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, and experimental TCP and TLS support
29 * Configuration file \link Config_sip sip.conf \endlink
31 * ********** IMPORTANT *
32 * \note TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration
33 * settings, dialplan commands and dialplans apps/functions
34 * See \ref sip_tcp_tls
37 * ******** General TODO:s
38 * \todo Better support of forking
39 * \todo VIA branch tag transaction checking
40 * \todo Transaction support
41 * \todo Asterisk should send a non-100 provisional response every minute to keep proxies
42 * from cancelling the transaction (RFC 3261 13.3.1.1). See bug #11157.
44 * ******** Wishlist: Improvements
45 * - Support of SIP domains for devices, so that we match on username@domain in the From: header
46 * - Connect registrations with a specific device on the incoming call. It's not done
47 * automatically in Asterisk
49 * \ingroup channel_drivers
51 * \par Overview of the handling of SIP sessions
52 * The SIP channel handles several types of SIP sessions, or dialogs,
53 * not all of them being "telephone calls".
54 * - Incoming calls that will be sent to the PBX core
55 * - Outgoing calls, generated by the PBX
56 * - SIP subscriptions and notifications of states and voicemail messages
57 * - SIP registrations, both inbound and outbound
58 * - SIP peer management (peerpoke, OPTIONS)
61 * In the SIP channel, there's a list of active SIP dialogs, which includes
62 * all of these when they are active. "sip show channels" in the CLI will
63 * show most of these, excluding subscriptions which are shown by
64 * "sip show subscriptions"
66 * \par incoming packets
67 * Incoming packets are received in the monitoring thread, then handled by
68 * sipsock_read() for udp only. In tcp, packets are read by the tcp_helper thread.
69 * sipsock_read() function parses the packet and matches an existing
70 * dialog or starts a new SIP dialog.
72 * sipsock_read sends the packet to handle_incoming(), that parses a bit more.
73 * If it is a response to an outbound request, the packet is sent to handle_response().
74 * If it is a request, handle_incoming() sends it to one of a list of functions
75 * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
76 * sipsock_read locks the ast_channel if it exists (an active call) and
77 * unlocks it after we have processed the SIP message.
79 * A new INVITE is sent to handle_request_invite(), that will end up
80 * starting a new channel in the PBX, the new channel after that executing
81 * in a separate channel thread. This is an incoming "call".
82 * When the call is answered, either by a bridged channel or the PBX itself
83 * the sip_answer() function is called.
85 * The actual media - Video or Audio - is mostly handled by the RTP subsystem
89 * Outbound calls are set up by the PBX through the sip_request_call()
90 * function. After that, they are activated by sip_call().
93 * The PBX issues a hangup on both incoming and outgoing calls through
94 * the sip_hangup() function
98 * \page sip_tcp_tls SIP TCP and TLS support
100 * \par tcpfixes TCP implementation changes needed
101 * \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
102 * \todo Save TCP/TLS sessions in registry
103 * If someone registers a SIPS uri, this forces us to set up a TLS connection back.
104 * \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
105 * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
106 * The tcpbindaddr config option should only be used to open ADDITIONAL ports
107 * So we should propably go back to
108 * bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
109 * if tlsenable=yes, open TLS port (provided we also have cert)
110 * tcpbindaddr = extra address for additional TCP connections
111 * tlsbindaddr = extra address for additional TCP/TLS connections
112 * udpbindaddr = extra address for additional UDP connections
113 * These three options should take multiple IP/port pairs
114 * Note: Since opening additional listen sockets is a *new* feature we do not have today
115 * the XXXbindaddr options needs to be disabled until we have support for it
117 * \todo re-evaluate the transport= setting in sip.conf. This is right now not well
118 * thought of. If a device in sip.conf contacts us via TCP, we should not switch transport,
119 * even if udp is the configured first transport.
121 * \todo Be prepared for one outbound and another incoming socket per pvt. This applies
122 * specially to communication with other peers (proxies).
123 * \todo We need to test TCP sessions with SIP proxies and in regards
124 * to the SIP outbound specs.
125 * \todo ;transport=tls was deprecated in RFC3261 and should not be used at all. See section 26.2.2.
127 * \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
128 * message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
129 * \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
130 * multiple domains in our TLS implementation, meaning one socket and one cert per domain
131 * \todo Selection of transport for a request needs to be done after we've parsed all route headers,
132 * also considering outbound proxy options.
133 * First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port: DNS naptr, srv, AAA)
134 * Intermediate requests: Outboundproxy(only when forced), routes, contact/uri
135 * DNS naptr support is crucial. A SIP uri might lead to a TLS connection.
136 * Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)
137 * \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote
138 * devices directly from the dialplan. UDP is only a fallback if no other method works,
139 * in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the
140 * MTU (like INIVTE with video, audio and RTT) TCP should be preferred.
142 * When dialling unconfigured peers (with no port number) or devices in external domains
143 * NAPTR records MUST be consulted to find configured transport. If they are not found,
144 * SRV records for both TCP and UDP should be checked. If there's a record for TCP, use that.
145 * If there's no record for TCP, then use UDP as a last resort. If there's no SRV records,
146 * \note this only applies if there's no outbound proxy configured for the session. If an outbound
147 * proxy is configured, these procedures might apply for locating the proxy and determining
148 * the transport to use for communication with the proxy.
149 * \par Other bugs to fix ----
150 * __set_address_from_contact(const char *fullcontact, struct sockaddr_in *sin, int tcp)
151 * - sets TLS port as default for all TCP connections, unless other port is given in contact.
152 * parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
153 * - assumes that the contact the UA registers is using the same transport as the REGISTER request, which is
155 * - Does not save any information about TCP/TLS connected devices, which is a severe BUG, as discussed on the mailing list.
156 * get_destination(struct sip_pvt *p, struct sip_request *oreq)
157 * - Doesn't store the information that we got an incoming SIPS request in the channel, so that
158 * we can require a secure signalling path OUT of Asterisk (on SIP or IAX2). Possibly, the call should
159 * fail on in-secure signalling paths if there's no override in our configuration. At least, provide a
160 * channel variable in the dialplan.
161 * get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
162 * - As above, if we have a SIPS: uri in the refer-to header
163 * - Does not check transport in refer_to uri.
167 <depend>chan_local</depend>
170 /*! \page sip_session_timers SIP Session Timers in Asterisk Chan_sip
172 The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
173 refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
174 request at a negotiated interval. If a session refresh fails then all the entities that support Session-
175 Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
176 the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
177 that do not support Session-Timers).
179 The Session-Timers can be configured on a system-wide, per-user, or per-peer basis. The peruser/
180 per-peer settings override the global settings. The following new parameters have been
181 added to the sip.conf file.
182 session-timers=["accept", "originate", "refuse"]
183 session-expires=[integer]
184 session-minse=[integer]
185 session-refresher=["uas", "uac"]
187 The session-timers parameter in sip.conf defines the mode of operation of SIP session-timers feature in
188 Asterisk. The Asterisk can be configured in one of the following three modes:
190 1. Accept :: In the "accept" mode, the Asterisk server honors session-timers requests
191 made by remote end-points. A remote end-point can request Asterisk to engage
192 session-timers by either sending it an INVITE request with a "Supported: timer"
193 header in it or by responding to Asterisk's INVITE with a 200 OK that contains
194 Session-Expires: header in it. In this mode, the Asterisk server does not
195 request session-timers from remote end-points. This is the default mode.
196 2. Originate :: In the "originate" mode, the Asterisk server requests the remote
197 end-points to activate session-timers in addition to honoring such requests
198 made by the remote end-pints. In order to get as much protection as possible
199 against hanging SIP channels due to network or end-point failures, Asterisk
200 resends periodic re-INVITEs even if a remote end-point does not support
201 the session-timers feature.
202 3. Refuse :: In the "refuse" mode, Asterisk acts as if it does not support session-
203 timers for inbound or outbound requests. If a remote end-point requests
204 session-timers in a dialog, then Asterisk ignores that request unless it's
205 noted as a requirement (Require: header), in which case the INVITE is
206 rejected with a 420 Bad Extension response.
210 #include "asterisk.h"
212 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
215 #include <sys/ioctl.h>
218 #include <sys/signal.h>
222 #include "asterisk/network.h"
223 #include "asterisk/paths.h" /* need ast_config_AST_SYSTEM_NAME */
225 #include "asterisk/lock.h"
226 #include "asterisk/channel.h"
227 #include "asterisk/config.h"
228 #include "asterisk/module.h"
229 #include "asterisk/pbx.h"
230 #include "asterisk/sched.h"
231 #include "asterisk/io.h"
232 #include "asterisk/rtp.h"
233 #include "asterisk/udptl.h"
234 #include "asterisk/acl.h"
235 #include "asterisk/manager.h"
236 #include "asterisk/callerid.h"
237 #include "asterisk/cli.h"
238 #include "asterisk/app.h"
239 #include "asterisk/musiconhold.h"
240 #include "asterisk/dsp.h"
241 #include "asterisk/features.h"
242 #include "asterisk/srv.h"
243 #include "asterisk/astdb.h"
244 #include "asterisk/causes.h"
245 #include "asterisk/utils.h"
246 #include "asterisk/file.h"
247 #include "asterisk/astobj.h"
249 Uncomment the define below, if you are having refcount related memory leaks.
250 With this uncommented, this module will generate a file, /tmp/refs, which contains
251 a history of the ao2_ref() calls. To be useful, all calls to ao2_* functions should
252 be modified to ao2_t_* calls, and include a tag describing what is happening with
253 enough detail, to make pairing up a reference count increment with its corresponding decrement.
254 The refcounter program in utils/ can be invaluable in highlighting objects that are not
255 balanced, along with the complete history for that object.
256 In normal operation, the macros defined will throw away the tags, so they do not
257 affect the speed of the program at all. They can be considered to be documentation.
259 /* #define REF_DEBUG 1 */
260 #include "asterisk/astobj2.h"
261 #include "asterisk/dnsmgr.h"
262 #include "asterisk/devicestate.h"
263 #include "asterisk/linkedlists.h"
264 #include "asterisk/stringfields.h"
265 #include "asterisk/monitor.h"
266 #include "asterisk/netsock.h"
267 #include "asterisk/localtime.h"
268 #include "asterisk/abstract_jb.h"
269 #include "asterisk/threadstorage.h"
270 #include "asterisk/translate.h"
271 #include "asterisk/ast_version.h"
272 #include "asterisk/event.h"
273 #include "asterisk/tcptls.h"
276 <application name="SIPDtmfMode" language="en_US">
278 Change the dtmfmode for a SIP call.
281 <parameter name="mode" required="true">
283 <enum name="inband" />
285 <enum name="rfc2833" />
290 <para>Changes the dtmfmode for a SIP call.</para>
293 <application name="SIPAddHeader" language="en_US">
295 Add a SIP header to the outbound call.
298 <parameter name="Header" required="true" />
299 <parameter name="Content" required="true" />
302 <para>Adds a header to a SIP call placed with DIAL.</para>
303 <para>Remember to use the X-header if you are adding non-standard SIP
304 headers, like <literal>X-Asterisk-Accountcode:</literal>. Use this with care.
305 Adding the wrong headers may jeopardize the SIP dialog.</para>
306 <para>Always returns <literal>0</literal>.</para>
309 <application name="SIPRemoveHeader" language="en_US">
311 Remove SIP headers previously added with SIPAddHeader
314 <parameter name="Header" required="false" />
317 <para>SIPRemoveHeader() allows you to remove headers which were previously
318 added with SIPAddHeader(). If no parameter is supplied, all previously added
319 headers will be removed. If a parameter is supplied, only the matching headers
320 will be removed.</para>
321 <para>For example you have added these 2 headers:</para>
322 <para>SIPAddHeader(P-Asserted-Identity: sip:foo@bar);</para>
323 <para>SIPAddHeader(P-Preferred-Identity: sip:bar@foo);</para>
325 <para>// remove all headers</para>
326 <para>SIPRemoveHeader();</para>
327 <para>// remove all P- headers</para>
328 <para>SIPRemoveHeader(P-);</para>
329 <para>// remove only the PAI header (note the : at the end)</para>
330 <para>SIPRemoveHeader(P-Asserted-Identity:);</para>
332 <para>Always returns <literal>0</literal>.</para>
335 <function name="SIP_HEADER" language="en_US">
337 Gets the specified SIP header.
340 <parameter name="name" required="true" />
341 <parameter name="number">
342 <para>If not specified, defaults to <literal>1</literal>.</para>
346 <para>Since there are several headers (such as Via) which can occur multiple
347 times, SIP_HEADER takes an optional second argument to specify which header with
348 that name to retrieve. Headers start at offset <literal>1</literal>.</para>
351 <function name="SIPPEER" language="en_US">
353 Gets SIP peer information.
356 <parameter name="peername" required="true" />
357 <parameter name="item">
360 <para>(default) The ip address.</para>
363 <para>The port number.</para>
365 <enum name="mailbox">
366 <para>The configured mailbox.</para>
368 <enum name="context">
369 <para>The configured context.</para>
372 <para>The epoch time of the next expire.</para>
374 <enum name="dynamic">
375 <para>Is it dynamic? (yes/no).</para>
377 <enum name="callerid_name">
378 <para>The configured Caller ID name.</para>
380 <enum name="callerid_num">
381 <para>The configured Caller ID number.</para>
383 <enum name="callgroup">
384 <para>The configured Callgroup.</para>
386 <enum name="pickupgroup">
387 <para>The configured Pickupgroup.</para>
390 <para>The configured codecs.</para>
393 <para>Status (if qualify=yes).</para>
395 <enum name="regexten">
396 <para>Registration extension.</para>
399 <para>Call limit (call-limit).</para>
401 <enum name="busylevel">
402 <para>Configured call level for signalling busy.</para>
404 <enum name="curcalls">
405 <para>Current amount of calls. Only available if call-limit is set.</para>
407 <enum name="language">
408 <para>Default language for peer.</para>
410 <enum name="accountcode">
411 <para>Account code for this peer.</para>
413 <enum name="useragent">
414 <para>Current user agent id for peer.</para>
416 <enum name="chanvar[name]">
417 <para>A channel variable configured with setvar for this peer.</para>
419 <enum name="codec[x]">
420 <para>Preferred codec index number <replaceable>x</replaceable> (beginning with zero).</para>
427 <function name="SIPCHANINFO" language="en_US">
429 Gets the specified SIP parameter from the current channel.
432 <parameter name="item" required="true">
435 <para>The IP address of the peer.</para>
438 <para>The source IP address of the peer.</para>
441 <para>The URI from the <literal>From:</literal> header.</para>
444 <para>The URI from the <literal>Contact:</literal> header.</para>
446 <enum name="useragent">
447 <para>The useragent.</para>
449 <enum name="peername">
450 <para>The name of the peer.</para>
452 <enum name="t38passthrough">
453 <para><literal>1</literal> if T38 is offered or enabled in this channel,
454 otherwise <literal>0</literal>.</para>
461 <function name="CHECKSIPDOMAIN" language="en_US">
463 Checks if domain is a local domain.
466 <parameter name="domain" required="true" />
469 <para>This function checks if the <replaceable>domain</replaceable> in the argument is configured
470 as a local SIP domain that this Asterisk server is configured to handle.
471 Returns the domain name if it is locally handled, otherwise an empty string.
472 Check the <literal>domain=</literal> configuration in <filename>sip.conf</filename>.</para>
486 #define MAX(a,b) ((a) > (b) ? (a) : (b))
489 /* Arguments for find_peer */
490 #define FINDUSERS (1 << 0)
491 #define FINDPEERS (1 << 1)
492 #define FINDALLDEVICES (FINDUSERS | FINDPEERS)
494 #define SIPBUFSIZE 512 /*!< Buffer size for many operations */
496 #define XMIT_ERROR -2
498 #define SIP_RESERVED ";/?:@&=+$,# " /*!< Reserved characters in the username part of the URI */
500 /* #define VOCAL_DATA_HACK */
502 #define DEFAULT_DEFAULT_EXPIRY 120
503 #define DEFAULT_MIN_EXPIRY 60
504 #define DEFAULT_MAX_EXPIRY 3600
505 #define DEFAULT_MWI_EXPIRY 3600
506 #define DEFAULT_REGISTRATION_TIMEOUT 20
507 #define DEFAULT_MAX_FORWARDS "70"
509 /* guard limit must be larger than guard secs */
510 /* guard min must be < 1000, and should be >= 250 */
511 #define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
512 #define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
514 #define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
515 GUARD_PCT turns out to be lower than this, it
516 will use this time instead.
517 This is in milliseconds. */
518 #define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
519 below EXPIRY_GUARD_LIMIT */
520 #define DEFAULT_EXPIRY 900 /*!< Expire slowly */
522 static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
523 static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
524 static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
525 static int mwi_expiry = DEFAULT_MWI_EXPIRY;
527 #define DEFAULT_QUALIFY_GAP 100
528 #define DEFAULT_QUALIFY_PEERS 1
531 #define CALLERID_UNKNOWN "Unknown"
533 #define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
534 #define DEFAULT_QUALIFYFREQ 60 * 1000 /*!< Qualification: How often to check for the host to be up */
535 #define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */
537 #define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
538 #define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
539 #define DEFAULT_TIMER_T1 500 /*!< SIP timer T1 (according to RFC 3261) */
540 #define SIP_TRANS_TIMEOUT 64 * DEFAULT_TIMER_T1 /*!< SIP request timeout (rfc 3261) 64*T1
541 \todo Use known T1 for timeout (peerpoke)
543 #define DEFAULT_TRANS_TIMEOUT -1 /*!< Use default SIP transaction timeout */
544 #define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
546 #define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
547 #define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
548 #define SIP_MIN_PACKET 4096 /*!< Initialize size of memory to allocate for packets */
549 #define MAX_HISTORY_ENTRIES 50 /*!< Max entires in the history list for a sip_pvt */
551 #define INITIAL_CSEQ 101 /*!< Our initial sip sequence number */
553 #define DEFAULT_MAX_SE 1800 /*!< Session-Timer Default Session-Expires period (RFC 4028) */
554 #define DEFAULT_MIN_SE 90 /*!< Session-Timer Default Min-SE period (RFC 4028) */
556 #define SDP_MAX_RTPMAP_CODECS 32 /*!< Maximum number of codecs allowed in received SDP */
558 /*! \brief Global jitterbuffer configuration - by default, jb is disabled */
559 static struct ast_jb_conf default_jbconf =
563 .resync_threshold = -1,
566 static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration */
568 static const char config[] = "sip.conf"; /*!< Main configuration file */
569 static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
574 /*! \brief Authorization scheme for call transfers
576 \note Not a bitfield flag, since there are plans for other modes,
577 like "only allow transfers for authenticated devices" */
579 TRANSFER_OPENFORALL, /*!< Allow all SIP transfers */
580 TRANSFER_CLOSED, /*!< Allow no SIP transfers */
584 /*! \brief The result of a lot of functions */
586 AST_SUCCESS = 0, /*!< FALSE means success, funny enough */
587 AST_FAILURE = -1, /*!< Failure code */
590 /*! \brief States for the INVITE transaction, not the dialog
591 \note this is for the INVITE that sets up the dialog
594 INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
595 INV_CALLING = 1, /*!< Invite sent, no answer */
596 INV_PROCEEDING = 2, /*!< We got/sent 1xx message */
597 INV_EARLY_MEDIA = 3, /*!< We got 18x message with to-tag back */
598 INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
599 INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */
600 INV_TERMINATED = 6, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
601 The only way out of this is a BYE from one side */
602 INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
605 /*! \brief Readable descriptions of device states.
606 \note Should be aligned to above table as index */
607 static const struct invstate2stringtable {
608 const enum invitestates state;
610 } invitestate2string[] = {
612 {INV_CALLING, "Calling (Trying)"},
613 {INV_PROCEEDING, "Proceeding "},
614 {INV_EARLY_MEDIA, "Early media"},
615 {INV_COMPLETED, "Completed (done)"},
616 {INV_CONFIRMED, "Confirmed (up)"},
617 {INV_TERMINATED, "Done"},
618 {INV_CANCELLED, "Cancelled"}
621 /*! \brief When sending a SIP message, we can send with a few options, depending on
622 type of SIP request. UNRELIABLE is moslty used for responses to repeated requests,
623 where the original response would be sent RELIABLE in an INVITE transaction */
625 XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
626 If it fails, it's critical and will cause a teardown of the session */
627 XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
628 XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */
631 /*! \brief Results from the parse_register() function */
632 enum parse_register_result {
633 PARSE_REGISTER_FAILED,
634 PARSE_REGISTER_UPDATE,
635 PARSE_REGISTER_QUERY,
638 /*! \brief Type of subscription, based on the packages we do support, see \ref subscription_types */
639 enum subscriptiontype {
648 /*! \brief Subscription types that we support. We support
649 - dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
650 - SIMPLE presence used for device status
651 - Voicemail notification subscriptions
653 static const struct cfsubscription_types {
654 enum subscriptiontype type;
655 const char * const event;
656 const char * const mediatype;
657 const char * const text;
658 } subscription_types[] = {
659 { NONE, "-", "unknown", "unknown" },
660 /* RFC 4235: SIP Dialog event package */
661 { DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" },
662 { CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */
663 { PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */
664 { XPIDF_XML, "presence", "application/xpidf+xml", "xpidf+xml" }, /* Pre-RFC 3863 with MS additions */
665 { MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
669 /*! \brief Authentication types - proxy or www authentication
670 \note Endpoints, like Asterisk, should always use WWW authentication to
671 allow multiple authentications in the same call - to the proxy and
679 /*! \brief Authentication result from check_auth* functions */
680 enum check_auth_result {
681 AUTH_DONT_KNOW = -100, /*!< no result, need to check further */
682 /* XXX maybe this is the same as AUTH_NOT_FOUND */
685 AUTH_CHALLENGE_SENT = 1,
686 AUTH_SECRET_FAILED = -1,
687 AUTH_USERNAME_MISMATCH = -2,
688 AUTH_NOT_FOUND = -3, /*!< returned by register_verify */
690 AUTH_UNKNOWN_DOMAIN = -5,
691 AUTH_PEER_NOT_DYNAMIC = -6,
692 AUTH_ACL_FAILED = -7,
693 AUTH_BAD_TRANSPORT = -8,
696 /*! \brief States for outbound registrations (with register= lines in sip.conf */
697 enum sipregistrystate {
698 REG_STATE_UNREGISTERED = 0, /*!< We are not registered
699 * \note Initial state. We should have a timeout scheduled for the initial
700 * (or next) registration transmission, calling sip_reregister
703 REG_STATE_REGSENT, /*!< Registration request sent
704 * \note sent initial request, waiting for an ack or a timeout to
705 * retransmit the initial request.
708 REG_STATE_AUTHSENT, /*!< We have tried to authenticate
709 * \note entered after transmit_register with auth info,
710 * waiting for an ack.
713 REG_STATE_REGISTERED, /*!< Registered and done */
715 REG_STATE_REJECTED, /*!< Registration rejected *
716 * \note only used when the remote party has an expire larger than
717 * our max-expire. This is a final state from which we do not
718 * recover (not sure how correctly).
721 REG_STATE_TIMEOUT, /*!< Registration timed out *
722 * \note XXX unused */
724 REG_STATE_NOAUTH, /*!< We have no accepted credentials
725 * \note fatal - no chance to proceed */
727 REG_STATE_FAILED, /*!< Registration failed after several tries
728 * \note fatal - no chance to proceed */
731 /*! \brief Modes in which Asterisk can be configured to run SIP Session-Timers */
733 SESSION_TIMER_MODE_INVALID = 0, /*!< Invalid value */
734 SESSION_TIMER_MODE_ACCEPT, /*!< Honor inbound Session-Timer requests */
735 SESSION_TIMER_MODE_ORIGINATE, /*!< Originate outbound and honor inbound requests */
736 SESSION_TIMER_MODE_REFUSE /*!< Ignore inbound Session-Timers requests */
739 /*! \brief The entity playing the refresher role for Session-Timers */
741 SESSION_TIMER_REFRESHER_AUTO, /*!< Negotiated */
742 SESSION_TIMER_REFRESHER_UAC, /*!< Session is refreshed by the UAC */
743 SESSION_TIMER_REFRESHER_UAS /*!< Session is refreshed by the UAS */
746 /*! \brief Define some implemented SIP transports
747 \note Asterisk does not support SCTP or UDP/DTLS
750 SIP_TRANSPORT_UDP = 1, /*!< Unreliable transport for SIP, needs retransmissions */
751 SIP_TRANSPORT_TCP = 1 << 1, /*!< Reliable, but unsecure */
752 SIP_TRANSPORT_TLS = 1 << 2, /*!< TCP/TLS - reliable and secure transport for signalling */
755 /*! \brief definition of a sip proxy server
757 * For outbound proxies, a sip_peer will contain a reference to a
758 * dynamically allocated instance of a sip_proxy. A sip_pvt may also
759 * contain a reference to a peer's outboundproxy, or it may contain
760 * a reference to the sip_cfg.outboundproxy.
763 char name[MAXHOSTNAMELEN]; /*!< DNS name of domain/host or IP */
764 struct sockaddr_in ip; /*!< Currently used IP address and port */
765 time_t last_dnsupdate; /*!< When this was resolved */
766 enum sip_transport transport;
767 int force; /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
768 /* Room for a SRV record chain based on the name */
771 /*! \brief argument for the 'show channels|subscriptions' callback. */
772 struct __show_chan_arg {
775 int numchans; /* return value */
779 /*! \brief States whether a SIP message can create a dialog in Asterisk. */
780 enum can_create_dialog {
781 CAN_NOT_CREATE_DIALOG,
783 CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
786 /*! \brief SIP Request methods known by Asterisk
788 \note Do _NOT_ make any changes to this enum, or the array following it;
789 if you think you are doing the right thing, you are probably
790 not doing the right thing. If you think there are changes
791 needed, get someone else to review them first _before_
792 submitting a patch. If these two lists do not match properly
793 bad things will happen.
797 SIP_UNKNOWN, /*!< Unknown response */
798 SIP_RESPONSE, /*!< Not request, response to outbound request */
799 SIP_REGISTER, /*!< Registration to the mothership, tell us where you are located */
800 SIP_OPTIONS, /*!< Check capabilities of a device, used for "ping" too */
801 SIP_NOTIFY, /*!< Status update, Part of the event package standard, result of a SUBSCRIBE or a REFER */
802 SIP_INVITE, /*!< Set up a session */
803 SIP_ACK, /*!< End of a three-way handshake started with INVITE. */
804 SIP_PRACK, /*!< Reliable pre-call signalling. Not supported in Asterisk. */
805 SIP_BYE, /*!< End of a session */
806 SIP_REFER, /*!< Refer to another URI (transfer) */
807 SIP_SUBSCRIBE, /*!< Subscribe for updates (voicemail, session status, device status, presence) */
808 SIP_MESSAGE, /*!< Text messaging */
809 SIP_UPDATE, /*!< Update a dialog. We can send UPDATE; but not accept it */
810 SIP_INFO, /*!< Information updates during a session */
811 SIP_CANCEL, /*!< Cancel an INVITE */
812 SIP_PUBLISH, /*!< Not supported in Asterisk */
813 SIP_PING, /*!< Not supported at all, no standard but still implemented out there */
816 /*! \brief Settings for the 'notifycid' option, see sip.conf.sample for details. */
817 enum notifycid_setting {
823 /*! \brief The core structure to setup dialogs. We parse incoming messages by using
824 structure and then route the messages according to the type.
826 \note Note that sip_methods[i].id == i must hold or the code breaks */
827 static const struct cfsip_methods {
829 int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
831 enum can_create_dialog can_create;
833 { SIP_UNKNOWN, RTP, "-UNKNOWN-", CAN_CREATE_DIALOG },
834 { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
835 { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
836 { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
837 { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
838 { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
839 { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
840 { SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG },
841 { SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG },
842 { SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG },
843 { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE", CAN_CREATE_DIALOG },
844 { SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG },
845 { SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG },
846 { SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG },
847 { SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG },
848 { SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD },
849 { SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
852 /*! Define SIP option tags, used in Require: and Supported: headers
853 We need to be aware of these properties in the phones to use
854 the replace: header. We should not do that without knowing
855 that the other end supports it...
856 This is nothing we can configure, we learn by the dialog
857 Supported: header on the REGISTER (peer) or the INVITE
859 We are not using many of these today, but will in the future.
860 This is documented in RFC 3261
863 #define NOT_SUPPORTED 0
866 #define SIP_OPT_REPLACES (1 << 0)
867 #define SIP_OPT_100REL (1 << 1)
868 #define SIP_OPT_TIMER (1 << 2)
869 #define SIP_OPT_EARLY_SESSION (1 << 3)
870 #define SIP_OPT_JOIN (1 << 4)
871 #define SIP_OPT_PATH (1 << 5)
872 #define SIP_OPT_PREF (1 << 6)
873 #define SIP_OPT_PRECONDITION (1 << 7)
874 #define SIP_OPT_PRIVACY (1 << 8)
875 #define SIP_OPT_SDP_ANAT (1 << 9)
876 #define SIP_OPT_SEC_AGREE (1 << 10)
877 #define SIP_OPT_EVENTLIST (1 << 11)
878 #define SIP_OPT_GRUU (1 << 12)
879 #define SIP_OPT_TARGET_DIALOG (1 << 13)
880 #define SIP_OPT_NOREFERSUB (1 << 14)
881 #define SIP_OPT_HISTINFO (1 << 15)
882 #define SIP_OPT_RESPRIORITY (1 << 16)
883 #define SIP_OPT_FROMCHANGE (1 << 17)
884 #define SIP_OPT_RECLISTINV (1 << 18)
885 #define SIP_OPT_RECLISTSUB (1 << 19)
886 #define SIP_OPT_OUTBOUND (1 << 20)
887 #define SIP_OPT_UNKNOWN (1 << 21)
890 /*! \brief List of well-known SIP options. If we get this in a require,
891 we should check the list and answer accordingly. */
892 static const struct cfsip_options {
893 int id; /*!< Bitmap ID */
894 int supported; /*!< Supported by Asterisk ? */
895 char * const text; /*!< Text id, as in standard */
896 } sip_options[] = { /* XXX used in 3 places */
897 /* RFC3262: PRACK 100% reliability */
898 { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" },
899 /* RFC3959: SIP Early session support */
900 { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" },
901 /* SIMPLE events: RFC4662 */
902 { SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" },
903 /* RFC 4916- Connected line ID updates */
904 { SIP_OPT_FROMCHANGE, NOT_SUPPORTED, "from-change" },
905 /* GRUU: Globally Routable User Agent URI's */
906 { SIP_OPT_GRUU, NOT_SUPPORTED, "gruu" },
907 /* RFC4244 History info */
908 { SIP_OPT_HISTINFO, NOT_SUPPORTED, "histinfo" },
909 /* RFC3911: SIP Join header support */
910 { SIP_OPT_JOIN, NOT_SUPPORTED, "join" },
911 /* Disable the REFER subscription, RFC 4488 */
912 { SIP_OPT_NOREFERSUB, NOT_SUPPORTED, "norefersub" },
913 /* SIP outbound - the final NAT battle - draft-sip-outbound */
914 { SIP_OPT_OUTBOUND, NOT_SUPPORTED, "outbound" },
915 /* RFC3327: Path support */
916 { SIP_OPT_PATH, NOT_SUPPORTED, "path" },
917 /* RFC3840: Callee preferences */
918 { SIP_OPT_PREF, NOT_SUPPORTED, "pref" },
919 /* RFC3312: Precondition support */
920 { SIP_OPT_PRECONDITION, NOT_SUPPORTED, "precondition" },
921 /* RFC3323: Privacy with proxies*/
922 { SIP_OPT_PRIVACY, NOT_SUPPORTED, "privacy" },
923 /* RFC-ietf-sip-uri-list-conferencing-02.txt conference invite lists */
924 { SIP_OPT_RECLISTINV, NOT_SUPPORTED, "recipient-list-invite" },
925 /* RFC-ietf-sip-uri-list-subscribe-02.txt - subscription lists */
926 { SIP_OPT_RECLISTSUB, NOT_SUPPORTED, "recipient-list-subscribe" },
927 /* RFC3891: Replaces: header for transfer */
928 { SIP_OPT_REPLACES, SUPPORTED, "replaces" },
929 /* One version of Polycom firmware has the wrong label */
930 { SIP_OPT_REPLACES, SUPPORTED, "replace" },
931 /* RFC4412 Resource priorities */
932 { SIP_OPT_RESPRIORITY, NOT_SUPPORTED, "resource-priority" },
933 /* RFC3329: Security agreement mechanism */
934 { SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" },
935 /* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
936 { SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp-anat" },
937 /* RFC4028: SIP Session-Timers */
938 { SIP_OPT_TIMER, SUPPORTED, "timer" },
939 /* RFC4538: Target-dialog */
940 { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "tdialog" },
944 /*! \brief SIP Methods we support
945 \todo This string should be set dynamically. We only support REFER and SUBSCRIBE if we have
946 allowsubscribe and allowrefer on in sip.conf.
948 #define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
950 /*! \brief SIP Extensions we support
951 \note This should be generated based on the previous array
952 in combination with settings.
953 \todo We should not have "timer" if it's disabled in the configuration file.
955 #define SUPPORTED_EXTENSIONS "replaces, timer"
957 /*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
958 #define STANDARD_SIP_PORT 5060
959 /*! \brief Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS */
960 #define STANDARD_TLS_PORT 5061
962 /*! \note in many SIP headers, absence of a port number implies port 5060,
963 * and this is why we cannot change the above constant.
964 * There is a limited number of places in asterisk where we could,
965 * in principle, use a different "default" port number, but
966 * we do not support this feature at the moment.
967 * You can run Asterisk with SIP on a different port with a configuration
968 * option. If you change this value, the signalling will be incorrect.
971 /*! \name DefaultValues Default values, set and reset in reload_config before reading configuration
973 These are default values in the source. There are other recommended values in the
974 sip.conf.sample for new installations. These may differ to keep backwards compatibility,
975 yet encouraging new behaviour on new installations
978 #define DEFAULT_CONTEXT "default" /*!< The default context for [general] section as well as devices */
979 #define DEFAULT_MOHINTERPRET "default" /*!< The default music class */
980 #define DEFAULT_MOHSUGGEST ""
981 #define DEFAULT_VMEXTEN "asterisk" /*!< Default voicemail extension */
982 #define DEFAULT_CALLERID "asterisk" /*!< Default caller ID */
983 #define DEFAULT_NOTIFYMIME "application/simple-message-summary"
984 #define DEFAULT_ALLOWGUEST TRUE
985 #define DEFAULT_RTPKEEPALIVE 0 /*!< Default RTPkeepalive setting */
986 #define DEFAULT_CALLCOUNTER FALSE
987 #define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */
988 #define DEFAULT_COMPACTHEADERS FALSE /*!< Send compact (one-character) SIP headers. Default off */
989 #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. */
990 #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. */
991 #define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
992 #define DEFAULT_TOS_TEXT 0 /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
993 #define DEFAULT_COS_SIP 4 /*!< Level 2 class of service for SIP signalling */
994 #define DEFAULT_COS_AUDIO 5 /*!< Level 2 class of service for audio media */
995 #define DEFAULT_COS_VIDEO 6 /*!< Level 2 class of service for video media */
996 #define DEFAULT_COS_TEXT 5 /*!< Level 2 class of service for text media (T.140) */
997 #define DEFAULT_ALLOW_EXT_DOM TRUE /*!< Allow external domains */
998 #define DEFAULT_REALM "asterisk" /*!< Realm for HTTP digest authentication */
999 #define DEFAULT_NOTIFYRINGING TRUE /*!< Notify devicestate system on ringing state */
1000 #define DEFAULT_NOTIFYCID DISABLED /*!< Include CID with ringing notifications */
1001 #define DEFAULT_PEDANTIC FALSE /*!< Avoid following SIP standards for dialog matching */
1002 #define DEFAULT_AUTOCREATEPEER FALSE /*!< Don't create peers automagically */
1003 #define DEFAULT_MATCHEXTERNIPLOCALLY FALSE /*!< Match extern IP locally default setting */
1004 #define DEFAULT_QUALIFY FALSE /*!< Don't monitor devices */
1005 #define DEFAULT_CALLEVENTS FALSE /*!< Extra manager SIP call events */
1006 #define DEFAULT_ALWAYSAUTHREJECT FALSE /*!< Don't reject authentication requests always */
1007 #define DEFAULT_REGEXTENONQUALIFY FALSE
1008 #define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
1009 #define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
1010 #ifndef DEFAULT_USERAGENT
1011 #define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
1012 #define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
1013 #define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
1017 /*! \name DefaultSettings
1018 Default setttings are used as a channel setting and as a default when
1022 static char default_language[MAX_LANGUAGE];
1023 static char default_callerid[AST_MAX_EXTENSION];
1024 static char default_fromdomain[AST_MAX_EXTENSION];
1025 static char default_notifymime[AST_MAX_EXTENSION];
1026 static int default_qualify; /*!< Default Qualify= setting */
1027 static char default_vmexten[AST_MAX_EXTENSION];
1028 static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
1029 static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
1030 * a bridged channel on hold */
1031 static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
1032 static int default_maxcallbitrate; /*!< Maximum bitrate for call */
1033 static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
1034 static unsigned int default_transports; /*!< Default Transports (enum sip_transport) that are acceptable */
1035 static unsigned int default_primary_transport; /*!< Default primary Transport (enum sip_transport) for outbound connections to devices */
1039 /*! \name GlobalSettings
1040 Global settings apply to the channel (often settings you can change in the general section
1044 /*! \brief a place to store all global settings for the sip channel driver
1045 These are settings that will be possibly to apply on a group level later on.
1046 \note Do not add settings that only apply to the channel itself and can't
1047 be applied to devices (trunks, services, phones)
1049 struct sip_settings {
1050 int peer_rtupdate; /*!< G: Update database with registration data for peer? */
1051 int rtsave_sysname; /*!< G: Save system name at registration? */
1052 int ignore_regexpire; /*!< G: Ignore expiration of peer */
1053 int rtautoclear; /*!< Realtime ?? */
1054 int directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */
1055 int pedanticsipchecking; /*!< Extra checking ? Default off */
1056 int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
1057 int srvlookup; /*!< SRV Lookup on or off. Default is on */
1058 int allowguest; /*!< allow unauthenticated peers to connect? */
1059 int alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
1060 int compactheaders; /*!< send compact sip headers */
1061 int allow_external_domains; /*!< Accept calls to external SIP domains? */
1062 int callevents; /*!< Whether we send manager events or not */
1063 int regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
1064 int matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
1065 int notifyringing; /*!< Send notifications on ringing */
1066 int notifyhold; /*!< Send notifications on hold */
1067 enum notifycid_setting notifycid; /*!< Send CID with ringing notifications */
1068 enum transfermodes allowtransfer; /*!< SIP Refer restriction scheme */
1069 int allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
1070 the global setting is in globals_flags[1] */
1071 char realm[MAXHOSTNAMELEN]; /*!< Default realm */
1072 struct sip_proxy outboundproxy; /*!< Outbound proxy */
1073 char default_context[AST_MAX_CONTEXT];
1074 char default_subscribecontext[AST_MAX_CONTEXT];
1077 static struct sip_settings sip_cfg;
1079 static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
1081 static int global_relaxdtmf; /*!< Relax DTMF */
1082 static int global_rtptimeout; /*!< Time out call if no RTP */
1083 static int global_rtpholdtimeout; /*!< Time out call if no RTP during hold */
1084 static int global_rtpkeepalive; /*!< Send RTP keepalives */
1085 static int global_reg_timeout;
1086 static int global_regattempts_max; /*!< Registration attempts before giving up */
1087 static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer
1088 call-limit to 999. When we remove the call-limit from the code, we can make it
1089 with just a boolean flag in the device structure */
1090 static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
1091 static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
1092 static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
1093 static unsigned int global_tos_text; /*!< IP type of service for text RTP packets */
1094 static unsigned int global_cos_sip; /*!< 802.1p class of service for SIP packets */
1095 static unsigned int global_cos_audio; /*!< 802.1p class of service for audio RTP packets */
1096 static unsigned int global_cos_video; /*!< 802.1p class of service for video RTP packets */
1097 static unsigned int global_cos_text; /*!< 802.1p class of service for text RTP packets */
1098 static int recordhistory; /*!< Record SIP history. Off by default */
1099 static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
1100 static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
1101 static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
1102 static char global_sdpsession[AST_MAX_EXTENSION]; /*!< SDP session name for the SIP channel */
1103 static char global_sdpowner[AST_MAX_EXTENSION]; /*!< SDP owner name for the SIP channel */
1104 static int global_authfailureevents; /*!< Whether we send authentication failure manager events or not. Default no. */
1105 static int global_t1; /*!< T1 time */
1106 static int global_t1min; /*!< T1 roundtrip time minimum */
1107 static int global_timer_b; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
1108 static int global_autoframing; /*!< Turn autoframing on or off. */
1109 static int global_qualifyfreq; /*!< Qualify frequency */
1110 static int global_qualify_gap; /*!< Time between our group of peer pokes */
1111 static int global_qualify_peers; /*!< Number of peers to poke at a given time */
1114 /*! \brief Codecs that we support by default: */
1115 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
1117 static enum st_mode global_st_mode; /*!< Mode of operation for Session-Timers */
1118 static enum st_refresher global_st_refresher; /*!< Session-Timer refresher */
1119 static int global_min_se; /*!< Lowest threshold for session refresh interval */
1120 static int global_max_se; /*!< Highest threshold for session refresh interval */
1124 /*! \brief Global list of addresses dynamic peers are not allowed to use */
1125 static struct ast_ha *global_contact_ha = NULL;
1126 static int global_dynamic_exclude_static = 0;
1128 /*! \name Object counters @{
1129 * \bug These counters are not handled in a thread-safe way ast_atomic_fetchadd_int()
1130 * should be used to modify these values. */
1131 static int speerobjs = 0; /*!< Static peers */
1132 static int rpeerobjs = 0; /*!< Realtime peers */
1133 static int apeerobjs = 0; /*!< Autocreated peer objects */
1134 static int regobjs = 0; /*!< Registry objects */
1137 static struct ast_flags global_flags[2] = {{0}}; /*!< global SIP_ flags */
1138 static char used_context[AST_MAX_CONTEXT]; /*!< name of automatically created context for unloading */
1141 AST_MUTEX_DEFINE_STATIC(netlock);
1143 /*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
1144 when it's doing something critical. */
1145 AST_MUTEX_DEFINE_STATIC(monlock);
1147 AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
1149 /*! \brief This is the thread for the monitor which checks for input on the channels
1150 which are not currently in use. */
1151 static pthread_t monitor_thread = AST_PTHREADT_NULL;
1153 static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
1154 static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
1156 static struct sched_context *sched; /*!< The scheduling context */
1157 static struct io_context *io; /*!< The IO context */
1158 static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
1160 #define DEC_CALL_LIMIT 0
1161 #define INC_CALL_LIMIT 1
1162 #define DEC_CALL_RINGING 2
1163 #define INC_CALL_RINGING 3
1165 /*! \brief The SIP socket definition */
1167 enum sip_transport type; /*!< UDP, TCP or TLS */
1168 int fd; /*!< Filed descriptor, the actual socket */
1170 struct ast_tcptls_session_instance *tcptls_session; /* If tcp or tls, a socket manager */
1173 /*! \brief sip_request: The data grabbed from the UDP socket
1176 * Incoming messages: we first store the data from the socket in data[],
1177 * adding a trailing \0 to make string parsing routines happy.
1178 * Then call parse_request() and req.method = find_sip_method();
1179 * to initialize the other fields. The \r\n at the end of each line is
1180 * replaced by \0, so that data[] is not a conforming SIP message anymore.
1181 * After this processing, rlPart1 is set to non-NULL to remember
1182 * that we can run get_header() on this kind of packet.
1184 * parse_request() splits the first line as follows:
1185 * Requests have in the first line method uri SIP/2.0
1186 * rlPart1 = method; rlPart2 = uri;
1187 * Responses have in the first line SIP/2.0 NNN description
1188 * rlPart1 = SIP/2.0; rlPart2 = NNN + description;
1190 * For outgoing packets, we initialize the fields with init_req() or init_resp()
1191 * (which fills the first line to "METHOD uri SIP/2.0" or "SIP/2.0 code text"),
1192 * and then fill the rest with add_header() and add_line().
1193 * The \r\n at the end of the line are still there, so the get_header()
1194 * and similar functions don't work on these packets.
1197 struct sip_request {
1198 ptrdiff_t rlPart1; /*!< Offset of the SIP Method Name or "SIP/2.0" protocol version */
1199 ptrdiff_t rlPart2; /*!< Offset of the Request URI or Response Status */
1200 int len; /*!< bytes used in data[], excluding trailing null terminator. Rarely used. */
1201 int headers; /*!< # of SIP Headers */
1202 int method; /*!< Method of this request */
1203 int lines; /*!< Body Content */
1204 unsigned int sdp_start; /*!< the line number where the SDP begins */
1205 unsigned int sdp_end; /*!< the line number where the SDP ends */
1206 char debug; /*!< print extra debugging if non zero */
1207 char has_to_tag; /*!< non-zero if packet has To: tag */
1208 char ignore; /*!< if non-zero This is a re-transmit, ignore it */
1209 /* Array of offsets into the request string of each SIP header*/
1210 ptrdiff_t header[SIP_MAX_HEADERS];
1211 /* Array of offsets into the request string of each SDP line*/
1212 ptrdiff_t line[SIP_MAX_LINES];
1213 struct ast_str *data;
1214 /* XXX Do we need to unref socket.ser when the request goes away? */
1215 struct sip_socket socket; /*!< The socket used for this request */
1216 AST_LIST_ENTRY(sip_request) next;
1219 /* \brief given a sip_request and an offset, return the char * that resides there
1221 * It used to be that rlPart1, rlPart2, and the header and line arrays were character
1222 * pointers. They are now offsets into the ast_str portion of the sip_request structure.
1223 * To avoid adding a bunch of redundant pointer arithmetic to the code, this macro is
1224 * provided to retrieve the string at a particular offset within the request's buffer
1226 #define REQ_OFFSET_TO_STR(req,offset) (ast_str_buffer((req)->data) + ((req)->offset))
1228 /*! \brief structure used in transfers */
1230 struct ast_channel *chan1; /*!< First channel involved */
1231 struct ast_channel *chan2; /*!< Second channel involved */
1232 struct sip_request req; /*!< Request that caused the transfer (REFER) */
1233 int seqno; /*!< Sequence number */
1238 /*! \brief Parameters to the transmit_invite function */
1239 struct sip_invite_param {
1240 int addsipheaders; /*!< Add extra SIP headers */
1241 const char *uri_options; /*!< URI options to add to the URI */
1242 const char *vxml_url; /*!< VXML url for Cisco phones */
1243 char *auth; /*!< Authentication */
1244 char *authheader; /*!< Auth header */
1245 enum sip_auth_type auth_type; /*!< Authentication type */
1246 const char *replaces; /*!< Replaces header for call transfers */
1247 int transfer; /*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */
1250 /*! \brief Structure to save routing information for a SIP session */
1252 struct sip_route *next;
1256 /*! \brief Modes for SIP domain handling in the PBX */
1258 SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */
1259 SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */
1262 /*! \brief Domain data structure.
1263 \note In the future, we will connect this to a configuration tree specific
1267 char domain[MAXHOSTNAMELEN]; /*!< SIP domain we are responsible for */
1268 char context[AST_MAX_EXTENSION]; /*!< Incoming context for this domain */
1269 enum domain_mode mode; /*!< How did we find this domain? */
1270 AST_LIST_ENTRY(domain) list; /*!< List mechanics */
1273 static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */
1276 /*! \brief sip_history: Structure for saving transactions within a SIP dialog */
1277 struct sip_history {
1278 AST_LIST_ENTRY(sip_history) list;
1279 char event[0]; /* actually more, depending on needs */
1282 AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
1284 /*! \brief sip_auth: Credentials for authentication to other SIP services */
1286 char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */
1287 char username[256]; /*!< Username */
1288 char secret[256]; /*!< Secret */
1289 char md5secret[256]; /*!< MD5Secret */
1290 struct sip_auth *next; /*!< Next auth structure in list */
1294 Various flags for the flags field in the pvt structure
1295 Trying to sort these up (one or more of the following):
1299 When flags are used by multiple structures, it is important that
1300 they have a common layout so it is easy to copy them.
1303 #define SIP_OUTGOING (1 << 0) /*!< D: Direction of the last transaction in this dialog */
1304 #define SIP_RINGING (1 << 2) /*!< D: Have sent 180 ringing */
1305 #define SIP_PROGRESS_SENT (1 << 3) /*!< D: Have sent 183 message progress */
1306 #define SIP_NEEDREINVITE (1 << 4) /*!< D: Do we need to send another reinvite? */
1307 #define SIP_PENDINGBYE (1 << 5) /*!< D: Need to send bye after we ack? */
1308 #define SIP_GOTREFER (1 << 6) /*!< D: Got a refer? */
1309 #define SIP_CALL_LIMIT (1 << 7) /*!< D: Call limit enforced for this call */
1310 #define SIP_INC_COUNT (1 << 8) /*!< D: Did this dialog increment the counter of in-use calls? */
1311 #define SIP_INC_RINGING (1 << 9) /*!< D: Did this connection increment the counter of in-use calls? */
1312 #define SIP_DEFER_BYE_ON_TRANSFER (1 << 10) /*!< D: Do not hangup at first ast_hangup */
1314 #define SIP_PROMISCREDIR (1 << 11) /*!< DP: Promiscuous redirection */
1315 #define SIP_TRUSTRPID (1 << 12) /*!< DP: Trust RPID headers? */
1316 #define SIP_USEREQPHONE (1 << 13) /*!< DP: Add user=phone to numeric URI. Default off */
1317 #define SIP_USECLIENTCODE (1 << 14) /*!< DP: Trust X-ClientCode info message */
1319 /* DTMF flags - see str2dtmfmode() and dtmfmode2str() */
1320 #define SIP_DTMF (7 << 15) /*!< DP: DTMF Support: five settings, uses three bits */
1321 #define SIP_DTMF_RFC2833 (0 << 15) /*!< DP: DTMF Support: RTP DTMF - "rfc2833" */
1322 #define SIP_DTMF_INBAND (1 << 15) /*!< DP: DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
1323 #define SIP_DTMF_INFO (2 << 15) /*!< DP: DTMF Support: SIP Info messages - "info" */
1324 #define SIP_DTMF_AUTO (3 << 15) /*!< DP: DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
1325 #define SIP_DTMF_SHORTINFO (4 << 15) /*!< DP: DTMF Support: SIP Info messages - "info" - short variant */
1327 /* NAT settings - see nat2str() */
1328 #define SIP_NAT (3 << 18) /*!< DP: four settings, uses two bits */
1329 #define SIP_NAT_NEVER (0 << 18) /*!< DP: No nat support */
1330 #define SIP_NAT_RFC3581 (1 << 18) /*!< DP: NAT RFC3581 */
1331 #define SIP_NAT_ROUTE (2 << 18) /*!< DP: NAT Only ROUTE */
1332 #define SIP_NAT_ALWAYS (3 << 18) /*!< DP: NAT Both ROUTE and RFC3581 */
1334 /* re-INVITE related settings */
1335 #define SIP_REINVITE (7 << 20) /*!< DP: four settings, uses three bits */
1336 #define SIP_REINVITE_NONE (0 << 20) /*!< DP: no reinvite allowed */
1337 #define SIP_CAN_REINVITE (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */
1338 #define SIP_CAN_REINVITE_NAT (2 << 20) /*!< DP: allow media reinvite when new peer is behind NAT */
1339 #define SIP_REINVITE_UPDATE (4 << 20) /*!< DP: use UPDATE (RFC3311) when reinviting this peer */
1341 /* "insecure" settings - see insecure2str() */
1342 #define SIP_INSECURE (3 << 23) /*!< DP: three settings, uses two bits */
1343 #define SIP_INSECURE_NONE (0 << 23) /*!< DP: secure mode */
1344 #define SIP_INSECURE_PORT (1 << 23) /*!< DP: don't require matching port for incoming requests */
1345 #define SIP_INSECURE_INVITE (1 << 24) /*!< DP: don't require authentication for incoming INVITEs */
1347 /* Sending PROGRESS in-band settings */
1348 #define SIP_PROG_INBAND (3 << 25) /*!< DP: three settings, uses two bits */
1349 #define SIP_PROG_INBAND_NEVER (0 << 25)
1350 #define SIP_PROG_INBAND_NO (1 << 25)
1351 #define SIP_PROG_INBAND_YES (2 << 25)
1353 #define SIP_SENDRPID (1 << 29) /*!< DP: Remote Party-ID Support */
1354 #define SIP_G726_NONSTANDARD (1 << 31) /*!< DP: Use non-standard packing for G726-32 data */
1356 /*! \brief Flags to copy from peer/user to dialog */
1357 #define SIP_FLAGS_TO_COPY \
1358 (SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
1359 SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
1360 SIP_USEREQPHONE | SIP_INSECURE)
1364 a second page of flags (for flags[1] */
1366 /* realtime flags */
1367 #define SIP_PAGE2_RTCACHEFRIENDS (1 << 0) /*!< GP: Should we keep RT objects in memory for extended time? */
1368 #define SIP_PAGE2_RTAUTOCLEAR (1 << 2) /*!< GP: Should we clean memory from peers after expiry? */
1369 /* Space for addition of other realtime flags in the future */
1370 #define SIP_PAGE2_STATECHANGEQUEUE (1 << 9) /*!< D: Unsent state pending change exists */
1372 #define SIP_PAGE2_PREFERRED_CODEC (1 << 13) /*!< GDP: Only respond with single most preferred joint codec */
1373 #define SIP_PAGE2_VIDEOSUPPORT (1 << 14) /*!< DP: Video supported if offered? */
1374 #define SIP_PAGE2_TEXTSUPPORT (1 << 15) /*!< GDP: Global text enable */
1375 #define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 16) /*!< GP: Allow subscriptions from this peer? */
1376 #define SIP_PAGE2_ALLOWOVERLAP (1 << 17) /*!< DP: Allow overlap dialing ? */
1377 #define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 18) /*!< GP: Only issue MWI notification if subscribed to */
1378 #define SIP_PAGE2_IGNORESDPVERSION (1 << 19) /*!< GDP: Ignore the SDP session version number we receive and treat all sessions as new */
1380 #define SIP_PAGE2_T38SUPPORT (7 << 20) /*!< GDP: T38 Fax Passthrough Support */
1381 #define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 20) /*!< GDP: T38 Fax Passthrough Support */
1382 #define SIP_PAGE2_T38SUPPORT_RTP (2 << 20) /*!< GDP: T38 Fax Passthrough Support (not implemented) */
1383 #define SIP_PAGE2_T38SUPPORT_TCP (4 << 20) /*!< GDP: T38 Fax Passthrough Support (not implemented) */
1385 #define SIP_PAGE2_CALL_ONHOLD (3 << 23) /*!< D: Call hold states: */
1386 #define SIP_PAGE2_CALL_ONHOLD_ACTIVE (1 << 23) /*!< D: Active hold */
1387 #define SIP_PAGE2_CALL_ONHOLD_ONEDIR (2 << 23) /*!< D: One directional hold */
1388 #define SIP_PAGE2_CALL_ONHOLD_INACTIVE (3 << 23) /*!< D: Inactive hold */
1390 #define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< DP: Compensate for buggy RFC2833 implementations */
1391 #define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< DP: Buggy CISCO MWI fix */
1392 #define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 27) /*!< 29: Has a dialog been established? */
1393 #define SIP_PAGE2_FAX_DETECT (1 << 28) /*!< DP: Fax Detection support */
1394 #define SIP_PAGE2_REGISTERTRYING (1 << 29) /*!< DP: Send 100 Trying on REGISTER attempts */
1395 #define SIP_PAGE2_UDPTL_DESTINATION (1 << 30) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
1396 #define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 31) /*!< DP: Always set up video, even if endpoints don't support it */
1398 #define SIP_PAGE2_FLAGS_TO_COPY \
1399 (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
1400 SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
1401 SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
1402 SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC)
1406 /*! \name SIPflagsT38
1407 T.38 set of flags */
1410 #define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0 (unset)*/
1411 #define T38FAX_TRANSCODING_MMR (1 << 1) /*!< Default: 0 (unset)*/
1412 #define T38FAX_TRANSCODING_JBIG (1 << 2) /*!< Default: 0 (unset)*/
1413 /* Rate management */
1414 #define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF (0 << 3)
1415 #define T38FAX_RATE_MANAGEMENT_LOCAL_TCF (1 << 3) /*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */
1416 /* UDP Error correction */
1417 #define T38FAX_UDP_EC_NONE (0 << 4) /*!< two bits, if unset NO t38UDPEC field in T38 SDP*/
1418 #define T38FAX_UDP_EC_FEC (1 << 4) /*!< Set for t38UDPFEC */
1419 #define T38FAX_UDP_EC_REDUNDANCY (2 << 4) /*!< Set for t38UDPRedundancy */
1420 /* T38 Spec version */
1421 #define T38FAX_VERSION (3 << 6) /*!< two bits, 2 values so far, up to 4 values max */
1422 #define T38FAX_VERSION_0 (0 << 6) /*!< Version 0 */
1423 #define T38FAX_VERSION_1 (1 << 6) /*!< Version 1 */
1424 /* Maximum Fax Rate */
1425 #define T38FAX_RATE_2400 (1 << 8) /*!< 2400 bps t38FaxRate */
1426 #define T38FAX_RATE_4800 (1 << 9) /*!< 4800 bps t38FaxRate */
1427 #define T38FAX_RATE_7200 (1 << 10) /*!< 7200 bps t38FaxRate */
1428 #define T38FAX_RATE_9600 (1 << 11) /*!< 9600 bps t38FaxRate */
1429 #define T38FAX_RATE_12000 (1 << 12) /*!< 12000 bps t38FaxRate */
1430 #define T38FAX_RATE_14400 (1 << 13) /*!< 14400 bps t38FaxRate */
1432 /*!< This is default: NO MMR and JBIG transcoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
1433 static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
1436 /*! \brief debugging state
1437 * We store separately the debugging requests from the config file
1438 * and requests from the CLI. Debugging is enabled if either is set
1439 * (which means that if sipdebug is set in the config file, we can
1440 * only turn it off by reloading the config).
1444 sip_debug_config = 1,
1445 sip_debug_console = 2,
1448 static enum sip_debug_e sipdebug;
1450 /*! \brief extra debugging for 'text' related events.
1451 * At the moment this is set together with sip_debug_console.
1452 * \note It should either go away or be implemented properly.
1454 static int sipdebug_text;
1456 /*! \brief T38 States for a call */
1458 T38_DISABLED = 0, /*!< Not enabled */
1459 T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */
1460 T38_PEER_DIRECT, /*!< Offered from peer */
1461 T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */
1462 T38_ENABLED /*!< Negotiated (enabled) */
1465 /*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
1466 struct t38properties {
1467 struct ast_flags t38support; /*!< Flag for udptl, rtp or tcp support for this session */
1468 int capability; /*!< Our T38 capability */
1469 int peercapability; /*!< Peers T38 capability */
1470 int jointcapability; /*!< Supported T38 capability at both ends */
1471 enum t38state state; /*!< T.38 state */
1472 unsigned int direct:1; /*!< Whether the T38 came from the initial invite or not */
1475 /*! \brief Parameters to know status of transfer */
1477 REFER_IDLE, /*!< No REFER is in progress */
1478 REFER_SENT, /*!< Sent REFER to transferee */
1479 REFER_RECEIVED, /*!< Received REFER from transferrer */
1480 REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING (unused) */
1481 REFER_ACCEPTED, /*!< Accepted by transferee */
1482 REFER_RINGING, /*!< Target Ringing */
1483 REFER_200OK, /*!< Answered by transfer target */
1484 REFER_FAILED, /*!< REFER declined - go on */
1485 REFER_NOAUTH /*!< We had no auth for REFER */
1488 /*! \brief generic struct to map between strings and integers.
1489 * Fill it with x-s pairs, terminate with an entry with s = NULL;
1490 * Then you can call map_x_s(...) to map an integer to a string,
1491 * and map_s_x() for the string -> integer mapping.
1498 static const struct _map_x_s referstatusstrings[] = {
1499 { REFER_IDLE, "<none>" },
1500 { REFER_SENT, "Request sent" },
1501 { REFER_RECEIVED, "Request received" },
1502 { REFER_CONFIRMED, "Confirmed" },
1503 { REFER_ACCEPTED, "Accepted" },
1504 { REFER_RINGING, "Target ringing" },
1505 { REFER_200OK, "Done" },
1506 { REFER_FAILED, "Failed" },
1507 { REFER_NOAUTH, "Failed - auth failure" },
1508 { -1, NULL} /* terminator */
1511 /*! \brief Structure to handle SIP transfers. Dynamically allocated when needed
1512 \note OEJ: Should be moved to string fields */
1514 char refer_to[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO extension */
1515 char refer_to_domain[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO domain */
1516 char refer_to_urioption[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO uri options */
1517 char refer_to_context[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO context */
1518 char referred_by[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
1519 char referred_by_name[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
1520 char refer_contact[AST_MAX_EXTENSION]; /*!< Place to store Contact info from a REFER extension */
1521 char replaces_callid[SIPBUFSIZE]; /*!< Replace info: callid */
1522 char replaces_callid_totag[SIPBUFSIZE/2]; /*!< Replace info: to-tag */
1523 char replaces_callid_fromtag[SIPBUFSIZE/2]; /*!< Replace info: from-tag */
1524 struct sip_pvt *refer_call; /*!< Call we are referring. This is just a reference to a
1525 * dialog owned by someone else, so we should not destroy
1526 * it when the sip_refer object goes.
1528 int attendedtransfer; /*!< Attended or blind transfer? */
1529 int localtransfer; /*!< Transfer to local domain? */
1530 enum referstatus status; /*!< REFER status */
1534 /*! \brief Structure that encapsulates all attributes related to running
1535 * SIP Session-Timers feature on a per dialog basis.
1538 int st_active; /*!< Session-Timers on/off */
1539 int st_interval; /*!< Session-Timers negotiated session refresh interval */
1540 int st_schedid; /*!< Session-Timers ast_sched scheduler id */
1541 enum st_refresher st_ref; /*!< Session-Timers session refresher */
1542 int st_expirys; /*!< Session-Timers number of expirys */
1543 int st_active_peer_ua; /*!< Session-Timers on/off in peer UA */
1544 int st_cached_min_se; /*!< Session-Timers cached Min-SE */
1545 int st_cached_max_se; /*!< Session-Timers cached Session-Expires */
1546 enum st_mode st_cached_mode; /*!< Session-Timers cached M.O. */
1547 enum st_refresher st_cached_ref; /*!< Session-Timers cached refresher */
1551 /*! \brief Structure that encapsulates all attributes related to configuration
1552 * of SIP Session-Timers feature on a per user/peer basis.
1555 enum st_mode st_mode_oper; /*!< Mode of operation for Session-Timers */
1556 enum st_refresher st_ref; /*!< Session-Timer refresher */
1557 int st_min_se; /*!< Lowest threshold for session refresh interval */
1558 int st_max_se; /*!< Highest threshold for session refresh interval */
1564 /*! \brief Structure used for each SIP dialog, ie. a call, a registration, a subscribe.
1565 * Created and initialized by sip_alloc(), the descriptor goes into the list of
1566 * descriptors (dialoglist).
1569 struct sip_pvt *next; /*!< Next dialog in chain */
1570 enum invitestates invitestate; /*!< Track state of SIP_INVITEs */
1571 int method; /*!< SIP method that opened this dialog */
1572 AST_DECLARE_STRING_FIELDS(
1573 AST_STRING_FIELD(callid); /*!< Global CallID */
1574 AST_STRING_FIELD(randdata); /*!< Random data */
1575 AST_STRING_FIELD(accountcode); /*!< Account code */
1576 AST_STRING_FIELD(realm); /*!< Authorization realm */
1577 AST_STRING_FIELD(nonce); /*!< Authorization nonce */
1578 AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
1579 AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
1580 AST_STRING_FIELD(domain); /*!< Authorization domain */
1581 AST_STRING_FIELD(from); /*!< The From: header */
1582 AST_STRING_FIELD(useragent); /*!< User agent in SIP request */
1583 AST_STRING_FIELD(exten); /*!< Extension where to start */
1584 AST_STRING_FIELD(context); /*!< Context for this call */
1585 AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */
1586 AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */
1587 AST_STRING_FIELD(fromdomain); /*!< Domain to show in the from field */
1588 AST_STRING_FIELD(fromuser); /*!< User to show in the user field */
1589 AST_STRING_FIELD(fromname); /*!< Name to show in the user field */
1590 AST_STRING_FIELD(tohost); /*!< Host we should put in the "to" field */
1591 AST_STRING_FIELD(todnid); /*!< DNID of this call (overrides host) */
1592 AST_STRING_FIELD(language); /*!< Default language for this call */
1593 AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */
1594 AST_STRING_FIELD(mohsuggest); /*!< MOH class to suggest when putting a peer on hold */
1595 AST_STRING_FIELD(rdnis); /*!< Referring DNIS */
1596 AST_STRING_FIELD(redircause); /*!< Referring cause */
1597 AST_STRING_FIELD(theirtag); /*!< Their tag */
1598 AST_STRING_FIELD(username); /*!< [user] name */
1599 AST_STRING_FIELD(peername); /*!< [peer] name, not set if [user] */
1600 AST_STRING_FIELD(authname); /*!< Who we use for authentication */
1601 AST_STRING_FIELD(uri); /*!< Original requested URI */
1602 AST_STRING_FIELD(okcontacturi); /*!< URI from the 200 OK on INVITE */
1603 AST_STRING_FIELD(peersecret); /*!< Password */
1604 AST_STRING_FIELD(peermd5secret);
1605 AST_STRING_FIELD(cid_num); /*!< Caller*ID number */
1606 AST_STRING_FIELD(cid_name); /*!< Caller*ID name */
1607 AST_STRING_FIELD(fullcontact); /*!< The Contact: that the UA registers with us */
1608 /* we only store the part in <brackets> in this field. */
1609 AST_STRING_FIELD(our_contact); /*!< Our contact header */
1610 AST_STRING_FIELD(rpid); /*!< Our RPID header */
1611 AST_STRING_FIELD(rpid_from); /*!< Our RPID From header */
1612 AST_STRING_FIELD(url); /*!< URL to be sent with next message to peer */
1613 AST_STRING_FIELD(parkinglot); /*!< Parkinglot */
1615 char via[128]; /*!< Via: header */
1616 struct sip_socket socket; /*!< The socket used for this dialog */
1617 unsigned int ocseq; /*!< Current outgoing seqno */
1618 unsigned int icseq; /*!< Current incoming seqno */
1619 ast_group_t callgroup; /*!< Call group */
1620 ast_group_t pickupgroup; /*!< Pickup group */
1621 int lastinvite; /*!< Last Cseq of invite */
1622 int lastnoninvite; /*!< Last Cseq of non-invite */
1623 struct ast_flags flags[2]; /*!< SIP_ flags */
1625 /* boolean or small integers that don't belong in flags */
1626 char do_history; /*!< Set if we want to record history */
1627 char alreadygone; /*!< already destroyed by our peer */
1628 char needdestroy; /*!< need to be destroyed by the monitor thread */
1629 char outgoing_call; /*!< this is an outgoing call */
1630 char answered_elsewhere; /*!< This call is cancelled due to answer on another channel */
1631 char novideo; /*!< Didn't get video in invite, don't offer */
1632 char notext; /*!< Text not supported (?) */
1634 int timer_t1; /*!< SIP timer T1, ms rtt */
1635 int timer_b; /*!< SIP timer B, ms */
1636 unsigned int sipoptions; /*!< Supported SIP options on the other end */
1637 unsigned int reqsipoptions; /*!< Required SIP options on the other end */
1638 struct ast_codec_pref prefs; /*!< codec prefs */
1639 int capability; /*!< Special capability (codec) */
1640 int jointcapability; /*!< Supported capability at both ends (codecs) */
1641 int peercapability; /*!< Supported peer capability */
1642 int prefcodec; /*!< Preferred codec (outbound only) */
1643 int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
1644 int jointnoncodeccapability; /*!< Joint Non codec capability */
1645 int redircodecs; /*!< Redirect codecs */
1646 int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
1647 struct sip_proxy *outboundproxy; /*!< Outbound proxy for this dialog. Use ref_proxy to set this instead of setting it directly*/
1648 struct t38properties t38; /*!< T38 settings */
1649 struct sockaddr_in udptlredirip; /*!< Where our T.38 UDPTL should be going if not to us */
1650 struct ast_udptl *udptl; /*!< T.38 UDPTL session */
1651 int callingpres; /*!< Calling presentation */
1652 int authtries; /*!< Times we've tried to authenticate */
1653 int expiry; /*!< How long we take to expire */
1654 long branch; /*!< The branch identifier of this session */
1655 long invite_branch; /*!< The branch used when we sent the initial INVITE */
1656 char tag[11]; /*!< Our tag for this session */
1657 int sessionid; /*!< SDP Session ID */
1658 int sessionversion; /*!< SDP Session Version */
1659 uint64_t sessionversion_remote; /*!< Remote UA's SDP Session Version */
1660 int session_modify; /*!< Session modification request true/false */
1661 struct sockaddr_in sa; /*!< Our peer */
1662 struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
1663 struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
1664 struct sockaddr_in tredirip; /*!< Where our Text RTP should be going if not to us */
1665 time_t lastrtprx; /*!< Last RTP received */
1666 time_t lastrtptx; /*!< Last RTP sent */
1667 int rtptimeout; /*!< RTP timeout time */
1668 struct sockaddr_in recv; /*!< Received as */
1669 struct sockaddr_in ourip; /*!< Our IP (as seen from the outside) */
1670 struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
1671 struct sip_route *route; /*!< Head of linked list of routing steps (fm Record-Route) */
1672 int route_persistant; /*!< Is this the "real" route? */
1673 struct ast_variable *notify_headers; /*!< Custom notify type */
1674 struct sip_auth *peerauth; /*!< Realm authentication */
1675 int noncecount; /*!< Nonce-count */
1676 char lastmsg[256]; /*!< Last Message sent/received */
1677 int amaflags; /*!< AMA Flags */
1678 int pendinginvite; /*!< Any pending INVITE or state NOTIFY (in subscribe pvt's) ? (seqno of this) */
1679 struct sip_request initreq; /*!< Latest request that opened a new transaction
1681 NOT the request that opened the dialog
1684 int initid; /*!< Auto-congest ID if appropriate (scheduler) */
1685 int waitid; /*!< Wait ID for scheduler after 491 or other delays */
1686 int autokillid; /*!< Auto-kill ID (scheduler) */
1687 int t38id; /*!< T.38 Response ID */
1688 enum transfermodes allowtransfer; /*!< REFER: restriction scheme */
1689 struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
1690 enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
1691 int stateid; /*!< SUBSCRIBE: ID for devicestate subscriptions */
1692 int laststate; /*!< SUBSCRIBE: Last known extension state */
1693 int dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */
1695 struct ast_dsp *vad; /*!< Inband DTMF Detection dsp */
1697 struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one
1698 Used in peerpoke, mwi subscriptions */
1699 struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */
1700 struct ast_rtp *rtp; /*!< RTP Session */
1701 struct ast_rtp *vrtp; /*!< Video RTP session */
1702 struct ast_rtp *trtp; /*!< Text RTP session */
1703 struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
1704 struct sip_history_head *history; /*!< History of this SIP dialog */
1705 size_t history_entries; /*!< Number of entires in the history */
1706 struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
1707 AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */
1708 int request_queue_sched_id; /*!< Scheduler ID of any scheduled action to process queued requests */
1709 struct sip_invite_param *options; /*!< Options for INVITE */
1710 int autoframing; /*!< The number of Asters we group in a Pyroflax
1711 before strolling to the Grokyzpå
1712 (A bit unsure of this, please correct if
1714 struct sip_st_dlg *stimer; /*!< SIP Session-Timers */
1716 int red; /*!< T.140 RTP Redundancy */
1717 int hangupcause; /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
1719 struct sip_subscription_mwi *mwi; /*!< If this is a subscription MWI dialog, to which subscription */
1724 * Here we implement the container for dialogs (sip_pvt), defining
1725 * generic wrapper functions to ease the transition from the current
1726 * implementation (a single linked list) to a different container.
1727 * In addition to a reference to the container, we need functions to lock/unlock
1728 * the container and individual items, and functions to add/remove
1729 * references to the individual items.
1731 struct ao2_container *dialogs;
1733 #define sip_pvt_lock(x) ao2_lock(x)
1734 #define sip_pvt_trylock(x) ao2_trylock(x)
1735 #define sip_pvt_unlock(x) ao2_unlock(x)
1738 * when we create or delete references, make sure to use these
1739 * functions so we keep track of the refcounts.
1740 * To simplify the code, we allow a NULL to be passed to dialog_unref().
1743 #define dialog_ref(arg1,arg2) dialog_ref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
1744 #define dialog_unref(arg1,arg2) dialog_unref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
1746 static struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
1749 _ao2_ref_debug(p, 1, tag, file, line, func);
1751 ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
1755 static struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
1758 _ao2_ref_debug(p, -1, tag, file, line, func);
1762 static struct sip_pvt *dialog_ref(struct sip_pvt *p, char *tag)
1767 ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
1771 static struct sip_pvt *dialog_unref(struct sip_pvt *p, char *tag)
1779 /*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission
1780 * Packets are linked in a list, whose head is in the struct sip_pvt they belong to.
1781 * Each packet holds a reference to the parent struct sip_pvt.
1782 * This structure is allocated in __sip_reliable_xmit() and only for packets that
1783 * require retransmissions.
1786 struct sip_pkt *next; /*!< Next packet in linked list */
1787 int retrans; /*!< Retransmission number */
1788 int method; /*!< SIP method for this packet */
1789 int seqno; /*!< Sequence number */
1790 char is_resp; /*!< 1 if this is a response packet (e.g. 200 OK), 0 if it is a request */
1791 char is_fatal; /*!< non-zero if there is a fatal error */
1792 struct sip_pvt *owner; /*!< Owner AST call */
1793 int retransid; /*!< Retransmission ID */
1794 int timer_a; /*!< SIP timer A, retransmission timer */
1795 int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */
1796 int packetlen; /*!< Length of packet */
1797 struct ast_str *data;
1801 * \brief A peer's mailbox
1803 * We could use STRINGFIELDS here, but for only two strings, it seems like
1804 * too much effort ...
1806 struct sip_mailbox {
1809 /*! Associated MWI subscription */
1810 struct ast_event_sub *event_sub;
1811 AST_LIST_ENTRY(sip_mailbox) entry;
1814 enum sip_peer_type {
1815 SIP_TYPE_PEER = (1 << 0),
1816 SIP_TYPE_USER = (1 << 1),
1819 /*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host)
1821 /* XXX field 'name' must be first otherwise sip_addrcmp() will fail, as will astobj2 hashing of the structure */
1823 char name[80]; /*!< the unique name of this object */
1824 AST_DECLARE_STRING_FIELDS(
1825 AST_STRING_FIELD(secret); /*!< Password for inbound auth */
1826 AST_STRING_FIELD(md5secret); /*!< Password in MD5 */
1827 AST_STRING_FIELD(remotesecret); /*!< Remote secret (trunks, remote devices) */
1828 AST_STRING_FIELD(context); /*!< Default context for incoming calls */
1829 AST_STRING_FIELD(subscribecontext); /*!< Default context for subscriptions */
1830 AST_STRING_FIELD(username); /*!< Temporary username until registration */
1831 AST_STRING_FIELD(accountcode); /*!< Account code */
1832 AST_STRING_FIELD(tohost); /*!< If not dynamic, IP address */
1833 AST_STRING_FIELD(regexten); /*!< Extension to register (if regcontext is used) */
1834 AST_STRING_FIELD(fromuser); /*!< From: user when calling this peer */
1835 AST_STRING_FIELD(fromdomain); /*!< From: domain when calling this peer */
1836 AST_STRING_FIELD(fullcontact); /*!< Contact registered with us (not in sip.conf) */
1837 AST_STRING_FIELD(cid_num); /*!< Caller ID num */
1838 AST_STRING_FIELD(cid_name); /*!< Caller ID name */
1839 AST_STRING_FIELD(vmexten); /*!< Dialplan extension for MWI notify message*/
1840 AST_STRING_FIELD(language); /*!< Default language for prompts */
1841 AST_STRING_FIELD(mohinterpret); /*!< Music on Hold class */
1842 AST_STRING_FIELD(mohsuggest); /*!< Music on Hold class */
1843 AST_STRING_FIELD(parkinglot); /*!< Parkinglot */
1844 AST_STRING_FIELD(useragent); /*!< User agent in SIP request (saved from registration) */
1846 struct sip_socket socket; /*!< Socket used for this peer */
1847 unsigned int transports:3; /*!< Transports (enum sip_transport) that are acceptable for this peer */
1848 struct sip_auth *auth; /*!< Realm authentication list */
1849 int amaflags; /*!< AMA Flags (for billing) */
1850 int callingpres; /*!< Calling id presentation */
1851 int inUse; /*!< Number of calls in use */
1852 int inRinging; /*!< Number of calls ringing */
1853 int onHold; /*!< Peer has someone on hold */
1854 int call_limit; /*!< Limit of concurrent calls */
1855 int busy_level; /*!< Level of active channels where we signal busy */
1856 enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
1857 struct ast_codec_pref prefs; /*!< codec prefs */
1859 unsigned int sipoptions; /*!< Supported SIP options */
1860 struct ast_flags flags[2]; /*!< SIP_ flags */
1862 /*! Mailboxes that this peer cares about */
1863 AST_LIST_HEAD_NOLOCK(, sip_mailbox) mailboxes;
1865 /* things that don't belong in flags */
1866 char is_realtime; /*!< this is a 'realtime' peer */
1867 char rt_fromcontact; /*!< copy fromcontact from realtime */
1868 char host_dynamic; /*!< Dynamic Peers register with Asterisk */
1869 char selfdestruct; /*!< Automatic peers need to destruct themselves */
1870 char the_mark; /*!< moved out of ASTOBJ into struct proper; That which bears the_mark should be deleted! */
1872 int expire; /*!< When to expire this peer registration */
1873 int capability; /*!< Codec capability */
1874 int rtptimeout; /*!< RTP timeout */
1875 int rtpholdtimeout; /*!< RTP Hold Timeout */
1876 int rtpkeepalive; /*!< Send RTP packets for keepalive */
1877 ast_group_t callgroup; /*!< Call group */
1878 ast_group_t pickupgroup; /*!< Pickup group */
1879 struct sip_proxy *outboundproxy; /*!< Outbound proxy for this peer */
1880 struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
1881 struct sockaddr_in addr; /*!< IP address of peer */
1882 int maxcallbitrate; /*!< Maximum Bitrate for a video call */
1885 struct sip_pvt *call; /*!< Call pointer */
1886 int pokeexpire; /*!< When to expire poke (qualify= checking) */
1887 int lastms; /*!< How long last response took (in ms), or -1 for no response */
1888 int maxms; /*!< Max ms we will accept for the host to be up, 0 to not monitor */
1889 int qualifyfreq; /*!< Qualification: How often to check for the host to be up */
1890 struct timeval ps; /*!< Time for sending SIP OPTION in sip_pke_peer() */
1891 struct sockaddr_in defaddr; /*!< Default IP address, used until registration */
1892 struct ast_ha *ha; /*!< Access control list */
1893 struct ast_ha *contactha; /*!< Restrict what IPs are allowed in the Contact header (for registration) */
1894 struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
1895 struct sip_pvt *mwipvt; /*!< Subscription for MWI */
1897 struct sip_st_cfg stimer; /*!< SIP Session-Timers */
1898 int timer_t1; /*!< The maximum T1 value for the peer */
1899 int timer_b; /*!< The maximum timer B (transaction timeouts) */
1900 int deprecated_username; /*!< If it's a realtime peer, are they using the deprecated "username" instead of "defaultuser" */
1902 /*XXX Seems like we suddenly have two flags with the same content. Why? To be continued... */
1903 enum sip_peer_type type; /*!< Distinguish between "user" and "peer" types. This is used solely for CLI and manager commands */
1908 * \brief Registrations with other SIP proxies
1910 * Created by sip_register(), the entry is linked in the 'regl' list,
1911 * and never deleted (other than at 'sip reload' or module unload times).
1912 * The entry always has a pending timeout, either waiting for an ACK to
1913 * the REGISTER message (in which case we have to retransmit the request),
1914 * or waiting for the next REGISTER message to be sent (either the initial one,
1915 * or once the previously completed registration one expires).
1916 * The registration can be in one of many states, though at the moment
1917 * the handling is a bit mixed.
1919 * XXX \todo Reference count handling for this object has some problems with
1920 * respect to scheduler entries. The ref count is handled in some places,
1921 * but not all of them. There are some places where references get leaked
1922 * when this scheduler entry gets cancelled. At worst, this would cause
1923 * memory leaks on reloads if registrations get removed from configuration.
1925 struct sip_registry {
1926 ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1);
1927 AST_DECLARE_STRING_FIELDS(
1928 AST_STRING_FIELD(callid); /*!< Global Call-ID */
1929 AST_STRING_FIELD(realm); /*!< Authorization realm */
1930 AST_STRING_FIELD(nonce); /*!< Authorization nonce */
1931 AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
1932 AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
1933 AST_STRING_FIELD(domain); /*!< Authorization domain */
1934 AST_STRING_FIELD(username); /*!< Who we are registering as */
1935 AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */
1936 AST_STRING_FIELD(hostname); /*!< Domain or host we register to */
1937 AST_STRING_FIELD(secret); /*!< Password in clear text */
1938 AST_STRING_FIELD(md5secret); /*!< Password in md5 */
1939 AST_STRING_FIELD(callback); /*!< Contact extension */
1940 AST_STRING_FIELD(random);
1942 enum sip_transport transport; /*!< Transport for this registration UDP, TCP or TLS */
1943 int portno; /*!< Optional port override */
1944 int expire; /*!< Sched ID of expiration */
1945 int expiry; /*!< Value to use for the Expires header */
1946 int regattempts; /*!< Number of attempts (since the last success) */
1947 int timeout; /*!< sched id of sip_reg_timeout */
1948 int refresh; /*!< How often to refresh */
1949 struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
1950 enum sipregistrystate regstate; /*!< Registration state (see above) */
1951 struct timeval regtime; /*!< Last successful registration time */
1952 int callid_valid; /*!< 0 means we haven't chosen callid for this registry yet. */
1953 unsigned int ocseq; /*!< Sequence number we got to for REGISTERs for this registry */
1954 struct ast_dnsmgr_entry *dnsmgr; /*!< DNS refresh manager for register */
1955 struct sockaddr_in us; /*!< Who the server thinks we are */
1956 int noncecount; /*!< Nonce-count */
1957 char lastmsg[256]; /*!< Last Message sent/received */
1960 /*! \brief Definition of a thread that handles a socket */
1961 struct sip_threadinfo {
1964 struct ast_tcptls_session_instance *tcptls_session;
1965 enum sip_transport type; /*!< We keep a copy of the type here so we can display it in the connection list */
1966 AST_LIST_ENTRY(sip_threadinfo) list;
1969 /*! \brief Definition of an MWI subscription to another server */
1970 struct sip_subscription_mwi {
1971 ASTOBJ_COMPONENTS_FULL(struct sip_subscription_mwi,1,1);
1972 AST_DECLARE_STRING_FIELDS(
1973 AST_STRING_FIELD(username); /*!< Who we are sending the subscription as */
1974 AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */
1975 AST_STRING_FIELD(hostname); /*!< Domain or host we subscribe to */
1976 AST_STRING_FIELD(secret); /*!< Password in clear text */
1977 AST_STRING_FIELD(mailbox); /*!< Mailbox store to put MWI into */
1979 enum sip_transport transport; /*!< Transport to use */
1980 int portno; /*!< Optional port override */
1981 int resub; /*!< Sched ID of resubscription */
1982 unsigned int subscribed:1; /*!< Whether we are currently subscribed or not */
1983 struct sip_pvt *call; /*!< Outbound subscription dialog */
1984 struct ast_dnsmgr_entry *dnsmgr; /*!< DNS refresh manager for subscription */
1985 struct sockaddr_in us; /*!< Who the server thinks we are */
1988 /* --- Hash tables of various objects --------*/
1991 static int hash_peer_size = 17;
1992 static int hash_dialog_size = 17;
1993 static int hash_user_size = 17;
1995 static int hash_peer_size = 563; /*!< Size of peer hash table, prime number preferred! */
1996 static int hash_dialog_size = 563;
1997 static int hash_user_size = 563;
2000 /*! \brief The thread list of TCP threads */
2001 static AST_LIST_HEAD_STATIC(threadl, sip_threadinfo);
2003 /*! \brief The peer list: Users, Peers and Friends */
2004 struct ao2_container *peers;
2005 struct ao2_container *peers_by_ip;
2007 /*! \brief The register list: Other SIP proxies we register with and place calls to */
2008 static struct ast_register_list {
2009 ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
2013 /*! \brief The MWI subscription list */
2014 static struct ast_subscription_mwi_list {
2015 ASTOBJ_CONTAINER_COMPONENTS(struct sip_subscription_mwi);
2019 * \note The only member of the peer used here is the name field
2021 static int peer_hash_cb(const void *obj, const int flags)
2023 const struct sip_peer *peer = obj;
2025 return ast_str_case_hash(peer->name);
2029 * \note The only member of the peer used here is the name field
2031 static int peer_cmp_cb(void *obj, void *arg, int flags)
2033 struct sip_peer *peer = obj, *peer2 = arg;
2035 return !strcasecmp(peer->name, peer2->name) ? CMP_MATCH | CMP_STOP : 0;
2039 * \note the peer's addr struct provides to fields combined to make a key: the sin_addr.s_addr and sin_port fields.
2041 static int peer_iphash_cb(const void *obj, const int flags)
2043 const struct sip_peer *peer = obj;
2044 int ret1 = peer->addr.sin_addr.s_addr;
2048 if (ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT)) {
2051 return ret1 + peer->addr.sin_port;
2056 * \note the peer's addr struct provides to fields combined to make a key: the sin_addr.s_addr and sin_port fields.
2058 static int peer_ipcmp_cb(void *obj, void *arg, int flags)
2060 struct sip_peer *peer = obj, *peer2 = arg;
2062 if (peer->addr.sin_addr.s_addr != peer2->addr.sin_addr.s_addr)
2065 if (!ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT) && !ast_test_flag(&peer2->flags[0], SIP_INSECURE_PORT)) {
2066 if (peer->addr.sin_port == peer2->addr.sin_port)
2067 return CMP_MATCH | CMP_STOP;
2071 return CMP_MATCH | CMP_STOP;
2075 * \note The only member of the dialog used here callid string
2077 static int dialog_hash_cb(const void *obj, const int flags)
2079 const struct sip_pvt *pvt = obj;
2081 return ast_str_case_hash(pvt->callid);
2085 * \note The only member of the dialog used here callid string
2087 static int dialog_cmp_cb(void *obj, void *arg, int flags)
2089 struct sip_pvt *pvt = obj, *pvt2 = arg;
2091 return !strcasecmp(pvt->callid, pvt2->callid) ? CMP_MATCH | CMP_STOP : 0;
2094 static int temp_pvt_init(void *);
2095 static void temp_pvt_cleanup(void *);
2097 /*! \brief A per-thread temporary pvt structure */
2098 AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
2101 static void ts_ast_rtp_destroy(void *);
2103 AST_THREADSTORAGE_CUSTOM(ts_audio_rtp, NULL, ts_ast_rtp_destroy);
2104 AST_THREADSTORAGE_CUSTOM(ts_video_rtp, NULL, ts_ast_rtp_destroy);
2105 AST_THREADSTORAGE_CUSTOM(ts_text_rtp, NULL, ts_ast_rtp_destroy);
2108 /*! \brief Authentication list for realm authentication
2109 * \todo Move the sip_auth list to AST_LIST */
2110 static struct sip_auth *authl = NULL;
2113 /* --- Sockets and networking --------------*/
2115 /*! \brief Main socket for UDP SIP communication.
2117 * sipsock is shared between the SIP manager thread (which handles reload
2118 * requests), the udp io handler (sipsock_read()) and the user routines that
2119 * issue udp writes (using __sip_xmit()).
2120 * The socket is -1 only when opening fails (this is a permanent condition),
2121 * or when we are handling a reload() that changes its address (this is
2122 * a transient situation during which we might have a harmless race, see
2123 * below). Because the conditions for the race to be possible are extremely
2124 * rare, we don't want to pay the cost of locking on every I/O.
2125 * Rather, we remember that when the race may occur, communication is
2126 * bound to fail anyways, so we just live with this event and let
2127 * the protocol handle this above us.
2129 static int sipsock = -1;
2131 static struct sockaddr_in bindaddr; /*!< UDP: The address we bind to */
2133 /*! \brief our (internal) default address/port to put in SIP/SDP messages
2134 * internip is initialized picking a suitable address from one of the
2135 * interfaces, and the same port number we bind to. It is used as the
2136 * default address/port in SIP messages, and as the default address
2137 * (but not port) in SDP messages.
2139 static struct sockaddr_in internip;
2141 /*! \brief our external IP address/port for SIP sessions.
2142 * externip.sin_addr is only set when we know we might be behind
2143 * a NAT, and this is done using a variety of (mutually exclusive)
2144 * ways from the config file:
2146 * + with "externip = host[:port]" we specify the address/port explicitly.
2147 * The address is looked up only once when (re)loading the config file;
2149 * + with "externhost = host[:port]" we do a similar thing, but the
2150 * hostname is stored in externhost, and the hostname->IP mapping
2151 * is refreshed every 'externrefresh' seconds;
2153 * + with "stunaddr = host[:port]" we run queries every externrefresh seconds
2154 * to the specified server, and store the result in externip.
2156 * Other variables (externhost, externexpire, externrefresh) are used
2157 * to support the above functions.
2159 static struct sockaddr_in externip; /*!< External IP address if we are behind NAT */
2161 static char externhost[MAXHOSTNAMELEN]; /*!< External host name */
2162 static time_t externexpire; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
2163 static int externrefresh = 10;
2164 static struct sockaddr_in stunaddr; /*!< stun server address */
2166 /*! \brief List of local networks
2167 * We store "localnet" addresses from the config file into an access list,
2168 * marked as 'DENY', so the call to ast_apply_ha() will return
2169 * AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local'
2170 * (i.e. presumably public) addresses.
2172 static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
2174 static int ourport_tcp; /*!< The port used for TCP connections */
2175 static int ourport_tls; /*!< The port used for TCP/TLS connections */
2176 static struct sockaddr_in debugaddr;
2178 static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
2180 /*! some list management macros. */
2182 #define UNLINK(element, head, prev) do { \
2184 (prev)->next = (element)->next; \
2186 (head) = (element)->next; \
2189 enum t38_action_flag {
2190 SDP_T38_NONE = 0, /*!< Do not modify T38 information at all */
2191 SDP_T38_INITIATE, /*!< Remote side has requested T38 with us */
2192 SDP_T38_ACCEPT, /*!< Remote side accepted our T38 request */
2195 /*---------------------------- Forward declarations of functions in chan_sip.c */
2196 /* Note: This is added to help splitting up chan_sip.c into several files
2197 in coming releases. */
2199 /*--- PBX interface functions */
2200 static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
2201 static int sip_devicestate(void *data);
2202 static int sip_sendtext(struct ast_channel *ast, const char *text);
2203 static int sip_call(struct ast_channel *ast, char *dest, int timeout);
2204 static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data, int datalen);
2205 static int sip_hangup(struct ast_channel *ast);
2206 static int sip_answer(struct ast_channel *ast);
2207 static struct ast_frame *sip_read(struct ast_channel *ast);
2208 static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
2209 static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
2210 static int sip_transfer(struct ast_channel *ast, const char *dest);
2211 static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
2212 static int sip_senddigit_begin(struct ast_channel *ast, char digit);
2213 static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration);
2214 static int sip_queryoption(struct ast_channel *chan, int option, void *data, int *datalen);
2215 static const char *sip_get_callid(struct ast_channel *chan);
2217 static int handle_request_do(struct sip_request *req, struct sockaddr_in *sin);
2218 static int sip_standard_port(enum sip_transport type, int port);
2219 static int sip_prepare_socket(struct sip_pvt *p);
2220 static int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport);
2222 /*--- Transmitting responses and requests */
2223 static int sipsock_read(int *id, int fd, short events, void *ignore);
2224 static int __sip_xmit(struct sip_pvt *p, struct ast_str *data, int len);
2225 static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, struct ast_str *data, int len, int fatal, int sipmethod);
2226 static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
2227 static int retrans_pkt(const void *data);
2228 static int transmit_response_using_temp(ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
2229 static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
2230 static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
2231 static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
2232 static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp);
2233 static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
2234 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);
2235 static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
2236 static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable);
2237 static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
2238 static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
2239 static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init);
2240 static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int oldsdp);
2241 static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration);
2242 static int transmit_info_with_vidupdate(struct sip_pvt *p);
2243 static int transmit_message_with_text(struct sip_pvt *p, const char *text);
2244 static int transmit_refer(struct sip_pvt *p, const char *dest);
2245 static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten);
2246 static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
2247 static int transmit_notify_custom(struct sip_pvt *p, struct ast_variable *vars);
2248 static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
2249 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
2250 static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
2251 static void copy_request(struct sip_request *dst, const struct sip_request *src);
2252 static void receive_message(struct sip_pvt *p, struct sip_request *req);
2253 static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req);
2254 static int sip_send_mwi_to_peer(struct sip_peer *peer, const struct ast_event *event, int cache_only);
2256 /*--- Dialog management */
2257 static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
2258 int useglobal_nat, const int intended_method);
2259 static int __sip_autodestruct(const void *data);
2260 static void sip_scheddestroy(struct sip_pvt *p, int ms);
2261 static int sip_cancel_destroy(struct sip_pvt *p);
2262 static struct sip_pvt *sip_destroy(struct sip_pvt *p);
2263 static void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist);
2264 static void *registry_unref(struct sip_registry *reg, char *tag);
2265 static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist);
2266 static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
2267 static void __sip_pretend_ack(struct sip_pvt *p);
2268 static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
2269 static int auto_congest(const void *arg);
2270 static int update_call_counter(struct sip_pvt *fup, int event);
2271 static int hangup_sip2cause(int cause);
2272 static const char *hangup_cause2sip(int cause);
2273 static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method);
2274 static void free_old_route(struct sip_route *route);
2275 static void list_route(struct sip_route *route);
2276 static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards);
2277 static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin,
2278 struct sip_request *req, char *uri);
2279 static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
2280 static void check_pendings(struct sip_pvt *p);
2281 static void *sip_park_thread(void *stuff);
2282 static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno);
2283 static int sip_sipredirect(struct sip_pvt *p, const char *dest);
2285 /*--- Codec handling / SDP */
2286 static void try_suggested_sip_codec(struct sip_pvt *p);
2287 static const char* get_sdp_iterate(int* start, struct sip_request *req, const char *name);
2288 static const char *get_sdp(struct sip_request *req, const char *name);
2289 static int find_sdp(struct sip_request *req);
2290 static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action);
2291 static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
2292 struct ast_str **m_buf, struct ast_str **a_buf,
2293 int debug, int *min_packet_size);
2294 static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
2295 struct ast_str **m_buf, struct ast_str **a_buf,
2297 static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38);
2298 static void do_setnat(struct sip_pvt *p, int natflags);
2299 static void stop_media_flows(struct sip_pvt *p);
2301 /*--- Authentication stuff */
2302 static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
2303 static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
2304 static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
2305 const char *secret, const char *md5secret, int sipmethod,
2306 char *uri, enum xmittype reliable, int ignore);
2307 static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
2308 int sipmethod, char *uri, enum xmittype reliable,
2309 struct sockaddr_in *sin, struct sip_peer **authpeer);
2310 static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);
2312 /*--- Domain handling */
2313 static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
2314 static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context);
2315 static void clear_sip_domains(void);
2317 /*--- SIP realm authentication */
2318 static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, const char *configuration, int lineno);
2319 static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
2320 static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm);
2322 /*--- Misc functions */
2323 static void check_rtp_timeout(struct sip_pvt *dialog, time_t t);
2324 static int sip_do_reload(enum channelreloadreason reason);
2325 static int reload_config(enum channelreloadreason reason);
2326 static int expire_register(const void *data);
2327 static void *do_monitor(void *data);
2328 static int restart_monitor(void);
2329 static void peer_mailboxes_to_str(struct ast_str **mailbox_str, struct sip_peer *peer);
2330 static struct ast_variable *copy_vars(struct ast_variable *src);
2331 /* static int sip_addrcmp(char *name, struct sockaddr_in *sin); Support for peer matching */
2332 static int sip_refer_allocate(struct sip_pvt *p);
2333 static void ast_quiet_chan(struct ast_channel *chan);
2334 static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
2335 static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
2338 * \brief generic function for determining if a correct transport is being
2339 * used to contact a peer
2341 * this is done as a macro so that the "tmpl" var can be passed either a
2342 * sip_request or a sip_peer
2344 #define check_request_transport(peer, tmpl) ({ \
2346 if (peer->socket.type == tmpl->socket.type) \
2348 else if (!(peer->transports & tmpl->socket.type)) {\
2349 ast_log(LOG_ERROR, \
2350 "'%s' is not a valid transport for '%s'. we only use '%s'! ending call.\n", \
2351 get_transport(tmpl->socket.type), peer->name, get_transport_list(peer->transports) \
2354 } else if (peer->socket.type & SIP_TRANSPORT_TLS) { \
2355 ast_log(LOG_WARNING, \
2356 "peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)!\n", \
2357 peer->name, get_transport(tmpl->socket.type) \
2361 "peer '%s' has contacted us over %s even though we prefer %s.\n", \
2362 peer->name, get_transport(tmpl->socket.type), get_transport(peer->socket.type) \
2369 /*--- Device monitoring and Device/extension state/event handling */
2370 static int cb_extensionstate(char *context, char* exten, int state, void *data);
2371 static int sip_devicestate(void *data);
2372 static int sip_poke_noanswer(const void *data);
2373 static int sip_poke_peer(struct sip_peer *peer, int force);
2374 static void sip_poke_all_peers(void);
2375 static void sip_peer_hold(struct sip_pvt *p, int hold);
2376 static void mwi_event_cb(const struct ast_event *, void *);
2378 /*--- Applications, functions, CLI and manager command helpers */
2379 static const char *sip_nat_mode(const struct sip_pvt *p);
2380 static char *sip_show_inuse(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2381 static char *transfermode2str(enum transfermodes mode) attribute_const;
2382 static const char *nat2str(int nat) attribute_const;
2383 static int peer_status(struct sip_peer *peer, char *status, int statuslen);
2384 static char *sip_show_sched(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2385 static char * _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]);
2386 static char *sip_show_peers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2387 static char *sip_show_objects(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2388 static void print_group(int fd, ast_group_t group, int crlf);
2389 static const char *dtmfmode2str(int mode) attribute_const;
2390 static int str2dtmfmode(const char *str) attribute_unused;
2391 static const char *insecure2str(int mode) attribute_const;
2392 static void cleanup_stale_contexts(char *new, char *old);
2393 static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
2394 static const char *domain_mode_to_text(const enum domain_mode mode);
2395 static char *sip_show_domains(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2396 static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
2397 static char *sip_show_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2398 static char *_sip_qualify_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
2399 static char *sip_qualify_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2400 static char *sip_show_registry(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2401 static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2402 static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2403 static char *sip_show_mwi(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2404 static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
2405 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
2406 static char *complete_sip_peer(const char *word, int state, int flags2);
2407 static char *complete_sip_registered_peer(const char *word, int state, int flags2);
2408 static char *complete_sip_show_history(const char *line, const char *word, int pos, int state);
2409 static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
2410 static char *complete_sip_unregister(const char *line, const char *word, int pos, int state);
2411 static char *complete_sipnotify(const char *line, const char *word, int pos, int state);
2412 static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2413 static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2414 static char *sip_show_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2415 static char *sip_do_debug_ip(int fd, char *arg);
2416 static char *sip_do_debug_peer(int fd, char *arg);
2417 static char *sip_do_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2418 static char *sip_cli_notify(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2419 static char *sip_set_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2420 static int sip_dtmfmode(struct ast_channel *chan, void *data);
2421 static int sip_addheader(struct ast_channel *chan, void *data);
2422 static int sip_do_reload(enum channelreloadreason reason);
2423 static char *sip_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2424 static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen);
2427 Functions for enabling debug per IP or fully, or enabling history logging for
2430 static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to debuglog at end of dialog, before destroying data */
2431 static inline int sip_debug_test_addr(const struct sockaddr_in *addr);
2432 static inline int sip_debug_test_pvt(struct sip_pvt *p);
2435 /*! \brief Append to SIP dialog history
2436 \return Always returns 0 */
2437 #define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
2438 static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
2439 static void sip_dump_history(struct sip_pvt *dialog);
2441 /*--- Device object handling */
2442 static struct sip_peer *temp_peer(const char *name);
2443 static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime);
2444 static int update_call_counter(struct sip_pvt *fup, int event);
2445 static void sip_destroy_peer(struct sip_peer *peer);
2446 static void sip_destroy_peer_fn(void *peer);
2447 static void set_peer_defaults(struct sip_peer *peer);
2448 static struct sip_peer *temp_peer(const char *name);
2449 static void register_peer_exten(struct sip_peer *peer, int onoff);
2450 static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime, int forcenamematch, int devstate_only);
2451 static int sip_poke_peer_s(const void *data);
2452 static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
2453 static void reg_source_db(struct sip_peer *peer);
2454 static void destroy_association(struct sip_peer *peer);
2455 static void set_insecure_flags(struct ast_flags *flags, const char *value, int lineno);
2456 static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);
2458 /* Realtime device support */
2459 static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, const char *useragent, int expirey, int deprecated_username, int lastms);
2460 static void update_peer(struct sip_peer *p, int expire);
2461 static struct ast_variable *get_insecure_variable_from_config(struct ast_config *config);
2462 static const char *get_name_from_variable(struct ast_variable *var, const char *newpeername);
2463 static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin, int devstate_only);
2464 static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
2466 /*--- Internal UA client handling (outbound registrations) */
2467 static void ast_sip_ouraddrfor(struct in_addr *them, struct sockaddr_in *us);
2468 static void sip_registry_destroy(struct sip_registry *reg);
2469 static int sip_register(const char *value, int lineno);
2470 static const char *regstate2str(enum sipregistrystate regstate) attribute_const;
2471 static int sip_reregister(const void *data);
2472 static int __sip_do_register(struct sip_registry *r);
2473 static int sip_reg_timeout(const void *data);
2474 static void sip_send_all_registers(void);
2475 static int sip_reinvite_retry(const void *data);
2477 /*--- Parsing SIP requests and responses */
2478 static void append_date(struct sip_request *req); /* Append date to SIP packet */
2479 static int determine_firstline_parts(struct sip_request *req);
2480 static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
2481 static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
2482 static int find_sip_method(const char *msg);
2483 static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported);
2484 static int parse_request(struct sip_request *req);
2485 static const char *get_header(const struct sip_request *req, const char *name);
2486 static const char *referstatus2str(enum referstatus rstatus) attribute_pure;
2487 static int method_match(enum sipmethod id, const char *name);
2488 static void parse_copy(struct sip_request *dst, const struct sip_request *src);
2489 static char *get_in_brackets(char *tmp);
2490 static const char *find_alias(const char *name, const char *_default);
2491 static const char *__get_header(const struct sip_request *req, const char *name, int *start);
2492 static int lws2sws(char *msgbuf, int len);
2493 static void extract_uri(struct sip_pvt *p, struct sip_request *req);
2494 static char *remove_uri_parameters(char *uri);
2495 static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req);
2496 static int get_also_info(struct sip_pvt *p, struct sip_request *oreq);
2497 static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
2498 static int set_address_from_contact(struct sip_pvt *pvt);
2499 static void check_via(struct sip_pvt *p, struct sip_request *req);
2500 static char *get_calleridname(const char *input, char *output, size_t outputsize);
2501 static int get_rpid_num(const char *input, char *output, int maxlen);
2502 static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq);
2503 static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
2504 static int get_msg_text(char *buf, int len, struct sip_request *req, int addnewline);
2505 static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout);
2507 /*-- TCP connection handling ---*/
2508 static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *tcptls_session);
2509 static void *sip_tcp_worker_fn(void *);
2511 /*--- Constructing requests and responses */
2512 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
2513 static int init_req(struct sip_request *req, int sipmethod, const char *recip);
2514 static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch);
2515 static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod);
2516 static int init_resp(struct sip_request *resp, const char *msg);
2517 static inline int resp_needs_contact(const char *msg, enum sipmethod method);
2518 static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req);
2519 static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p);
2520 static void build_via(struct sip_pvt *p);
2521 static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
2522 static int create_addr(struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin, int newdialog);
2523 static char *generate_random_string(char *buf, size_t size);
2524 static void build_callid_pvt(struct sip_pvt *pvt);
2525 static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
2526 static void make_our_tag(char *tagbuf, size_t len);
2527 static int add_header(struct sip_request *req, const char *var, const char *value);
2528 static int add_header_contentLength(struct sip_request *req, int len);
2529 static int add_line(struct sip_request *req, const char *line);
2530 static int add_text(struct sip_request *req, const char *text);
2531 static int add_digit(struct sip_request *req, char digit, unsigned int duration, int mode);
2532 static int add_vidupdate(struct sip_request *req);
2533 static void add_route(struct sip_request *req, struct sip_route *route);
2534 static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field);
2535 static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field);
2536 static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field);
2537 static void set_destination(struct sip_pvt *p, char *uri);
2538 static void append_date(struct sip_request *req);
2539 static void build_contact(struct sip_pvt *p);
2540 static void build_rpid(struct sip_pvt *p);
2542 /*------Request handling functions */
2543 static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock);
2544 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, int *nounlock);
2545 static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, int *nounlock);
2546 static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
2547 static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e);
2548 static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
2549 static int handle_request_message(struct sip_pvt *p, struct sip_request *req);
2550 static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
2551 static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
2552 static int handle_request_options(struct sip_pvt *p, struct sip_request *req);
2553 static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin);
2554 static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
2555 static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno);
2557 /*------Response handling functions */
2558 static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
2559 static void handle_response_notify(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
2560 static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
2561 static void handle_response_subscribe(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
2562 static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
2563 static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
2565 /*----- RTP interface functions */
2566 static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active);
2567 static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
2568 static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
2569 static enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
2570 static int sip_get_codec(struct ast_channel *chan);
2571 static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
2573 /*------ T38 Support --------- */
2574 static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
2575 static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
2576 static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
2577 static void change_t38_state(struct sip_pvt *p, int state);
2579 /*------ Session-Timers functions --------- */
2580 static void proc_422_rsp(struct sip_pvt *p, struct sip_request *rsp);
2581 static int proc_session_timer(const void *vp);
2582 static void stop_session_timer(struct sip_pvt *p);
2583 static void start_session_timer(struct sip_pvt *p);
2584 static void restart_session_timer(struct sip_pvt *p);
2585 static const char *strefresher2str(enum st_refresher r);
2586 static int parse_session_expires(const char *p_hdrval, int *const p_interval, enum st_refresher *const p_ref);
2587 static int parse_minse(const char *p_hdrval, int *const p_interval);
2588 static int st_get_se(struct sip_pvt *, int max);
2589 static enum st_refresher st_get_refresher(struct sip_pvt *);
2590 static enum st_mode st_get_mode(struct sip_pvt *);
2591 static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p);
2593 /*!--- SIP MWI Subscription support */
2594 static int sip_subscribe_mwi(const char *value, int lineno);
2595 static void sip_subscribe_mwi_destroy(struct sip_subscription_mwi *mwi);
2596 static void sip_send_all_mwi_subscriptions(void);
2597 static int sip_subscribe_mwi_do(const void *data);
2598 static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi);
2600 /*! \brief Definition of this channel for PBX channel registration */
2601 static const struct ast_channel_tech sip_tech = {
2603 .description = "Session Initiation Protocol (SIP)",
2604 .capabilities = AST_FORMAT_AUDIO_MASK, /* all audio formats */
2605 .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
2606 .requester = sip_request_call, /* called with chan unlocked */
2607 .devicestate = sip_devicestate, /* called with chan unlocked (not chan-specific) */
2608 .call = sip_call, /* called with chan locked */
2609 .send_html = sip_sendhtml,
2610 .hangup = sip_hangup, /* called with chan locked */
2611 .answer = sip_answer, /* called with chan locked */
2612 .read = sip_read, /* called with chan locked */
2613 .write = sip_write, /* called with chan locked */
2614 .write_video = sip_write, /* called with chan locked */
2615 .write_text = sip_write,
2616 .indicate = sip_indicate, /* called with chan locked */
2617 .transfer = sip_transfer, /* called with chan locked */
2618 .fixup = sip_fixup, /* called with chan locked */
2619 .send_digit_begin = sip_senddigit_begin, /* called with chan unlocked */
2620 .send_digit_end = sip_senddigit_end,
2621 .bridge = ast_rtp_bridge, /* XXX chan unlocked ? */
2622 .early_bridge = ast_rtp_early_bridge,
2623 .send_text = sip_sendtext, /* called with chan locked */
2624 .func_channel_read = acf_channel_read,
2625 .queryoption = sip_queryoption,
2626 .get_pvt_uniqueid = sip_get_callid,
2629 /*! \brief This version of the sip channel tech has no send_digit_begin
2630 * callback so that the core knows that the channel does not want
2631 * DTMF BEGIN frames.
2632 * The struct is initialized just before registering the channel driver,
2633 * and is for use with channels using SIP INFO DTMF.
2635 static struct ast_channel_tech sip_tech_info;
2638 /*! \brief Working TLS connection configuration */
2639 static struct ast_tls_config sip_tls_cfg;
2641 /*! \brief Default TLS connection configuration */
2642 static struct ast_tls_config default_tls_cfg;
2644 /*! \brief The TCP server definition */
2645 static struct ast_tcptls_session_args sip_tcp_desc = {
2647 .master = AST_PTHREADT_NULL,
2650 .name = "SIP TCP server",
2651 .accept_fn = ast_tcptls_server_root,
2652 .worker_fn = sip_tcp_worker_fn,
2655 /*! \brief The TCP/TLS server definition */
2656 static struct ast_tcptls_session_args sip_tls_desc = {
2658 .master = AST_PTHREADT_NULL,
2659 .tls_cfg = &sip_tls_cfg,
2661 .name = "SIP TLS server",
2662 .accept_fn = ast_tcptls_server_root,
2663 .worker_fn = sip_tcp_worker_fn,
2666 /* wrapper macro to tell whether t points to one of the sip_tech descriptors */
2667 #define IS_SIP_TECH(t) ((t) == &sip_tech || (t) == &sip_tech_info)
2669 /*! \brief map from an integer value to a string.
2670 * If no match is found, return errorstring
2672 static const char *map_x_s(const struct _map_x_s *table, int x, const char *errorstring)
2674 const struct _map_x_s *cur;
2676 for (cur = table; cur->s; cur++)
2682 /*! \brief map from a string to an integer value, case insensitive.
2683 * If no match is found, return errorvalue.
2685 static int map_s_x(const struct _map_x_s *table, const char *s, int errorvalue)
2687 const struct _map_x_s *cur;
2689 for (cur = table; cur->s; cur++)
2690 if (!strcasecmp(cur->s, s))
2696 /*! \brief Interface structure with callbacks used to connect to RTP module */
2697 static struct ast_rtp_protocol sip_rtp = {
2699 .get_rtp_info = sip_get_rtp_peer,
2700 .get_vrtp_info = sip_get_vrtp_peer,
2701 .get_trtp_info = sip_get_trtp_peer,
2702 .set_rtp_peer = sip_set_rtp_peer,
2703 .get_codec = sip_get_codec,
2707 * duplicate a list of channel variables, \return the copy.
2709 static struct ast_variable *copy_vars(struct ast_variable *src)
2711 struct ast_variable *res = NULL, *tmp, *v = NULL;
2713 for (v = src ; v ; v = v->next) {
2714 if ((tmp = ast_variable_new(v->name, v->value, v->file))) {
2722 /*! \brief SIP TCP connection handler */
2723 static void *sip_tcp_worker_fn(void *data)
2725 struct ast_tcptls_session_instance *tcptls_session = data;
2727 return _sip_tcp_helper_thread(NULL, tcptls_session);
2730 /*! \brief SIP TCP thread management function
2731 This function reads from the socket, parses the packet into a request
2733 static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *tcptls_session)
2736 struct sip_request req = { 0, } , reqcpy = { 0, };
2737 struct sip_threadinfo *me;
2738 char buf[1024] = "";
2740 me = ast_calloc(1, sizeof(*me));
2745 me->threadid = pthread_self();
2746 me->tcptls_session = tcptls_session;
2747 if (tcptls_session->ssl)
2748 me->type = SIP_TRANSPORT_TLS;
2750 me->type = SIP_TRANSPORT_TCP;
2752 ast_debug(2, "Starting thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");
2754 AST_LIST_LOCK(&threadl);
2755 AST_LIST_INSERT_TAIL(&threadl, me, list);
2756 AST_LIST_UNLOCK(&threadl);
2758 if (!(req.data = ast_str_create(SIP_MIN_PACKET)))
2760 if (!(reqcpy.data = ast_str_create(SIP_MIN_PACKET)))
2764 struct ast_str *str_save;
2766 str_save = req.data;
2767 memset(&req, 0, sizeof(req));
2768 req.data = str_save;
2769 ast_str_reset(req.data);
2771 str_save = reqcpy.data;
2772 memset(&reqcpy, 0, sizeof(reqcpy));
2773 reqcpy.data = str_save;
2774 ast_str_reset(reqcpy.data);
2776 req.socket.fd = tcptls_session->fd;
2777 if (tcptls_session->ssl) {
2778 req.socket.type = SIP_TRANSPORT_TLS;
2779 req.socket.port = htons(ourport_tls);
2781 req.socket.type = SIP_TRANSPORT_TCP;
2782 req.socket.port = htons(ourport_tcp);
2784 res = ast_wait_for_input(tcptls_session->fd, -1);
2786 ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", tcptls_session->ssl ? "SSL": "TCP", res);
2790 /* Read in headers one line at a time */
2791 while (req.len < 4 || strncmp(REQ_OFFSET_TO_STR(&req, len - 4), "\r\n\r\n", 4)) {
2792 ast_mutex_lock(&tcptls_session->lock);
2793 if (!fgets(buf, sizeof(buf), tcptls_session->f)) {
2794 ast_mutex_unlock(&tcptls_session->lock);
2797 ast_mutex_unlock(&tcptls_session->lock);
2800 ast_str_append(&req.data, 0, "%s", buf);
2801 req.len = req.data->used;
2803 copy_request(&reqcpy, &req);
2804 parse_request(&reqcpy);
2805 /* In order to know how much to read, we need the content-length header */
2806 if (sscanf(get_header(&reqcpy, "Content-Length"), "%d", &cl)) {
2808 ast_mutex_lock(&tcptls_session->lock);
2809 if (!fread(buf, (cl < sizeof(buf)) ? cl : sizeof(buf), 1, tcptls_session->f)) {
2810 ast_mutex_unlock(&tcptls_session->lock);
2813 ast_mutex_unlock(&tcptls_session->lock);
2817 ast_str_append(&req.data, 0, "%s", buf);
2818 req.len = req.data->used;
2821 /*! \todo XXX If there's no Content-Length or if the content-length and what
2822 we receive is not the same - we should generate an error */
2824 req.socket.tcptls_session = tcptls_session;
2825 handle_request_do(&req, &tcptls_session->remote_address);
2829 AST_LIST_LOCK(&threadl);
2830 AST_LIST_REMOVE(&threadl, me, list);
2831 AST_LIST_UNLOCK(&threadl);
2834 fclose(tcptls_session->f);
2835 tcptls_session->f = NULL;
2836 tcptls_session->fd = -1;
2838 ast_free(reqcpy.data);
2846 ast_debug(2, "Shutting down thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");
2849 ao2_ref(tcptls_session, -1);
2850 tcptls_session = NULL;
2857 * helper functions to unreference various types of objects.
2858 * By handling them this way, we don't have to declare the
2859 * destructor on each call, which removes the chance of errors.
2861 static void *unref_peer(struct sip_peer *peer, char *tag)
2863 ao2_t_ref(peer, -1, tag);
2867 static struct sip_peer *ref_peer(struct sip_peer *peer, char *tag)
2869 ao2_t_ref(peer, 1, tag);
2873 /*! \brief maintain proper refcounts for a sip_pvt's outboundproxy
2875 * This function sets pvt's outboundproxy pointer to the one referenced
2876 * by the proxy parameter. Because proxy may be a refcounted object, and
2877 * because pvt's old outboundproxy may also be a refcounted object, we need
2878 * to maintain the proper refcounts.
2880 * \param pvt The sip_pvt for which we wish to set the outboundproxy
2881 * \param proxy The sip_proxy which we will point pvt towards.
2882 * \return Returns void
2884 static void ref_proxy(struct sip_pvt *pvt, struct sip_proxy *proxy)
2886 struct sip_proxy *old_obproxy = pvt->outboundproxy;
2887 /* The sip_cfg.outboundproxy is statically allocated, and so
2888 * we don't ever need to adjust refcounts for it
2890 if (proxy && proxy != &sip_cfg.outboundproxy) {
2893 pvt->outboundproxy = proxy;
2894 if (old_obproxy && old_obproxy != &sip_cfg.outboundproxy) {
2895 ao2_ref(old_obproxy, -1);
2900 * \brief Unlink a dialog from the dialogs container, as well as any other places
2901 * that it may be currently stored.
2903 * \note A reference to the dialog must be held before calling this function, and this
2904 * function does not release that reference.
2906 static void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist)
2910 dialog_ref(dialog, "Let's bump the count in the unlink so it doesn't accidentally become dead before we are done");
2912 ao2_t_unlink(dialogs, dialog, "unlinking dialog via ao2_unlink");
2914 /* Unlink us from the owner (channel) if we have one */
2915 if (dialog->owner) {
2917 ast_channel_lock(dialog->owner);
2918 ast_debug(1, "Detaching from channel %s\n", dialog->owner->name);
2919 dialog->owner->tech_pvt = dialog_unref(dialog->owner->tech_pvt, "resetting channel dialog ptr in unlink_all");
2921 ast_channel_unlock(dialog->owner);
2923 if (dialog->registry) {
2924 if (dialog->registry->call == dialog)
2925 dialog->registry->call = dialog_unref(dialog->registry->call, "nulling out the registry's call dialog field in unlink_all");
2926 dialog->registry = registry_unref(dialog->registry, "delete dialog->registry");
2928 if (dialog->stateid > -1) {
2929 ast_extension_state_del(dialog->stateid, NULL);
2930 dialog_unref(dialog, "removing extension_state, should unref the associated dialog ptr that was stored there.");
2931 dialog->stateid = -1; /* shouldn't we 'zero' this out? */
2933 /* Remove link from peer to subscription of MWI */
2934 if (dialog->relatedpeer && dialog->relatedpeer->mwipvt == dialog)
2935 dialog->relatedpeer->mwipvt = dialog_unref(dialog->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
2936 if (dialog->relatedpeer && dialog->relatedpeer->call == dialog)
2937 dialog->relatedpeer->call = dialog_unref(dialog->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
2939 /* remove all current packets in this dialog */
2940 while((cp = dialog->packets)) {
2941 dialog->packets = dialog->packets->next;