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/lock.h"
117 #include "asterisk/sched.h"
118 #include "asterisk/io.h"
119 #include "asterisk/rtp.h"
120 #include "asterisk/udptl.h"
121 #include "asterisk/acl.h"
122 #include "asterisk/manager.h"
123 #include "asterisk/callerid.h"
124 #include "asterisk/cli.h"
125 #include "asterisk/app.h"
126 #include "asterisk/musiconhold.h"
127 #include "asterisk/dsp.h"
128 #include "asterisk/features.h"
129 #include "asterisk/acl.h"
130 #include "asterisk/srv.h"
131 #include "asterisk/astdb.h"
132 #include "asterisk/causes.h"
133 #include "asterisk/utils.h"
134 #include "asterisk/file.h"
135 #include "asterisk/astobj.h"
136 #include "asterisk/dnsmgr.h"
137 #include "asterisk/devicestate.h"
138 #include "asterisk/linkedlists.h"
139 #include "asterisk/stringfields.h"
140 #include "asterisk/monitor.h"
141 #include "asterisk/localtime.h"
142 #include "asterisk/abstract_jb.h"
143 #include "asterisk/compiler.h"
153 #define VIDEO_CODEC_MASK 0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
154 #ifndef IPTOS_MINCOST
155 #define IPTOS_MINCOST 0x02
158 /* #define VOCAL_DATA_HACK */
160 #define DEFAULT_DEFAULT_EXPIRY 120
161 #define DEFAULT_MIN_EXPIRY 60
162 #define DEFAULT_MAX_EXPIRY 3600
163 #define DEFAULT_REGISTRATION_TIMEOUT 20
164 #define DEFAULT_MAX_FORWARDS "70"
166 /* guard limit must be larger than guard secs */
167 /* guard min must be < 1000, and should be >= 250 */
168 #define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
169 #define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
171 #define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
172 GUARD_PCT turns out to be lower than this, it
173 will use this time instead.
174 This is in milliseconds. */
175 #define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
176 below EXPIRY_GUARD_LIMIT */
177 #define DEFAULT_EXPIRY 900 /*!< Expire slowly */
179 static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
180 static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
181 static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
182 static int expiry = DEFAULT_EXPIRY;
185 #define MAX(a,b) ((a) > (b) ? (a) : (b))
188 #define CALLERID_UNKNOWN "Unknown"
190 #define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
191 #define DEFAULT_FREQ_OK 60 * 1000 /*!< Qualification: How often to check for the host to be up */
192 #define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */
194 #define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
195 #define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
196 #define SIP_TRANS_TIMEOUT 32000 /*!< SIP request timeout (rfc 3261) 64*T1
197 \todo Use known T1 for timeout (peerpoke)
199 #define DEFAULT_TRANS_TIMEOUT -1 /* Use default SIP transaction timeout */
200 #define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
202 #define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
203 #define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
204 #define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */
206 #define INITIAL_CSEQ 101 /*!< our initial sip sequence number */
208 /*! \brief Global jitterbuffer configuration - by default, jb is disabled */
209 static struct ast_jb_conf default_jbconf =
213 .resync_threshold = -1,
216 static struct ast_jb_conf global_jbconf;
218 static const char config[] = "sip.conf";
219 static const char notify_config[] = "sip_notify.conf";
220 static int usecnt = 0;
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 /* Do _NOT_ make any changes to this enum, or the array following it;
241 if you think you are doing the right thing, you are probably
242 not doing the right thing. If you think there are changes
243 needed, get someone else to review them first _before_
244 submitting a patch. If these two lists do not match properly
245 bad things will happen.
249 XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
250 If it fails, it's critical and will cause a teardown of the session */
251 XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
252 XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */
255 enum parse_register_result {
256 PARSE_REGISTER_FAILED,
257 PARSE_REGISTER_UPDATE,
258 PARSE_REGISTER_QUERY,
261 enum subscriptiontype {
271 static const struct cfsubscription_types {
272 enum subscriptiontype type;
273 const char * const event;
274 const char * const mediatype;
275 const char * const text;
276 } subscription_types[] = {
277 { NONE, "-", "unknown", "unknown" },
278 /* RFC 4235: SIP Dialog event package */
279 { DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" },
280 { CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */
281 { PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */
282 { XPIDF_XML, "presence", "application/xpidf+xml", "xpidf+xml" }, /* Pre-RFC 3863 with MS additions */
283 { MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
286 /*! \brief SIP Request methods known by Asterisk */
288 SIP_UNKNOWN, /* Unknown response */
289 SIP_RESPONSE, /* Not request, response to outbound request */
295 SIP_PRACK, /* Not supported at all */
300 SIP_UPDATE, /* We can send UPDATE; but not accept it */
303 SIP_PUBLISH, /* Not supported at all */
306 /*! \brief Authentication types - proxy or www authentication
307 \note Endpoints, like Asterisk, should always use WWW authentication to
308 allow multiple authentications in the same call - to the proxy and
316 /*! \brief Authentication result from check_auth* functions */
317 enum check_auth_result {
319 AUTH_CHALLENGE_SENT = 1,
320 AUTH_SECRET_FAILED = -1,
321 AUTH_USERNAME_MISMATCH = -2,
324 AUTH_UNKNOWN_DOMAIN = -5,
327 /*! \brief States for outbound registrations (with register= lines in sip.conf */
328 enum sipregistrystate {
329 REG_STATE_UNREGISTERED = 0, /*!< We are not registred */
330 REG_STATE_REGSENT, /*!< Registration request sent */
331 REG_STATE_AUTHSENT, /*!< We have tried to authenticate */
332 REG_STATE_REGISTERED, /*!< Registred and done */
333 REG_STATE_REJECTED, /*!< Registration rejected */
334 REG_STATE_TIMEOUT, /*!< Registration timed out */
335 REG_STATE_NOAUTH, /*!< We have no accepted credentials */
336 REG_STATE_FAILED, /*!< Registration failed after several tries */
340 /*! XXX Note that sip_methods[i].id == i must hold or the code breaks */
341 static const struct cfsip_methods {
343 int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
346 { SIP_UNKNOWN, RTP, "-UNKNOWN-" },
347 { SIP_RESPONSE, NO_RTP, "SIP/2.0" },
348 { SIP_REGISTER, NO_RTP, "REGISTER" },
349 { SIP_OPTIONS, NO_RTP, "OPTIONS" },
350 { SIP_NOTIFY, NO_RTP, "NOTIFY" },
351 { SIP_INVITE, RTP, "INVITE" },
352 { SIP_ACK, NO_RTP, "ACK" },
353 { SIP_PRACK, NO_RTP, "PRACK" },
354 { SIP_BYE, NO_RTP, "BYE" },
355 { SIP_REFER, NO_RTP, "REFER" },
356 { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE" },
357 { SIP_MESSAGE, NO_RTP, "MESSAGE" },
358 { SIP_UPDATE, NO_RTP, "UPDATE" },
359 { SIP_INFO, NO_RTP, "INFO" },
360 { SIP_CANCEL, NO_RTP, "CANCEL" },
361 { SIP_PUBLISH, NO_RTP, "PUBLISH" }
364 /*! Define SIP option tags, used in Require: and Supported: headers
365 We need to be aware of these properties in the phones to use
366 the replace: header. We should not do that without knowing
367 that the other end supports it...
368 This is nothing we can configure, we learn by the dialog
369 Supported: header on the REGISTER (peer) or the INVITE
371 We are not using many of these today, but will in the future.
372 This is documented in RFC 3261
375 #define NOT_SUPPORTED 0
377 #define SIP_OPT_REPLACES (1 << 0)
378 #define SIP_OPT_100REL (1 << 1)
379 #define SIP_OPT_TIMER (1 << 2)
380 #define SIP_OPT_EARLY_SESSION (1 << 3)
381 #define SIP_OPT_JOIN (1 << 4)
382 #define SIP_OPT_PATH (1 << 5)
383 #define SIP_OPT_PREF (1 << 6)
384 #define SIP_OPT_PRECONDITION (1 << 7)
385 #define SIP_OPT_PRIVACY (1 << 8)
386 #define SIP_OPT_SDP_ANAT (1 << 9)
387 #define SIP_OPT_SEC_AGREE (1 << 10)
388 #define SIP_OPT_EVENTLIST (1 << 11)
389 #define SIP_OPT_GRUU (1 << 12)
390 #define SIP_OPT_TARGET_DIALOG (1 << 13)
391 #define SIP_OPT_NOREFERSUB (1 << 14)
392 #define SIP_OPT_HISTINFO (1 << 15)
393 #define SIP_OPT_RESPRIORITY (1 << 16)
395 /*! \brief List of well-known SIP options. If we get this in a require,
396 we should check the list and answer accordingly. */
397 static const struct cfsip_options {
398 int id; /*!< Bitmap ID */
399 int supported; /*!< Supported by Asterisk ? */
400 char * const text; /*!< Text id, as in standard */
401 } sip_options[] = { /* XXX used in 3 places */
402 /* RFC3891: Replaces: header for transfer */
403 { SIP_OPT_REPLACES, SUPPORTED, "replaces" },
404 /* One version of Polycom firmware has the wrong label */
405 { SIP_OPT_REPLACES, SUPPORTED, "replace" },
406 /* RFC3262: PRACK 100% reliability */
407 { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" },
408 /* RFC4028: SIP Session Timers */
409 { SIP_OPT_TIMER, NOT_SUPPORTED, "timer" },
410 /* RFC3959: SIP Early session support */
411 { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" },
412 /* RFC3911: SIP Join header support */
413 { SIP_OPT_JOIN, NOT_SUPPORTED, "join" },
414 /* RFC3327: Path support */
415 { SIP_OPT_PATH, NOT_SUPPORTED, "path" },
416 /* RFC3840: Callee preferences */
417 { SIP_OPT_PREF, NOT_SUPPORTED, "pref" },
418 /* RFC3312: Precondition support */
419 { SIP_OPT_PRECONDITION, NOT_SUPPORTED, "precondition" },
420 /* RFC3323: Privacy with proxies*/
421 { SIP_OPT_PRIVACY, NOT_SUPPORTED, "privacy" },
422 /* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
423 { SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp-anat" },
424 /* RFC3329: Security agreement mechanism */
425 { SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" },
426 /* SIMPLE events: draft-ietf-simple-event-list-07.txt */
427 { SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" },
428 /* GRUU: Globally Routable User Agent URI's */
429 { SIP_OPT_GRUU, NOT_SUPPORTED, "gruu" },
430 /* Target-dialog: draft-ietf-sip-target-dialog-03.txt */
431 { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "tdialog" },
432 /* Disable the REFER subscription, RFC 4488 */
433 { SIP_OPT_NOREFERSUB, NOT_SUPPORTED, "norefersub" },
434 /* ietf-sip-history-info-06.txt */
435 { SIP_OPT_HISTINFO, NOT_SUPPORTED, "histinfo" },
436 /* ietf-sip-resource-priority-10.txt */
437 { SIP_OPT_RESPRIORITY, NOT_SUPPORTED, "resource-priority" },
441 /*! \brief SIP Methods we support */
442 #define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
444 /*! \brief SIP Extensions we support */
445 #define SUPPORTED_EXTENSIONS "replaces"
448 /* Default values, set and reset in reload_config before reading configuration */
449 /* These are default values in the source. There are other recommended values in the
450 sip.conf.sample for new installations. These may differ to keep backwards compatibility,
451 yet encouraging new behaviour on new installations
453 #define DEFAULT_SIP_PORT 5060 /*!< Our default port to listen to. From RFC 3261 (former 2543) */
454 #define DEFAULT_CONTEXT "default"
455 #define DEFAULT_MOHINTERPRET "default"
456 #define DEFAULT_MOHSUGGEST ""
457 #define DEFAULT_VMEXTEN "asterisk"
458 #define DEFAULT_CALLERID "asterisk"
459 #define DEFAULT_NOTIFYMIME "application/simple-message-summary"
460 #define DEFAULT_MWITIME 10
461 #define DEFAULT_ALLOWGUEST TRUE
462 #define DEFAULT_SRVLOOKUP FALSE /*!< Recommended setting is ON */
463 #define DEFAULT_COMPACTHEADERS FALSE
464 #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. */
465 #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. */
466 #define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
467 #define DEFAULT_ALLOW_EXT_DOM TRUE
468 #define DEFAULT_REALM "asterisk"
469 #define DEFAULT_NOTIFYRINGING TRUE
470 #define DEFAULT_PEDANTIC FALSE
471 #define DEFAULT_AUTOCREATEPEER FALSE
472 #define DEFAULT_QUALIFY FALSE
473 #define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
474 #define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
475 #ifndef DEFAULT_USERAGENT
476 #define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
480 /* Default setttings are used as a channel setting and as a default when
481 configuring devices */
482 static char default_context[AST_MAX_CONTEXT];
483 static char default_subscribecontext[AST_MAX_CONTEXT];
484 static char default_language[MAX_LANGUAGE];
485 static char default_callerid[AST_MAX_EXTENSION];
486 static char default_fromdomain[AST_MAX_EXTENSION];
487 static char default_notifymime[AST_MAX_EXTENSION];
488 static int default_qualify; /*!< Default Qualify= setting */
489 static char default_vmexten[AST_MAX_EXTENSION];
490 static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
491 static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
492 * a bridged channel on hold */
493 static int default_maxcallbitrate; /*!< Maximum bitrate for call */
494 static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
496 /* Global settings only apply to the channel */
497 static int global_rtautoclear;
498 static int global_notifyringing; /*!< Send notifications on ringing */
499 static int global_alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
500 static int srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
501 static int pedanticsipchecking; /*!< Extra checking ? Default off */
502 static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
503 static int global_relaxdtmf; /*!< Relax DTMF */
504 static int global_rtptimeout; /*!< Time out call if no RTP */
505 static int global_rtpholdtimeout;
506 static int global_rtpkeepalive; /*!< Send RTP keepalives */
507 static int global_reg_timeout;
508 static int global_regattempts_max; /*!< Registration attempts before giving up */
509 static int global_allowguest; /*!< allow unauthenticated users/peers to connect? */
510 static int global_allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
511 the global setting is in globals_flags[1] */
512 static int global_mwitime; /*!< Time between MWI checks for peers */
513 static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
514 static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
515 static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
516 static int compactheaders; /*!< send compact sip headers */
517 static int recordhistory; /*!< Record SIP history. Off by default */
518 static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
519 static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
520 static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
521 static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
522 static int allow_external_domains; /*!< Accept calls to external SIP domains? */
523 static int global_callevents; /*!< Whether we send manager events or not */
524 static int global_t1min; /*!< T1 roundtrip time minimum */
525 static int global_autoframing; /*!< ?????????? */
526 static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */
528 /*! \brief Codecs that we support by default: */
529 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
530 static int noncodeccapability = AST_RTP_DTMF;
532 /* Object counters */
533 static int suserobjs = 0; /*!< Static users */
534 static int ruserobjs = 0; /*!< Realtime users */
535 static int speerobjs = 0; /*!< Statis peers */
536 static int rpeerobjs = 0; /*!< Realtime peers */
537 static int apeerobjs = 0; /*!< Autocreated peer objects */
538 static int regobjs = 0; /*!< Registry objects */
540 static struct ast_flags global_flags[2] = {{0}}; /*!< global SIP_ flags */
543 /*! \brief Protect the SIP dialog list (of sip_pvt's) */
544 AST_MUTEX_DEFINE_STATIC(iflock);
546 /*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
547 when it's doing something critical. */
548 AST_MUTEX_DEFINE_STATIC(netlock);
550 AST_MUTEX_DEFINE_STATIC(monlock);
552 AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
554 /*! \brief This is the thread for the monitor which checks for input on the channels
555 which are not currently in use. */
556 static pthread_t monitor_thread = AST_PTHREADT_NULL;
558 static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
559 static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
561 static struct sched_context *sched; /*!< The scheduling context */
562 static struct io_context *io; /*!< The IO context */
563 static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
565 #define DEC_CALL_LIMIT 0
566 #define INC_CALL_LIMIT 1
567 #define DEC_CALL_RINGING 2
568 #define INC_CALL_RINGING 3
570 /*! \brief sip_request: The data grabbed from the UDP socket */
572 char *rlPart1; /*!< SIP Method Name or "SIP/2.0" protocol version */
573 char *rlPart2; /*!< The Request URI or Response Status */
574 int len; /*!< Length */
575 int headers; /*!< # of SIP Headers */
576 int method; /*!< Method of this request */
577 int lines; /*!< Body Content */
578 unsigned int flags; /*!< SIP_PKT Flags for this packet */
579 char *header[SIP_MAX_HEADERS];
580 char *line[SIP_MAX_LINES];
581 char data[SIP_MAX_PACKET];
582 unsigned int sdp_start; /*!< the line number where the SDP begins */
583 unsigned int sdp_end; /*!< the line number where the SDP ends */
587 * A sip packet is stored into the data[] buffer, with the header followed
588 * by an empty line and the body of the message.
589 * On outgoing packets, data is accumulated in data[] with len reflecting
590 * the next available byte, headers and lines count the number of lines
591 * in both parts. There are no '\0' in data[0..len-1].
593 * On received packet, the input read from the socket is copied into data[],
594 * len is set and the string is NUL-terminated. Then a parser fills up
595 * the other fields -header[] and line[] to point to the lines of the
596 * message, rlPart1 and rlPart2 parse the first lnie as below:
598 * Requests have in the first line METHOD URI SIP/2.0
599 * rlPart1 = method; rlPart2 = uri;
600 * Responses have in the first line SIP/2.0 code description
601 * rlPart1 = SIP/2.0; rlPart2 = code + description;
605 /*! \brief structure used in transfers */
607 struct ast_channel *chan1; /*!< First channel involved */
608 struct ast_channel *chan2; /*!< Second channel involved */
609 struct sip_request req; /*!< Request that caused the transfer (REFER) */
610 int seqno; /*!< Sequence number */
615 /*! \brief Parameters to the transmit_invite function */
616 struct sip_invite_param {
617 int addsipheaders; /*!< Add extra SIP headers */
618 const char *uri_options; /*!< URI options to add to the URI */
619 const char *vxml_url; /*!< VXML url for Cisco phones */
620 char *auth; /*!< Authentication */
621 char *authheader; /*!< Auth header */
622 enum sip_auth_type auth_type; /*!< Authentication type */
623 const char *replaces; /*!< Replaces header for call transfers */
624 int transfer; /*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */
627 /*! \brief Structure to save routing information for a SIP session */
629 struct sip_route *next;
633 /*! \brief Modes for SIP domain handling in the PBX */
635 SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */
636 SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */
639 /*! \brief Domain data structure.
640 \note In the future, we will connect this to a configuration tree specific
644 char domain[MAXHOSTNAMELEN]; /*!< SIP domain we are responsible for */
645 char context[AST_MAX_EXTENSION]; /*!< Incoming context for this domain */
646 enum domain_mode mode; /*!< How did we find this domain? */
647 AST_LIST_ENTRY(domain) list; /*!< List mechanics */
650 static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */
653 /*! \brief sip_history: Structure for saving transactions within a SIP dialog */
655 AST_LIST_ENTRY(sip_history) list;
656 char event[0]; /* actually more, depending on needs */
659 AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
661 /*! \brief sip_auth: Creadentials for authentication to other SIP services */
663 char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */
664 char username[256]; /*!< Username */
665 char secret[256]; /*!< Secret */
666 char md5secret[256]; /*!< MD5Secret */
667 struct sip_auth *next; /*!< Next auth structure in list */
670 /*--- Various flags for the flags field in the pvt structure */
671 #define SIP_ALREADYGONE (1 << 0) /*!< Whether or not we've already been destroyed by our peer */
672 #define SIP_NEEDDESTROY (1 << 1) /*!< if we need to be destroyed by the monitor thread */
673 #define SIP_NOVIDEO (1 << 2) /*!< Didn't get video in invite, don't offer */
674 #define SIP_RINGING (1 << 3) /*!< Have sent 180 ringing */
675 #define SIP_PROGRESS_SENT (1 << 4) /*!< Have sent 183 message progress */
676 #define SIP_NEEDREINVITE (1 << 5) /*!< Do we need to send another reinvite? */
677 #define SIP_PENDINGBYE (1 << 6) /*!< Need to send bye after we ack? */
678 #define SIP_GOTREFER (1 << 7) /*!< Got a refer? */
679 #define SIP_PROMISCREDIR (1 << 8) /*!< Promiscuous redirection */
680 #define SIP_TRUSTRPID (1 << 9) /*!< Trust RPID headers? */
681 #define SIP_USEREQPHONE (1 << 10) /*!< Add user=phone to numeric URI. Default off */
682 #define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
683 #define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
684 #define SIP_OUTGOING (1 << 13) /*!< Is this an outgoing call? */
685 #define SIP_CAN_BYE (1 << 14) /*!< Can we send BYE on this dialog? */
686 #define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */
687 #define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
688 #define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */
689 #define SIP_DTMF_INBAND (1 << 16) /*!< DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
690 #define SIP_DTMF_INFO (2 << 16) /*!< DTMF Support: SIP Info messages - "info" */
691 #define SIP_DTMF_AUTO (3 << 16) /*!< DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
693 #define SIP_NAT (3 << 18) /*!< four settings, uses two bits */
694 #define SIP_NAT_NEVER (0 << 18) /*!< No nat support */
695 #define SIP_NAT_RFC3581 (1 << 18) /*!< NAT RFC3581 */
696 #define SIP_NAT_ROUTE (2 << 18) /*!< NAT Only ROUTE */
697 #define SIP_NAT_ALWAYS (3 << 18) /*!< NAT Both ROUTE and RFC3581 */
698 /* re-INVITE related settings */
699 #define SIP_REINVITE (7 << 20) /*!< three bits used */
700 #define SIP_CAN_REINVITE (1 << 20) /*!< allow peers to be reinvited to send media directly p2p */
701 #define SIP_CAN_REINVITE_NAT (2 << 20) /*!< allow media reinvite when new peer is behind NAT */
702 #define SIP_REINVITE_UPDATE (4 << 20) /*!< use UPDATE (RFC3311) when reinviting this peer */
703 /* "insecure" settings */
704 #define SIP_INSECURE_PORT (1 << 23) /*!< don't require matching port for incoming requests */
705 #define SIP_INSECURE_INVITE (1 << 24) /*!< don't require authentication for incoming INVITEs */
706 /* Sending PROGRESS in-band settings */
707 #define SIP_PROG_INBAND (3 << 25) /*!< three settings, uses two bits */
708 #define SIP_PROG_INBAND_NEVER (0 << 25)
709 #define SIP_PROG_INBAND_NO (1 << 25)
710 #define SIP_PROG_INBAND_YES (2 << 25)
711 #define SIP_FREE_BIT (1 << 27) /*!< Undefined bit - not in use */
712 #define SIP_CALL_LIMIT (1 << 28) /*!< Call limit enforced for this call */
713 #define SIP_SENDRPID (1 << 29) /*!< Remote Party-ID Support */
714 #define SIP_INC_COUNT (1 << 30) /*!< Did this connection increment the counter of in-use calls? */
715 #define SIP_G726_NONSTANDARD (1 << 31) /*!< Use non-standard packing for G726-32 data */
717 #define SIP_FLAGS_TO_COPY \
718 (SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
719 SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
720 SIP_USEREQPHONE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE)
722 /*--- a new page of flags (for flags[1] */
724 #define SIP_PAGE2_RTCACHEFRIENDS (1 << 0)
725 #define SIP_PAGE2_RTUPDATE (1 << 1)
726 #define SIP_PAGE2_RTAUTOCLEAR (1 << 2)
727 #define SIP_PAGE2_RT_FROMCONTACT (1 << 4)
728 #define SIP_PAGE2_RTSAVE_SYSNAME (1 << 5)
729 /* Space for addition of other realtime flags in the future */
730 #define SIP_PAGE2_IGNOREREGEXPIRE (1 << 10)
731 #define SIP_PAGE2_DEBUG (3 << 11)
732 #define SIP_PAGE2_DEBUG_CONFIG (1 << 11)
733 #define SIP_PAGE2_DEBUG_CONSOLE (1 << 12)
734 #define SIP_PAGE2_DYNAMIC (1 << 13) /*!< Dynamic Peers register with Asterisk */
735 #define SIP_PAGE2_SELFDESTRUCT (1 << 14) /*!< Automatic peers need to destruct themselves */
736 #define SIP_PAGE2_VIDEOSUPPORT (1 << 15)
737 #define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 16) /*!< Allow subscriptions from this peer? */
738 #define SIP_PAGE2_ALLOWOVERLAP (1 << 17) /*!< Allow overlap dialing ? */
739 #define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 18) /*!< Only issue MWI notification if subscribed to */
740 #define SIP_PAGE2_INC_RINGING (1 << 19) /*!< Did this connection increment the counter of in-use calls? */
741 #define SIP_PAGE2_T38SUPPORT (7 << 20) /*!< T38 Fax Passthrough Support */
742 #define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 20) /*!< 20: T38 Fax Passthrough Support */
743 #define SIP_PAGE2_T38SUPPORT_RTP (2 << 20) /*!< 21: T38 Fax Passthrough Support */
744 #define SIP_PAGE2_T38SUPPORT_TCP (4 << 20) /*!< 22: T38 Fax Passthrough Support */
745 #define SIP_PAGE2_CALL_ONHOLD (3 << 23) /*!< Call states */
746 #define SIP_PAGE2_CALL_ONHOLD_ONEDIR (1 << 23) /*!< 23: One directional hold */
747 #define SIP_PAGE2_CALL_ONHOLD_INACTIVE (2 << 24) /*!< 24: Inactive */
748 #define SIP_PAGE2_RFC2833_COMPENSATE (1 << 26)
750 #define SIP_PAGE2_FLAGS_TO_COPY \
751 (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE)
753 /* SIP packet flags */
754 #define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
755 #define SIP_PKT_WITH_TOTAG (1 << 1) /*!< This packet has a to-tag */
756 #define SIP_PKT_IGNORE (1 << 2) /*!< This is a re-transmit, ignore it */
757 #define SIP_PKT_IGNORE_RESP (1 << 3) /*!< Resp ignore - ??? */
758 #define SIP_PKT_IGNORE_REQ (1 << 4) /*!< Req ignore - ??? */
760 /* T.38 set of flags */
761 #define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0 (unset)*/
762 #define T38FAX_TRANSCODING_MMR (1 << 1) /*!< Default: 0 (unset)*/
763 #define T38FAX_TRANSCODING_JBIG (1 << 2) /*!< Default: 0 (unset)*/
764 /* Rate management */
765 #define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF (0 << 3)
766 #define T38FAX_RATE_MANAGEMENT_LOCAL_TCF (1 << 3) /*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */
767 /* UDP Error correction */
768 #define T38FAX_UDP_EC_NONE (0 << 4) /*!< two bits, if unset NO t38UDPEC field in T38 SDP*/
769 #define T38FAX_UDP_EC_FEC (1 << 4) /*!< Set for t38UDPFEC */
770 #define T38FAX_UDP_EC_REDUNDANCY (2 << 4) /*!< Set for t38UDPRedundancy */
771 /* T38 Spec version */
772 #define T38FAX_VERSION (3 << 6) /*!< two bits, 2 values so far, up to 4 values max */
773 #define T38FAX_VERSION_0 (0 << 6) /*!< Version 0 */
774 #define T38FAX_VERSION_1 (1 << 6) /*!< Version 1 */
775 /* Maximum Fax Rate */
776 #define T38FAX_RATE_2400 (1 << 8) /*!< 2400 bps t38FaxRate */
777 #define T38FAX_RATE_4800 (1 << 9) /*!< 4800 bps t38FaxRate */
778 #define T38FAX_RATE_7200 (1 << 10) /*!< 7200 bps t38FaxRate */
779 #define T38FAX_RATE_9600 (1 << 11) /*!< 9600 bps t38FaxRate */
780 #define T38FAX_RATE_12000 (1 << 12) /*!< 12000 bps t38FaxRate */
781 #define T38FAX_RATE_14400 (1 << 13) /*!< 14400 bps t38FaxRate */
783 /*!< This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
784 static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
786 #define sipdebug ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
787 #define sipdebug_config ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)
788 #define sipdebug_console ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)
790 /*! \brief T38 States for a call */
792 T38_DISABLED = 0, /*!< Not enabled */
793 T38_LOCAL_DIRECT, /*!< Offered from local */
794 T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */
795 T38_PEER_DIRECT, /*!< Offered from peer */
796 T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */
797 T38_ENABLED /*!< Negotiated (enabled) */
800 /*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
801 struct t38properties {
802 struct ast_flags t38support; /*!< Flag for udptl, rtp or tcp support for this session */
803 int capability; /*!< Our T38 capability */
804 int peercapability; /*!< Peers T38 capability */
805 int jointcapability; /*!< Supported T38 capability at both ends */
806 enum t38state state; /*!< T.38 state */
809 /*! \brief Parameters to know status of transfer */
811 REFER_IDLE, /*!< No REFER is in progress */
812 REFER_SENT, /*!< Sent REFER to transferee */
813 REFER_RECEIVED, /*!< Received REFER from transferer */
814 REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING */
815 REFER_ACCEPTED, /*!< Accepted by transferee */
816 REFER_RINGING, /*!< Target Ringing */
817 REFER_200OK, /*!< Answered by transfer target */
818 REFER_FAILED, /*!< REFER declined - go on */
819 REFER_NOAUTH /*!< We had no auth for REFER */
822 static const struct c_referstatusstring {
823 enum referstatus status;
825 } referstatusstrings[] = {
826 { REFER_IDLE, "<none>" },
827 { REFER_SENT, "Request sent" },
828 { REFER_RECEIVED, "Request received" },
829 { REFER_ACCEPTED, "Accepted" },
830 { REFER_RINGING, "Target ringing" },
831 { REFER_200OK, "Done" },
832 { REFER_FAILED, "Failed" },
833 { REFER_NOAUTH, "Failed - auth failure" }
836 /*! \brief Structure to handle SIP transfers. Dynamically allocated when needed */
837 /* OEJ: Should be moved to string fields */
839 char refer_to[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO extension */
840 char refer_to_domain[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO domain */
841 char refer_to_urioption[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO uri options */
842 char refer_to_context[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO context */
843 char referred_by[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
844 char referred_by_name[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
845 char refer_contact[AST_MAX_EXTENSION]; /*!< Place to store Contact info from a REFER extension */
846 char replaces_callid[BUFSIZ]; /*!< Replace info: callid */
847 char replaces_callid_totag[BUFSIZ/2]; /*!< Replace info: to-tag */
848 char replaces_callid_fromtag[BUFSIZ/2]; /*!< Replace info: from-tag */
849 struct sip_pvt *refer_call; /*!< Call we are referring */
850 int attendedtransfer; /*!< Attended or blind transfer? */
851 int localtransfer; /*!< Transfer to local domain? */
852 enum referstatus status; /*!< REFER status */
855 /*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe */
856 static struct sip_pvt {
857 ast_mutex_t lock; /*!< Dialog private lock */
858 int method; /*!< SIP method that opened this dialog */
859 AST_DECLARE_STRING_FIELDS(
860 AST_STRING_FIELD(callid); /*!< Global CallID */
861 AST_STRING_FIELD(randdata); /*!< Random data */
862 AST_STRING_FIELD(accountcode); /*!< Account code */
863 AST_STRING_FIELD(realm); /*!< Authorization realm */
864 AST_STRING_FIELD(nonce); /*!< Authorization nonce */
865 AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
866 AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
867 AST_STRING_FIELD(domain); /*!< Authorization domain */
868 AST_STRING_FIELD(from); /*!< The From: header */
869 AST_STRING_FIELD(useragent); /*!< User agent in SIP request */
870 AST_STRING_FIELD(exten); /*!< Extension where to start */
871 AST_STRING_FIELD(context); /*!< Context for this call */
872 AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */
873 AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */
874 AST_STRING_FIELD(fromdomain); /*!< Domain to show in the from field */
875 AST_STRING_FIELD(fromuser); /*!< User to show in the user field */
876 AST_STRING_FIELD(fromname); /*!< Name to show in the user field */
877 AST_STRING_FIELD(tohost); /*!< Host we should put in the "to" field */
878 AST_STRING_FIELD(language); /*!< Default language for this call */
879 AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */
880 AST_STRING_FIELD(mohsuggest); /*!< MOH class to suggest when putting a peer on hold */
881 AST_STRING_FIELD(rdnis); /*!< Referring DNIS */
882 AST_STRING_FIELD(theirtag); /*!< Their tag */
883 AST_STRING_FIELD(username); /*!< [user] name */
884 AST_STRING_FIELD(peername); /*!< [peer] name, not set if [user] */
885 AST_STRING_FIELD(authname); /*!< Who we use for authentication */
886 AST_STRING_FIELD(uri); /*!< Original requested URI */
887 AST_STRING_FIELD(okcontacturi); /*!< URI from the 200 OK on INVITE */
888 AST_STRING_FIELD(peersecret); /*!< Password */
889 AST_STRING_FIELD(peermd5secret);
890 AST_STRING_FIELD(cid_num); /*!< Caller*ID number */
891 AST_STRING_FIELD(cid_name); /*!< Caller*ID name */
892 AST_STRING_FIELD(via); /*!< Via: header */
893 AST_STRING_FIELD(fullcontact); /*!< The Contact: that the UA registers with us */
894 AST_STRING_FIELD(our_contact); /*!< Our contact header */
895 AST_STRING_FIELD(rpid); /*!< Our RPID header */
896 AST_STRING_FIELD(rpid_from); /*!< Our RPID From header */
898 unsigned int ocseq; /*!< Current outgoing seqno */
899 unsigned int icseq; /*!< Current incoming seqno */
900 ast_group_t callgroup; /*!< Call group */
901 ast_group_t pickupgroup; /*!< Pickup group */
902 int lastinvite; /*!< Last Cseq of invite */
903 struct ast_flags flags[2]; /*!< SIP_ flags */
904 int timer_t1; /*!< SIP timer T1, ms rtt */
905 unsigned int sipoptions; /*!< Supported SIP options on the other end */
906 struct ast_codec_pref prefs; /*!< codec prefs */
907 int capability; /*!< Special capability (codec) */
908 int jointcapability; /*!< Supported capability at both ends (codecs ) */
909 int peercapability; /*!< Supported peer capability */
910 int prefcodec; /*!< Preferred codec (outbound only) */
911 int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
912 int redircodecs; /*!< Redirect codecs */
913 int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
914 struct t38properties t38; /*!< T38 settings */
915 struct sockaddr_in udptlredirip; /*!< Where our T.38 UDPTL should be going if not to us */
916 struct ast_udptl *udptl; /*!< T.38 UDPTL session */
917 int callingpres; /*!< Calling presentation */
918 int authtries; /*!< Times we've tried to authenticate */
919 int expiry; /*!< How long we take to expire */
920 long branch; /*!< The branch identifier of this session */
921 char tag[11]; /*!< Our tag for this session */
922 int sessionid; /*!< SDP Session ID */
923 int sessionversion; /*!< SDP Session Version */
924 struct sockaddr_in sa; /*!< Our peer */
925 struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
926 struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
927 time_t lastrtprx; /*!< Last RTP received */
928 time_t lastrtptx; /*!< Last RTP sent */
929 int rtptimeout; /*!< RTP timeout time */
930 int rtpholdtimeout; /*!< RTP timeout when on hold */
931 int rtpkeepalive; /*!< Send RTP packets for keepalive */
932 struct sockaddr_in recv; /*!< Received as */
933 struct in_addr ourip; /*!< Our IP */
934 struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
935 struct sip_route *route; /*!< Head of linked list of routing steps (fm Record-Route) */
936 int route_persistant; /*!< Is this the "real" route? */
937 struct sip_auth *peerauth; /*!< Realm authentication */
938 int noncecount; /*!< Nonce-count */
939 char lastmsg[256]; /*!< Last Message sent/received */
940 int amaflags; /*!< AMA Flags */
941 int pendinginvite; /*!< Any pending invite ? (seqno of this) */
942 struct sip_request initreq; /*!< Initial request that opened the SIP dialog */
944 int maxtime; /*!< Max time for first response */
945 int initid; /*!< Auto-congest ID if appropriate (scheduler) */
946 int autokillid; /*!< Auto-kill ID (scheduler) */
947 enum transfermodes allowtransfer; /*!< REFER: restriction scheme */
948 struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
949 enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
950 int stateid; /*!< SUBSCRIBE: ID for devicestate subscriptions */
951 int laststate; /*!< SUBSCRIBE: Last known extension state */
952 int dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */
954 struct ast_dsp *vad; /*!< Voice Activation Detection dsp */
956 struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one
957 Used in peerpoke, mwi subscriptions */
958 struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */
959 struct ast_rtp *rtp; /*!< RTP Session */
960 struct ast_rtp *vrtp; /*!< Video RTP session */
961 struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
962 struct sip_history_head *history; /*!< History of this SIP dialog */
963 struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
964 struct sip_pvt *next; /*!< Next dialog in chain */
965 struct sip_invite_param *options; /*!< Options for INVITE */
969 #define FLAG_RESPONSE (1 << 0)
970 #define FLAG_FATAL (1 << 1)
972 /*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission */
974 struct sip_pkt *next; /*!< Next packet in linked list */
975 int retrans; /*!< Retransmission number */
976 int method; /*!< SIP method for this packet */
977 int seqno; /*!< Sequence number */
978 unsigned int flags; /*!< non-zero if this is a response packet (e.g. 200 OK) */
979 struct sip_pvt *owner; /*!< Owner AST call */
980 int retransid; /*!< Retransmission ID */
981 int timer_a; /*!< SIP timer A, retransmission timer */
982 int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */
983 int packetlen; /*!< Length of packet */
987 /*! \brief Structure for SIP user data. User's place calls to us */
989 /* Users who can access various contexts */
990 ASTOBJ_COMPONENTS(struct sip_user);
991 char secret[80]; /*!< Password */
992 char md5secret[80]; /*!< Password in md5 */
993 char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
994 char subscribecontext[AST_MAX_CONTEXT]; /* Default context for subscriptions */
995 char cid_num[80]; /*!< Caller ID num */
996 char cid_name[80]; /*!< Caller ID name */
997 char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */
998 char language[MAX_LANGUAGE]; /*!< Default language for this user */
999 char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
1000 char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
1001 char useragent[256]; /*!< User agent in SIP request */
1002 struct ast_codec_pref prefs; /*!< codec prefs */
1003 ast_group_t callgroup; /*!< Call group */
1004 ast_group_t pickupgroup; /*!< Pickup Group */
1005 unsigned int sipoptions; /*!< Supported SIP options */
1006 struct ast_flags flags[2]; /*!< SIP_ flags */
1007 int amaflags; /*!< AMA flags for billing */
1008 int callingpres; /*!< Calling id presentation */
1009 int capability; /*!< Codec capability */
1010 int inUse; /*!< Number of calls in use */
1011 int call_limit; /*!< Limit of concurrent calls */
1012 enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
1013 struct ast_ha *ha; /*!< ACL setting */
1014 struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
1015 int maxcallbitrate; /*!< Maximum Bitrate for a video call */
1019 /*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host) */
1020 /* XXX field 'name' must be first otherwise sip_addrcmp() will fail */
1022 ASTOBJ_COMPONENTS(struct sip_peer); /*!< name, refcount, objflags, object pointers */
1023 /*!< peer->name is the unique name of this object */
1024 char secret[80]; /*!< Password */
1025 char md5secret[80]; /*!< Password in MD5 */
1026 struct sip_auth *auth; /*!< Realm authentication list */
1027 char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
1028 char subscribecontext[AST_MAX_CONTEXT]; /*!< Default context for subscriptions */
1029 char username[80]; /*!< Temporary username until registration */
1030 char accountcode[AST_MAX_ACCOUNT_CODE]; /*!< Account code */
1031 int amaflags; /*!< AMA Flags (for billing) */
1032 char tohost[MAXHOSTNAMELEN]; /*!< If not dynamic, IP address */
1033 char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */
1034 char fromuser[80]; /*!< From: user when calling this peer */
1035 char fromdomain[MAXHOSTNAMELEN]; /*!< From: domain when calling this peer */
1036 char fullcontact[256]; /*!< Contact registered with us (not in sip.conf) */
1037 char cid_num[80]; /*!< Caller ID num */
1038 char cid_name[80]; /*!< Caller ID name */
1039 int callingpres; /*!< Calling id presentation */
1040 int inUse; /*!< Number of calls in use */
1041 int inRinging; /*!< Number of calls ringing */
1042 int onHold; /*!< Peer has someone on hold */
1043 int call_limit; /*!< Limit of concurrent calls */
1044 enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
1045 char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
1046 char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
1047 char language[MAX_LANGUAGE]; /*!< Default language for prompts */
1048 char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
1049 char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
1050 char useragent[256]; /*!< User agent in SIP request (saved from registration) */
1051 struct ast_codec_pref prefs; /*!< codec prefs */
1053 time_t lastmsgcheck; /*!< Last time we checked for MWI */
1054 unsigned int sipoptions; /*!< Supported SIP options */
1055 struct ast_flags flags[2]; /*!< SIP_ flags */
1056 int expire; /*!< When to expire this peer registration */
1057 int capability; /*!< Codec capability */
1058 int rtptimeout; /*!< RTP timeout */
1059 int rtpholdtimeout; /*!< RTP Hold Timeout */
1060 int rtpkeepalive; /*!< Send RTP packets for keepalive */
1061 ast_group_t callgroup; /*!< Call group */
1062 ast_group_t pickupgroup; /*!< Pickup group */
1063 struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
1064 struct sockaddr_in addr; /*!< IP address of peer */
1065 int maxcallbitrate; /*!< Maximum Bitrate for a video call */
1068 struct sip_pvt *call; /*!< Call pointer */
1069 int pokeexpire; /*!< When to expire poke (qualify= checking) */
1070 int lastms; /*!< How long last response took (in ms), or -1 for no response */
1071 int maxms; /*!< Max ms we will accept for the host to be up, 0 to not monitor */
1072 struct timeval ps; /*!< Ping send time */
1074 struct sockaddr_in defaddr; /*!< Default IP address, used until registration */
1075 struct ast_ha *ha; /*!< Access control list */
1076 struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
1077 struct sip_pvt *mwipvt; /*!< Subscription for MWI */
1084 /*! \brief Registrations with other SIP proxies */
1085 struct sip_registry {
1086 ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1);
1087 AST_DECLARE_STRING_FIELDS(
1088 AST_STRING_FIELD(callid); /*!< Global Call-ID */
1089 AST_STRING_FIELD(realm); /*!< Authorization realm */
1090 AST_STRING_FIELD(nonce); /*!< Authorization nonce */
1091 AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
1092 AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
1093 AST_STRING_FIELD(domain); /*!< Authorization domain */
1094 AST_STRING_FIELD(username); /*!< Who we are registering as */
1095 AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */
1096 AST_STRING_FIELD(hostname); /*!< Domain or host we register to */
1097 AST_STRING_FIELD(secret); /*!< Password in clear text */
1098 AST_STRING_FIELD(md5secret); /*!< Password in md5 */
1099 AST_STRING_FIELD(contact); /*!< Contact extension */
1100 AST_STRING_FIELD(random);
1102 int portno; /*!< Optional port override */
1103 int expire; /*!< Sched ID of expiration */
1104 int regattempts; /*!< Number of attempts (since the last success) */
1105 int timeout; /*!< sched id of sip_reg_timeout */
1106 int refresh; /*!< How often to refresh */
1107 struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
1108 enum sipregistrystate regstate; /*!< Registration state (see above) */
1109 time_t regtime; /*!< Last succesful registration time */
1110 int callid_valid; /*!< 0 means we haven't chosen callid for this registry yet. */
1111 unsigned int ocseq; /*!< Sequence number we got to for REGISTERs for this registry */
1112 struct sockaddr_in us; /*!< Who the server thinks we are */
1113 int noncecount; /*!< Nonce-count */
1114 char lastmsg[256]; /*!< Last Message sent/received */
1117 /* --- Linked lists of various objects --------*/
1119 /*! \brief The user list: Users and friends */
1120 static struct ast_user_list {
1121 ASTOBJ_CONTAINER_COMPONENTS(struct sip_user);
1124 /*! \brief The peer list: Peers and Friends */
1125 static struct ast_peer_list {
1126 ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer);
1129 /*! \brief The register list: Other SIP proxys we register with and place calls to */
1130 static struct ast_register_list {
1131 ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
1135 /*! \todo Move the sip_auth list to AST_LIST */
1136 static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */
1139 /* --- Sockets and networking --------------*/
1140 static int sipsock = -1; /*!< Main socket for SIP network communication */
1141 static struct sockaddr_in bindaddr = { 0, }; /*!< The address we bind to */
1142 static struct sockaddr_in externip; /*!< External IP address if we are behind NAT */
1143 static char externhost[MAXHOSTNAMELEN]; /*!< External host name (possibly with dynamic DNS and DHCP */
1144 static time_t externexpire = 0; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
1145 static int externrefresh = 10;
1146 static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
1147 static struct in_addr __ourip;
1148 static struct sockaddr_in outboundproxyip;
1150 static struct sockaddr_in debugaddr;
1152 static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
1154 /*---------------------------- Forward declarations of functions in chan_sip.c */
1155 /*! \note This is added to help splitting up chan_sip.c into several files
1156 in coming releases */
1158 /*--- PBX interface functions */
1159 static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
1160 static int sip_devicestate(void *data);
1161 static int sip_sendtext(struct ast_channel *ast, const char *text);
1162 static int sip_call(struct ast_channel *ast, char *dest, int timeout);
1163 static int sip_hangup(struct ast_channel *ast);
1164 static int sip_answer(struct ast_channel *ast);
1165 static struct ast_frame *sip_read(struct ast_channel *ast);
1166 static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
1167 static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
1168 static int sip_transfer(struct ast_channel *ast, const char *dest);
1169 static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
1170 static int sip_senddigit_begin(struct ast_channel *ast, char digit);
1171 static int sip_senddigit_end(struct ast_channel *ast, char digit);
1173 /*--- Transmitting responses and requests */
1174 static int sipsock_read(int *id, int fd, short events, void *ignore);
1175 static int __sip_xmit(struct sip_pvt *p, char *data, int len);
1176 static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod);
1177 static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1178 static int retrans_pkt(void *data);
1179 static int transmit_sip_request(struct sip_pvt *p, struct sip_request *req);
1180 static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1181 static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1182 static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1183 static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1184 static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
1185 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);
1186 static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1187 static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, int reliable);
1188 static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
1189 static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
1190 static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init);
1191 static int transmit_reinvite_with_sdp(struct sip_pvt *p);
1192 static int transmit_info_with_digit(struct sip_pvt *p, const char digit);
1193 static int transmit_info_with_vidupdate(struct sip_pvt *p);
1194 static int transmit_message_with_text(struct sip_pvt *p, const char *text);
1195 static int transmit_refer(struct sip_pvt *p, const char *dest);
1196 static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten);
1197 static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
1198 static int transmit_state_notify(struct sip_pvt *p, int state, int full);
1199 static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
1200 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
1201 static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
1202 static void copy_request(struct sip_request *dst, const struct sip_request *src);
1203 static void receive_message(struct sip_pvt *p, struct sip_request *req);
1204 static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req);
1205 static int sip_send_mwi_to_peer(struct sip_peer *peer);
1206 static int does_peer_need_mwi(struct sip_peer *peer);
1208 /*--- Dialog management */
1209 static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
1210 int useglobal_nat, const int intended_method);
1211 static int __sip_autodestruct(void *data);
1212 static void sip_scheddestroy(struct sip_pvt *p, int ms);
1213 static void sip_cancel_destroy(struct sip_pvt *p);
1214 static void sip_destroy(struct sip_pvt *p);
1215 static void __sip_destroy(struct sip_pvt *p, int lockowner);
1216 static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod, int reset);
1217 static void __sip_pretend_ack(struct sip_pvt *p);
1218 static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
1219 static int auto_congest(void *nothing);
1220 static int update_call_counter(struct sip_pvt *fup, int event);
1221 static int hangup_sip2cause(int cause);
1222 static const char *hangup_cause2sip(int cause);
1223 static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method);
1224 static void free_old_route(struct sip_route *route);
1225 static void list_route(struct sip_route *route);
1226 static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards);
1227 static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin,
1228 struct sip_request *req, char *uri);
1229 static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
1230 static void check_pendings(struct sip_pvt *p);
1231 static void *sip_park_thread(void *stuff);
1232 static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno);
1233 static int sip_sipredirect(struct sip_pvt *p, const char *dest);
1235 /*--- Codec handling / SDP */
1236 static void try_suggested_sip_codec(struct sip_pvt *p);
1237 static const char* get_sdp_iterate(int* start, struct sip_request *req, const char *name);
1238 static const char *get_sdp(struct sip_request *req, const char *name);
1239 static int find_sdp(struct sip_request *req);
1240 static int process_sdp(struct sip_pvt *p, struct sip_request *req);
1241 static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
1242 char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
1244 static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
1245 char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
1247 static int add_sdp(struct sip_request *resp, struct sip_pvt *p);
1249 /*--- Authentication stuff */
1250 static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, int sipmethod, int init);
1251 static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
1252 static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
1253 static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
1254 static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); /* Add realm authentication in list */
1255 static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm); /* Find authentication for a specific realm */
1256 static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
1257 const char *secret, const char *md5secret, int sipmethod,
1258 char *uri, enum xmittype reliable, int ignore);
1259 static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
1260 int sipmethod, char *uri, enum xmittype reliable,
1261 struct sockaddr_in *sin, struct sip_peer **authpeer);
1262 static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);
1263 static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, int sipmethod, int init);
1264 static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int digest_len);
1266 /*--- Domain handling */
1267 static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
1268 static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context);
1269 static void clear_sip_domains(void);
1271 /*--- SIP realm authentication */
1272 static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno);
1273 static int clear_realm_authentication(struct sip_auth *authlist);
1274 static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm);
1276 /*--- Misc functions */
1277 static int sip_do_reload(enum channelreloadreason reason);
1278 static int reload_config(enum channelreloadreason reason);
1279 static int expire_register(void *data);
1280 static int sip_sipredirect(struct sip_pvt *p, const char *dest);
1281 static void *do_monitor(void *data);
1282 static int restart_monitor(void);
1283 static int sip_send_mwi_to_peer(struct sip_peer *peer);
1284 static void sip_destroy(struct sip_pvt *p);
1285 static int sip_addrcmp(char *name, struct sockaddr_in *sin); /* Support for peer matching */
1286 static int sip_refer_allocate(struct sip_pvt *p);
1287 static void ast_quiet_chan(struct ast_channel *chan);
1288 static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
1290 /*--- Device monitoring and Device/extension state handling */
1291 static int cb_extensionstate(char *context, char* exten, int state, void *data);
1292 static int sip_devicestate(void *data);
1293 static int sip_poke_noanswer(void *data);
1294 static int sip_poke_peer(struct sip_peer *peer);
1295 static void sip_poke_all_peers(void);
1296 static void sip_peer_hold(struct sip_pvt *p, int hold);
1298 /*--- Applications, functions, CLI and manager command helpers */
1299 static const char *sip_nat_mode(const struct sip_pvt *p);
1300 static int sip_show_inuse(int fd, int argc, char *argv[]);
1301 static char *transfermode2str(enum transfermodes mode) attribute_const;
1302 static char *nat2str(int nat) attribute_const;
1303 static int peer_status(struct sip_peer *peer, char *status, int statuslen);
1304 static int sip_show_users(int fd, int argc, char *argv[]);
1305 static int _sip_show_peers(int fd, int *total, struct mansession *s, struct message *m, int argc, char *argv[]);
1306 static int manager_sip_show_peers( struct mansession *s, struct message *m );
1307 static int sip_show_peers(int fd, int argc, char *argv[]);
1308 static int sip_show_objects(int fd, int argc, char *argv[]);
1309 static void print_group(int fd, ast_group_t group, int crlf);
1310 static const char *dtmfmode2str(int mode) attribute_const;
1311 static const char *insecure2str(int port, int invite) attribute_const;
1312 static void cleanup_stale_contexts(char *new, char *old);
1313 static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
1314 static const char *domain_mode_to_text(const enum domain_mode mode);
1315 static int sip_show_domains(int fd, int argc, char *argv[]);
1316 static int _sip_show_peer(int type, int fd, struct mansession *s, struct message *m, int argc, char *argv[]);
1317 static int manager_sip_show_peer( struct mansession *s, struct message *m);
1318 static int sip_show_peer(int fd, int argc, char *argv[]);
1319 static int _sip_show_peer(int type, int fd, struct mansession *s, struct message *m, int argc, char *argv[]);
1320 static int sip_show_user(int fd, int argc, char *argv[]);
1321 static int sip_show_registry(int fd, int argc, char *argv[]);
1322 static int sip_show_settings(int fd, int argc, char *argv[]);
1323 static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
1324 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
1325 static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
1326 static int sip_show_channels(int fd, int argc, char *argv[]);
1327 static int sip_show_subscriptions(int fd, int argc, char *argv[]);
1328 static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
1329 static char *complete_sipch(const char *line, const char *word, int pos, int state);
1330 static char *complete_sip_peer(const char *word, int state, int flags2);
1331 static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
1332 static char *complete_sip_debug_peer(const char *line, const char *word, int pos, int state);
1333 static char *complete_sip_user(const char *word, int state, int flags2);
1334 static char *complete_sip_show_user(const char *line, const char *word, int pos, int state);
1335 static char *complete_sipnotify(const char *line, const char *word, int pos, int state);
1336 static char *complete_sip_prune_realtime_peer(const char *line, const char *word, int pos, int state);
1337 static char *complete_sip_prune_realtime_user(const char *line, const char *word, int pos, int state);
1338 static int sip_show_channel(int fd, int argc, char *argv[]);
1339 static int sip_show_history(int fd, int argc, char *argv[]);
1340 static int sip_do_debug_ip(int fd, int argc, char *argv[]);
1341 static int sip_do_debug_peer(int fd, int argc, char *argv[]);
1342 static int sip_do_debug(int fd, int argc, char *argv[]);
1343 static int sip_no_debug(int fd, int argc, char *argv[]);
1344 static int sip_notify(int fd, int argc, char *argv[]);
1345 static int sip_do_history(int fd, int argc, char *argv[]);
1346 static int sip_no_history(int fd, int argc, char *argv[]);
1347 static int func_header_read(struct ast_channel *chan, char *function, char *data, char *buf, size_t len);
1348 static int func_check_sipdomain(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len);
1349 static int function_sippeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len);
1350 static int function_sipchaninfo_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len);
1351 static int sip_dtmfmode(struct ast_channel *chan, void *data);
1352 static int sip_addheader(struct ast_channel *chan, void *data);
1353 static int sip_do_reload(enum channelreloadreason reason);
1354 static int sip_reload(int fd, int argc, char *argv[]);
1357 Functions for enabling debug per IP or fully, or enabling history logging for
1360 static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to LOG_DEBUG at end of dialog, before destroying data */
1361 static inline int sip_debug_test_addr(const struct sockaddr_in *addr);
1362 static inline int sip_debug_test_pvt(struct sip_pvt *p);
1363 static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
1364 static void sip_dump_history(struct sip_pvt *dialog);
1366 /*--- Device object handling */
1367 static struct sip_peer *temp_peer(const char *name);
1368 static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime);
1369 static struct sip_user *build_user(const char *name, struct ast_variable *v, int realtime);
1370 static int update_call_counter(struct sip_pvt *fup, int event);
1371 static void sip_destroy_peer(struct sip_peer *peer);
1372 static void sip_destroy_user(struct sip_user *user);
1373 static int sip_poke_peer(struct sip_peer *peer);
1374 static void set_peer_defaults(struct sip_peer *peer);
1375 static struct sip_peer *temp_peer(const char *name);
1376 static void register_peer_exten(struct sip_peer *peer, int onoff);
1377 static void sip_destroy_peer(struct sip_peer *peer);
1378 static void sip_destroy_user(struct sip_user *user);
1379 static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime);
1380 static struct sip_user *find_user(const char *name, int realtime);
1381 static int sip_poke_peer_s(void *data);
1382 static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
1383 static int expire_register(void *data);
1384 static void reg_source_db(struct sip_peer *peer);
1385 static void destroy_association(struct sip_peer *peer);
1386 static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);
1388 /* Realtime device support */
1389 static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey);
1390 static struct sip_user *realtime_user(const char *username);
1391 static void update_peer(struct sip_peer *p, int expiry);
1392 static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin);
1393 static int sip_prune_realtime(int fd, int argc, char *argv[]);
1395 /*--- Internal UA client handling (outbound registrations) */
1396 static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us);
1397 static void sip_registry_destroy(struct sip_registry *reg);
1398 static int sip_register(char *value, int lineno);
1399 static char *regstate2str(enum sipregistrystate regstate) attribute_const;
1400 static int sip_reregister(void *data);
1401 static int __sip_do_register(struct sip_registry *r);
1402 static int sip_reg_timeout(void *data);
1403 static int do_register_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader);
1404 static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
1405 static void sip_send_all_registers(void);
1407 /*--- Parsing SIP requests and responses */
1408 static void append_date(struct sip_request *req); /* Append date to SIP packet */
1409 static int determine_firstline_parts(struct sip_request *req);
1410 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
1411 static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
1412 static int find_sip_method(const char *msg);
1413 static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported);
1414 static void parse_request(struct sip_request *req);
1415 static const char *get_header(const struct sip_request *req, const char *name);
1416 static char *referstatus2str(enum referstatus rstatus) attribute_pure;
1417 static int method_match(enum sipmethod id, const char *name);
1418 static void parse_copy(struct sip_request *dst, const struct sip_request *src);
1419 static char *get_in_brackets(char *tmp);
1420 static const char *find_alias(const char *name, const char *_default);
1421 static const char *__get_header(const struct sip_request *req, const char *name, int *start);
1422 static const char *get_header(const struct sip_request *req, const char *name);
1423 static int lws2sws(char *msgbuf, int len);
1424 static void extract_uri(struct sip_pvt *p, struct sip_request *req);
1425 static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req);
1426 static int get_also_info(struct sip_pvt *p, struct sip_request *oreq);
1427 static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
1428 static int set_address_from_contact(struct sip_pvt *pvt);
1429 static void check_via(struct sip_pvt *p, struct sip_request *req);
1430 static char *get_calleridname(const char *input, char *output, size_t outputsize);
1431 static int get_rpid_num(const char *input, char *output, int maxlen);
1432 static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq);
1433 static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
1434 static int get_msg_text(char *buf, int len, struct sip_request *req);
1435 static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
1436 static void free_old_route(struct sip_route *route);
1438 /*--- Constructing requests and responses */
1439 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
1440 static int init_req(struct sip_request *req, int sipmethod, const char *recip);
1441 static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch);
1442 static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod);
1443 static int init_resp(struct sip_request *resp, const char *msg);
1444 static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req);
1445 static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p);
1446 static void build_via(struct sip_pvt *p);
1447 static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
1448 static int create_addr(struct sip_pvt *dialog, const char *opeer);
1449 static char *generate_random_string(char *buf, size_t size);
1450 static void build_callid_pvt(struct sip_pvt *pvt);
1451 static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
1452 static void make_our_tag(char *tagbuf, size_t len);
1453 static int add_header(struct sip_request *req, const char *var, const char *value);
1454 static int add_header_contentLength(struct sip_request *req, int len);
1455 static int add_line(struct sip_request *req, const char *line);
1456 static int add_text(struct sip_request *req, const char *text);
1457 static int add_digit(struct sip_request *req, char digit);
1458 static int add_vidupdate(struct sip_request *req);
1459 static void add_route(struct sip_request *req, struct sip_route *route);
1460 static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field);
1461 static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field);
1462 static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field);
1463 static void set_destination(struct sip_pvt *p, char *uri);
1464 static void append_date(struct sip_request *req);
1465 static void build_contact(struct sip_pvt *p);
1466 static void build_rpid(struct sip_pvt *p);
1468 /*------Request handling functions */
1469 static int handle_request(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock);
1470 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);
1471 static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, int *nounlock);
1472 static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
1473 static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e);
1474 static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
1475 static int handle_request_message(struct sip_pvt *p, struct sip_request *req);
1476 static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
1477 static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
1478 static int handle_request_options(struct sip_pvt *p, struct sip_request *req);
1479 static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin);
1480 static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
1481 static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin);
1482 static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno);
1484 /*------Response handling functions */
1485 static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1486 static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1487 static int handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_request *req);
1488 static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno);
1489 static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno);
1491 /*----- RTP interface functions */
1492 static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active);
1493 static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
1494 static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
1495 static int sip_get_codec(struct ast_channel *chan);
1496 static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
1498 /*------ T38 Support --------- */
1499 static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */
1500 static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
1501 static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p);
1502 static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
1503 static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
1505 /*! \brief Definition of this channel for PBX channel registration */
1506 static const struct ast_channel_tech sip_tech = {
1508 .description = "Session Initiation Protocol (SIP)",
1509 .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
1510 .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
1511 .requester = sip_request_call,
1512 .devicestate = sip_devicestate,
1514 .hangup = sip_hangup,
1515 .answer = sip_answer,
1518 .write_video = sip_write,
1519 .indicate = sip_indicate,
1520 .transfer = sip_transfer,
1522 .send_digit_begin = sip_senddigit_begin,
1523 .send_digit_end = sip_senddigit_end,
1524 .bridge = ast_rtp_bridge,
1525 .early_bridge = ast_rtp_early_bridge,
1526 .send_text = sip_sendtext,
1529 /**--- some list management macros. **/
1531 #define UNLINK(element, head, prev) do { \
1533 (prev)->next = (element)->next; \
1535 (head) = (element)->next; \
1538 /*! \brief Interface structure with callbacks used to connect to RTP module */
1539 static struct ast_rtp_protocol sip_rtp = {
1541 get_rtp_info: sip_get_rtp_peer,
1542 get_vrtp_info: sip_get_vrtp_peer,
1543 set_rtp_peer: sip_set_rtp_peer,
1544 get_codec: sip_get_codec,
1547 /*! \brief Interface structure with callbacks used to connect to UDPTL module*/
1548 static struct ast_udptl_protocol sip_udptl = {
1550 get_udptl_info: sip_get_udptl_peer,
1551 set_udptl_peer: sip_set_udptl_peer,
1554 /*! \brief Convert transfer status to string */
1555 static char *referstatus2str(enum referstatus rstatus)
1557 int i = (sizeof(referstatusstrings) / sizeof(referstatusstrings[0]));
1560 for (x = 0; x < i; x++) {
1561 if (referstatusstrings[x].status == rstatus)
1562 return (char *) referstatusstrings[x].text;
1567 /*! \brief Initialize the initital request packet in the pvt structure.
1568 This packet is used for creating replies and future requests in
1570 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req)
1572 if (p->initreq.headers && option_debug) {
1573 ast_log(LOG_DEBUG, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
1575 /* Use this as the basis */
1576 copy_request(&p->initreq, req);
1577 parse_request(&p->initreq);
1578 if (ast_test_flag(req, SIP_PKT_DEBUG))
1579 ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
1583 /*! \brief returns true if 'name' (with optional trailing whitespace)
1584 * matches the sip method 'id'.
1585 * Strictly speaking, SIP methods are case SENSITIVE, but we do
1586 * a case-insensitive comparison to be more tolerant.
1587 * following Jon Postel's rule: Be gentle in what you accept, strict with what you send
1589 static int method_match(enum sipmethod id, const char *name)
1591 int len = strlen(sip_methods[id].text);
1592 int l_name = name ? strlen(name) : 0;
1593 /* true if the string is long enough, and ends with whitespace, and matches */
1594 return (l_name >= len && name[len] < 33 &&
1595 !strncasecmp(sip_methods[id].text, name, len));
1598 /*! \brief find_sip_method: Find SIP method from header */
1599 static int find_sip_method(const char *msg)
1603 if (ast_strlen_zero(msg))
1605 for (i = 1; i < (sizeof(sip_methods) / sizeof(sip_methods[0])) && !res; i++) {
1606 if (method_match(i, msg))
1607 res = sip_methods[i].id;
1612 /*! \brief Parse supported header in incoming packet */
1613 static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported)
1616 char *temp = ast_strdupa(supported);
1617 unsigned int profile = 0;
1620 if (ast_strlen_zero(supported) )
1623 if (option_debug > 2 && sipdebug)
1624 ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported);
1626 for (next = temp; next; next = sep) {
1628 if ( (sep = strchr(next, ',')) != NULL)
1630 next = ast_skip_blanks(next);
1631 if (option_debug > 2 && sipdebug)
1632 ast_log(LOG_DEBUG, "Found SIP option: -%s-\n", next);
1633 for (i=0; i < (sizeof(sip_options) / sizeof(sip_options[0])); i++) {
1634 if (!strcasecmp(next, sip_options[i].text)) {
1635 profile |= sip_options[i].id;
1637 if (option_debug > 2 && sipdebug)
1638 ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next);
1642 if (!found && option_debug > 2 && sipdebug) {
1643 if (!strncasecmp(next, "x-", 2))
1644 ast_log(LOG_DEBUG, "Found private SIP option, not supported: %s\n", next);
1646 ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next);
1651 pvt->sipoptions = profile;
1655 /*! \brief See if we pass debug IP filter */
1656 static inline int sip_debug_test_addr(const struct sockaddr_in *addr)
1660 if (debugaddr.sin_addr.s_addr) {
1661 if (((ntohs(debugaddr.sin_port) != 0)
1662 && (debugaddr.sin_port != addr->sin_port))
1663 || (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
1669 /*! \brief The real destination address for a write */
1670 static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p)
1672 return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? &p->recv : &p->sa;
1675 /*! \brief Display SIP nat mode */
1676 static const char *sip_nat_mode(const struct sip_pvt *p)
1678 return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? "NAT" : "no NAT";
1681 /*! \brief Test PVT for debugging output */
1682 static inline int sip_debug_test_pvt(struct sip_pvt *p)
1686 return sip_debug_test_addr(sip_real_dst(p));
1689 /*! \brief Transmit SIP message */
1690 static int __sip_xmit(struct sip_pvt *p, char *data, int len)
1693 const struct sockaddr_in *dst = sip_real_dst(p);
1694 res = sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
1697 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));
1702 /*! \brief Build a Via header for a request */
1703 static void build_via(struct sip_pvt *p)
1705 /* Work around buggy UNIDEN UIP200 firmware */
1706 const char *rport = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : "";
1708 /* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */
1709 ast_string_field_build(p, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s",
1710 ast_inet_ntoa(p->ourip), ourport, p->branch, rport);
1713 /*! \brief NAT fix - decide which IP address to use for ASterisk server?
1715 * Using the localaddr structure built up with localnet statements in sip.conf
1716 * apply it to their address to see if we need to substitute our
1717 * externip or can get away with our internal bindaddr
1719 static enum sip_result ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us)
1721 struct sockaddr_in theirs, ours;
1723 /* Get our local information */
1724 ast_ouraddrfor(them, us);
1725 theirs.sin_addr = *them;
1726 ours.sin_addr = *us;
1728 if (localaddr && externip.sin_addr.s_addr &&
1729 ast_apply_ha(localaddr, &theirs) &&
1730 !ast_apply_ha(localaddr, &ours)) {
1731 if (externexpire && time(NULL) >= externexpire) {
1732 struct ast_hostent ahp;
1735 externexpire = time(NULL) + externrefresh;
1736 if ((hp = ast_gethostbyname(externhost, &ahp))) {
1737 memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr));
1739 ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
1741 *us = externip.sin_addr;
1743 ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n",
1744 ast_inet_ntoa(*(struct in_addr *)&them->s_addr));
1746 } else if (bindaddr.sin_addr.s_addr)
1747 *us = bindaddr.sin_addr;
1751 /*! \brief Append to SIP dialog history
1752 \return Always returns 0 */
1753 #define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
1755 static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
1756 __attribute__ ((format (printf, 2, 3)));
1758 /*! \brief Append to SIP dialog history with arg list */
1759 static void append_history_va(struct sip_pvt *p, const char *fmt, va_list ap)
1761 char buf[80], *c = buf; /* max history length */
1762 struct sip_history *hist;
1765 vsnprintf(buf, sizeof(buf), fmt, ap);
1766 strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
1767 l = strlen(buf) + 1;
1768 if (!(hist = ast_calloc(1, sizeof(*hist) + l)))
1770 if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
1774 memcpy(hist->event, buf, l);
1775 AST_LIST_INSERT_TAIL(p->history, hist, list);
1778 /*! \brief Append to SIP dialog history with arg list */
1779 static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
1783 if (!recordhistory || !p)
1786 append_history_va(p, fmt, ap);
1792 /*! \brief Retransmit SIP message if no answer (Called from scheduler) */
1793 static int retrans_pkt(void *data)
1795 struct sip_pkt *pkt = data, *prev, *cur = NULL;
1796 int reschedule = DEFAULT_RETRANS;
1798 /* Lock channel PVT */
1799 ast_mutex_lock(&pkt->owner->lock);
1801 if (pkt->retrans < MAX_RETRANS) {
1803 if (!pkt->timer_t1) { /* Re-schedule using timer_a and timer_t1 */
1804 if (sipdebug && option_debug > 3)
1805 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);
1809 if (sipdebug && option_debug > 3)
1810 ast_log(LOG_DEBUG, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method);
1814 pkt->timer_a = 2 * pkt->timer_a;
1816 /* For non-invites, a maximum of 4 secs */
1817 siptimer_a = pkt->timer_t1 * pkt->timer_a; /* Double each time */
1818 if (pkt->method != SIP_INVITE && siptimer_a > 4000)
1821 /* Reschedule re-transmit */
1822 reschedule = siptimer_a;
1823 if (option_debug > 3)
1824 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);
1827 if (sip_debug_test_pvt(pkt->owner)) {
1828 const struct sockaddr_in *dst = sip_real_dst(pkt->owner);
1829 ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n",
1830 pkt->retrans, sip_nat_mode(pkt->owner),
1831 ast_inet_ntoa(dst->sin_addr),
1832 ntohs(dst->sin_port), pkt->data);
1835 append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data);
1836 __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
1837 ast_mutex_unlock(&pkt->owner->lock);
1840 /* Too many retries */
1841 if (pkt->owner && pkt->method != SIP_OPTIONS) {
1842 if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */
1843 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");
1845 if ((pkt->method == SIP_OPTIONS) && sipdebug)
1846 ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid);
1848 append_history(pkt->owner, "MaxRetries", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
1850 pkt->retransid = -1;
1852 if (ast_test_flag(pkt, FLAG_FATAL)) {
1853 while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
1854 ast_mutex_unlock(&pkt->owner->lock); /* SIP_PVT, not channel */
1856 ast_mutex_lock(&pkt->owner->lock);
1858 if (pkt->owner->owner) {
1859 ast_set_flag(&pkt->owner->flags[0], SIP_ALREADYGONE);
1860 ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
1861 ast_queue_hangup(pkt->owner->owner);
1862 ast_channel_unlock(pkt->owner->owner);
1864 /* If no channel owner, destroy now */
1865 ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY);
1868 /* In any case, go ahead and remove the packet */
1869 for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
1875 prev->next = cur->next;
1877 pkt->owner->packets = cur->next;
1878 ast_mutex_unlock(&pkt->owner->lock);
1882 ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
1884 ast_mutex_unlock(&pkt->owner->lock);
1888 /*! \brief Transmit packet with retransmits
1889 \return 0 on success, -1 on failure to allocate packet
1891 static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod)
1893 struct sip_pkt *pkt;
1894 int siptimer_a = DEFAULT_RETRANS;
1896 if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
1898 memcpy(pkt->data, data, len);
1899 pkt->method = sipmethod;
1900 pkt->packetlen = len;
1901 pkt->next = p->packets;
1905 pkt->data[len] = '\0';
1906 pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */
1908 ast_set_flag(pkt, FLAG_FATAL);
1910 siptimer_a = pkt->timer_t1 * 2;
1912 /* Schedule retransmission */
1913 pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1);
1914 if (option_debug > 3 && sipdebug)
1915 ast_log(LOG_DEBUG, "*** SIP TIMER: Initalizing retransmit timer on packet: Id #%d\n", pkt->retransid);
1916 pkt->next = p->packets;
1919 __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */
1920 if (sipmethod == SIP_INVITE) {
1921 /* Note this is a pending invite */
1922 p->pendinginvite = seqno;
1927 /*! \brief Kill a SIP dialog (called by scheduler) */
1928 static int __sip_autodestruct(void *data)
1930 struct sip_pvt *p = data;
1932 /* If this is a subscription, tell the phone that we got a timeout */
1933 if (p->subscribed) {
1934 p->subscribed = TIMEOUT;
1935 transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1); /* Send last notification */
1936 p->subscribed = NONE;
1937 append_history(p, "Subscribestatus", "timeout");
1938 if (option_debug > 2)
1939 ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP subsription %s\n", p->callid ? p->callid : "<unknown>");
1940 return 10000; /* Reschedule this destruction so that we know that it's gone */
1943 /* Reset schedule ID */
1947 ast_log(LOG_DEBUG, "Auto destroying SIP dialog '%s'\n", p->callid);
1948 append_history(p, "AutoDestroy", "%s", p->callid);
1950 ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
1951 ast_queue_hangup(p->owner);
1952 } else if (p->refer) {
1953 transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
1960 /*! \brief Schedule destruction of SIP dialog */
1961 static void sip_scheddestroy(struct sip_pvt *p, int ms)
1964 if (p->timer_t1 == 0)
1965 p->timer_t1 = 500; /* Set timer T1 if not set (RFC 3261) */
1966 ms = p->timer_t1 * 64;
1968 if (sip_debug_test_pvt(p))
1969 ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
1971 append_history(p, "SchedDestroy", "%d ms", ms);
1973 if (p->autokillid > -1)
1974 ast_sched_del(sched, p->autokillid);
1975 p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, p);
1978 /*! \brief Cancel destruction of SIP dialog */
1979 static void sip_cancel_destroy(struct sip_pvt *p)
1981 if (p->autokillid > -1) {
1982 ast_sched_del(sched, p->autokillid);
1983 append_history(p, "CancelDestroy", "");
1988 /*! \brief Acknowledges receipt of a packet and stops retransmission */
1989 static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod, int reset)
1991 struct sip_pkt *cur, *prev = NULL;
1993 /* Just in case... */
1997 msg = sip_methods[sipmethod].text;
1999 ast_mutex_lock(&p->lock);
2000 for (cur = p->packets; cur; prev = cur, cur = cur->next) {
2001 if ((cur->seqno == seqno) && ((ast_test_flag(cur, FLAG_RESPONSE)) == resp) &&
2002 ((ast_test_flag(cur, FLAG_RESPONSE)) ||
2003 (!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
2004 if (!resp && (seqno == p->pendinginvite)) {
2006 ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
2007 p->pendinginvite = 0;
2009 /* this is our baby */
2011 UNLINK(cur, p->packets, prev);
2012 if (cur->retransid > -1) {
2013 if (sipdebug && option_debug > 3)
2014 ast_log(LOG_DEBUG, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
2015 ast_sched_del(sched, cur->retransid);
2022 ast_mutex_unlock(&p->lock);
2024 ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
2027 /*! \brief Pretend to ack all packets
2028 * maybe the lock on p is not strictly necessary but there might be a race */
2029 static void __sip_pretend_ack(struct sip_pvt *p)
2031 struct sip_pkt *cur = NULL;
2033 while (p->packets) {
2035 if (cur == p->packets) {
2036 ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
2040 method = (cur->method) ? cur->method : find_sip_method(cur->data);
2041 __sip_ack(p, cur->seqno, ast_test_flag(cur, FLAG_RESPONSE), method, FALSE);
2045 /*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
2046 static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
2048 struct sip_pkt *cur;
2051 for (cur = p->packets; cur; cur = cur->next) {
2052 if (cur->seqno == seqno && ast_test_flag(cur, FLAG_RESPONSE) == resp &&
2053 (ast_test_flag(cur, FLAG_RESPONSE) || method_match(sipmethod, cur->data))) {
2054 /* this is our baby */
2055 if (cur->retransid > -1) {
2056 if (option_debug > 3 && sipdebug)
2057 ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
2058 ast_sched_del(sched, cur->retransid);
2060 cur->retransid = -1;
2066 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");
2071 /*! \brief Copy SIP request, parse it */
2072 static void parse_copy(struct sip_request *dst, const struct sip_request *src)
2074 memset(dst, 0, sizeof(*dst));
2075 memcpy(dst->data, src->data, sizeof(dst->data));
2076 dst->len = src->len;
2080 /*! \brief add a blank line if no body */
2081 static void add_blank(struct sip_request *req)
2084 /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
2085 snprintf(req->data + req->len, sizeof(req->data) - req->len, "\r\n");
2086 req->len += strlen(req->data + req->len);
2090 /*! \brief Transmit response on SIP request*/
2091 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
2096 if (sip_debug_test_pvt(p)) {
2097 const struct sockaddr_in *dst = sip_real_dst(p);
2099 ast_verbose("%sTransmitting (%s) to %s:%d:\n%s\n---\n",
2100 reliable ? "Reliably " : "", sip_nat_mode(p),
2101 ast_inet_ntoa(dst->sin_addr),
2102 ntohs(dst->sin_port), req->data);
2104 if (recordhistory) {
2105 struct sip_request tmp;
2106 parse_copy(&tmp, req);
2107 append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"),
2108 (tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? tmp.rlPart2 : sip_methods[tmp.method].text);
2111 __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
2112 __sip_xmit(p, req->data, req->len);
2118 /*! \brief Send SIP Request to the other part of the dialogue */
2119 static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
2124 if (sip_debug_test_pvt(p)) {
2125 if (ast_test_flag(&p->flags[0], SIP_NAT_ROUTE))
2126 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);
2128 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);
2130 if (recordhistory) {
2131 struct sip_request tmp;
2132 parse_copy(&tmp, req);
2133 append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
2136 __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1), req->method) :
2137 __sip_xmit(p, req->data, req->len);
2141 /*! \brief Locate closing quote in a string, skipping escaped quotes.
2142 * optionally with a limit on the search.
2143 * start must be past the first quote.
2145 static const char *find_closing_quote(const char *start, const char *lim)
2147 char last_char = '\0';
2149 for (s = start; *s && s != lim; last_char = *s++) {
2150 if (*s == '"' && last_char != '\\')
2156 /*! \brief Pick out text in brackets from character string
2157 \return pointer to terminated stripped string
2158 \param tmp input string that will be modified
2161 "foo" <bar> valid input, returns bar
2162 foo returns the whole string
2163 < "foo ... > returns the string between brackets
2164 < "foo... bogus (missing closing bracket), returns the whole string
2165 XXX maybe should still skip the opening bracket
2167 static char *get_in_brackets(char *tmp)
2169 const char *parse = tmp;
2170 char *first_bracket;
2173 * Skip any quoted text until we find the part in brackets.
2174 * On any error give up and return the full string.
2176 while ( (first_bracket = strchr(parse, '<')) ) {
2177 char *first_quote = strchr(parse, '"');
2179 if (!first_quote || first_quote > first_bracket)
2180 break; /* no need to look at quoted part */
2181 /* the bracket is within quotes, so ignore it */
2182 parse = find_closing_quote(first_quote + 1, NULL);
2183 if (!*parse) { /* not found, return full string ? */
2184 /* XXX or be robust and return in-bracket part ? */
2185 ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp);
2190 if (first_bracket) {
2191 char *second_bracket = strchr(first_bracket + 1, '>');
2192 if (second_bracket) {
2193 *second_bracket = '\0';
2194 tmp = first_bracket + 1;
2196 ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp);
2202 /*! \brief Send SIP MESSAGE text within a call
2203 Called from PBX core sendtext() application */
2204 static int sip_sendtext(struct ast_channel *ast, const char *text)
2206 struct sip_pvt *p = ast->tech_pvt;
2207 int debug = sip_debug_test_pvt(p);
2210 ast_verbose("Sending text %s on %s\n", text, ast->name);
2213 if (ast_strlen_zero(text))
2216 ast_verbose("Really sending text %s on %s\n", text, ast->name);
2217 transmit_message_with_text(p, text);
2221 /*! \brief Update peer object in realtime storage
2222 If the Asterisk system name is set in asterisk.conf, we will use
2223 that name and store that in the "regserver" field in the sippeers
2224 table to facilitate multi-server setups.
2226 static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey)
2229 char ipaddr[INET_ADDRSTRLEN];
2230 char regseconds[20];
2232 char *sysname = ast_config_AST_SYSTEM_NAME;
2233 char *syslabel = NULL;
2235 time_t nowtime = time(NULL) + expirey;
2236 const char *fc = fullcontact ? "fullcontact" : NULL;
2238 snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime); /* Expiration time */
2239 ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
2240 snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port));
2242 if (ast_strlen_zero(sysname)) /* No system name, disable this */
2244 else if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTSAVE_SYSNAME))
2245 syslabel = "regserver";
2248 ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr,
2249 "port", port, "regseconds", regseconds,
2250 "username", username, fc, fullcontact, syslabel, sysname, NULL); /* note fc and syslabel _can_ be NULL */
2252 ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr,
2253 "port", port, "regseconds", regseconds,
2254 "username", username, syslabel, sysname, NULL); /* note syslabel _can_ be NULL */
2257 /*! \brief Automatically add peer extension to dial plan */
2258 static void register_peer_exten(struct sip_peer *peer, int onoff)
2261 char *stringp, *ext, *context;
2263 /* XXX note that global_regcontext is both a global 'enable' flag and
2264 * the name of the global regexten context, if not specified
2267 if (ast_strlen_zero(global_regcontext))
2270 ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi));
2272 while ((ext = strsep(&stringp, "&"))) {
2273 if ((context = strchr(ext, '@'))) {
2274 *context++ = '\0'; /* split ext@context */
2275 if (!ast_context_find(context)) {
2276 ast_log(LOG_WARNING, "Context %s must exist in regcontext= in sip.conf!\n", context);
2280 context = global_regcontext;
2283 ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop",
2284 ast_strdup(peer->name), ast_free, "SIP");
2286 ast_context_remove_extension(context, ext, 1, NULL);
2290 /*! \brief Destroy peer object from memory */
2291 static void sip_destroy_peer(struct sip_peer *peer)
2293 if (option_debug > 2)
2294 ast_log(LOG_DEBUG, "Destroying SIP peer %s\n", peer->name);
2296 /* Delete it, it needs to disappear */
2298 sip_destroy(peer->call);
2300 if (peer->mwipvt) /* We have an active subscription, delete it */
2301 sip_destroy(peer->mwipvt);
2303 if (peer->chanvars) {
2304 ast_variables_destroy(peer->chanvars);
2305 peer->chanvars = NULL;
2307 if (peer->expire > -1)
2308 ast_sched_del(sched, peer->expire);
2309 if (peer->pokeexpire > -1)
2310 ast_sched_del(sched, peer->pokeexpire);
2311 register_peer_exten(peer, FALSE);
2312 ast_free_ha(peer->ha);
2313 if (ast_test_flag(&peer->flags[1], SIP_PAGE2_SELFDESTRUCT))
2315 else if (ast_test_flag(&peer->flags[0], SIP_REALTIME))
2319 clear_realm_authentication(peer->auth);
2322 ast_dnsmgr_release(peer->dnsmgr);
2326 /*! \brief Update peer data in database (if used) */
2327 static void update_peer(struct sip_peer *p, int expiry)
2329 int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
2330 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTUPDATE) &&
2331 (ast_test_flag(&p->flags[0], SIP_REALTIME) || rtcachefriends)) {
2332 realtime_update_peer(p->name, &p->addr, p->username, rtcachefriends ? p->fullcontact : NULL, expiry);
2337 /*! \brief realtime_peer: Get peer from realtime storage
2338 * Checks the "sippeers" realtime family from extconfig.conf
2339 * \todo Consider adding check of port address when matching here to follow the same
2340 * algorithm as for static peers. Will we break anything by adding that?
2342 static struct sip_peer *realtime_peer(const char *newpeername, struct sockaddr_in *sin)
2344 struct sip_peer *peer;
2345 struct ast_variable *var = NULL;
2346 struct ast_variable *tmp;
2347 char ipaddr[INET_ADDRSTRLEN];
2349 /* First check on peer name */
2351 var = ast_load_realtime("sippeers", "name", newpeername, NULL);
2352 else if (sin) { /* Then check on IP address for dynamic peers */
2353 ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
2354 var = ast_load_realtime("sippeers", "host", ipaddr, NULL); /* First check for fixed IP hosts */
2356 var = ast_load_realtime("sippeers", "ipaddr", ipaddr, NULL); /* Then check for registred hosts */
2362 for (tmp = var; tmp; tmp = tmp->next) {
2363 /* If this is type=user, then skip this object. */
2364 if (!strcasecmp(tmp->name, "type") &&
2365 !strcasecmp(tmp->value, "user")) {
2366 ast_variables_destroy(var);
2368 } else if (!newpeername && !strcasecmp(tmp->name, "name")) {
2369 newpeername = tmp->value;
2373 if (!newpeername) { /* Did not find peer in realtime */
2374 ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", ipaddr);
2375 ast_variables_destroy(var);
2379 /* Peer found in realtime, now build it in memory */
2380 peer = build_peer(newpeername, var, NULL, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
2382 ast_variables_destroy(var);
2386 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
2388 ast_copy_flags(&peer->flags[1],&global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
2389 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
2390 if (peer->expire > -1) {
2391 ast_sched_del(sched, peer->expire);
2393 peer->expire = ast_sched_add(sched, (global_rtautoclear) * 1000, expire_register, (void *)peer);
2395 ASTOBJ_CONTAINER_LINK(&peerl,peer);
2397 ast_set_flag(&peer->flags[0], SIP_REALTIME);
2399 ast_variables_destroy(var);
2404 /*! \brief Support routine for find_peer */
2405 static int sip_addrcmp(char *name, struct sockaddr_in *sin)
2407 /* We know name is the first field, so we can cast */
2408 struct sip_peer *p = (struct sip_peer *) name;
2409 return !(!inaddrcmp(&p->addr, sin) ||
2410 (ast_test_flag(&p->flags[0], SIP_INSECURE_PORT) &&
2411 (p->addr.sin_addr.s_addr == sin->sin_addr.s_addr)));
2414 /*! \brief Locate peer by name or ip address
2415 * This is used on incoming SIP message to find matching peer on ip
2416 or outgoing message to find matching peer on name */
2417 static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime)
2419 struct sip_peer *p = NULL;
2422 p = ASTOBJ_CONTAINER_FIND(&peerl, peer);
2424 p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp);
2427 p = realtime_peer(peer, sin);
2432 /*! \brief Remove user object from in-memory storage */
2433 static void sip_destroy_user(struct sip_user *user)
2435 if (option_debug > 2)
2436 ast_log(LOG_DEBUG, "Destroying user object from memory: %s\n", user->name);
2437 ast_free_ha(user->ha);
2438 if (user->chanvars) {
2439 ast_variables_destroy(user->chanvars);
2440 user->chanvars = NULL;
2442 if (ast_test_flag(&user->flags[0], SIP_REALTIME))
2449 /*! \brief Load user from realtime storage
2450 * Loads user from "sipusers" category in realtime (extconfig.conf)
2451 * Users are matched on From: user name (the domain in skipped) */
2452 static struct sip_user *realtime_user(const char *username)
2454 struct ast_variable *var;
2455 struct ast_variable *tmp;
2456 struct sip_user *user = NULL;
2458 var = ast_load_realtime("sipusers", "name", username, NULL);
2463 for (tmp = var; tmp; tmp = tmp->next) {
2464 if (!strcasecmp(tmp->name, "type") &&
2465 !strcasecmp(tmp->value, "peer")) {
2466 ast_variables_destroy(var);
2471 user = build_user(username, var, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
2473 if (!user) { /* No user found */
2474 ast_variables_destroy(var);
2478 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
2479 ast_set_flag(&user->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
2481 ASTOBJ_CONTAINER_LINK(&userl,user);
2483 /* Move counter from s to r... */
2486 ast_set_flag(&user->flags[0], SIP_REALTIME);
2488 ast_variables_destroy(var);
2492 /*! \brief Locate user by name
2493 * Locates user by name (From: sip uri user name part) first
2494 * from in-memory list (static configuration) then from
2495 * realtime storage (defined in extconfig.conf) */
2496 static struct sip_user *find_user(const char *name, int realtime)
2498 struct sip_user *u = ASTOBJ_CONTAINER_FIND(&userl, name);
2500 u = realtime_user(name);
2504 /*! \brief Set nat mode on the various data sockets */
2505 static void do_setnat(struct sip_pvt *p, int natflags)
2507 const char *mode = natflags ? "On" : "Off";
2511 ast_log(LOG_DEBUG, "Setting NAT on RTP to %s\n", mode);
2512 ast_rtp_setnat(p->rtp, natflags);
2516 ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", mode);
2517 ast_rtp_setnat(p->vrtp, natflags);
2521 ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", mode);
2522 ast_udptl_setnat(p->udptl, natflags);
2526 /*! \brief Create address structure from peer reference.
2527 * return -1 on error, 0 on success.
2529 static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
2531 if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) &&
2532 (!peer->maxms || ((peer->lastms >= 0) && (peer->lastms <= peer->maxms)))) {
2533 dialog->sa = (peer->addr.sin_addr.s_addr) ? peer->addr : peer->defaddr;
2534 dialog->recv = dialog->sa;
2538 ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
2539 ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
2540 dialog->capability = peer->capability;
2541 if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && dialog->vrtp) {
2542 ast_rtp_destroy(dialog->vrtp);
2543 dialog->vrtp = NULL;
2545 dialog->prefs = peer->prefs;
2546 if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
2547 dialog->t38.capability = global_t38_capability;
2548 if (dialog->udptl) {
2549 if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_FEC )
2550 dialog->t38.capability |= T38FAX_UDP_EC_FEC;
2551 else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY )
2552 dialog->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
2553 else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_NONE )
2554 dialog->t38.capability |= T38FAX_UDP_EC_NONE;
2555 dialog->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
2556 if (option_debug > 1)
2557 ast_log(LOG_DEBUG,"Our T38 capability (%d)\n", dialog->t38.capability);
2559 dialog->t38.jointcapability = dialog->t38.capability;
2560 } else if (dialog->udptl) {
2561 ast_udptl_destroy(dialog->udptl);
2562 dialog->udptl = NULL;
2564 do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE );
2567 ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
2568 ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
2571 ast_rtp_setdtmf(dialog->vrtp, 0);
2572 ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
2575 /* Set Frame packetization */
2577 ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
2578 dialog->autoframing = peer->autoframing;
2580 ast_string_field_set(dialog, peername, peer->username);
2581 ast_string_field_set(dialog, authname, peer->username);
2582 ast_string_field_set(dialog, username, peer->username);
2583 ast_string_field_set(dialog, peersecret, peer->secret);
2584 ast_string_field_set(dialog, peermd5secret, peer->md5secret);
2585 ast_string_field_set(dialog, tohost, peer->tohost);
2586 ast_string_field_set(dialog, fullcontact, peer->fullcontact);
2587 if (!dialog->initreq.headers && !ast_strlen_zero(peer->fromdomain)) {
2590 tmpcall = ast_strdupa(dialog->callid);
2591 c = strchr(tmpcall, '@');
2594 ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
2597 if (ast_strlen_zero(dialog->tohost))
2598 ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
2599 if (!ast_strlen_zero(peer->fromdomain))
2600 ast_string_field_set(dialog, fromdomain, peer->fromdomain);
2601 if (!ast_strlen_zero(peer->fromuser))
2602 ast_string_field_set(dialog, fromuser, peer->fromuser);
2603 dialog->maxtime = peer->maxms;
2604 dialog->callgroup = peer->callgroup;
2605 dialog->pickupgroup = peer->pickupgroup;
2606 dialog->allowtransfer = peer->allowtransfer;
2607 /* Set timer T1 to RTT for this peer (if known by qualify=) */
2608 /* Minimum is settable or default to 100 ms */
2609 if (peer->maxms && peer->lastms)
2610 dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
2611 if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
2612 (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
2613 dialog->noncodeccapability |= AST_RTP_DTMF;
2615 dialog->noncodeccapability &= ~AST_RTP_DTMF;
2616 ast_string_field_set(dialog, context, peer->context);
2617 dialog->rtptimeout = peer->rtptimeout;
2618 dialog->rtpholdtimeout = peer->rtpholdtimeout;
2619 dialog->rtpkeepalive = peer->rtpkeepalive;
2620 if (peer->call_limit)
2621 ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
2622 dialog->maxcallbitrate = peer->maxcallbitrate;
2627 /*! \brief create address structure from peer name
2628 * Or, if peer not found, find it in the global DNS
2629 * returns TRUE (-1) on failure, FALSE on success */
2630 static int create_addr(struct sip_pvt *dialog, const char *opeer)
2633 struct ast_hostent ahp;
2637 char host[MAXHOSTNAMELEN], *hostn;
2640 ast_copy_string(peer, opeer, sizeof(peer));
2641 port = strchr(peer, ':');
2644 dialog->sa.sin_family = AF_INET;
2645 dialog->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
2646 p = find_peer(peer, NULL, 1);
2649 int res = create_addr_from_peer(dialog, p);
2650 ASTOBJ_UNREF(p, sip_destroy_peer);
2654 portno = port ? atoi(port) : DEFAULT_SIP_PORT;
2656 char service[MAXHOSTNAMELEN];
2660 snprintf(service, sizeof(service), "_sip._udp.%s", peer);
2661 ret = ast_get_srv(NULL, host, sizeof(host), &tportno, service);
2667 hp = ast_gethostbyname(hostn, &ahp);
2669 ast_log(LOG_WARNING, "No such host: %s\n", peer);
2672 ast_string_field_set(dialog, tohost, peer);
2673 memcpy(&dialog->sa.sin_addr, hp->h_addr, sizeof(dialog->sa.sin_addr));
2674 dialog->sa.sin_port = htons(portno);
2675 dialog->recv = dialog->sa;
2679 /*! \brief Scheduled congestion on a call */
2680 static int auto_congest(void *nothing)
2682 struct sip_pvt *p = nothing;
2684 ast_mutex_lock(&p->lock);
2687 /* XXX fails on possible deadlock */
2688 if (!ast_channel_trylock(p->owner)) {
2689 ast_log(LOG_NOTICE, "Auto-congesting %s\n", p->owner->name);
2690 append_history(p, "Cong", "Auto-congesting (timer)");
2691 ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
2692 ast_channel_unlock(p->owner);
2695 ast_mutex_unlock(&p->lock);
2700 /*! \brief Initiate SIP call from PBX
2701 * used from the dial() application */
2702 static int sip_call(struct ast_channel *ast, char *dest, int timeout)
2706 struct varshead *headp;
2707 struct ast_var_t *current;
2708 const char *referer = NULL; /* SIP refererer */
2711 if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
2712 ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
2716 /* Check whether there is vxml_url, distinctive ring variables */
2717 headp=&ast->varshead;
2718 AST_LIST_TRAVERSE(headp,current,entries) {
2719 /* Check whether there is a VXML_URL variable */
2720 if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
2721 p->options->vxml_url = ast_var_value(current);
2722 } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
2723 p->options->uri_options = ast_var_value(current);
2724 } else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
2725 /* Check whether there is a variable with a name starting with SIPADDHEADER */
2726 p->options->addsipheaders = 1;
2727 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
2728 /* This is a transfered call */
2729 p->options->transfer = 1;
2730 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
2731 /* This is the referer */
2732 referer = ast_var_value(current);
2733 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
2734 /* We're replacing a call. */
2735 p->options->replaces = ast_var_value(current);
2736 } else if (!strcasecmp(ast_var_name(current), "T38CALL")) {
2737 p->t38.state = T38_LOCAL_DIRECT;
2739 ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
2745 ast_set_flag(&p->flags[0], SIP_OUTGOING);
2747 if (p->options->transfer) {
2751 if (sipdebug && option_debug > 2)
2752 ast_log(LOG_DEBUG, "Call for %s transfered by %s\n", p->username, referer);
2753 snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
2755 snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
2756 ast_string_field_set(p, cid_name, buf);
2759 ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username);
2761 res = update_call_counter(p, INC_CALL_RINGING);
2763 p->callingpres = ast->cid.cid_pres;
2764 p->jointcapability = p->capability;
2765 p->t38.jointcapability = p->t38.capability;
2767 ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
2768 transmit_invite(p, SIP_INVITE, 1, 2);
2770 /* Initialize auto-congest time */
2771 p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p);
2773 p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
2778 /*! \brief Destroy registry object
2779 Objects created with the register= statement in static configuration */
2780 static void sip_registry_destroy(struct sip_registry *reg)
2783 if (option_debug > 2)
2784 ast_log(LOG_DEBUG, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname);
2787 /* Clear registry before destroying to ensure
2788 we don't get reentered trying to grab the registry lock */
2789 reg->call->registry = NULL;
2790 if (option_debug > 2)
2791 ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
2792 sip_destroy(reg->call);
2794 if (reg->expire > -1)
2795 ast_sched_del(sched, reg->expire);
2796 if (reg->timeout > -1)
2797 ast_sched_del(sched, reg->timeout);
2798 ast_string_field_free_all(reg);
2804 /*! \brief Execute destruction of SIP dialog structure, release memory */
2805 static void __sip_destroy(struct sip_pvt *p, int lockowner)
2807 struct sip_pvt *cur, *prev = NULL;
2810 if (sip_debug_test_pvt(p) || option_debug > 2)
2811 ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
2813 /* Remove link from peer to subscription of MWI */
2814 if (p->relatedpeer && p->relatedpeer->mwipvt)
2815 p->relatedpeer->mwipvt = NULL;
2818 sip_dump_history(p);
2823 if (p->stateid > -1)
2824 ast_extension_state_del(p->stateid, NULL);
2826 ast_sched_del(sched, p->initid);
2827 if (p->autokillid > -1)
2828 ast_sched_del(sched, p->autokillid);
2831 ast_rtp_destroy(p->rtp);
2833 ast_rtp_destroy(p->vrtp);
2835 ast_udptl_destroy(p->udptl);
2839 free_old_route(p->route);
2843 if (p->registry->call == p)
2844 p->registry->call = NULL;
2845 ASTOBJ_UNREF(p->registry, sip_registry_destroy);
2848 /* Unlink us from the owner if we have one */
2851 ast_channel_lock(p->owner);
2853 ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
2854 p->owner->tech_pvt = NULL;
2856 ast_channel_unlock(p->owner);
2860 struct sip_history *hist;
2861 while( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) )
2867 for (prev = NULL, cur = iflist; cur; prev = cur, cur = cur->next) {
2869 UNLINK(cur, iflist, prev);
2874 ast_log(LOG_WARNING, "Trying to destroy \"%s\", not found in dialog list?!?! \n", p->callid);
2878 /* remove all current packets in this dialog */
2879 while((cp = p->packets)) {
2880 p->packets = p->packets->next;
2881 if (cp->retransid > -1)
2882 ast_sched_del(sched, cp->retransid);
2886 ast_variables_destroy(p->chanvars);
2889 ast_mutex_destroy(&p->lock);
2891 ast_string_field_free_all(p);
2896 /*! \brief update_call_counter: Handle call_limit for SIP users
2897 * Setting a call-limit will cause calls above the limit not to be accepted.
2899 * Remember that for a type=friend, there's one limit for the user and
2900 * another for the peer, not a combined call limit.
2901 * This will cause unexpected behaviour in subscriptions, since a "friend"
2902 * is *two* devices in Asterisk, not one.
2904 * Thought: For realtime, we should propably update storage with inuse counter...
2906 * \return 0 if call is ok (no call limit, below treshold)
2907 * -1 on rejection of call
2910 static int update_call_counter(struct sip_pvt *fup, int event)
2913 int *inuse, *call_limit, *inringing = NULL;
2914 int outgoing = ast_test_flag(&fup->flags[0], SIP_OUTGOING);
2915 struct sip_user *u = NULL;
2916 struct sip_peer *p = NULL;
2918 if (option_debug > 2)
2919 ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
2920 /* Test if we need to check call limits, in order to avoid
2921 realtime lookups if we do not need it */
2922 if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT))
2925 ast_copy_string(name, fup->username, sizeof(name));
2927 /* Check the list of users */
2928 if (!outgoing) /* Only check users for incoming calls */
2929 u = find_user(name, 1);
2933 call_limit = &u->call_limit;
2936 /* Try to find peer */
2938 p = find_peer(fup->peername, NULL, 1);
2941 call_limit = &p->call_limit;
2942 inringing = &p->inRinging;
2943 ast_copy_string(name, fup->peername, sizeof(name));
2945 if (option_debug > 1)
2946 ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name);
2951 /* incoming and outgoing affects the inUse counter */
2952 case DEC_CALL_LIMIT:
2954 if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT))
2960 if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
2964 ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", fup->peername);
2965 ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
2968 if (option_debug > 1 || sipdebug) {
2969 ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
2972 case INC_CALL_RINGING:
2973 case INC_CALL_LIMIT:
2974 if (*call_limit > 0 ) {
2975 if (*inuse >= *call_limit) {
2976 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);
2978 ASTOBJ_UNREF(u, sip_destroy_user);
2980 ASTOBJ_UNREF(p, sip_destroy_peer);
2984 if (inringing && (event == INC_CALL_RINGING)) {
2985 if (!ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
2987 ast_set_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
2992 ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
2993 if (option_debug > 1 || sipdebug) {
2994 ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
2997 case DEC_CALL_RINGING:
2999 if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
3003 ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", p->name);
3004 ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
3009 ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
3012 ast_device_state_changed("SIP/%s", p->name);
3014 ASTOBJ_UNREF(u, sip_destroy_user);
3016 ASTOBJ_UNREF(p, sip_destroy_peer);
3020 /*! \brief Destroy SIP call structure */
3021 static void sip_destroy(struct sip_pvt *p)
3023 ast_mutex_lock(&iflock);
3024 if (option_debug > 2)
3025 ast_log(LOG_DEBUG, "Destroying SIP dialog %s\n", p->callid);
3026 __sip_destroy(p, 1);
3027 ast_mutex_unlock(&iflock);
3030 /*! \brief Convert SIP hangup causes to Asterisk hangup causes */
3031 static int hangup_sip2cause(int cause)
3033 /* Possible values taken from causes.h */
3036 case 401: /* Unauthorized */