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"
447 /*! \brief Standard SIP port from RFC 3261. DO NOT CHANGE THIS */
448 #define STANDARD_SIP_PORT 5060
449 /* Note: in many SIP headers, absence of a port number implies port 5060,
450 * and this is why we cannot change the above constant.
451 * There is a limited number of places in asterisk where we could,
452 * in principle, use a different "default" port number, but
453 * we do not support this feature at the moment.
456 /* Default values, set and reset in reload_config before reading configuration */
457 /* These are default values in the source. There are other recommended values in the
458 sip.conf.sample for new installations. These may differ to keep backwards compatibility,
459 yet encouraging new behaviour on new installations
461 #define DEFAULT_CONTEXT "default"
462 #define DEFAULT_MOHINTERPRET "default"
463 #define DEFAULT_MOHSUGGEST ""
464 #define DEFAULT_VMEXTEN "asterisk"
465 #define DEFAULT_CALLERID "asterisk"
466 #define DEFAULT_NOTIFYMIME "application/simple-message-summary"
467 #define DEFAULT_MWITIME 10
468 #define DEFAULT_ALLOWGUEST TRUE
469 #define DEFAULT_SRVLOOKUP FALSE /*!< Recommended setting is ON */
470 #define DEFAULT_COMPACTHEADERS FALSE
471 #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. */
472 #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. */
473 #define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
474 #define DEFAULT_ALLOW_EXT_DOM TRUE
475 #define DEFAULT_REALM "asterisk"
476 #define DEFAULT_NOTIFYRINGING TRUE
477 #define DEFAULT_PEDANTIC FALSE
478 #define DEFAULT_AUTOCREATEPEER FALSE
479 #define DEFAULT_QUALIFY FALSE
480 #define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
481 #define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
482 #ifndef DEFAULT_USERAGENT
483 #define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
487 /* Default setttings are used as a channel setting and as a default when
488 configuring devices */
489 static char default_context[AST_MAX_CONTEXT];
490 static char default_subscribecontext[AST_MAX_CONTEXT];
491 static char default_language[MAX_LANGUAGE];
492 static char default_callerid[AST_MAX_EXTENSION];
493 static char default_fromdomain[AST_MAX_EXTENSION];
494 static char default_notifymime[AST_MAX_EXTENSION];
495 static int default_qualify; /*!< Default Qualify= setting */
496 static char default_vmexten[AST_MAX_EXTENSION];
497 static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
498 static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
499 * a bridged channel on hold */
500 static int default_maxcallbitrate; /*!< Maximum bitrate for call */
501 static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
503 /* Global settings only apply to the channel */
504 static int global_rtautoclear;
505 static int global_notifyringing; /*!< Send notifications on ringing */
506 static int global_alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
507 static int srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
508 static int pedanticsipchecking; /*!< Extra checking ? Default off */
509 static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
510 static int global_relaxdtmf; /*!< Relax DTMF */
511 static int global_rtptimeout; /*!< Time out call if no RTP */
512 static int global_rtpholdtimeout;
513 static int global_rtpkeepalive; /*!< Send RTP keepalives */
514 static int global_reg_timeout;
515 static int global_regattempts_max; /*!< Registration attempts before giving up */
516 static int global_allowguest; /*!< allow unauthenticated users/peers to connect? */
517 static int global_allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
518 the global setting is in globals_flags[1] */
519 static int global_mwitime; /*!< Time between MWI checks for peers */
520 static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
521 static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
522 static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
523 static int compactheaders; /*!< send compact sip headers */
524 static int recordhistory; /*!< Record SIP history. Off by default */
525 static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
526 static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
527 static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
528 static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
529 static int allow_external_domains; /*!< Accept calls to external SIP domains? */
530 static int global_callevents; /*!< Whether we send manager events or not */
531 static int global_t1min; /*!< T1 roundtrip time minimum */
532 static int global_autoframing; /*!< ?????????? */
533 static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */
535 /*! \brief Codecs that we support by default: */
536 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
537 static int noncodeccapability = AST_RTP_DTMF;
539 /* Object counters */
540 static int suserobjs = 0; /*!< Static users */
541 static int ruserobjs = 0; /*!< Realtime users */
542 static int speerobjs = 0; /*!< Statis peers */
543 static int rpeerobjs = 0; /*!< Realtime peers */
544 static int apeerobjs = 0; /*!< Autocreated peer objects */
545 static int regobjs = 0; /*!< Registry objects */
547 static struct ast_flags global_flags[2] = {{0}}; /*!< global SIP_ flags */
550 /*! \brief Protect the SIP dialog list (of sip_pvt's) */
551 AST_MUTEX_DEFINE_STATIC(iflock);
553 /*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
554 when it's doing something critical. */
555 AST_MUTEX_DEFINE_STATIC(netlock);
557 AST_MUTEX_DEFINE_STATIC(monlock);
559 AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
561 /*! \brief This is the thread for the monitor which checks for input on the channels
562 which are not currently in use. */
563 static pthread_t monitor_thread = AST_PTHREADT_NULL;
565 static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
566 static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
568 static struct sched_context *sched; /*!< The scheduling context */
569 static struct io_context *io; /*!< The IO context */
570 static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
572 #define DEC_CALL_LIMIT 0
573 #define INC_CALL_LIMIT 1
574 #define DEC_CALL_RINGING 2
575 #define INC_CALL_RINGING 3
577 /*! \brief sip_request: The data grabbed from the UDP socket */
579 char *rlPart1; /*!< SIP Method Name or "SIP/2.0" protocol version */
580 char *rlPart2; /*!< The Request URI or Response Status */
581 int len; /*!< Length */
582 int headers; /*!< # of SIP Headers */
583 int method; /*!< Method of this request */
584 int lines; /*!< Body Content */
585 unsigned int flags; /*!< SIP_PKT Flags for this packet */
586 char *header[SIP_MAX_HEADERS];
587 char *line[SIP_MAX_LINES];
588 char data[SIP_MAX_PACKET];
589 unsigned int sdp_start; /*!< the line number where the SDP begins */
590 unsigned int sdp_end; /*!< the line number where the SDP ends */
594 * A sip packet is stored into the data[] buffer, with the header followed
595 * by an empty line and the body of the message.
596 * On outgoing packets, data is accumulated in data[] with len reflecting
597 * the next available byte, headers and lines count the number of lines
598 * in both parts. There are no '\0' in data[0..len-1].
600 * On received packet, the input read from the socket is copied into data[],
601 * len is set and the string is NUL-terminated. Then a parser fills up
602 * the other fields -header[] and line[] to point to the lines of the
603 * message, rlPart1 and rlPart2 parse the first lnie as below:
605 * Requests have in the first line METHOD URI SIP/2.0
606 * rlPart1 = method; rlPart2 = uri;
607 * Responses have in the first line SIP/2.0 code description
608 * rlPart1 = SIP/2.0; rlPart2 = code + description;
612 /*! \brief structure used in transfers */
614 struct ast_channel *chan1; /*!< First channel involved */
615 struct ast_channel *chan2; /*!< Second channel involved */
616 struct sip_request req; /*!< Request that caused the transfer (REFER) */
617 int seqno; /*!< Sequence number */
622 /*! \brief Parameters to the transmit_invite function */
623 struct sip_invite_param {
624 int addsipheaders; /*!< Add extra SIP headers */
625 const char *uri_options; /*!< URI options to add to the URI */
626 const char *vxml_url; /*!< VXML url for Cisco phones */
627 char *auth; /*!< Authentication */
628 char *authheader; /*!< Auth header */
629 enum sip_auth_type auth_type; /*!< Authentication type */
630 const char *replaces; /*!< Replaces header for call transfers */
631 int transfer; /*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */
634 /*! \brief Structure to save routing information for a SIP session */
636 struct sip_route *next;
640 /*! \brief Modes for SIP domain handling in the PBX */
642 SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */
643 SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */
646 /*! \brief Domain data structure.
647 \note In the future, we will connect this to a configuration tree specific
651 char domain[MAXHOSTNAMELEN]; /*!< SIP domain we are responsible for */
652 char context[AST_MAX_EXTENSION]; /*!< Incoming context for this domain */
653 enum domain_mode mode; /*!< How did we find this domain? */
654 AST_LIST_ENTRY(domain) list; /*!< List mechanics */
657 static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */
660 /*! \brief sip_history: Structure for saving transactions within a SIP dialog */
662 AST_LIST_ENTRY(sip_history) list;
663 char event[0]; /* actually more, depending on needs */
666 AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
668 /*! \brief sip_auth: Creadentials for authentication to other SIP services */
670 char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */
671 char username[256]; /*!< Username */
672 char secret[256]; /*!< Secret */
673 char md5secret[256]; /*!< MD5Secret */
674 struct sip_auth *next; /*!< Next auth structure in list */
677 /*--- Various flags for the flags field in the pvt structure */
678 #define SIP_ALREADYGONE (1 << 0) /*!< Whether or not we've already been destroyed by our peer */
679 #define SIP_NEEDDESTROY (1 << 1) /*!< if we need to be destroyed by the monitor thread */
680 #define SIP_NOVIDEO (1 << 2) /*!< Didn't get video in invite, don't offer */
681 #define SIP_RINGING (1 << 3) /*!< Have sent 180 ringing */
682 #define SIP_PROGRESS_SENT (1 << 4) /*!< Have sent 183 message progress */
683 #define SIP_NEEDREINVITE (1 << 5) /*!< Do we need to send another reinvite? */
684 #define SIP_PENDINGBYE (1 << 6) /*!< Need to send bye after we ack? */
685 #define SIP_GOTREFER (1 << 7) /*!< Got a refer? */
686 #define SIP_PROMISCREDIR (1 << 8) /*!< Promiscuous redirection */
687 #define SIP_TRUSTRPID (1 << 9) /*!< Trust RPID headers? */
688 #define SIP_USEREQPHONE (1 << 10) /*!< Add user=phone to numeric URI. Default off */
689 #define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
690 #define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
691 #define SIP_OUTGOING (1 << 13) /*!< Is this an outgoing call? */
692 #define SIP_CAN_BYE (1 << 14) /*!< Can we send BYE on this dialog? */
693 #define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */
694 #define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
695 #define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */
696 #define SIP_DTMF_INBAND (1 << 16) /*!< DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
697 #define SIP_DTMF_INFO (2 << 16) /*!< DTMF Support: SIP Info messages - "info" */
698 #define SIP_DTMF_AUTO (3 << 16) /*!< DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
700 #define SIP_NAT (3 << 18) /*!< four settings, uses two bits */
701 #define SIP_NAT_NEVER (0 << 18) /*!< No nat support */
702 #define SIP_NAT_RFC3581 (1 << 18) /*!< NAT RFC3581 */
703 #define SIP_NAT_ROUTE (2 << 18) /*!< NAT Only ROUTE */
704 #define SIP_NAT_ALWAYS (3 << 18) /*!< NAT Both ROUTE and RFC3581 */
705 /* re-INVITE related settings */
706 #define SIP_REINVITE (7 << 20) /*!< three bits used */
707 #define SIP_CAN_REINVITE (1 << 20) /*!< allow peers to be reinvited to send media directly p2p */
708 #define SIP_CAN_REINVITE_NAT (2 << 20) /*!< allow media reinvite when new peer is behind NAT */
709 #define SIP_REINVITE_UPDATE (4 << 20) /*!< use UPDATE (RFC3311) when reinviting this peer */
710 /* "insecure" settings */
711 #define SIP_INSECURE_PORT (1 << 23) /*!< don't require matching port for incoming requests */
712 #define SIP_INSECURE_INVITE (1 << 24) /*!< don't require authentication for incoming INVITEs */
713 /* Sending PROGRESS in-band settings */
714 #define SIP_PROG_INBAND (3 << 25) /*!< three settings, uses two bits */
715 #define SIP_PROG_INBAND_NEVER (0 << 25)
716 #define SIP_PROG_INBAND_NO (1 << 25)
717 #define SIP_PROG_INBAND_YES (2 << 25)
718 #define SIP_FREE_BIT (1 << 27) /*!< Undefined bit - not in use */
719 #define SIP_CALL_LIMIT (1 << 28) /*!< Call limit enforced for this call */
720 #define SIP_SENDRPID (1 << 29) /*!< Remote Party-ID Support */
721 #define SIP_INC_COUNT (1 << 30) /*!< Did this connection increment the counter of in-use calls? */
722 #define SIP_G726_NONSTANDARD (1 << 31) /*!< Use non-standard packing for G726-32 data */
724 #define SIP_FLAGS_TO_COPY \
725 (SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
726 SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
727 SIP_USEREQPHONE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE)
729 /*--- a new page of flags (for flags[1] */
731 #define SIP_PAGE2_RTCACHEFRIENDS (1 << 0)
732 #define SIP_PAGE2_RTUPDATE (1 << 1)
733 #define SIP_PAGE2_RTAUTOCLEAR (1 << 2)
734 #define SIP_PAGE2_RT_FROMCONTACT (1 << 4)
735 #define SIP_PAGE2_RTSAVE_SYSNAME (1 << 5)
736 /* Space for addition of other realtime flags in the future */
737 #define SIP_PAGE2_IGNOREREGEXPIRE (1 << 10)
738 #define SIP_PAGE2_DEBUG (3 << 11)
739 #define SIP_PAGE2_DEBUG_CONFIG (1 << 11)
740 #define SIP_PAGE2_DEBUG_CONSOLE (1 << 12)
741 #define SIP_PAGE2_DYNAMIC (1 << 13) /*!< Dynamic Peers register with Asterisk */
742 #define SIP_PAGE2_SELFDESTRUCT (1 << 14) /*!< Automatic peers need to destruct themselves */
743 #define SIP_PAGE2_VIDEOSUPPORT (1 << 15)
744 #define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 16) /*!< Allow subscriptions from this peer? */
745 #define SIP_PAGE2_ALLOWOVERLAP (1 << 17) /*!< Allow overlap dialing ? */
746 #define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 18) /*!< Only issue MWI notification if subscribed to */
747 #define SIP_PAGE2_INC_RINGING (1 << 19) /*!< Did this connection increment the counter of in-use calls? */
748 #define SIP_PAGE2_T38SUPPORT (7 << 20) /*!< T38 Fax Passthrough Support */
749 #define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 20) /*!< 20: T38 Fax Passthrough Support */
750 #define SIP_PAGE2_T38SUPPORT_RTP (2 << 20) /*!< 21: T38 Fax Passthrough Support */
751 #define SIP_PAGE2_T38SUPPORT_TCP (4 << 20) /*!< 22: T38 Fax Passthrough Support */
752 #define SIP_PAGE2_CALL_ONHOLD (3 << 23) /*!< Call states */
753 #define SIP_PAGE2_CALL_ONHOLD_ONEDIR (1 << 23) /*!< 23: One directional hold */
754 #define SIP_PAGE2_CALL_ONHOLD_INACTIVE (2 << 24) /*!< 24: Inactive */
755 #define SIP_PAGE2_RFC2833_COMPENSATE (1 << 26)
757 #define SIP_PAGE2_FLAGS_TO_COPY \
758 (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE)
760 /* SIP packet flags */
761 #define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
762 #define SIP_PKT_WITH_TOTAG (1 << 1) /*!< This packet has a to-tag */
763 #define SIP_PKT_IGNORE (1 << 2) /*!< This is a re-transmit, ignore it */
764 #define SIP_PKT_IGNORE_RESP (1 << 3) /*!< Resp ignore - ??? */
765 #define SIP_PKT_IGNORE_REQ (1 << 4) /*!< Req ignore - ??? */
767 /* T.38 set of flags */
768 #define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0 (unset)*/
769 #define T38FAX_TRANSCODING_MMR (1 << 1) /*!< Default: 0 (unset)*/
770 #define T38FAX_TRANSCODING_JBIG (1 << 2) /*!< Default: 0 (unset)*/
771 /* Rate management */
772 #define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF (0 << 3)
773 #define T38FAX_RATE_MANAGEMENT_LOCAL_TCF (1 << 3) /*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */
774 /* UDP Error correction */
775 #define T38FAX_UDP_EC_NONE (0 << 4) /*!< two bits, if unset NO t38UDPEC field in T38 SDP*/
776 #define T38FAX_UDP_EC_FEC (1 << 4) /*!< Set for t38UDPFEC */
777 #define T38FAX_UDP_EC_REDUNDANCY (2 << 4) /*!< Set for t38UDPRedundancy */
778 /* T38 Spec version */
779 #define T38FAX_VERSION (3 << 6) /*!< two bits, 2 values so far, up to 4 values max */
780 #define T38FAX_VERSION_0 (0 << 6) /*!< Version 0 */
781 #define T38FAX_VERSION_1 (1 << 6) /*!< Version 1 */
782 /* Maximum Fax Rate */
783 #define T38FAX_RATE_2400 (1 << 8) /*!< 2400 bps t38FaxRate */
784 #define T38FAX_RATE_4800 (1 << 9) /*!< 4800 bps t38FaxRate */
785 #define T38FAX_RATE_7200 (1 << 10) /*!< 7200 bps t38FaxRate */
786 #define T38FAX_RATE_9600 (1 << 11) /*!< 9600 bps t38FaxRate */
787 #define T38FAX_RATE_12000 (1 << 12) /*!< 12000 bps t38FaxRate */
788 #define T38FAX_RATE_14400 (1 << 13) /*!< 14400 bps t38FaxRate */
790 /*!< This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
791 static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
793 #define sipdebug ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
794 #define sipdebug_config ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)
795 #define sipdebug_console ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)
797 /*! \brief T38 States for a call */
799 T38_DISABLED = 0, /*!< Not enabled */
800 T38_LOCAL_DIRECT, /*!< Offered from local */
801 T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */
802 T38_PEER_DIRECT, /*!< Offered from peer */
803 T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */
804 T38_ENABLED /*!< Negotiated (enabled) */
807 /*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
808 struct t38properties {
809 struct ast_flags t38support; /*!< Flag for udptl, rtp or tcp support for this session */
810 int capability; /*!< Our T38 capability */
811 int peercapability; /*!< Peers T38 capability */
812 int jointcapability; /*!< Supported T38 capability at both ends */
813 enum t38state state; /*!< T.38 state */
816 /*! \brief Parameters to know status of transfer */
818 REFER_IDLE, /*!< No REFER is in progress */
819 REFER_SENT, /*!< Sent REFER to transferee */
820 REFER_RECEIVED, /*!< Received REFER from transferer */
821 REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING */
822 REFER_ACCEPTED, /*!< Accepted by transferee */
823 REFER_RINGING, /*!< Target Ringing */
824 REFER_200OK, /*!< Answered by transfer target */
825 REFER_FAILED, /*!< REFER declined - go on */
826 REFER_NOAUTH /*!< We had no auth for REFER */
829 static const struct c_referstatusstring {
830 enum referstatus status;
832 } referstatusstrings[] = {
833 { REFER_IDLE, "<none>" },
834 { REFER_SENT, "Request sent" },
835 { REFER_RECEIVED, "Request received" },
836 { REFER_ACCEPTED, "Accepted" },
837 { REFER_RINGING, "Target ringing" },
838 { REFER_200OK, "Done" },
839 { REFER_FAILED, "Failed" },
840 { REFER_NOAUTH, "Failed - auth failure" }
843 /*! \brief Structure to handle SIP transfers. Dynamically allocated when needed */
844 /* OEJ: Should be moved to string fields */
846 char refer_to[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO extension */
847 char refer_to_domain[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO domain */
848 char refer_to_urioption[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO uri options */
849 char refer_to_context[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO context */
850 char referred_by[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
851 char referred_by_name[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
852 char refer_contact[AST_MAX_EXTENSION]; /*!< Place to store Contact info from a REFER extension */
853 char replaces_callid[BUFSIZ]; /*!< Replace info: callid */
854 char replaces_callid_totag[BUFSIZ/2]; /*!< Replace info: to-tag */
855 char replaces_callid_fromtag[BUFSIZ/2]; /*!< Replace info: from-tag */
856 struct sip_pvt *refer_call; /*!< Call we are referring */
857 int attendedtransfer; /*!< Attended or blind transfer? */
858 int localtransfer; /*!< Transfer to local domain? */
859 enum referstatus status; /*!< REFER status */
862 /*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe */
863 static struct sip_pvt {
864 ast_mutex_t lock; /*!< Dialog private lock */
865 int method; /*!< SIP method that opened this dialog */
866 AST_DECLARE_STRING_FIELDS(
867 AST_STRING_FIELD(callid); /*!< Global CallID */
868 AST_STRING_FIELD(randdata); /*!< Random data */
869 AST_STRING_FIELD(accountcode); /*!< Account code */
870 AST_STRING_FIELD(realm); /*!< Authorization realm */
871 AST_STRING_FIELD(nonce); /*!< Authorization nonce */
872 AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
873 AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
874 AST_STRING_FIELD(domain); /*!< Authorization domain */
875 AST_STRING_FIELD(from); /*!< The From: header */
876 AST_STRING_FIELD(useragent); /*!< User agent in SIP request */
877 AST_STRING_FIELD(exten); /*!< Extension where to start */
878 AST_STRING_FIELD(context); /*!< Context for this call */
879 AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */
880 AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */
881 AST_STRING_FIELD(fromdomain); /*!< Domain to show in the from field */
882 AST_STRING_FIELD(fromuser); /*!< User to show in the user field */
883 AST_STRING_FIELD(fromname); /*!< Name to show in the user field */
884 AST_STRING_FIELD(tohost); /*!< Host we should put in the "to" field */
885 AST_STRING_FIELD(language); /*!< Default language for this call */
886 AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */
887 AST_STRING_FIELD(mohsuggest); /*!< MOH class to suggest when putting a peer on hold */
888 AST_STRING_FIELD(rdnis); /*!< Referring DNIS */
889 AST_STRING_FIELD(theirtag); /*!< Their tag */
890 AST_STRING_FIELD(username); /*!< [user] name */
891 AST_STRING_FIELD(peername); /*!< [peer] name, not set if [user] */
892 AST_STRING_FIELD(authname); /*!< Who we use for authentication */
893 AST_STRING_FIELD(uri); /*!< Original requested URI */
894 AST_STRING_FIELD(okcontacturi); /*!< URI from the 200 OK on INVITE */
895 AST_STRING_FIELD(peersecret); /*!< Password */
896 AST_STRING_FIELD(peermd5secret);
897 AST_STRING_FIELD(cid_num); /*!< Caller*ID number */
898 AST_STRING_FIELD(cid_name); /*!< Caller*ID name */
899 AST_STRING_FIELD(via); /*!< Via: header */
900 AST_STRING_FIELD(fullcontact); /*!< The Contact: that the UA registers with us */
901 AST_STRING_FIELD(our_contact); /*!< Our contact header */
902 AST_STRING_FIELD(rpid); /*!< Our RPID header */
903 AST_STRING_FIELD(rpid_from); /*!< Our RPID From header */
905 unsigned int ocseq; /*!< Current outgoing seqno */
906 unsigned int icseq; /*!< Current incoming seqno */
907 ast_group_t callgroup; /*!< Call group */
908 ast_group_t pickupgroup; /*!< Pickup group */
909 int lastinvite; /*!< Last Cseq of invite */
910 struct ast_flags flags[2]; /*!< SIP_ flags */
911 int timer_t1; /*!< SIP timer T1, ms rtt */
912 unsigned int sipoptions; /*!< Supported SIP options on the other end */
913 struct ast_codec_pref prefs; /*!< codec prefs */
914 int capability; /*!< Special capability (codec) */
915 int jointcapability; /*!< Supported capability at both ends (codecs ) */
916 int peercapability; /*!< Supported peer capability */
917 int prefcodec; /*!< Preferred codec (outbound only) */
918 int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
919 int redircodecs; /*!< Redirect codecs */
920 int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
921 struct t38properties t38; /*!< T38 settings */
922 struct sockaddr_in udptlredirip; /*!< Where our T.38 UDPTL should be going if not to us */
923 struct ast_udptl *udptl; /*!< T.38 UDPTL session */
924 int callingpres; /*!< Calling presentation */
925 int authtries; /*!< Times we've tried to authenticate */
926 int expiry; /*!< How long we take to expire */
927 long branch; /*!< The branch identifier of this session */
928 char tag[11]; /*!< Our tag for this session */
929 int sessionid; /*!< SDP Session ID */
930 int sessionversion; /*!< SDP Session Version */
931 struct sockaddr_in sa; /*!< Our peer */
932 struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
933 struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
934 time_t lastrtprx; /*!< Last RTP received */
935 time_t lastrtptx; /*!< Last RTP sent */
936 int rtptimeout; /*!< RTP timeout time */
937 int rtpholdtimeout; /*!< RTP timeout when on hold */
938 int rtpkeepalive; /*!< Send RTP packets for keepalive */
939 struct sockaddr_in recv; /*!< Received as */
940 struct in_addr ourip; /*!< Our IP */
941 struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
942 struct sip_route *route; /*!< Head of linked list of routing steps (fm Record-Route) */
943 int route_persistant; /*!< Is this the "real" route? */
944 struct sip_auth *peerauth; /*!< Realm authentication */
945 int noncecount; /*!< Nonce-count */
946 char lastmsg[256]; /*!< Last Message sent/received */
947 int amaflags; /*!< AMA Flags */
948 int pendinginvite; /*!< Any pending invite ? (seqno of this) */
949 struct sip_request initreq; /*!< Initial request that opened the SIP dialog */
951 int maxtime; /*!< Max time for first response */
952 int initid; /*!< Auto-congest ID if appropriate (scheduler) */
953 int autokillid; /*!< Auto-kill ID (scheduler) */
954 enum transfermodes allowtransfer; /*!< REFER: restriction scheme */
955 struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
956 enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
957 int stateid; /*!< SUBSCRIBE: ID for devicestate subscriptions */
958 int laststate; /*!< SUBSCRIBE: Last known extension state */
959 int dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */
961 struct ast_dsp *vad; /*!< Voice Activation Detection dsp */
963 struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one
964 Used in peerpoke, mwi subscriptions */
965 struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */
966 struct ast_rtp *rtp; /*!< RTP Session */
967 struct ast_rtp *vrtp; /*!< Video RTP session */
968 struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
969 struct sip_history_head *history; /*!< History of this SIP dialog */
970 struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
971 struct sip_pvt *next; /*!< Next dialog in chain */
972 struct sip_invite_param *options; /*!< Options for INVITE */
976 #define FLAG_RESPONSE (1 << 0)
977 #define FLAG_FATAL (1 << 1)
979 /*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission */
981 struct sip_pkt *next; /*!< Next packet in linked list */
982 int retrans; /*!< Retransmission number */
983 int method; /*!< SIP method for this packet */
984 int seqno; /*!< Sequence number */
985 unsigned int flags; /*!< non-zero if this is a response packet (e.g. 200 OK) */
986 struct sip_pvt *owner; /*!< Owner AST call */
987 int retransid; /*!< Retransmission ID */
988 int timer_a; /*!< SIP timer A, retransmission timer */
989 int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */
990 int packetlen; /*!< Length of packet */
994 /*! \brief Structure for SIP user data. User's place calls to us */
996 /* Users who can access various contexts */
997 ASTOBJ_COMPONENTS(struct sip_user);
998 char secret[80]; /*!< Password */
999 char md5secret[80]; /*!< Password in md5 */
1000 char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
1001 char subscribecontext[AST_MAX_CONTEXT]; /* Default context for subscriptions */
1002 char cid_num[80]; /*!< Caller ID num */
1003 char cid_name[80]; /*!< Caller ID name */
1004 char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */
1005 char language[MAX_LANGUAGE]; /*!< Default language for this user */
1006 char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
1007 char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
1008 char useragent[256]; /*!< User agent in SIP request */
1009 struct ast_codec_pref prefs; /*!< codec prefs */
1010 ast_group_t callgroup; /*!< Call group */
1011 ast_group_t pickupgroup; /*!< Pickup Group */
1012 unsigned int sipoptions; /*!< Supported SIP options */
1013 struct ast_flags flags[2]; /*!< SIP_ flags */
1014 int amaflags; /*!< AMA flags for billing */
1015 int callingpres; /*!< Calling id presentation */
1016 int capability; /*!< Codec capability */
1017 int inUse; /*!< Number of calls in use */
1018 int call_limit; /*!< Limit of concurrent calls */
1019 enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
1020 struct ast_ha *ha; /*!< ACL setting */
1021 struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
1022 int maxcallbitrate; /*!< Maximum Bitrate for a video call */
1026 /*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host) */
1027 /* XXX field 'name' must be first otherwise sip_addrcmp() will fail */
1029 ASTOBJ_COMPONENTS(struct sip_peer); /*!< name, refcount, objflags, object pointers */
1030 /*!< peer->name is the unique name of this object */
1031 char secret[80]; /*!< Password */
1032 char md5secret[80]; /*!< Password in MD5 */
1033 struct sip_auth *auth; /*!< Realm authentication list */
1034 char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
1035 char subscribecontext[AST_MAX_CONTEXT]; /*!< Default context for subscriptions */
1036 char username[80]; /*!< Temporary username until registration */
1037 char accountcode[AST_MAX_ACCOUNT_CODE]; /*!< Account code */
1038 int amaflags; /*!< AMA Flags (for billing) */
1039 char tohost[MAXHOSTNAMELEN]; /*!< If not dynamic, IP address */
1040 char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */
1041 char fromuser[80]; /*!< From: user when calling this peer */
1042 char fromdomain[MAXHOSTNAMELEN]; /*!< From: domain when calling this peer */
1043 char fullcontact[256]; /*!< Contact registered with us (not in sip.conf) */
1044 char cid_num[80]; /*!< Caller ID num */
1045 char cid_name[80]; /*!< Caller ID name */
1046 int callingpres; /*!< Calling id presentation */
1047 int inUse; /*!< Number of calls in use */
1048 int inRinging; /*!< Number of calls ringing */
1049 int onHold; /*!< Peer has someone on hold */
1050 int call_limit; /*!< Limit of concurrent calls */
1051 enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
1052 char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
1053 char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
1054 char language[MAX_LANGUAGE]; /*!< Default language for prompts */
1055 char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
1056 char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
1057 char useragent[256]; /*!< User agent in SIP request (saved from registration) */
1058 struct ast_codec_pref prefs; /*!< codec prefs */
1060 time_t lastmsgcheck; /*!< Last time we checked for MWI */
1061 unsigned int sipoptions; /*!< Supported SIP options */
1062 struct ast_flags flags[2]; /*!< SIP_ flags */
1063 int expire; /*!< When to expire this peer registration */
1064 int capability; /*!< Codec capability */
1065 int rtptimeout; /*!< RTP timeout */
1066 int rtpholdtimeout; /*!< RTP Hold Timeout */
1067 int rtpkeepalive; /*!< Send RTP packets for keepalive */
1068 ast_group_t callgroup; /*!< Call group */
1069 ast_group_t pickupgroup; /*!< Pickup group */
1070 struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
1071 struct sockaddr_in addr; /*!< IP address of peer */
1072 int maxcallbitrate; /*!< Maximum Bitrate for a video call */
1075 struct sip_pvt *call; /*!< Call pointer */
1076 int pokeexpire; /*!< When to expire poke (qualify= checking) */
1077 int lastms; /*!< How long last response took (in ms), or -1 for no response */
1078 int maxms; /*!< Max ms we will accept for the host to be up, 0 to not monitor */
1079 struct timeval ps; /*!< Ping send time */
1081 struct sockaddr_in defaddr; /*!< Default IP address, used until registration */
1082 struct ast_ha *ha; /*!< Access control list */
1083 struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
1084 struct sip_pvt *mwipvt; /*!< Subscription for MWI */
1091 /*! \brief Registrations with other SIP proxies */
1092 struct sip_registry {
1093 ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1);
1094 AST_DECLARE_STRING_FIELDS(
1095 AST_STRING_FIELD(callid); /*!< Global Call-ID */
1096 AST_STRING_FIELD(realm); /*!< Authorization realm */
1097 AST_STRING_FIELD(nonce); /*!< Authorization nonce */
1098 AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
1099 AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
1100 AST_STRING_FIELD(domain); /*!< Authorization domain */
1101 AST_STRING_FIELD(username); /*!< Who we are registering as */
1102 AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */
1103 AST_STRING_FIELD(hostname); /*!< Domain or host we register to */
1104 AST_STRING_FIELD(secret); /*!< Password in clear text */
1105 AST_STRING_FIELD(md5secret); /*!< Password in md5 */
1106 AST_STRING_FIELD(contact); /*!< Contact extension */
1107 AST_STRING_FIELD(random);
1109 int portno; /*!< Optional port override */
1110 int expire; /*!< Sched ID of expiration */
1111 int regattempts; /*!< Number of attempts (since the last success) */
1112 int timeout; /*!< sched id of sip_reg_timeout */
1113 int refresh; /*!< How often to refresh */
1114 struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
1115 enum sipregistrystate regstate; /*!< Registration state (see above) */
1116 time_t regtime; /*!< Last succesful registration time */
1117 int callid_valid; /*!< 0 means we haven't chosen callid for this registry yet. */
1118 unsigned int ocseq; /*!< Sequence number we got to for REGISTERs for this registry */
1119 struct sockaddr_in us; /*!< Who the server thinks we are */
1120 int noncecount; /*!< Nonce-count */
1121 char lastmsg[256]; /*!< Last Message sent/received */
1124 /* --- Linked lists of various objects --------*/
1126 /*! \brief The user list: Users and friends */
1127 static struct ast_user_list {
1128 ASTOBJ_CONTAINER_COMPONENTS(struct sip_user);
1131 /*! \brief The peer list: Peers and Friends */
1132 static struct ast_peer_list {
1133 ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer);
1136 /*! \brief The register list: Other SIP proxys we register with and place calls to */
1137 static struct ast_register_list {
1138 ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
1142 /*! \todo Move the sip_auth list to AST_LIST */
1143 static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */
1146 /* --- Sockets and networking --------------*/
1147 static int sipsock = -1; /*!< Main socket for SIP network communication */
1148 static struct sockaddr_in bindaddr = { 0, }; /*!< The address we bind to */
1149 static struct sockaddr_in externip; /*!< External IP address if we are behind NAT */
1150 static char externhost[MAXHOSTNAMELEN]; /*!< External host name (possibly with dynamic DNS and DHCP */
1151 static time_t externexpire = 0; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
1152 static int externrefresh = 10;
1153 static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
1154 static struct in_addr __ourip;
1155 static struct sockaddr_in outboundproxyip;
1157 static struct sockaddr_in debugaddr;
1159 static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
1161 /*---------------------------- Forward declarations of functions in chan_sip.c */
1162 /*! \note This is added to help splitting up chan_sip.c into several files
1163 in coming releases */
1165 /*--- PBX interface functions */
1166 static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
1167 static int sip_devicestate(void *data);
1168 static int sip_sendtext(struct ast_channel *ast, const char *text);
1169 static int sip_call(struct ast_channel *ast, char *dest, int timeout);
1170 static int sip_hangup(struct ast_channel *ast);
1171 static int sip_answer(struct ast_channel *ast);
1172 static struct ast_frame *sip_read(struct ast_channel *ast);
1173 static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
1174 static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
1175 static int sip_transfer(struct ast_channel *ast, const char *dest);
1176 static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
1177 static int sip_senddigit_begin(struct ast_channel *ast, char digit);
1178 static int sip_senddigit_end(struct ast_channel *ast, char digit);
1180 /*--- Transmitting responses and requests */
1181 static int sipsock_read(int *id, int fd, short events, void *ignore);
1182 static int __sip_xmit(struct sip_pvt *p, char *data, int len);
1183 static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod);
1184 static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1185 static int retrans_pkt(void *data);
1186 static int transmit_sip_request(struct sip_pvt *p, struct sip_request *req);
1187 static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1188 static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1189 static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
1190 static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1191 static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
1192 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);
1193 static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
1194 static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, int reliable);
1195 static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
1196 static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
1197 static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init);
1198 static int transmit_reinvite_with_sdp(struct sip_pvt *p);
1199 static int transmit_info_with_digit(struct sip_pvt *p, const char digit);
1200 static int transmit_info_with_vidupdate(struct sip_pvt *p);
1201 static int transmit_message_with_text(struct sip_pvt *p, const char *text);
1202 static int transmit_refer(struct sip_pvt *p, const char *dest);
1203 static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten);
1204 static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
1205 static int transmit_state_notify(struct sip_pvt *p, int state, int full);
1206 static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
1207 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
1208 static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
1209 static void copy_request(struct sip_request *dst, const struct sip_request *src);
1210 static void receive_message(struct sip_pvt *p, struct sip_request *req);
1211 static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req);
1212 static int sip_send_mwi_to_peer(struct sip_peer *peer);
1213 static int does_peer_need_mwi(struct sip_peer *peer);
1215 /*--- Dialog management */
1216 static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
1217 int useglobal_nat, const int intended_method);
1218 static int __sip_autodestruct(void *data);
1219 static void sip_scheddestroy(struct sip_pvt *p, int ms);
1220 static void sip_cancel_destroy(struct sip_pvt *p);
1221 static void sip_destroy(struct sip_pvt *p);
1222 static void __sip_destroy(struct sip_pvt *p, int lockowner);
1223 static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod, int reset);
1224 static void __sip_pretend_ack(struct sip_pvt *p);
1225 static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
1226 static int auto_congest(void *nothing);
1227 static int update_call_counter(struct sip_pvt *fup, int event);
1228 static int hangup_sip2cause(int cause);
1229 static const char *hangup_cause2sip(int cause);
1230 static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method);
1231 static void free_old_route(struct sip_route *route);
1232 static void list_route(struct sip_route *route);
1233 static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards);
1234 static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin,
1235 struct sip_request *req, char *uri);
1236 static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
1237 static void check_pendings(struct sip_pvt *p);
1238 static void *sip_park_thread(void *stuff);
1239 static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno);
1240 static int sip_sipredirect(struct sip_pvt *p, const char *dest);
1242 /*--- Codec handling / SDP */
1243 static void try_suggested_sip_codec(struct sip_pvt *p);
1244 static const char* get_sdp_iterate(int* start, struct sip_request *req, const char *name);
1245 static const char *get_sdp(struct sip_request *req, const char *name);
1246 static int find_sdp(struct sip_request *req);
1247 static int process_sdp(struct sip_pvt *p, struct sip_request *req);
1248 static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
1249 char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
1250 int debug, int *min_packet_size);
1251 static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
1252 char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
1254 static int add_sdp(struct sip_request *resp, struct sip_pvt *p);
1255 static void do_setnat(struct sip_pvt *p, int natflags);
1257 /*--- Authentication stuff */
1258 static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
1259 static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
1260 static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
1261 static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); /* Add realm authentication in list */
1262 static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm); /* Find authentication for a specific realm */
1263 static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
1264 const char *secret, const char *md5secret, int sipmethod,
1265 char *uri, enum xmittype reliable, int ignore);
1266 static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
1267 int sipmethod, char *uri, enum xmittype reliable,
1268 struct sockaddr_in *sin, struct sip_peer **authpeer);
1269 static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);
1271 /*--- Domain handling */
1272 static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
1273 static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context);
1274 static void clear_sip_domains(void);
1276 /*--- SIP realm authentication */
1277 static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno);
1278 static int clear_realm_authentication(struct sip_auth *authlist);
1279 static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm);
1281 /*--- Misc functions */
1282 static int sip_do_reload(enum channelreloadreason reason);
1283 static int reload_config(enum channelreloadreason reason);
1284 static int expire_register(void *data);
1285 static int sip_sipredirect(struct sip_pvt *p, const char *dest);
1286 static void *do_monitor(void *data);
1287 static int restart_monitor(void);
1288 static int sip_send_mwi_to_peer(struct sip_peer *peer);
1289 static void sip_destroy(struct sip_pvt *p);
1290 static int sip_addrcmp(char *name, struct sockaddr_in *sin); /* Support for peer matching */
1291 static int sip_refer_allocate(struct sip_pvt *p);
1292 static void ast_quiet_chan(struct ast_channel *chan);
1293 static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
1295 /*--- Device monitoring and Device/extension state handling */
1296 static int cb_extensionstate(char *context, char* exten, int state, void *data);
1297 static int sip_devicestate(void *data);
1298 static int sip_poke_noanswer(void *data);
1299 static int sip_poke_peer(struct sip_peer *peer);
1300 static void sip_poke_all_peers(void);
1301 static void sip_peer_hold(struct sip_pvt *p, int hold);
1303 /*--- Applications, functions, CLI and manager command helpers */
1304 static const char *sip_nat_mode(const struct sip_pvt *p);
1305 static int sip_show_inuse(int fd, int argc, char *argv[]);
1306 static char *transfermode2str(enum transfermodes mode) attribute_const;
1307 static char *nat2str(int nat) attribute_const;
1308 static int peer_status(struct sip_peer *peer, char *status, int statuslen);
1309 static int sip_show_users(int fd, int argc, char *argv[]);
1310 static int _sip_show_peers(int fd, int *total, struct mansession *s, struct message *m, int argc, char *argv[]);
1311 static int manager_sip_show_peers( struct mansession *s, struct message *m );
1312 static int sip_show_peers(int fd, int argc, char *argv[]);
1313 static int sip_show_objects(int fd, int argc, char *argv[]);
1314 static void print_group(int fd, ast_group_t group, int crlf);
1315 static const char *dtmfmode2str(int mode) attribute_const;
1316 static const char *insecure2str(int port, int invite) attribute_const;
1317 static void cleanup_stale_contexts(char *new, char *old);
1318 static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
1319 static const char *domain_mode_to_text(const enum domain_mode mode);
1320 static int sip_show_domains(int fd, int argc, char *argv[]);
1321 static int _sip_show_peer(int type, int fd, struct mansession *s, struct message *m, int argc, char *argv[]);
1322 static int manager_sip_show_peer( struct mansession *s, struct message *m);
1323 static int sip_show_peer(int fd, int argc, char *argv[]);
1324 static int _sip_show_peer(int type, int fd, struct mansession *s, struct message *m, int argc, char *argv[]);
1325 static int sip_show_user(int fd, int argc, char *argv[]);
1326 static int sip_show_registry(int fd, int argc, char *argv[]);
1327 static int sip_show_settings(int fd, int argc, char *argv[]);
1328 static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
1329 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
1330 static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
1331 static int sip_show_channels(int fd, int argc, char *argv[]);
1332 static int sip_show_subscriptions(int fd, int argc, char *argv[]);
1333 static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
1334 static char *complete_sipch(const char *line, const char *word, int pos, int state);
1335 static char *complete_sip_peer(const char *word, int state, int flags2);
1336 static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
1337 static char *complete_sip_debug_peer(const char *line, const char *word, int pos, int state);
1338 static char *complete_sip_user(const char *word, int state, int flags2);
1339 static char *complete_sip_show_user(const char *line, const char *word, int pos, int state);
1340 static char *complete_sipnotify(const char *line, const char *word, int pos, int state);
1341 static char *complete_sip_prune_realtime_peer(const char *line, const char *word, int pos, int state);
1342 static char *complete_sip_prune_realtime_user(const char *line, const char *word, int pos, int state);
1343 static int sip_show_channel(int fd, int argc, char *argv[]);
1344 static int sip_show_history(int fd, int argc, char *argv[]);
1345 static int sip_do_debug_ip(int fd, int argc, char *argv[]);
1346 static int sip_do_debug_peer(int fd, int argc, char *argv[]);
1347 static int sip_do_debug(int fd, int argc, char *argv[]);
1348 static int sip_no_debug(int fd, int argc, char *argv[]);
1349 static int sip_notify(int fd, int argc, char *argv[]);
1350 static int sip_do_history(int fd, int argc, char *argv[]);
1351 static int sip_no_history(int fd, int argc, char *argv[]);
1352 static int func_header_read(struct ast_channel *chan, char *function, char *data, char *buf, size_t len);
1353 static int func_check_sipdomain(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len);
1354 static int function_sippeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len);
1355 static int function_sipchaninfo_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len);
1356 static int sip_dtmfmode(struct ast_channel *chan, void *data);
1357 static int sip_addheader(struct ast_channel *chan, void *data);
1358 static int sip_do_reload(enum channelreloadreason reason);
1359 static int sip_reload(int fd, int argc, char *argv[]);
1362 Functions for enabling debug per IP or fully, or enabling history logging for
1365 static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to LOG_DEBUG at end of dialog, before destroying data */
1366 static inline int sip_debug_test_addr(const struct sockaddr_in *addr);
1367 static inline int sip_debug_test_pvt(struct sip_pvt *p);
1368 static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
1369 static void sip_dump_history(struct sip_pvt *dialog);
1371 /*--- Device object handling */
1372 static struct sip_peer *temp_peer(const char *name);
1373 static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime);
1374 static struct sip_user *build_user(const char *name, struct ast_variable *v, int realtime);
1375 static int update_call_counter(struct sip_pvt *fup, int event);
1376 static void sip_destroy_peer(struct sip_peer *peer);
1377 static void sip_destroy_user(struct sip_user *user);
1378 static int sip_poke_peer(struct sip_peer *peer);
1379 static void set_peer_defaults(struct sip_peer *peer);
1380 static struct sip_peer *temp_peer(const char *name);
1381 static void register_peer_exten(struct sip_peer *peer, int onoff);
1382 static void sip_destroy_peer(struct sip_peer *peer);
1383 static void sip_destroy_user(struct sip_user *user);
1384 static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime);
1385 static struct sip_user *find_user(const char *name, int realtime);
1386 static int sip_poke_peer_s(void *data);
1387 static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
1388 static int expire_register(void *data);
1389 static void reg_source_db(struct sip_peer *peer);
1390 static void destroy_association(struct sip_peer *peer);
1391 static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);
1393 /* Realtime device support */
1394 static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey);
1395 static struct sip_user *realtime_user(const char *username);
1396 static void update_peer(struct sip_peer *p, int expiry);
1397 static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin);
1398 static int sip_prune_realtime(int fd, int argc, char *argv[]);
1400 /*--- Internal UA client handling (outbound registrations) */
1401 static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us);
1402 static void sip_registry_destroy(struct sip_registry *reg);
1403 static int sip_register(char *value, int lineno);
1404 static char *regstate2str(enum sipregistrystate regstate) attribute_const;
1405 static int sip_reregister(void *data);
1406 static int __sip_do_register(struct sip_registry *r);
1407 static int sip_reg_timeout(void *data);
1408 static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
1409 static void sip_send_all_registers(void);
1411 /*--- Parsing SIP requests and responses */
1412 static void append_date(struct sip_request *req); /* Append date to SIP packet */
1413 static int determine_firstline_parts(struct sip_request *req);
1414 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
1415 static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
1416 static int find_sip_method(const char *msg);
1417 static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported);
1418 static void parse_request(struct sip_request *req);
1419 static const char *get_header(const struct sip_request *req, const char *name);
1420 static char *referstatus2str(enum referstatus rstatus) attribute_pure;
1421 static int method_match(enum sipmethod id, const char *name);
1422 static void parse_copy(struct sip_request *dst, const struct sip_request *src);
1423 static char *get_in_brackets(char *tmp);
1424 static const char *find_alias(const char *name, const char *_default);
1425 static const char *__get_header(const struct sip_request *req, const char *name, int *start);
1426 static const char *get_header(const struct sip_request *req, const char *name);
1427 static int lws2sws(char *msgbuf, int len);
1428 static void extract_uri(struct sip_pvt *p, struct sip_request *req);
1429 static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req);
1430 static int get_also_info(struct sip_pvt *p, struct sip_request *oreq);
1431 static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
1432 static int set_address_from_contact(struct sip_pvt *pvt);
1433 static void check_via(struct sip_pvt *p, struct sip_request *req);
1434 static char *get_calleridname(const char *input, char *output, size_t outputsize);
1435 static int get_rpid_num(const char *input, char *output, int maxlen);
1436 static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq);
1437 static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
1438 static int get_msg_text(char *buf, int len, struct sip_request *req);
1439 static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
1440 static void free_old_route(struct sip_route *route);
1442 /*--- Constructing requests and responses */
1443 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
1444 static int init_req(struct sip_request *req, int sipmethod, const char *recip);
1445 static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch);
1446 static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod);
1447 static int init_resp(struct sip_request *resp, const char *msg);
1448 static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req);
1449 static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p);
1450 static void build_via(struct sip_pvt *p);
1451 static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
1452 static int create_addr(struct sip_pvt *dialog, const char *opeer);
1453 static char *generate_random_string(char *buf, size_t size);
1454 static void build_callid_pvt(struct sip_pvt *pvt);
1455 static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
1456 static void make_our_tag(char *tagbuf, size_t len);
1457 static int add_header(struct sip_request *req, const char *var, const char *value);
1458 static int add_header_contentLength(struct sip_request *req, int len);
1459 static int add_line(struct sip_request *req, const char *line);
1460 static int add_text(struct sip_request *req, const char *text);
1461 static int add_digit(struct sip_request *req, char digit);
1462 static int add_vidupdate(struct sip_request *req);
1463 static void add_route(struct sip_request *req, struct sip_route *route);
1464 static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field);
1465 static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field);
1466 static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field);
1467 static void set_destination(struct sip_pvt *p, char *uri);
1468 static void append_date(struct sip_request *req);
1469 static void build_contact(struct sip_pvt *p);
1470 static void build_rpid(struct sip_pvt *p);
1472 /*------Request handling functions */
1473 static int handle_request(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock);
1474 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);
1475 static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, int *nounlock);
1476 static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
1477 static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e);
1478 static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
1479 static int handle_request_message(struct sip_pvt *p, struct sip_request *req);
1480 static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
1481 static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
1482 static int handle_request_options(struct sip_pvt *p, struct sip_request *req);
1483 static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin);
1484 static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
1485 static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin);
1486 static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno);
1488 /*------Response handling functions */
1489 static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1490 static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
1491 static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno);
1492 static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno);
1494 /*----- RTP interface functions */
1495 static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active);
1496 static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
1497 static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
1498 static int sip_get_codec(struct ast_channel *chan);
1499 static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
1501 /*------ T38 Support --------- */
1502 static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */
1503 static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
1504 static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p);
1505 static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
1506 static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
1508 /*! \brief Definition of this channel for PBX channel registration */
1509 static const struct ast_channel_tech sip_tech = {
1511 .description = "Session Initiation Protocol (SIP)",
1512 .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
1513 .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
1514 .requester = sip_request_call,
1515 .devicestate = sip_devicestate,
1517 .hangup = sip_hangup,
1518 .answer = sip_answer,
1521 .write_video = sip_write,
1522 .indicate = sip_indicate,
1523 .transfer = sip_transfer,
1525 .send_digit_begin = sip_senddigit_begin,
1526 .send_digit_end = sip_senddigit_end,
1527 .bridge = ast_rtp_bridge,
1528 .early_bridge = ast_rtp_early_bridge,
1529 .send_text = sip_sendtext,
1532 /**--- some list management macros. **/
1534 #define UNLINK(element, head, prev) do { \
1536 (prev)->next = (element)->next; \
1538 (head) = (element)->next; \
1541 /*! \brief Interface structure with callbacks used to connect to RTP module */
1542 static struct ast_rtp_protocol sip_rtp = {
1544 get_rtp_info: sip_get_rtp_peer,
1545 get_vrtp_info: sip_get_vrtp_peer,
1546 set_rtp_peer: sip_set_rtp_peer,
1547 get_codec: sip_get_codec,
1550 /*! \brief Interface structure with callbacks used to connect to UDPTL module*/
1551 static struct ast_udptl_protocol sip_udptl = {
1553 get_udptl_info: sip_get_udptl_peer,
1554 set_udptl_peer: sip_set_udptl_peer,
1557 /*! \brief Convert transfer status to string */
1558 static char *referstatus2str(enum referstatus rstatus)
1560 int i = (sizeof(referstatusstrings) / sizeof(referstatusstrings[0]));
1563 for (x = 0; x < i; x++) {
1564 if (referstatusstrings[x].status == rstatus)
1565 return (char *) referstatusstrings[x].text;
1570 /*! \brief Initialize the initital request packet in the pvt structure.
1571 This packet is used for creating replies and future requests in
1573 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req)
1575 if (p->initreq.headers && option_debug) {
1576 ast_log(LOG_DEBUG, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
1578 /* Use this as the basis */
1579 copy_request(&p->initreq, req);
1580 parse_request(&p->initreq);
1581 if (ast_test_flag(req, SIP_PKT_DEBUG))
1582 ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
1586 /*! \brief returns true if 'name' (with optional trailing whitespace)
1587 * matches the sip method 'id'.
1588 * Strictly speaking, SIP methods are case SENSITIVE, but we do
1589 * a case-insensitive comparison to be more tolerant.
1590 * following Jon Postel's rule: Be gentle in what you accept, strict with what you send
1592 static int method_match(enum sipmethod id, const char *name)
1594 int len = strlen(sip_methods[id].text);
1595 int l_name = name ? strlen(name) : 0;
1596 /* true if the string is long enough, and ends with whitespace, and matches */
1597 return (l_name >= len && name[len] < 33 &&
1598 !strncasecmp(sip_methods[id].text, name, len));
1601 /*! \brief find_sip_method: Find SIP method from header */
1602 static int find_sip_method(const char *msg)
1606 if (ast_strlen_zero(msg))
1608 for (i = 1; i < (sizeof(sip_methods) / sizeof(sip_methods[0])) && !res; i++) {
1609 if (method_match(i, msg))
1610 res = sip_methods[i].id;
1615 /*! \brief Parse supported header in incoming packet */
1616 static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported)
1619 char *temp = ast_strdupa(supported);
1620 unsigned int profile = 0;
1623 if (ast_strlen_zero(supported) )
1626 if (option_debug > 2 && sipdebug)
1627 ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported);
1629 for (next = temp; next; next = sep) {
1631 if ( (sep = strchr(next, ',')) != NULL)
1633 next = ast_skip_blanks(next);
1634 if (option_debug > 2 && sipdebug)
1635 ast_log(LOG_DEBUG, "Found SIP option: -%s-\n", next);
1636 for (i=0; i < (sizeof(sip_options) / sizeof(sip_options[0])); i++) {
1637 if (!strcasecmp(next, sip_options[i].text)) {
1638 profile |= sip_options[i].id;
1640 if (option_debug > 2 && sipdebug)
1641 ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next);
1645 if (!found && option_debug > 2 && sipdebug) {
1646 if (!strncasecmp(next, "x-", 2))
1647 ast_log(LOG_DEBUG, "Found private SIP option, not supported: %s\n", next);
1649 ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next);
1654 pvt->sipoptions = profile;
1658 /*! \brief See if we pass debug IP filter */
1659 static inline int sip_debug_test_addr(const struct sockaddr_in *addr)
1663 if (debugaddr.sin_addr.s_addr) {
1664 if (((ntohs(debugaddr.sin_port) != 0)
1665 && (debugaddr.sin_port != addr->sin_port))
1666 || (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
1672 /*! \brief The real destination address for a write */
1673 static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p)
1675 return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? &p->recv : &p->sa;
1678 /*! \brief Display SIP nat mode */
1679 static const char *sip_nat_mode(const struct sip_pvt *p)
1681 return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? "NAT" : "no NAT";
1684 /*! \brief Test PVT for debugging output */
1685 static inline int sip_debug_test_pvt(struct sip_pvt *p)
1689 return sip_debug_test_addr(sip_real_dst(p));
1692 /*! \brief Transmit SIP message */
1693 static int __sip_xmit(struct sip_pvt *p, char *data, int len)
1696 const struct sockaddr_in *dst = sip_real_dst(p);
1697 res = sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
1700 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));
1705 /*! \brief Build a Via header for a request */
1706 static void build_via(struct sip_pvt *p)
1708 /* Work around buggy UNIDEN UIP200 firmware */
1709 const char *rport = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : "";
1711 /* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */
1712 ast_string_field_build(p, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s",
1713 ast_inet_ntoa(p->ourip), ourport, p->branch, rport);
1716 /*! \brief NAT fix - decide which IP address to use for ASterisk server?
1718 * Using the localaddr structure built up with localnet statements in sip.conf
1719 * apply it to their address to see if we need to substitute our
1720 * externip or can get away with our internal bindaddr
1722 static enum sip_result ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us)
1724 struct sockaddr_in theirs, ours;
1726 /* Get our local information */
1727 ast_ouraddrfor(them, us);
1728 theirs.sin_addr = *them;
1729 ours.sin_addr = *us;
1731 if (localaddr && externip.sin_addr.s_addr &&
1732 ast_apply_ha(localaddr, &theirs) &&
1733 !ast_apply_ha(localaddr, &ours)) {
1734 if (externexpire && time(NULL) >= externexpire) {
1735 struct ast_hostent ahp;
1738 externexpire = time(NULL) + externrefresh;
1739 if ((hp = ast_gethostbyname(externhost, &ahp))) {
1740 memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr));
1742 ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
1744 *us = externip.sin_addr;
1746 ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n",
1747 ast_inet_ntoa(*(struct in_addr *)&them->s_addr));
1749 } else if (bindaddr.sin_addr.s_addr)
1750 *us = bindaddr.sin_addr;
1754 /*! \brief Append to SIP dialog history
1755 \return Always returns 0 */
1756 #define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
1758 static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
1759 __attribute__ ((format (printf, 2, 3)));
1761 /*! \brief Append to SIP dialog history with arg list */
1762 static void append_history_va(struct sip_pvt *p, const char *fmt, va_list ap)
1764 char buf[80], *c = buf; /* max history length */
1765 struct sip_history *hist;
1768 vsnprintf(buf, sizeof(buf), fmt, ap);
1769 strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
1770 l = strlen(buf) + 1;
1771 if (!(hist = ast_calloc(1, sizeof(*hist) + l)))
1773 if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
1777 memcpy(hist->event, buf, l);
1778 AST_LIST_INSERT_TAIL(p->history, hist, list);
1781 /*! \brief Append to SIP dialog history with arg list */
1782 static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
1786 if (!recordhistory || !p)
1789 append_history_va(p, fmt, ap);
1795 /*! \brief Retransmit SIP message if no answer (Called from scheduler) */
1796 static int retrans_pkt(void *data)
1798 struct sip_pkt *pkt = data, *prev, *cur = NULL;
1799 int reschedule = DEFAULT_RETRANS;
1801 /* Lock channel PVT */
1802 ast_mutex_lock(&pkt->owner->lock);
1804 if (pkt->retrans < MAX_RETRANS) {
1806 if (!pkt->timer_t1) { /* Re-schedule using timer_a and timer_t1 */
1807 if (sipdebug && option_debug > 3)
1808 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);
1812 if (sipdebug && option_debug > 3)
1813 ast_log(LOG_DEBUG, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method);
1817 pkt->timer_a = 2 * pkt->timer_a;
1819 /* For non-invites, a maximum of 4 secs */
1820 siptimer_a = pkt->timer_t1 * pkt->timer_a; /* Double each time */
1821 if (pkt->method != SIP_INVITE && siptimer_a > 4000)
1824 /* Reschedule re-transmit */
1825 reschedule = siptimer_a;
1826 if (option_debug > 3)
1827 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);
1830 if (sip_debug_test_pvt(pkt->owner)) {
1831 const struct sockaddr_in *dst = sip_real_dst(pkt->owner);
1832 ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n",
1833 pkt->retrans, sip_nat_mode(pkt->owner),
1834 ast_inet_ntoa(dst->sin_addr),
1835 ntohs(dst->sin_port), pkt->data);
1838 append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data);
1839 __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
1840 ast_mutex_unlock(&pkt->owner->lock);
1843 /* Too many retries */
1844 if (pkt->owner && pkt->method != SIP_OPTIONS) {
1845 if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */
1846 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");
1848 if ((pkt->method == SIP_OPTIONS) && sipdebug)
1849 ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid);
1851 append_history(pkt->owner, "MaxRetries", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
1853 pkt->retransid = -1;
1855 if (ast_test_flag(pkt, FLAG_FATAL)) {
1856 while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
1857 ast_mutex_unlock(&pkt->owner->lock); /* SIP_PVT, not channel */
1859 ast_mutex_lock(&pkt->owner->lock);
1861 if (pkt->owner->owner) {
1862 ast_set_flag(&pkt->owner->flags[0], SIP_ALREADYGONE);
1863 ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
1864 ast_queue_hangup(pkt->owner->owner);
1865 ast_channel_unlock(pkt->owner->owner);
1867 /* If no channel owner, destroy now */
1868 ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY);
1871 /* In any case, go ahead and remove the packet */
1872 for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
1878 prev->next = cur->next;
1880 pkt->owner->packets = cur->next;
1881 ast_mutex_unlock(&pkt->owner->lock);
1885 ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
1887 ast_mutex_unlock(&pkt->owner->lock);
1891 /*! \brief Transmit packet with retransmits
1892 \return 0 on success, -1 on failure to allocate packet
1894 static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod)
1896 struct sip_pkt *pkt;
1897 int siptimer_a = DEFAULT_RETRANS;
1899 if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
1901 memcpy(pkt->data, data, len);
1902 pkt->method = sipmethod;
1903 pkt->packetlen = len;
1904 pkt->next = p->packets;
1908 pkt->data[len] = '\0';
1909 pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */
1911 ast_set_flag(pkt, FLAG_FATAL);
1913 siptimer_a = pkt->timer_t1 * 2;
1915 /* Schedule retransmission */
1916 pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1);
1917 if (option_debug > 3 && sipdebug)
1918 ast_log(LOG_DEBUG, "*** SIP TIMER: Initalizing retransmit timer on packet: Id #%d\n", pkt->retransid);
1919 pkt->next = p->packets;
1922 __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */
1923 if (sipmethod == SIP_INVITE) {
1924 /* Note this is a pending invite */
1925 p->pendinginvite = seqno;
1930 /*! \brief Kill a SIP dialog (called by scheduler) */
1931 static int __sip_autodestruct(void *data)
1933 struct sip_pvt *p = data;
1935 /* If this is a subscription, tell the phone that we got a timeout */
1936 if (p->subscribed) {
1937 p->subscribed = TIMEOUT;
1938 transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1); /* Send last notification */
1939 p->subscribed = NONE;
1940 append_history(p, "Subscribestatus", "timeout");
1941 if (option_debug > 2)
1942 ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP subsription %s\n", p->callid ? p->callid : "<unknown>");
1943 return 10000; /* Reschedule this destruction so that we know that it's gone */
1946 /* Reset schedule ID */
1950 ast_log(LOG_DEBUG, "Auto destroying SIP dialog '%s'\n", p->callid);
1951 append_history(p, "AutoDestroy", "%s", p->callid);
1953 ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
1954 ast_queue_hangup(p->owner);
1955 } else if (p->refer) {
1956 transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
1963 /*! \brief Schedule destruction of SIP dialog */
1964 static void sip_scheddestroy(struct sip_pvt *p, int ms)
1967 if (p->timer_t1 == 0)
1968 p->timer_t1 = 500; /* Set timer T1 if not set (RFC 3261) */
1969 ms = p->timer_t1 * 64;
1971 if (sip_debug_test_pvt(p))
1972 ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
1974 append_history(p, "SchedDestroy", "%d ms", ms);
1976 if (p->autokillid > -1)
1977 ast_sched_del(sched, p->autokillid);
1978 p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, p);
1981 /*! \brief Cancel destruction of SIP dialog */
1982 static void sip_cancel_destroy(struct sip_pvt *p)
1984 if (p->autokillid > -1) {
1985 ast_sched_del(sched, p->autokillid);
1986 append_history(p, "CancelDestroy", "");
1991 /*! \brief Acknowledges receipt of a packet and stops retransmission */
1992 static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod, int reset)
1994 struct sip_pkt *cur, *prev = NULL;
1996 /* Just in case... */
2000 msg = sip_methods[sipmethod].text;
2002 ast_mutex_lock(&p->lock);
2003 for (cur = p->packets; cur; prev = cur, cur = cur->next) {
2004 if ((cur->seqno == seqno) && ((ast_test_flag(cur, FLAG_RESPONSE)) == resp) &&
2005 ((ast_test_flag(cur, FLAG_RESPONSE)) ||
2006 (!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
2007 if (!resp && (seqno == p->pendinginvite)) {
2009 ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
2010 p->pendinginvite = 0;
2012 /* this is our baby */
2014 UNLINK(cur, p->packets, prev);
2015 if (cur->retransid > -1) {
2016 if (sipdebug && option_debug > 3)
2017 ast_log(LOG_DEBUG, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
2018 ast_sched_del(sched, cur->retransid);
2025 ast_mutex_unlock(&p->lock);
2027 ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
2030 /*! \brief Pretend to ack all packets
2031 * maybe the lock on p is not strictly necessary but there might be a race */
2032 static void __sip_pretend_ack(struct sip_pvt *p)
2034 struct sip_pkt *cur = NULL;
2036 while (p->packets) {
2038 if (cur == p->packets) {
2039 ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
2043 method = (cur->method) ? cur->method : find_sip_method(cur->data);
2044 __sip_ack(p, cur->seqno, ast_test_flag(cur, FLAG_RESPONSE), method, FALSE);
2048 /*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
2049 static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
2051 struct sip_pkt *cur;
2054 for (cur = p->packets; cur; cur = cur->next) {
2055 if (cur->seqno == seqno && ast_test_flag(cur, FLAG_RESPONSE) == resp &&
2056 (ast_test_flag(cur, FLAG_RESPONSE) || method_match(sipmethod, cur->data))) {
2057 /* this is our baby */
2058 if (cur->retransid > -1) {
2059 if (option_debug > 3 && sipdebug)
2060 ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
2061 ast_sched_del(sched, cur->retransid);
2063 cur->retransid = -1;
2069 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");
2074 /*! \brief Copy SIP request, parse it */
2075 static void parse_copy(struct sip_request *dst, const struct sip_request *src)
2077 memset(dst, 0, sizeof(*dst));
2078 memcpy(dst->data, src->data, sizeof(dst->data));
2079 dst->len = src->len;
2083 /*! \brief add a blank line if no body */
2084 static void add_blank(struct sip_request *req)
2087 /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
2088 snprintf(req->data + req->len, sizeof(req->data) - req->len, "\r\n");
2089 req->len += strlen(req->data + req->len);
2093 /*! \brief Transmit response on SIP request*/
2094 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
2099 if (sip_debug_test_pvt(p)) {
2100 const struct sockaddr_in *dst = sip_real_dst(p);
2102 ast_verbose("\n<--- %sTransmitting (%s) to %s:%d --->\n%s\n<------------>\n",
2103 reliable ? "Reliably " : "", sip_nat_mode(p),
2104 ast_inet_ntoa(dst->sin_addr),
2105 ntohs(dst->sin_port), req->data);
2107 if (recordhistory) {
2108 struct sip_request tmp;
2109 parse_copy(&tmp, req);
2110 append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"),
2111 (tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? tmp.rlPart2 : sip_methods[tmp.method].text);
2114 __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
2115 __sip_xmit(p, req->data, req->len);
2121 /*! \brief Send SIP Request to the other part of the dialogue */
2122 static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
2127 if (sip_debug_test_pvt(p)) {
2128 if (ast_test_flag(&p->flags[0], SIP_NAT_ROUTE))
2129 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);
2131 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);
2133 if (recordhistory) {
2134 struct sip_request tmp;
2135 parse_copy(&tmp, req);
2136 append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
2139 __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1), req->method) :
2140 __sip_xmit(p, req->data, req->len);
2144 /*! \brief Locate closing quote in a string, skipping escaped quotes.
2145 * optionally with a limit on the search.
2146 * start must be past the first quote.
2148 static const char *find_closing_quote(const char *start, const char *lim)
2150 char last_char = '\0';
2152 for (s = start; *s && s != lim; last_char = *s++) {
2153 if (*s == '"' && last_char != '\\')
2159 /*! \brief Pick out text in brackets from character string
2160 \return pointer to terminated stripped string
2161 \param tmp input string that will be modified
2164 "foo" <bar> valid input, returns bar
2165 foo returns the whole string
2166 < "foo ... > returns the string between brackets
2167 < "foo... bogus (missing closing bracket), returns the whole string
2168 XXX maybe should still skip the opening bracket
2170 static char *get_in_brackets(char *tmp)
2172 const char *parse = tmp;
2173 char *first_bracket;
2176 * Skip any quoted text until we find the part in brackets.
2177 * On any error give up and return the full string.
2179 while ( (first_bracket = strchr(parse, '<')) ) {
2180 char *first_quote = strchr(parse, '"');
2182 if (!first_quote || first_quote > first_bracket)
2183 break; /* no need to look at quoted part */
2184 /* the bracket is within quotes, so ignore it */
2185 parse = find_closing_quote(first_quote + 1, NULL);
2186 if (!*parse) { /* not found, return full string ? */
2187 /* XXX or be robust and return in-bracket part ? */
2188 ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp);
2193 if (first_bracket) {
2194 char *second_bracket = strchr(first_bracket + 1, '>');
2195 if (second_bracket) {
2196 *second_bracket = '\0';
2197 tmp = first_bracket + 1;
2199 ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp);
2205 /*! \brief Send SIP MESSAGE text within a call
2206 Called from PBX core sendtext() application */
2207 static int sip_sendtext(struct ast_channel *ast, const char *text)
2209 struct sip_pvt *p = ast->tech_pvt;
2210 int debug = sip_debug_test_pvt(p);
2213 ast_verbose("Sending text %s on %s\n", text, ast->name);
2216 if (ast_strlen_zero(text))
2219 ast_verbose("Really sending text %s on %s\n", text, ast->name);
2220 transmit_message_with_text(p, text);
2224 /*! \brief Update peer object in realtime storage
2225 If the Asterisk system name is set in asterisk.conf, we will use
2226 that name and store that in the "regserver" field in the sippeers
2227 table to facilitate multi-server setups.
2229 static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey)
2232 char ipaddr[INET_ADDRSTRLEN];
2233 char regseconds[20];
2235 char *sysname = ast_config_AST_SYSTEM_NAME;
2236 char *syslabel = NULL;
2238 time_t nowtime = time(NULL) + expirey;
2239 const char *fc = fullcontact ? "fullcontact" : NULL;
2241 snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime); /* Expiration time */
2242 ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
2243 snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port));
2245 if (ast_strlen_zero(sysname)) /* No system name, disable this */
2247 else if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTSAVE_SYSNAME))
2248 syslabel = "regserver";
2251 ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr,
2252 "port", port, "regseconds", regseconds,
2253 "username", username, fc, fullcontact, syslabel, sysname, NULL); /* note fc and syslabel _can_ be NULL */
2255 ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr,
2256 "port", port, "regseconds", regseconds,
2257 "username", username, syslabel, sysname, NULL); /* note syslabel _can_ be NULL */
2260 /*! \brief Automatically add peer extension to dial plan */
2261 static void register_peer_exten(struct sip_peer *peer, int onoff)
2264 char *stringp, *ext, *context;
2266 /* XXX note that global_regcontext is both a global 'enable' flag and
2267 * the name of the global regexten context, if not specified
2270 if (ast_strlen_zero(global_regcontext))
2273 ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi));
2275 while ((ext = strsep(&stringp, "&"))) {
2276 if ((context = strchr(ext, '@'))) {
2277 *context++ = '\0'; /* split ext@context */
2278 if (!ast_context_find(context)) {
2279 ast_log(LOG_WARNING, "Context %s must exist in regcontext= in sip.conf!\n", context);
2283 context = global_regcontext;
2286 ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop",
2287 ast_strdup(peer->name), ast_free, "SIP");
2289 ast_context_remove_extension(context, ext, 1, NULL);
2293 /*! \brief Destroy peer object from memory */
2294 static void sip_destroy_peer(struct sip_peer *peer)
2296 if (option_debug > 2)
2297 ast_log(LOG_DEBUG, "Destroying SIP peer %s\n", peer->name);
2299 /* Delete it, it needs to disappear */
2301 sip_destroy(peer->call);
2303 if (peer->mwipvt) /* We have an active subscription, delete it */
2304 sip_destroy(peer->mwipvt);
2306 if (peer->chanvars) {
2307 ast_variables_destroy(peer->chanvars);
2308 peer->chanvars = NULL;
2310 if (peer->expire > -1)
2311 ast_sched_del(sched, peer->expire);
2312 if (peer->pokeexpire > -1)
2313 ast_sched_del(sched, peer->pokeexpire);
2314 register_peer_exten(peer, FALSE);
2315 ast_free_ha(peer->ha);
2316 if (ast_test_flag(&peer->flags[1], SIP_PAGE2_SELFDESTRUCT))
2318 else if (ast_test_flag(&peer->flags[0], SIP_REALTIME))
2322 clear_realm_authentication(peer->auth);
2325 ast_dnsmgr_release(peer->dnsmgr);
2329 /*! \brief Update peer data in database (if used) */
2330 static void update_peer(struct sip_peer *p, int expiry)
2332 int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
2333 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTUPDATE) &&
2334 (ast_test_flag(&p->flags[0], SIP_REALTIME) || rtcachefriends)) {
2335 realtime_update_peer(p->name, &p->addr, p->username, rtcachefriends ? p->fullcontact : NULL, expiry);
2340 /*! \brief realtime_peer: Get peer from realtime storage
2341 * Checks the "sippeers" realtime family from extconfig.conf
2342 * \todo Consider adding check of port address when matching here to follow the same
2343 * algorithm as for static peers. Will we break anything by adding that?
2345 static struct sip_peer *realtime_peer(const char *newpeername, struct sockaddr_in *sin)
2347 struct sip_peer *peer;
2348 struct ast_variable *var = NULL;
2349 struct ast_variable *tmp;
2350 char ipaddr[INET_ADDRSTRLEN];
2352 /* First check on peer name */
2354 var = ast_load_realtime("sippeers", "name", newpeername, NULL);
2355 else if (sin) { /* Then check on IP address for dynamic peers */
2356 ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
2357 var = ast_load_realtime("sippeers", "host", ipaddr, NULL); /* First check for fixed IP hosts */
2359 var = ast_load_realtime("sippeers", "ipaddr", ipaddr, NULL); /* Then check for registred hosts */
2365 for (tmp = var; tmp; tmp = tmp->next) {
2366 /* If this is type=user, then skip this object. */
2367 if (!strcasecmp(tmp->name, "type") &&
2368 !strcasecmp(tmp->value, "user")) {
2369 ast_variables_destroy(var);
2371 } else if (!newpeername && !strcasecmp(tmp->name, "name")) {
2372 newpeername = tmp->value;
2376 if (!newpeername) { /* Did not find peer in realtime */
2377 ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", ipaddr);
2378 ast_variables_destroy(var);
2382 /* Peer found in realtime, now build it in memory */
2383 peer = build_peer(newpeername, var, NULL, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
2385 ast_variables_destroy(var);
2389 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
2391 ast_copy_flags(&peer->flags[1],&global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
2392 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
2393 if (peer->expire > -1) {
2394 ast_sched_del(sched, peer->expire);
2396 peer->expire = ast_sched_add(sched, (global_rtautoclear) * 1000, expire_register, (void *)peer);
2398 ASTOBJ_CONTAINER_LINK(&peerl,peer);
2400 ast_set_flag(&peer->flags[0], SIP_REALTIME);
2402 ast_variables_destroy(var);
2407 /*! \brief Support routine for find_peer */
2408 static int sip_addrcmp(char *name, struct sockaddr_in *sin)
2410 /* We know name is the first field, so we can cast */
2411 struct sip_peer *p = (struct sip_peer *) name;
2412 return !(!inaddrcmp(&p->addr, sin) ||
2413 (ast_test_flag(&p->flags[0], SIP_INSECURE_PORT) &&
2414 (p->addr.sin_addr.s_addr == sin->sin_addr.s_addr)));
2417 /*! \brief Locate peer by name or ip address
2418 * This is used on incoming SIP message to find matching peer on ip
2419 or outgoing message to find matching peer on name */
2420 static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime)
2422 struct sip_peer *p = NULL;
2425 p = ASTOBJ_CONTAINER_FIND(&peerl, peer);
2427 p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp);
2430 p = realtime_peer(peer, sin);
2435 /*! \brief Remove user object from in-memory storage */
2436 static void sip_destroy_user(struct sip_user *user)
2438 if (option_debug > 2)
2439 ast_log(LOG_DEBUG, "Destroying user object from memory: %s\n", user->name);
2440 ast_free_ha(user->ha);
2441 if (user->chanvars) {
2442 ast_variables_destroy(user->chanvars);
2443 user->chanvars = NULL;
2445 if (ast_test_flag(&user->flags[0], SIP_REALTIME))
2452 /*! \brief Load user from realtime storage
2453 * Loads user from "sipusers" category in realtime (extconfig.conf)
2454 * Users are matched on From: user name (the domain in skipped) */
2455 static struct sip_user *realtime_user(const char *username)
2457 struct ast_variable *var;
2458 struct ast_variable *tmp;
2459 struct sip_user *user = NULL;
2461 var = ast_load_realtime("sipusers", "name", username, NULL);
2466 for (tmp = var; tmp; tmp = tmp->next) {
2467 if (!strcasecmp(tmp->name, "type") &&
2468 !strcasecmp(tmp->value, "peer")) {
2469 ast_variables_destroy(var);
2474 user = build_user(username, var, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
2476 if (!user) { /* No user found */
2477 ast_variables_destroy(var);
2481 if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
2482 ast_set_flag(&user->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
2484 ASTOBJ_CONTAINER_LINK(&userl,user);
2486 /* Move counter from s to r... */
2489 ast_set_flag(&user->flags[0], SIP_REALTIME);
2491 ast_variables_destroy(var);
2495 /*! \brief Locate user by name
2496 * Locates user by name (From: sip uri user name part) first
2497 * from in-memory list (static configuration) then from
2498 * realtime storage (defined in extconfig.conf) */
2499 static struct sip_user *find_user(const char *name, int realtime)
2501 struct sip_user *u = ASTOBJ_CONTAINER_FIND(&userl, name);
2503 u = realtime_user(name);
2507 /*! \brief Set nat mode on the various data sockets */
2508 static void do_setnat(struct sip_pvt *p, int natflags)
2510 const char *mode = natflags ? "On" : "Off";
2514 ast_log(LOG_DEBUG, "Setting NAT on RTP to %s\n", mode);
2515 ast_rtp_setnat(p->rtp, natflags);
2519 ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", mode);
2520 ast_rtp_setnat(p->vrtp, natflags);
2524 ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", mode);
2525 ast_udptl_setnat(p->udptl, natflags);
2529 /*! \brief Create address structure from peer reference.
2530 * return -1 on error, 0 on success.
2532 static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
2534 if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) &&
2535 (!peer->maxms || ((peer->lastms >= 0) && (peer->lastms <= peer->maxms)))) {
2536 dialog->sa = (peer->addr.sin_addr.s_addr) ? peer->addr : peer->defaddr;
2537 dialog->recv = dialog->sa;
2541 ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
2542 ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
2543 dialog->capability = peer->capability;
2544 if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && dialog->vrtp) {
2545 ast_rtp_destroy(dialog->vrtp);
2546 dialog->vrtp = NULL;
2548 dialog->prefs = peer->prefs;
2549 if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
2550 dialog->t38.capability = global_t38_capability;
2551 if (dialog->udptl) {
2552 if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_FEC )
2553 dialog->t38.capability |= T38FAX_UDP_EC_FEC;
2554 else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY )
2555 dialog->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
2556 else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_NONE )
2557 dialog->t38.capability |= T38FAX_UDP_EC_NONE;
2558 dialog->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
2559 if (option_debug > 1)
2560 ast_log(LOG_DEBUG,"Our T38 capability (%d)\n", dialog->t38.capability);
2562 dialog->t38.jointcapability = dialog->t38.capability;
2563 } else if (dialog->udptl) {
2564 ast_udptl_destroy(dialog->udptl);
2565 dialog->udptl = NULL;
2567 do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE );
2570 ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
2571 ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
2574 ast_rtp_setdtmf(dialog->vrtp, 0);
2575 ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
2578 /* Set Frame packetization */
2580 ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
2581 dialog->autoframing = peer->autoframing;
2583 ast_string_field_set(dialog, peername, peer->username);
2584 ast_string_field_set(dialog, authname, peer->username);
2585 ast_string_field_set(dialog, username, peer->username);
2586 ast_string_field_set(dialog, peersecret, peer->secret);
2587 ast_string_field_set(dialog, peermd5secret, peer->md5secret);
2588 ast_string_field_set(dialog, tohost, peer->tohost);
2589 ast_string_field_set(dialog, fullcontact, peer->fullcontact);
2590 if (!dialog->initreq.headers && !ast_strlen_zero(peer->fromdomain)) {
2593 tmpcall = ast_strdupa(dialog->callid);
2594 c = strchr(tmpcall, '@');
2597 ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
2600 if (ast_strlen_zero(dialog->tohost))
2601 ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
2602 if (!ast_strlen_zero(peer->fromdomain))
2603 ast_string_field_set(dialog, fromdomain, peer->fromdomain);
2604 if (!ast_strlen_zero(peer->fromuser))
2605 ast_string_field_set(dialog, fromuser, peer->fromuser);
2606 dialog->maxtime = peer->maxms;
2607 dialog->callgroup = peer->callgroup;
2608 dialog->pickupgroup = peer->pickupgroup;
2609 dialog->allowtransfer = peer->allowtransfer;
2610 /* Set timer T1 to RTT for this peer (if known by qualify=) */
2611 /* Minimum is settable or default to 100 ms */
2612 if (peer->maxms && peer->lastms)
2613 dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
2614 if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
2615 (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
2616 dialog->noncodeccapability |= AST_RTP_DTMF;
2618 dialog->noncodeccapability &= ~AST_RTP_DTMF;
2619 ast_string_field_set(dialog, context, peer->context);
2620 dialog->rtptimeout = peer->rtptimeout;
2621 dialog->rtpholdtimeout = peer->rtpholdtimeout;
2622 dialog->rtpkeepalive = peer->rtpkeepalive;
2623 if (peer->call_limit)
2624 ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
2625 dialog->maxcallbitrate = peer->maxcallbitrate;
2630 /*! \brief create address structure from peer name
2631 * Or, if peer not found, find it in the global DNS
2632 * returns TRUE (-1) on failure, FALSE on success */
2633 static int create_addr(struct sip_pvt *dialog, const char *opeer)
2636 struct ast_hostent ahp;
2640 char host[MAXHOSTNAMELEN], *hostn;
2643 ast_copy_string(peer, opeer, sizeof(peer));
2644 port = strchr(peer, ':');
2647 dialog->sa.sin_family = AF_INET;
2648 dialog->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
2649 p = find_peer(peer, NULL, 1);
2652 int res = create_addr_from_peer(dialog, p);
2653 ASTOBJ_UNREF(p, sip_destroy_peer);
2657 portno = port ? atoi(port) : STANDARD_SIP_PORT;
2659 char service[MAXHOSTNAMELEN];
2663 snprintf(service, sizeof(service), "_sip._udp.%s", peer);
2664 ret = ast_get_srv(NULL, host, sizeof(host), &tportno, service);
2670 hp = ast_gethostbyname(hostn, &ahp);
2672 ast_log(LOG_WARNING, "No such host: %s\n", peer);
2675 ast_string_field_set(dialog, tohost, peer);
2676 memcpy(&dialog->sa.sin_addr, hp->h_addr, sizeof(dialog->sa.sin_addr));
2677 dialog->sa.sin_port = htons(portno);
2678 dialog->recv = dialog->sa;
2682 /*! \brief Scheduled congestion on a call */
2683 static int auto_congest(void *nothing)
2685 struct sip_pvt *p = nothing;
2687 ast_mutex_lock(&p->lock);
2690 /* XXX fails on possible deadlock */
2691 if (!ast_channel_trylock(p->owner)) {
2692 ast_log(LOG_NOTICE, "Auto-congesting %s\n", p->owner->name);
2693 append_history(p, "Cong", "Auto-congesting (timer)");
2694 ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
2695 ast_channel_unlock(p->owner);
2698 ast_mutex_unlock(&p->lock);
2703 /*! \brief Initiate SIP call from PBX
2704 * used from the dial() application */
2705 static int sip_call(struct ast_channel *ast, char *dest, int timeout)
2709 struct varshead *headp;
2710 struct ast_var_t *current;
2711 const char *referer = NULL; /* SIP refererer */
2714 if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
2715 ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
2719 /* Check whether there is vxml_url, distinctive ring variables */
2720 headp=&ast->varshead;
2721 AST_LIST_TRAVERSE(headp,current,entries) {
2722 /* Check whether there is a VXML_URL variable */
2723 if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
2724 p->options->vxml_url = ast_var_value(current);
2725 } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
2726 p->options->uri_options = ast_var_value(current);
2727 } else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
2728 /* Check whether there is a variable with a name starting with SIPADDHEADER */
2729 p->options->addsipheaders = 1;
2730 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
2731 /* This is a transfered call */
2732 p->options->transfer = 1;
2733 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
2734 /* This is the referer */
2735 referer = ast_var_value(current);
2736 } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
2737 /* We're replacing a call. */
2738 p->options->replaces = ast_var_value(current);
2739 } else if (!strcasecmp(ast_var_name(current), "T38CALL")) {
2740 p->t38.state = T38_LOCAL_DIRECT;
2742 ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
2748 ast_set_flag(&p->flags[0], SIP_OUTGOING);
2750 if (p->options->transfer) {
2754 if (sipdebug && option_debug > 2)
2755 ast_log(LOG_DEBUG, "Call for %s transfered by %s\n", p->username, referer);
2756 snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
2758 snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
2759 ast_string_field_set(p, cid_name, buf);
2762 ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username);
2764 res = update_call_counter(p, INC_CALL_RINGING);
2766 p->callingpres = ast->cid.cid_pres;
2767 p->jointcapability = p->capability;
2768 p->t38.jointcapability = p->t38.capability;
2770 ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
2771 transmit_invite(p, SIP_INVITE, 1, 2);
2773 /* Initialize auto-congest time */
2774 p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p);
2776 p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
2781 /*! \brief Destroy registry object
2782 Objects created with the register= statement in static configuration */
2783 static void sip_registry_destroy(struct sip_registry *reg)
2786 if (option_debug > 2)
2787 ast_log(LOG_DEBUG, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname);
2790 /* Clear registry before destroying to ensure
2791 we don't get reentered trying to grab the registry lock */
2792 reg->call->registry = NULL;
2793 if (option_debug > 2)
2794 ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
2795 sip_destroy(reg->call);
2797 if (reg->expire > -1)
2798 ast_sched_del(sched, reg->expire);
2799 if (reg->timeout > -1)
2800 ast_sched_del(sched, reg->timeout);
2801 ast_string_field_free_all(reg);
2807 /*! \brief Execute destruction of SIP dialog structure, release memory */
2808 static void __sip_destroy(struct sip_pvt *p, int lockowner)
2810 struct sip_pvt *cur, *prev = NULL;
2813 if (sip_debug_test_pvt(p) || option_debug > 2)
2814 ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
2816 /* Remove link from peer to subscription of MWI */
2817 if (p->relatedpeer && p->relatedpeer->mwipvt)
2818 p->relatedpeer->mwipvt = NULL;
2821 sip_dump_history(p);
2826 if (p->stateid > -1)
2827 ast_extension_state_del(p->stateid, NULL);
2829 ast_sched_del(sched, p->initid);
2830 if (p->autokillid > -1)
2831 ast_sched_del(sched, p->autokillid);
2834 ast_rtp_destroy(p->rtp);
2836 ast_rtp_destroy(p->vrtp);
2838 ast_udptl_destroy(p->udptl);
2842 free_old_route(p->route);
2846 if (p->registry->call == p)
2847 p->registry->call = NULL;
2848 ASTOBJ_UNREF(p->registry, sip_registry_destroy);
2851 /* Unlink us from the owner if we have one */
2854 ast_channel_lock(p->owner);
2856 ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
2857 p->owner->tech_pvt = NULL;
2859 ast_channel_unlock(p->owner);
2863 struct sip_history *hist;
2864 while( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) )
2870 for (prev = NULL, cur = iflist; cur; prev = cur, cur = cur->next) {
2872 UNLINK(cur, iflist, prev);
2877 ast_log(LOG_WARNING, "Trying to destroy \"%s\", not found in dialog list?!?! \n", p->callid);
2881 /* remove all current packets in this dialog */
2882 while((cp = p->packets)) {
2883 p->packets = p->packets->next;
2884 if (cp->retransid > -1)
2885 ast_sched_del(sched, cp->retransid);
2889 ast_variables_destroy(p->chanvars);
2892 ast_mutex_destroy(&p->lock);
2894 ast_string_field_free_all(p);
2899 /*! \brief update_call_counter: Handle call_limit for SIP users
2900 * Setting a call-limit will cause calls above the limit not to be accepted.
2902 * Remember that for a type=friend, there's one limit for the user and
2903 * another for the peer, not a combined call limit.
2904 * This will cause unexpected behaviour in subscriptions, since a "friend"
2905 * is *two* devices in Asterisk, not one.
2907 * Thought: For realtime, we should propably update storage with inuse counter...
2909 * \return 0 if call is ok (no call limit, below treshold)
2910 * -1 on rejection of call
2913 static int update_call_counter(struct sip_pvt *fup, int event)
2916 int *inuse, *call_limit, *inringing;
2917 int outgoing = ast_test_flag(&fup->flags[0], SIP_OUTGOING);
2918 struct sip_user *u = NULL;
2919 struct sip_peer *p = NULL;
2921 if (option_debug > 2)
2922 ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
2923 /* Test if we need to check call limits, in order to avoid
2924 realtime lookups if we do not need it */
2925 if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT))
2928 ast_copy_string(name, fup->username, sizeof(name));
2930 /* Check the list of users only for incoming calls */
2931 if (!outgoing && (u = find_user(name, 1)) ) {
2933 call_limit = &u->call_limit;
2935 } else if ( (p = find_peer(fup->peername, NULL, 1) ) ) { /* Try to find peer */
2937 call_limit = &p->call_limit;
2938 inringing = &p->inRinging;
2939 ast_copy_string(name, fup->peername, sizeof(name));
2941 if (option_debug > 1)
2942 ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name);
2947 /* incoming and outgoing affects the inUse counter */
2948 case DEC_CALL_LIMIT:
2950 if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT))
2956 if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
2960 ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", fup->peername);
2961 ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
2964 if (option_debug > 1 || sipdebug) {
2965 ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
2969 case INC_CALL_RINGING:
2970 case INC_CALL_LIMIT:
2971 if (*call_limit > 0 ) {
2972 if (*inuse >= *call_limit) {
2973 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);
2975 ASTOBJ_UNREF(u, sip_destroy_user);
2977 ASTOBJ_UNREF(p, sip_destroy_peer);
2981 if (inringing && (event == INC_CALL_RINGING)) {
2982 if (!ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
2984 ast_set_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
2989 ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
2990 if (option_debug > 1 || sipdebug) {
2991 ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
2995 case DEC_CALL_RINGING:
2997 if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
3001 ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", p->name);
3002 ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
3008 ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
3011 ast_device_state_changed("SIP/%s", p->name);
3012 ASTOBJ_UNREF(p, sip_destroy_peer);
3013 } else /* u must be set */
3014 ASTOBJ_UNREF(u, sip_destroy_user);
3018 /*! \brief Destroy SIP call structure */
3019 static void sip_destroy(struct sip_pvt *p)
3021 ast_mutex_lock(&iflock);
3022 if (option_debug > 2)
3023 ast_log(LOG_DEBUG, "Destroying SIP dialog %s\n", p->callid);
3024 __sip_destroy(p, 1);
3025 ast_mutex_unlock(&iflock);
3028 /*! \brief Convert SIP hangup causes to Asterisk hangup causes */
3029 static int hangup_sip2cause(int cause)
3031 /* Possible values taken from causes.h */
3034 case 401: /* Unauthorized */
3035 return AST_CAUSE_CALL_REJECTED;
3036 case 403: /* Not found */
3037 return AST_CAUSE_CALL_REJECTED;
3038 case 404: /* Not found */
3039 return AST_CAUSE_UNALLOCATED;
3040 case 405: /* Method not allowed */
3041 return AST_CAUSE_INTERWORKING;
3042 case 407: /* Proxy authentication required */
3043 return AST_CAUSE_CALL_REJECTED;
3044 case 408: /* No reaction */
3045 return AST_CAUSE_NO_USER_RESPONSE;