) { - if (p_old == p) - continue; - if (p_old->initreq.method != SIP_SUBSCRIBE) - continue; - if (p_old->subscribed == NONE) - continue; - ast_mutex_lock(&p_old->lock); - if (!strcmp(p_old->username, p->username)) { - if (!strcmp(p_old->exten, p->exten) && - !strcmp(p_old->context, p->context)) { - ast_set_flag(p_old, SIP_NEEDDESTROY); - ast_mutex_unlock(&p_old->lock); - break; - } - } - ast_mutex_unlock(&p_old->lock); - } - ast_mutex_unlock(&iflock); - } - if (!p->expiry) - ast_set_flag(p, SIP_NEEDDESTROY); - } - return 1; -} - -/*! \brief 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 REGISTER 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' - %s\n", get_header(req, "To"), ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr), (res == -1) ? "Wrong password" : (res == -2 ? "Username/auth name mismatch" : "Not a local SIP domain")); - if (res < 1) { - /* Destroy the session, but keep us around for just a bit in case they don't - get our 200 OK */ - sip_scheddestroy(p, 15*1000); - } - return res; -} - -/*! \brief 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 *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; - int error = 0; - - /* 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)) { - ast_log(LOG_ERROR, "Missing Cseq. Dropping this SIP message, it's incomplete.\n"); - error = 1; - } - if (!error && sscanf(cseq, "%d%n", &seqno, &len) != 1) { - ast_log(LOG_ERROR, "No seqno in '%s'. Dropping incomplete message.\n", cmd); - error = 1; - } - if (error) { - if (!p->initreq.header) /* New call */ - ast_set_flag(p, SIP_NEEDDESTROY); /* Make sure we destroy this dialog */ - return -1; - } - /* Get the command XXX */ - - cmd = req->rlPart1; - e = req->rlPart2; - - /* Save useragent of the client */ - useragent = get_header(req, "User-Agent"); - if (!ast_strlen_zero(useragent)) - ast_copy_string(p->useragent, useragent, sizeof(p->useragent)); - - /* Find out SIP method for incoming request */ - if (req->method == SIP_RESPONSE) { /* Response to our request */ - /* Response to our request -- Do some sanity checks */ - if (!p->initreq.headers) { - 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; - } - - e = ast_skip_blanks(e); - if (sscanf(e, "%d %n", &respid, &len) != 1) { - ast_log(LOG_WARNING, "Invalid response: '%s'\n", e); - } else { - /* More SIP ridiculousness, we have to ignore bogus contacts in 100 etc responses */ - if ((respid == 200) || ((respid >= 300) && (respid <= 399))) - extract_uri(p, req); - handle_response(p, respid, e + len, req, ignore, seqno); - } - return 0; - } - - /* New SIP request coming in - (could be new request in existing SIP dialog as well...) - */ - - p->method = req->method; /* Find out which SIP method they are using */ - if (option_debug > 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)) { - if (option_debug) - ast_log(LOG_DEBUG, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq); - if (req->method != SIP_ACK) - transmit_response(p, "503 Server error", req); /* We must respond according to RFC 3261 sec 12.2 */ - return -1; - } else if (p->icseq && (p->icseq == seqno) && req->method != SIP_ACK &&(p->method != SIP_CANCEL|| ast_test_flag(p, SIP_ALREADYGONE))) { - /* ignore means "don't do anything with it" but still have to - respond appropriately. We do this if we receive a repeat of - the last sequence number */ - ignore=2; - if (option_debug > 2) - ast_log(LOG_DEBUG, "Ignoring SIP message because of retransmit (%s Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno); - } - - if (seqno >= p->icseq) - /* Next should follow monotonically (but not necessarily - incrementally -- thanks again to the genius authors of SIP -- - increasing */ - p->icseq = seqno; - - /* Find their tag if we haven't got it */ - if (ast_strlen_zero(p->theirtag)) { - gettag(req, "From", p->theirtag, sizeof(p->theirtag)); - } - snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd); - - if (pedanticsipchecking) { - /* If this is a request packet without a from tag, it's not - correct according to RFC 3261 */ - /* Check if this a new request in a new dialog with a totag already attached to it, - RFC 3261 - section 12.2 - and we don't want to mess with recovery */ - if (!p->initreq.headers && ast_test_flag(req, SIP_PKT_WITH_TOTAG)) { - /* If this is a first request and it got a to-tag, it is not for us */ - if (!ignore && req->method == SIP_INVITE) { - transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req, 1); - /* Will cease to exist after ACK */ - } else { - transmit_response(p, "481 Call/Transaction Does Not Exist", req); - ast_set_flag(p, SIP_NEEDDESTROY); - } - return res; - } - } - - /* 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, ignore); - 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 INFO!\n"); - handle_request_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, 0); - 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; -} - -/*! \brief 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; - 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; - } - if (res == sizeof(req.data)) { - ast_log(LOG_DEBUG, "Received packet exceeds buffer. Data is possibly lost\n"); - } - req.data[res] = '\0'; - req.len = res; - if(sip_debug_test_addr(&sin)) - ast_set_flag(&req, SIP_PKT_DEBUG); - if (pedanticsipchecking) - req.len = lws2sws(req.data, req.len); /* Fix multiline headers */ - if (ast_test_flag(&req, SIP_PKT_DEBUG)) { - ast_verbose("\n<-- SIP read from %s:%d: \n%s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), req.data); - } - parse_request(&req); - req.method = find_sip_method(req.rlPart1); - if (ast_test_flag(&req, SIP_PKT_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; - } - - - /* Process request, with netlock held */ -retrylock: - ast_mutex_lock(&netlock); - p = find_call(&req, &sin, req.method); - 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; - if (handle_request(p, &req, &sin, &recount, &nounlock) == -1) { - /* Request failed */ - ast_log(LOG_DEBUG, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>"); - } - - if (p->owner && !nounlock) - ast_mutex_unlock(&p->owner->lock); - ast_mutex_unlock(&p->lock); - } - ast_mutex_unlock(&netlock); - if (recount) - ast_update_use_count(); - - return 1; -} - -/*! \brief 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, peer->vmexten); - sip_scheddestroy(p, 15000); - return 0; -} - -/*! \brief 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); - if (res > 20) - ast_log(LOG_DEBUG, "chan_sip: ast_io_wait ran %d all at once\n", res); - ast_mutex_lock(&monlock); - if (res >= 0) { - res = ast_sched_runq(sched); - if (res >= 20) - ast_log(LOG_DEBUG, "chan_sip: ast_sched_runq ran %d all at once\n", res); - } - - /* 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; - -} - -/*! \brief restart_monitor: Start the channel monitor thread ---*/ -static int restart_monitor(void) -{ - /* If we're supposed to be stopped -- stay stopped */ - if (monitor_thread == AST_PTHREADT_STOP) - return 0; - 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 { - /* Start a new monitor */ - if (ast_pthread_create(&monitor_thread, NULL, do_monitor, NULL) < 0) { - ast_mutex_unlock(&monlock); - ast_log(LOG_ERROR, "Unable to start monitor thread.\n"); - return -1; - } - } - ast_mutex_unlock(&monlock); - return 0; -} - -/*! \brief 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; -} - -/*! \brief 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) { - if (sipdebug) - ast_log(LOG_NOTICE, "Still have a QUALIFY dialog active, deleting\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 dialog 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, 2); -#else - transmit_invite(p, SIP_OPTIONS, 0, 2); -#endif - gettimeofday(&peer->ps, NULL); - peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2, sip_poke_noanswer, peer); - - return 0; -} - -/*! \brief sip_devicestate: Part of PBX channel interface ---*/ - -/* Return values:--- - If we have qualify on and the device is not reachable, regardless of registration - state we return AST_DEVICE_UNAVAILABLE - - For peers with call limit: - not registered AST_DEVICE_UNAVAILABLE - registered, no call AST_DEVICE_NOT_INUSE - registered, calls possible AST_DEVICE_INUSE - registered, call limit reached AST_DEVICE_BUSY - For peers without call limit: - not registered AST_DEVICE_UNAVAILABLE - registered AST_DEVICE_UNKNOWN -*/ -static int sip_devicestate(void *data) -{ - char *host; - char *tmp; - - struct hostent *hp; - struct ast_hostent ahp; - struct sip_peer *p; - - int res = AST_DEVICE_INVALID; - - host = ast_strdupa(data); - if ((tmp = strchr(host, '@'))) - host = tmp + 1; - - if (option_debug > 2) - ast_log(LOG_DEBUG, "Checking device state for peer %s\n", host); - - if ((p = find_peer(host, NULL, 1))) { - if (p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) { - /* we have an address for the peer */ - /* if qualify is turned on, check the status */ - if (p->maxms && (p->lastms > p->maxms)) { - res = AST_DEVICE_UNAVAILABLE; - } else { - /* qualify is not on, or the peer is responding properly */ - /* check call limit */ - if (p->call_limit && (p->inUse == p->call_limit)) - res = AST_DEVICE_BUSY; - else if (p->call_limit && p->inUse) - res = AST_DEVICE_INUSE; - else if (p->call_limit) - res = AST_DEVICE_NOT_INUSE; - else - res = AST_DEVICE_UNKNOWN; - } - } else { - /* there is no address, it's unavailable */ - res = AST_DEVICE_UNAVAILABLE; - } - ASTOBJ_UNREF(p,sip_destroy_peer); - } else { - hp = ast_gethostbyname(host, &ahp); - if (hp) - res = AST_DEVICE_UNKNOWN; - } - - return res; -} - -/*! \brief sip_request: PBX interface function -build SIP pvt structure ---*/ -/* SIP calls initiated by the PBX arrive here */ -static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause) -{ - int oldformat; - struct sip_pvt *p; - struct ast_channel *tmpc = NULL; - char *ext, *host; - char tmp[256]; - char *dest = data; - - oldformat = format; - 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; - } - - p->options = calloc(1, sizeof(*p->options)); - if (!p->options) { - ast_log(LOG_ERROR, "Out of memory\n"); - 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; - } - } - - 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 registered 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 : "<none>", 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; -} - -/*! \brief 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, "sendrpid")) { - ast_set_flag(mask, SIP_SENDRPID); - ast_set2_flag(flags, ast_true(v->value), SIP_SENDRPID); - 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 if (!strcasecmp(v->value, "auto")) - ast_set_flag(flags, SIP_DTMF_AUTO); - else { - ast_log(LOG_WARNING, "Unknown dtmf mode '%s' on line %d, using rfc2833\n", v->value, v->lineno); - ast_set_flag(flags, 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 (ast_true(v->value)) - ast_set_flag(flags, SIP_PROG_INBAND_YES); - else if (strcasecmp(v->value, "never")) - ast_set_flag(flags, SIP_PROG_INBAND_NO); - } 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, "proxy")) - ast_set_flag(flags, SIP_OSPAUTH_PROXY); - else if (!strcasecmp(v->value, "gateway")) - ast_set_flag(flags, SIP_OSPAUTH_GATEWAY); - else if(!strcasecmp (v->value, "exclusive")) - ast_set_flag(flags, SIP_OSPAUTH_EXCLUSIVE); -#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; -} - -/*! \brief add_sip_domain: Add SIP domain to list of domains we are responsible for */ -static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context) -{ - struct domain *d; - - if (ast_strlen_zero(domain)) { - ast_log(LOG_WARNING, "Zero length domain.\n"); - return 1; - } - - d = calloc(1, sizeof(*d)); - if (!d) { - ast_log(LOG_ERROR, "Allocation of domain structure failed, Out of memory\n"); - return 0; - } - - ast_copy_string(d->domain, domain, sizeof(d->domain)); - - if (!ast_strlen_zero(context)) - ast_copy_string(d->context, context, sizeof(d->context)); - - d->mode = mode; - - AST_LIST_LOCK(&domain_list); - AST_LIST_INSERT_TAIL(&domain_list, d, list); - AST_LIST_UNLOCK(&domain_list); - - if (sipdebug) - ast_log(LOG_DEBUG, "Added local SIP domain '%s'\n", domain); - - return 1; -} - -/*! \brief check_sip_domain: Check if domain part of uri is local to our server */ -static int check_sip_domain(const char *domain, char *context, size_t len) -{ - struct domain *d; - int result = 0; - - AST_LIST_LOCK(&domain_list); - AST_LIST_TRAVERSE(&domain_list, d, list) { - if (strcasecmp(d->domain, domain)) - continue; - - if (len && !ast_strlen_zero(d->context)) - ast_copy_string(context, d->context, len); - - result = 1; - break; - } - AST_LIST_UNLOCK(&domain_list); - - return result; -} - -/*! \brief clear_sip_domains: Clear our domain list (at reload) */ -static void clear_sip_domains(void) -{ - struct domain *d; - - AST_LIST_LOCK(&domain_list); - while ((d = AST_LIST_REMOVE_HEAD(&domain_list, list))) - free(d); - AST_LIST_UNLOCK(&domain_list); -} - - -/*! \brief add_realm_authentication: 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 (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 (ast_strlen_zero(username) || ast_strlen_zero(realm)) { - ast_log(LOG_WARNING, "Format for authentication entry is user[:secret]@realm at line %d\n", lineno); - return authlist; - } - stringp = username; - username = strsep(&stringp, ":"); - if (username) { - secret = strsep(&stringp, ":"); - if (!secret) { - stringp = username; - md5secret = strsep(&stringp,"#"); - } - } - 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; - -} - -/*! \brief 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; -} - -/*! \brief 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; -} - -/*! \brief 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; - ast_copy_flags(user, &global_flags, SIP_FLAGS_TO_COPY); - 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, "subscribecontext")) { - ast_copy_string(user->subscribecontext, v->value, sizeof(user->subscribecontext)); - } 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, "musicclass") || !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, "call-limit") || !strcasecmp(v->name, "incominglimit")) { - user->call_limit = atoi(v->value); - if (user->call_limit < 0) - user->call_limit = 0; - } else if (!strcasecmp(v->name, "amaflags")) { - format = ast_cdr_amaflags2int(v->value); - if (format < 0) { - ast_log(LOG_WARNING, "Invalid AMA Flags: %s at line %d\n", v->value, v->lineno); - } else { - user->amaflags = format; - } - } else if (!strcasecmp(v->name, "allow")) { - ast_parse_allow_disallow(&user->prefs, &user->capability, v->value, 1); - } else if (!strcasecmp(v->name, "disallow")) { - ast_parse_allow_disallow(&user->prefs, &user->capability, v->value, 0); - } else if (!strcasecmp(v->name, "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; -} - -/*! \brief 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_FLAGS_TO_COPY); - strcpy(peer->context, default_context); - strcpy(peer->subscribecontext, default_subscribecontext); - 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->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; -} - -/*! \brief 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 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 */ - /* We also use a case-sensitive comparison (unlike find_peer) so - that case changes made to the peer name will be properly handled - during reload - */ - peer = ASTOBJ_CONTAINER_FIND_UNLINK_FULL(&peerl, name, name, 0, 0, strcmp); - - if (peer) { - /* Already in the list, remove it and it will be added back (or FREE'd) */ - found++; - } else { - peer = malloc(sizeof(*peer)); - if (peer) { - memset(peer, 0, sizeof(*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; - } - /* 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->subscribecontext, default_subscribecontext); - strcpy(peer->vmexten, global_vmexten); - 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_FLAGS_TO_COPY); - 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, "%ld", (time_t *)®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 (realtime && !strcasecmp(v->name, "fullcontact")) { - ast_copy_string(peer->fullcontact, v->value, sizeof(peer->fullcontact)); - ast_set_flag((&peer->flags_page2), SIP_PAGE2_RT_FROMCONTACT); - } else if (!strcasecmp(v->name, "secret")) - ast_copy_string(peer->secret, v->value, sizeof(peer->secret)); - else if (!strcasecmp(v->name, "md5secret")) - ast_copy_string(peer->md5secret, v->value, sizeof(peer->md5secret)); - else if (!strcasecmp(v->name, "auth")) - peer->auth = add_realm_authentication(peer->auth, v->value, v->lineno); - else if (!strcasecmp(v->name, "callerid")) { - ast_callerid_split(v->value, peer->cid_name, sizeof(peer->cid_name), peer->cid_num, sizeof(peer->cid_num)); - } else if (!strcasecmp(v->name, "context")) { - ast_copy_string(peer->context, v->value, sizeof(peer->context)); - } else if (!strcasecmp(v->name, "subscribecontext")) { - ast_copy_string(peer->subscribecontext, v->value, sizeof(peer->subscribecontext)); - } else if (!strcasecmp(v->name, "fromdomain")) - ast_copy_string(peer->fromdomain, v->value, sizeof(peer->fromdomain)); - else if (!strcasecmp(v->name, "usereqphone")) - ast_set2_flag(peer, 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 (!peer->addr.sin_port) - peer->addr.sin_port = htons(DEFAULT_SIP_PORT); - } - } - } 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, "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, "call-limit") || !strcasecmp(v->name, "incominglimit")) { - peer->call_limit = atoi(v->value); - if (peer->call_limit < 0) - peer->call_limit = 0; - } else if (!strcasecmp(v->name, "amaflags")) { - format = ast_cdr_amaflags2int(v->value); - if (format < 0) { - ast_log(LOG_WARNING, "Invalid AMA Flags for peer: %s at line %d\n", v->value, v->lineno); - } else { - peer->amaflags = format; - } - } else if (!strcasecmp(v->name, "accountcode")) { - ast_copy_string(peer->accountcode, v->value, sizeof(peer->accountcode)); - } else if (!strcasecmp(v->name, "musicclass") || !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, "vmexten")) { - ast_copy_string(peer->vmexten, v->value, sizeof(peer->vmexten)); - } else if (!strcasecmp(v->name, "callgroup")) { - peer->callgroup = ast_get_group(v->value); - } else if (!strcasecmp(v->name, "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 (!ast_test_flag((&global_flags_page2), SIP_PAGE2_IGNOREREGEXPIRE) && ast_test_flag(peer, SIP_DYNAMIC) && realtime) { - time_t nowtime; - - time(&nowtime); - if ((nowtime - regseconds) > 0) { - destroy_association(peer); - memset(&peer->addr, 0, sizeof(peer->addr)); - if (option_debug) - ast_log(LOG_DEBUG, "Bah, we're expired (%d/%d/%d)!\n", (int)(nowtime - regseconds), (int)regseconds, (int)nowtime); - } - } - ast_copy_flags(peer, &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; -} - -/*! \brief 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; - char iabuf[INET_ADDRSTRLEN]; - struct ast_flags dummy; - int auto_sip_domains = 0; - struct sockaddr_in old_bindaddr = bindaddr; - - 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)); - sipdebug &= ~SIP_DEBUG_CONFIG; - - /* Initialize some reasonable defaults at SIP reload */ - ast_copy_string(default_context, DEFAULT_CONTEXT, sizeof(default_context)); - default_subscribecontext[0] = '\0'; - default_language[0] = '\0'; - default_fromdomain[0] = '\0'; - default_qualify = 0; - allow_external_domains = 1; /* Allow external invites */ - externhost[0] = '\0'; - externexpire = 0; - externrefresh = 10; - ast_copy_string(default_useragent, DEFAULT_USERAGENT, sizeof(default_useragent)); - ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime)); - global_notifyringing = 1; - 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; - dumphistory = 0; - recordhistory = 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 = 0; - 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); - ast_set_flag(&global_flags_page2, SIP_PAGE2_RTUPDATE); - global_mwitime = DEFAULT_MWITIME; - strcpy(global_vmexten, DEFAULT_VMEXTEN); - 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, "rtupdate")) { - ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_RTUPDATE); - } else if (!strcasecmp(v->name, "ignoreregexpire")) { - ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_IGNOREREGEXPIRE); - } 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, "vmexten")) { - ast_copy_string(global_vmexten, v->value, sizeof(global_vmexten)); - } else if (!strcasecmp(v->name, "rtptimeout")) { - if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) { - ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno); - global_rtptimeout = 0; - } - } else if (!strcasecmp(v->name, "rtpholdtimeout")) { - if ((sscanf(v->value, "%d", &global_rtpholdtimeout) != 1) || (global_rtpholdtimeout < 0)) { - ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno); - global_rtpholdtimeout = 0; - } - } else if (!strcasecmp(v->name, "rtpkeepalive")) { - if ((sscanf(v->value, "%d", &global_rtpkeepalive) != 1) || (global_rtpkeepalive < 0)) { - ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno); - global_rtpkeepalive = 0; - } - } else if (!strcasecmp(v->name, "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, "notifyringing")) { - global_notifyringing = ast_true(v->value); - } 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")) { - if (ast_true(v->value)) - sipdebug |= SIP_DEBUG_CONFIG; - } else if (!strcasecmp(v->name, "dumphistory")) { - dumphistory = ast_true(v->value); - } else if (!strcasecmp(v->name, "recordhistory")) { - recordhistory = ast_true(v->value); - } else if (!strcasecmp(v->name, "registertimeout")) { - global_reg_timeout = atoi(v->value); - if (global_reg_timeout < 1) - global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT; - } else if (!strcasecmp(v->name, "registerattempts")) { - global_regattempts_max = atoi(v->value); - } else if (!strcasecmp(v->name, "bindaddr")) { - if (!(hp = ast_gethostbyname(v->value, &ahp))) { - ast_log(LOG_WARNING, "Invalid address: %s\n", v->value); - } else { - memcpy(&bindaddr.sin_addr, hp->h_addr, sizeof(bindaddr.sin_addr)); - } - } else if (!strcasecmp(v->name, "localnet")) { - struct ast_ha *na; - if (!(na = ast_append_ha("d", v->value, localaddr))) - ast_log(LOG_WARNING, "Invalid localnet value: %s\n", v->value); - else - localaddr = na; - } else if (!strcasecmp(v->name, "localmask")) { - ast_log(LOG_WARNING, "Use of localmask is no long supported -- use localnet with mask syntax\n"); - } else if (!strcasecmp(v->name, "externip")) { - if (!(hp = ast_gethostbyname(v->value, &ahp))) - ast_log(LOG_WARNING, "Invalid address for externip keyword: %s\n", v->value); - else - memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr)); - externexpire = 0; - } else if (!strcasecmp(v->name, "externhost")) { - ast_copy_string(externhost, v->value, sizeof(externhost)); - if (!(hp = ast_gethostbyname(externhost, &ahp))) - ast_log(LOG_WARNING, "Invalid address for externhost keyword: %s\n", externhost); - else - memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr)); - 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, "allowexternaldomains")) { - allow_external_domains = ast_true(v->value); - } else if (!strcasecmp(v->name, "autodomain")) { - auto_sip_domains = ast_true(v->value); - } else if (!strcasecmp(v->name, "domain")) { - char *domain = ast_strdupa(v->value); - char *context = strchr(domain, ','); - - if (context) - *context++ = '\0'; - - if (ast_strlen_zero(domain)) - ast_log(LOG_WARNING, "Empty domain specified at line %d\n", v->lineno); - else if (ast_strlen_zero(context)) - ast_log(LOG_WARNING, "Empty context specified at line %d for domain '%s'\n", v->lineno, domain); - else - add_sip_domain(ast_strip(domain), SIP_DOMAIN_CONFIG, context ? ast_strip(context) : ""); - } else if (!strcasecmp(v->name, "register")) { - sip_register(v->value, v->lineno); - } else if (!strcasecmp(v->name, "tos")) { - if (ast_str2tos(v->value, &tos)) - 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; - } - - if (!allow_external_domains && AST_LIST_EMPTY(&domain_list)) { - ast_log(LOG_WARNING, "To disallow external domains, you need to configure local SIP domains.\n"); - allow_external_domains = 1; - } - - /* Build list of authentication to various SIP realms, i.e. service providers */ - 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) && (memcmp(&old_bindaddr, &bindaddr, sizeof(struct sockaddr_in)))) { - close(sipsock); - sipsock = -1; - } - if (sipsock < 0) { - sipsock = socket(AF_INET, SOCK_DGRAM, 0); - if (sipsock < 0) { - ast_log(LOG_WARNING, "Unable to create SIP socket: %s\n", strerror(errno)); - } 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); - - /* Add default domains - host name, IP address and IP:port */ - /* Only do this if user added any sip domain with "localdomains" */ - /* In order to *not* break backwards compatibility */ - /* Some phones address us at IP only, some with additional port number */ - if (auto_sip_domains) { - char temp[MAXHOSTNAMELEN]; - - /* First our default IP address */ - if (bindaddr.sin_addr.s_addr) { - ast_inet_ntoa(temp, sizeof(temp), bindaddr.sin_addr); - add_sip_domain(temp, SIP_DOMAIN_AUTO, NULL); - } else { - ast_log(LOG_NOTICE, "Can't add wildcard IP address to domain list, please add IP address to domain manually.\n"); - } - - /* Our extern IP address, if configured */ - if (externip.sin_addr.s_addr) { - ast_inet_ntoa(temp, sizeof(temp), externip.sin_addr); - add_sip_domain(temp, SIP_DOMAIN_AUTO, NULL); - } - - /* Extern host name (NAT traversal support) */ - if (!ast_strlen_zero(externhost)) - add_sip_domain(externhost, SIP_DOMAIN_AUTO, NULL); - - /* Our host name */ - if (!gethostname(temp, sizeof(temp))) - add_sip_domain(temp, SIP_DOMAIN_AUTO, NULL); - } - - /* Release configuration from memory */ - ast_config_destroy(cfg); - - /* Load the list of manual NOTIFY types to support */ - if (notify_types) - ast_config_destroy(notify_types); - notify_types = ast_config_load(notify_config); - - return 0; -} - -/*! \brief 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; -#ifdef SIP_MIDCOM - if (m_cb) - m_cb->ast_rtp_nat_us_audio_hook(rtp, p->r); /* change the ip port in rtp */ -#endif - } - ast_mutex_unlock(&p->lock); - return rtp; -} - -/*! \brief 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; -#ifdef SIP_MIDCOM - if (m_cb) - m_cb->ast_rtp_nat_us_video_hook(rtp, p->r); /* change the ip port in rtp */ -#endif - } - ast_mutex_unlock(&p->lock); - return rtp; -} - -/*! \brief sip_set_rtp_peer: Set the RTP peer for this call ---*/ -static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active) -{ - struct sip_pvt *p; - - p = chan->tech_pvt; - if (!p) - return -1; - ast_mutex_lock(&p->lock); - if (rtp) { - ast_rtp_get_peer(rtp, &p->redirip); -#ifdef SIP_MIDCOM - if (m_cb) - m_cb->ast_rtp_get_their_nat_audio_hook(rtp, p->r); -#endif - } - else - memset(&p->redirip, 0, sizeof(p->redirip)); - if (vrtp) { - ast_rtp_get_peer(vrtp, &p->vredirip); -#ifdef SIP_MIDCOM - if (m_cb) - m_cb->ast_rtp_get_their_nat_video_hook(vrtp, p->r); -#endif - } - else - memset(&p->vredirip, 0, sizeof(p->vredirip)); - p->redircodecs = codecs; - if (!ast_test_flag(p, SIP_GOTREFER)) { - if (!p->pendinginvite) { - if (option_debug > 2) { - char iabuf[INET_ADDRSTRLEN]; - ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip)); - } - transmit_reinvite_with_sdp(p); - } else if (!ast_test_flag(p, SIP_PENDINGBYE)) { - if (option_debug > 2) { - char iabuf[INET_ADDRSTRLEN]; - ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip)); - } - 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-Accountcode:\". 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"; - -/*! \brief 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; -} - -/*! \brief sip_addheader: Add a SIP header ---*/ -static int sip_addheader(struct ast_channel *chan, void *data) -{ - int no = 0; - int ok = 0; - char varbuf[128]; - - if (ast_strlen_zero((char *)data)) { - ast_log(LOG_WARNING, "This application requires the argument: Header\n"); - return 0; - } - ast_mutex_lock(&chan->lock); - - /* Check for headers */ - while (!ok && no <= 50) { - no++; - snprintf(varbuf, sizeof(varbuf), "_SIPADDHEADER%02d", no); - if (ast_strlen_zero(pbx_builtin_getvar_helper(chan, varbuf + 1))) - ok = 1; - } - if (ok) { - pbx_builtin_setvar_helper (chan, varbuf, (char *)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; -} - -/*! \brief 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; -} - -/*! \brief 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; - } - } - } - } - - snprintf(p->our_contact, sizeof(p->our_contact), "Transfer <sip:%s@%s%s%s>", extension, host, port ? ":" : "", port ? port : ""); - transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq, 1); - - /* this is all that we want to send to that SIP device */ - ast_set_flag(p, SIP_ALREADYGONE); - - /* hangup here */ - return -1; -} - -/*! \brief 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; -} - -/*! \brief 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, -}; - -#ifdef SIP_MIDCOM -/*! \brief sip_helper: Interface structure with callbacks used to connect to midcom module --*/ -static struct ast_sip_helper_cb sip_helper = { - ast_rtp_get_peer_audio_helper: sip_rtp_get_peer_audio_helper, - ast_rtp_get_peer_video_helper: sip_rtp_get_peer_video_helper, - ast_rtp_get_us_audio_helper: sip_rtp_get_us_audio_helper, - ast_rtp_get_us_video_helper: sip_rtp_get_us_video_helper, - ast_map_hook_struct: sip_map_hook_struct, - ast_get_hook_struct: sip_get_hook_struct, - ast_get_flag_novideo: sip_get_flag_novideo, - ast_cmp_sa_addr: sip_cmp_sa_addr, - ast_get_recv_addr: sip_get_recv_addr, - ast_get_username: sip_get_username, - ast_channel_helper: sip_channel_helper, - ast_bridged_channel_helper: sip_bridged_channel_helper, - ast_get_capability_helper: sip_get_capability_helper, - ast_softhangup_helper: sip_softhangup_helper, -}; -#endif - -/*! \brief 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) - ); -} - -/*! \brief sip_send_all_registers: Send all known registrations */ -static void sip_send_all_registers(void) -{ - int ms; - int regspacing; - if (!regobjs) - return; - regspacing = default_expiry * 1000/regobjs; - if (regspacing > 100) - regspacing = 100; - ms = regspacing; - ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do { - ASTOBJ_WRLOCK(iterator); - if (iterator->expire > -1) - ast_sched_del(sched, iterator->expire); - ms += regspacing; - iterator->expire = ast_sched_add(sched, ms, sip_reregister, iterator); - ASTOBJ_UNLOCK(iterator); - } while (0) - ); -} - -/*! \brief sip_do_reload: Reload module */ -static int sip_do_reload(void) -{ - clear_realm_authentication(authl); - clear_sip_domains(); - authl = NULL; - - /* First, destroy all outstanding registry calls */ - /* This is needed, since otherwise active registry entries will not be destroyed */ - ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do { - ASTOBJ_RDLOCK(iterator); - if (iterator->call) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", iterator->username, iterator->hostname); - /* This will also remove references to the registry */ - sip_destroy(iterator->call); - } - ASTOBJ_UNLOCK(iterator); - } while(0)); - - 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; -} - -/*! \brief 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; -} - -/*! \brief reload: Part of Asterisk module interface ---*/ -int reload(void) -{ - return sip_reload(0, 0, NULL); -} - -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", "show", "domains", NULL }, sip_show_domains, "List our local SIP domains.", show_domains_usage }, - { { "sip", "show", "settings", NULL }, sip_show_settings, "Show SIP global settings", show_settings_usage }, - { { "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 }, -}; - -/*! \brief load_module: PBX load module - initialization ---*/ -int load_module() -{ - ASTOBJ_CONTAINER_INIT(&userl); /* User object list */ - ASTOBJ_CONTAINER_INIT(&peerl); /* Peer object list */ - ASTOBJ_CONTAINER_INIT(®l); /* Registry object list */ - - 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"); - } - - reload_config(); /* Load the configuration from sip.conf */ - - /* 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; - } - - /* Register all CLI functions for SIP */ - ast_cli_register_multiple(my_clis, sizeof(my_clis)/ sizeof(my_clis[0])); - - /* Tell the RTP subdriver that we're here */ - ast_rtp_proto_register(&sip_rtp); - -#ifdef SIP_MIDCOM - /* Register the sip helper functions */ - if (m_cb) - m_cb->ast_sip_helper_register(&sip_helper); -#endif - - /* Register dialplan applications */ - ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode); - - /* These will be removed soon */ - ast_register_application(app_sipaddheader, sip_addheader, synopsis_sipaddheader, descrip_sipaddheader); - ast_register_application(app_sipgetheader, sip_getheader, synopsis_sipgetheader, descrip_sipgetheader); - - /* Register dialplan functions */ - ast_custom_function_register(&sip_header_function); - ast_custom_function_register(&sippeer_function); - ast_custom_function_register(&sipchaninfo_function); - ast_custom_function_register(&checksipdomain_function); - - /* Register manager commands */ - ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM, manager_sip_show_peers, - "List SIP peers (text format)", mandescr_show_peers); - ast_manager_register2("SIPshowpeer", EVENT_FLAG_SYSTEM, manager_sip_show_peer, - "Show SIP peer (text format)", mandescr_show_peer); - - sip_poke_all_peers(); - sip_send_all_registers(); - - /* And start the monitor for the first time */ - restart_monitor(); - - return 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(&sipchaninfo_function); - ast_custom_function_unregister(&sippeer_function); - ast_custom_function_unregister(&sip_header_function); - ast_custom_function_unregister(&checksipdomain_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); - -#ifdef SIP_MIDCOM - /* Unregister the sip helper functions */ - if (m_cb) - m_cb->ast_sip_helper_unregister(); -#endif - - 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; - } - 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); - clear_sip_domains(); - close(sipsock); - sched_context_destroy(sched); - - return 0; -} - -int usecount() -{ - return usecnt; -} - -char *key() -{ - return ASTERISK_GPL_KEY; -} - -char *description() -{ - return (char *) desc; -} - -#ifdef SIP_MIDCOM -static void sip_rtp_get_peer_audio_helper(void *p, struct sockaddr_in *them) -{ - ast_rtp_get_peer(((struct sip_pvt*)p)->rtp, them); -} - -static void sip_rtp_get_peer_video_helper(void *p, struct sockaddr_in *them) -{ - ast_rtp_get_peer(((struct sip_pvt*)p)->vrtp, them); -} - -static void sip_rtp_get_us_audio_helper(void *p, struct sockaddr_in *sin) -{ - ast_rtp_get_us(((struct sip_pvt*)p)->rtp, sin); - sin->sin_addr = ((struct sip_pvt*)p)->ourip; -} - -static void sip_rtp_get_us_video_helper(void *p, struct sockaddr_in *vsin) -{ - ast_rtp_get_us(((struct sip_pvt*)p)->vrtp, vsin); - vsin->sin_addr = ((struct sip_pvt*)p)->ourip; -} - -static void sip_map_hook_struct(void *p, void *r) -{ - ((struct sip_pvt*)p)->r = r; -} - -static void *sip_get_hook_struct(void *p) -{ - return ((struct sip_pvt*)p)->r; -} - -static int sip_get_flag_novideo(void *p) -{ - return ast_test_flag((struct sip_pvt*)p, SIP_NOVIDEO); -} - -static int sip_cmp_sa_addr(void *p, struct sockaddr_in *addr) -{ - return (((struct sip_pvt*)p)->sa.sin_addr.s_addr == addr->sin_addr.s_addr); -} - -static void sip_get_recv_addr(void *p, struct in_addr *addr) -{ - memcpy(addr, &((struct sip_pvt *)p)->recv.sin_addr, sizeof(struct in_addr)); -} - -static char *sip_get_username(void *p) -{ - return ((struct sip_pvt*)p)->username; -} - -static struct ast_channel *sip_channel_helper(void *p) -{ - return ((struct sip_pvt*)p)->owner; -} - -static struct ast_channel *sip_bridged_channel_helper(void *p) -{ - return ast_bridged_channel(((struct sip_pvt*)p)->owner); -} - -static int sip_get_capability_helper(void *p) -{ - return ((struct sip_pvt*)p)->jointcapability; -} - -static void sip_softhangup_helper(void *p) -{ - if (p && ((struct sip_pvt *)p)->owner) - ast_softhangup(((struct sip_pvt *)p)->owner, AST_SOFTHANGUP_APPUNLOAD); -} -#endif - |