res_pjsip: Documentation improvement for Endpoint and AOR mailbox options.
[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                         </configObject>
901                         <configObject name="aor">
902                                 <synopsis>The configuration for a location of an endpoint</synopsis>
903                                 <description><para>
904                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
905                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
906                                         Beyond that, an AoR has other uses within Asterisk, such as inbound
907                                         registration.
908                                         </para><para>
909                                         An <literal>AoR</literal> is a way to allow dialing a group
910                                         of <literal>Contacts</literal> that all use the same
911                                         <literal>endpoint</literal> for calls.
912                                         </para><para>
913                                         This can be used as another way of grouping a list of contacts to dial
914                                         rather than specifing them each directly when dialing via the dialplan.
915                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
916                                         </para><para>
917                                         Registrations: For Asterisk to match an inbound registration to an endpoint,
918                                         the AoR object name must match the user portion of the SIP URI in the "To:"
919                                         header of the inbound SIP registration. That will usually be equivalent
920                                         to the "user name" set in your hard or soft phones configuration.
921                                 </para></description>
922                                 <configOption name="contact">
923                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
924                                         <description><para>
925                                                 Contacts specified will be called whenever referenced
926                                                 by <literal>chan_pjsip</literal>.
927                                                 </para><para>
928                                                 Use a separate "contact=" entry for each contact required. Contacts
929                                                 are specified using a SIP URI.
930                                         </para></description>
931                                 </configOption>
932                                 <configOption name="default_expiration" default="3600">
933                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
934                                 </configOption>
935                                 <configOption name="mailboxes">
936                                         <synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
937                                         <description><para>This option applies when an external entity subscribes to an AoR
938                                                 for Message Waiting Indications. The mailboxes specified will be subscribed to.
939                                                 More than one mailbox can be specified with a comma-delimited string.
940                                                 app_voicemail mailboxes must be specified as mailbox@context;
941                                                 for example: mailboxes=6001@default. For mailboxes provided by external sources,
942                                                 such as through the res_external_mwi module, you must specify strings supported by
943                                                 the external system.
944                                         </para><para>
945                                                 For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
946                                                 endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
947                                         </para></description>
948                                 </configOption>
949                                 <configOption name="maximum_expiration" default="7200">
950                                         <synopsis>Maximum time to keep an AoR</synopsis>
951                                         <description><para>
952                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
953                                         </para></description>
954                                 </configOption>
955                                 <configOption name="max_contacts" default="0">
956                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
957                                         <description><para>
958                                                 Maximum number of contacts that can associate with this AoR. This value does
959                                                 not affect the number of contacts that can be added with the "contact" option.
960                                                 It only limits contacts added through external interaction, such as
961                                                 registration.
962                                                 </para>
963                                                 <note><para>This should be set to <literal>1</literal> and
964                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
965                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
966                                                 </para></note>
967                                         </description>
968                                 </configOption>
969                                 <configOption name="minimum_expiration" default="60">
970                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
971                                         <description><para>
972                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
973                                         </para></description>
974                                 </configOption>
975                                 <configOption name="remove_existing" default="no">
976                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
977                                         <description><para>
978                                                 On receiving a new registration to the AoR should it remove
979                                                 the existing contact that was registered against it?
980                                                 </para>
981                                                 <note><para>This should be set to <literal>yes</literal> and
982                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
983                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
984                                                 </para></note>
985                                         </description>
986                                 </configOption>
987                                 <configOption name="type">
988                                         <synopsis>Must be of type 'aor'.</synopsis>
989                                 </configOption>
990                                 <configOption name="qualify_frequency" default="0">
991                                         <synopsis>Interval at which to qualify an AoR</synopsis>
992                                         <description><para>
993                                                 Interval between attempts to qualify the AoR for reachability.
994                                                 If <literal>0</literal> never qualify. Time in seconds.
995                                         </para></description>
996                                 </configOption>
997                                 <configOption name="authenticate_qualify" default="no">
998                                         <synopsis>Authenticates a qualify request if needed</synopsis>
999                                         <description><para>
1000                                                 If true and a qualify request receives a challenge or authenticate response
1001                                                 authentication is attempted before declaring the contact available.
1002                                         </para></description>
1003                                 </configOption>
1004                                 <configOption name="outbound_proxy">
1005                                         <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
1006                                         <description><para>
1007                                                 If set the provided URI will be used as the outbound proxy when an
1008                                                 OPTIONS request is sent to a contact for qualify purposes.
1009                                         </para></description>
1010                                 </configOption>
1011                                 <configOption name="support_path">
1012                                         <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
1013                                         <description><para>
1014                                                 When this option is enabled, the Path headers in register requests will be saved
1015                                                 and its contents will be used in Route headers for outbound out-of-dialog requests
1016                                                 and in Path headers for outbound 200 responses. Path support will also be indicated
1017                                                 in the Supported header.
1018                                         </para></description>
1019                                 </configOption>
1020                         </configObject>
1021                         <configObject name="system">
1022                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
1023                                 <description><para>
1024                                         The settings in this section are global. In addition to being global, the values will
1025                                         not be re-evaluated when a reload is performed. This is because the values must be set
1026                                         before the SIP stack is initialized. The only way to reset these values is to either
1027                                         restart Asterisk, or unload res_pjsip.so and then load it again.
1028                                 </para></description>
1029                                 <configOption name="timer_t1" default="500">
1030                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
1031                                         <description><para>
1032                                                 Timer T1 is the base for determining how long to wait before retransmitting
1033                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
1034                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
1035                                         </para></description>
1036                                 </configOption>
1037                                 <configOption name="timer_b" default="32000">
1038                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
1039                                         <description><para>
1040                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
1041                                                 request before terminating the transaction. It is recommended that this be set
1042                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
1043                                                 this timer, see RFC 3261, Section 17.1.1.1.
1044                                         </para></description>
1045                                 </configOption>
1046                                 <configOption name="compact_headers" default="no">
1047                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
1048                                 </configOption>
1049                                 <configOption name="threadpool_initial_size" default="0">
1050                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
1051                                 </configOption>
1052                                 <configOption name="threadpool_auto_increment" default="5">
1053                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1054                                 </configOption>
1055                                 <configOption name="threadpool_idle_timeout" default="60">
1056                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1057                                 </configOption>
1058                                 <configOption name="threadpool_max_size" default="0">
1059                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
1060                                         A value of 0 indicates no maximum.</synopsis>
1061                                 </configOption>
1062                                 <configOption name="type">
1063                                         <synopsis>Must be of type 'system'.</synopsis>
1064                                 </configOption>
1065                         </configObject>
1066                         <configObject name="global">
1067                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
1068                                 <description><para>
1069                                         The settings in this section are global. Unlike options in the <literal>system</literal>
1070                                         section, these options can be refreshed by performing a reload.
1071                                 </para></description>
1072                                 <configOption name="max_forwards" default="70">
1073                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1074                                 </configOption>
1075                                 <configOption name="type">
1076                                         <synopsis>Must be of type 'global'.</synopsis>
1077                                 </configOption>
1078                                 <configOption name="user_agent" default="Asterisk &lt;Asterisk Version&gt;">
1079                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1080                                 </configOption>
1081                                 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1082                                         <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1083                                 </configOption>
1084
1085                         </configObject>
1086                 </configFile>
1087         </configInfo>
1088         <manager name="PJSIPQualify" language="en_US">
1089                 <synopsis>
1090                         Qualify a chan_pjsip endpoint.
1091                 </synopsis>
1092                 <syntax>
1093                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1094                         <parameter name="Endpoint" required="true">
1095                                 <para>The endpoint you want to qualify.</para>
1096                         </parameter>
1097                 </syntax>
1098                 <description>
1099                         <para>Qualify a chan_pjsip endpoint.</para>
1100                 </description>
1101         </manager>
1102         <manager name="PJSIPShowEndpoints" language="en_US">
1103                 <synopsis>
1104                         Lists PJSIP endpoints.
1105                 </synopsis>
1106                 <syntax />
1107                 <description>
1108                         <para>
1109                         Provides a listing of all endpoints.  For each endpoint an <literal>EndpointList</literal> event
1110                         is raised that contains relevant attributes and status information.  Once all
1111                         endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1112                         </para>
1113                 </description>
1114         </manager>
1115         <manager name="PJSIPShowEndpoint" language="en_US">
1116                 <synopsis>
1117                         Detail listing of an endpoint and its objects.
1118                 </synopsis>
1119                 <syntax>
1120                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1121                         <parameter name="Endpoint" required="true">
1122                                 <para>The endpoint to list.</para>
1123                         </parameter>
1124                 </syntax>
1125                 <description>
1126                         <para>
1127                         Provides a detailed listing of options for a given endpoint.  Events are issued
1128                         showing the configuration and status of the endpoint and associated objects.  These
1129                         events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1130                         <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1131                         <literal>IdentifyDetail</literal>.  Some events may be listed multiple times if multiple objects are
1132                         associated (for instance AoRs).  Once all detail events have been raised a final
1133                         <literal>EndpointDetailComplete</literal> event is issued.
1134                         </para>
1135                 </description>
1136         </manager>
1137  ***/
1138
1139 #define MOD_DATA_CONTACT "contact"
1140
1141 static pjsip_endpoint *ast_pjsip_endpoint;
1142
1143 static struct ast_threadpool *sip_threadpool;
1144
1145 static int register_service(void *data)
1146 {
1147         pjsip_module **module = data;
1148         if (!ast_pjsip_endpoint) {
1149                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1150                 return -1;
1151         }
1152         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1153                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1154                 return -1;
1155         }
1156         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1157         ast_module_ref(ast_module_info->self);
1158         return 0;
1159 }
1160
1161 int ast_sip_register_service(pjsip_module *module)
1162 {
1163         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1164 }
1165
1166 static int unregister_service(void *data)
1167 {
1168         pjsip_module **module = data;
1169         ast_module_unref(ast_module_info->self);
1170         if (!ast_pjsip_endpoint) {
1171                 return -1;
1172         }
1173         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1174         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1175         return 0;
1176 }
1177
1178 void ast_sip_unregister_service(pjsip_module *module)
1179 {
1180         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1181 }
1182
1183 static struct ast_sip_authenticator *registered_authenticator;
1184
1185 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1186 {
1187         if (registered_authenticator) {
1188                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1189                 return -1;
1190         }
1191         registered_authenticator = auth;
1192         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1193         ast_module_ref(ast_module_info->self);
1194         return 0;
1195 }
1196
1197 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1198 {
1199         if (registered_authenticator != auth) {
1200                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1201                                 auth, registered_authenticator);
1202                 return;
1203         }
1204         registered_authenticator = NULL;
1205         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1206         ast_module_unref(ast_module_info->self);
1207 }
1208
1209 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1210 {
1211         if (!registered_authenticator) {
1212                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1213                 return 0;
1214         }
1215
1216         return registered_authenticator->requires_authentication(endpoint, rdata);
1217 }
1218
1219 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1220                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1221 {
1222         if (!registered_authenticator) {
1223                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1224                 return 0;
1225         }
1226         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1227 }
1228
1229 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1230
1231 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1232 {
1233         if (registered_outbound_authenticator) {
1234                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1235                 return -1;
1236         }
1237         registered_outbound_authenticator = auth;
1238         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1239         ast_module_ref(ast_module_info->self);
1240         return 0;
1241 }
1242
1243 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1244 {
1245         if (registered_outbound_authenticator != auth) {
1246                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1247                                 auth, registered_outbound_authenticator);
1248                 return;
1249         }
1250         registered_outbound_authenticator = NULL;
1251         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1252         ast_module_unref(ast_module_info->self);
1253 }
1254
1255 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1256                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1257 {
1258         if (!registered_outbound_authenticator) {
1259                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1260                 return -1;
1261         }
1262         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1263 }
1264
1265 struct endpoint_identifier_list {
1266         struct ast_sip_endpoint_identifier *identifier;
1267         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1268 };
1269
1270 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1271
1272 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1273 {
1274         struct endpoint_identifier_list *id_list_item;
1275         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1276
1277         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1278         if (!id_list_item) {
1279                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1280                 return -1;
1281         }
1282         id_list_item->identifier = identifier;
1283
1284         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1285         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1286
1287         ast_module_ref(ast_module_info->self);
1288         return 0;
1289 }
1290
1291 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1292 {
1293         struct endpoint_identifier_list *iter;
1294         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1295         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1296                 if (iter->identifier == identifier) {
1297                         AST_RWLIST_REMOVE_CURRENT(list);
1298                         ast_free(iter);
1299                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1300                         ast_module_unref(ast_module_info->self);
1301                         break;
1302                 }
1303         }
1304         AST_RWLIST_TRAVERSE_SAFE_END;
1305 }
1306
1307 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1308 {
1309         struct endpoint_identifier_list *iter;
1310         struct ast_sip_endpoint *endpoint = NULL;
1311         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1312         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1313                 ast_assert(iter->identifier->identify_endpoint != NULL);
1314                 endpoint = iter->identifier->identify_endpoint(rdata);
1315                 if (endpoint) {
1316                         break;
1317                 }
1318         }
1319         return endpoint;
1320 }
1321
1322 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1323
1324 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1325 {
1326         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1327         AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1328         ast_module_ref(ast_module_info->self);
1329         return 0;
1330 }
1331
1332 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1333 {
1334         struct ast_sip_endpoint_formatter *i;
1335         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1336         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1337                 if (i == obj) {
1338                         AST_RWLIST_REMOVE_CURRENT(next);
1339                         ast_module_unref(ast_module_info->self);
1340                         break;
1341                 }
1342         }
1343         AST_RWLIST_TRAVERSE_SAFE_END;
1344 }
1345
1346 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1347                                 struct ast_sip_ami *ami, int *count)
1348 {
1349         int res = 0;
1350         struct ast_sip_endpoint_formatter *i;
1351         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1352         *count = 0;
1353         AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1354                 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1355                         return res;
1356                 }
1357
1358                 if (!res) {
1359                         (*count)++;
1360                 }
1361         }
1362         return 0;
1363 }
1364
1365 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1366 {
1367         return ast_pjsip_endpoint;
1368 }
1369
1370 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)
1371 {
1372         pj_str_t tmp, local_addr;
1373         pjsip_uri *uri;
1374         pjsip_sip_uri *sip_uri;
1375         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1376         int local_port;
1377         char uuid_str[AST_UUID_STR_LEN];
1378
1379         if (ast_strlen_zero(user)) {
1380                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1381                 if (!uuid) {
1382                         return -1;
1383                 }
1384                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1385         }
1386
1387         /* Parse the provided target URI so we can determine what transport it will end up using */
1388         pj_strdup_with_null(pool, &tmp, target);
1389
1390         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1391             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1392                 return -1;
1393         }
1394
1395         sip_uri = pjsip_uri_get_uri(uri);
1396
1397         /* Determine the transport type to use */
1398         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1399                 type = PJSIP_TRANSPORT_TLS;
1400         } else if (!sip_uri->transport_param.slen) {
1401                 type = PJSIP_TRANSPORT_UDP;
1402         } else {
1403                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1404         }
1405
1406         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1407                 return -1;
1408         }
1409
1410         /* If the host is IPv6 turn the transport into an IPv6 version */
1411         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1412                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1413         }
1414
1415         if (!ast_strlen_zero(domain)) {
1416                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1417                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1418                                 "<sip:%s@%s%s%s>",
1419                                 user,
1420                                 domain,
1421                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1422                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1423                 return 0;
1424         }
1425
1426         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1427         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1428                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1429
1430                 /* If no local address can be retrieved using the transport manager use the host one */
1431                 pj_strdup(pool, &local_addr, pj_gethostname());
1432                 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1433         }
1434
1435         /* If IPv6 was specified in the transport, set the proper type */
1436         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1437                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1438         }
1439
1440         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1441         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1442                                       "<sip:%s@%s%.*s%s:%d%s%s>",
1443                                       user,
1444                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1445                                       (int)local_addr.slen,
1446                                       local_addr.ptr,
1447                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1448                                       local_port,
1449                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1450                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1451
1452         return 0;
1453 }
1454
1455 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1456 {
1457         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1458         const char *transport_name = endpoint->transport;
1459
1460         if (ast_strlen_zero(transport_name)) {
1461                 return 0;
1462         }
1463
1464         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1465
1466         if (!transport || !transport->state) {
1467                 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1468                         transport_name, ast_sorcery_object_get_id(endpoint));
1469                 return -1;
1470         }
1471
1472         if (transport->state->transport) {
1473                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1474                 selector->u.transport = transport->state->transport;
1475         } else if (transport->state->factory) {
1476                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1477                 selector->u.listener = transport->state->factory;
1478         } else {
1479                 return -1;
1480         }
1481
1482         return 0;
1483 }
1484
1485 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1486 {
1487         char enclosed_uri[PJSIP_MAX_URL_SIZE];
1488         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1489         pjsip_dialog *dlg = NULL;
1490         const char *outbound_proxy = endpoint->outbound_proxy;
1491         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1492         static const pj_str_t HCONTACT = { "Contact", 7 };
1493
1494         snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1495         pj_cstr(&remote_uri, enclosed_uri);
1496
1497         pj_cstr(&target_uri, uri);
1498
1499         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1500                 return NULL;
1501         }
1502
1503         if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1504                 pjsip_dlg_terminate(dlg);
1505                 return NULL;
1506         }
1507
1508         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1509                 pjsip_dlg_terminate(dlg);
1510                 return NULL;
1511         }
1512
1513         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1514         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1515         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1516         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1517
1518         /* If a request user has been specified and we are permitted to change it, do so */
1519         if (!ast_strlen_zero(request_user)) {
1520                 pjsip_sip_uri *sip_uri;
1521
1522                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1523                         sip_uri = pjsip_uri_get_uri(dlg->target);
1524                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1525                 }
1526                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1527                         sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1528                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1529                 }
1530         }
1531
1532         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1533         dlg->sess_count++;
1534
1535         pjsip_dlg_set_transport(dlg, &selector);
1536
1537         if (!ast_strlen_zero(outbound_proxy)) {
1538                 pjsip_route_hdr route_set, *route;
1539                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1540                 pj_str_t tmp;
1541
1542                 pj_list_init(&route_set);
1543
1544                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1545                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1546                         dlg->sess_count--;
1547                         pjsip_dlg_terminate(dlg);
1548                         return NULL;
1549                 }
1550                 pj_list_insert_nodes_before(&route_set, route);
1551
1552                 pjsip_dlg_set_route_set(dlg, &route_set);
1553         }
1554
1555         dlg->sess_count--;
1556
1557         return dlg;
1558 }
1559
1560 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1561 {
1562         pjsip_dialog *dlg;
1563         pj_str_t contact;
1564         pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1565         pj_status_t status;
1566
1567         contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1568         contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1569                         "<sip:%s%.*s%s:%d%s%s>",
1570                         (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1571                         (int)rdata->tp_info.transport->local_name.host.slen,
1572                         rdata->tp_info.transport->local_name.host.ptr,
1573                         (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1574                         rdata->tp_info.transport->local_name.port,
1575                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1576                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1577
1578         status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1579         if (status != PJ_SUCCESS) {
1580                 char err[PJ_ERR_MSG_SIZE];
1581
1582                 pj_strerror(status, err, sizeof(err));
1583                 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1584                                 ast_sorcery_object_get_id(endpoint), err);
1585                 return NULL;
1586         }
1587
1588         return dlg;
1589 }
1590
1591 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1592 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1593 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1594
1595 static struct {
1596         const char *method;
1597         const pjsip_method *pmethod;
1598 } methods [] = {
1599         { "INVITE", &pjsip_invite_method },
1600         { "CANCEL", &pjsip_cancel_method },
1601         { "ACK", &pjsip_ack_method },
1602         { "BYE", &pjsip_bye_method },
1603         { "REGISTER", &pjsip_register_method },
1604         { "OPTIONS", &pjsip_options_method },
1605         { "SUBSCRIBE", &pjsip_subscribe_method },
1606         { "NOTIFY", &pjsip_notify_method },
1607         { "PUBLISH", &pjsip_publish_method },
1608         { "INFO", &info_method },
1609         { "MESSAGE", &message_method },
1610 };
1611
1612 static const pjsip_method *get_pjsip_method(const char *method)
1613 {
1614         int i;
1615         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1616                 if (!strcmp(method, methods[i].method)) {
1617                         return methods[i].pmethod;
1618                 }
1619         }
1620         return NULL;
1621 }
1622
1623 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1624 {
1625         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1626                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1627                 return -1;
1628         }
1629
1630         return 0;
1631 }
1632
1633 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata);
1634 static pjsip_module supplement_module = {
1635         .name = { "Out of dialog supplement hook", 29 },
1636         .id = -1,
1637         .priority = PJSIP_MOD_PRIORITY_APPLICATION - 1,
1638         .on_rx_request = supplement_on_rx_request,
1639 };
1640
1641 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1642                 const char *uri, struct ast_sip_contact *provided_contact, pjsip_tx_data **tdata)
1643 {
1644         RAII_VAR(struct ast_sip_contact *, contact, ao2_bump(provided_contact), ao2_cleanup);
1645         pj_str_t remote_uri;
1646         pj_str_t from;
1647         pj_pool_t *pool;
1648         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1649
1650         if (ast_strlen_zero(uri)) {
1651                 if (!endpoint && !contact) {
1652                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1653                         return -1;
1654                 }
1655
1656                 if (!contact) {
1657                         contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1658                 }
1659                 if (!contact || ast_strlen_zero(contact->uri)) {
1660                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1661                                         ast_sorcery_object_get_id(endpoint));
1662                         return -1;
1663                 }
1664
1665                 pj_cstr(&remote_uri, contact->uri);
1666         } else {
1667                 pj_cstr(&remote_uri, uri);
1668         }
1669
1670         if (endpoint) {
1671                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1672                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1673                                 ast_sorcery_object_get_id(endpoint));
1674                         return -1;
1675                 }
1676         }
1677
1678         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1679
1680         if (!pool) {
1681                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1682                 return -1;
1683         }
1684
1685         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1686                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1687                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1688                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1689                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1690                 return -1;
1691         }
1692
1693         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1694                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1695                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1696                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1697                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1698                 return -1;
1699         }
1700
1701         /* If an outbound proxy is specified on the endpoint apply it to this request */
1702         if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
1703                 ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
1704                 ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
1705                         (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1706                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1707                 return -1;
1708         }
1709
1710         ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
1711
1712         /* We can release this pool since request creation copied all the necessary
1713          * data into the outbound request's pool
1714          */
1715         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1716         return 0;
1717 }
1718
1719 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1720                 struct ast_sip_endpoint *endpoint, const char *uri,
1721                 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
1722 {
1723         const pjsip_method *pmethod = get_pjsip_method(method);
1724
1725         if (!pmethod) {
1726                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1727                 return -1;
1728         }
1729
1730         if (dlg) {
1731                 return create_in_dialog_request(pmethod, dlg, tdata);
1732         } else {
1733                 return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
1734         }
1735 }
1736
1737 AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
1738
1739 int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
1740 {
1741         struct ast_sip_supplement *iter;
1742         int inserted = 0;
1743         SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1744
1745         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1746                 if (iter->priority > supplement->priority) {
1747                         AST_RWLIST_INSERT_BEFORE_CURRENT(supplement, next);
1748                         inserted = 1;
1749                         break;
1750                 }
1751         }
1752         AST_RWLIST_TRAVERSE_SAFE_END;
1753
1754         if (!inserted) {
1755                 AST_RWLIST_INSERT_TAIL(&supplements, supplement, next);
1756         }
1757         ast_module_ref(ast_module_info->self);
1758         return 0;
1759 }
1760
1761 void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
1762 {
1763         struct ast_sip_supplement *iter;
1764         SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1765         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1766                 if (supplement == iter) {
1767                         AST_RWLIST_REMOVE_CURRENT(next);
1768                         ast_module_unref(ast_module_info->self);
1769                         break;
1770                 }
1771         }
1772         AST_RWLIST_TRAVERSE_SAFE_END;
1773 }
1774
1775 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1776 {
1777         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1778                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1779                 return -1;
1780         }
1781         return 0;
1782 }
1783
1784 static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
1785 {
1786         pj_str_t method;
1787
1788         if (ast_strlen_zero(supplement_method)) {
1789                 return PJ_TRUE;
1790         }
1791
1792         pj_cstr(&method, supplement_method);
1793
1794         return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
1795 }
1796
1797 /*! \brief Structure to hold information about an outbound request */
1798 struct send_request_data {
1799         struct ast_sip_endpoint *endpoint;              /*! The endpoint associated with this request */
1800         void *token;                                    /*! Information to be provided to the callback upon receipt of a response */
1801         void (*callback)(void *token, pjsip_event *e);  /*! The callback to be called upon receipt of a response */
1802 };
1803
1804 static void send_request_data_destroy(void *obj)
1805 {
1806         struct send_request_data *req_data = obj;
1807         ao2_cleanup(req_data->endpoint);
1808 }
1809
1810 static struct send_request_data *send_request_data_alloc(struct ast_sip_endpoint *endpoint,
1811         void *token, void (*callback)(void *token, pjsip_event *e))
1812 {
1813         struct send_request_data *req_data = ao2_alloc(sizeof(*req_data), send_request_data_destroy);
1814
1815         if (!req_data) {
1816                 return NULL;
1817         }
1818
1819         req_data->endpoint = ao2_bump(endpoint);
1820         req_data->token = token;
1821         req_data->callback = callback;
1822
1823         return req_data;
1824 }
1825
1826 static void send_request_cb(void *token, pjsip_event *e)
1827 {
1828         RAII_VAR(struct send_request_data *, req_data, token, ao2_cleanup);
1829         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1830         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1831         pjsip_tx_data *tdata;
1832         struct ast_sip_supplement *supplement;
1833
1834         AST_RWLIST_RDLOCK(&supplements);
1835         AST_LIST_TRAVERSE(&supplements, supplement, next) {
1836                 if (supplement->incoming_response && does_method_match(&challenge->msg_info.cseq->method.name, supplement->method)) {
1837                         supplement->incoming_response(req_data->endpoint, challenge);
1838                 }
1839         }
1840         AST_RWLIST_UNLOCK(&supplements);
1841
1842         if (tsx->status_code == 401 || tsx->status_code == 407) {
1843                 if (!ast_sip_create_request_with_auth(&req_data->endpoint->outbound_auths, challenge, tsx, &tdata)) {
1844                         pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data->token, req_data->callback);
1845                 }
1846                 return;
1847         }
1848
1849         if (req_data->callback) {
1850                 req_data->callback(req_data->token, e);
1851         }
1852 }
1853
1854 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint,
1855         void *token, void (*callback)(void *token, pjsip_event *e))
1856 {
1857         struct ast_sip_supplement *supplement;
1858         struct send_request_data *req_data = send_request_data_alloc(endpoint, token, callback);
1859         struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
1860
1861         if (!req_data) {
1862                 return -1;
1863         }
1864
1865         AST_RWLIST_RDLOCK(&supplements);
1866         AST_LIST_TRAVERSE(&supplements, supplement, next) {
1867                 if (supplement->outgoing_request && does_method_match(&tdata->msg->line.req.method.name, supplement->method)) {
1868                         supplement->outgoing_request(endpoint, contact, tdata);
1869                 }
1870         }
1871         AST_RWLIST_UNLOCK(&supplements);
1872
1873         ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
1874         ao2_cleanup(contact);
1875
1876         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data, send_request_cb) != PJ_SUCCESS) {
1877                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1878                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1879                                 pj_strbuf(&tdata->msg->line.req.method.name),
1880                                 ast_sorcery_object_get_id(endpoint));
1881                 ao2_cleanup(req_data);
1882                 return -1;
1883         }
1884
1885         return 0;
1886 }
1887
1888 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
1889         struct ast_sip_endpoint *endpoint, void *token,
1890         void (*callback)(void *token, pjsip_event *e))
1891 {
1892         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1893
1894         if (dlg) {
1895                 return send_in_dialog_request(tdata, dlg);
1896         } else {
1897                 return send_out_of_dialog_request(tdata, endpoint, token, callback);
1898         }
1899 }
1900
1901 int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
1902 {
1903         pjsip_route_hdr *route;
1904         static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1905         pj_str_t tmp;
1906
1907         pj_strdup2_with_null(tdata->pool, &tmp, proxy);
1908         if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1909                 return -1;
1910         }
1911
1912         pj_list_insert_nodes_before(&tdata->msg->hdr, (pjsip_hdr*)route);
1913
1914         return 0;
1915 }
1916
1917 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1918 {
1919         pj_str_t hdr_name;
1920         pj_str_t hdr_value;
1921         pjsip_generic_string_hdr *hdr;
1922
1923         pj_cstr(&hdr_name, name);
1924         pj_cstr(&hdr_value, value);
1925
1926         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1927
1928         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1929         return 0;
1930 }
1931
1932 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1933 {
1934         pj_str_t type;
1935         pj_str_t subtype;
1936         pj_str_t body_text;
1937
1938         pj_cstr(&type, body->type);
1939         pj_cstr(&subtype, body->subtype);
1940         pj_cstr(&body_text, body->body_text);
1941
1942         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1943 }
1944
1945 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1946 {
1947         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1948         tdata->msg->body = pjsip_body;
1949         return 0;
1950 }
1951
1952 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1953 {
1954         int i;
1955         /* NULL for type and subtype automatically creates "multipart/mixed" */
1956         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1957
1958         for (i = 0; i < num_bodies; ++i) {
1959                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1960                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1961                 pjsip_multipart_add_part(tdata->pool, body, part);
1962         }
1963
1964         tdata->msg->body = body;
1965         return 0;
1966 }
1967
1968 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1969 {
1970         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1971         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1972
1973         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1974
1975         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1976         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1977         tdata->msg->body->len = combined_size;
1978
1979         return 0;
1980 }
1981
1982 struct ast_taskprocessor *ast_sip_create_serializer(void)
1983 {
1984         struct ast_taskprocessor *serializer;
1985         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1986         char name[AST_UUID_STR_LEN];
1987
1988         if (!uuid) {
1989                 return NULL;
1990         }
1991
1992         ast_uuid_to_str(uuid, name, sizeof(name));
1993
1994         serializer = ast_threadpool_serializer(name, sip_threadpool);
1995         if (!serializer) {
1996                 return NULL;
1997         }
1998         return serializer;
1999 }
2000
2001 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2002 {
2003         if (serializer) {
2004                 return ast_taskprocessor_push(serializer, sip_task, task_data);
2005         } else {
2006                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
2007         }
2008 }
2009
2010 struct sync_task_data {
2011         ast_mutex_t lock;
2012         ast_cond_t cond;
2013         int complete;
2014         int fail;
2015         int (*task)(void *);
2016         void *task_data;
2017 };
2018
2019 static int sync_task(void *data)
2020 {
2021         struct sync_task_data *std = data;
2022         std->fail = std->task(std->task_data);
2023
2024         ast_mutex_lock(&std->lock);
2025         std->complete = 1;
2026         ast_cond_signal(&std->cond);
2027         ast_mutex_unlock(&std->lock);
2028         return std->fail;
2029 }
2030
2031 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2032 {
2033         /* This method is an onion */
2034         struct sync_task_data std;
2035
2036         if (ast_sip_thread_is_servant()) {
2037                 return sip_task(task_data);
2038         }
2039
2040         ast_mutex_init(&std.lock);
2041         ast_cond_init(&std.cond, NULL);
2042         std.fail = std.complete = 0;
2043         std.task = sip_task;
2044         std.task_data = task_data;
2045
2046         if (serializer) {
2047                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
2048                         return -1;
2049                 }
2050         } else {
2051                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
2052                         return -1;
2053                 }
2054         }
2055
2056         ast_mutex_lock(&std.lock);
2057         while (!std.complete) {
2058                 ast_cond_wait(&std.cond, &std.lock);
2059         }
2060         ast_mutex_unlock(&std.lock);
2061
2062         ast_mutex_destroy(&std.lock);
2063         ast_cond_destroy(&std.cond);
2064         return std.fail;
2065 }
2066
2067 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
2068 {
2069         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
2070         memcpy(dest, pj_strbuf(src), chars_to_copy);
2071         dest[chars_to_copy] = '\0';
2072 }
2073
2074 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
2075 {
2076         pjsip_media_type compare;
2077
2078         if (!content_type) {
2079                 return 0;
2080         }
2081
2082         pjsip_media_type_init2(&compare, type, subtype);
2083
2084         return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
2085 }
2086
2087 pj_caching_pool caching_pool;
2088 pj_pool_t *memory_pool;
2089 pj_thread_t *monitor_thread;
2090 static int monitor_continue;
2091
2092 static void *monitor_thread_exec(void *endpt)
2093 {
2094         while (monitor_continue) {
2095                 const pj_time_val delay = {0, 10};
2096                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
2097         }
2098         return NULL;
2099 }
2100
2101 static void stop_monitor_thread(void)
2102 {
2103         monitor_continue = 0;
2104         pj_thread_join(monitor_thread);
2105 }
2106
2107 AST_THREADSTORAGE(pj_thread_storage);
2108 AST_THREADSTORAGE(servant_id_storage);
2109 #define SIP_SERVANT_ID 0x5E2F1D
2110
2111 static void sip_thread_start(void)
2112 {
2113         pj_thread_desc *desc;
2114         pj_thread_t *thread;
2115         uint32_t *servant_id;
2116
2117         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2118         if (!servant_id) {
2119                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
2120                 return;
2121         }
2122         *servant_id = SIP_SERVANT_ID;
2123
2124         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
2125         if (!desc) {
2126                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
2127                 return;
2128         }
2129         pj_bzero(*desc, sizeof(*desc));
2130
2131         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
2132                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
2133         }
2134 }
2135
2136 int ast_sip_thread_is_servant(void)
2137 {
2138         uint32_t *servant_id;
2139
2140         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2141         if (!servant_id) {
2142                 return 0;
2143         }
2144
2145         return *servant_id == SIP_SERVANT_ID;
2146 }
2147
2148 void *ast_sip_dict_get(void *ht, const char *key)
2149 {
2150         unsigned int hval = 0;
2151
2152         if (!ht) {
2153                 return NULL;
2154         }
2155
2156         return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
2157 }
2158
2159 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
2160                        const char *key, void *val)
2161 {
2162         if (!ht) {
2163                 ht = pj_hash_create(pool, 11);
2164         }
2165
2166         pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
2167
2168         return ht;
2169 }
2170
2171 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata)
2172 {
2173         struct ast_sip_supplement *supplement;
2174
2175         if (pjsip_rdata_get_dlg(rdata)) {
2176                 return PJ_FALSE;
2177         }
2178
2179         AST_RWLIST_RDLOCK(&supplements);
2180         AST_LIST_TRAVERSE(&supplements, supplement, next) {
2181                 if (supplement->incoming_request && does_method_match(&rdata->msg_info.msg->line.req.method.name, supplement->method)) {
2182                         supplement->incoming_request(ast_pjsip_rdata_get_endpoint(rdata), rdata);
2183                 }
2184         }
2185         AST_RWLIST_UNLOCK(&supplements);
2186
2187         return PJ_FALSE;
2188 }
2189
2190 int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
2191 {
2192         struct ast_sip_supplement *supplement;
2193         pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
2194         struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
2195
2196         AST_RWLIST_RDLOCK(&supplements);
2197         AST_LIST_TRAVERSE(&supplements, supplement, next) {
2198                 if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
2199                         supplement->outgoing_response(sip_endpoint, contact, tdata);
2200                 }
2201         }
2202         AST_RWLIST_UNLOCK(&supplements);
2203
2204         ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
2205         ao2_cleanup(contact);
2206
2207         return pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
2208 }
2209
2210 int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
2211         struct ast_sip_contact *contact, pjsip_tx_data **tdata)
2212 {
2213         int res = pjsip_endpt_create_response(ast_sip_get_pjsip_endpoint(), rdata, st_code, NULL, tdata);
2214
2215         if (!res) {
2216                 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
2217         }
2218
2219         return res;
2220 }
2221
2222 static void remove_request_headers(pjsip_endpoint *endpt)
2223 {
2224         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
2225         pjsip_hdr *iter = request_headers->next;
2226
2227         while (iter != request_headers) {
2228                 pjsip_hdr *to_erase = iter;
2229                 iter = iter->next;
2230                 pj_list_erase(to_erase);
2231         }
2232 }
2233
2234 static int load_module(void)
2235 {
2236         /* The third parameter is just copied from
2237          * example code from PJLIB. This can be adjusted
2238          * if necessary.
2239          */
2240         pj_status_t status;
2241         struct ast_threadpool_options options;
2242
2243         if (pj_init() != PJ_SUCCESS) {
2244                 return AST_MODULE_LOAD_DECLINE;
2245         }
2246
2247         if (pjlib_util_init() != PJ_SUCCESS) {
2248                 pj_shutdown();
2249                 return AST_MODULE_LOAD_DECLINE;
2250         }
2251
2252         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
2253         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
2254                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
2255                 pj_caching_pool_destroy(&caching_pool);
2256                 return AST_MODULE_LOAD_DECLINE;
2257         }
2258
2259         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2260          * we need to stop PJSIP from doing it automatically
2261          */
2262         remove_request_headers(ast_pjsip_endpoint);
2263
2264         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2265         if (!memory_pool) {
2266                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2267                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2268                 ast_pjsip_endpoint = NULL;
2269                 pj_caching_pool_destroy(&caching_pool);
2270                 return AST_MODULE_LOAD_DECLINE;
2271         }
2272
2273         if (ast_sip_initialize_system()) {
2274                 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2275                 pj_pool_release(memory_pool);
2276                 memory_pool = NULL;
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         sip_get_threadpool_options(&options);
2284         options.thread_start = sip_thread_start;
2285         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2286         if (!sip_threadpool) {
2287                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2288                 pj_pool_release(memory_pool);
2289                 memory_pool = NULL;
2290                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2291                 ast_pjsip_endpoint = NULL;
2292                 pj_caching_pool_destroy(&caching_pool);
2293                 return AST_MODULE_LOAD_DECLINE;
2294         }
2295
2296         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2297         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2298
2299         monitor_continue = 1;
2300         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2301                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2302         if (status != PJ_SUCCESS) {
2303                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2304                 pj_pool_release(memory_pool);
2305                 memory_pool = NULL;
2306                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2307                 ast_pjsip_endpoint = NULL;
2308                 pj_caching_pool_destroy(&caching_pool);
2309                 return AST_MODULE_LOAD_DECLINE;
2310         }
2311
2312         ast_sip_initialize_global_headers();
2313
2314         if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2315                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2316                 ast_sip_destroy_global_headers();
2317                 stop_monitor_thread();
2318                 pj_pool_release(memory_pool);
2319                 memory_pool = NULL;
2320                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2321                 ast_pjsip_endpoint = NULL;
2322                 pj_caching_pool_destroy(&caching_pool);
2323                 return AST_MODULE_LOAD_DECLINE;
2324         }
2325
2326         if (ast_sip_initialize_distributor()) {
2327                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2328                 ast_res_pjsip_destroy_configuration();
2329                 ast_sip_destroy_global_headers();
2330                 stop_monitor_thread();
2331                 pj_pool_release(memory_pool);
2332                 memory_pool = NULL;
2333                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2334                 ast_pjsip_endpoint = NULL;
2335                 pj_caching_pool_destroy(&caching_pool);
2336                 return AST_MODULE_LOAD_DECLINE;
2337         }
2338
2339         if (ast_sip_register_service(&supplement_module)) {
2340                 ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
2341                 ast_sip_destroy_distributor();
2342                 ast_res_pjsip_destroy_configuration();
2343                 ast_sip_destroy_global_headers();
2344                 stop_monitor_thread();
2345                 pj_pool_release(memory_pool);
2346                 memory_pool = NULL;
2347                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2348                 ast_pjsip_endpoint = NULL;
2349                 pj_caching_pool_destroy(&caching_pool);
2350                 return AST_MODULE_LOAD_DECLINE;
2351         }
2352
2353         if (ast_sip_initialize_outbound_authentication()) {
2354                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2355                 ast_sip_unregister_service(&supplement_module);
2356                 ast_sip_destroy_distributor();
2357                 ast_res_pjsip_destroy_configuration();
2358                 ast_sip_destroy_global_headers();
2359                 stop_monitor_thread();
2360                 pj_pool_release(memory_pool);
2361                 memory_pool = NULL;
2362                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2363                 ast_pjsip_endpoint = NULL;
2364                 pj_caching_pool_destroy(&caching_pool);
2365                 return AST_MODULE_LOAD_DECLINE;
2366         }
2367
2368         ast_res_pjsip_init_options_handling(0);
2369
2370         ast_module_ref(ast_module_info->self);
2371
2372         return AST_MODULE_LOAD_SUCCESS;
2373 }
2374
2375 static int reload_module(void)
2376 {
2377         if (ast_res_pjsip_reload_configuration()) {
2378                 return AST_MODULE_LOAD_DECLINE;
2379         }
2380         ast_res_pjsip_init_options_handling(1);
2381         return 0;
2382 }
2383
2384 static int unload_module(void)
2385 {
2386         /* This will never get called as this module can't be unloaded */
2387         return 0;
2388 }
2389
2390 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2391                 .load = load_module,
2392                 .unload = unload_module,
2393                 .reload = reload_module,
2394                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
2395 );