2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 1999 - 2006, Digium, Inc.
6 * Mark Spencer <markster@digium.com>
8 * See http://www.asterisk.org for more information about
9 * the Asterisk project. Please do not directly contact
10 * any of the maintainers of this project for assistance;
11 * the project provides a web site, mailing lists and IRC
12 * channels for your use.
14 * This program is free software, distributed under the terms of
15 * the GNU General Public License Version 2. See the LICENSE file
16 * at the top of the source tree.
21 * \brief Implementation of Session Initiation Protocol
23 * \author Mark Spencer <markster@digium.com>
26 * \arg \ref AstCREDITS
28 * Implementation of RFC 3261 - without S/MIME, TCP and TLS support
29 * Configuration file \link Config_sip sip.conf \endlink
34 * \todo Better support of forking
35 * \todo VIA branch tag transaction checking
36 * \todo Transaction support
38 * \ingroup channel_drivers
40 * \par Overview of the handling of SIP sessions
41 * The SIP channel handles several types of SIP sessions, or dialogs,
42 * not all of them being "telephone calls".
43 * - Incoming calls that will be sent to the PBX core
44 * - Outgoing calls, generated by the PBX
45 * - SIP subscriptions and notifications of states and voicemail messages
46 * - SIP registrations, both inbound and outbound
47 * - SIP peer management (peerpoke, OPTIONS)
50 * In the SIP channel, there's a list of active SIP dialogs, which includes
51 * all of these when they are active. "sip show channels" in the CLI will
52 * show most of these, excluding subscriptions which are shown by
53 * "sip show subscriptions"
55 * \par incoming packets
56 * Incoming packets are received in the monitoring thread, then handled by
57 * sipsock_read(). This function parses the packet and matches an existing
58 * dialog or starts a new SIP dialog.
60 * sipsock_read sends the packet to handle_request(), that parses a bit more.
61 * if it's a response to an outbound request, it's sent to handle_response().
62 * If it is a request, handle_request sends it to one of a list of functions
63 * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
64 * sipsock_read locks the ast_channel if it exists (an active call) and
65 * unlocks it after we have processed the SIP message.
67 * A new INVITE is sent to handle_request_invite(), that will end up
68 * starting a new channel in the PBX, the new channel after that executing
69 * in a separate channel thread. This is an incoming "call".
70 * When the call is answered, either by a bridged channel or the PBX itself
71 * the sip_answer() function is called.
73 * The actual media - Video or Audio - is mostly handled by the RTP subsystem
77 * Outbound calls are set up by the PBX through the sip_request_call()
78 * function. After that, they are activated by sip_call().
81 * The PBX issues a hangup on both incoming and outgoing calls through
82 * the sip_hangup() function
88 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
94 #include <sys/socket.h>
95 #include <sys/ioctl.h>
102 #include <sys/signal.h>
103 #include <netinet/in.h>
104 #include <netinet/in_systm.h>
105 #include <arpa/inet.h>
106 #include <netinet/ip.h>
109 #include "asterisk/lock.h"
110 #include "asterisk/channel.h"
111 #include "asterisk/config.h"
112 #include "asterisk/logger.h"
113 #include "asterisk/module.h"
114 #include "asterisk/pbx.h"
115 #include "asterisk/options.h"
116 #include "asterisk/sched.h"
117 #include "asterisk/io.h"
118 #include "asterisk/rtp.h"
119 #include "asterisk/udptl.h"
120 #include "asterisk/acl.h"
121 #include "asterisk/manager.h"
122 #include "asterisk/callerid.h"
123 #include "asterisk/cli.h"
124 #include "asterisk/app.h"
125 #include "asterisk/musiconhold.h"
126 #include "asterisk/dsp.h"
127 #include "asterisk/features.h"
128 #include "asterisk/srv.h"
129 #include "asterisk/astdb.h"
130 #include "asterisk/causes.h"
131 #include "asterisk/utils.h"
132 #include "asterisk/file.h"
133 #include "asterisk/astobj.h"
134 #include "asterisk/dnsmgr.h"
135 #include "asterisk/devicestate.h"
136 #include "asterisk/linkedlists.h"
137 #include "asterisk/stringfields.h"
138 #include "asterisk/monitor.h"
139 #include "asterisk/localtime.h"
140 #include "asterisk/abstract_jb.h"
141 #include "asterisk/compiler.h"
142 #include "asterisk/threadstorage.h"
143 #include "asterisk/translate.h"
144 #include "asterisk/version.h"
154 #define VIDEO_CODEC_MASK 0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
155 #ifndef IPTOS_MINCOST
156 #define IPTOS_MINCOST 0x02
159 /* #define VOCAL_DATA_HACK */
161 #define DEFAULT_DEFAULT_EXPIRY 120
162 #define DEFAULT_MIN_EXPIRY 60
163 #define DEFAULT_MAX_EXPIRY 3600
164 #define DEFAULT_REGISTRATION_TIMEOUT 20
165 #define DEFAULT_MAX_FORWARDS "70"
167 /* guard limit must be larger than guard secs */
168 /* guard min must be < 1000, and should be >= 250 */
169 #define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
170 #define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
172 #define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
173 GUARD_PCT turns out to be lower than this, it
174 will use this time instead.
175 This is in milliseconds. */
176 #define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
177 below EXPIRY_GUARD_LIMIT */
178 #define DEFAULT_EXPIRY 900 /*!< Expire slowly */
180 static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
181 static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
182 static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
183 static int expiry = DEFAULT_EXPIRY;
186 #define MAX(a,b) ((a) > (b) ? (a) : (b))
189 #define CALLERID_UNKNOWN "Unknown"
191 #define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
192 #define DEFAULT_FREQ_OK 60 * 1000 /*!< Qualification: How often to check for the host to be up */
193 #define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */
195 #define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
196 #define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
197 #define SIP_TIMER_T1 500 /* SIP timer T1 (according to RFC 3261) */
198 #define SIP_TRANS_TIMEOUT 32000 /*!< SIP request timeout (rfc 3261) 64*T1
199 \todo Use known T1 for timeout (peerpoke)
201 #define DEFAULT_TRANS_TIMEOUT -1 /* Use default SIP transaction timeout */
202 #define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
204 #define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
205 #define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
206 #define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */
208 #define INITIAL_CSEQ 101 /*!< our initial sip sequence number */
210 /*! \brief Global jitterbuffer configuration - by default, jb is disabled */
211 static struct ast_jb_conf default_jbconf =
215 .resync_threshold = -1,
218 static struct ast_jb_conf global_jbconf;
220 static const char config[] = "sip.conf";
221 static const char notify_config[] = "sip_notify.conf";
226 /*! \brief Authorization scheme for call transfers
227 \note Not a bitfield flag, since there are plans for other modes,
228 like "only allow transfers for authenticated devices" */
230 TRANSFER_OPENFORALL, /*!< Allow all SIP transfers */
231 TRANSFER_CLOSED, /*!< Allow no SIP transfers */
240 /*! \brief States for the INVITE transaction, not the dialog
241 \note this is for the INVITE that sets up the dialog
244 INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
245 INV_CALLING = 1, /*!< Invite sent, no answer */
246 INV_PROCEEDING = 2, /*!< We got/sent 1xx message */
247 INV_EARLY_MEDIA = 3, /*!< We got 18x message with to-tag back */
248 INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
249 INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */
250 INV_TERMINATED = 6, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
251 The only way out of this is a BYE from one side */
252 INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
255 /* Do _NOT_ make any changes to this enum, or the array following it;
256 if you think you are doing the right thing, you are probably
257 not doing the right thing. If you think there are changes
258 needed, get someone else to review them first _before_
259 submitting a patch. If these two lists do not match properly
260 bad things will happen.
264 XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
265 If it fails, it's critical and will cause a teardown of the session */
266 XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
267 XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */
270 enum parse_register_result {
271 PARSE_REGISTER_FAILED,
272 PARSE_REGISTER_UPDATE,
273 PARSE_REGISTER_QUERY,
276 enum subscriptiontype {
285 static const struct cfsubscription_types {
286 enum subscriptiontype type;
287 const char * const event;
288 const char * const mediatype;
289 const char * const text;
290 } subscription_types[] = {
291 { NONE, "-", "unknown", "unknown" },
292 /* RFC 4235: SIP Dialog event package */
293 { DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" },
294 { CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */
295 { PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */
296 { XPIDF_XML, "presence", "application/xpidf+xml", "xpidf+xml" }, /* Pre-RFC 3863 with MS additions */
297 { MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
300 /*! \brief SIP Request methods known by Asterisk */
302 SIP_UNKNOWN, /* Unknown response */
303 SIP_RESPONSE, /* Not request, response to outbound request */
309 SIP_PRACK, /* Not supported at all */
314 SIP_UPDATE, /* We can send UPDATE; but not accept it */
317 SIP_PUBLISH, /* Not supported at all */
318 SIP_PING, /* Not supported at all, no standard but still implemented out there */
321 /*! \brief Authentication types - proxy or www authentication
322 \note Endpoints, like Asterisk, should always use WWW authentication to
323 allow multiple authentications in the same call - to the proxy and
331 /*! \brief Authentication result from check_auth* functions */
332 enum check_auth_result {
333 AUTH_DONT_KNOW = -100, /*!< no result, need to check further */
334 /* XXX maybe this is the same as AUTH_NOT_FOUND */
337 AUTH_CHALLENGE_SENT = 1,
338 AUTH_SECRET_FAILED = -1,
339 AUTH_USERNAME_MISMATCH = -2,
340 AUTH_NOT_FOUND = -3, /* returned by register_verify */
342 AUTH_UNKNOWN_DOMAIN = -5,
345 /*! \brief States for outbound registrations (with register= lines in sip.conf */
346 enum sipregistrystate {
347 REG_STATE_UNREGISTERED = 0, /*!< We are not registred */
348 REG_STATE_REGSENT, /*!< Registration request sent */
349 REG_STATE_AUTHSENT, /*!< We have tried to authenticate */
350 REG_STATE_REGISTERED, /*!< Registered and done */
351 REG_STATE_REJECTED, /*!< Registration rejected */
352 REG_STATE_TIMEOUT, /*!< Registration timed out */
353 REG_STATE_NOAUTH, /*!< We have no accepted credentials */
354 REG_STATE_FAILED, /*!< Registration failed after several tries */
357 /*! \brief definition of a sip proxy server
359 * For outbound proxies, this is allocated in the SIP peer dynamically or
360 * statically as the global_outboundproxy. The pointer in a SIP message is just
361 * a pointer and should *not* be de-allocated.
364 char name[MAXHOSTNAMELEN]; /*!< DNS name of domain/host or IP */
365 struct sockaddr_in ip; /*!< Currently used IP address and port */
366 time_t last_dnsupdate; /*!< When this was resolved */
367 int force; /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
368 /* Room for a SRV record chain based on the name */
371 enum can_create_dialog {
372 CAN_NOT_CREATE_DIALOG,
374 CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
377 /*! XXX Note that sip_methods[i].id == i must hold or the code breaks */
378 static const struct cfsip_methods {
380 int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
382 enum can_create_dialog can_create;
384 { SIP_UNKNOWN, RTP, "-UNKNOWN-", CAN_CREATE_DIALOG },
385 { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
386 { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
387 { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
388 { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
389 { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
390 { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
391 { SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG },
392 { SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG },
393 { SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG },
394 { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE", CAN_CREATE_DIALOG },
395 { SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG },
396 { SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG },
397 { SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG },
398 { SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG },
399 { SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD },
400 { SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
403 /*! Define SIP option tags, used in Require: and Supported: headers
404 We need to be aware of these properties in the phones to use
405 the replace: header. We should not do that without knowing
406 that the other end supports it...
407 This is nothing we can configure, we learn by the dialog
408 Supported: header on the REGISTER (peer) or the INVITE
410 We are not using many of these today, but will in the future.
411 This is documented in RFC 3261
414 #define NOT_SUPPORTED 0
416 #define SIP_OPT_REPLACES (1 << 0)
417 #define SIP_OPT_100REL (1 << 1)
418 #define SIP_OPT_TIMER (1 << 2)
419 #define SIP_OPT_EARLY_SESSION (1 << 3)
420 #define SIP_OPT_JOIN (1 << 4)
421 #define SIP_OPT_PATH (1 << 5)
422 #define SIP_OPT_PREF (1 << 6)
423 #define SIP_OPT_PRECONDITION (1 << 7)
424 #define SIP_OPT_PRIVACY (1 << 8)
425 #define SIP_OPT_SDP_ANAT (1 << 9)
426 #define SIP_OPT_SEC_AGREE (1 << 10)
427 #define SIP_OPT_EVENTLIST (1 << 11)
428 #define SIP_OPT_GRUU (1 << 12)
429 #define SIP_OPT_TARGET_DIALOG (1 << 13)
430 #define SIP_OPT_NOREFERSUB (1 << 14)
431 #define SIP_OPT_HISTINFO (1 << 15)
432 #define SIP_OPT_RESPRIORITY (1 << 16)
434 /*! \brief List of well-known SIP options. If we get this in a require,
435 we should check the list and answer accordingly. */
436 static const struct cfsip_options {
437 int id; /*!< Bitmap ID */
438 int supported; /*!< Supported by Asterisk ? */
439 char * const text; /*!< Text id, as in standard */
440 } sip_options[] = { /* XXX used in 3 places */
441 /* RFC3891: Replaces: header for transfer */
442 { SIP_OPT_REPLACES, SUPPORTED, "replaces" },
443 /* One version of Polycom firmware has the wrong label */
444 { SIP_OPT_REPLACES, SUPPORTED, "replace" },
445 /* RFC3262: PRACK 100% reliability */
446 { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" },
447 /* RFC4028: SIP Session Timers */
448 { SIP_OPT_TIMER, NOT_SUPPORTED, "timer" },
449 /* RFC3959: SIP Early session support */
450 { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" },
451 /* RFC3911: SIP Join header support */
452 { SIP_OPT_JOIN, NOT_SUPPORTED, "join" },
453 /* RFC3327: Path support */
454 { SIP_OPT_PATH, NOT_SUPPORTED, "path" },
455 /* RFC3840: Callee preferences */
456 { SIP_OPT_PREF, NOT_SUPPORTED, "pref" },
457 /* RFC3312: Precondition support */
458 { SIP_OPT_PRECONDITION, NOT_SUPPORTED, "precondition" },
459 /* RFC3323: Privacy with proxies*/
460 { SIP_OPT_PRIVACY, NOT_SUPPORTED, "privacy" },
461 /* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
462 { SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp-anat" },
463 /* RFC3329: Security agreement mechanism */
464 { SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" },
465 /* SIMPLE events: RFC4662 */
466 { SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" },
467 /* GRUU: Globally Routable User Agent URI's */
468 { SIP_OPT_GRUU, NOT_SUPPORTED, "gruu" },
469 /* RFC4538: Target-dialog */
470 { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "tdialog" },
471 /* Disable the REFER subscription, RFC 4488 */
472 { SIP_OPT_NOREFERSUB, NOT_SUPPORTED, "norefersub" },
473 /* ietf-sip-history-info-06.txt */
474 { SIP_OPT_HISTINFO, NOT_SUPPORTED, "histinfo" },
475 /* ietf-sip-resource-priority-10.txt */
476 { SIP_OPT_RESPRIORITY, NOT_SUPPORTED, "resource-priority" },
480 /*! \brief SIP Methods we support */
481 #define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
483 /*! \brief SIP Extensions we support */
484 #define SUPPORTED_EXTENSIONS "replaces"
486 /*! \brief Standard SIP port from RFC 3261. DO NOT CHANGE THIS */
487 #define STANDARD_SIP_PORT 5060
488 /* Note: in many SIP headers, absence of a port number implies port 5060,
489 * and this is why we cannot change the above constant.
490 * There is a limited number of places in asterisk where we could,
491 * in principle, use a different "default" port number, but
492 * we do not support this feature at the moment.
495 /* Default values, set and reset in reload_config before reading configuration */
496 /* These are default values in the source. There are other recommended values in the
497 sip.conf.sample for new installations. These may differ to keep backwards compatibility,
498 yet encouraging new behaviour on new installations
500 #define DEFAULT_CONTEXT "default"
501 #define DEFAULT_MOHINTERPRET "default"
502 #define DEFAULT_MOHSUGGEST ""
503 #define DEFAULT_VMEXTEN "asterisk"
504 #define DEFAULT_CALLERID "asterisk"
505 #define DEFAULT_NOTIFYMIME "application/simple-message-summary"
506 #define DEFAULT_MWITIME 10
507 #define DEFAULT_ALLOWGUEST TRUE
508 #define DEFAULT_SRVLOOKUP FALSE /*!< Recommended setting is ON */
509 #define DEFAULT_COMPACTHEADERS FALSE
510 #define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
511 #define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
512 #define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
513 #define DEFAULT_TOS_TEXT 0 /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
514 #define DEFAULT_ALLOW_EXT_DOM TRUE
515 #define DEFAULT_REALM "asterisk"
516 #define DEFAULT_NOTIFYRINGING TRUE
517 #define DEFAULT_PEDANTIC FALSE
518 #define DEFAULT_AUTOCREATEPEER FALSE
519 #define DEFAULT_QUALIFY FALSE
520 #define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
521 #define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
522 #ifndef DEFAULT_USERAGENT
523 #define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
527 /* Default setttings are used as a channel setting and as a default when
528 configuring devices */
529 static char default_context[AST_MAX_CONTEXT];
530 static char default_subscribecontext[AST_MAX_CONTEXT];
531 static char default_language[MAX_LANGUAGE];
532 static char default_callerid[AST_MAX_EXTENSION];
533 static char default_fromdomain[AST_MAX_EXTENSION];
534 static char default_notifymime[AST_MAX_EXTENSION];
535 static int default_qualify; /*!< Default Qualify= setting */
536 static char default_vmexten[AST_MAX_EXTENSION];
537 static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
538 static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
539 * a bridged channel on hold */
540 static int default_maxcallbitrate; /*!< Maximum bitrate for call */
541 static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
543 /* Global settings only apply to the channel */
544 static int global_directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */
545 static int global_limitonpeers; /*!< Match call limit on peers only */
546 static int global_rtautoclear;
547 static int global_notifyringing; /*!< Send notifications on ringing */
548 static int global_notifyhold; /*!< Send notifications on hold */
549 static int global_alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
550 static int global_srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
551 static int pedanticsipchecking; /*!< Extra checking ? Default off */
552 static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
553 static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
554 static int global_relaxdtmf; /*!< Relax DTMF */
555 static int global_rtptimeout; /*!< Time out call if no RTP */
556 static int global_rtpholdtimeout;
557 static int global_rtpkeepalive; /*!< Send RTP keepalives */
558 static int global_reg_timeout;
559 static int global_regattempts_max; /*!< Registration attempts before giving up */
560 static int global_allowguest; /*!< allow unauthenticated users/peers to connect? */
561 static int global_allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
562 the global setting is in globals_flags[1] */
563 static int global_mwitime; /*!< Time between MWI checks for peers */
564 static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
565 static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
566 static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
567 static unsigned int global_tos_text; /*!< IP type of service for text RTP packets */
568 static int compactheaders; /*!< send compact sip headers */
569 static int recordhistory; /*!< Record SIP history. Off by default */
570 static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
571 static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
572 static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
573 static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
574 static int allow_external_domains; /*!< Accept calls to external SIP domains? */
575 static int global_callevents; /*!< Whether we send manager events or not */
576 static int global_t1min; /*!< T1 roundtrip time minimum */
577 static int global_autoframing; /*!< Turn autoframing on or off. */
578 static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */
579 static struct sip_proxy global_outboundproxy; /*!< Outbound proxy */
581 static int global_matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
583 /*! \brief Codecs that we support by default: */
584 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
586 /* Object counters */
587 static int suserobjs = 0; /*!< Static users */
588 static int ruserobjs = 0; /*!< Realtime users */
589 static int speerobjs = 0; /*!< Statis peers */
590 static int rpeerobjs = 0; /*!< Realtime peers */
591 static int apeerobjs = 0; /*!< Autocreated peer objects */
592 static int regobjs = 0; /*!< Registry objects */
594 static struct ast_flags global_flags[2] = {{0}}; /*!< global SIP_ flags */
596 AST_MUTEX_DEFINE_STATIC(netlock);
598 /*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
599 when it's doing something critical. */
601 AST_MUTEX_DEFINE_STATIC(monlock);
603 AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
605 /*! \brief This is the thread for the monitor which checks for input on the channels
606 which are not currently in use. */
607 static pthread_t monitor_thread = AST_PTHREADT_NULL;
609 static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
610 static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
612 static struct sched_context *sched; /*!< The scheduling context */
613 static struct io_context *io; /*!< The IO context */
614 static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
616 #define DEC_CALL_LIMIT 0
617 #define INC_CALL_LIMIT 1
618 #define DEC_CALL_RINGING 2
619 #define INC_CALL_RINGING 3
621 /*! \brief sip_request: The data grabbed from the UDP socket */
623 char *rlPart1; /*!< SIP Method Name or "SIP/2.0" protocol version */
624 char *rlPart2; /*!< The Request URI or Response Status */
625 int len; /*!< Length */
626 int headers; /*!< # of SIP Headers */
627 int method; /*!< Method of this request */
628 int lines; /*!< Body Content */
629 unsigned int flags; /*!< SIP_PKT Flags for this packet */
630 char *header[SIP_MAX_HEADERS];
631 char *line[SIP_MAX_LINES];
632 char data[SIP_MAX_PACKET];
633 unsigned int sdp_start; /*!< the line number where the SDP begins */
634 unsigned int sdp_end; /*!< the line number where the SDP ends */
638 * A sip packet is stored into the data[] buffer, with the header followed
639 * by an empty line and the body of the message.
640 * On outgoing packets, data is accumulated in data[] with len reflecting
641 * the next available byte, headers and lines count the number of lines
642 * in both parts. There are no '\0' in data[0..len-1].
644 * On received packet, the input read from the socket is copied into data[],
645 * len is set and the string is NUL-terminated. Then a parser fills up
646 * the other fields -header[] and line[] to point to the lines of the
647 * message, rlPart1 and rlPart2 parse the first lnie as below:
649 * Requests have in the first line METHOD URI SIP/2.0
650 * rlPart1 = method; rlPart2 = uri;
651 * Responses have in the first line SIP/2.0 code description
652 * rlPart1 = SIP/2.0; rlPart2 = code + description;
656 /*! \brief structure used in transfers */
658 struct ast_channel *chan1; /*!< First channel involved */
659 struct ast_channel *chan2; /*!< Second channel involved */
660 struct sip_request req; /*!< Request that caused the transfer (REFER) */
661 int seqno; /*!< Sequence number */
666 /*! \brief Parameters to the transmit_invite function */
667 struct sip_invite_param {
668 int addsipheaders; /*!< Add extra SIP headers */
669 const char *uri_options; /*!< URI options to add to the URI */
670 const char *vxml_url; /*!< VXML url for Cisco phones */
671 char *auth; /*!< Authentication */
672 char *authheader; /*!< Auth header */
673 enum sip_auth_type auth_type; /*!< Authentication type */
674 const char *replaces; /*!< Replaces header for call transfers */
675 int transfer; /*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */
678 /*! \brief Structure to save routing information for a SIP session */
680 struct sip_route *next;
684 /*! \brief Modes for SIP domain handling in the PBX */
686 SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */
687 SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */
690 /*! \brief Domain data structure.
691 \note In the future, we will connect this to a configuration tree specific
695 char domain[MAXHOSTNAMELEN]; /*!< SIP domain we are responsible for */
696 char context[AST_MAX_EXTENSION]; /*!< Incoming context for this domain */
697 enum domain_mode mode; /*!< How did we find this domain? */
698 AST_LIST_ENTRY(domain) list; /*!< List mechanics */
701 static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */
704 /*! \brief sip_history: Structure for saving transactions within a SIP dialog */
706 AST_LIST_ENTRY(sip_history) list;
707 char event[0]; /* actually more, depending on needs */
710 AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
712 /*! \brief sip_auth: Credentials for authentication to other SIP services */
714 char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */
715 char username[256]; /*!< Username */
716 char secret[256]; /*!< Secret */
717 char md5secret[256]; /*!< MD5Secret */
718 struct sip_auth *next; /*!< Next auth structure in list */
721 /*--- Various flags for the flags field in the pvt structure */
722 #define SIP_ALREADYGONE (1 << 0) /*!< Whether or not we've already been destroyed by our peer */
723 #define SIP_NEEDDESTROY (1 << 1) /*!< if we need to be destroyed by the monitor thread */
724 #define SIP_NOVIDEO (1 << 2) /*!< Didn't get video in invite, don't offer */
725 #define SIP_RINGING (1 << 3) /*!< Have sent 180 ringing */
726 #define SIP_PROGRESS_SENT (1 << 4) /*!< Have sent 183 message progress */
727 #define SIP_NEEDREINVITE (1 << 5) /*!< Do we need to send another reinvite? */
728 #define SIP_PENDINGBYE (1 << 6) /*!< Need to send bye after we ack? */
729 #define SIP_GOTREFER (1 << 7) /*!< Got a refer? */
730 #define SIP_PROMISCREDIR (1 << 8) /*!< Promiscuous redirection */
731 #define SIP_TRUSTRPID (1 << 9) /*!< Trust RPID headers? */
732 #define SIP_USEREQPHONE (1 << 10) /*!< Add user=phone to numeric URI. Default off */
733 #define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
734 #define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
735 #define SIP_OUTGOING (1 << 13) /*!< Direction of the last transaction in this dialog */
736 #define SIP_FREE_BIT (1 << 14) /*!< ---- */
737 #define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */
738 #define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
739 #define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */
740 #define SIP_DTMF_INBAND (1 << 16) /*!< DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
741 #define SIP_DTMF_INFO (2 << 16) /*!< DTMF Support: SIP Info messages - "info" */
742 #define SIP_DTMF_AUTO (3 << 16) /*!< DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
744 #define SIP_NAT (3 << 18) /*!< four settings, uses two bits */
745 #define SIP_NAT_NEVER (0 << 18) /*!< No nat support */
746 #define SIP_NAT_RFC3581 (1 << 18) /*!< NAT RFC3581 */
747 #define SIP_NAT_ROUTE (2 << 18) /*!< NAT Only ROUTE */
748 #define SIP_NAT_ALWAYS (3 << 18) /*!< NAT Both ROUTE and RFC3581 */
749 /* re-INVITE related settings */
750 #define SIP_REINVITE (7 << 20) /*!< three bits used */
751 #define SIP_CAN_REINVITE (1 << 20) /*!< allow peers to be reinvited to send media directly p2p */
752 #define SIP_CAN_REINVITE_NAT (2 << 20) /*!< allow media reinvite when new peer is behind NAT */
753 #define SIP_REINVITE_UPDATE (4 << 20) /*!< use UPDATE (RFC3311) when reinviting this peer */
754 /* "insecure" settings */
755 #define SIP_INSECURE_PORT (1 << 23) /*!< don't require matching port for incoming requests */
756 #define SIP_INSECURE_INVITE (1 << 24) /*!< don't require authentication for incoming INVITEs */
757 /* Sending PROGRESS in-band settings */
758 #define SIP_PROG_INBAND (3 << 25) /*!< three settings, uses two bits */
759 #define SIP_PROG_INBAND_NEVER (0 << 25)
760 #define SIP_PROG_INBAND_NO (1 << 25)
761 #define SIP_PROG_INBAND_YES (2 << 25)
762 #define SIP_NO_HISTORY (1 << 27) /*!< Suppress recording request/response history */
763 #define SIP_CALL_LIMIT (1 << 28) /*!< Call limit enforced for this call */
764 #define SIP_SENDRPID (1 << 29) /*!< Remote Party-ID Support */
765 #define SIP_INC_COUNT (1 << 30) /*!< Did this connection increment the counter of in-use calls? */
766 #define SIP_G726_NONSTANDARD (1 << 31) /*!< Use non-standard packing for G726-32 data */
768 #define SIP_FLAGS_TO_COPY \
769 (SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
770 SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
771 SIP_USEREQPHONE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE)
773 /*--- a new page of flags (for flags[1] */
775 #define SIP_PAGE2_RTCACHEFRIENDS (1 << 0)
776 #define SIP_PAGE2_RTUPDATE (1 << 1)
777 #define SIP_PAGE2_RTAUTOCLEAR (1 << 2)
778 #define SIP_PAGE2_RT_FROMCONTACT (1 << 4)
779 #define SIP_PAGE2_RTSAVE_SYSNAME (1 << 5)
780 /* Space for addition of other realtime flags in the future */
781 #define SIP_PAGE2_IGNOREREGEXPIRE (1 << 10)
782 #define SIP_PAGE2_DEBUG (3 << 11)
783 #define SIP_PAGE2_DEBUG_CONFIG (1 << 11)
784 #define SIP_PAGE2_DEBUG_CONSOLE (1 << 12)
785 #define SIP_PAGE2_DYNAMIC (1 << 13) /*!< Dynamic Peers register with Asterisk */
786 #define SIP_PAGE2_SELFDESTRUCT (1 << 14) /*!< Automatic peers need to destruct themselves */
787 #define SIP_PAGE2_VIDEOSUPPORT (1 << 15)
788 #define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 16) /*!< Allow subscriptions from this peer? */
789 #define SIP_PAGE2_ALLOWOVERLAP (1 << 17) /*!< Allow overlap dialing ? */
790 #define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 18) /*!< Only issue MWI notification if subscribed to */
791 #define SIP_PAGE2_INC_RINGING (1 << 19) /*!< Did this connection increment the counter of in-use calls? */
792 #define SIP_PAGE2_T38SUPPORT (7 << 20) /*!< T38 Fax Passthrough Support */
793 #define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 20) /*!< 20: T38 Fax Passthrough Support */
794 #define SIP_PAGE2_T38SUPPORT_RTP (2 << 20) /*!< 21: T38 Fax Passthrough Support (not implemented) */
795 #define SIP_PAGE2_T38SUPPORT_TCP (4 << 20) /*!< 22: T38 Fax Passthrough Support (not implemented) */
796 #define SIP_PAGE2_CALL_ONHOLD (3 << 23) /*!< Call states */
797 #define SIP_PAGE2_CALL_ONHOLD_ONEDIR (1 << 23) /*!< 23: One directional hold */
798 #define SIP_PAGE2_CALL_ONHOLD_INACTIVE (1 << 24) /*!< 24: Inactive */
799 #define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< 25: Compensate for buggy RFC2833 implementations */
800 #define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< 26: Buggy CISCO MWI fix */
801 #define SIP_PAGE2_NOTEXT (1 << 27) /*!< 27: Text not supported */
802 #define SIP_PAGE2_TEXTSUPPORT (1 << 28) /*!< 28: Global text enable */
803 #define SIP_PAGE2_DEBUG_TEXT (1 << 29) /*!< 29: Global text debug */
804 #define SIP_PAGE2_OUTGOING_CALL (1 << 30) /*!< 30: Is this an outgoing call? */
806 #define SIP_PAGE2_FLAGS_TO_COPY \
807 (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
808 SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
809 SIP_PAGE2_TEXTSUPPORT )
811 /* SIP packet flags */
812 #define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
813 #define SIP_PKT_WITH_TOTAG (1 << 1) /*!< This packet has a to-tag */
814 #define SIP_PKT_IGNORE (1 << 2) /*!< This is a re-transmit, ignore it */
816 /* T.38 set of flags */
817 #define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0 (unset)*/
818 #define T38FAX_TRANSCODING_MMR (1 << 1) /*!< Default: 0 (unset)*/
819 #define T38FAX_TRANSCODING_JBIG (1 << 2) /*!< Default: 0 (unset)*/
820 /* Rate management */
821 #define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF (0 << 3)
822 #define T38FAX_RATE_MANAGEMENT_LOCAL_TCF (1 << 3) /*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */
823 /* UDP Error correction */
824 #define T38FAX_UDP_EC_NONE (0 << 4) /*!< two bits, if unset NO t38UDPEC field in T38 SDP*/
825 #define T38FAX_UDP_EC_FEC (1 << 4) /*!< Set for t38UDPFEC */
826 #define T38FAX_UDP_EC_REDUNDANCY (2 << 4) /*!< Set for t38UDPRedundancy */
827 /* T38 Spec version */
828 #define T38FAX_VERSION (3 << 6) /*!< two bits, 2 values so far, up to 4 values max */
829 #define T38FAX_VERSION_0 (0 << 6) /*!< Version 0 */
830 #define T38FAX_VERSION_1 (1 << 6) /*!< Version 1 */
831 /* Maximum Fax Rate */
832 #define T38FAX_RATE_2400 (1 << 8) /*!< 2400 bps t38FaxRate */
833 #define T38FAX_RATE_4800 (1 << 9) /*!< 4800 bps t38FaxRate */
834 #define T38FAX_RATE_7200 (1 << 10) /*!< 7200 bps t38FaxRate */
835 #define T38FAX_RATE_9600 (1 << 11) /*!< 9600 bps t38FaxRate */
836 #define T38FAX_RATE_12000 (1 << 12) /*!< 12000 bps t38FaxRate */
837 #define T38FAX_RATE_14400 (1 << 13) /*!< 14400 bps t38FaxRate */
839 /*!< This is default: NO MMR and JBIG transcoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
840 static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
842 #define sipdebug ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
843 #define sipdebug_config ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)
844 #define sipdebug_console ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)
845 #define sipdebug_text ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_TEXT)
847 /*! \brief T38 States for a call */
849 T38_DISABLED = 0, /*!< Not enabled */
850 T38_LOCAL_DIRECT, /*!< Offered from local */
851 T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */
852 T38_PEER_DIRECT, /*!< Offered from peer */
853 T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */
854 T38_ENABLED /*!< Negotiated (enabled) */
857 /*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
858 struct t38properties {
859 struct ast_flags t38support; /*!< Flag for udptl, rtp or tcp support for this session */
860 int capability; /*!< Our T38 capability */
861 int peercapability; /*!< Peers T38 capability */
862 int jointcapability; /*!< Supported T38 capability at both ends */
863 enum t38state state; /*!< T.38 state */
866 /*! \brief Parameters to know status of transfer */
868 REFER_IDLE, /*!< No REFER is in progress */
869 REFER_SENT, /*!< Sent REFER to transferee */
870 REFER_RECEIVED, /*!< Received REFER from transferrer */
871 REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING */
872 REFER_ACCEPTED, /*!< Accepted by transferee */
873 REFER_RINGING, /*!< Target Ringing */
874 REFER_200OK, /*!< Answered by transfer target */
875 REFER_FAILED, /*!< REFER declined - go on */
876 REFER_NOAUTH /*!< We had no auth for REFER */
879 static const struct c_referstatusstring {
880 enum referstatus status;
882 } referstatusstrings[] = {
883 { REFER_IDLE, "<none>" },
884 { REFER_SENT, "Request sent" },
885 { REFER_RECEIVED, "Request received" },
886 { REFER_ACCEPTED, "Accepted" },
887 { REFER_RINGING, "Target ringing" },
888 { REFER_200OK, "Done" },
889 { REFER_FAILED, "Failed" },
890 { REFER_NOAUTH, "Failed - auth failure" }
893 /*! \brief Structure to handle SIP transfers. Dynamically allocated when needed */
894 /* OEJ: Should be moved to string fields */
896 char refer_to[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO extension */
897 char refer_to_domain[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO domain */
898 char refer_to_urioption[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO uri options */
899 char refer_to_context[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO context */
900 char referred_by[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
901 char referred_by_name[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
902 char refer_contact[AST_MAX_EXTENSION]; /*!< Place to store Contact info from a REFER extension */
903 char replaces_callid[BUFSIZ]; /*!< Replace info: callid */
904 char replaces_callid_totag[BUFSIZ/2]; /*!< Replace info: to-tag */
905 char replaces_callid_fromtag[BUFSIZ/2]; /*!< Replace info: from-tag */
906 struct sip_pvt *refer_call; /*!< Call we are referring */
907 int attendedtransfer; /*!< Attended or blind transfer? */
908 int localtransfer; /*!< Transfer to local domain? */
909 enum referstatus status; /*!< REFER status */
912 /*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe */
914 ast_mutex_t pvt_lock; /*!< Dialog private lock */
915 enum invitestates invitestate; /*!< Track state of SIP_INVITEs */
916 int method; /*!< SIP method that opened this dialog */
917 AST_DECLARE_STRING_FIELDS(
918 AST_STRING_FIELD(callid); /*!< Global CallID */
919 AST_STRING_FIELD(randdata); /*!< Random data */
920 AST_STRING_FIELD(accountcode); /*!< Account code */
921 AST_STRING_FIELD(realm); /*!< Authorization realm */
922 AST_STRING_FIELD(nonce); /*!< Authorization nonce */
923 AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
924 AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
925 AST_STRING_FIELD(domain); /*!< Authorization domain */
926 AST_STRING_FIELD(from); /*!< The From: header */
927 AST_STRING_FIELD(useragent); /*!< User agent in SIP request */
928 AST_STRING_FIELD(exten); /*!< Extension where to start */
929 AST_STRING_FIELD(context); /*!< Context for this call */
930 AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */
931 AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */
932 AST_STRING_FIELD(fromdomain); /*!< Domain to show in the from field */
933 AST_STRING_FIELD(fromuser); /*!< User to show in the user field */
934 AST_STRING_FIELD(fromname); /*!< Name to show in the user field */
935 AST_STRING_FIELD(tohost); /*!< Host we should put in the "to" field */
936 AST_STRING_FIELD(language); /*!< Default language for this call */
937 AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */
938 AST_STRING_FIELD(mohsuggest); /*!< MOH class to suggest when putting a peer on hold */
939 AST_STRING_FIELD(rdnis); /*!< Referring DNIS */
940 AST_STRING_FIELD(redircause); /*!< Referring cause */
941 AST_STRING_FIELD(theirtag); /*!< Their tag */
942 AST_STRING_FIELD(username); /*!< [user] name */
943 AST_STRING_FIELD(peername); /*!< [peer] name, not set if [user] */
944 AST_STRING_FIELD(authname); /*!< Who we use for authentication */
945 AST_STRING_FIELD(uri); /*!< Original requested URI */
946 AST_STRING_FIELD(okcontacturi); /*!< URI from the 200 OK on INVITE */
947 AST_STRING_FIELD(peersecret); /*!< Password */
948 AST_STRING_FIELD(peermd5secret);
949 AST_STRING_FIELD(cid_num); /*!< Caller*ID number */
950 AST_STRING_FIELD(cid_name); /*!< Caller*ID name */
951 AST_STRING_FIELD(via); /*!< Via: header */
952 AST_STRING_FIELD(fullcontact); /*!< The Contact: that the UA registers with us */
953 /* we only store the part in <brackets> in this field. */
954 AST_STRING_FIELD(our_contact); /*!< Our contact header */
955 AST_STRING_FIELD(rpid); /*!< Our RPID header */
956 AST_STRING_FIELD(rpid_from); /*!< Our RPID From header */
958 unsigned int ocseq; /*!< Current outgoing seqno */
959 unsigned int icseq; /*!< Current incoming seqno */
960 ast_group_t callgroup; /*!< Call group */
961 ast_group_t pickupgroup; /*!< Pickup group */
962 int lastinvite; /*!< Last Cseq of invite */
963 struct ast_flags flags[2]; /*!< SIP_ flags */
964 int timer_t1; /*!< SIP timer T1, ms rtt */
965 unsigned int sipoptions; /*!< Supported SIP options on the other end */
966 struct ast_codec_pref prefs; /*!< codec prefs */
967 int capability; /*!< Special capability (codec) */
968 int jointcapability; /*!< Supported capability at both ends (codecs) */
969 int peercapability; /*!< Supported peer capability */
970 int prefcodec; /*!< Preferred codec (outbound only) */
971 int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
972 int jointnoncodeccapability; /*!< Joint Non codec capability */
973 int redircodecs; /*!< Redirect codecs */
974 int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
975 struct sip_proxy *outboundproxy; /*!< Outbound proxy for this dialog */
976 struct t38properties t38; /*!< T38 settings */
977 struct sockaddr_in udptlredirip; /*!< Where our T.38 UDPTL should be going if not to us */
978 struct ast_udptl *udptl; /*!< T.38 UDPTL session */
979 int callingpres; /*!< Calling presentation */
980 int authtries; /*!< Times we've tried to authenticate */
981 int expiry; /*!< How long we take to expire */
982 long branch; /*!< The branch identifier of this session */
983 char tag[11]; /*!< Our tag for this session */
984 int sessionid; /*!< SDP Session ID */
985 int sessionversion; /*!< SDP Session Version */
986 struct sockaddr_in sa; /*!< Our peer */
987 struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
988 struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
989 struct sockaddr_in tredirip; /*!< Where our Text RTP should be going if not to us */
990 time_t lastrtprx; /*!< Last RTP received */
991 time_t lastrtptx; /*!< Last RTP sent */
992 int rtptimeout; /*!< RTP timeout time */
993 struct sockaddr_in recv; /*!< Received as */
994 struct in_addr ourip; /*!< Our IP */
995 struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
996 struct sip_route *route; /*!< Head of linked list of routing steps (fm Record-Route) */
997 int route_persistant; /*!< Is this the "real" route? */
998 struct sip_auth *peerauth; /*!< Realm authentication */
999 int noncecount; /*!< Nonce-count */
1000 char lastmsg[256]; /*!< Last Message sent/received */
1001 int amaflags; /*!< AMA Flags */
1002 int pendinginvite; /*!< Any pending invite ? (seqno of this) */
1003 struct sip_request initreq; /*!< Latest request that opened a new transaction
1005 NOT the request that opened the dialog
1008 int initid; /*!< Auto-congest ID if appropriate (scheduler) */
1009 int autokillid; /*!< Auto-kill ID (scheduler) */
1010 enum transfermodes allowtransfer; /*!< REFER: restriction scheme */
1011 struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
1012 enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
1013 int stateid; /*!< SUBSCRIBE: ID for devicestate subscriptions */
1014 int laststate; /*!< SUBSCRIBE: Last known extension state */
1015 int dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */
1017 struct ast_dsp *vad; /*!< Inband DTMF Detection dsp */
1019 struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one
1020 Used in peerpoke, mwi subscriptions */
1021 struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */
1022 struct ast_rtp *rtp; /*!< RTP Session */
1023 struct ast_rtp *vrtp; /*!< Video RTP session */
1024 struct ast_rtp *trtp; /*!< Text RTP session */
1025 struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
1026 struct sip_history_head *history; /*!< History of this SIP dialog */
1027 struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
1028 struct sip_pvt *next; /*!< Next dialog in chain */
1029 struct sip_invite_param *options; /*!< Options for INVITE */
1030 int autoframing; /*!< The number of Asters we group in a Pyroflax
1031 before strolling to the Grokyzpå
1032 (A bit unsure of this, please correct if
1036 static struct sip_pvt *dialoglist = NULL;
1038 /*! \brief Protect the SIP dialog list (of sip_pvt's) */
1039 AST_MUTEX_DEFINE_STATIC(dialoglock);
1041 /*! \brief hide the way the list is locked/unlocked */
1042 static void dialoglist_lock(void)
1044 ast_mutex_lock(&dialoglock);
1047 static void dialoglist_unlock(void)
1049 ast_mutex_unlock(&dialoglock);
1052 #define FLAG_RESPONSE (1 << 0)
1053 #define FLAG_FATAL (1 << 1)
1055 /*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission */
1057 struct sip_pkt *next; /*!< Next packet in linked list */
1058 int retrans; /*!< Retransmission number */
1059 int method; /*!< SIP method for this packet */
1060 int seqno; /*!< Sequence number */
1061 unsigned int flags; /*!< non-zero if this is a response packet (e.g. 200 OK) */
1062 struct sip_pvt *owner; /*!< Owner AST call */
1063 int retransid; /*!< Retransmission ID */
1064 int timer_a; /*!< SIP timer A, retransmission timer */
1065 int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */
1066 int packetlen; /*!< Length of packet */
1070 /*! \brief Structure for SIP user data. User's place calls to us */
1072 /* Users who can access various contexts */
1073 ASTOBJ_COMPONENTS(struct sip_user);
1074 char secret[80]; /*!< Password */
1075 char md5secret[80]; /*!< Password in md5 */
1076 char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
1077 char subscribecontext[AST_MAX_CONTEXT]; /* Default context for subscriptions */
1078 char cid_num[80]; /*!< Caller ID num */
1079 char cid_name[80]; /*!< Caller ID name */
1080 char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */
1081 char language[MAX_LANGUAGE]; /*!< Default language for this user */
1082 char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
1083 char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
1084 char useragent[256]; /*!< User agent in SIP request */
1085 struct ast_codec_pref prefs; /*!< codec prefs */
1086 ast_group_t callgroup; /*!< Call group */
1087 ast_group_t pickupgroup; /*!< Pickup Group */
1088 unsigned int sipoptions; /*!< Supported SIP options */
1089 struct ast_flags flags[2]; /*!< SIP_ flags */
1090 int amaflags; /*!< AMA flags for billing */
1091 int callingpres; /*!< Calling id presentation */
1092 int capability; /*!< Codec capability */
1093 int inUse; /*!< Number of calls in use */
1094 int call_limit; /*!< Limit of concurrent calls */
1095 enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
1096 struct ast_ha *ha; /*!< ACL setting */
1097 struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
1098 int maxcallbitrate; /*!< Maximum Bitrate for a video call */
1102 /*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host) */
1103 /* XXX field 'name' must be first otherwise sip_addrcmp() will fail */
1105 ASTOBJ_COMPONENTS(struct sip_peer); /*!< name, refcount, objflags, object pointers */
1106 /*!< peer->name is the unique name of this object */
1107 char secret[80]; /*!< Password */
1108 char md5secret[80]; /*!< Password in MD5 */
1109 struct sip_auth *auth; /*!< Realm authentication list */
1110 char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
1111 char subscribecontext[AST_MAX_CONTEXT]; /*!< Default context for subscriptions */
1112 char username[80]; /*!< Temporary username until registration */
1113 char accountcode[AST_MAX_ACCOUNT_CODE]; /*!< Account code */
1114 int amaflags; /*!< AMA Flags (for billing) */
1115 char tohost[MAXHOSTNAMELEN]; /*!< If not dynamic, IP address */
1116 char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */
1117 char fromuser[80]; /*!< From: user when calling this peer */
1118 char fromdomain[MAXHOSTNAMELEN]; /*!< From: domain when calling this peer */
1119 char fullcontact[256]; /*!< Contact registered with us (not in sip.conf) */
1120 char cid_num[80]; /*!< Caller ID num */
1121 char cid_name[80]; /*!< Caller ID name */
1122 int callingpres; /*!< Calling id presentation */
1123 int inUse; /*!< Number of calls in use */
1124 int inRinging; /*!< Number of calls ringing */
1125 int onHold; /*!< Peer has someone on hold */
1126 int call_limit; /*!< Limit of concurrent calls */
1127 int busy_level; /*!< Level of active channels where we signal busy */
1128 enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
1129 char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
1130 char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
1131 char language[MAX_LANGUAGE]; /*!< Default language for prompts */
1132 char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
1133 char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
1134 char useragent[256]; /*!< User agent in SIP request (saved from registration) */
1135 struct ast_codec_pref prefs; /*!< codec prefs */
1137 time_t lastmsgcheck; /*!< Last time we checked for MWI */
1138 unsigned int sipoptions; /*!< Supported SIP options */
1139 struct ast_flags flags[2]; /*!< SIP_ flags */
1140 int expire; /*!< When to expire this peer registration */
1141 int capability; /*!< Codec capability */
1142 int rtptimeout; /*!< RTP timeout */
1143 int rtpholdtimeout; /*!< RTP Hold Timeout */
1144 int rtpkeepalive; /*!< Send RTP packets for keepalive */
1145 ast_group_t callgroup; /*!< Call group */
1146 ast_group_t pickupgroup; /*!< Pickup group */
1147 struct sip_proxy *outboundproxy; /*!< Outbound proxy for this peer */
1148 struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
1149 struct sockaddr_in addr; /*!< IP address of peer */
1150 int maxcallbitrate; /*!< Maximum Bitrate for a video call */
1153 struct sip_pvt *call; /*!< Call pointer */
1154 int pokeexpire; /*!< When to expire poke (qualify= checking) */
1155 int lastms; /*!< How long last response took (in ms), or -1 for no response */
1156 int maxms; /*!< Max ms we will accept for the host to be up, 0 to not monitor */
1157 struct timeval ps; /*!< Time for sending SIP OPTION in sip_pke_peer() */
1158 struct sockaddr_in defaddr; /*!< Default IP address, used until registration */
1159 struct ast_ha *ha; /*!< Access control list */
1160 struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
1161 struct sip_pvt *mwipvt; /*!< Subscription for MWI */
1167 /*! \brief Registrations with other SIP proxies */
1168 struct sip_registry {
1169 ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1);
1170 AST_DECLARE_STRING_FIELDS(
1171 AST_STRING_FIELD(callid); /*!< Global Call-ID */
1172 AST_STRING_FIELD(realm); /*!< Authorization realm */
1173 AST_STRING_FIELD(nonce); /*!< Authorization nonce */
1174 AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
1175 AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
1176 AST_STRING_FIELD(domain); /*!< Authorization domain */
1177 AST_STRING_FIELD(username); /*!< Who we are registering as */
1178 AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */
1179 AST_STRING_FIELD(hostname); /*!< Domain or host we register to */
1180 AST_STRING_FIELD(secret); /*!< Password in clear text */
1181 AST_STRING_FIELD(md5secret); /*!< Password in md5 */
1182 AST_STRING_FIELD(callback); /*!< Contact extension */
1183 AST_STRING_FIELD(random);
1185 int portno; /*!< Optional port override */
1186 int expire; /*!< Sched ID of expiration */
1187 int expiry; /*!< Value to use for the Expires header */
1188 int regattempts; /*!< Number of attempts (since the last success) */
1189 int timeout; /*!< sched id of sip_reg_timeout */
1190 int refresh; /*!< How often to refresh */
1191 struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
1192 enum sipregistrystate regstate; /*!< Registration state (see above) */
1193 time_t regtime; /*!< Last successful registration time */
1194 int callid_valid; /*!< 0 means we haven't chosen callid for this registry yet. */
1195 unsigned int ocseq; /*!< Sequence number we got to for REGISTERs for this registry */
1196 struct sockaddr_in us; /*!< Who the server thinks we are */
1197 int noncecount; /*!< Nonce-count */
1198 char lastmsg[256]; /*!< Last Message sent/received */
1201 /* --- Linked lists of various objects --------*/
1203 /*! \brief The user list: Users and friends */
1204 static struct ast_user_list {
1205 ASTOBJ_CONTAINER_COMPONENTS(struct sip_user);
1208 /*! \brief The peer list: Peers and Friends */
1209 static struct ast_peer_list {
1210 ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer);
1213 /*! \brief The register list: Other SIP proxies we register with and place calls to */
1214 static struct ast_register_list {
1215 ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
1219 static int temp_pvt_init(void *);
1220 static void temp_pvt_cleanup(void *);
1222 /*! \brief A per-thread temporary pvt structure */
1223 AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
1225 /*! \todo Move the sip_auth list to AST_LIST */
1226 static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */
1229 /* --- Sockets and networking --------------*/
1230 static int sipsock = -1; /*!< Main socket for SIP network communication */
1231 static struct sockaddr_in bindaddr = { 0, }; /*!< The address we bind to */
1232 static struct sockaddr_in externip; /*!< External IP address if we are behind NAT */
1233 static char externhost[MAXHOSTNAMELEN]; /*!< External host name (possibly with dynamic DNS and DHCP */
1234 static time_t externexpire = 0; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
1235 static int externrefresh = 10;
1236 static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
1237 static struct in_addr __ourip;
1239 static struct sockaddr_in debugaddr;
1241 static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
1243 /*---------------------------- Forward declarations of functions in chan_sip.c */
1244 /*! \note This is added to help splitting up chan_sip.c into several files
1245 in coming releases */
1247 /*--- PBX interface functions */
1248 static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
1249 static int sip_devicestate(void *data);
1250 static int sip_sendtext(struct ast_channel *ast, const char *text);
1251 static int sip_call(struct ast_channel *ast, char *dest, int timeout);
1252 static int sip_hangup(struct ast_channel *ast);
1253 static int sip_answer(struct ast_channel *ast);
1254 static struct ast_frame *sip_read(struct ast_channel *ast);
1255 static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
1256 static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
1257 static int sip_transfer(struct ast_channel *ast, const char *dest);
1258 static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
1259 static int sip_senddigit_begin(struct ast_channel *ast, char digit);
1260 static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration);
1262 /*--- Transmitting responses and requests */
1263 static int sipsock_read(int *id, int fd, short events, void *ignore);
1264 static int __sip_xmit(struct sip_pvt *p, char *data, int len);
1265 static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod);
1266 static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1267 static int retrans_pkt(void *data);
1268 static int transmit_sip_request(struct sip_pvt *p, struct sip_request *req);
1269 static int transmit_response_using_temp(ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
1270 static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1271 static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1272 static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1273 static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1274 static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
1275 static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
1276 static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1277 static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, int reliable);
1278 static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
1279 static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
1280 static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init);
1281 static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version);
1282 static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration);
1283 static int transmit_info_with_vidupdate(struct sip_pvt *p);
1284 static int transmit_message_with_text(struct sip_pvt *p, const char *text);
1285 static int transmit_refer(struct sip_pvt *p, const char *dest);
1286 static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten);
1287 static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
1288 static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
1289 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
1290 static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
1291 static void copy_request(struct sip_request *dst, const struct sip_request *src);
1292 static void receive_message(struct sip_pvt *p, struct sip_request *req);
1293 static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req);
1294 static int sip_send_mwi_to_peer(struct sip_peer *peer);
1295 static int does_peer_need_mwi(struct sip_peer *peer);
1297 /*--- Dialog management */
1298 static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
1299 int useglobal_nat, const int intended_method);
1300 static int __sip_autodestruct(void *data);
1301 static void sip_scheddestroy(struct sip_pvt *p, int ms);
1302 static void sip_cancel_destroy(struct sip_pvt *p);
1303 static void sip_destroy(struct sip_pvt *p);
1304 static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist);
1305 static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
1306 static void __sip_pretend_ack(struct sip_pvt *p);
1307 static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
1308 static int auto_congest(void *nothing);
1309 static int update_call_counter(struct sip_pvt *fup, int event);
1310 static int hangup_sip2cause(int cause);
1311 static const char *hangup_cause2sip(int cause);
1312 static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method);
1313 static void free_old_route(struct sip_route *route);
1314 static void list_route(struct sip_route *route);
1315 static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards);
1316 static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin,
1317 struct sip_request *req, char *uri);
1318 static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
1319 static void check_pendings(struct sip_pvt *p);
1320 static void *sip_park_thread(void *stuff);
1321 static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno);
1322 static int sip_sipredirect(struct sip_pvt *p, const char *dest);
1324 /*--- Codec handling / SDP */
1325 static void try_suggested_sip_codec(struct sip_pvt *p);
1326 static const char* get_sdp_iterate(int* start, struct sip_request *req, const char *name);
1327 static const char *get_sdp(struct sip_request *req, const char *name);
1328 static int find_sdp(struct sip_request *req);
1329 static int process_sdp(struct sip_pvt *p, struct sip_request *req);
1330 static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
1331 char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
1332 int debug, int *min_packet_size);
1333 static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
1334 char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
1336 static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p);
1337 static void do_setnat(struct sip_pvt *p, int natflags);
1339 /*--- Authentication stuff */
1340 static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
1341 static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
1342 static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
1343 const char *secret, const char *md5secret, int sipmethod,
1344 char *uri, enum xmittype reliable, int ignore);
1345 static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
1346 int sipmethod, char *uri, enum xmittype reliable,
1347 struct sockaddr_in *sin, struct sip_peer **authpeer);
1348 static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);
1350 /*--- Domain handling */
1351 static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
1352 static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context);
1353 static void clear_sip_domains(void);
1355 /*--- SIP realm authentication */
1356 static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno);
1357 static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
1358 static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm);
1360 /*--- Misc functions */
1361 static int sip_do_reload(enum channelreloadreason reason);
1362 static int reload_config(enum channelreloadreason reason);
1363 static int expire_register(void *data);
1364 static void *do_monitor(void *data);
1365 static int restart_monitor(void);
1366 static int sip_send_mwi_to_peer(struct sip_peer *peer);
1367 static void sip_destroy(struct sip_pvt *p);
1368 static int sip_addrcmp(char *name, struct sockaddr_in *sin); /* Support for peer matching */
1369 static int sip_refer_allocate(struct sip_pvt *p);
1370 static void ast_quiet_chan(struct ast_channel *chan);
1371 static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
1373 /*--- Device monitoring and Device/extension state handling */
1374 static int cb_extensionstate(char *context, char* exten, int state, void *data);
1375 static int sip_devicestate(void *data);
1376 static int sip_poke_noanswer(void *data);
1377 static int sip_poke_peer(struct sip_peer *peer);
1378 static void sip_poke_all_peers(void);
1379 static void sip_peer_hold(struct sip_pvt *p, int hold);
1381 /*--- Applications, functions, CLI and manager command helpers */
1382 static const char *sip_nat_mode(const struct sip_pvt *p);
1383 static int sip_show_inuse(int fd, int argc, char *argv[]);
1384 static char *transfermode2str(enum transfermodes mode) attribute_const;
1385 static char *nat2str(int nat) attribute_const;
1386 static int peer_status(struct sip_peer *peer, char *status, int statuslen);
1387 static int sip_show_users(int fd, int argc, char *argv[]);
1388 static int _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]);
1389 static int sip_show_peers(int fd, int argc, char *argv[]);
1390 static int sip_show_objects(int fd, int argc, char *argv[]);
1391 static void print_group(int fd, ast_group_t group, int crlf);
1392 static const char *dtmfmode2str(int mode) attribute_const;
1393 static const char *insecure2str(int port, int invite) attribute_const;
1394 static void cleanup_stale_contexts(char *new, char *old);
1395 static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
1396 static const char *domain_mode_to_text(const enum domain_mode mode);
1397 static int sip_show_domains(int fd, int argc, char *argv[]);
1398 static int _sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
1399 static int sip_show_peer(int fd, int argc, char *argv[]);
1400 static int sip_show_user(int fd, int argc, char *argv[]);
1401 static int sip_show_registry(int fd, int argc, char *argv[]);
1402 static int sip_show_settings(int fd, int argc, char *argv[]);
1403 static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
1404 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
1405 static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
1406 static int sip_show_channels(int fd, int argc, char *argv[]);
1407 static int sip_show_subscriptions(int fd, int argc, char *argv[]);
1408 static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
1409 static char *complete_sipch(const char *line, const char *word, int pos, int state);
1410 static char *complete_sip_peer(const char *word, int state, int flags2);
1411 static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
1412 static char *complete_sip_debug_peer(const char *line, const char *word, int pos, int state);
1413 static char *complete_sip_user(const char *word, int state, int flags2);
1414 static char *complete_sip_show_user(const char *line, const char *word, int pos, int state);
1415 static char *complete_sipnotify(const char *line, const char *word, int pos, int state);
1416 static char *complete_sip_prune_realtime_peer(const char *line, const char *word, int pos, int state);
1417 static char *complete_sip_prune_realtime_user(const char *line, const char *word, int pos, int state);
1418 static int sip_show_channel(int fd, int argc, char *argv[]);
1419 static int sip_show_history(int fd, int argc, char *argv[]);
1420 static int sip_do_debug_ip(int fd, int argc, char *argv[]);
1421 static int sip_do_debug_peer(int fd, int argc, char *argv[]);
1422 static int sip_do_debug(int fd, int argc, char *argv[]);
1423 static int sip_no_debug(int fd, int argc, char *argv[]);
1424 static int sip_notify(int fd, int argc, char *argv[]);
1425 static int sip_do_history(int fd, int argc, char *argv[]);
1426 static int sip_no_history(int fd, int argc, char *argv[]);
1427 static int sip_dtmfmode(struct ast_channel *chan, void *data);
1428 static int sip_addheader(struct ast_channel *chan, void *data);
1429 static int sip_do_reload(enum channelreloadreason reason);
1430 static int sip_reload(int fd, int argc, char *argv[]);
1433 Functions for enabling debug per IP or fully, or enabling history logging for
1436 static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to LOG_DEBUG at end of dialog, before destroying data */
1437 static inline int sip_debug_test_addr(const struct sockaddr_in *addr);
1438 static inline int sip_debug_test_pvt(struct sip_pvt *p);
1439 static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
1440 static void sip_dump_history(struct sip_pvt *dialog);
1442 /*--- Device object handling */
1443 static struct sip_peer *temp_peer(const char *name);
1444 static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime);
1445 static struct sip_user *build_user(const char *name, struct ast_variable *v, int realtime);
1446 static int update_call_counter(struct sip_pvt *fup, int event);
1447 static void sip_destroy_peer(struct sip_peer *peer);
1448 static void sip_destroy_user(struct sip_user *user);
1449 static int sip_poke_peer(struct sip_peer *peer);
1450 static void set_peer_defaults(struct sip_peer *peer);
1451 static struct sip_peer *temp_peer(const char *name);
1452 static void register_peer_exten(struct sip_peer *peer, int onoff);
1453 static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime);
1454 static struct sip_user *find_user(const char *name, int realtime);
1455 static int sip_poke_peer_s(void *data);
1456 static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
1457 static void reg_source_db(struct sip_peer *peer);
1458 static void destroy_association(struct sip_peer *peer);
1459 static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);
1461 /* Realtime device support */
1462 static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey);
1463 static struct sip_user *realtime_user(const char *username);
1464 static void update_peer(struct sip_peer *p, int expiry);
1465 static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin);
1466 static int sip_prune_realtime(int fd, int argc, char *argv[]);
1468 /*--- Internal UA client handling (outbound registrations) */
1469 static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us);
1470 static void sip_registry_destroy(struct sip_registry *reg);
1471 static int sip_register(char *value, int lineno);
1472 static char *regstate2str(enum sipregistrystate regstate) attribute_const;
1473 static int sip_reregister(void *data);
1474 static int __sip_do_register(struct sip_registry *r);
1475 static int sip_reg_timeout(void *data);
1476 static void sip_send_all_registers(void);
1478 /*--- Parsing SIP requests and responses */
1479 static void append_date(struct sip_request *req); /* Append date to SIP packet */
1480 static int determine_firstline_parts(struct sip_request *req);
1481 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
1482 static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
1483 static int find_sip_method(const char *msg);
1484 static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported);
1485 static void parse_request(struct sip_request *req);
1486 static const char *get_header(const struct sip_request *req, const char *name);
1487 static const char *referstatus2str(enum referstatus rstatus) attribute_pure;
1488 static int method_match(enum sipmethod id, const char *name);
1489 static void parse_copy(struct sip_request *dst, const struct sip_request *src);
1490 static char *get_in_brackets(char *tmp);
1491 static const char *find_alias(const char *name, const char *_default);
1492 static const char *__get_header(const struct sip_request *req, const char *name, int *start);
1493 static int lws2sws(char *msgbuf, int len);
1494 static void extract_uri(struct sip_pvt *p, struct sip_request *req);
1495 static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req);
1496 static int get_also_info(struct sip_pvt *p, struct sip_request *oreq);
1497 static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
1498 static int set_address_from_contact(struct sip_pvt *pvt);
1499 static void check_via(struct sip_pvt *p, struct sip_request *req);
1500 static char *get_calleridname(const char *input, char *output, size_t outputsize);
1501 static int get_rpid_num(const char *input, char *output, int maxlen);
1502 static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq);
1503 static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
1504 static int get_msg_text(char *buf, int len, struct sip_request *req);
1505 static void free_old_route(struct sip_route *route);
1506 static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout);
1508 /*--- Constructing requests and responses */
1509 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
1510 static int init_req(struct sip_request *req, int sipmethod, const char *recip);
1511 static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch);
1512 static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod);
1513 static int init_resp(struct sip_request *resp, const char *msg);
1514 static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req);
1515 static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p);
1516 static void build_via(struct sip_pvt *p);
1517 static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
1518 static int create_addr(struct sip_pvt *dialog, const char *opeer);
1519 static char *generate_random_string(char *buf, size_t size);
1520 static void build_callid_pvt(struct sip_pvt *pvt);
1521 static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
1522 static void make_our_tag(char *tagbuf, size_t len);
1523 static int add_header(struct sip_request *req, const char *var, const char *value);
1524 static int add_header_contentLength(struct sip_request *req, int len);
1525 static int add_line(struct sip_request *req, const char *line);
1526 static int add_text(struct sip_request *req, const char *text);
1527 static int add_digit(struct sip_request *req, char digit, unsigned int duration);
1528 static int add_vidupdate(struct sip_request *req);
1529 static void add_route(struct sip_request *req, struct sip_route *route);
1530 static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field);
1531 static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field);
1532 static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field);
1533 static void set_destination(struct sip_pvt *p, char *uri);
1534 static void append_date(struct sip_request *req);
1535 static void build_contact(struct sip_pvt *p);
1536 static void build_rpid(struct sip_pvt *p);
1538 /*------Request handling functions */
1539 static int handle_request(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock);
1540 static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e);
1541 static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, int *nounlock);
1542 static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
1543 static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e);
1544 static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
1545 static int handle_request_message(struct sip_pvt *p, struct sip_request *req);
1546 static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
1547 static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
1548 static int handle_request_options(struct sip_pvt *p, struct sip_request *req);
1549 static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin);
1550 static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
1551 static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno);
1553 /*------Response handling functions */
1554 static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1555 static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1556 static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1557 static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1559 /*----- RTP interface functions */
1560 static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active);
1561 static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
1562 static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
1563 static enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
1564 static int sip_get_codec(struct ast_channel *chan);
1565 static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
1567 /*------ T38 Support --------- */
1568 static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */
1569 static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
1570 static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
1571 static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
1573 /*! \brief Definition of this channel for PBX channel registration */
1574 static const struct ast_channel_tech sip_tech = {
1576 .description = "Session Initiation Protocol (SIP)",
1577 .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
1578 .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
1579 .requester = sip_request_call,
1580 .devicestate = sip_devicestate,
1582 .hangup = sip_hangup,
1583 .answer = sip_answer,
1586 .write_video = sip_write,
1587 .write_text = sip_write,
1588 .indicate = sip_indicate,
1589 .transfer = sip_transfer,
1591 .send_digit_begin = sip_senddigit_begin,
1592 .send_digit_end = sip_senddigit_end,
1593 .bridge = ast_rtp_bridge,
1594 .early_bridge = ast_rtp_early_bridge,
1595 .send_text = sip_sendtext,
1598 /*! \brief This version of the sip channel tech has no send_digit_begin
1599 * callback. This is for use with channels using SIP INFO DTMF so that
1600 * the core knows that the channel doesn't want DTMF BEGIN frames. */
1601 static const struct ast_channel_tech sip_tech_info = {
1603 .description = "Session Initiation Protocol (SIP)",
1604 .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
1605 .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
1606 .requester = sip_request_call,
1607 .devicestate = sip_devicestate,
1609 .hangup = sip_hangup,
1610 .answer = sip_answer,
1613 .write_video = sip_write,
1614 .indicate = sip_indicate,
1615 .transfer = sip_transfer,
1617 .send_digit_end = sip_senddigit_end,
1618 .bridge = ast_rtp_bridge,
1619 .send_text = sip_sendtext,
1622 /**--- some list management macros. **/
1624 #define UNLINK(element, head, prev) do { \
1626 (prev)->next = (element)->next; \
1628 (head) = (element)->next; \
1631 /*! \brief Interface structure with callbacks used to connect to RTP module */
1632 static struct ast_rtp_protocol sip_rtp = {
1634 get_rtp_info: sip_get_rtp_peer,
1635 get_vrtp_info: sip_get_vrtp_peer,
1636 get_trtp_info: sip_get_trtp_peer,
1637 set_rtp_peer: sip_set_rtp_peer,
1638 get_codec: sip_get_codec,
1641 /*! \brief Helper function to lock, hiding the underlying locking mechanism. */
1642 static void sip_pvt_lock(struct sip_pvt *pvt)
1644 ast_mutex_lock(&pvt->pvt_lock);
1647 /*! \brief Helper function to unlock pvt, hiding the underlying locking mechanism. */
1648 static void sip_pvt_unlock(struct sip_pvt *pvt)
1650 ast_mutex_unlock(&pvt->pvt_lock);
1654 * helper functions to unreference various types of objects.
1655 * By handling them this way, we don't have to declare the
1656 * destructor on each call, which removes the chance of errors.
1658 static void unref_peer(struct sip_peer *peer)
1660 ASTOBJ_UNREF(peer, sip_destroy_peer);
1663 static void unref_user(struct sip_user *user)
1665 ASTOBJ_UNREF(user, sip_destroy_user);
1668 static void registry_unref(struct sip_registry *reg)
1670 if (option_debug > 2)
1671 ast_log(LOG_DEBUG, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1);
1672 ASTOBJ_UNREF(reg, sip_registry_destroy);
1675 /*! \brief Add object reference to SIP registry */
1676 static struct sip_registry *registry_addref(struct sip_registry *reg)
1678 if (option_debug > 2)
1679 ast_log(LOG_DEBUG, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount + 1);
1680 return ASTOBJ_REF(reg); /* Add pointer to registry in packet */
1683 /*! \brief Interface structure with callbacks used to connect to UDPTL module*/
1684 static struct ast_udptl_protocol sip_udptl = {
1686 get_udptl_info: sip_get_udptl_peer,
1687 set_udptl_peer: sip_set_udptl_peer,
1690 /*! \brief Append to SIP dialog history
1691 \return Always returns 0 */
1692 #define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
1694 static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
1695 __attribute__ ((format (printf, 2, 3)));
1698 /*! \brief Convert transfer status to string */
1699 static const char *referstatus2str(enum referstatus rstatus)
1701 int i = (sizeof(referstatusstrings) / sizeof(referstatusstrings[0]));
1704 for (x = 0; x < i; x++) {
1705 if (referstatusstrings[x].status == rstatus)
1706 return referstatusstrings[x].text;
1711 /*! \brief Initialize the initital request packet in the pvt structure.
1712 This packet is used for creating replies and future requests in
1714 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req)
1717 if (p->initreq.headers)
1718 ast_log(LOG_DEBUG, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
1720 ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
1722 /* Use this as the basis */
1723 copy_request(&p->initreq, req);
1724 parse_request(&p->initreq);
1725 if (ast_test_flag(req, SIP_PKT_DEBUG))
1726 ast_verbose("Initreq: %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
1729 /*! \brief Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging */
1730 static void sip_alreadygone(struct sip_pvt *dialog)
1732 if (option_debug > 2)
1733 ast_log(LOG_DEBUG, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
1734 ast_set_flag(&dialog->flags[0], SIP_ALREADYGONE);
1737 /*! Resolve DNS srv name or host name in a sip_proxy structure */
1738 static int proxy_update(struct sip_proxy *proxy)
1740 /* if it's actually an IP address and not a name,
1741 there's no need for a managed lookup */
1742 if (!inet_aton(proxy->name, &proxy->ip.sin_addr)) {
1743 /* Ok, not an IP address, then let's check if it's a domain or host */
1744 /* XXX Todo - if we have proxy port, don't do SRV */
1745 if (ast_get_ip_or_srv(&proxy->ip, proxy->name, global_srvlookup ? "_sip._udp" : NULL) < 0) {
1746 ast_log(LOG_WARNING, "Unable to locate host '%s'\n", proxy->name);
1750 proxy->last_dnsupdate = time(NULL);
1754 /*! \brief Allocate and initialize sip proxy */
1755 static struct sip_proxy *proxy_allocate(char *name, char *port, int force)
1757 struct sip_proxy *proxy;
1758 proxy = ast_calloc(1, sizeof(struct sip_proxy));
1761 proxy->force = force;
1762 ast_copy_string(proxy->name, name, sizeof(proxy->name));
1763 if (!ast_strlen_zero(port))
1764 proxy->ip.sin_port = htons(atoi(port));
1765 proxy_update(proxy);
1769 /*! \brief Get default outbound proxy or global proxy */
1770 static struct sip_proxy *obproxy_get(struct sip_pvt *dialog, struct sip_peer *peer)
1772 if (peer && peer->outboundproxy) {
1773 if (option_debug && sipdebug)
1774 ast_log(LOG_DEBUG, "OBPROXY: Applying peer OBproxy to this call\n");
1775 append_history(dialog, "OBproxy", "Using peer obproxy %s", peer->outboundproxy->name);
1776 return peer->outboundproxy;
1778 if (global_outboundproxy.name[0]) {
1779 if (option_debug && sipdebug)
1780 ast_log(LOG_DEBUG, "OBPROXY: Applying global OBproxy to this call\n");
1781 append_history(dialog, "OBproxy", "Using global obproxy %s", global_outboundproxy.name);
1782 return &global_outboundproxy;
1784 if (option_debug && sipdebug)
1785 ast_log(LOG_DEBUG, "OBPROXY: Not applying OBproxy to this call\n");
1789 /*! \brief returns true if 'name' (with optional trailing whitespace)
1790 * matches the sip method 'id'.
1791 * Strictly speaking, SIP methods are case SENSITIVE, but we do
1792 * a case-insensitive comparison to be more tolerant.
1793 * following Jon Postel's rule: Be gentle in what you accept, strict with what you send
1795 static int method_match(enum sipmethod id, const char *name)
1797 int len = strlen(sip_methods[id].text);
1798 int l_name = name ? strlen(name) : 0;
1799 /* true if the string is long enough, and ends with whitespace, and matches */
1800 return (l_name >= len && name[len] < 33 &&
1801 !strncasecmp(sip_methods[id].text, name, len));
1804 /*! \brief find_sip_method: Find SIP method from header */
1805 static int find_sip_method(const char *msg)
1809 if (ast_strlen_zero(msg))
1811 for (i = 1; i < (sizeof(sip_methods) / sizeof(sip_methods[0])) && !res; i++) {
1812 if (method_match(i, msg))
1813 res = sip_methods[i].id;
1818 /*! \brief Parse supported header in incoming packet */
1819 static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported)
1823 unsigned int profile = 0;
1826 if (ast_strlen_zero(supported) )
1828 temp = ast_strdupa(supported);
1830 if (option_debug > 2 && sipdebug)
1831 ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported);
1833 for (next = temp; next; next = sep) {
1835 if ( (sep = strchr(next, ',')) != NULL)
1837 next = ast_skip_blanks(next);
1838 if (option_debug > 2 && sipdebug)
1839 ast_log(LOG_DEBUG, "Found SIP option: -%s-\n", next);
1840 for (i=0; i < (sizeof(sip_options) / sizeof(sip_options[0])); i++) {
1841 if (!strcasecmp(next, sip_options[i].text)) {
1842 profile |= sip_options[i].id;
1844 if (option_debug > 2 && sipdebug)
1845 ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next);
1849 if (!found && option_debug > 2 && sipdebug) {
1850 if (!strncasecmp(next, "x-", 2))
1851 ast_log(LOG_DEBUG, "Found private SIP option, not supported: %s\n", next);
1853 ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next);
1858 pvt->sipoptions = profile;
1862 /*! \brief See if we pass debug IP filter */
1863 static inline int sip_debug_test_addr(const struct sockaddr_in *addr)
1867 if (debugaddr.sin_addr.s_addr) {
1868 if (((ntohs(debugaddr.sin_port) != 0)
1869 && (debugaddr.sin_port != addr->sin_port))
1870 || (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
1876 /*! \brief The real destination address for a write */
1877 static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p)
1879 if (p->outboundproxy)
1880 return &p->outboundproxy->ip;
1882 return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? &p->recv : &p->sa;
1885 /*! \brief Display SIP nat mode */
1886 static const char *sip_nat_mode(const struct sip_pvt *p)
1888 return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? "NAT" : "no NAT";
1891 /*! \brief Test PVT for debugging output */
1892 static inline int sip_debug_test_pvt(struct sip_pvt *p)
1896 return sip_debug_test_addr(sip_real_dst(p));
1899 /*! \brief Transmit SIP message */
1900 static int __sip_xmit(struct sip_pvt *p, char *data, int len)
1903 const struct sockaddr_in *dst = sip_real_dst(p);
1904 res = sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
1907 ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));
1912 /*! \brief Build a Via header for a request */
1913 static void build_via(struct sip_pvt *p)
1915 /* Work around buggy UNIDEN UIP200 firmware */
1916 const char *rport = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : "";
1918 /* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */
1919 ast_string_field_build(p, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s",
1920 ast_inet_ntoa(p->ourip), ourport, p->branch, rport);
1923 /*! \brief NAT fix - decide which IP address to use for ASterisk server?
1925 * Using the localaddr structure built up with localnet statements in sip.conf
1926 * apply it to their address to see if we need to substitute our
1927 * externip or can get away with our internal bindaddr
1929 static enum sip_result ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us)
1931 struct sockaddr_in theirs, ours;
1933 /* Get our local information */
1934 ast_ouraddrfor(them, us);
1935 theirs.sin_addr = *them;
1936 ours.sin_addr = *us;
1938 if (localaddr && externip.sin_addr.s_addr &&
1939 (ast_apply_ha(localaddr, &theirs)) &&
1940 (!global_matchexterniplocally || !ast_apply_ha(localaddr, &ours))) {
1941 if (externexpire && time(NULL) >= externexpire) {
1942 struct ast_hostent ahp;
1945 externexpire = time(NULL) + externrefresh;
1946 if ((hp = ast_gethostbyname(externhost, &ahp))) {
1947 memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr));
1949 ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
1951 *us = externip.sin_addr;
1953 ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n",
1954 ast_inet_ntoa(*(struct in_addr *)&them->s_addr));
1956 } else if (bindaddr.sin_addr.s_addr)
1957 *us = bindaddr.sin_addr;
1961 /*! \brief Append to SIP dialog history with arg list */
1962 static void append_history_va(struct sip_pvt *p, const char *fmt, va_list ap)
1964 char buf[80], *c = buf; /* max history length */
1965 struct sip_history *hist;
1968 vsnprintf(buf, sizeof(buf), fmt, ap);
1969 strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
1970 l = strlen(buf) + 1;
1971 if (!(hist = ast_calloc(1, sizeof(*hist) + l)))
1973 if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
1977 memcpy(hist->event, buf, l);
1978 AST_LIST_INSERT_TAIL(p->history, hist, list);
1981 /*! \brief Append to SIP dialog history with arg list */
1982 static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
1989 append_history_va(p, fmt, ap);
1995 /*! \brief Retransmit SIP message if no answer (Called from scheduler) */
1996 static int retrans_pkt(void *data)
1998 struct sip_pkt *pkt = data, *prev, *cur = NULL;
1999 int reschedule = DEFAULT_RETRANS;
2001 /* Lock channel PVT */
2002 sip_pvt_lock(pkt->owner);
2004 if (pkt->retrans < MAX_RETRANS) {
2006 if (!pkt->timer_t1) { /* Re-schedule using timer_a and timer_t1 */
2007 if (sipdebug && option_debug > 3)
2008 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);
2012 if (sipdebug && option_debug > 3)
2013 ast_log(LOG_DEBUG, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method);
2017 pkt->timer_a = 2 * pkt->timer_a;
2019 /* For non-invites, a maximum of 4 secs */
2020 siptimer_a = pkt->timer_t1 * pkt->timer_a; /* Double each time */
2021 if (pkt->method != SIP_INVITE && siptimer_a > 4000)
2024 /* Reschedule re-transmit */
2025 reschedule = siptimer_a;
2026 if (option_debug > 3)
2027 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);
2030 if (sip_debug_test_pvt(pkt->owner)) {
2031 const struct sockaddr_in *dst = sip_real_dst(pkt->owner);
2032 ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n",
2033 pkt->retrans, sip_nat_mode(pkt->owner),
2034 ast_inet_ntoa(dst->sin_addr),
2035 ntohs(dst->sin_port), pkt->data);
2038 append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data);
2039 __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
2040 sip_pvt_unlock(pkt->owner);
2043 /* Too many retries */
2044 if (pkt->owner && pkt->method != SIP_OPTIONS) {
2045 if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */
2046 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");
2048 if ((pkt->method == SIP_OPTIONS) && sipdebug)
2049 ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid);
2051 append_history(pkt->owner, "MaxRetries", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
2053 pkt->retransid = -1;
2055 if (ast_test_flag(pkt, FLAG_FATAL)) {
2056 while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
2057 sip_pvt_unlock(pkt->owner); /* SIP_PVT, not channel */
2059 sip_pvt_lock(pkt->owner);
2061 if (pkt->owner->owner) {
2062 sip_alreadygone(pkt->owner);
2063 ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
2064 ast_queue_hangup(pkt->owner->owner);
2065 ast_channel_unlock(pkt->owner->owner);
2067 /* If no channel owner, destroy now */
2069 /* Let the peerpoke system expire packets when the timer expires for poke_noanswer */
2070 if (pkt->method != SIP_OPTIONS && pkt->method != SIP_REGISTER)
2071 ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY);
2074 /* Remove the packet */
2075 for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
2077 UNLINK(cur, pkt->owner->packets, prev);
2078 sip_pvt_unlock(pkt->owner);
2084 ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
2085 sip_pvt_unlock(pkt->owner);
2089 /*! \brief Transmit packet with retransmits
2090 \return 0 on success, -1 on failure to allocate packet
2092 static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod)
2094 struct sip_pkt *pkt;
2095 int siptimer_a = DEFAULT_RETRANS;
2097 if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
2099 memcpy(pkt->data, data, len);
2100 pkt->method = sipmethod;
2101 pkt->packetlen = len;
2102 pkt->next = p->packets;
2106 ast_set_flag(pkt, FLAG_RESPONSE);
2107 pkt->data[len] = '\0';
2108 pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */
2110 ast_set_flag(pkt, FLAG_FATAL);
2112 siptimer_a = pkt->timer_t1 * 2;
2114 /* Schedule retransmission */
2115 pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1);
2116 if (option_debug > 3 && sipdebug)
2117 ast_log(LOG_DEBUG, "*** SIP TIMER: Initalizing retransmit timer on packet: Id #%d\n", pkt->retransid);
2118 pkt->next = p->packets;
2121 __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */
2122 if (sipmethod == SIP_INVITE) {
2123 /* Note this is a pending invite */
2124 p->pendinginvite = seqno;
2129 /*! \brief Kill a SIP dialog (called by scheduler) */
2130 static int __sip_autodestruct(void *data)
2132 struct sip_pvt *p = data;
2134 /* If this is a subscription, tell the phone that we got a timeout */
2135 if (p->subscribed) {
2136 transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, TRUE); /* Send last notification */
2137 p->subscribed = NONE;
2138 append_history(p, "Subscribestatus", "timeout");
2139 if (option_debug > 2)
2140 ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP subsription %s\n", p->callid ? p->callid : "<unknown>");
2141 return 10000; /* Reschedule this destruction so that we know that it's gone */
2144 if (p->subscribed == MWI_NOTIFICATION)
2146 unref_peer(p->relatedpeer); /* Remove link to peer. If it's realtime, make sure it's gone from memory) */
2148 /* Reset schedule ID */
2152 ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
2153 ast_queue_hangup(p->owner);
2154 } else if (p->refer) {
2155 if (option_debug > 2)
2156 ast_log(LOG_DEBUG, "Finally hanging up channel after transfer: %s\n", p->callid);
2157 transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
2158 append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
2159 sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
2161 append_history(p, "AutoDestroy", "%s", p->callid);
2163 ast_log(LOG_DEBUG, "Auto destroying SIP dialog '%s'\n", p->callid);
2164 sip_destroy(p); /* Go ahead and destroy dialog. All attempts to recover is done */
2169 /*! \brief Schedule destruction of SIP dialog */
2170 static void sip_scheddestroy(struct sip_pvt *p, int ms)
2173 if (p->timer_t1 == 0)
2174 p->timer_t1 = SIP_TIMER_T1; /* Set timer T1 if not set (RFC 3261) */
2175 ms = p->timer_t1 * 64;
2177 if (sip_debug_test_pvt(p))
2178 ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
2179 if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
2180 append_history(p, "SchedDestroy", "%d ms", ms);
2182 if (p->autokillid > -1)
2183 ast_sched_del(sched, p->autokillid);
2184 p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, p);
2187 /*! \brief Cancel destruction of SIP dialog */
2188 static void sip_cancel_destroy(struct sip_pvt *p)
2190 if (p->autokillid > -1) {
2191 ast_sched_del(sched, p->autokillid);
2192 append_history(p, "CancelDestroy", "");
2197 /*! \brief Acknowledges receipt of a packet and stops retransmission */
2198 static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
2200 struct sip_pkt *cur, *prev = NULL;
2201 const char *msg = "Not Found"; /* used only for debugging */
2205 /* If we have an outbound proxy for this dialog, then delete it now since
2206 the rest of the requests in this dialog needs to follow the routing.
2207 If obforcing is set, we will keep the outbound proxy during the whole
2208 dialog, regardless of what the SIP rfc says
2210 if (p->outboundproxy && !p->outboundproxy->force)
2211 p->outboundproxy = NULL;
2213 for (cur = p->packets; cur; prev = cur, cur = cur->next) {
2214 if (cur->seqno != seqno || ast_test_flag(cur, FLAG_RESPONSE) != resp)
2216 if (ast_test_flag(cur, FLAG_RESPONSE) || cur->method == sipmethod) {
2218 if (!resp && (seqno == p->pendinginvite)) {
2220 ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
2221 p->pendinginvite = 0;
2223 if (cur->retransid > -1) {
2224 if (sipdebug && option_debug > 3)
2225 ast_log(LOG_DEBUG, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
2226 ast_sched_del(sched, cur->retransid);
2227 cur->retransid = -1;
2229 UNLINK(cur, p->packets, prev);
2236 ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n",
2237 p->callid, resp ? "Response" : "Request", seqno, msg);
2240 /*! \brief Pretend to ack all packets
2241 * maybe the lock on p is not strictly necessary but there might be a race */
2242 static void __sip_pretend_ack(struct sip_pvt *p)
2244 struct sip_pkt *cur = NULL;
2246 while (p->packets) {
2248 if (cur == p->packets) {
2249 ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
2253 method = (cur->method) ? cur->method : find_sip_method(cur->data);
2254 __sip_ack(p, cur->seqno, ast_test_flag(cur, FLAG_RESPONSE), method);
2258 /*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
2259 static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
2261 struct sip_pkt *cur;
2264 for (cur = p->packets; cur; cur = cur->next) {
2265 if (cur->seqno == seqno && ast_test_flag(cur, FLAG_RESPONSE) == resp &&
2266 (ast_test_flag(cur, FLAG_RESPONSE) || method_match(sipmethod, cur->data))) {
2267 /* this is our baby */
2268 if (cur->retransid > -1) {
2269 if (option_debug > 3 && sipdebug)
2270 ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
2271 ast_sched_del(sched, cur->retransid);
2272 cur->retransid = -1;
2279 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");
2284 /*! \brief Copy SIP request, parse it */
2285 static void parse_copy(struct sip_request *dst, const struct sip_request *src)
2287 memset(dst, 0, sizeof(*dst));
2288 memcpy(dst->data, src->data, sizeof(dst->data));
2289 dst->len = src->len;
2293 /*! \brief add a blank line if no body */
2294 static void add_blank(struct sip_request *req)
2297 /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
2298 snprintf(req->data + req->len, sizeof(req->data) - req->len, "\r\n");
2299 req->len += strlen(req->data + req->len);
2303 /*! \brief Transmit response on SIP request*/
2304 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
2309 if (sip_debug_test_pvt(p)) {
2310 const struct sockaddr_in *dst = sip_real_dst(p);
2312 ast_verbose("\n<--- %sTransmitting (%s) to %s:%d --->\n%s\n<------------>\n",
2313 reliable ? "Reliably " : "", sip_nat_mode(p),
2314 ast_inet_ntoa(dst->sin_addr),
2315 ntohs(dst->sin_port), req->data);
2317 if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
2318 struct sip_request tmp;
2319 parse_copy(&tmp, req);
2320 append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"),
2321 (tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? tmp.rlPart2 : sip_methods[tmp.method].text);
2324 __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
2325 __sip_xmit(p, req->data, req->len);
2331 /*! \brief Send SIP Request to the other part of the dialogue */
2332 static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
2336 /* If we have an outbound proxy, reset peer address
2339 if (p->outboundproxy) {
2340 p->sa = p->outboundproxy->ip;
2344 if (sip_debug_test_pvt(p)) {
2345 if (ast_test_flag(&p->flags[0], SIP_NAT_ROUTE))
2346 ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port), req->data);
2348 ast_verbose("%sTransmitting (no NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->sa.sin_addr), ntohs(p->sa.sin_port), req->data);
2350 if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
2351 struct sip_request tmp;
2352 parse_copy(&tmp, req);
2353 append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
2356 __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1), req->method) :
2357 __sip_xmit(p, req->data, req->len);
2361 /*! \brief Locate closing quote in a string, skipping escaped quotes.
2362 * optionally with a limit on the search.
2363 * start must be past the first quote.
2365 static const char *find_closing_quote(const char *start, const char *lim)
2367 char last_char = '\0';
2369 for (s = start; *s && s != lim; last_char = *s++) {
2370 if (*s == '"' && last_char != '\\')
2376 /*! \brief Pick out text in brackets from character string
2377 \return pointer to terminated stripped string
2378 \param tmp input string that will be modified
2381 "foo" <bar> valid input, returns bar
2382 foo returns the whole string
2383 < "foo ... > returns the string between brackets
2384 < "foo... bogus (missing closing bracket), returns the whole string
2385 XXX maybe should still skip the opening bracket
2387 static char *get_in_brackets(char *tmp)
2389 const char *parse = tmp;
2390 char *first_bracket;
2393 * Skip any quoted text until we find the part in brackets.
2394 * On any error give up and return the full string.
2396 while ( (first_bracket = strchr(parse, '<')) ) {
2397 char *first_quote = strchr(parse, '"');
2399 if (!first_quote || first_quote > first_bracket)
2400 break; /* no need to look at quoted part */
2401 /* the bracket is within quotes, so ignore it */
2402 parse = find_closing_quote(first_quote + 1, NULL);
2403 if (!*parse) { /* not found, return full string ? */
2404 /* XXX or be robust and return in-bracket part ? */
2405 ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp);
2410 if (first_bracket) {
2411 char *second_bracket = strchr(first_bracket + 1, '>');
2412 if (second_bracket) {
2413 *second_bracket = '\0';
2414 tmp = first_bracket + 1;
2416 ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp);
2423 * parses a URI in its components.
2424 * If scheme is specified, drop it from the top.
2425 * If a component is not requested, do not split around it.
2426 * This means that if we don't have domain, we cannot split
2427 * name:pass and domain:port.
2428 * It is safe to call with ret_name, pass, domain, port
2429 * pointing all to the same place.
2430 * Init pointers to empty string so we never get NULL dereferencing.
2431 * Overwrites the string.
2432 * return 0 on success, other values on error.
2434 static int parse_uri(char *uri, char *scheme,
2435 char **ret_name, char **pass, char **domain, char **port, char **options)
2440 /* init field as required */
2445 name = strsep(&uri, ";"); /* remove options */
2447 int l = strlen(scheme);
2448 if (!strncmp(name, scheme, l))
2451 ast_log(LOG_NOTICE, "Missing scheme '%s' in '%s'\n", scheme, name);
2456 /* if we don't want to split around domain, keep everything as a name,
2457 * so we need to do nothing here, except remember why.
2460 /* store the result in a temp. variable to avoid it being
2461 * overwritten if arguments point to the same place.
2465 if ((c = strchr(name, '@')) == NULL) {
2466 /* domain-only URI, according to the SIP RFC. */
2473 if (port && (c = strchr(dom, ':'))) { /* Remove :port */
2477 if (pass && (c = strchr(name, ':'))) { /* user:password */
2483 if (ret_name) /* same as for domain, store the result only at the end */
2486 *options = uri ? uri : "";
2491 /*! \brief Send SIP MESSAGE text within a call
2492 Called from PBX core sendtext() application */
2493 static int sip_sendtext(struct ast_channel *ast, const char *text)
2495 struct sip_pvt *p = ast->tech_pvt;
2496 int debug = sip_debug_test_pvt(p);
2499 ast_verbose("Sending text %s on %s\n", text, ast->name);
2502 if (ast_strlen_zero(text))
2505 ast_verbose("Really sending text %s on %s\n", text, ast->name);
2506 transmit_message_with_text(p, text);
2510 /*! \brief Update peer object in realtime storage
2511 If the Asterisk system name is set in asterisk.conf, we will use
2512 that name and store that in the "regserver" field in the sippeers
2513 table to facilitate multi-server setups.
2515 static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey)
2518 char ipaddr[INET_ADDRSTRLEN];
2519 char regseconds[20];
2520 char *tablename = NULL;
2522 char *sysname = ast_config_AST_SYSTEM_NAME;
2523 char *syslabel = NULL;
2525 time_t nowtime = time(NULL) + expirey;
2526 const char *fc = fullcontact ? "fullcontact" : NULL;
2528 int realtimeregs = ast_check_realtime("sipregs");
2530 tablename = realtimeregs ? "sipregs" : "sippeers";
2532 snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime); /* Expiration time */
2533 ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
2534 snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port));
2536 if (ast_strlen_zero(sysname)) /* No system name, disable this */
2538 else if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTSAVE_SYSNAME))
2539 syslabel = "regserver";
2542 ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
2543 "port", port, "regseconds", regseconds,
2544 "username", username, fc, fullcontact, syslabel, sysname, NULL); /* note fc and syslabel _can_ be NULL */
2546 ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
2547 "port", port, "regseconds", regseconds,
2548 "username", username, syslabel, sysname, NULL); /* note syslabel _can_ be NULL */
2551 /*! \brief Automatically add peer extension to dial plan */
2552 static void register_peer_exten(struct sip_peer *peer, int onoff)
2555 char *stringp, *ext, *context;
2557 /* XXX note that global_regcontext is both a global 'enable' flag and
2558 * the name of the global regexten context, if not specified
2561 if (ast_strlen_zero(global_regcontext))
2564 ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi));
2566 while ((ext = strsep(&stringp, "&"))) {
2567 if ((context = strchr(ext, '@'))) {
2568 *context++ = '\0'; /* split ext@context */
2569 if (!ast_context_find(context)) {
2570 ast_log(LOG_WARNING, "Context %s must exist in regcontext= in sip.conf!\n", context);
2574 context = global_regcontext;
2577 ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop",
2578 ast_strdup(peer->name), ast_free, "SIP");
2580 ast_context_remove_extension(context, ext, 1, NULL);
2584 /*! \brief Destroy peer object from memory */
2585 static void sip_destroy_peer(struct sip_peer *peer)
2587 if (option_debug > 2)
2588 ast_log(LOG_DEBUG, "Destroying SIP peer %s\n", peer->name);
2590 if (peer->outboundproxy)
2591 free(peer->outboundproxy);
2593 /* Delete it, it needs to disappear */
2595 sip_destroy(peer->call);
2597 if (peer->mwipvt) /* We have an active subscription, delete it */
2598 sip_destroy(peer->mwipvt);
2600 if (peer->chanvars) {
2601 ast_variables_destroy(peer->chanvars);
2602 peer->chanvars = NULL;
2604 if (peer->expire > -1)
2605 ast_sched_del(sched, peer->expire);
2607 if (peer->pokeexpire > -1)
2608 ast_sched_del(sched, peer->pokeexpire);
2609 register_peer_exten(peer, FALSE);
2610 ast_free_ha(peer->ha);
2611 if (ast_test_flag(&peer->flags[1], SIP_PAGE2_SELFDESTRUCT))
2613 else if (ast_test_flag(&peer->flags[0], SIP_REALTIME)) {
2615 if (option_debug > 2)
2616 ast_log(LOG_DEBUG,"-REALTIME- peer Destroyed. Name: %s. Realtime Peer objects: %d\n", peer->name, rpeerobjs);
2619 clear_realm_authentication(peer->auth);
2622 ast_dnsmgr_release(peer->dnsmgr);
2626 /*! \brief Update peer data in database (if used) */
2627 static void update_peer(struct sip_peer *p, int expiry)
2629 int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
2630 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTUPDATE) &&
2631 (ast_test_flag(&p->flags[0], SIP_REALTIME) || rtcachefriends)) {
2632 realtime_update_peer(p->name, &p->addr, p->username, rtcachefriends ? p->fullcontact : NULL, expiry);
2637 /*! \brief realtime_peer: Get peer from realtime storage
2638 * Checks the "sippeers" realtime family from extconfig.conf
2639 * Checks the "sipregs" realtime family from extconfig.conf if it's configured.
2640 * \todo Consider adding check of port address when matching here to follow the same
2641 * algorithm as for static peers. Will we break anything by adding that?
2643 static struct sip_peer *realtime_peer(const char *newpeername, struct sockaddr_in *sin)
2645 struct sip_peer *peer;
2646 struct ast_variable *var = NULL;
2647 struct ast_variable *varregs = NULL;
2648 struct ast_variable *tmp;
2649 char ipaddr[INET_ADDRSTRLEN];
2650 int realtimeregs = ast_check_realtime("sipregs");
2652 /* First check on peer name */
2654 var = ast_load_realtime("sippeers", "name", newpeername, NULL);
2656 varregs = ast_load_realtime("sipregs", "name", newpeername, NULL);
2657 } else if (sin) { /* Then check on IP address for dynamic peers */
2658 ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
2659 var = ast_load_realtime("sippeers", "host", ipaddr, NULL); /* First check for fixed IP hosts */
2660 if (var && realtimeregs) {
2663 if (!newpeername && !strcasecmp(tmp->name, "name"))
2664 newpeername = tmp->value;
2667 varregs = ast_load_realtime("sipregs", "name", newpeername, NULL);
2670 varregs = ast_load_realtime("sipregs", "ipaddr", ipaddr, NULL); /* Then check for registered hosts */
2672 var = ast_load_realtime("sippeers", "ipaddr", ipaddr, NULL); /* Then check for registered hosts */
2676 if (!newpeername && !strcasecmp(tmp->name, "name"))
2677 newpeername = tmp->value;
2680 var = ast_load_realtime("sippeers", "name", newpeername, NULL);
2688 for (tmp = var; tmp; tmp = tmp->next) {
2689 /* If this is type=user, then skip this object. */
2690 if (!strcasecmp(tmp->name, "type") &&
2691 !strcasecmp(tmp->value, "user")) {
2692 ast_variables_destroy(var);
2693 ast_variables_destroy(varregs);
2695 } else if (!newpeername && !strcasecmp(tmp->name, "name")) {
2696 newpeername = tmp->value;
2700 if (!newpeername) { /* Did not find peer in realtime */
2701 ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", ipaddr);
2702 ast_variables_destroy(var);
2707 /* Peer found in realtime, now build it in memory */
2708 peer = build_peer(newpeername, var, varregs, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
2710 ast_variables_destroy(var);
2711 ast_variables_destroy(varregs);
2715 if (option_debug > 2)
2716 ast_log(LOG_DEBUG,"-REALTIME- loading peer from database to memory. Name: %s. Peer objects: %d\n", peer->name, rpeerobjs);
2718 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
2720 ast_copy_flags(&peer->flags[1],&global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
2721 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
2722 if (peer->expire > -1) {
2723 ast_sched_del(sched, peer->expire);
2725 peer->expire = ast_sched_add(sched, (global_rtautoclear) * 1000, expire_register, (void *)peer);
2727 ASTOBJ_CONTAINER_LINK(&peerl,peer);
2729 ast_set_flag(&peer->flags[0], SIP_REALTIME);
2731 ast_variables_destroy(var);
2732 ast_variables_destroy(varregs);
2737 /*! \brief Support routine for find_peer */
2738 static int sip_addrcmp(char *name, struct sockaddr_in *sin)
2740 /* We know name is the first field, so we can cast */
2741 struct sip_peer *p = (struct sip_peer *) name;
2742 return !(!inaddrcmp(&p->addr, sin) ||
2743 (ast_test_flag(&p->flags[0], SIP_INSECURE_PORT) &&
2744 (p->addr.sin_addr.s_addr == sin->sin_addr.s_addr)));
2747 /*! \brief Locate peer by name or ip address
2748 * This is used on incoming SIP message to find matching peer on ip
2749 or outgoing message to find matching peer on name */
2750 static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime)
2752 struct sip_peer *p = NULL;
2755 p = ASTOBJ_CONTAINER_FIND(&peerl, peer);
2757 p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp);
2760 p = realtime_peer(peer, sin);
2765 /*! \brief Remove user object from in-memory storage */
2766 static void sip_destroy_user(struct sip_user *user)
2768 if (option_debug > 2)
2769 ast_log(LOG_DEBUG, "Destroying user object from memory: %s\n", user->name);
2770 ast_free_ha(user->ha);
2771 if (user->chanvars) {
2772 ast_variables_destroy(user->chanvars);
2773 user->chanvars = NULL;
2775 if (ast_test_flag(&user->flags[0], SIP_REALTIME))
2782 /*! \brief Load user from realtime storage
2783 * Loads user from "sipusers" category in realtime (extconfig.conf)
2784 * Users are matched on From: user name (the domain in skipped) */
2785 static struct sip_user *realtime_user(const char *username)
2787 struct ast_variable *var;
2788 struct ast_variable *tmp;
2789 struct sip_user *user = NULL;
2791 var = ast_load_realtime("sipusers", "name", username, NULL);
2796 for (tmp = var; tmp; tmp = tmp->next) {
2797 if (!strcasecmp(tmp->name, "type") &&
2798 !strcasecmp(tmp->value, "peer")) {
2799 ast_variables_destroy(var);
2804 user = build_user(username, var, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
2806 if (!user) { /* No user found */
2807 ast_variables_destroy(var);
2811 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
2812 ast_set_flag(&user->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
2814 ASTOBJ_CONTAINER_LINK(&userl,user);
2816 /* Move counter from s to r... */
2819 ast_set_flag(&user->flags[0], SIP_REALTIME);
2821 ast_variables_destroy(var);
2825 /*! \brief Locate user by name
2826 * Locates user by name (From: sip uri user name part) first
2827 * from in-memory list (static configuration) then from
2828 * realtime storage (defined in extconfig.conf) */
2829 static struct sip_user *find_user(const char *name, int realtime)
2831 struct sip_user *u = ASTOBJ_CONTAINER_FIND(&userl, name);
2833 u = realtime_user(name);
2837 /*! \brief Set nat mode on the various data sockets */
2838 static void do_setnat(struct sip_pvt *p, int natflags)
2840 const char *mode = natflags ? "On" : "Off";
2844 ast_log(LOG_DEBUG, "Setting NAT on RTP to %s\n", mode);
2845 ast_rtp_setnat(p->rtp, natflags);
2849 ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", mode);
2850 ast_rtp_setnat(p->vrtp, natflags);
2854 ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", mode);
2855 ast_udptl_setnat(p->udptl, natflags);
2859 ast_log(LOG_DEBUG, "Setting NAT on TRTP to %s\n", mode);
2860 ast_rtp_setnat(p->trtp, natflags);
2864 /*! \brief Create address structure from peer reference.
2865 * return -1 on error, 0 on success.
2867 static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
2869 if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) &&
2870 (!peer->maxms || ((peer->lastms >= 0) && (peer->lastms <= peer->maxms)))) {
2871 dialog->sa = (peer->addr.sin_addr.s_addr) ? peer->addr : peer->defaddr;
2872 dialog->recv = dialog->sa;
2876 ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
2877 ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
2878 dialog->capability = peer->capability;
2879 if ((!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) || !(dialog->capability & AST_FORMAT_VIDEO_MASK)) && dialog->vrtp) {
2880 ast_rtp_destroy(dialog->vrtp);
2881 dialog->vrtp = NULL;
2883 if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT) && dialog->trtp) {
2884 ast_rtp_destroy(dialog->trtp);
2885 dialog->trtp = NULL;
2887 dialog->prefs = peer->prefs;
2888 if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
2889 dialog->t38.capability = global_t38_capability;
2890 if (dialog->udptl) {
2891 if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_FEC )
2892 dialog->t38.capability |= T38FAX_UDP_EC_FEC;
2893 else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY )
2894 dialog->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
2895 else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_NONE )
2896 dialog->t38.capability |= T38FAX_UDP_EC_NONE;
2897 dialog->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
2898 if (option_debug > 1)
2899 ast_log(LOG_DEBUG,"Our T38 capability (%d)\n", dialog->t38.capability);
2901 dialog->t38.jointcapability = dialog->t38.capability;
2902 } else if (dialog->udptl) {
2903 ast_udptl_destroy(dialog->udptl);
2904 dialog->udptl = NULL;
2906 do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE );
2909 ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
2910 ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
2911 ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
2912 ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
2913 ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive);
2914 /* Set Frame packetization */
2915 ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
2916 dialog->autoframing = peer->autoframing;
2919 ast_rtp_setdtmf(dialog->vrtp, 0);
2920 ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
2921 ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
2922 ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
2923 ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
2926 ast_rtp_setdtmf(dialog->trtp, 0);
2927 ast_rtp_setdtmfcompensate(dialog->trtp, 0);
2928 ast_rtp_set_rtptimeout(dialog->trtp, peer->rtptimeout);
2929 ast_rtp_set_rtpholdtimeout(dialog->trtp, peer->rtpholdtimeout);
2930 ast_rtp_set_rtpkeepalive(dialog->trtp, peer->rtpkeepalive);
2933 ast_string_field_set(dialog, peername, peer->username);
2934 ast_string_field_set(dialog, authname, peer->username);
2935 ast_string_field_set(dialog, username, peer->username);
2936 ast_string_field_set(dialog, peersecret, peer->secret);
2937 ast_string_field_set(dialog, peermd5secret, peer->md5secret);
2938 ast_string_field_set(dialog, mohsuggest, peer->mohsuggest);
2939 ast_string_field_set(dialog, mohinterpret, peer->mohinterpret);
2940 ast_string_field_set(dialog, tohost, peer->tohost);
2941 ast_string_field_set(dialog, fullcontact, peer->fullcontact);
2942 if (!dialog->initreq.headers && !ast_strlen_zero(peer->fromdomain)) {
2945 tmpcall = ast_strdupa(dialog->callid);
2946 c = strchr(tmpcall, '@');
2949 ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
2952 dialog->outboundproxy = obproxy_get(dialog, peer);
2953 if (ast_strlen_zero(dialog->tohost))
2954 ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
2955 if (!ast_strlen_zero(peer->fromdomain))
2956 ast_string_field_set(dialog, fromdomain, peer->fromdomain);
2957 if (!ast_strlen_zero(peer->fromuser))
2958 ast_string_field_set(dialog, fromuser, peer->fromuser);
2959 dialog->callgroup = peer->callgroup;
2960 dialog->pickupgroup = peer->pickupgroup;
2961 dialog->allowtransfer = peer->allowtransfer;
2962 /* Set timer T1 to RTT for this peer (if known by qualify=) */
2963 /* Minimum is settable or default to 100 ms */
2964 if (peer->maxms && peer->lastms)
2965 dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
2966 if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
2967 (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
2968 dialog->noncodeccapability |= AST_RTP_DTMF;
2970 dialog->noncodeccapability &= ~AST_RTP_DTMF;
2971 ast_string_field_set(dialog, context, peer->context);
2972 dialog->rtptimeout = peer->rtptimeout;
2973 if (peer->call_limit)
2974 ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
2975 dialog->maxcallbitrate = peer->maxcallbitrate;
2980 /*! \brief create address structure from peer name
2981 * Or, if peer not found, find it in the global DNS
2982 * returns TRUE (-1) on failure, FALSE on success */
2983 static int create_addr(struct sip_pvt *dialog, const char *opeer)
2986 struct ast_hostent ahp;
2987 struct sip_peer *peer;
2990 char host[MAXHOSTNAMELEN], *hostn;
2993 ast_copy_string(peername, opeer, sizeof(peername));
2994 port = strchr(peername, ':');
2997 dialog->sa.sin_family = AF_INET;
2998 dialog->timer_t1 = SIP_TIMER_T1; /* Default SIP retransmission timer T1 (RFC 3261) */
2999 peer = find_peer(peername, NULL, 1);
3002 int res = create_addr_from_peer(dialog, peer);
3007 ast_string_field_set(dialog, tohost, peername);
3009 /* Get the outbound proxy information */
3010 dialog->outboundproxy = obproxy_get(dialog, NULL);
3012 /* If we have an outbound proxy, don't bother with DNS resolution at all */
3013 if (dialog->outboundproxy)
3016 /* Let's see if we can find the host in DNS. First try DNS SRV records,
3017 then hostname lookup */
3020 portno = port ? atoi(port) : STANDARD_SIP_PORT;
3021 if (global_srvlookup) {
3022 char service[MAXHOSTNAMELEN];
3026 snprintf(service, sizeof(service), "_sip._udp.%s", peername);
3027 ret = ast_get_srv(NULL, host, sizeof(host), &tportno, service);
3033 hp = ast_gethostbyname(hostn, &ahp);
3035 ast_log(LOG_WARNING, "No such host: %s\n", peername);
3038 memcpy(&dialog->sa.sin_addr, hp->h_addr, sizeof(dialog->sa.sin_addr));
3039 dialog->sa.sin_port = htons(portno);
3040 dialog->recv = dialog->sa;
3044 /*! \brief Scheduled congestion on a call */
3045 static int auto_congest(void *nothing)
3047 struct sip_pvt *p = nothing;
3052 /* XXX fails on possible deadlock */
3053 if (!ast_channel_trylock(p->owner)) {
3054 ast_log(LOG_NOTICE, "Auto-congesting %s\n", p->owner->name);
3055 append_history(p, "Cong", "Auto-congesting (timer)");
3056 ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
3057 ast_channel_unlock(p->owner);
3065 /*! \brief Initiate SIP call from PBX
3066 * used from the dial() application */
3067 static int sip_call(struct ast_channel *ast, char *dest, int timeout)
3071 struct varshead *headp;
3072 struct ast_var_t *current;
3073 const char *referer = NULL; /* SIP referrer */
3076 if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
3077 ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
3081 /* Check whether there is vxml_url, distinctive ring variables */
3082 headp=&ast->varshead;
3083 AST_LIST_TRAVERSE(headp,current,entries) {
3084 /* Check whether there is a VXML_URL variable */
3085 if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
3086 p->options->vxml_url = ast_var_value(current);
3087 } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
3088 p->options->uri_options = ast_var_value(current);
3089 } else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
3090 /* Check whether there is a variable with a name starting with SIPADDHEADER */
3091 p->options->addsipheaders = 1;
3092 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
3093 /* This is a transfered call */
3094 p->options->transfer = 1;
3095 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
3096 /* This is the referrer */
3097 referer = ast_var_value(current);
3098 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
3099 /* We're replacing a call. */
3100 p->options->replaces = ast_var_value(current);
3101 } else if (!strcasecmp(ast_var_name(current), "T38CALL")) {
3102 p->t38.state = T38_LOCAL_DIRECT;
3104 ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
3110 ast_set_flag(&p->flags[0], SIP_OUTGOING);
3112 if (p->options->transfer) {
3116 if (sipdebug && option_debug > 2)
3117 ast_log(LOG_DEBUG, "Call for %s transfered by %s\n", p->username, referer);
3118 snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
3120 snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
3121 ast_string_field_set(p, cid_name, buf);
3124 ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username);
3126 res = update_call_counter(p, INC_CALL_RINGING);
3131 p->callingpres = ast->cid.cid_pres;
3132 p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
3133 p->jointnoncodeccapability = p->noncodeccapability;
3135 /* If there are no audio formats left to offer, punt */
3136 if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
3137 ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
3140 p->t38.jointcapability = p->t38.capability;
3141 if (option_debug > 1)
3142 ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
3143 transmit_invite(p, SIP_INVITE, 1, 2);
3144 p->invitestate = INV_CALLING;
3146 /* Initialize auto-congest time */
3147 p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
3153 /*! \brief Destroy registry object
3154 Objects created with the register= statement in static configuration */
3155 static void sip_registry_destroy(struct sip_registry *reg)
3158 if (option_debug > 2)
3159 ast_log(LOG_DEBUG, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname);
3162 /* Clear registry before destroying to ensure
3163 we don't get reentered trying to grab the registry lock */
3164 reg->call->registry = NULL;
3165 if (option_debug > 2)
3166 ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
3167 sip_destroy(reg->call);
3169 if (reg->expire > -1)
3170 ast_sched_del(sched, reg->expire);
3171 if (reg->timeout > -1)
3172 ast_sched_del(sched, reg->timeout);
3173 ast_string_field_free_pools(reg);
3179 /*! \brief Execute destruction of SIP dialog structure, release memory */
3180 static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
3182 struct sip_pvt *cur, *prev = NULL;
3185 if (sip_debug_test_pvt(p) || option_debug > 2)
3186 ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
3188 if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) {
3189 update_call_counter(p, DEC_CALL_LIMIT);
3190 if (option_debug > 1)
3191 ast_log(LOG_DEBUG, "This call did not properly clean up call limits. Call ID %s\n", p->callid);
3194 /* Remove link from peer to subscription of MWI */
3195 if (p->relatedpeer && p->relatedpeer->mwipvt)
3196 p->relatedpeer->mwipvt = NULL;
3199 sip_dump_history(p);
3204 if (p->stateid > -1)
3205 ast_extension_state_del(p->stateid, NULL);
3207 ast_sched_del(sched, p->initid);
3208 if (p->autokillid > -1)
3209 ast_sched_del(sched, p->autokillid);
3212 ast_rtp_destroy(p->rtp);
3214 ast_rtp_destroy(p->vrtp);
3216 ast_rtp_destroy(p->trtp);
3218 ast_udptl_destroy(p->udptl);
3222 free_old_route(p->route);
3226 if (p->registry->call == p)
3227 p->registry->call = NULL;
3228 registry_unref(p->registry);
3231 /* Unlink us from the owner if we have one */
3234 ast_channel_lock(p->owner);
3236 ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
3237 p->owner->tech_pvt = NULL;
3239 ast_channel_unlock(p->owner);
3243 struct sip_history *hist;
3244 while( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) )
3250 /* Lock dialog list before removing ourselves from the list */
3253 for (prev = NULL, cur = dialoglist; cur; prev = cur, cur = cur->next) {
3255 UNLINK(cur, dialoglist, prev);
3260 dialoglist_unlock();
3262 ast_log(LOG_WARNING, "Trying to destroy \"%s\", not found in dialog list?!?! \n", p->callid);
3266 /* remove all current packets in this dialog */
3267 while((cp = p->packets)) {
3268 p->packets = p->packets->next;
3269 if (cp->retransid > -1)
3270 ast_sched_del(sched, cp->retransid);
3274 ast_variables_destroy(p->chanvars);
3277 ast_mutex_destroy(&p->pvt_lock);
3279 ast_string_field_free_pools(p);
3284 /*! \brief update_call_counter: Handle call_limit for SIP users
3285 * Setting a call-limit will cause calls above the limit not to be accepted.
3287 * Remember that for a type=friend, there's one limit for the user and
3288 * another for the peer, not a combined call limit.
3289 * This will cause unexpected behaviour in subscriptions, since a "friend"
3290 * is *two* devices in Asterisk, not one.
3292 * Thought: For realtime, we should probably update storage with inuse counter...
3294 * \return 0 if call is ok (no call limit, below threshold)
3295 * -1 on rejection of call
3298 static int update_call_counter(struct sip_pvt *fup, int event)
3301 int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
3302 int outgoing = ast_test_flag(&fup->flags[1], SIP_PAGE2_OUTGOING_CALL);
3303 struct sip_user *u = NULL;
3304 struct sip_peer *p = NULL;
3306 if (option_debug > 2)
3307 ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
3309 /* Test if we need to check call limits, in order to avoid
3310 realtime lookups if we do not need it */
3311 if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT))
3314 ast_copy_string(name, fup->username, sizeof(name));
3316 /* Check the list of users only for incoming calls */
3317 if (global_limitonpeers == FALSE && !outgoing && (u = find_user(name, 1))) {
3319 call_limit = &u->call_limit;
3321 } else if ( (p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, 1) ) ) { /* Try to find peer */
3323 call_limit = &p->call_limit;
3324 inringing = &p->inRinging;
3325 ast_copy_string(name, fup->peername, sizeof(name));
3328 if (option_debug > 1)
3329 ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name);
3334 /* incoming and outgoing affects the inUse counter */
3335 case DEC_CALL_LIMIT:
3336 /* Decrement inuse count if applicable */
3337 if (inuse && ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
3338 ast_atomic_fetchadd_int(inuse, -1);
3339 ast_clear_flag(&fup->flags[0], SIP_INC_COUNT);
3342 /* Decrement ringing count if applicable */
3343 if (inringing && ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
3344 ast_atomic_fetchadd_int(inringing, -1);
3345 ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
3347 /* Decrement onhold count if applicable */
3348 if (ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD) && global_notifyhold)
3349 sip_peer_hold(fup, FALSE);
3350 if (option_debug > 1 || sipdebug)
3351 ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
3354 case INC_CALL_RINGING:
3355 case INC_CALL_LIMIT:
3356 /* If call limit is active and we have reached the limit, reject the call */
3357 if (*call_limit > 0 ) {
3358 if (*inuse >= *call_limit) {
3359 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);
3367 if (inringing && (event == INC_CALL_RINGING)) {
3368 if (!ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
3369 ast_atomic_fetchadd_int(inringing, +1);
3370 ast_set_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
3374 ast_atomic_fetchadd_int(inuse, +1);
3375 ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
3376 if (option_debug > 1 || sipdebug) {
3377 ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
3381 case DEC_CALL_RINGING:
3382 if (inringing && ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
3383 ast_atomic_fetchadd_int(inringing, -1);
3384 ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
3389 ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
3392 ast_device_state_changed("SIP/%s", p->name);
3394 } else /* u must be set */
3399 /*! \brief Destroy SIP call structure */
3400 static void sip_destroy(struct sip_pvt *p)
3402 if (option_debug > 2)
3403 ast_log(LOG_DEBUG, "Destroying SIP dialog %s\n", p->callid);
3404 __sip_destroy(p, TRUE, TRUE);
3407 /*! \brief Convert SIP hangup causes to Asterisk hangup causes */
3408 static int hangup_sip2cause(int cause)
3410 /* Possible values taken from causes.h */
3413 case 401: /* Unauthorized */
3414 return AST_CAUSE_CALL_REJECTED;
3415 case 403: /* Not found */
3416 return AST_CAUSE_CALL_REJECTED;
3417 case 404: /* Not found */
3418 return AST_CAUSE_UNALLOCATED;
3419 case 405: /* Method not allowed */
3420 return AST_CAUSE_INTERWORKING;
3421 case 407: /* Proxy authentication required */
3422 return AST_CAUSE_CALL_REJECTED;
3423 case 408: /* No reaction */
3424 return AST_CAUSE_NO_USER_RESPONSE;
3425 case 409: /* Conflict */
3426 return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
3427 case 410: /* Gone */
3428 return AST_CAUSE_UNALLOCATED;
3429 case 411: /* Length required */
3430 return AST_CAUSE_INTERWORKING;
3431 case 413: /* Request entity too large */
3432 return AST_CAUSE_INTERWORKING;
3433 case 414: /* Request URI too large */
3434 return AST_CAUSE_INTERWORKING;
3435 case 415: /* Unsupported media type */
3436 return AST_CAUSE_INTERWORKING;
3437 case 420: /* Bad extension */
3438 return AST_CAUSE_NO_ROUTE_DESTINATION;
3439 case 480: /* No answer */
3440 return AST_CAUSE_NO_ANSWER;
3441 case 481: /* No answer */
3442 return AST_CAUSE_INTERWORKING;
3443 case 482: /* Loop detected */
3444 return AST_CAUSE_INTERWORKING;
3445 case 483: /* Too many hops */
3446 return AST_CAUSE_NO_ANSWER;
3447 case 484: /* Address incomplete */
3448 return AST_CAUSE_INVALID_NUMBER_FORMAT;
3449 case 485: /* Ambiguous */
3450 return AST_CAUSE_UNALLOCATED;
3451 case 486: /* Busy everywhere */
3452 return AST_CAUSE_BUSY;
3453 case 487: /* Request terminated */
3454 return AST_CAUSE_INTERWORKING;
3455 case 488: /* No codecs approved */
3456 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
3457 case 491: /* Request pending */
3458 return AST_CAUSE_INTERWORKING;
3459 case 493: /* Undecipherable */
3460 return AST_CAUSE_INTERWORKING;
3461 case 500: /* Server internal failure */
3462 return AST_CAUSE_FAILURE;
3463 case 501: /* Call rejected */
3464 return AST_CAUSE_FACILITY_REJECTED;
3466 return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
3467 case 503: /* Service unavailable */
3468 return AST_CAUSE_CONGESTION;
3469 case 504: /* Gateway timeout */
3470 return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
3471 case 505: /* SIP version not supported */
3472 return AST_CAUSE_INTERWORKING;
3473 case 600: /* Busy everywhere */
3474 return AST_CAUSE_USER_BUSY;
3475 case 603: /* Decline */
3476 return AST_CAUSE_CALL_REJECTED;
3477 case 604: /* Does not exist anywhere */
3478 return AST_CAUSE_UNALLOCATED;
3479 case 606: /* Not acceptable */
3480 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;