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, 13481 insertions, 0 deletions
diff --git a/1.2-netsec/channels/chan_sip.c b/1.2-netsec/channels/chan_sip.c new file mode 100644 index 000000000..9960b7174 --- /dev/null +++ b/1.2-netsec/channels/chan_sip.c @@ -0,0 +1,13481 @@ +/* + * 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 + |