Store SIP User-Agent information in contacts.
[asterisk/asterisk.git] / res / res_pjsip.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2013, Digium, Inc.
5  *
6  * Mark Michelson <mmichelson@digium.com>
7  *
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.
13  *
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.
17  */
18
19 #include "asterisk.h"
20
21 #include <pjsip.h>
22 /* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
23 #include <pjsip_simple.h>
24 #include <pjlib.h>
25
26 #include "asterisk/res_pjsip.h"
27 #include "res_pjsip/include/res_pjsip_private.h"
28 #include "asterisk/linkedlists.h"
29 #include "asterisk/logger.h"
30 #include "asterisk/lock.h"
31 #include "asterisk/utils.h"
32 #include "asterisk/astobj2.h"
33 #include "asterisk/module.h"
34 #include "asterisk/threadpool.h"
35 #include "asterisk/taskprocessor.h"
36 #include "asterisk/uuid.h"
37 #include "asterisk/sorcery.h"
38
39 /*** MODULEINFO
40         <depend>pjproject</depend>
41         <depend>res_sorcery_config</depend>
42         <support_level>core</support_level>
43  ***/
44
45 /*** DOCUMENTATION
46         <configInfo name="res_pjsip" language="en_US">
47                 <synopsis>SIP Resource using PJProject</synopsis>
48                 <configFile name="pjsip.conf">
49                         <configObject name="endpoint">
50                                 <synopsis>Endpoint</synopsis>
51                                 <description><para>
52                                         The <emphasis>Endpoint</emphasis> is the primary configuration object.
53                                         It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
54                                         dialable entries of their own. Communication with another SIP device is
55                                         accomplished via Addresses of Record (AoRs) which have one or more
56                                         contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
57                                         use a <literal>transport</literal> will default to first transport found
58                                         in <filename>pjsip.conf</filename> that matches its type.
59                                         </para>
60                                         <para>Example: An Endpoint has been configured with no transport.
61                                         When it comes time to call an AoR, PJSIP will find the
62                                         first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
63                                         will use the first IPv6 transport and try to send the request.
64                                         </para>
65                                         <para>If the anonymous endpoint identifier is in use an endpoint with the name
66                                         "anonymous@domain" will be searched for as a last resort. If this is not found
67                                         it will fall back to searching for "anonymous". If neither endpoints are found
68                                         the anonymous endpoint identifier will not return an endpoint and anonymous
69                                         calling will not be possible.
70                                         </para>
71                                 </description>
72                                 <configOption name="100rel" default="yes">
73                                         <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
74                                         <description>
75                                                 <enumlist>
76                                                         <enum name="no" />
77                                                         <enum name="required" />
78                                                         <enum name="yes" />
79                                                 </enumlist>
80                                         </description>
81                                 </configOption>
82                                 <configOption name="aggregate_mwi" default="yes">
83                                         <synopsis></synopsis>
84                                         <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
85                                         waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
86                                         individual NOTIFYs are sent for each mailbox.</para></description>
87                                 </configOption>
88                                 <configOption name="allow">
89                                         <synopsis>Media Codec(s) to allow</synopsis>
90                                 </configOption>
91                                 <configOption name="aors">
92                                         <synopsis>AoR(s) to be used with the endpoint</synopsis>
93                                         <description><para>
94                                                 List of comma separated AoRs that the endpoint should be associated with.
95                                         </para></description>
96                                 </configOption>
97                                 <configOption name="auth">
98                                         <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
99                                         <description><para>
100                                                 This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
101                                                 in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
102                                                 </para><para>
103                                                 Endpoints without an <literal>authentication</literal> object
104                                                 configured will allow connections without vertification.
105                                         </para></description>
106                                 </configOption>
107                                 <configOption name="callerid">
108                                         <synopsis>CallerID information for the endpoint</synopsis>
109                                         <description><para>
110                                                 Must be in the format <literal>Name &lt;Number&gt;</literal>,
111                                                 or only <literal>&lt;Number&gt;</literal>.
112                                         </para></description>
113                                 </configOption>
114                                 <configOption name="callerid_privacy">
115                                         <synopsis>Default privacy level</synopsis>
116                                         <description>
117                                                 <enumlist>
118                                                         <enum name="allowed_not_screened" />
119                                                         <enum name="allowed_passed_screened" />
120                                                         <enum name="allowed_failed_screened" />
121                                                         <enum name="allowed" />
122                                                         <enum name="prohib_not_screened" />
123                                                         <enum name="prohib_passed_screened" />
124                                                         <enum name="prohib_failed_screened" />
125                                                         <enum name="prohib" />
126                                                         <enum name="unavailable" />
127                                                 </enumlist>
128                                         </description>
129                                 </configOption>
130                                 <configOption name="callerid_tag">
131                                         <synopsis>Internal id_tag for the endpoint</synopsis>
132                                 </configOption>
133                                 <configOption name="context">
134                                         <synopsis>Dialplan context for inbound sessions</synopsis>
135                                 </configOption>
136                                 <configOption name="direct_media_glare_mitigation" default="none">
137                                         <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
138                                         <description>
139                                                 <para>
140                                                 This setting attempts to avoid creating INVITE glare scenarios
141                                                 by disabling direct media reINVITEs in one direction thereby allowing
142                                                 designated servers (according to this option) to initiate direct
143                                                 media reINVITEs without contention and significantly reducing call
144                                                 setup time.
145                                                 </para>
146                                                 <para>
147                                                 A more detailed description of how this option functions can be found on
148                                                 the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
149                                                 </para>
150                                                 <enumlist>
151                                                         <enum name="none" />
152                                                         <enum name="outgoing" />
153                                                         <enum name="incoming" />
154                                                 </enumlist>
155                                         </description>
156                                 </configOption>
157                                 <configOption name="direct_media_method" default="invite">
158                                         <synopsis>Direct Media method type</synopsis>
159                                         <description>
160                                                 <para>Method for setting up Direct Media between endpoints.</para>
161                                                 <enumlist>
162                                                         <enum name="invite" />
163                                                         <enum name="reinvite">
164                                                                 <para>Alias for the <literal>invite</literal> value.</para>
165                                                         </enum>
166                                                         <enum name="update" />
167                                                 </enumlist>
168                                         </description>
169                                 </configOption>
170                                 <configOption name="connected_line_method" default="invite">
171                                         <synopsis>Connected line method type</synopsis>
172                                         <description>
173                                                 <para>Method used when updating connected line information.</para>
174                                                 <enumlist>
175                                                         <enum name="invite" />
176                                                         <enum name="reinvite">
177                                                                 <para>Alias for the <literal>invite</literal> value.</para>
178                                                         </enum>
179                                                         <enum name="update" />
180                                                 </enumlist>
181                                         </description>
182                                 </configOption>
183                                 <configOption name="direct_media" default="yes">
184                                         <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
185                                 </configOption>
186                                 <configOption name="disable_direct_media_on_nat" default="no">
187                                         <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
188                                 </configOption>
189                                 <configOption name="disallow">
190                                         <synopsis>Media Codec(s) to disallow</synopsis>
191                                 </configOption>
192                                 <configOption name="dtmf_mode" default="rfc4733">
193                                         <synopsis>DTMF mode</synopsis>
194                                         <description>
195                                                 <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
196                                                 <enumlist>
197                                                         <enum name="rfc4733">
198                                                                 <para>DTMF is sent out of band of the main audio stream.This
199                                                                 supercedes the older <emphasis>RFC-2833</emphasis> used within
200                                                                 the older <literal>chan_sip</literal>.</para>
201                                                         </enum>
202                                                         <enum name="inband">
203                                                                 <para>DTMF is sent as part of audio stream.</para>
204                                                         </enum>
205                                                         <enum name="info">
206                                                                 <para>DTMF is sent as SIP INFO packets.</para>
207                                                         </enum>
208                                                 </enumlist>
209                                         </description>
210                                 </configOption>
211                                 <configOption name="media_address">
212                                         <synopsis>IP address used in SDP for media handling</synopsis>
213                                         <description><para>
214                                                 At the time of SDP creation, the IP address defined here will be used as
215                                                 the media address for individual streams in the SDP.
216                                         </para>
217                                         <note><para>
218                                                 Be aware that the <literal>external_media_address</literal> option, set in Transport
219                                                 configuration, can also affect the final media address used in the SDP.
220                                         </para></note>
221                                         </description>
222                                 </configOption>
223                                 <configOption name="force_rport" default="yes">
224                                         <synopsis>Force use of return port</synopsis>
225                                 </configOption>
226                                 <configOption name="ice_support" default="no">
227                                         <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
228                                 </configOption>
229                                 <configOption name="identify_by" default="username,location">
230                                         <synopsis>Way(s) for Endpoint to be identified</synopsis>
231                                         <description><para>
232                                                 An endpoint can be identified in multiple ways. Currently, the only supported
233                                                 option is <literal>username</literal>, which matches the endpoint based on the
234                                                 username in the From header.
235                                                 </para>
236                                                 <note><para>Endpoints can also be identified by IP address; however, that method
237                                                 of identification is not handled by this configuration option. See the documentation
238                                                 for the <literal>identify</literal> configuration section for more details on that
239                                                 method of endpoint identification. If this option is set to <literal>username</literal>
240                                                 and an <literal>identify</literal> configuration section exists for the endpoint, then
241                                                 the endpoint can be identified in multiple ways.</para></note>
242                                                 <enumlist>
243                                                         <enum name="username" />
244                                                 </enumlist>
245                                         </description>
246                                 </configOption>
247                                 <configOption name="redirect_method">
248                                         <synopsis>How redirects received from an endpoint are handled</synopsis>
249                                         <description><para>
250                                                 When a redirect is received from an endpoint there are multiple ways it can be handled.
251                                                 If this option is set to <literal>user</literal> the user portion of the redirect target
252                                                 is treated as an extension within the dialplan and dialed using a Local channel. If this option
253                                                 is set to <literal>uri_core</literal> the target URI is returned to the dialing application
254                                                 which dials it using the PJSIP channel driver and endpoint originally used. If this option is
255                                                 set to <literal>uri_pjsip</literal> the redirect occurs within chan_pjsip itself and is not exposed
256                                                 to the core at all. The <literal>uri_pjsip</literal> option has the benefit of being more efficient
257                                                 and also supporting multiple potential redirect targets. The con is that since redirection occurs
258                                                 within chan_pjsip redirecting information is not forwarded and redirection can not be
259                                                 prevented.
260                                                 </para>
261                                                 <enumlist>
262                                                         <enum name="user" />
263                                                         <enum name="uri_core" />
264                                                         <enum name="uri_pjsip" />
265                                                 </enumlist>
266                                         </description>
267                                 </configOption>
268                                 <configOption name="mailboxes">
269                                         <synopsis>NOTIFY the endpoint when state changes for any of the specified mailboxes</synopsis>
270                                         <description><para>
271                                                 Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state
272                                                 changes happen for any of the specified mailboxes. More than one mailbox can be
273                                                 specified with a comma-delimited string. app_voicemail mailboxes must be specified
274                                                 as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by
275                                                 external sources, such as through the res_external_mwi module, you must specify
276                                                 strings supported by the external system.
277                                         </para><para>
278                                                 For endpoints that SUBSCRIBE for MWI, use the <literal>mailboxes</literal> option in your AOR
279                                                 configuration.
280                                         </para></description>
281                                 </configOption>
282                                 <configOption name="moh_suggest" default="default">
283                                         <synopsis>Default Music On Hold class</synopsis>
284                                 </configOption>
285                                 <configOption name="outbound_auth">
286                                         <synopsis>Authentication object used for outbound requests</synopsis>
287                                 </configOption>
288                                 <configOption name="outbound_proxy">
289                                         <synopsis>Proxy through which to send requests, a full SIP URI must be provided</synopsis>
290                                 </configOption>
291                                 <configOption name="rewrite_contact">
292                                         <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
293                                         <description><para>
294                                                 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
295                                                 source IP address and port. This option does not affect outbound messages send to this
296                                                 endpoint.
297                                         </para></description>
298                                 </configOption>
299                                 <configOption name="rtp_ipv6" default="no">
300                                         <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
301                                 </configOption>
302                                 <configOption name="rtp_symmetric" default="no">
303                                         <synopsis>Enforce that RTP must be symmetric</synopsis>
304                                 </configOption>
305                                 <configOption name="send_diversion" default="yes">
306                                         <synopsis>Send the Diversion header, conveying the diversion
307                                         information to the called user agent</synopsis>
308                                 </configOption>
309                                 <configOption name="send_pai" default="no">
310                                         <synopsis>Send the P-Asserted-Identity header</synopsis>
311                                 </configOption>
312                                 <configOption name="send_rpid" default="no">
313                                         <synopsis>Send the Remote-Party-ID header</synopsis>
314                                 </configOption>
315                                 <configOption name="timers_min_se" default="90">
316                                         <synopsis>Minimum session timers expiration period</synopsis>
317                                         <description><para>
318                                                 Minimium session timer expiration period. Time in seconds.
319                                         </para></description>
320                                 </configOption>
321                                 <configOption name="timers" default="yes">
322                                         <synopsis>Session timers for SIP packets</synopsis>
323                                         <description>
324                                                 <enumlist>
325                                                         <enum name="forced" />
326                                                         <enum name="no" />
327                                                         <enum name="required" />
328                                                         <enum name="yes" />
329                                                 </enumlist>
330                                         </description>
331                                 </configOption>
332                                 <configOption name="timers_sess_expires" default="1800">
333                                         <synopsis>Maximum session timer expiration period</synopsis>
334                                         <description><para>
335                                                 Maximium session timer expiration period. Time in seconds.
336                                         </para></description>
337                                 </configOption>
338                                 <configOption name="transport">
339                                         <synopsis>Desired transport configuration</synopsis>
340                                         <description><para>
341                                                 This will set the desired transport configuration to send SIP data through.
342                                                 </para>
343                                                 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
344                                                 to the first configured transport in <filename>pjsip.conf</filename> which is
345                                                 valid for the URI we are trying to contact.
346                                                 </para></warning>
347                                                 <warning><para>Transport configuration is not affected by reloads. In order to
348                                                 change transports, a full Asterisk restart is required</para></warning>
349                                         </description>
350                                 </configOption>
351                                 <configOption name="trust_id_inbound" default="no">
352                                         <synopsis>Accept identification information received from this endpoint</synopsis>
353                                         <description><para>This option determines whether Asterisk will accept
354                                         identification from the endpoint from headers such as P-Asserted-Identity
355                                         or Remote-Party-ID header. This option applies both to calls originating from the
356                                         endpoint and calls originating from Asterisk. If <literal>no</literal>, the
357                                         configured Caller-ID from pjsip.conf will always be used as the identity for
358                                         the endpoint.</para></description>
359                                 </configOption>
360                                 <configOption name="trust_id_outbound" default="no">
361                                         <synopsis>Send private identification details to the endpoint.</synopsis>
362                                         <description><para>This option determines whether res_pjsip will send private
363                                         identification information to the endpoint. If <literal>no</literal>,
364                                         private Caller-ID information will not be forwarded to the endpoint.
365                                         "Private" in this case refers to any method of restricting identification.
366                                         Example: setting <replaceable>callerid_privacy</replaceable> to any
367                                         <literal>prohib</literal> variation.
368                                         Example: If <replaceable>trust_id_inbound</replaceable> is set to
369                                         <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
370                                         header in a SIP request or response would indicate the identification
371                                         provided in the request is private.</para></description>
372                                 </configOption>
373                                 <configOption name="type">
374                                         <synopsis>Must be of type 'endpoint'.</synopsis>
375                                 </configOption>
376                                 <configOption name="use_ptime" default="no">
377                                         <synopsis>Use Endpoint's requested packetisation interval</synopsis>
378                                 </configOption>
379                                 <configOption name="use_avpf" default="no">
380                                         <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
381                                         endpoint.</synopsis>
382                                         <description><para>
383                                                 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
384                                                 profile for all media offers on outbound calls and media updates and will
385                                                 decline media offers not using the AVPF or SAVPF profile.
386                                         </para><para>
387                                                 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
388                                                 profile for all media offers on outbound calls and media updates and will
389                                                 decline media offers not using the AVP or SAVP profile.
390                                         </para></description>
391                                 </configOption>
392                                 <configOption name="media_encryption" default="no">
393                                         <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
394                                         for this endpoint.</synopsis>
395                                         <description>
396                                                 <enumlist>
397                                                         <enum name="no"><para>
398                                                                 res_pjsip will offer no encryption and allow no encryption to be setup.
399                                                         </para></enum>
400                                                         <enum name="sdes"><para>
401                                                                 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
402                                                                 transport should be used in conjunction with this option to prevent
403                                                                 exposure of media encryption keys.
404                                                         </para></enum>
405                                                         <enum name="dtls"><para>
406                                                                 res_pjsip will offer DTLS-SRTP setup.
407                                                         </para></enum>
408                                                 </enumlist>
409                                         </description>
410                                 </configOption>
411                                 <configOption name="inband_progress" default="no">
412                                         <synopsis>Determines whether chan_pjsip will indicate ringing using inband
413                                             progress.</synopsis>
414                                         <description><para>
415                                                 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
416                                                 when told to indicate ringing and will immediately start sending ringing
417                                                 as audio.
418                                         </para><para>
419                                                 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
420                                                 to indicate ringing and will NOT send it as audio.
421                                         </para></description>
422                                 </configOption>
423                                 <configOption name="call_group">
424                                         <synopsis>The numeric pickup groups for a channel.</synopsis>
425                                         <description><para>
426                                                 Can be set to a comma separated list of numbers or ranges between the values
427                                                 of 0-63 (maximum of 64 groups).
428                                         </para></description>
429                                 </configOption>
430                                 <configOption name="pickup_group">
431                                         <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
432                                         <description><para>
433                                                 Can be set to a comma separated list of numbers or ranges between the values
434                                                 of 0-63 (maximum of 64 groups).
435                                         </para></description>
436                                 </configOption>
437                                 <configOption name="named_call_group">
438                                         <synopsis>The named pickup groups for a channel.</synopsis>
439                                         <description><para>
440                                                 Can be set to a comma separated list of case sensitive strings limited by
441                                                 supported line length.
442                                         </para></description>
443                                 </configOption>
444                                 <configOption name="named_pickup_group">
445                                         <synopsis>The named pickup groups that a channel can pickup.</synopsis>
446                                         <description><para>
447                                                 Can be set to a comma separated list of case sensitive strings limited by
448                                                 supported line length.
449                                         </para></description>
450                                 </configOption>
451                                 <configOption name="device_state_busy_at" default="0">
452                                         <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
453                                         <description><para>
454                                                 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
455                                                 PJSIP channel driver will return busy as the device state instead of in use.
456                                         </para></description>
457                                 </configOption>
458                                 <configOption name="t38_udptl" default="no">
459                                         <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
460                                         <description><para>
461                                                 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
462                                                 and relayed.
463                                         </para></description>
464                                 </configOption>
465                                 <configOption name="t38_udptl_ec" default="none">
466                                         <synopsis>T.38 UDPTL error correction method</synopsis>
467                                         <description>
468                                                 <enumlist>
469                                                         <enum name="none"><para>
470                                                                 No error correction should be used.
471                                                         </para></enum>
472                                                         <enum name="fec"><para>
473                                                                 Forward error correction should be used.
474                                                         </para></enum>
475                                                         <enum name="redundancy"><para>
476                                                                 Redundacy error correction should be used.
477                                                         </para></enum>
478                                                 </enumlist>
479                                         </description>
480                                 </configOption>
481                                 <configOption name="t38_udptl_maxdatagram" default="0">
482                                         <synopsis>T.38 UDPTL maximum datagram size</synopsis>
483                                         <description><para>
484                                                 This option can be set to override the maximum datagram of a remote endpoint for broken
485                                                 endpoints.
486                                         </para></description>
487                                 </configOption>
488                                 <configOption name="fax_detect" default="no">
489                                         <synopsis>Whether CNG tone detection is enabled</synopsis>
490                                         <description><para>
491                                                 This option can be set to send the session to the fax extension when a CNG tone is
492                                                 detected.
493                                         </para></description>
494                                 </configOption>
495                                 <configOption name="t38_udptl_nat" default="no">
496                                         <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
497                                         <description><para>
498                                                 When enabled the UDPTL stack will send UDPTL packets to the source address of
499                                                 received packets.
500                                         </para></description>
501                                 </configOption>
502                                 <configOption name="t38_udptl_ipv6" default="no">
503                                         <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
504                                         <description><para>
505                                                 When enabled the UDPTL stack will use IPv6.
506                                         </para></description>
507                                 </configOption>
508                                 <configOption name="tone_zone">
509                                         <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
510                                 </configOption>
511                                 <configOption name="language">
512                                         <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
513                                 </configOption>
514                                 <configOption name="one_touch_recording" default="no">
515                                         <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
516                                         <see-also>
517                                                 <ref type="configOption">recordonfeature</ref>
518                                                 <ref type="configOption">recordofffeature</ref>
519                                         </see-also>
520                                 </configOption>
521                                 <configOption name="record_on_feature" default="automixmon">
522                                         <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
523                                         <description>
524                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
525                                                 feature will be enabled for the channel. The feature designated here can be any built-in
526                                                 or dynamic feature defined in features.conf.</para>
527                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
528                                         </description>
529                                         <see-also>
530                                                 <ref type="configOption">one_touch_recording</ref>
531                                                 <ref type="configOption">recordofffeature</ref>
532                                         </see-also>
533                                 </configOption>
534                                 <configOption name="record_off_feature" default="automixmon">
535                                         <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
536                                         <description>
537                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
538                                                 feature will be enabled for the channel. The feature designated here can be any built-in
539                                                 or dynamic feature defined in features.conf.</para>
540                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
541                                         </description>
542                                         <see-also>
543                                                 <ref type="configOption">one_touch_recording</ref>
544                                                 <ref type="configOption">recordonfeature</ref>
545                                         </see-also>
546                                 </configOption>
547                                 <configOption name="rtp_engine" default="asterisk">
548                                         <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
549                                 </configOption>
550                                 <configOption name="allow_transfer" default="yes">
551                                         <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
552                                 </configOption>
553                                 <configOption name="sdp_owner" default="-">
554                                         <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
555                                 </configOption>
556                                 <configOption name="sdp_session" default="Asterisk">
557                                         <synopsis>String used for the SDP session (s=) line.</synopsis>
558                                 </configOption>
559                                 <configOption name="tos_audio">
560                                         <synopsis>DSCP TOS bits for audio streams</synopsis>
561                                         <description><para>
562                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
563                                         </para></description>
564                                 </configOption>
565                                 <configOption name="tos_video">
566                                         <synopsis>DSCP TOS bits for video streams</synopsis>
567                                         <description><para>
568                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
569                                         </para></description>
570                                 </configOption>
571                                 <configOption name="cos_audio">
572                                         <synopsis>Priority for audio streams</synopsis>
573                                         <description><para>
574                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
575                                         </para></description>
576                                 </configOption>
577                                 <configOption name="cos_video">
578                                         <synopsis>Priority for video streams</synopsis>
579                                         <description><para>
580                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
581                                         </para></description>
582                                 </configOption>
583                                 <configOption name="allow_subscribe" default="yes">
584                                         <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
585                                 </configOption>
586                                 <configOption name="sub_min_expiry" default="60">
587                                         <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
588                                 </configOption>
589                                 <configOption name="from_user">
590                                         <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
591                                 </configOption>
592                                 <configOption name="mwi_from_user">
593                                         <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
594                                 </configOption>
595                                 <configOption name="from_domain">
596                                         <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
597                                 </configOption>
598                                 <configOption name="dtls_verify">
599                                         <synopsis>Verify that the provided peer certificate is valid</synopsis>
600                                         <description><para>
601                                                 This option only applies if <replaceable>media_encryption</replaceable> is
602                                                 set to <literal>dtls</literal>.
603                                         </para></description>
604                                 </configOption>
605                                 <configOption name="dtls_rekey">
606                                         <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
607                                         <description><para>
608                                                 This option only applies if <replaceable>media_encryption</replaceable> is
609                                                 set to <literal>dtls</literal>.
610                                         </para><para>
611                                                 If this is not set or the value provided is 0 rekeying will be disabled.
612                                         </para></description>
613                                 </configOption>
614                                 <configOption name="dtls_cert_file">
615                                         <synopsis>Path to certificate file to present to peer</synopsis>
616                                         <description><para>
617                                                 This option only applies if <replaceable>media_encryption</replaceable> is
618                                                 set to <literal>dtls</literal>.
619                                         </para></description>
620                                 </configOption>
621                                 <configOption name="dtls_private_key">
622                                         <synopsis>Path to private key for certificate file</synopsis>
623                                         <description><para>
624                                                 This option only applies if <replaceable>media_encryption</replaceable> is
625                                                 set to <literal>dtls</literal>.
626                                         </para></description>
627                                 </configOption>
628                                 <configOption name="dtls_cipher">
629                                         <synopsis>Cipher to use for DTLS negotiation</synopsis>
630                                         <description><para>
631                                                 This option only applies if <replaceable>media_encryption</replaceable> is
632                                                 set to <literal>dtls</literal>.
633                                         </para><para>
634                                                 Many options for acceptable ciphers. See link for more:
635                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
636                                         </para></description>
637                                 </configOption>
638                                 <configOption name="dtls_ca_file">
639                                         <synopsis>Path to certificate authority certificate</synopsis>
640                                         <description><para>
641                                                 This option only applies if <replaceable>media_encryption</replaceable> is
642                                                 set to <literal>dtls</literal>.
643                                         </para></description>
644                                 </configOption>
645                                 <configOption name="dtls_ca_path">
646                                         <synopsis>Path to a directory containing certificate authority certificates</synopsis>
647                                         <description><para>
648                                                 This option only applies if <replaceable>media_encryption</replaceable> is
649                                                 set to <literal>dtls</literal>.
650                                         </para></description>
651                                 </configOption>
652                                 <configOption name="dtls_setup">
653                                         <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
654                                         <description>
655                                                 <para>
656                                                         This option only applies if <replaceable>media_encryption</replaceable> is
657                                                         set to <literal>dtls</literal>.
658                                                 </para>
659                                                 <enumlist>
660                                                         <enum name="active"><para>
661                                                                 res_pjsip will make a connection to the peer.
662                                                         </para></enum>
663                                                         <enum name="passive"><para>
664                                                                 res_pjsip will accept connections from the peer.
665                                                         </para></enum>
666                                                         <enum name="actpass"><para>
667                                                                 res_pjsip will offer and accept connections from the peer.
668                                                         </para></enum>
669                                                 </enumlist>
670                                         </description>
671                                 </configOption>
672                                 <configOption name="srtp_tag_32">
673                                         <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
674                                         <description><para>
675                                                 This option only applies if <replaceable>media_encryption</replaceable> is
676                                                 set to <literal>sdes</literal> or <literal>dtls</literal>.
677                                         </para></description>
678                                 </configOption>
679                                 <configOption name="set_var">
680                                         <synopsis>Variable set on a channel involving the endpoint.</synopsis>
681                                         <description><para>
682                                                 When a new channel is created using the endpoint set the specified
683                                                 variable(s) on that channel. For multiple channel variables specify
684                                                 multiple 'set_var'(s).
685                                         </para></description>
686                                 </configOption>
687                         </configObject>
688                         <configObject name="auth">
689                                 <synopsis>Authentication type</synopsis>
690                                 <description><para>
691                                         Authentication objects hold the authentication information for use
692                                         by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
693                                         This also allows for multiple objects to use a single auth object. See
694                                         the <literal>auth_type</literal> config option for password style choices.
695                                 </para></description>
696                                 <configOption name="auth_type" default="userpass">
697                                         <synopsis>Authentication type</synopsis>
698                                         <description><para>
699                                                 This option specifies which of the password style config options should be read
700                                                 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
701                                                 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
702                                                 from 'md5_cred'.
703                                                 </para>
704                                                 <enumlist>
705                                                         <enum name="md5"/>
706                                                         <enum name="userpass"/>
707                                                 </enumlist>
708                                         </description>
709                                 </configOption>
710                                 <configOption name="nonce_lifetime" default="32">
711                                         <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
712                                 </configOption>
713                                 <configOption name="md5_cred">
714                                         <synopsis>MD5 Hash used for authentication.</synopsis>
715                                         <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
716                                 </configOption>
717                                 <configOption name="password">
718                                         <synopsis>PlainText password used for authentication.</synopsis>
719                                         <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
720                                 </configOption>
721                                 <configOption name="realm" default="asterisk">
722                                         <synopsis>SIP realm for endpoint</synopsis>
723                                 </configOption>
724                                 <configOption name="type">
725                                         <synopsis>Must be 'auth'</synopsis>
726                                 </configOption>
727                                 <configOption name="username">
728                                         <synopsis>Username to use for account</synopsis>
729                                 </configOption>
730                         </configObject>
731                         <configObject name="domain_alias">
732                                 <synopsis>Domain Alias</synopsis>
733                                 <description><para>
734                                         Signifies that a domain is an alias. If the domain on a session is
735                                         not found to match an AoR then this object is used to see if we have
736                                         an alias for the AoR to which the endpoint is binding. This objects
737                                         name as defined in configuration should be the domain alias and a
738                                         config option is provided to specify the domain to be aliased.
739                                 </para></description>
740                                 <configOption name="type">
741                                         <synopsis>Must be of type 'domain_alias'.</synopsis>
742                                 </configOption>
743                                 <configOption name="domain">
744                                         <synopsis>Domain to be aliased</synopsis>
745                                 </configOption>
746                         </configObject>
747                         <configObject name="transport">
748                                 <synopsis>SIP Transport</synopsis>
749                                 <description><para>
750                                         <emphasis>Transports</emphasis>
751                                         </para>
752                                         <para>There are different transports and protocol derivatives
753                                                 supported by <literal>res_pjsip</literal>. They are in order of
754                                                 preference: UDP, TCP, and WebSocket (WS).</para>
755                                         <note><para>Changes to transport configuration in pjsip.conf will only be
756                                                 effected on a complete restart of Asterisk. A module reload
757                                                 will not suffice.</para></note>
758                                 </description>
759                                 <configOption name="async_operations" default="1">
760                                         <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
761                                 </configOption>
762                                 <configOption name="bind">
763                                         <synopsis>IP Address and optional port to bind to for this transport</synopsis>
764                                 </configOption>
765                                 <configOption name="ca_list_file">
766                                         <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
767                                 </configOption>
768                                 <configOption name="cert_file">
769                                         <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
770                                 </configOption>
771                                 <configOption name="cipher">
772                                         <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
773                                         <description><para>
774                                                 Many options for acceptable ciphers see link for more:
775                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
776                                         </para></description>
777                                 </configOption>
778                                 <configOption name="domain">
779                                         <synopsis>Domain the transport comes from</synopsis>
780                                 </configOption>
781                                 <configOption name="external_media_address">
782                                         <synopsis>External IP address to use in RTP handling</synopsis>
783                                         <description><para>
784                                                 When a request or response is sent out, if the destination of the
785                                                 message is outside the IP network defined in the option <literal>localnet</literal>,
786                                                 and the media address in the SDP is within the localnet network, then the
787                                                 media address in the SDP will be rewritten to the value defined for
788                                                 <literal>external_media_address</literal>.
789                                         </para></description>
790                                 </configOption>
791                                 <configOption name="external_signaling_address">
792                                         <synopsis>External address for SIP signalling</synopsis>
793                                 </configOption>
794                                 <configOption name="external_signaling_port" default="0">
795                                         <synopsis>External port for SIP signalling</synopsis>
796                                 </configOption>
797                                 <configOption name="method">
798                                         <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
799                                         <description>
800                                                 <enumlist>
801                                                         <enum name="default" />
802                                                         <enum name="unspecified" />
803                                                         <enum name="tlsv1" />
804                                                         <enum name="sslv2" />
805                                                         <enum name="sslv3" />
806                                                         <enum name="sslv23" />
807                                                 </enumlist>
808                                         </description>
809                                 </configOption>
810                                 <configOption name="local_net">
811                                         <synopsis>Network to consider local (used for NAT purposes).</synopsis>
812                                         <description><para>This must be in CIDR or dotted decimal format with the IP
813                                         and mask separated with a slash ('/').</para></description>
814                                 </configOption>
815                                 <configOption name="password">
816                                         <synopsis>Password required for transport</synopsis>
817                                 </configOption>
818                                 <configOption name="priv_key_file">
819                                         <synopsis>Private key file (TLS ONLY)</synopsis>
820                                 </configOption>
821                                 <configOption name="protocol" default="udp">
822                                         <synopsis>Protocol to use for SIP traffic</synopsis>
823                                         <description>
824                                                 <enumlist>
825                                                         <enum name="udp" />
826                                                         <enum name="tcp" />
827                                                         <enum name="tls" />
828                                                         <enum name="ws" />
829                                                         <enum name="wss" />
830                                                 </enumlist>
831                                         </description>
832                                 </configOption>
833                                 <configOption name="require_client_cert" default="false">
834                                         <synopsis>Require client certificate (TLS ONLY)</synopsis>
835                                 </configOption>
836                                 <configOption name="type">
837                                         <synopsis>Must be of type 'transport'.</synopsis>
838                                 </configOption>
839                                 <configOption name="verify_client" default="false">
840                                         <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
841                                 </configOption>
842                                 <configOption name="verify_server" default="false">
843                                         <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
844                                 </configOption>
845                                 <configOption name="tos" default="false">
846                                         <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
847                                         <description>
848                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
849                                         for more information on this parameter.</para>
850                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
851                                         or the <replaceable>wss</replaceable> protocols.</para></note>
852                                         </description>
853                                 </configOption>
854                                 <configOption name="cos" default="false">
855                                         <synopsis>Enable COS for the signalling sent over this transport</synopsis>
856                                         <description>
857                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
858                                         for more information on this parameter.</para>
859                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
860                                         or the <replaceable>wss</replaceable> protocols.</para></note>
861                                         </description>
862                                 </configOption>
863                         </configObject>
864                         <configObject name="contact">
865                                 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
866                                 <description><para>
867                                         Contacts are a way to hide SIP URIs from the dialplan directly.
868                                         They are also used to make a group of contactable parties when
869                                         in use with <literal>AoR</literal> lists.
870                                 </para></description>
871                                 <configOption name="type">
872                                         <synopsis>Must be of type 'contact'.</synopsis>
873                                 </configOption>
874                                 <configOption name="uri">
875                                         <synopsis>SIP URI to contact peer</synopsis>
876                                 </configOption>
877                                 <configOption name="expiration_time">
878                                         <synopsis>Time to keep alive a contact</synopsis>
879                                         <description><para>
880                                                 Time to keep alive a contact. String style specification.
881                                         </para></description>
882                                 </configOption>
883                                 <configOption name="qualify_frequency" default="0">
884                                         <synopsis>Interval at which to qualify a contact</synopsis>
885                                         <description><para>
886                                                 Interval between attempts to qualify the contact for reachability.
887                                                 If <literal>0</literal> never qualify. Time in seconds.
888                                         </para></description>
889                                 </configOption>
890                                 <configOption name="outbound_proxy">
891                                         <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
892                                         <description><para>
893                                                 If set the provided URI will be used as the outbound proxy when an
894                                                 OPTIONS request is sent to a contact for qualify purposes.
895                                         </para></description>
896                                 </configOption>
897                                 <configOption name="path">
898                                         <synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
899                                 </configOption>
900                                 <configOption name="user_agent">
901                                         <synopsis>User-Agent header from registration.</synopsis>
902                                         <description><para>
903                                                 The User-Agent is automatically stored based on data present in incoming SIP
904                                                 REGISTER requests and is not intended to be configured manually.
905                                         </para></description>
906                                 </configOption>
907                         </configObject>
908                         <configObject name="aor">
909                                 <synopsis>The configuration for a location of an endpoint</synopsis>
910                                 <description><para>
911                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
912                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
913                                         Beyond that, an AoR has other uses within Asterisk, such as inbound
914                                         registration.
915                                         </para><para>
916                                         An <literal>AoR</literal> is a way to allow dialing a group
917                                         of <literal>Contacts</literal> that all use the same
918                                         <literal>endpoint</literal> for calls.
919                                         </para><para>
920                                         This can be used as another way of grouping a list of contacts to dial
921                                         rather than specifing them each directly when dialing via the dialplan.
922                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
923                                         </para><para>
924                                         Registrations: For Asterisk to match an inbound registration to an endpoint,
925                                         the AoR object name must match the user portion of the SIP URI in the "To:"
926                                         header of the inbound SIP registration. That will usually be equivalent
927                                         to the "user name" set in your hard or soft phones configuration.
928                                 </para></description>
929                                 <configOption name="contact">
930                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
931                                         <description><para>
932                                                 Contacts specified will be called whenever referenced
933                                                 by <literal>chan_pjsip</literal>.
934                                                 </para><para>
935                                                 Use a separate "contact=" entry for each contact required. Contacts
936                                                 are specified using a SIP URI.
937                                         </para></description>
938                                 </configOption>
939                                 <configOption name="default_expiration" default="3600">
940                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
941                                 </configOption>
942                                 <configOption name="mailboxes">
943                                         <synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
944                                         <description><para>This option applies when an external entity subscribes to an AoR
945                                                 for Message Waiting Indications. The mailboxes specified will be subscribed to.
946                                                 More than one mailbox can be specified with a comma-delimited string.
947                                                 app_voicemail mailboxes must be specified as mailbox@context;
948                                                 for example: mailboxes=6001@default. For mailboxes provided by external sources,
949                                                 such as through the res_external_mwi module, you must specify strings supported by
950                                                 the external system.
951                                         </para><para>
952                                                 For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
953                                                 endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
954                                         </para></description>
955                                 </configOption>
956                                 <configOption name="maximum_expiration" default="7200">
957                                         <synopsis>Maximum time to keep an AoR</synopsis>
958                                         <description><para>
959                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
960                                         </para></description>
961                                 </configOption>
962                                 <configOption name="max_contacts" default="0">
963                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
964                                         <description><para>
965                                                 Maximum number of contacts that can associate with this AoR. This value does
966                                                 not affect the number of contacts that can be added with the "contact" option.
967                                                 It only limits contacts added through external interaction, such as
968                                                 registration.
969                                                 </para>
970                                                 <note><para>This should be set to <literal>1</literal> and
971                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
972                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
973                                                 </para></note>
974                                         </description>
975                                 </configOption>
976                                 <configOption name="minimum_expiration" default="60">
977                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
978                                         <description><para>
979                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
980                                         </para></description>
981                                 </configOption>
982                                 <configOption name="remove_existing" default="no">
983                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
984                                         <description><para>
985                                                 On receiving a new registration to the AoR should it remove
986                                                 the existing contact that was registered against it?
987                                                 </para>
988                                                 <note><para>This should be set to <literal>yes</literal> and
989                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
990                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
991                                                 </para></note>
992                                         </description>
993                                 </configOption>
994                                 <configOption name="type">
995                                         <synopsis>Must be of type 'aor'.</synopsis>
996                                 </configOption>
997                                 <configOption name="qualify_frequency" default="0">
998                                         <synopsis>Interval at which to qualify an AoR</synopsis>
999                                         <description><para>
1000                                                 Interval between attempts to qualify the AoR for reachability.
1001                                                 If <literal>0</literal> never qualify. Time in seconds.
1002                                         </para></description>
1003                                 </configOption>
1004                                 <configOption name="authenticate_qualify" default="no">
1005                                         <synopsis>Authenticates a qualify request if needed</synopsis>
1006                                         <description><para>
1007                                                 If true and a qualify request receives a challenge or authenticate response
1008                                                 authentication is attempted before declaring the contact available.
1009                                         </para></description>
1010                                 </configOption>
1011                                 <configOption name="outbound_proxy">
1012                                         <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
1013                                         <description><para>
1014                                                 If set the provided URI will be used as the outbound proxy when an
1015                                                 OPTIONS request is sent to a contact for qualify purposes.
1016                                         </para></description>
1017                                 </configOption>
1018                                 <configOption name="support_path">
1019                                         <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
1020                                         <description><para>
1021                                                 When this option is enabled, the Path headers in register requests will be saved
1022                                                 and its contents will be used in Route headers for outbound out-of-dialog requests
1023                                                 and in Path headers for outbound 200 responses. Path support will also be indicated
1024                                                 in the Supported header.
1025                                         </para></description>
1026                                 </configOption>
1027                         </configObject>
1028                         <configObject name="system">
1029                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
1030                                 <description><para>
1031                                         The settings in this section are global. In addition to being global, the values will
1032                                         not be re-evaluated when a reload is performed. This is because the values must be set
1033                                         before the SIP stack is initialized. The only way to reset these values is to either
1034                                         restart Asterisk, or unload res_pjsip.so and then load it again.
1035                                 </para></description>
1036                                 <configOption name="timer_t1" default="500">
1037                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
1038                                         <description><para>
1039                                                 Timer T1 is the base for determining how long to wait before retransmitting
1040                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
1041                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
1042                                         </para></description>
1043                                 </configOption>
1044                                 <configOption name="timer_b" default="32000">
1045                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
1046                                         <description><para>
1047                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
1048                                                 request before terminating the transaction. It is recommended that this be set
1049                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
1050                                                 this timer, see RFC 3261, Section 17.1.1.1.
1051                                         </para></description>
1052                                 </configOption>
1053                                 <configOption name="compact_headers" default="no">
1054                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
1055                                 </configOption>
1056                                 <configOption name="threadpool_initial_size" default="0">
1057                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
1058                                 </configOption>
1059                                 <configOption name="threadpool_auto_increment" default="5">
1060                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1061                                 </configOption>
1062                                 <configOption name="threadpool_idle_timeout" default="60">
1063                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1064                                 </configOption>
1065                                 <configOption name="threadpool_max_size" default="0">
1066                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
1067                                         A value of 0 indicates no maximum.</synopsis>
1068                                 </configOption>
1069                                 <configOption name="type">
1070                                         <synopsis>Must be of type 'system'.</synopsis>
1071                                 </configOption>
1072                         </configObject>
1073                         <configObject name="global">
1074                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
1075                                 <description><para>
1076                                         The settings in this section are global. Unlike options in the <literal>system</literal>
1077                                         section, these options can be refreshed by performing a reload.
1078                                 </para></description>
1079                                 <configOption name="max_forwards" default="70">
1080                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1081                                 </configOption>
1082                                 <configOption name="type">
1083                                         <synopsis>Must be of type 'global'.</synopsis>
1084                                 </configOption>
1085                                 <configOption name="user_agent" default="Asterisk &lt;Asterisk Version&gt;">
1086                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1087                                 </configOption>
1088                                 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1089                                         <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1090                                 </configOption>
1091                                 <configOption name="debug" default="no">
1092                                         <synopsis>Enable/Disable SIP debug logging.  Valid options include yes|no or
1093                                         a host address</synopsis>
1094                                 </configOption>
1095                         </configObject>
1096                 </configFile>
1097         </configInfo>
1098         <manager name="PJSIPQualify" language="en_US">
1099                 <synopsis>
1100                         Qualify a chan_pjsip endpoint.
1101                 </synopsis>
1102                 <syntax>
1103                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1104                         <parameter name="Endpoint" required="true">
1105                                 <para>The endpoint you want to qualify.</para>
1106                         </parameter>
1107                 </syntax>
1108                 <description>
1109                         <para>Qualify a chan_pjsip endpoint.</para>
1110                 </description>
1111         </manager>
1112         <manager name="PJSIPShowEndpoints" language="en_US">
1113                 <synopsis>
1114                         Lists PJSIP endpoints.
1115                 </synopsis>
1116                 <syntax />
1117                 <description>
1118                         <para>
1119                         Provides a listing of all endpoints.  For each endpoint an <literal>EndpointList</literal> event
1120                         is raised that contains relevant attributes and status information.  Once all
1121                         endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1122                         </para>
1123                 </description>
1124         </manager>
1125         <manager name="PJSIPShowEndpoint" language="en_US">
1126                 <synopsis>
1127                         Detail listing of an endpoint and its objects.
1128                 </synopsis>
1129                 <syntax>
1130                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1131                         <parameter name="Endpoint" required="true">
1132                                 <para>The endpoint to list.</para>
1133                         </parameter>
1134                 </syntax>
1135                 <description>
1136                         <para>
1137                         Provides a detailed listing of options for a given endpoint.  Events are issued
1138                         showing the configuration and status of the endpoint and associated objects.  These
1139                         events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1140                         <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1141                         <literal>IdentifyDetail</literal>.  Some events may be listed multiple times if multiple objects are
1142                         associated (for instance AoRs).  Once all detail events have been raised a final
1143                         <literal>EndpointDetailComplete</literal> event is issued.
1144                         </para>
1145                 </description>
1146         </manager>
1147  ***/
1148
1149 #define MOD_DATA_CONTACT "contact"
1150
1151 static pjsip_endpoint *ast_pjsip_endpoint;
1152
1153 static struct ast_threadpool *sip_threadpool;
1154
1155 static int register_service(void *data)
1156 {
1157         pjsip_module **module = data;
1158         if (!ast_pjsip_endpoint) {
1159                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1160                 return -1;
1161         }
1162         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1163                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1164                 return -1;
1165         }
1166         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1167         ast_module_ref(ast_module_info->self);
1168         return 0;
1169 }
1170
1171 int ast_sip_register_service(pjsip_module *module)
1172 {
1173         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1174 }
1175
1176 static int unregister_service(void *data)
1177 {
1178         pjsip_module **module = data;
1179         ast_module_unref(ast_module_info->self);
1180         if (!ast_pjsip_endpoint) {
1181                 return -1;
1182         }
1183         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1184         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1185         return 0;
1186 }
1187
1188 void ast_sip_unregister_service(pjsip_module *module)
1189 {
1190         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1191 }
1192
1193 static struct ast_sip_authenticator *registered_authenticator;
1194
1195 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1196 {
1197         if (registered_authenticator) {
1198                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1199                 return -1;
1200         }
1201         registered_authenticator = auth;
1202         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1203         ast_module_ref(ast_module_info->self);
1204         return 0;
1205 }
1206
1207 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1208 {
1209         if (registered_authenticator != auth) {
1210                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1211                                 auth, registered_authenticator);
1212                 return;
1213         }
1214         registered_authenticator = NULL;
1215         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1216         ast_module_unref(ast_module_info->self);
1217 }
1218
1219 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1220 {
1221         if (!registered_authenticator) {
1222                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1223                 return 0;
1224         }
1225
1226         return registered_authenticator->requires_authentication(endpoint, rdata);
1227 }
1228
1229 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1230                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1231 {
1232         if (!registered_authenticator) {
1233                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1234                 return 0;
1235         }
1236         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1237 }
1238
1239 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1240
1241 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1242 {
1243         if (registered_outbound_authenticator) {
1244                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1245                 return -1;
1246         }
1247         registered_outbound_authenticator = auth;
1248         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1249         ast_module_ref(ast_module_info->self);
1250         return 0;
1251 }
1252
1253 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1254 {
1255         if (registered_outbound_authenticator != auth) {
1256                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1257                                 auth, registered_outbound_authenticator);
1258                 return;
1259         }
1260         registered_outbound_authenticator = NULL;
1261         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1262         ast_module_unref(ast_module_info->self);
1263 }
1264
1265 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1266                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1267 {
1268         if (!registered_outbound_authenticator) {
1269                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1270                 return -1;
1271         }
1272         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1273 }
1274
1275 struct endpoint_identifier_list {
1276         struct ast_sip_endpoint_identifier *identifier;
1277         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1278 };
1279
1280 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1281
1282 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1283 {
1284         struct endpoint_identifier_list *id_list_item;
1285         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1286
1287         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1288         if (!id_list_item) {
1289                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1290                 return -1;
1291         }
1292         id_list_item->identifier = identifier;
1293
1294         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1295         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1296
1297         ast_module_ref(ast_module_info->self);
1298         return 0;
1299 }
1300
1301 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1302 {
1303         struct endpoint_identifier_list *iter;
1304         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1305         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1306                 if (iter->identifier == identifier) {
1307                         AST_RWLIST_REMOVE_CURRENT(list);
1308                         ast_free(iter);
1309                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1310                         ast_module_unref(ast_module_info->self);
1311                         break;
1312                 }
1313         }
1314         AST_RWLIST_TRAVERSE_SAFE_END;
1315 }
1316
1317 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1318 {
1319         struct endpoint_identifier_list *iter;
1320         struct ast_sip_endpoint *endpoint = NULL;
1321         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1322         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1323                 ast_assert(iter->identifier->identify_endpoint != NULL);
1324                 endpoint = iter->identifier->identify_endpoint(rdata);
1325                 if (endpoint) {
1326                         break;
1327                 }
1328         }
1329         return endpoint;
1330 }
1331
1332 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1333
1334 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1335 {
1336         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1337         AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1338         ast_module_ref(ast_module_info->self);
1339         return 0;
1340 }
1341
1342 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1343 {
1344         struct ast_sip_endpoint_formatter *i;
1345         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1346         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1347                 if (i == obj) {
1348                         AST_RWLIST_REMOVE_CURRENT(next);
1349                         ast_module_unref(ast_module_info->self);
1350                         break;
1351                 }
1352         }
1353         AST_RWLIST_TRAVERSE_SAFE_END;
1354 }
1355
1356 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1357                                 struct ast_sip_ami *ami, int *count)
1358 {
1359         int res = 0;
1360         struct ast_sip_endpoint_formatter *i;
1361         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1362         *count = 0;
1363         AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1364                 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1365                         return res;
1366                 }
1367
1368                 if (!res) {
1369                         (*count)++;
1370                 }
1371         }
1372         return 0;
1373 }
1374
1375 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1376 {
1377         return ast_pjsip_endpoint;
1378 }
1379
1380 static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user, const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
1381 {
1382         pj_str_t tmp, local_addr;
1383         pjsip_uri *uri;
1384         pjsip_sip_uri *sip_uri;
1385         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1386         int local_port;
1387         char uuid_str[AST_UUID_STR_LEN];
1388
1389         if (ast_strlen_zero(user)) {
1390                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1391                 if (!uuid) {
1392                         return -1;
1393                 }
1394                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1395         }
1396
1397         /* Parse the provided target URI so we can determine what transport it will end up using */
1398         pj_strdup_with_null(pool, &tmp, target);
1399
1400         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1401             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1402                 return -1;
1403         }
1404
1405         sip_uri = pjsip_uri_get_uri(uri);
1406
1407         /* Determine the transport type to use */
1408         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1409                 type = PJSIP_TRANSPORT_TLS;
1410         } else if (!sip_uri->transport_param.slen) {
1411                 type = PJSIP_TRANSPORT_UDP;
1412         } else {
1413                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1414         }
1415
1416         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1417                 return -1;
1418         }
1419
1420         /* If the host is IPv6 turn the transport into an IPv6 version */
1421         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1422                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1423         }
1424
1425         if (!ast_strlen_zero(domain)) {
1426                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1427                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1428                                 "<sip:%s@%s%s%s>",
1429                                 user,
1430                                 domain,
1431                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1432                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1433                 return 0;
1434         }
1435
1436         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1437         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1438                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1439
1440                 /* If no local address can be retrieved using the transport manager use the host one */
1441                 pj_strdup(pool, &local_addr, pj_gethostname());
1442                 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1443         }
1444
1445         /* If IPv6 was specified in the transport, set the proper type */
1446         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1447                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1448         }
1449
1450         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1451         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1452                                       "<sip:%s@%s%.*s%s:%d%s%s>",
1453                                       user,
1454                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1455                                       (int)local_addr.slen,
1456                                       local_addr.ptr,
1457                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1458                                       local_port,
1459                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1460                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1461
1462         return 0;
1463 }
1464
1465 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1466 {
1467         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1468         const char *transport_name = endpoint->transport;
1469
1470         if (ast_strlen_zero(transport_name)) {
1471                 return 0;
1472         }
1473
1474         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1475
1476         if (!transport || !transport->state) {
1477                 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1478                         transport_name, ast_sorcery_object_get_id(endpoint));
1479                 return -1;
1480         }
1481
1482         if (transport->state->transport) {
1483                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1484                 selector->u.transport = transport->state->transport;
1485         } else if (transport->state->factory) {
1486                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1487                 selector->u.listener = transport->state->factory;
1488         } else {
1489                 return -1;
1490         }
1491
1492         return 0;
1493 }
1494
1495 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1496 {
1497         char enclosed_uri[PJSIP_MAX_URL_SIZE];
1498         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1499         pjsip_dialog *dlg = NULL;
1500         const char *outbound_proxy = endpoint->outbound_proxy;
1501         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1502         static const pj_str_t HCONTACT = { "Contact", 7 };
1503
1504         snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1505         pj_cstr(&remote_uri, enclosed_uri);
1506
1507         pj_cstr(&target_uri, uri);
1508
1509         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1510                 return NULL;
1511         }
1512
1513         if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1514                 pjsip_dlg_terminate(dlg);
1515                 return NULL;
1516         }
1517
1518         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1519                 pjsip_dlg_terminate(dlg);
1520                 return NULL;
1521         }
1522
1523         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1524         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1525         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1526         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1527
1528         /* If a request user has been specified and we are permitted to change it, do so */
1529         if (!ast_strlen_zero(request_user)) {
1530                 pjsip_sip_uri *sip_uri;
1531
1532                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1533                         sip_uri = pjsip_uri_get_uri(dlg->target);
1534                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1535                 }
1536                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1537                         sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1538                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1539                 }
1540         }
1541
1542         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1543         dlg->sess_count++;
1544
1545         pjsip_dlg_set_transport(dlg, &selector);
1546
1547         if (!ast_strlen_zero(outbound_proxy)) {
1548                 pjsip_route_hdr route_set, *route;
1549                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1550                 pj_str_t tmp;
1551
1552                 pj_list_init(&route_set);
1553
1554                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1555                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1556                         dlg->sess_count--;
1557                         pjsip_dlg_terminate(dlg);
1558                         return NULL;
1559                 }
1560                 pj_list_insert_nodes_before(&route_set, route);
1561
1562                 pjsip_dlg_set_route_set(dlg, &route_set);
1563         }
1564
1565         dlg->sess_count--;
1566
1567         return dlg;
1568 }
1569
1570 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1571 {
1572         pjsip_dialog *dlg;
1573         pj_str_t contact;
1574         pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1575         pj_status_t status;
1576
1577         contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1578         contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1579                         "<sip:%s%.*s%s:%d%s%s>",
1580                         (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1581                         (int)rdata->tp_info.transport->local_name.host.slen,
1582                         rdata->tp_info.transport->local_name.host.ptr,
1583                         (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1584                         rdata->tp_info.transport->local_name.port,
1585                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1586                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1587
1588         status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1589         if (status != PJ_SUCCESS) {
1590                 char err[PJ_ERR_MSG_SIZE];
1591
1592                 pj_strerror(status, err, sizeof(err));
1593                 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1594                                 ast_sorcery_object_get_id(endpoint), err);
1595                 return NULL;
1596         }
1597
1598         return dlg;
1599 }
1600
1601 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1602 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1603 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1604
1605 static struct {
1606         const char *method;
1607         const pjsip_method *pmethod;
1608 } methods [] = {
1609         { "INVITE", &pjsip_invite_method },
1610         { "CANCEL", &pjsip_cancel_method },
1611         { "ACK", &pjsip_ack_method },
1612         { "BYE", &pjsip_bye_method },
1613         { "REGISTER", &pjsip_register_method },
1614         { "OPTIONS", &pjsip_options_method },
1615         { "SUBSCRIBE", &pjsip_subscribe_method },
1616         { "NOTIFY", &pjsip_notify_method },
1617         { "PUBLISH", &pjsip_publish_method },
1618         { "INFO", &info_method },
1619         { "MESSAGE", &message_method },
1620 };
1621
1622 static const pjsip_method *get_pjsip_method(const char *method)
1623 {
1624         int i;
1625         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1626                 if (!strcmp(method, methods[i].method)) {
1627                         return methods[i].pmethod;
1628                 }
1629         }
1630         return NULL;
1631 }
1632
1633 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1634 {
1635         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1636                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1637                 return -1;
1638         }
1639
1640         return 0;
1641 }
1642
1643 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata);
1644 static pjsip_module supplement_module = {
1645         .name = { "Out of dialog supplement hook", 29 },
1646         .id = -1,
1647         .priority = PJSIP_MOD_PRIORITY_APPLICATION - 1,
1648         .on_rx_request = supplement_on_rx_request,
1649 };
1650
1651 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1652                 const char *uri, struct ast_sip_contact *provided_contact, pjsip_tx_data **tdata)
1653 {
1654         RAII_VAR(struct ast_sip_contact *, contact, ao2_bump(provided_contact), ao2_cleanup);
1655         pj_str_t remote_uri;
1656         pj_str_t from;
1657         pj_pool_t *pool;
1658         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1659
1660         if (ast_strlen_zero(uri)) {
1661                 if (!endpoint && !contact) {
1662                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1663                         return -1;
1664                 }
1665
1666                 if (!contact) {
1667                         contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1668                 }
1669                 if (!contact || ast_strlen_zero(contact->uri)) {
1670                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1671                                         ast_sorcery_object_get_id(endpoint));
1672                         return -1;
1673                 }
1674
1675                 pj_cstr(&remote_uri, contact->uri);
1676         } else {
1677                 pj_cstr(&remote_uri, uri);
1678         }
1679
1680         if (endpoint) {
1681                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1682                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1683                                 ast_sorcery_object_get_id(endpoint));
1684                         return -1;
1685                 }
1686         }
1687
1688         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1689
1690         if (!pool) {
1691                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1692                 return -1;
1693         }
1694
1695         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1696                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1697                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1698                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1699                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1700                 return -1;
1701         }
1702
1703         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1704                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1705                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1706                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1707                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1708                 return -1;
1709         }
1710
1711         /* If an outbound proxy is specified on the endpoint apply it to this request */
1712         if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
1713                 ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
1714                 ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
1715                         (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1716                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1717                 return -1;
1718         }
1719
1720         ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
1721
1722         /* We can release this pool since request creation copied all the necessary
1723          * data into the outbound request's pool
1724          */
1725         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1726         return 0;
1727 }
1728
1729 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1730                 struct ast_sip_endpoint *endpoint, const char *uri,
1731                 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
1732 {
1733         const pjsip_method *pmethod = get_pjsip_method(method);
1734
1735         if (!pmethod) {
1736                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1737                 return -1;
1738         }
1739
1740         if (dlg) {
1741                 return create_in_dialog_request(pmethod, dlg, tdata);
1742         } else {
1743                 return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
1744         }
1745 }
1746
1747 AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
1748
1749 int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
1750 {
1751         struct ast_sip_supplement *iter;
1752         int inserted = 0;
1753         SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1754
1755         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1756                 if (iter->priority > supplement->priority) {
1757                         AST_RWLIST_INSERT_BEFORE_CURRENT(supplement, next);
1758                         inserted = 1;
1759                         break;
1760                 }
1761         }
1762         AST_RWLIST_TRAVERSE_SAFE_END;
1763
1764         if (!inserted) {
1765                 AST_RWLIST_INSERT_TAIL(&supplements, supplement, next);
1766         }
1767         ast_module_ref(ast_module_info->self);
1768         return 0;
1769 }
1770
1771 void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
1772 {
1773         struct ast_sip_supplement *iter;
1774         SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1775         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1776                 if (supplement == iter) {
1777                         AST_RWLIST_REMOVE_CURRENT(next);
1778                         ast_module_unref(ast_module_info->self);
1779                         break;
1780                 }
1781         }
1782         AST_RWLIST_TRAVERSE_SAFE_END;
1783 }
1784
1785 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1786 {
1787         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1788                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1789                 return -1;
1790         }
1791         return 0;
1792 }
1793
1794 static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
1795 {
1796         pj_str_t method;
1797
1798         if (ast_strlen_zero(supplement_method)) {
1799                 return PJ_TRUE;
1800         }
1801
1802         pj_cstr(&method, supplement_method);
1803
1804         return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
1805 }
1806
1807 /*! \brief Structure to hold information about an outbound request */
1808 struct send_request_data {
1809         struct ast_sip_endpoint *endpoint;              /*! The endpoint associated with this request */
1810         void *token;                                    /*! Information to be provided to the callback upon receipt of a response */
1811         void (*callback)(void *token, pjsip_event *e);  /*! The callback to be called upon receipt of a response */
1812 };
1813
1814 static void send_request_data_destroy(void *obj)
1815 {
1816         struct send_request_data *req_data = obj;
1817         ao2_cleanup(req_data->endpoint);
1818 }
1819
1820 static struct send_request_data *send_request_data_alloc(struct ast_sip_endpoint *endpoint,
1821         void *token, void (*callback)(void *token, pjsip_event *e))
1822 {
1823         struct send_request_data *req_data = ao2_alloc(sizeof(*req_data), send_request_data_destroy);
1824
1825         if (!req_data) {
1826                 return NULL;
1827         }
1828
1829         req_data->endpoint = ao2_bump(endpoint);
1830         req_data->token = token;
1831         req_data->callback = callback;
1832
1833         return req_data;
1834 }
1835
1836 static void send_request_cb(void *token, pjsip_event *e)
1837 {
1838         RAII_VAR(struct send_request_data *, req_data, token, ao2_cleanup);
1839         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1840         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1841         pjsip_tx_data *tdata;
1842         struct ast_sip_supplement *supplement;
1843
1844         AST_RWLIST_RDLOCK(&supplements);
1845         AST_LIST_TRAVERSE(&supplements, supplement, next) {
1846                 if (supplement->incoming_response && does_method_match(&challenge->msg_info.cseq->method.name, supplement->method)) {
1847                         supplement->incoming_response(req_data->endpoint, challenge);
1848                 }
1849         }
1850         AST_RWLIST_UNLOCK(&supplements);
1851
1852         if (tsx->status_code == 401 || tsx->status_code == 407) {
1853                 if (!ast_sip_create_request_with_auth(&req_data->endpoint->outbound_auths, challenge, tsx, &tdata)) {
1854                         pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data->token, req_data->callback);
1855                 }
1856                 return;
1857         }
1858
1859         if (req_data->callback) {
1860                 req_data->callback(req_data->token, e);
1861         }
1862 }
1863
1864 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint,
1865         void *token, void (*callback)(void *token, pjsip_event *e))
1866 {
1867         struct ast_sip_supplement *supplement;
1868         struct send_request_data *req_data = send_request_data_alloc(endpoint, token, callback);
1869         struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
1870
1871         if (!req_data) {
1872                 return -1;
1873         }
1874
1875         AST_RWLIST_RDLOCK(&supplements);
1876         AST_LIST_TRAVERSE(&supplements, supplement, next) {
1877                 if (supplement->outgoing_request && does_method_match(&tdata->msg->line.req.method.name, supplement->method)) {
1878                         supplement->outgoing_request(endpoint, contact, tdata);
1879                 }
1880         }
1881         AST_RWLIST_UNLOCK(&supplements);
1882
1883         ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
1884         ao2_cleanup(contact);
1885
1886         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data, send_request_cb) != PJ_SUCCESS) {
1887                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1888                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1889                                 pj_strbuf(&tdata->msg->line.req.method.name),
1890                                 ast_sorcery_object_get_id(endpoint));
1891                 ao2_cleanup(req_data);
1892                 return -1;
1893         }
1894
1895         return 0;
1896 }
1897
1898 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
1899         struct ast_sip_endpoint *endpoint, void *token,
1900         void (*callback)(void *token, pjsip_event *e))
1901 {
1902         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1903
1904         if (dlg) {
1905                 return send_in_dialog_request(tdata, dlg);
1906         } else {
1907                 return send_out_of_dialog_request(tdata, endpoint, token, callback);
1908         }
1909 }
1910
1911 int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
1912 {
1913         pjsip_route_hdr *route;
1914         static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1915         pj_str_t tmp;
1916
1917         pj_strdup2_with_null(tdata->pool, &tmp, proxy);
1918         if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1919                 return -1;
1920         }
1921
1922         pj_list_insert_nodes_before(&tdata->msg->hdr, (pjsip_hdr*)route);
1923
1924         return 0;
1925 }
1926
1927 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1928 {
1929         pj_str_t hdr_name;
1930         pj_str_t hdr_value;
1931         pjsip_generic_string_hdr *hdr;
1932
1933         pj_cstr(&hdr_name, name);
1934         pj_cstr(&hdr_value, value);
1935
1936         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1937
1938         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1939         return 0;
1940 }
1941
1942 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1943 {
1944         pj_str_t type;
1945         pj_str_t subtype;
1946         pj_str_t body_text;
1947
1948         pj_cstr(&type, body->type);
1949         pj_cstr(&subtype, body->subtype);
1950         pj_cstr(&body_text, body->body_text);
1951
1952         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1953 }
1954
1955 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1956 {
1957         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1958         tdata->msg->body = pjsip_body;
1959         return 0;
1960 }
1961
1962 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1963 {
1964         int i;
1965         /* NULL for type and subtype automatically creates "multipart/mixed" */
1966         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1967
1968         for (i = 0; i < num_bodies; ++i) {
1969                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1970                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1971                 pjsip_multipart_add_part(tdata->pool, body, part);
1972         }
1973
1974         tdata->msg->body = body;
1975         return 0;
1976 }
1977
1978 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1979 {
1980         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1981         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1982
1983         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1984
1985         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1986         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1987         tdata->msg->body->len = combined_size;
1988
1989         return 0;
1990 }
1991
1992 struct ast_taskprocessor *ast_sip_create_serializer(void)
1993 {
1994         struct ast_taskprocessor *serializer;
1995         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1996         char name[AST_UUID_STR_LEN];
1997
1998         if (!uuid) {
1999                 return NULL;
2000         }
2001
2002         ast_uuid_to_str(uuid, name, sizeof(name));
2003
2004         serializer = ast_threadpool_serializer(name, sip_threadpool);
2005         if (!serializer) {
2006                 return NULL;
2007         }
2008         return serializer;
2009 }
2010
2011 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2012 {
2013         if (serializer) {
2014                 return ast_taskprocessor_push(serializer, sip_task, task_data);
2015         } else {
2016                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
2017         }
2018 }
2019
2020 struct sync_task_data {
2021         ast_mutex_t lock;
2022         ast_cond_t cond;
2023         int complete;
2024         int fail;
2025         int (*task)(void *);
2026         void *task_data;
2027 };
2028
2029 static int sync_task(void *data)
2030 {
2031         struct sync_task_data *std = data;
2032         std->fail = std->task(std->task_data);
2033
2034         ast_mutex_lock(&std->lock);
2035         std->complete = 1;
2036         ast_cond_signal(&std->cond);
2037         ast_mutex_unlock(&std->lock);
2038         return std->fail;
2039 }
2040
2041 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2042 {
2043         /* This method is an onion */
2044         struct sync_task_data std;
2045
2046         if (ast_sip_thread_is_servant()) {
2047                 return sip_task(task_data);
2048         }
2049
2050         ast_mutex_init(&std.lock);
2051         ast_cond_init(&std.cond, NULL);
2052         std.fail = std.complete = 0;
2053         std.task = sip_task;
2054         std.task_data = task_data;
2055
2056         if (serializer) {
2057                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
2058                         return -1;
2059                 }
2060         } else {
2061                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
2062                         return -1;
2063                 }
2064         }
2065
2066         ast_mutex_lock(&std.lock);
2067         while (!std.complete) {
2068                 ast_cond_wait(&std.cond, &std.lock);
2069         }
2070         ast_mutex_unlock(&std.lock);
2071
2072         ast_mutex_destroy(&std.lock);
2073         ast_cond_destroy(&std.cond);
2074         return std.fail;
2075 }
2076
2077 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
2078 {
2079         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
2080         memcpy(dest, pj_strbuf(src), chars_to_copy);
2081         dest[chars_to_copy] = '\0';
2082 }
2083
2084 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
2085 {
2086         pjsip_media_type compare;
2087
2088         if (!content_type) {
2089                 return 0;
2090         }
2091
2092         pjsip_media_type_init2(&compare, type, subtype);
2093
2094         return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
2095 }
2096
2097 pj_caching_pool caching_pool;
2098 pj_pool_t *memory_pool;
2099 pj_thread_t *monitor_thread;
2100 static int monitor_continue;
2101
2102 static void *monitor_thread_exec(void *endpt)
2103 {
2104         while (monitor_continue) {
2105                 const pj_time_val delay = {0, 10};
2106                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
2107         }
2108         return NULL;
2109 }
2110
2111 static void stop_monitor_thread(void)
2112 {
2113         monitor_continue = 0;
2114         pj_thread_join(monitor_thread);
2115 }
2116
2117 AST_THREADSTORAGE(pj_thread_storage);
2118 AST_THREADSTORAGE(servant_id_storage);
2119 #define SIP_SERVANT_ID 0x5E2F1D
2120
2121 static void sip_thread_start(void)
2122 {
2123         pj_thread_desc *desc;
2124         pj_thread_t *thread;
2125         uint32_t *servant_id;
2126
2127         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2128         if (!servant_id) {
2129                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
2130                 return;
2131         }
2132         *servant_id = SIP_SERVANT_ID;
2133
2134         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
2135         if (!desc) {
2136                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
2137                 return;
2138         }
2139         pj_bzero(*desc, sizeof(*desc));
2140
2141         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
2142                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
2143         }
2144 }
2145
2146 int ast_sip_thread_is_servant(void)
2147 {
2148         uint32_t *servant_id;
2149
2150         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2151         if (!servant_id) {
2152                 return 0;
2153         }
2154
2155         return *servant_id == SIP_SERVANT_ID;
2156 }
2157
2158 void *ast_sip_dict_get(void *ht, const char *key)
2159 {
2160         unsigned int hval = 0;
2161
2162         if (!ht) {
2163                 return NULL;
2164         }
2165
2166         return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
2167 }
2168
2169 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
2170                        const char *key, void *val)
2171 {
2172         if (!ht) {
2173                 ht = pj_hash_create(pool, 11);
2174         }
2175
2176         pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
2177
2178         return ht;
2179 }
2180
2181 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata)
2182 {
2183         struct ast_sip_supplement *supplement;
2184
2185         if (pjsip_rdata_get_dlg(rdata)) {
2186                 return PJ_FALSE;
2187         }
2188
2189         AST_RWLIST_RDLOCK(&supplements);
2190         AST_LIST_TRAVERSE(&supplements, supplement, next) {
2191                 if (supplement->incoming_request && does_method_match(&rdata->msg_info.msg->line.req.method.name, supplement->method)) {
2192                         supplement->incoming_request(ast_pjsip_rdata_get_endpoint(rdata), rdata);
2193                 }
2194         }
2195         AST_RWLIST_UNLOCK(&supplements);
2196
2197         return PJ_FALSE;
2198 }
2199
2200 int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
2201 {
2202         struct ast_sip_supplement *supplement;
2203         pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
2204         struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
2205
2206         AST_RWLIST_RDLOCK(&supplements);
2207         AST_LIST_TRAVERSE(&supplements, supplement, next) {
2208                 if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
2209                         supplement->outgoing_response(sip_endpoint, contact, tdata);
2210                 }
2211         }
2212         AST_RWLIST_UNLOCK(&supplements);
2213
2214         ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
2215         ao2_cleanup(contact);
2216
2217         return pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
2218 }
2219
2220 int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
2221         struct ast_sip_contact *contact, pjsip_tx_data **tdata)
2222 {
2223         int res = pjsip_endpt_create_response(ast_sip_get_pjsip_endpoint(), rdata, st_code, NULL, tdata);
2224
2225         if (!res) {
2226                 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
2227         }
2228
2229         return res;
2230 }
2231
2232 static void remove_request_headers(pjsip_endpoint *endpt)
2233 {
2234         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
2235         pjsip_hdr *iter = request_headers->next;
2236
2237         while (iter != request_headers) {
2238                 pjsip_hdr *to_erase = iter;
2239                 iter = iter->next;
2240                 pj_list_erase(to_erase);
2241         }
2242 }
2243
2244 static int load_module(void)
2245 {
2246         /* The third parameter is just copied from
2247          * example code from PJLIB. This can be adjusted
2248          * if necessary.
2249          */
2250         pj_status_t status;
2251         struct ast_threadpool_options options;
2252
2253         if (pj_init() != PJ_SUCCESS) {
2254                 return AST_MODULE_LOAD_DECLINE;
2255         }
2256
2257         if (pjlib_util_init() != PJ_SUCCESS) {
2258                 pj_shutdown();
2259                 return AST_MODULE_LOAD_DECLINE;
2260         }
2261
2262         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
2263         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
2264                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
2265                 pj_caching_pool_destroy(&caching_pool);
2266                 return AST_MODULE_LOAD_DECLINE;
2267         }
2268
2269         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2270          * we need to stop PJSIP from doing it automatically
2271          */
2272         remove_request_headers(ast_pjsip_endpoint);
2273
2274         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2275         if (!memory_pool) {
2276                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2277                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2278                 ast_pjsip_endpoint = NULL;
2279                 pj_caching_pool_destroy(&caching_pool);
2280                 return AST_MODULE_LOAD_DECLINE;
2281         }
2282
2283         if (ast_sip_initialize_system()) {
2284                 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2285                 pj_pool_release(memory_pool);
2286                 memory_pool = NULL;
2287                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2288                 ast_pjsip_endpoint = NULL;
2289                 pj_caching_pool_destroy(&caching_pool);
2290                 return AST_MODULE_LOAD_DECLINE;
2291         }
2292
2293         sip_get_threadpool_options(&options);
2294         options.thread_start = sip_thread_start;
2295         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2296         if (!sip_threadpool) {
2297                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2298                 pj_pool_release(memory_pool);
2299                 memory_pool = NULL;
2300                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2301                 ast_pjsip_endpoint = NULL;
2302                 pj_caching_pool_destroy(&caching_pool);
2303                 return AST_MODULE_LOAD_DECLINE;
2304         }
2305
2306         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2307         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2308
2309         monitor_continue = 1;
2310         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2311                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2312         if (status != PJ_SUCCESS) {
2313                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2314                 pj_pool_release(memory_pool);
2315                 memory_pool = NULL;
2316                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2317                 ast_pjsip_endpoint = NULL;
2318                 pj_caching_pool_destroy(&caching_pool);
2319                 return AST_MODULE_LOAD_DECLINE;
2320         }
2321
2322         ast_sip_initialize_global_headers();
2323
2324         if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2325                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2326                 ast_sip_destroy_global_headers();
2327                 stop_monitor_thread();
2328                 pj_pool_release(memory_pool);
2329                 memory_pool = NULL;
2330                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2331                 ast_pjsip_endpoint = NULL;
2332                 pj_caching_pool_destroy(&caching_pool);
2333                 return AST_MODULE_LOAD_DECLINE;
2334         }
2335
2336         if (ast_sip_initialize_distributor()) {
2337                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2338                 ast_res_pjsip_destroy_configuration();
2339                 ast_sip_destroy_global_headers();
2340                 stop_monitor_thread();
2341                 pj_pool_release(memory_pool);
2342                 memory_pool = NULL;
2343                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2344                 ast_pjsip_endpoint = NULL;
2345                 pj_caching_pool_destroy(&caching_pool);
2346                 return AST_MODULE_LOAD_DECLINE;
2347         }
2348
2349         if (ast_sip_register_service(&supplement_module)) {
2350                 ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
2351                 ast_sip_destroy_distributor();
2352                 ast_res_pjsip_destroy_configuration();
2353                 ast_sip_destroy_global_headers();
2354                 stop_monitor_thread();
2355                 pj_pool_release(memory_pool);
2356                 memory_pool = NULL;
2357                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2358                 ast_pjsip_endpoint = NULL;
2359                 pj_caching_pool_destroy(&caching_pool);
2360                 return AST_MODULE_LOAD_DECLINE;
2361         }
2362
2363         if (ast_sip_initialize_outbound_authentication()) {
2364                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2365                 ast_sip_unregister_service(&supplement_module);
2366                 ast_sip_destroy_distributor();
2367                 ast_res_pjsip_destroy_configuration();
2368                 ast_sip_destroy_global_headers();
2369                 stop_monitor_thread();
2370                 pj_pool_release(memory_pool);
2371                 memory_pool = NULL;
2372                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2373                 ast_pjsip_endpoint = NULL;
2374                 pj_caching_pool_destroy(&caching_pool);
2375                 return AST_MODULE_LOAD_DECLINE;
2376         }
2377
2378         ast_res_pjsip_init_options_handling(0);
2379
2380         ast_module_ref(ast_module_info->self);
2381
2382         return AST_MODULE_LOAD_SUCCESS;
2383 }
2384
2385 static int reload_module(void)
2386 {
2387         if (ast_res_pjsip_reload_configuration()) {
2388                 return AST_MODULE_LOAD_DECLINE;
2389         }
2390         ast_res_pjsip_init_options_handling(1);
2391         return 0;
2392 }
2393
2394 static int unload_module(void)
2395 {
2396         /* This will never get called as this module can't be unloaded */
2397         return 0;
2398 }
2399
2400 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2401                 .load = load_module,
2402                 .unload = unload_module,
2403                 .reload = reload_module,
2404                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
2405 );