08c4552429dcce684feeb23871c2ea55925e6b1c
[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>Mailbox(es) to be associated with</synopsis>
270                                 </configOption>
271                                 <configOption name="moh_suggest" default="default">
272                                         <synopsis>Default Music On Hold class</synopsis>
273                                 </configOption>
274                                 <configOption name="outbound_auth">
275                                         <synopsis>Authentication object used for outbound requests</synopsis>
276                                 </configOption>
277                                 <configOption name="outbound_proxy">
278                                         <synopsis>Proxy through which to send requests, a full SIP URI must be provided</synopsis>
279                                 </configOption>
280                                 <configOption name="rewrite_contact">
281                                         <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
282                                         <description><para>
283                                                 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
284                                                 source IP address and port. This option does not affect outbound messages send to this
285                                                 endpoint.
286                                         </para></description>
287                                 </configOption>
288                                 <configOption name="rtp_ipv6" default="no">
289                                         <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
290                                 </configOption>
291                                 <configOption name="rtp_symmetric" default="no">
292                                         <synopsis>Enforce that RTP must be symmetric</synopsis>
293                                 </configOption>
294                                 <configOption name="send_diversion" default="yes">
295                                         <synopsis>Send the Diversion header, conveying the diversion
296                                         information to the called user agent</synopsis>
297                                 </configOption>
298                                 <configOption name="send_pai" default="no">
299                                         <synopsis>Send the P-Asserted-Identity header</synopsis>
300                                 </configOption>
301                                 <configOption name="send_rpid" default="no">
302                                         <synopsis>Send the Remote-Party-ID header</synopsis>
303                                 </configOption>
304                                 <configOption name="timers_min_se" default="90">
305                                         <synopsis>Minimum session timers expiration period</synopsis>
306                                         <description><para>
307                                                 Minimium session timer expiration period. Time in seconds.
308                                         </para></description>
309                                 </configOption>
310                                 <configOption name="timers" default="yes">
311                                         <synopsis>Session timers for SIP packets</synopsis>
312                                         <description>
313                                                 <enumlist>
314                                                         <enum name="forced" />
315                                                         <enum name="no" />
316                                                         <enum name="required" />
317                                                         <enum name="yes" />
318                                                 </enumlist>
319                                         </description>
320                                 </configOption>
321                                 <configOption name="timers_sess_expires" default="1800">
322                                         <synopsis>Maximum session timer expiration period</synopsis>
323                                         <description><para>
324                                                 Maximium session timer expiration period. Time in seconds.
325                                         </para></description>
326                                 </configOption>
327                                 <configOption name="transport">
328                                         <synopsis>Desired transport configuration</synopsis>
329                                         <description><para>
330                                                 This will set the desired transport configuration to send SIP data through.
331                                                 </para>
332                                                 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
333                                                 to the first configured transport in <filename>pjsip.conf</filename> which is
334                                                 valid for the URI we are trying to contact.
335                                                 </para></warning>
336                                                 <warning><para>Transport configuration is not affected by reloads. In order to
337                                                 change transports, a full Asterisk restart is required</para></warning>
338                                         </description>
339                                 </configOption>
340                                 <configOption name="trust_id_inbound" default="no">
341                                         <synopsis>Accept identification information received from this endpoint</synopsis>
342                                         <description><para>This option determines whether Asterisk will accept
343                                         identification from the endpoint from headers such as P-Asserted-Identity
344                                         or Remote-Party-ID header. This option applies both to calls originating from the
345                                         endpoint and calls originating from Asterisk. If <literal>no</literal>, the
346                                         configured Caller-ID from pjsip.conf will always be used as the identity for
347                                         the endpoint.</para></description>
348                                 </configOption>
349                                 <configOption name="trust_id_outbound" default="no">
350                                         <synopsis>Send private identification details to the endpoint.</synopsis>
351                                         <description><para>This option determines whether res_pjsip will send private
352                                         identification information to the endpoint. If <literal>no</literal>,
353                                         private Caller-ID information will not be forwarded to the endpoint.
354                                         "Private" in this case refers to any method of restricting identification.
355                                         Example: setting <replaceable>callerid_privacy</replaceable> to any
356                                         <literal>prohib</literal> variation.
357                                         Example: If <replaceable>trust_id_inbound</replaceable> is set to
358                                         <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
359                                         header in a SIP request or response would indicate the identification
360                                         provided in the request is private.</para></description>
361                                 </configOption>
362                                 <configOption name="type">
363                                         <synopsis>Must be of type 'endpoint'.</synopsis>
364                                 </configOption>
365                                 <configOption name="use_ptime" default="no">
366                                         <synopsis>Use Endpoint's requested packetisation interval</synopsis>
367                                 </configOption>
368                                 <configOption name="use_avpf" default="no">
369                                         <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
370                                         endpoint.</synopsis>
371                                         <description><para>
372                                                 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
373                                                 profile for all media offers on outbound calls and media updates and will
374                                                 decline media offers not using the AVPF or SAVPF profile.
375                                         </para><para>
376                                                 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
377                                                 profile for all media offers on outbound calls and media updates and will
378                                                 decline media offers not using the AVP or SAVP profile.
379                                         </para></description>
380                                 </configOption>
381                                 <configOption name="media_encryption" default="no">
382                                         <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
383                                         for this endpoint.</synopsis>
384                                         <description>
385                                                 <enumlist>
386                                                         <enum name="no"><para>
387                                                                 res_pjsip will offer no encryption and allow no encryption to be setup.
388                                                         </para></enum>
389                                                         <enum name="sdes"><para>
390                                                                 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
391                                                                 transport should be used in conjunction with this option to prevent
392                                                                 exposure of media encryption keys.
393                                                         </para></enum>
394                                                         <enum name="dtls"><para>
395                                                                 res_pjsip will offer DTLS-SRTP setup.
396                                                         </para></enum>
397                                                 </enumlist>
398                                         </description>
399                                 </configOption>
400                                 <configOption name="inband_progress" default="no">
401                                         <synopsis>Determines whether chan_pjsip will indicate ringing using inband
402                                             progress.</synopsis>
403                                         <description><para>
404                                                 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
405                                                 when told to indicate ringing and will immediately start sending ringing
406                                                 as audio.
407                                         </para><para>
408                                                 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
409                                                 to indicate ringing and will NOT send it as audio.
410                                         </para></description>
411                                 </configOption>
412                                 <configOption name="call_group">
413                                         <synopsis>The numeric pickup groups for a channel.</synopsis>
414                                         <description><para>
415                                                 Can be set to a comma separated list of numbers or ranges between the values
416                                                 of 0-63 (maximum of 64 groups).
417                                         </para></description>
418                                 </configOption>
419                                 <configOption name="pickup_group">
420                                         <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
421                                         <description><para>
422                                                 Can be set to a comma separated list of numbers or ranges between the values
423                                                 of 0-63 (maximum of 64 groups).
424                                         </para></description>
425                                 </configOption>
426                                 <configOption name="named_call_group">
427                                         <synopsis>The named pickup groups for a channel.</synopsis>
428                                         <description><para>
429                                                 Can be set to a comma separated list of case sensitive strings limited by
430                                                 supported line length.
431                                         </para></description>
432                                 </configOption>
433                                 <configOption name="named_pickup_group">
434                                         <synopsis>The named pickup groups that a channel can pickup.</synopsis>
435                                         <description><para>
436                                                 Can be set to a comma separated list of case sensitive strings limited by
437                                                 supported line length.
438                                         </para></description>
439                                 </configOption>
440                                 <configOption name="device_state_busy_at" default="0">
441                                         <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
442                                         <description><para>
443                                                 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
444                                                 PJSIP channel driver will return busy as the device state instead of in use.
445                                         </para></description>
446                                 </configOption>
447                                 <configOption name="t38_udptl" default="no">
448                                         <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
449                                         <description><para>
450                                                 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
451                                                 and relayed.
452                                         </para></description>
453                                 </configOption>
454                                 <configOption name="t38_udptl_ec" default="none">
455                                         <synopsis>T.38 UDPTL error correction method</synopsis>
456                                         <description>
457                                                 <enumlist>
458                                                         <enum name="none"><para>
459                                                                 No error correction should be used.
460                                                         </para></enum>
461                                                         <enum name="fec"><para>
462                                                                 Forward error correction should be used.
463                                                         </para></enum>
464                                                         <enum name="redundancy"><para>
465                                                                 Redundacy error correction should be used.
466                                                         </para></enum>
467                                                 </enumlist>
468                                         </description>
469                                 </configOption>
470                                 <configOption name="t38_udptl_maxdatagram" default="0">
471                                         <synopsis>T.38 UDPTL maximum datagram size</synopsis>
472                                         <description><para>
473                                                 This option can be set to override the maximum datagram of a remote endpoint for broken
474                                                 endpoints.
475                                         </para></description>
476                                 </configOption>
477                                 <configOption name="fax_detect" default="no">
478                                         <synopsis>Whether CNG tone detection is enabled</synopsis>
479                                         <description><para>
480                                                 This option can be set to send the session to the fax extension when a CNG tone is
481                                                 detected.
482                                         </para></description>
483                                 </configOption>
484                                 <configOption name="t38_udptl_nat" default="no">
485                                         <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
486                                         <description><para>
487                                                 When enabled the UDPTL stack will send UDPTL packets to the source address of
488                                                 received packets.
489                                         </para></description>
490                                 </configOption>
491                                 <configOption name="t38_udptl_ipv6" default="no">
492                                         <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
493                                         <description><para>
494                                                 When enabled the UDPTL stack will use IPv6.
495                                         </para></description>
496                                 </configOption>
497                                 <configOption name="tone_zone">
498                                         <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
499                                 </configOption>
500                                 <configOption name="language">
501                                         <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
502                                 </configOption>
503                                 <configOption name="one_touch_recording" default="no">
504                                         <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
505                                         <see-also>
506                                                 <ref type="configOption">recordonfeature</ref>
507                                                 <ref type="configOption">recordofffeature</ref>
508                                         </see-also>
509                                 </configOption>
510                                 <configOption name="record_on_feature" default="automixmon">
511                                         <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
512                                         <description>
513                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
514                                                 feature will be enabled for the channel. The feature designated here can be any built-in
515                                                 or dynamic feature defined in features.conf.</para>
516                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
517                                         </description>
518                                         <see-also>
519                                                 <ref type="configOption">one_touch_recording</ref>
520                                                 <ref type="configOption">recordofffeature</ref>
521                                         </see-also>
522                                 </configOption>
523                                 <configOption name="record_off_feature" default="automixmon">
524                                         <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
525                                         <description>
526                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
527                                                 feature will be enabled for the channel. The feature designated here can be any built-in
528                                                 or dynamic feature defined in features.conf.</para>
529                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
530                                         </description>
531                                         <see-also>
532                                                 <ref type="configOption">one_touch_recording</ref>
533                                                 <ref type="configOption">recordonfeature</ref>
534                                         </see-also>
535                                 </configOption>
536                                 <configOption name="rtp_engine" default="asterisk">
537                                         <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
538                                 </configOption>
539                                 <configOption name="allow_transfer" default="yes">
540                                         <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
541                                 </configOption>
542                                 <configOption name="sdp_owner" default="-">
543                                         <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
544                                 </configOption>
545                                 <configOption name="sdp_session" default="Asterisk">
546                                         <synopsis>String used for the SDP session (s=) line.</synopsis>
547                                 </configOption>
548                                 <configOption name="tos_audio">
549                                         <synopsis>DSCP TOS bits for audio streams</synopsis>
550                                         <description><para>
551                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
552                                         </para></description>
553                                 </configOption>
554                                 <configOption name="tos_video">
555                                         <synopsis>DSCP TOS bits for video streams</synopsis>
556                                         <description><para>
557                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
558                                         </para></description>
559                                 </configOption>
560                                 <configOption name="cos_audio">
561                                         <synopsis>Priority for audio streams</synopsis>
562                                         <description><para>
563                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
564                                         </para></description>
565                                 </configOption>
566                                 <configOption name="cos_video">
567                                         <synopsis>Priority for video streams</synopsis>
568                                         <description><para>
569                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
570                                         </para></description>
571                                 </configOption>
572                                 <configOption name="allow_subscribe" default="yes">
573                                         <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
574                                 </configOption>
575                                 <configOption name="sub_min_expiry" default="60">
576                                         <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
577                                 </configOption>
578                                 <configOption name="from_user">
579                                         <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
580                                 </configOption>
581                                 <configOption name="mwi_from_user">
582                                         <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
583                                 </configOption>
584                                 <configOption name="from_domain">
585                                         <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
586                                 </configOption>
587                                 <configOption name="dtls_verify">
588                                         <synopsis>Verify that the provided peer certificate is valid</synopsis>
589                                         <description><para>
590                                                 This option only applies if <replaceable>media_encryption</replaceable> is
591                                                 set to <literal>dtls</literal>.
592                                         </para></description>
593                                 </configOption>
594                                 <configOption name="dtls_rekey">
595                                         <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
596                                         <description><para>
597                                                 This option only applies if <replaceable>media_encryption</replaceable> is
598                                                 set to <literal>dtls</literal>.
599                                         </para><para>
600                                                 If this is not set or the value provided is 0 rekeying will be disabled.
601                                         </para></description>
602                                 </configOption>
603                                 <configOption name="dtls_cert_file">
604                                         <synopsis>Path to certificate file to present to peer</synopsis>
605                                         <description><para>
606                                                 This option only applies if <replaceable>media_encryption</replaceable> is
607                                                 set to <literal>dtls</literal>.
608                                         </para></description>
609                                 </configOption>
610                                 <configOption name="dtls_private_key">
611                                         <synopsis>Path to private key for certificate file</synopsis>
612                                         <description><para>
613                                                 This option only applies if <replaceable>media_encryption</replaceable> is
614                                                 set to <literal>dtls</literal>.
615                                         </para></description>
616                                 </configOption>
617                                 <configOption name="dtls_cipher">
618                                         <synopsis>Cipher to use for DTLS negotiation</synopsis>
619                                         <description><para>
620                                                 This option only applies if <replaceable>media_encryption</replaceable> is
621                                                 set to <literal>dtls</literal>.
622                                         </para><para>
623                                                 Many options for acceptable ciphers. See link for more:
624                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
625                                         </para></description>
626                                 </configOption>
627                                 <configOption name="dtls_ca_file">
628                                         <synopsis>Path to certificate authority certificate</synopsis>
629                                         <description><para>
630                                                 This option only applies if <replaceable>media_encryption</replaceable> is
631                                                 set to <literal>dtls</literal>.
632                                         </para></description>
633                                 </configOption>
634                                 <configOption name="dtls_ca_path">
635                                         <synopsis>Path to a directory containing certificate authority certificates</synopsis>
636                                         <description><para>
637                                                 This option only applies if <replaceable>media_encryption</replaceable> is
638                                                 set to <literal>dtls</literal>.
639                                         </para></description>
640                                 </configOption>
641                                 <configOption name="dtls_setup">
642                                         <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
643                                         <description>
644                                                 <para>
645                                                         This option only applies if <replaceable>media_encryption</replaceable> is
646                                                         set to <literal>dtls</literal>.
647                                                 </para>
648                                                 <enumlist>
649                                                         <enum name="active"><para>
650                                                                 res_pjsip will make a connection to the peer.
651                                                         </para></enum>
652                                                         <enum name="passive"><para>
653                                                                 res_pjsip will accept connections from the peer.
654                                                         </para></enum>
655                                                         <enum name="actpass"><para>
656                                                                 res_pjsip will offer and accept connections from the peer.
657                                                         </para></enum>
658                                                 </enumlist>
659                                         </description>
660                                 </configOption>
661                                 <configOption name="srtp_tag_32">
662                                         <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
663                                         <description><para>
664                                                 This option only applies if <replaceable>media_encryption</replaceable> is
665                                                 set to <literal>sdes</literal> or <literal>dtls</literal>.
666                                         </para></description>
667                                 </configOption>
668                         </configObject>
669                         <configObject name="auth">
670                                 <synopsis>Authentication type</synopsis>
671                                 <description><para>
672                                         Authentication objects hold the authentication information for use
673                                         by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
674                                         This also allows for multiple objects to use a single auth object. See
675                                         the <literal>auth_type</literal> config option for password style choices.
676                                 </para></description>
677                                 <configOption name="auth_type" default="userpass">
678                                         <synopsis>Authentication type</synopsis>
679                                         <description><para>
680                                                 This option specifies which of the password style config options should be read
681                                                 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
682                                                 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
683                                                 from 'md5_cred'.
684                                                 </para>
685                                                 <enumlist>
686                                                         <enum name="md5"/>
687                                                         <enum name="userpass"/>
688                                                 </enumlist>
689                                         </description>
690                                 </configOption>
691                                 <configOption name="nonce_lifetime" default="32">
692                                         <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
693                                 </configOption>
694                                 <configOption name="md5_cred">
695                                         <synopsis>MD5 Hash used for authentication.</synopsis>
696                                         <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
697                                 </configOption>
698                                 <configOption name="password">
699                                         <synopsis>PlainText password used for authentication.</synopsis>
700                                         <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
701                                 </configOption>
702                                 <configOption name="realm" default="asterisk">
703                                         <synopsis>SIP realm for endpoint</synopsis>
704                                 </configOption>
705                                 <configOption name="type">
706                                         <synopsis>Must be 'auth'</synopsis>
707                                 </configOption>
708                                 <configOption name="username">
709                                         <synopsis>Username to use for account</synopsis>
710                                 </configOption>
711                         </configObject>
712                         <configObject name="domain_alias">
713                                 <synopsis>Domain Alias</synopsis>
714                                 <description><para>
715                                         Signifies that a domain is an alias. If the domain on a session is
716                                         not found to match an AoR then this object is used to see if we have
717                                         an alias for the AoR to which the endpoint is binding. This objects
718                                         name as defined in configuration should be the domain alias and a
719                                         config option is provided to specify the domain to be aliased.
720                                 </para></description>
721                                 <configOption name="type">
722                                         <synopsis>Must be of type 'domain_alias'.</synopsis>
723                                 </configOption>
724                                 <configOption name="domain">
725                                         <synopsis>Domain to be aliased</synopsis>
726                                 </configOption>
727                         </configObject>
728                         <configObject name="transport">
729                                 <synopsis>SIP Transport</synopsis>
730                                 <description><para>
731                                         <emphasis>Transports</emphasis>
732                                         </para>
733                                         <para>There are different transports and protocol derivatives
734                                                 supported by <literal>res_pjsip</literal>. They are in order of
735                                                 preference: UDP, TCP, and WebSocket (WS).</para>
736                                         <note><para>Changes to transport configuration in pjsip.conf will only be
737                                                 effected on a complete restart of Asterisk. A module reload
738                                                 will not suffice.</para></note>
739                                 </description>
740                                 <configOption name="async_operations" default="1">
741                                         <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
742                                 </configOption>
743                                 <configOption name="bind">
744                                         <synopsis>IP Address and optional port to bind to for this transport</synopsis>
745                                 </configOption>
746                                 <configOption name="ca_list_file">
747                                         <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
748                                 </configOption>
749                                 <configOption name="cert_file">
750                                         <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
751                                 </configOption>
752                                 <configOption name="cipher">
753                                         <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
754                                         <description><para>
755                                                 Many options for acceptable ciphers see link for more:
756                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
757                                         </para></description>
758                                 </configOption>
759                                 <configOption name="domain">
760                                         <synopsis>Domain the transport comes from</synopsis>
761                                 </configOption>
762                                 <configOption name="external_media_address">
763                                         <synopsis>External IP address to use in RTP handling</synopsis>
764                                         <description><para>
765                                                 When a request or response is sent out, if the destination of the
766                                                 message is outside the IP network defined in the option <literal>localnet</literal>,
767                                                 and the media address in the SDP is within the localnet network, then the
768                                                 media address in the SDP will be rewritten to the value defined for
769                                                 <literal>external_media_address</literal>.
770                                         </para></description>
771                                 </configOption>
772                                 <configOption name="external_signaling_address">
773                                         <synopsis>External address for SIP signalling</synopsis>
774                                 </configOption>
775                                 <configOption name="external_signaling_port" default="0">
776                                         <synopsis>External port for SIP signalling</synopsis>
777                                 </configOption>
778                                 <configOption name="method">
779                                         <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
780                                         <description>
781                                                 <enumlist>
782                                                         <enum name="default" />
783                                                         <enum name="unspecified" />
784                                                         <enum name="tlsv1" />
785                                                         <enum name="sslv2" />
786                                                         <enum name="sslv3" />
787                                                         <enum name="sslv23" />
788                                                 </enumlist>
789                                         </description>
790                                 </configOption>
791                                 <configOption name="local_net">
792                                         <synopsis>Network to consider local (used for NAT purposes).</synopsis>
793                                         <description><para>This must be in CIDR or dotted decimal format with the IP
794                                         and mask separated with a slash ('/').</para></description>
795                                 </configOption>
796                                 <configOption name="password">
797                                         <synopsis>Password required for transport</synopsis>
798                                 </configOption>
799                                 <configOption name="priv_key_file">
800                                         <synopsis>Private key file (TLS ONLY)</synopsis>
801                                 </configOption>
802                                 <configOption name="protocol" default="udp">
803                                         <synopsis>Protocol to use for SIP traffic</synopsis>
804                                         <description>
805                                                 <enumlist>
806                                                         <enum name="udp" />
807                                                         <enum name="tcp" />
808                                                         <enum name="tls" />
809                                                         <enum name="ws" />
810                                                         <enum name="wss" />
811                                                 </enumlist>
812                                         </description>
813                                 </configOption>
814                                 <configOption name="require_client_cert" default="false">
815                                         <synopsis>Require client certificate (TLS ONLY)</synopsis>
816                                 </configOption>
817                                 <configOption name="type">
818                                         <synopsis>Must be of type 'transport'.</synopsis>
819                                 </configOption>
820                                 <configOption name="verify_client" default="false">
821                                         <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
822                                 </configOption>
823                                 <configOption name="verify_server" default="false">
824                                         <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
825                                 </configOption>
826                                 <configOption name="tos" default="false">
827                                         <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
828                                         <description>
829                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
830                                         for more information on this parameter.</para>
831                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
832                                         or the <replaceable>wss</replaceable> protocols.</para></note>
833                                         </description>
834                                 </configOption>
835                                 <configOption name="cos" default="false">
836                                         <synopsis>Enable COS for the signalling sent over this transport</synopsis>
837                                         <description>
838                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
839                                         for more information on this parameter.</para>
840                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
841                                         or the <replaceable>wss</replaceable> protocols.</para></note>
842                                         </description>
843                                 </configOption>
844                         </configObject>
845                         <configObject name="contact">
846                                 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
847                                 <description><para>
848                                         Contacts are a way to hide SIP URIs from the dialplan directly.
849                                         They are also used to make a group of contactable parties when
850                                         in use with <literal>AoR</literal> lists.
851                                 </para></description>
852                                 <configOption name="type">
853                                         <synopsis>Must be of type 'contact'.</synopsis>
854                                 </configOption>
855                                 <configOption name="uri">
856                                         <synopsis>SIP URI to contact peer</synopsis>
857                                 </configOption>
858                                 <configOption name="expiration_time">
859                                         <synopsis>Time to keep alive a contact</synopsis>
860                                         <description><para>
861                                                 Time to keep alive a contact. String style specification.
862                                         </para></description>
863                                 </configOption>
864                                 <configOption name="qualify_frequency" default="0">
865                                         <synopsis>Interval at which to qualify a contact</synopsis>
866                                         <description><para>
867                                                 Interval between attempts to qualify the contact for reachability.
868                                                 If <literal>0</literal> never qualify. Time in seconds.
869                                         </para></description>
870                                 </configOption>
871                         </configObject>
872                         <configObject name="aor">
873                                 <synopsis>The configuration for a location of an endpoint</synopsis>
874                                 <description><para>
875                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
876                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
877                                         Beyond that, an AoR has other uses within Asterisk, such as inbound
878                                         registration.
879                                         </para><para>
880                                         An <literal>AoR</literal> is a way to allow dialing a group
881                                         of <literal>Contacts</literal> that all use the same
882                                         <literal>endpoint</literal> for calls.
883                                         </para><para>
884                                         This can be used as another way of grouping a list of contacts to dial
885                                         rather than specifing them each directly when dialing via the dialplan.
886                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
887                                         </para><para>
888                                         Registrations: For Asterisk to match an inbound registration to an endpoint,
889                                         the AoR object name must match the user portion of the SIP URI in the "To:"
890                                         header of the inbound SIP registration. That will usually be equivalent
891                                         to the "user name" set in your hard or soft phones configuration.
892                                 </para></description>
893                                 <configOption name="contact">
894                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
895                                         <description><para>
896                                                 Contacts specified will be called whenever referenced
897                                                 by <literal>chan_pjsip</literal>.
898                                                 </para><para>
899                                                 Use a separate "contact=" entry for each contact required. Contacts
900                                                 are specified using a SIP URI.
901                                         </para></description>
902                                 </configOption>
903                                 <configOption name="default_expiration" default="3600">
904                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
905                                 </configOption>
906                                 <configOption name="mailboxes">
907                                         <synopsis>Mailbox(es) to be associated with</synopsis>
908                                         <description><para>This option applies when an external entity subscribes to an AoR
909                                         for message waiting indications. The mailboxes specified will be subscribed to.
910                                         More than one mailbox can be specified with a comma-delimited string.</para></description>
911                                 </configOption>
912                                 <configOption name="maximum_expiration" default="7200">
913                                         <synopsis>Maximum time to keep an AoR</synopsis>
914                                         <description><para>
915                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
916                                         </para></description>
917                                 </configOption>
918                                 <configOption name="max_contacts" default="0">
919                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
920                                         <description><para>
921                                                 Maximum number of contacts that can associate with this AoR. This value does
922                                                 not affect the number of contacts that can be added with the "contact" option.
923                                                 It only limits contacts added through external interaction, such as
924                                                 registration.
925                                                 </para>
926                                                 <note><para>This should be set to <literal>1</literal> and
927                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
928                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
929                                                 </para></note>
930                                         </description>
931                                 </configOption>
932                                 <configOption name="minimum_expiration" default="60">
933                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
934                                         <description><para>
935                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
936                                         </para></description>
937                                 </configOption>
938                                 <configOption name="remove_existing" default="no">
939                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
940                                         <description><para>
941                                                 On receiving a new registration to the AoR should it remove
942                                                 the existing contact that was registered against it?
943                                                 </para>
944                                                 <note><para>This should be set to <literal>yes</literal> and
945                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
946                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
947                                                 </para></note>
948                                         </description>
949                                 </configOption>
950                                 <configOption name="type">
951                                         <synopsis>Must be of type 'aor'.</synopsis>
952                                 </configOption>
953                                 <configOption name="qualify_frequency" default="0">
954                                         <synopsis>Interval at which to qualify an AoR</synopsis>
955                                         <description><para>
956                                                 Interval between attempts to qualify the AoR for reachability.
957                                                 If <literal>0</literal> never qualify. Time in seconds.
958                                         </para></description>
959                                 </configOption>
960                                 <configOption name="authenticate_qualify" default="no">
961                                         <synopsis>Authenticates a qualify request if needed</synopsis>
962                                         <description><para>
963                                                 If true and a qualify request receives a challenge or authenticate response
964                                                 authentication is attempted before declaring the contact available.
965                                         </para></description>
966                                 </configOption>
967                         </configObject>
968                         <configObject name="system">
969                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
970                                 <description><para>
971                                         The settings in this section are global. In addition to being global, the values will
972                                         not be re-evaluated when a reload is performed. This is because the values must be set
973                                         before the SIP stack is initialized. The only way to reset these values is to either
974                                         restart Asterisk, or unload res_pjsip.so and then load it again.
975                                 </para></description>
976                                 <configOption name="timer_t1" default="500">
977                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
978                                         <description><para>
979                                                 Timer T1 is the base for determining how long to wait before retransmitting
980                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
981                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
982                                         </para></description>
983                                 </configOption>
984                                 <configOption name="timer_b" default="32000">
985                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
986                                         <description><para>
987                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
988                                                 request before terminating the transaction. It is recommended that this be set
989                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
990                                                 this timer, see RFC 3261, Section 17.1.1.1.
991                                         </para></description>
992                                 </configOption>
993                                 <configOption name="compact_headers" default="no">
994                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
995                                 </configOption>
996                                 <configOption name="threadpool_initial_size" default="0">
997                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
998                                 </configOption>
999                                 <configOption name="threadpool_auto_increment" default="5">
1000                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1001                                 </configOption>
1002                                 <configOption name="threadpool_idle_timeout" default="60">
1003                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1004                                 </configOption>
1005                                 <configOption name="threadpool_max_size" default="0">
1006                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
1007                                         A value of 0 indicates no maximum.</synopsis>
1008                                 </configOption>
1009                                 <configOption name="type">
1010                                         <synopsis>Must be of type 'system'.</synopsis>
1011                                 </configOption>
1012                         </configObject>
1013                         <configObject name="global">
1014                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
1015                                 <description><para>
1016                                         The settings in this section are global. Unlike options in the <literal>system</literal>
1017                                         section, these options can be refreshed by performing a reload.
1018                                 </para></description>
1019                                 <configOption name="max_forwards" default="70">
1020                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1021                                 </configOption>
1022                                 <configOption name="type">
1023                                         <synopsis>Must be of type 'global'.</synopsis>
1024                                 </configOption>
1025                                 <configOption name="user_agent" default="Asterisk &lt;Asterisk Version&gt;">
1026                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1027                                 </configOption>
1028                                 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1029                                         <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1030                                 </configOption>
1031
1032                         </configObject>
1033                 </configFile>
1034         </configInfo>
1035         <manager name="PJSIPQualify" language="en_US">
1036                 <synopsis>
1037                         Qualify a chan_pjsip endpoint.
1038                 </synopsis>
1039                 <syntax>
1040                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1041                         <parameter name="Endpoint" required="true">
1042                                 <para>The endpoint you want to qualify.</para>
1043                         </parameter>
1044                 </syntax>
1045                 <description>
1046                         <para>Qualify a chan_pjsip endpoint.</para>
1047                 </description>
1048         </manager>
1049         <manager name="PJSIPShowEndpoints" language="en_US">
1050                 <synopsis>
1051                         Lists PJSIP endpoints.
1052                 </synopsis>
1053                 <syntax />
1054                 <description>
1055                         <para>
1056                         Provides a listing of all endpoints.  For each endpoint an <literal>EndpointList</literal> event
1057                         is raised that contains relevant attributes and status information.  Once all
1058                         endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1059                         </para>
1060                 </description>
1061         </manager>
1062         <manager name="PJSIPShowEndpoint" language="en_US">
1063                 <synopsis>
1064                         Detail listing of an endpoint and its objects.
1065                 </synopsis>
1066                 <syntax>
1067                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1068                         <parameter name="Endpoint" required="true">
1069                                 <para>The endpoint to list.</para>
1070                         </parameter>
1071                 </syntax>
1072                 <description>
1073                         <para>
1074                         Provides a detailed listing of options for a given endpoint.  Events are issued
1075                         showing the configuration and status of the endpoint and associated objects.  These
1076                         events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1077                         <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1078                         <literal>IdentifyDetail</literal>.  Some events may be listed multiple times if multiple objects are
1079                         associated (for instance AoRs).  Once all detail events have been raised a final
1080                         <literal>EndpointDetailComplete</literal> event is issued.
1081                         </para>
1082                 </description>
1083         </manager>
1084  ***/
1085
1086
1087 static pjsip_endpoint *ast_pjsip_endpoint;
1088
1089 static struct ast_threadpool *sip_threadpool;
1090
1091 static int register_service(void *data)
1092 {
1093         pjsip_module **module = data;
1094         if (!ast_pjsip_endpoint) {
1095                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1096                 return -1;
1097         }
1098         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1099                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1100                 return -1;
1101         }
1102         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1103         ast_module_ref(ast_module_info->self);
1104         return 0;
1105 }
1106
1107 int ast_sip_register_service(pjsip_module *module)
1108 {
1109         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1110 }
1111
1112 static int unregister_service(void *data)
1113 {
1114         pjsip_module **module = data;
1115         ast_module_unref(ast_module_info->self);
1116         if (!ast_pjsip_endpoint) {
1117                 return -1;
1118         }
1119         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1120         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1121         return 0;
1122 }
1123
1124 void ast_sip_unregister_service(pjsip_module *module)
1125 {
1126         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1127 }
1128
1129 static struct ast_sip_authenticator *registered_authenticator;
1130
1131 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1132 {
1133         if (registered_authenticator) {
1134                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1135                 return -1;
1136         }
1137         registered_authenticator = auth;
1138         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1139         ast_module_ref(ast_module_info->self);
1140         return 0;
1141 }
1142
1143 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1144 {
1145         if (registered_authenticator != auth) {
1146                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1147                                 auth, registered_authenticator);
1148                 return;
1149         }
1150         registered_authenticator = NULL;
1151         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1152         ast_module_unref(ast_module_info->self);
1153 }
1154
1155 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1156 {
1157         if (!registered_authenticator) {
1158                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1159                 return 0;
1160         }
1161
1162         return registered_authenticator->requires_authentication(endpoint, rdata);
1163 }
1164
1165 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1166                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1167 {
1168         if (!registered_authenticator) {
1169                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1170                 return 0;
1171         }
1172         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1173 }
1174
1175 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1176
1177 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1178 {
1179         if (registered_outbound_authenticator) {
1180                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1181                 return -1;
1182         }
1183         registered_outbound_authenticator = auth;
1184         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1185         ast_module_ref(ast_module_info->self);
1186         return 0;
1187 }
1188
1189 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1190 {
1191         if (registered_outbound_authenticator != auth) {
1192                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1193                                 auth, registered_outbound_authenticator);
1194                 return;
1195         }
1196         registered_outbound_authenticator = NULL;
1197         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1198         ast_module_unref(ast_module_info->self);
1199 }
1200
1201 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1202                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1203 {
1204         if (!registered_outbound_authenticator) {
1205                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1206                 return -1;
1207         }
1208         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1209 }
1210
1211 struct endpoint_identifier_list {
1212         struct ast_sip_endpoint_identifier *identifier;
1213         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1214 };
1215
1216 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1217
1218 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1219 {
1220         struct endpoint_identifier_list *id_list_item;
1221         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1222
1223         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1224         if (!id_list_item) {
1225                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1226                 return -1;
1227         }
1228         id_list_item->identifier = identifier;
1229
1230         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1231         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1232
1233         ast_module_ref(ast_module_info->self);
1234         return 0;
1235 }
1236
1237 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1238 {
1239         struct endpoint_identifier_list *iter;
1240         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1241         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1242                 if (iter->identifier == identifier) {
1243                         AST_RWLIST_REMOVE_CURRENT(list);
1244                         ast_free(iter);
1245                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1246                         ast_module_unref(ast_module_info->self);
1247                         break;
1248                 }
1249         }
1250         AST_RWLIST_TRAVERSE_SAFE_END;
1251 }
1252
1253 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1254 {
1255         struct endpoint_identifier_list *iter;
1256         struct ast_sip_endpoint *endpoint = NULL;
1257         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1258         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1259                 ast_assert(iter->identifier->identify_endpoint != NULL);
1260                 endpoint = iter->identifier->identify_endpoint(rdata);
1261                 if (endpoint) {
1262                         break;
1263                 }
1264         }
1265         return endpoint;
1266 }
1267
1268 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1269
1270 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1271 {
1272         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1273         AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1274         ast_module_ref(ast_module_info->self);
1275         return 0;
1276 }
1277
1278 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1279 {
1280         struct ast_sip_endpoint_formatter *i;
1281         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1282         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1283                 if (i == obj) {
1284                         AST_RWLIST_REMOVE_CURRENT(next);
1285                         ast_module_unref(ast_module_info->self);
1286                         break;
1287                 }
1288         }
1289         AST_RWLIST_TRAVERSE_SAFE_END;
1290 }
1291
1292 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1293                                 struct ast_sip_ami *ami, int *count)
1294 {
1295         int res = 0;
1296         struct ast_sip_endpoint_formatter *i;
1297         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1298         *count = 0;
1299         AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1300                 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1301                         return res;
1302                 }
1303
1304                 if (!res) {
1305                         (*count)++;
1306                 }
1307         }
1308         return 0;
1309 }
1310
1311 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1312 {
1313         return ast_pjsip_endpoint;
1314 }
1315
1316 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)
1317 {
1318         pj_str_t tmp, local_addr;
1319         pjsip_uri *uri;
1320         pjsip_sip_uri *sip_uri;
1321         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1322         int local_port;
1323         char uuid_str[AST_UUID_STR_LEN];
1324
1325         if (ast_strlen_zero(user)) {
1326                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1327                 if (!uuid) {
1328                         return -1;
1329                 }
1330                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1331         }
1332
1333         /* Parse the provided target URI so we can determine what transport it will end up using */
1334         pj_strdup_with_null(pool, &tmp, target);
1335
1336         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1337             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1338                 return -1;
1339         }
1340
1341         sip_uri = pjsip_uri_get_uri(uri);
1342
1343         /* Determine the transport type to use */
1344         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1345                 type = PJSIP_TRANSPORT_TLS;
1346         } else if (!sip_uri->transport_param.slen) {
1347                 type = PJSIP_TRANSPORT_UDP;
1348         } else {
1349                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1350         }
1351
1352         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1353                 return -1;
1354         }
1355
1356         /* If the host is IPv6 turn the transport into an IPv6 version */
1357         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1358                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1359         }
1360
1361         if (!ast_strlen_zero(domain)) {
1362                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1363                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1364                                 "<%s:%s@%s%s%s>",
1365                                 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1366                                 user,
1367                                 domain,
1368                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1369                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1370                 return 0;
1371         }
1372
1373         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1374         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1375                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1376
1377                 /* If no local address can be retrieved using the transport manager use the host one */
1378                 pj_strdup(pool, &local_addr, pj_gethostname());
1379                 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1380         }
1381
1382         /* If IPv6 was specified in the transport, set the proper type */
1383         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1384                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1385         }
1386
1387         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1388         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1389                                       "<%s:%s@%s%.*s%s:%d%s%s>",
1390                                       (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1391                                       user,
1392                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1393                                       (int)local_addr.slen,
1394                                       local_addr.ptr,
1395                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1396                                       local_port,
1397                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1398                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1399
1400         return 0;
1401 }
1402
1403 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1404 {
1405         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1406         const char *transport_name = endpoint->transport;
1407
1408         if (ast_strlen_zero(transport_name)) {
1409                 return 0;
1410         }
1411
1412         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1413
1414         if (!transport || !transport->state) {
1415                 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1416                         transport_name, ast_sorcery_object_get_id(endpoint));
1417                 return -1;
1418         }
1419
1420         if (transport->state->transport) {
1421                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1422                 selector->u.transport = transport->state->transport;
1423         } else if (transport->state->factory) {
1424                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1425                 selector->u.listener = transport->state->factory;
1426         } else {
1427                 return -1;
1428         }
1429
1430         return 0;
1431 }
1432
1433 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1434 {
1435         char enclosed_uri[PJSIP_MAX_URL_SIZE];
1436         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1437         pjsip_dialog *dlg = NULL;
1438         const char *outbound_proxy = endpoint->outbound_proxy;
1439         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1440         static const pj_str_t HCONTACT = { "Contact", 7 };
1441
1442         snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1443         pj_cstr(&remote_uri, enclosed_uri);
1444
1445         pj_cstr(&target_uri, uri);
1446
1447         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1448                 return NULL;
1449         }
1450
1451         if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1452                 pjsip_dlg_terminate(dlg);
1453                 return NULL;
1454         }
1455
1456         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1457                 pjsip_dlg_terminate(dlg);
1458                 return NULL;
1459         }
1460
1461         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1462         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1463         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1464         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1465
1466         /* If a request user has been specified and we are permitted to change it, do so */
1467         if (!ast_strlen_zero(request_user)) {
1468                 pjsip_sip_uri *sip_uri;
1469
1470                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1471                         sip_uri = pjsip_uri_get_uri(dlg->target);
1472                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1473                 }
1474                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1475                         sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1476                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1477                 }
1478         }
1479
1480         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1481         dlg->sess_count++;
1482
1483         pjsip_dlg_set_transport(dlg, &selector);
1484
1485         if (!ast_strlen_zero(outbound_proxy)) {
1486                 pjsip_route_hdr route_set, *route;
1487                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1488                 pj_str_t tmp;
1489
1490                 pj_list_init(&route_set);
1491
1492                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1493                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1494                         dlg->sess_count--;
1495                         pjsip_dlg_terminate(dlg);
1496                         return NULL;
1497                 }
1498                 pj_list_push_back(&route_set, route);
1499
1500                 pjsip_dlg_set_route_set(dlg, &route_set);
1501         }
1502
1503         dlg->sess_count--;
1504
1505         return dlg;
1506 }
1507
1508 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1509 {
1510         pjsip_dialog *dlg;
1511         pj_str_t contact;
1512         pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1513         pj_status_t status;
1514
1515         contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1516         contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1517                         "<%s:%s%.*s%s:%d%s%s>",
1518                         (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1519                         (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1520                         (int)rdata->tp_info.transport->local_name.host.slen,
1521                         rdata->tp_info.transport->local_name.host.ptr,
1522                         (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1523                         rdata->tp_info.transport->local_name.port,
1524                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1525                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1526
1527         status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1528         if (status != PJ_SUCCESS) {
1529                 char err[PJ_ERR_MSG_SIZE];
1530
1531                 pj_strerror(status, err, sizeof(err));
1532                 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1533                                 ast_sorcery_object_get_id(endpoint), err);
1534                 return NULL;
1535         }
1536
1537         return dlg;
1538 }
1539
1540 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1541 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1542 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1543
1544 static struct {
1545         const char *method;
1546         const pjsip_method *pmethod;
1547 } methods [] = {
1548         { "INVITE", &pjsip_invite_method },
1549         { "CANCEL", &pjsip_cancel_method },
1550         { "ACK", &pjsip_ack_method },
1551         { "BYE", &pjsip_bye_method },
1552         { "REGISTER", &pjsip_register_method },
1553         { "OPTIONS", &pjsip_options_method },
1554         { "SUBSCRIBE", &pjsip_subscribe_method },
1555         { "NOTIFY", &pjsip_notify_method },
1556         { "PUBLISH", &pjsip_publish_method },
1557         { "INFO", &info_method },
1558         { "MESSAGE", &message_method },
1559 };
1560
1561 static const pjsip_method *get_pjsip_method(const char *method)
1562 {
1563         int i;
1564         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1565                 if (!strcmp(method, methods[i].method)) {
1566                         return methods[i].pmethod;
1567                 }
1568         }
1569         return NULL;
1570 }
1571
1572 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1573 {
1574         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1575                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1576                 return -1;
1577         }
1578
1579         return 0;
1580 }
1581
1582 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1583                 const char *uri, pjsip_tx_data **tdata)
1584 {
1585         RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1586         pj_str_t remote_uri;
1587         pj_str_t from;
1588         pj_pool_t *pool;
1589         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1590
1591         if (ast_strlen_zero(uri)) {
1592                 if (!endpoint) {
1593                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1594                         return -1;
1595                 }
1596
1597                 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1598                 if (!contact || ast_strlen_zero(contact->uri)) {
1599                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1600                                         ast_sorcery_object_get_id(endpoint));
1601                         return -1;
1602                 }
1603
1604                 pj_cstr(&remote_uri, contact->uri);
1605         } else {
1606                 pj_cstr(&remote_uri, uri);
1607         }
1608
1609         if (endpoint) {
1610                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1611                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1612                                 ast_sorcery_object_get_id(endpoint));
1613                         return -1;
1614                 }
1615         }
1616
1617         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1618
1619         if (!pool) {
1620                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1621                 return -1;
1622         }
1623
1624         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1625                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1626                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1627                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1628                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1629                 return -1;
1630         }
1631
1632         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1633                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1634                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1635                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1636                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1637                 return -1;
1638         }
1639
1640         /* We can release this pool since request creation copied all the necessary
1641          * data into the outbound request's pool
1642          */
1643         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1644         return 0;
1645 }
1646
1647 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1648                 struct ast_sip_endpoint *endpoint, const char *uri,
1649                 pjsip_tx_data **tdata)
1650 {
1651         const pjsip_method *pmethod = get_pjsip_method(method);
1652
1653         if (!pmethod) {
1654                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1655                 return -1;
1656         }
1657
1658         if (dlg) {
1659                 return create_in_dialog_request(pmethod, dlg, tdata);
1660         } else {
1661                 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1662         }
1663 }
1664
1665 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1666 {
1667         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1668                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1669                 return -1;
1670         }
1671         return 0;
1672 }
1673
1674 static void send_request_cb(void *token, pjsip_event *e)
1675 {
1676         RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1677         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1678         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1679         pjsip_tx_data *tdata;
1680
1681         if (tsx->status_code != 401 && tsx->status_code != 407) {
1682                 return;
1683         }
1684
1685         if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1686                 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1687         }
1688 }
1689
1690 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1691 {
1692         ao2_ref(endpoint, +1);
1693         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1694                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1695                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1696                                 pj_strbuf(&tdata->msg->line.req.method.name),
1697                                 ast_sorcery_object_get_id(endpoint));
1698                 ao2_ref(endpoint, -1);
1699                 return -1;
1700         }
1701
1702         return 0;
1703 }
1704
1705 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1706 {
1707         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1708
1709         if (dlg) {
1710                 return send_in_dialog_request(tdata, dlg);
1711         } else {
1712                 return send_out_of_dialog_request(tdata, endpoint);
1713         }
1714 }
1715
1716 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1717 {
1718         pj_str_t hdr_name;
1719         pj_str_t hdr_value;
1720         pjsip_generic_string_hdr *hdr;
1721
1722         pj_cstr(&hdr_name, name);
1723         pj_cstr(&hdr_value, value);
1724
1725         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1726
1727         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1728         return 0;
1729 }
1730
1731 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1732 {
1733         pj_str_t type;
1734         pj_str_t subtype;
1735         pj_str_t body_text;
1736
1737         pj_cstr(&type, body->type);
1738         pj_cstr(&subtype, body->subtype);
1739         pj_cstr(&body_text, body->body_text);
1740
1741         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1742 }
1743
1744 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1745 {
1746         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1747         tdata->msg->body = pjsip_body;
1748         return 0;
1749 }
1750
1751 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1752 {
1753         int i;
1754         /* NULL for type and subtype automatically creates "multipart/mixed" */
1755         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1756
1757         for (i = 0; i < num_bodies; ++i) {
1758                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1759                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1760                 pjsip_multipart_add_part(tdata->pool, body, part);
1761         }
1762
1763         tdata->msg->body = body;
1764         return 0;
1765 }
1766
1767 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1768 {
1769         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1770         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1771
1772         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1773
1774         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1775         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1776         tdata->msg->body->len = combined_size;
1777
1778         return 0;
1779 }
1780
1781 struct ast_taskprocessor *ast_sip_create_serializer(void)
1782 {
1783         struct ast_taskprocessor *serializer;
1784         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1785         char name[AST_UUID_STR_LEN];
1786
1787         if (!uuid) {
1788                 return NULL;
1789         }
1790
1791         ast_uuid_to_str(uuid, name, sizeof(name));
1792
1793         serializer = ast_threadpool_serializer(name, sip_threadpool);
1794         if (!serializer) {
1795                 return NULL;
1796         }
1797         return serializer;
1798 }
1799
1800 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1801 {
1802         if (serializer) {
1803                 return ast_taskprocessor_push(serializer, sip_task, task_data);
1804         } else {
1805                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1806         }
1807 }
1808
1809 struct sync_task_data {
1810         ast_mutex_t lock;
1811         ast_cond_t cond;
1812         int complete;
1813         int fail;
1814         int (*task)(void *);
1815         void *task_data;
1816 };
1817
1818 static int sync_task(void *data)
1819 {
1820         struct sync_task_data *std = data;
1821         std->fail = std->task(std->task_data);
1822
1823         ast_mutex_lock(&std->lock);
1824         std->complete = 1;
1825         ast_cond_signal(&std->cond);
1826         ast_mutex_unlock(&std->lock);
1827         return std->fail;
1828 }
1829
1830 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1831 {
1832         /* This method is an onion */
1833         struct sync_task_data std;
1834         ast_mutex_init(&std.lock);
1835         ast_cond_init(&std.cond, NULL);
1836         std.fail = std.complete = 0;
1837         std.task = sip_task;
1838         std.task_data = task_data;
1839
1840         if (serializer) {
1841                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1842                         return -1;
1843                 }
1844         } else {
1845                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1846                         return -1;
1847                 }
1848         }
1849
1850         ast_mutex_lock(&std.lock);
1851         while (!std.complete) {
1852                 ast_cond_wait(&std.cond, &std.lock);
1853         }
1854         ast_mutex_unlock(&std.lock);
1855
1856         ast_mutex_destroy(&std.lock);
1857         ast_cond_destroy(&std.cond);
1858         return std.fail;
1859 }
1860
1861 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1862 {
1863         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1864         memcpy(dest, pj_strbuf(src), chars_to_copy);
1865         dest[chars_to_copy] = '\0';
1866 }
1867
1868 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1869 {
1870         pjsip_media_type compare;
1871
1872         if (!content_type) {
1873                 return 0;
1874         }
1875
1876         pjsip_media_type_init2(&compare, type, subtype);
1877
1878         return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
1879 }
1880
1881 pj_caching_pool caching_pool;
1882 pj_pool_t *memory_pool;
1883 pj_thread_t *monitor_thread;
1884 static int monitor_continue;
1885
1886 static void *monitor_thread_exec(void *endpt)
1887 {
1888         while (monitor_continue) {
1889                 const pj_time_val delay = {0, 10};
1890                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1891         }
1892         return NULL;
1893 }
1894
1895 static void stop_monitor_thread(void)
1896 {
1897         monitor_continue = 0;
1898         pj_thread_join(monitor_thread);
1899 }
1900
1901 AST_THREADSTORAGE(pj_thread_storage);
1902 AST_THREADSTORAGE(servant_id_storage);
1903 #define SIP_SERVANT_ID 0x5E2F1D
1904
1905 static void sip_thread_start(void)
1906 {
1907         pj_thread_desc *desc;
1908         pj_thread_t *thread;
1909         uint32_t *servant_id;
1910
1911         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1912         if (!servant_id) {
1913                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1914                 return;
1915         }
1916         *servant_id = SIP_SERVANT_ID;
1917
1918         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1919         if (!desc) {
1920                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1921                 return;
1922         }
1923         pj_bzero(*desc, sizeof(*desc));
1924
1925         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1926                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1927         }
1928 }
1929
1930 int ast_sip_thread_is_servant(void)
1931 {
1932         uint32_t *servant_id;
1933
1934         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1935         if (!servant_id) {
1936                 return 0;
1937         }
1938
1939         return *servant_id == SIP_SERVANT_ID;
1940 }
1941
1942 void *ast_sip_dict_get(void *ht, const char *key)
1943 {
1944         unsigned int hval = 0;
1945
1946         if (!ht) {
1947                 return NULL;
1948         }
1949
1950         return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
1951 }
1952
1953 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
1954                        const char *key, void *val)
1955 {
1956         if (!ht) {
1957                 ht = pj_hash_create(pool, 11);
1958         }
1959
1960         pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
1961
1962         return ht;
1963 }
1964
1965 static void remove_request_headers(pjsip_endpoint *endpt)
1966 {
1967         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1968         pjsip_hdr *iter = request_headers->next;
1969
1970         while (iter != request_headers) {
1971                 pjsip_hdr *to_erase = iter;
1972                 iter = iter->next;
1973                 pj_list_erase(to_erase);
1974         }
1975 }
1976
1977 static int load_module(void)
1978 {
1979         /* The third parameter is just copied from
1980          * example code from PJLIB. This can be adjusted
1981          * if necessary.
1982          */
1983         pj_status_t status;
1984         struct ast_threadpool_options options;
1985
1986         if (pj_init() != PJ_SUCCESS) {
1987                 return AST_MODULE_LOAD_DECLINE;
1988         }
1989
1990         if (pjlib_util_init() != PJ_SUCCESS) {
1991                 pj_shutdown();
1992                 return AST_MODULE_LOAD_DECLINE;
1993         }
1994
1995         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1996         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1997                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1998                 pj_caching_pool_destroy(&caching_pool);
1999                 return AST_MODULE_LOAD_DECLINE;
2000         }
2001
2002         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2003          * we need to stop PJSIP from doing it automatically
2004          */
2005         remove_request_headers(ast_pjsip_endpoint);
2006
2007         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2008         if (!memory_pool) {
2009                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2010                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2011                 ast_pjsip_endpoint = NULL;
2012                 pj_caching_pool_destroy(&caching_pool);
2013                 return AST_MODULE_LOAD_DECLINE;
2014         }
2015
2016         if (ast_sip_initialize_system()) {
2017                 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2018                 pj_pool_release(memory_pool);
2019                 memory_pool = NULL;
2020                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2021                 ast_pjsip_endpoint = NULL;
2022                 pj_caching_pool_destroy(&caching_pool);
2023                 return AST_MODULE_LOAD_DECLINE;
2024         }
2025
2026         sip_get_threadpool_options(&options);
2027         options.thread_start = sip_thread_start;
2028         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2029         if (!sip_threadpool) {
2030                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2031                 pj_pool_release(memory_pool);
2032                 memory_pool = NULL;
2033                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2034                 ast_pjsip_endpoint = NULL;
2035                 pj_caching_pool_destroy(&caching_pool);
2036                 return AST_MODULE_LOAD_DECLINE;
2037         }
2038
2039         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2040         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2041
2042         monitor_continue = 1;
2043         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2044                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2045         if (status != PJ_SUCCESS) {
2046                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2047                 pj_pool_release(memory_pool);
2048                 memory_pool = NULL;
2049                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2050                 ast_pjsip_endpoint = NULL;
2051                 pj_caching_pool_destroy(&caching_pool);
2052                 return AST_MODULE_LOAD_DECLINE;
2053         }
2054
2055         ast_sip_initialize_global_headers();
2056
2057         if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2058                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2059                 ast_sip_destroy_global_headers();
2060                 stop_monitor_thread();
2061                 pj_pool_release(memory_pool);
2062                 memory_pool = NULL;
2063                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2064                 ast_pjsip_endpoint = NULL;
2065                 pj_caching_pool_destroy(&caching_pool);
2066                 return AST_MODULE_LOAD_DECLINE;
2067         }
2068
2069         if (ast_sip_initialize_distributor()) {
2070                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2071                 ast_res_pjsip_destroy_configuration();
2072                 ast_sip_destroy_global_headers();
2073                 stop_monitor_thread();
2074                 pj_pool_release(memory_pool);
2075                 memory_pool = NULL;
2076                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2077                 ast_pjsip_endpoint = NULL;
2078                 pj_caching_pool_destroy(&caching_pool);
2079                 return AST_MODULE_LOAD_DECLINE;
2080         }
2081
2082         if (ast_sip_initialize_outbound_authentication()) {
2083                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2084                 ast_sip_destroy_distributor();
2085                 ast_res_pjsip_destroy_configuration();
2086                 ast_sip_destroy_global_headers();
2087                 stop_monitor_thread();
2088                 pj_pool_release(memory_pool);
2089                 memory_pool = NULL;
2090                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2091                 ast_pjsip_endpoint = NULL;
2092                 pj_caching_pool_destroy(&caching_pool);
2093                 return AST_MODULE_LOAD_DECLINE;
2094         }
2095
2096         ast_res_pjsip_init_options_handling(0);
2097
2098         ast_module_ref(ast_module_info->self);
2099
2100         return AST_MODULE_LOAD_SUCCESS;
2101 }
2102
2103 static int reload_module(void)
2104 {
2105         if (ast_res_pjsip_reload_configuration()) {
2106                 return AST_MODULE_LOAD_DECLINE;
2107         }
2108         ast_res_pjsip_init_options_handling(1);
2109         return 0;
2110 }
2111
2112 static int unload_module(void)
2113 {
2114         /* This will never get called as this module can't be unloaded */
2115         return 0;
2116 }
2117
2118 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2119                 .load = load_module,
2120                 .unload = unload_module,
2121                 .reload = reload_module,
2122                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
2123 );