e6f00a8e9493f0672d7b60cb1ce5536865253468
[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                                 <configOption name="set_var">
669                                         <synopsis>Variable set on a channel involving the endpoint.</synopsis>
670                                         <description><para>
671                                                 When a new channel is created using the endpoint set the specified
672                                                 variable(s) on that channel. For multiple channel variables specify
673                                                 multiple 'set_var'(s).
674                                         </para></description>
675                                 </configOption>
676                         </configObject>
677                         <configObject name="auth">
678                                 <synopsis>Authentication type</synopsis>
679                                 <description><para>
680                                         Authentication objects hold the authentication information for use
681                                         by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
682                                         This also allows for multiple objects to use a single auth object. See
683                                         the <literal>auth_type</literal> config option for password style choices.
684                                 </para></description>
685                                 <configOption name="auth_type" default="userpass">
686                                         <synopsis>Authentication type</synopsis>
687                                         <description><para>
688                                                 This option specifies which of the password style config options should be read
689                                                 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
690                                                 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
691                                                 from 'md5_cred'.
692                                                 </para>
693                                                 <enumlist>
694                                                         <enum name="md5"/>
695                                                         <enum name="userpass"/>
696                                                 </enumlist>
697                                         </description>
698                                 </configOption>
699                                 <configOption name="nonce_lifetime" default="32">
700                                         <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
701                                 </configOption>
702                                 <configOption name="md5_cred">
703                                         <synopsis>MD5 Hash used for authentication.</synopsis>
704                                         <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
705                                 </configOption>
706                                 <configOption name="password">
707                                         <synopsis>PlainText password used for authentication.</synopsis>
708                                         <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
709                                 </configOption>
710                                 <configOption name="realm" default="asterisk">
711                                         <synopsis>SIP realm for endpoint</synopsis>
712                                 </configOption>
713                                 <configOption name="type">
714                                         <synopsis>Must be 'auth'</synopsis>
715                                 </configOption>
716                                 <configOption name="username">
717                                         <synopsis>Username to use for account</synopsis>
718                                 </configOption>
719                         </configObject>
720                         <configObject name="domain_alias">
721                                 <synopsis>Domain Alias</synopsis>
722                                 <description><para>
723                                         Signifies that a domain is an alias. If the domain on a session is
724                                         not found to match an AoR then this object is used to see if we have
725                                         an alias for the AoR to which the endpoint is binding. This objects
726                                         name as defined in configuration should be the domain alias and a
727                                         config option is provided to specify the domain to be aliased.
728                                 </para></description>
729                                 <configOption name="type">
730                                         <synopsis>Must be of type 'domain_alias'.</synopsis>
731                                 </configOption>
732                                 <configOption name="domain">
733                                         <synopsis>Domain to be aliased</synopsis>
734                                 </configOption>
735                         </configObject>
736                         <configObject name="transport">
737                                 <synopsis>SIP Transport</synopsis>
738                                 <description><para>
739                                         <emphasis>Transports</emphasis>
740                                         </para>
741                                         <para>There are different transports and protocol derivatives
742                                                 supported by <literal>res_pjsip</literal>. They are in order of
743                                                 preference: UDP, TCP, and WebSocket (WS).</para>
744                                         <note><para>Changes to transport configuration in pjsip.conf will only be
745                                                 effected on a complete restart of Asterisk. A module reload
746                                                 will not suffice.</para></note>
747                                 </description>
748                                 <configOption name="async_operations" default="1">
749                                         <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
750                                 </configOption>
751                                 <configOption name="bind">
752                                         <synopsis>IP Address and optional port to bind to for this transport</synopsis>
753                                 </configOption>
754                                 <configOption name="ca_list_file">
755                                         <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
756                                 </configOption>
757                                 <configOption name="cert_file">
758                                         <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
759                                 </configOption>
760                                 <configOption name="cipher">
761                                         <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
762                                         <description><para>
763                                                 Many options for acceptable ciphers see link for more:
764                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
765                                         </para></description>
766                                 </configOption>
767                                 <configOption name="domain">
768                                         <synopsis>Domain the transport comes from</synopsis>
769                                 </configOption>
770                                 <configOption name="external_media_address">
771                                         <synopsis>External IP address to use in RTP handling</synopsis>
772                                         <description><para>
773                                                 When a request or response is sent out, if the destination of the
774                                                 message is outside the IP network defined in the option <literal>localnet</literal>,
775                                                 and the media address in the SDP is within the localnet network, then the
776                                                 media address in the SDP will be rewritten to the value defined for
777                                                 <literal>external_media_address</literal>.
778                                         </para></description>
779                                 </configOption>
780                                 <configOption name="external_signaling_address">
781                                         <synopsis>External address for SIP signalling</synopsis>
782                                 </configOption>
783                                 <configOption name="external_signaling_port" default="0">
784                                         <synopsis>External port for SIP signalling</synopsis>
785                                 </configOption>
786                                 <configOption name="method">
787                                         <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
788                                         <description>
789                                                 <enumlist>
790                                                         <enum name="default" />
791                                                         <enum name="unspecified" />
792                                                         <enum name="tlsv1" />
793                                                         <enum name="sslv2" />
794                                                         <enum name="sslv3" />
795                                                         <enum name="sslv23" />
796                                                 </enumlist>
797                                         </description>
798                                 </configOption>
799                                 <configOption name="local_net">
800                                         <synopsis>Network to consider local (used for NAT purposes).</synopsis>
801                                         <description><para>This must be in CIDR or dotted decimal format with the IP
802                                         and mask separated with a slash ('/').</para></description>
803                                 </configOption>
804                                 <configOption name="password">
805                                         <synopsis>Password required for transport</synopsis>
806                                 </configOption>
807                                 <configOption name="priv_key_file">
808                                         <synopsis>Private key file (TLS ONLY)</synopsis>
809                                 </configOption>
810                                 <configOption name="protocol" default="udp">
811                                         <synopsis>Protocol to use for SIP traffic</synopsis>
812                                         <description>
813                                                 <enumlist>
814                                                         <enum name="udp" />
815                                                         <enum name="tcp" />
816                                                         <enum name="tls" />
817                                                         <enum name="ws" />
818                                                         <enum name="wss" />
819                                                 </enumlist>
820                                         </description>
821                                 </configOption>
822                                 <configOption name="require_client_cert" default="false">
823                                         <synopsis>Require client certificate (TLS ONLY)</synopsis>
824                                 </configOption>
825                                 <configOption name="type">
826                                         <synopsis>Must be of type 'transport'.</synopsis>
827                                 </configOption>
828                                 <configOption name="verify_client" default="false">
829                                         <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
830                                 </configOption>
831                                 <configOption name="verify_server" default="false">
832                                         <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
833                                 </configOption>
834                                 <configOption name="tos" default="false">
835                                         <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
836                                         <description>
837                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
838                                         for more information on this parameter.</para>
839                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
840                                         or the <replaceable>wss</replaceable> protocols.</para></note>
841                                         </description>
842                                 </configOption>
843                                 <configOption name="cos" default="false">
844                                         <synopsis>Enable COS for the signalling sent over this transport</synopsis>
845                                         <description>
846                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
847                                         for more information on this parameter.</para>
848                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
849                                         or the <replaceable>wss</replaceable> protocols.</para></note>
850                                         </description>
851                                 </configOption>
852                         </configObject>
853                         <configObject name="contact">
854                                 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
855                                 <description><para>
856                                         Contacts are a way to hide SIP URIs from the dialplan directly.
857                                         They are also used to make a group of contactable parties when
858                                         in use with <literal>AoR</literal> lists.
859                                 </para></description>
860                                 <configOption name="type">
861                                         <synopsis>Must be of type 'contact'.</synopsis>
862                                 </configOption>
863                                 <configOption name="uri">
864                                         <synopsis>SIP URI to contact peer</synopsis>
865                                 </configOption>
866                                 <configOption name="expiration_time">
867                                         <synopsis>Time to keep alive a contact</synopsis>
868                                         <description><para>
869                                                 Time to keep alive a contact. String style specification.
870                                         </para></description>
871                                 </configOption>
872                                 <configOption name="qualify_frequency" default="0">
873                                         <synopsis>Interval at which to qualify a contact</synopsis>
874                                         <description><para>
875                                                 Interval between attempts to qualify the contact for reachability.
876                                                 If <literal>0</literal> never qualify. Time in seconds.
877                                         </para></description>
878                                 </configOption>
879                                 <configOption name="outbound_proxy">
880                                         <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
881                                         <description><para>
882                                                 If set the provided URI will be used as the outbound proxy when an
883                                                 OPTIONS request is sent to a contact for qualify purposes.
884                                         </para></description>
885                                 </configOption>
886                         </configObject>
887                         <configObject name="aor">
888                                 <synopsis>The configuration for a location of an endpoint</synopsis>
889                                 <description><para>
890                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
891                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
892                                         Beyond that, an AoR has other uses within Asterisk, such as inbound
893                                         registration.
894                                         </para><para>
895                                         An <literal>AoR</literal> is a way to allow dialing a group
896                                         of <literal>Contacts</literal> that all use the same
897                                         <literal>endpoint</literal> for calls.
898                                         </para><para>
899                                         This can be used as another way of grouping a list of contacts to dial
900                                         rather than specifing them each directly when dialing via the dialplan.
901                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
902                                         </para><para>
903                                         Registrations: For Asterisk to match an inbound registration to an endpoint,
904                                         the AoR object name must match the user portion of the SIP URI in the "To:"
905                                         header of the inbound SIP registration. That will usually be equivalent
906                                         to the "user name" set in your hard or soft phones configuration.
907                                 </para></description>
908                                 <configOption name="contact">
909                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
910                                         <description><para>
911                                                 Contacts specified will be called whenever referenced
912                                                 by <literal>chan_pjsip</literal>.
913                                                 </para><para>
914                                                 Use a separate "contact=" entry for each contact required. Contacts
915                                                 are specified using a SIP URI.
916                                         </para></description>
917                                 </configOption>
918                                 <configOption name="default_expiration" default="3600">
919                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
920                                 </configOption>
921                                 <configOption name="mailboxes">
922                                         <synopsis>Mailbox(es) to be associated with</synopsis>
923                                         <description><para>This option applies when an external entity subscribes to an AoR
924                                         for message waiting indications. The mailboxes specified will be subscribed to.
925                                         More than one mailbox can be specified with a comma-delimited string.</para></description>
926                                 </configOption>
927                                 <configOption name="maximum_expiration" default="7200">
928                                         <synopsis>Maximum time to keep an AoR</synopsis>
929                                         <description><para>
930                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
931                                         </para></description>
932                                 </configOption>
933                                 <configOption name="max_contacts" default="0">
934                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
935                                         <description><para>
936                                                 Maximum number of contacts that can associate with this AoR. This value does
937                                                 not affect the number of contacts that can be added with the "contact" option.
938                                                 It only limits contacts added through external interaction, such as
939                                                 registration.
940                                                 </para>
941                                                 <note><para>This should be set to <literal>1</literal> and
942                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
943                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
944                                                 </para></note>
945                                         </description>
946                                 </configOption>
947                                 <configOption name="minimum_expiration" default="60">
948                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
949                                         <description><para>
950                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
951                                         </para></description>
952                                 </configOption>
953                                 <configOption name="remove_existing" default="no">
954                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
955                                         <description><para>
956                                                 On receiving a new registration to the AoR should it remove
957                                                 the existing contact that was registered against it?
958                                                 </para>
959                                                 <note><para>This should be set to <literal>yes</literal> and
960                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
961                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
962                                                 </para></note>
963                                         </description>
964                                 </configOption>
965                                 <configOption name="type">
966                                         <synopsis>Must be of type 'aor'.</synopsis>
967                                 </configOption>
968                                 <configOption name="qualify_frequency" default="0">
969                                         <synopsis>Interval at which to qualify an AoR</synopsis>
970                                         <description><para>
971                                                 Interval between attempts to qualify the AoR for reachability.
972                                                 If <literal>0</literal> never qualify. Time in seconds.
973                                         </para></description>
974                                 </configOption>
975                                 <configOption name="authenticate_qualify" default="no">
976                                         <synopsis>Authenticates a qualify request if needed</synopsis>
977                                         <description><para>
978                                                 If true and a qualify request receives a challenge or authenticate response
979                                                 authentication is attempted before declaring the contact available.
980                                         </para></description>
981                                 </configOption>
982                                 <configOption name="outbound_proxy">
983                                         <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
984                                         <description><para>
985                                                 If set the provided URI will be used as the outbound proxy when an
986                                                 OPTIONS request is sent to a contact for qualify purposes.
987                                         </para></description>
988                                 </configOption>
989                         </configObject>
990                         <configObject name="system">
991                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
992                                 <description><para>
993                                         The settings in this section are global. In addition to being global, the values will
994                                         not be re-evaluated when a reload is performed. This is because the values must be set
995                                         before the SIP stack is initialized. The only way to reset these values is to either
996                                         restart Asterisk, or unload res_pjsip.so and then load it again.
997                                 </para></description>
998                                 <configOption name="timer_t1" default="500">
999                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
1000                                         <description><para>
1001                                                 Timer T1 is the base for determining how long to wait before retransmitting
1002                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
1003                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
1004                                         </para></description>
1005                                 </configOption>
1006                                 <configOption name="timer_b" default="32000">
1007                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
1008                                         <description><para>
1009                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
1010                                                 request before terminating the transaction. It is recommended that this be set
1011                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
1012                                                 this timer, see RFC 3261, Section 17.1.1.1.
1013                                         </para></description>
1014                                 </configOption>
1015                                 <configOption name="compact_headers" default="no">
1016                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
1017                                 </configOption>
1018                                 <configOption name="threadpool_initial_size" default="0">
1019                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
1020                                 </configOption>
1021                                 <configOption name="threadpool_auto_increment" default="5">
1022                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1023                                 </configOption>
1024                                 <configOption name="threadpool_idle_timeout" default="60">
1025                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1026                                 </configOption>
1027                                 <configOption name="threadpool_max_size" default="0">
1028                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
1029                                         A value of 0 indicates no maximum.</synopsis>
1030                                 </configOption>
1031                                 <configOption name="type">
1032                                         <synopsis>Must be of type 'system'.</synopsis>
1033                                 </configOption>
1034                         </configObject>
1035                         <configObject name="global">
1036                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
1037                                 <description><para>
1038                                         The settings in this section are global. Unlike options in the <literal>system</literal>
1039                                         section, these options can be refreshed by performing a reload.
1040                                 </para></description>
1041                                 <configOption name="max_forwards" default="70">
1042                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1043                                 </configOption>
1044                                 <configOption name="type">
1045                                         <synopsis>Must be of type 'global'.</synopsis>
1046                                 </configOption>
1047                                 <configOption name="user_agent" default="Asterisk &lt;Asterisk Version&gt;">
1048                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1049                                 </configOption>
1050                                 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1051                                         <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1052                                 </configOption>
1053
1054                         </configObject>
1055                 </configFile>
1056         </configInfo>
1057         <manager name="PJSIPQualify" language="en_US">
1058                 <synopsis>
1059                         Qualify a chan_pjsip endpoint.
1060                 </synopsis>
1061                 <syntax>
1062                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1063                         <parameter name="Endpoint" required="true">
1064                                 <para>The endpoint you want to qualify.</para>
1065                         </parameter>
1066                 </syntax>
1067                 <description>
1068                         <para>Qualify a chan_pjsip endpoint.</para>
1069                 </description>
1070         </manager>
1071         <manager name="PJSIPShowEndpoints" language="en_US">
1072                 <synopsis>
1073                         Lists PJSIP endpoints.
1074                 </synopsis>
1075                 <syntax />
1076                 <description>
1077                         <para>
1078                         Provides a listing of all endpoints.  For each endpoint an <literal>EndpointList</literal> event
1079                         is raised that contains relevant attributes and status information.  Once all
1080                         endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1081                         </para>
1082                 </description>
1083         </manager>
1084         <manager name="PJSIPShowEndpoint" language="en_US">
1085                 <synopsis>
1086                         Detail listing of an endpoint and its objects.
1087                 </synopsis>
1088                 <syntax>
1089                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1090                         <parameter name="Endpoint" required="true">
1091                                 <para>The endpoint to list.</para>
1092                         </parameter>
1093                 </syntax>
1094                 <description>
1095                         <para>
1096                         Provides a detailed listing of options for a given endpoint.  Events are issued
1097                         showing the configuration and status of the endpoint and associated objects.  These
1098                         events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1099                         <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1100                         <literal>IdentifyDetail</literal>.  Some events may be listed multiple times if multiple objects are
1101                         associated (for instance AoRs).  Once all detail events have been raised a final
1102                         <literal>EndpointDetailComplete</literal> event is issued.
1103                         </para>
1104                 </description>
1105         </manager>
1106  ***/
1107
1108
1109 static pjsip_endpoint *ast_pjsip_endpoint;
1110
1111 static struct ast_threadpool *sip_threadpool;
1112
1113 static int register_service(void *data)
1114 {
1115         pjsip_module **module = data;
1116         if (!ast_pjsip_endpoint) {
1117                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1118                 return -1;
1119         }
1120         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1121                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1122                 return -1;
1123         }
1124         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1125         ast_module_ref(ast_module_info->self);
1126         return 0;
1127 }
1128
1129 int ast_sip_register_service(pjsip_module *module)
1130 {
1131         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1132 }
1133
1134 static int unregister_service(void *data)
1135 {
1136         pjsip_module **module = data;
1137         ast_module_unref(ast_module_info->self);
1138         if (!ast_pjsip_endpoint) {
1139                 return -1;
1140         }
1141         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1142         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1143         return 0;
1144 }
1145
1146 void ast_sip_unregister_service(pjsip_module *module)
1147 {
1148         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1149 }
1150
1151 static struct ast_sip_authenticator *registered_authenticator;
1152
1153 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1154 {
1155         if (registered_authenticator) {
1156                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1157                 return -1;
1158         }
1159         registered_authenticator = auth;
1160         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1161         ast_module_ref(ast_module_info->self);
1162         return 0;
1163 }
1164
1165 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1166 {
1167         if (registered_authenticator != auth) {
1168                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1169                                 auth, registered_authenticator);
1170                 return;
1171         }
1172         registered_authenticator = NULL;
1173         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1174         ast_module_unref(ast_module_info->self);
1175 }
1176
1177 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1178 {
1179         if (!registered_authenticator) {
1180                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1181                 return 0;
1182         }
1183
1184         return registered_authenticator->requires_authentication(endpoint, rdata);
1185 }
1186
1187 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1188                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1189 {
1190         if (!registered_authenticator) {
1191                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1192                 return 0;
1193         }
1194         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1195 }
1196
1197 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1198
1199 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1200 {
1201         if (registered_outbound_authenticator) {
1202                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1203                 return -1;
1204         }
1205         registered_outbound_authenticator = auth;
1206         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1207         ast_module_ref(ast_module_info->self);
1208         return 0;
1209 }
1210
1211 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1212 {
1213         if (registered_outbound_authenticator != auth) {
1214                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1215                                 auth, registered_outbound_authenticator);
1216                 return;
1217         }
1218         registered_outbound_authenticator = NULL;
1219         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1220         ast_module_unref(ast_module_info->self);
1221 }
1222
1223 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1224                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1225 {
1226         if (!registered_outbound_authenticator) {
1227                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1228                 return -1;
1229         }
1230         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1231 }
1232
1233 struct endpoint_identifier_list {
1234         struct ast_sip_endpoint_identifier *identifier;
1235         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1236 };
1237
1238 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1239
1240 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1241 {
1242         struct endpoint_identifier_list *id_list_item;
1243         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1244
1245         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1246         if (!id_list_item) {
1247                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1248                 return -1;
1249         }
1250         id_list_item->identifier = identifier;
1251
1252         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1253         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1254
1255         ast_module_ref(ast_module_info->self);
1256         return 0;
1257 }
1258
1259 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1260 {
1261         struct endpoint_identifier_list *iter;
1262         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1263         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1264                 if (iter->identifier == identifier) {
1265                         AST_RWLIST_REMOVE_CURRENT(list);
1266                         ast_free(iter);
1267                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1268                         ast_module_unref(ast_module_info->self);
1269                         break;
1270                 }
1271         }
1272         AST_RWLIST_TRAVERSE_SAFE_END;
1273 }
1274
1275 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1276 {
1277         struct endpoint_identifier_list *iter;
1278         struct ast_sip_endpoint *endpoint = NULL;
1279         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1280         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1281                 ast_assert(iter->identifier->identify_endpoint != NULL);
1282                 endpoint = iter->identifier->identify_endpoint(rdata);
1283                 if (endpoint) {
1284                         break;
1285                 }
1286         }
1287         return endpoint;
1288 }
1289
1290 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1291
1292 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1293 {
1294         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1295         AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1296         ast_module_ref(ast_module_info->self);
1297         return 0;
1298 }
1299
1300 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1301 {
1302         struct ast_sip_endpoint_formatter *i;
1303         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1304         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1305                 if (i == obj) {
1306                         AST_RWLIST_REMOVE_CURRENT(next);
1307                         ast_module_unref(ast_module_info->self);
1308                         break;
1309                 }
1310         }
1311         AST_RWLIST_TRAVERSE_SAFE_END;
1312 }
1313
1314 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1315                                 struct ast_sip_ami *ami, int *count)
1316 {
1317         int res = 0;
1318         struct ast_sip_endpoint_formatter *i;
1319         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1320         *count = 0;
1321         AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1322                 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1323                         return res;
1324                 }
1325
1326                 if (!res) {
1327                         (*count)++;
1328                 }
1329         }
1330         return 0;
1331 }
1332
1333 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1334 {
1335         return ast_pjsip_endpoint;
1336 }
1337
1338 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)
1339 {
1340         pj_str_t tmp, local_addr;
1341         pjsip_uri *uri;
1342         pjsip_sip_uri *sip_uri;
1343         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1344         int local_port;
1345         char uuid_str[AST_UUID_STR_LEN];
1346
1347         if (ast_strlen_zero(user)) {
1348                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1349                 if (!uuid) {
1350                         return -1;
1351                 }
1352                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1353         }
1354
1355         /* Parse the provided target URI so we can determine what transport it will end up using */
1356         pj_strdup_with_null(pool, &tmp, target);
1357
1358         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1359             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1360                 return -1;
1361         }
1362
1363         sip_uri = pjsip_uri_get_uri(uri);
1364
1365         /* Determine the transport type to use */
1366         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1367                 type = PJSIP_TRANSPORT_TLS;
1368         } else if (!sip_uri->transport_param.slen) {
1369                 type = PJSIP_TRANSPORT_UDP;
1370         } else {
1371                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1372         }
1373
1374         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1375                 return -1;
1376         }
1377
1378         /* If the host is IPv6 turn the transport into an IPv6 version */
1379         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1380                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1381         }
1382
1383         if (!ast_strlen_zero(domain)) {
1384                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1385                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1386                                 "<sip:%s@%s%s%s>",
1387                                 user,
1388                                 domain,
1389                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1390                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1391                 return 0;
1392         }
1393
1394         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1395         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1396                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1397
1398                 /* If no local address can be retrieved using the transport manager use the host one */
1399                 pj_strdup(pool, &local_addr, pj_gethostname());
1400                 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1401         }
1402
1403         /* If IPv6 was specified in the transport, set the proper type */
1404         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1405                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1406         }
1407
1408         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1409         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1410                                       "<sip:%s@%s%.*s%s:%d%s%s>",
1411                                       user,
1412                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1413                                       (int)local_addr.slen,
1414                                       local_addr.ptr,
1415                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1416                                       local_port,
1417                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1418                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1419
1420         return 0;
1421 }
1422
1423 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1424 {
1425         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1426         const char *transport_name = endpoint->transport;
1427
1428         if (ast_strlen_zero(transport_name)) {
1429                 return 0;
1430         }
1431
1432         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1433
1434         if (!transport || !transport->state) {
1435                 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1436                         transport_name, ast_sorcery_object_get_id(endpoint));
1437                 return -1;
1438         }
1439
1440         if (transport->state->transport) {
1441                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1442                 selector->u.transport = transport->state->transport;
1443         } else if (transport->state->factory) {
1444                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1445                 selector->u.listener = transport->state->factory;
1446         } else {
1447                 return -1;
1448         }
1449
1450         return 0;
1451 }
1452
1453 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1454 {
1455         char enclosed_uri[PJSIP_MAX_URL_SIZE];
1456         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1457         pjsip_dialog *dlg = NULL;
1458         const char *outbound_proxy = endpoint->outbound_proxy;
1459         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1460         static const pj_str_t HCONTACT = { "Contact", 7 };
1461
1462         snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1463         pj_cstr(&remote_uri, enclosed_uri);
1464
1465         pj_cstr(&target_uri, uri);
1466
1467         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1468                 return NULL;
1469         }
1470
1471         if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1472                 pjsip_dlg_terminate(dlg);
1473                 return NULL;
1474         }
1475
1476         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1477                 pjsip_dlg_terminate(dlg);
1478                 return NULL;
1479         }
1480
1481         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1482         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1483         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1484         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1485
1486         /* If a request user has been specified and we are permitted to change it, do so */
1487         if (!ast_strlen_zero(request_user)) {
1488                 pjsip_sip_uri *sip_uri;
1489
1490                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1491                         sip_uri = pjsip_uri_get_uri(dlg->target);
1492                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1493                 }
1494                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1495                         sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1496                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1497                 }
1498         }
1499
1500         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1501         dlg->sess_count++;
1502
1503         pjsip_dlg_set_transport(dlg, &selector);
1504
1505         if (!ast_strlen_zero(outbound_proxy)) {
1506                 pjsip_route_hdr route_set, *route;
1507                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1508                 pj_str_t tmp;
1509
1510                 pj_list_init(&route_set);
1511
1512                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1513                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1514                         dlg->sess_count--;
1515                         pjsip_dlg_terminate(dlg);
1516                         return NULL;
1517                 }
1518                 pj_list_push_back(&route_set, route);
1519
1520                 pjsip_dlg_set_route_set(dlg, &route_set);
1521         }
1522
1523         dlg->sess_count--;
1524
1525         return dlg;
1526 }
1527
1528 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1529 {
1530         pjsip_dialog *dlg;
1531         pj_str_t contact;
1532         pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1533         pj_status_t status;
1534
1535         contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1536         contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1537                         "<sip:%s%.*s%s:%d%s%s>",
1538                         (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1539                         (int)rdata->tp_info.transport->local_name.host.slen,
1540                         rdata->tp_info.transport->local_name.host.ptr,
1541                         (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1542                         rdata->tp_info.transport->local_name.port,
1543                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1544                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1545
1546         status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1547         if (status != PJ_SUCCESS) {
1548                 char err[PJ_ERR_MSG_SIZE];
1549
1550                 pj_strerror(status, err, sizeof(err));
1551                 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1552                                 ast_sorcery_object_get_id(endpoint), err);
1553                 return NULL;
1554         }
1555
1556         return dlg;
1557 }
1558
1559 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1560 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1561 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1562
1563 static struct {
1564         const char *method;
1565         const pjsip_method *pmethod;
1566 } methods [] = {
1567         { "INVITE", &pjsip_invite_method },
1568         { "CANCEL", &pjsip_cancel_method },
1569         { "ACK", &pjsip_ack_method },
1570         { "BYE", &pjsip_bye_method },
1571         { "REGISTER", &pjsip_register_method },
1572         { "OPTIONS", &pjsip_options_method },
1573         { "SUBSCRIBE", &pjsip_subscribe_method },
1574         { "NOTIFY", &pjsip_notify_method },
1575         { "PUBLISH", &pjsip_publish_method },
1576         { "INFO", &info_method },
1577         { "MESSAGE", &message_method },
1578 };
1579
1580 static const pjsip_method *get_pjsip_method(const char *method)
1581 {
1582         int i;
1583         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1584                 if (!strcmp(method, methods[i].method)) {
1585                         return methods[i].pmethod;
1586                 }
1587         }
1588         return NULL;
1589 }
1590
1591 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1592 {
1593         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1594                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1595                 return -1;
1596         }
1597
1598         return 0;
1599 }
1600
1601 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1602                 const char *uri, pjsip_tx_data **tdata)
1603 {
1604         RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1605         pj_str_t remote_uri;
1606         pj_str_t from;
1607         pj_pool_t *pool;
1608         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1609
1610         if (ast_strlen_zero(uri)) {
1611                 if (!endpoint) {
1612                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1613                         return -1;
1614                 }
1615
1616                 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1617                 if (!contact || ast_strlen_zero(contact->uri)) {
1618                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1619                                         ast_sorcery_object_get_id(endpoint));
1620                         return -1;
1621                 }
1622
1623                 pj_cstr(&remote_uri, contact->uri);
1624         } else {
1625                 pj_cstr(&remote_uri, uri);
1626         }
1627
1628         if (endpoint) {
1629                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1630                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1631                                 ast_sorcery_object_get_id(endpoint));
1632                         return -1;
1633                 }
1634         }
1635
1636         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1637
1638         if (!pool) {
1639                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1640                 return -1;
1641         }
1642
1643         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1644                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1645                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1646                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1647                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1648                 return -1;
1649         }
1650
1651         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1652                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1653                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1654                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1655                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1656                 return -1;
1657         }
1658
1659         /* If an outbound proxy is specified on the endpoint apply it to this request */
1660         if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
1661                 ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
1662                 ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
1663                         (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1664                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1665                 return -1;
1666         }
1667
1668         /* We can release this pool since request creation copied all the necessary
1669          * data into the outbound request's pool
1670          */
1671         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1672         return 0;
1673 }
1674
1675 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1676                 struct ast_sip_endpoint *endpoint, const char *uri,
1677                 pjsip_tx_data **tdata)
1678 {
1679         const pjsip_method *pmethod = get_pjsip_method(method);
1680
1681         if (!pmethod) {
1682                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1683                 return -1;
1684         }
1685
1686         if (dlg) {
1687                 return create_in_dialog_request(pmethod, dlg, tdata);
1688         } else {
1689                 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1690         }
1691 }
1692
1693 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1694 {
1695         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1696                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1697                 return -1;
1698         }
1699         return 0;
1700 }
1701
1702 static void send_request_cb(void *token, pjsip_event *e)
1703 {
1704         RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1705         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1706         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1707         pjsip_tx_data *tdata;
1708
1709         if (tsx->status_code != 401 && tsx->status_code != 407) {
1710                 return;
1711         }
1712
1713         if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1714                 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1715         }
1716 }
1717
1718 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1719 {
1720         ao2_ref(endpoint, +1);
1721         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1722                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1723                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1724                                 pj_strbuf(&tdata->msg->line.req.method.name),
1725                                 ast_sorcery_object_get_id(endpoint));
1726                 ao2_ref(endpoint, -1);
1727                 return -1;
1728         }
1729
1730         return 0;
1731 }
1732
1733 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1734 {
1735         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1736
1737         if (dlg) {
1738                 return send_in_dialog_request(tdata, dlg);
1739         } else {
1740                 return send_out_of_dialog_request(tdata, endpoint);
1741         }
1742 }
1743
1744 int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
1745 {
1746         pjsip_route_hdr *route;
1747         static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1748         pj_str_t tmp;
1749
1750         pj_strdup2_with_null(tdata->pool, &tmp, proxy);
1751         if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1752                 return -1;
1753         }
1754
1755         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)route);
1756
1757         return 0;
1758 }
1759
1760 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1761 {
1762         pj_str_t hdr_name;
1763         pj_str_t hdr_value;
1764         pjsip_generic_string_hdr *hdr;
1765
1766         pj_cstr(&hdr_name, name);
1767         pj_cstr(&hdr_value, value);
1768
1769         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1770
1771         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1772         return 0;
1773 }
1774
1775 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1776 {
1777         pj_str_t type;
1778         pj_str_t subtype;
1779         pj_str_t body_text;
1780
1781         pj_cstr(&type, body->type);
1782         pj_cstr(&subtype, body->subtype);
1783         pj_cstr(&body_text, body->body_text);
1784
1785         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1786 }
1787
1788 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1789 {
1790         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1791         tdata->msg->body = pjsip_body;
1792         return 0;
1793 }
1794
1795 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1796 {
1797         int i;
1798         /* NULL for type and subtype automatically creates "multipart/mixed" */
1799         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1800
1801         for (i = 0; i < num_bodies; ++i) {
1802                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1803                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1804                 pjsip_multipart_add_part(tdata->pool, body, part);
1805         }
1806
1807         tdata->msg->body = body;
1808         return 0;
1809 }
1810
1811 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1812 {
1813         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1814         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1815
1816         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1817
1818         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1819         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1820         tdata->msg->body->len = combined_size;
1821
1822         return 0;
1823 }
1824
1825 struct ast_taskprocessor *ast_sip_create_serializer(void)
1826 {
1827         struct ast_taskprocessor *serializer;
1828         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1829         char name[AST_UUID_STR_LEN];
1830
1831         if (!uuid) {
1832                 return NULL;
1833         }
1834
1835         ast_uuid_to_str(uuid, name, sizeof(name));
1836
1837         serializer = ast_threadpool_serializer(name, sip_threadpool);
1838         if (!serializer) {
1839                 return NULL;
1840         }
1841         return serializer;
1842 }
1843
1844 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1845 {
1846         if (serializer) {
1847                 return ast_taskprocessor_push(serializer, sip_task, task_data);
1848         } else {
1849                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1850         }
1851 }
1852
1853 struct sync_task_data {
1854         ast_mutex_t lock;
1855         ast_cond_t cond;
1856         int complete;
1857         int fail;
1858         int (*task)(void *);
1859         void *task_data;
1860 };
1861
1862 static int sync_task(void *data)
1863 {
1864         struct sync_task_data *std = data;
1865         std->fail = std->task(std->task_data);
1866
1867         ast_mutex_lock(&std->lock);
1868         std->complete = 1;
1869         ast_cond_signal(&std->cond);
1870         ast_mutex_unlock(&std->lock);
1871         return std->fail;
1872 }
1873
1874 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1875 {
1876         /* This method is an onion */
1877         struct sync_task_data std;
1878
1879         if (ast_sip_thread_is_servant()) {
1880                 return sip_task(task_data);
1881         }
1882
1883         ast_mutex_init(&std.lock);
1884         ast_cond_init(&std.cond, NULL);
1885         std.fail = std.complete = 0;
1886         std.task = sip_task;
1887         std.task_data = task_data;
1888
1889         if (serializer) {
1890                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1891                         return -1;
1892                 }
1893         } else {
1894                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1895                         return -1;
1896                 }
1897         }
1898
1899         ast_mutex_lock(&std.lock);
1900         while (!std.complete) {
1901                 ast_cond_wait(&std.cond, &std.lock);
1902         }
1903         ast_mutex_unlock(&std.lock);
1904
1905         ast_mutex_destroy(&std.lock);
1906         ast_cond_destroy(&std.cond);
1907         return std.fail;
1908 }
1909
1910 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1911 {
1912         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1913         memcpy(dest, pj_strbuf(src), chars_to_copy);
1914         dest[chars_to_copy] = '\0';
1915 }
1916
1917 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1918 {
1919         pjsip_media_type compare;
1920
1921         if (!content_type) {
1922                 return 0;
1923         }
1924
1925         pjsip_media_type_init2(&compare, type, subtype);
1926
1927         return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
1928 }
1929
1930 pj_caching_pool caching_pool;
1931 pj_pool_t *memory_pool;
1932 pj_thread_t *monitor_thread;
1933 static int monitor_continue;
1934
1935 static void *monitor_thread_exec(void *endpt)
1936 {
1937         while (monitor_continue) {
1938                 const pj_time_val delay = {0, 10};
1939                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1940         }
1941         return NULL;
1942 }
1943
1944 static void stop_monitor_thread(void)
1945 {
1946         monitor_continue = 0;
1947         pj_thread_join(monitor_thread);
1948 }
1949
1950 AST_THREADSTORAGE(pj_thread_storage);
1951 AST_THREADSTORAGE(servant_id_storage);
1952 #define SIP_SERVANT_ID 0x5E2F1D
1953
1954 static void sip_thread_start(void)
1955 {
1956         pj_thread_desc *desc;
1957         pj_thread_t *thread;
1958         uint32_t *servant_id;
1959
1960         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1961         if (!servant_id) {
1962                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1963                 return;
1964         }
1965         *servant_id = SIP_SERVANT_ID;
1966
1967         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1968         if (!desc) {
1969                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1970                 return;
1971         }
1972         pj_bzero(*desc, sizeof(*desc));
1973
1974         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1975                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1976         }
1977 }
1978
1979 int ast_sip_thread_is_servant(void)
1980 {
1981         uint32_t *servant_id;
1982
1983         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1984         if (!servant_id) {
1985                 return 0;
1986         }
1987
1988         return *servant_id == SIP_SERVANT_ID;
1989 }
1990
1991 void *ast_sip_dict_get(void *ht, const char *key)
1992 {
1993         unsigned int hval = 0;
1994
1995         if (!ht) {
1996                 return NULL;
1997         }
1998
1999         return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
2000 }
2001
2002 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
2003                        const char *key, void *val)
2004 {
2005         if (!ht) {
2006                 ht = pj_hash_create(pool, 11);
2007         }
2008
2009         pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
2010
2011         return ht;
2012 }
2013
2014 static void remove_request_headers(pjsip_endpoint *endpt)
2015 {
2016         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
2017         pjsip_hdr *iter = request_headers->next;
2018
2019         while (iter != request_headers) {
2020                 pjsip_hdr *to_erase = iter;
2021                 iter = iter->next;
2022                 pj_list_erase(to_erase);
2023         }
2024 }
2025
2026 static int load_module(void)
2027 {
2028         /* The third parameter is just copied from
2029          * example code from PJLIB. This can be adjusted
2030          * if necessary.
2031          */
2032         pj_status_t status;
2033         struct ast_threadpool_options options;
2034
2035         if (pj_init() != PJ_SUCCESS) {
2036                 return AST_MODULE_LOAD_DECLINE;
2037         }
2038
2039         if (pjlib_util_init() != PJ_SUCCESS) {
2040                 pj_shutdown();
2041                 return AST_MODULE_LOAD_DECLINE;
2042         }
2043
2044         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
2045         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
2046                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
2047                 pj_caching_pool_destroy(&caching_pool);
2048                 return AST_MODULE_LOAD_DECLINE;
2049         }
2050
2051         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2052          * we need to stop PJSIP from doing it automatically
2053          */
2054         remove_request_headers(ast_pjsip_endpoint);
2055
2056         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2057         if (!memory_pool) {
2058                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2059                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2060                 ast_pjsip_endpoint = NULL;
2061                 pj_caching_pool_destroy(&caching_pool);
2062                 return AST_MODULE_LOAD_DECLINE;
2063         }
2064
2065         if (ast_sip_initialize_system()) {
2066                 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2067                 pj_pool_release(memory_pool);
2068                 memory_pool = NULL;
2069                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2070                 ast_pjsip_endpoint = NULL;
2071                 pj_caching_pool_destroy(&caching_pool);
2072                 return AST_MODULE_LOAD_DECLINE;
2073         }
2074
2075         sip_get_threadpool_options(&options);
2076         options.thread_start = sip_thread_start;
2077         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2078         if (!sip_threadpool) {
2079                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2080                 pj_pool_release(memory_pool);
2081                 memory_pool = NULL;
2082                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2083                 ast_pjsip_endpoint = NULL;
2084                 pj_caching_pool_destroy(&caching_pool);
2085                 return AST_MODULE_LOAD_DECLINE;
2086         }
2087
2088         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2089         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2090
2091         monitor_continue = 1;
2092         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2093                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2094         if (status != PJ_SUCCESS) {
2095                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2096                 pj_pool_release(memory_pool);
2097                 memory_pool = NULL;
2098                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2099                 ast_pjsip_endpoint = NULL;
2100                 pj_caching_pool_destroy(&caching_pool);
2101                 return AST_MODULE_LOAD_DECLINE;
2102         }
2103
2104         ast_sip_initialize_global_headers();
2105
2106         if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2107                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2108                 ast_sip_destroy_global_headers();
2109                 stop_monitor_thread();
2110                 pj_pool_release(memory_pool);
2111                 memory_pool = NULL;
2112                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2113                 ast_pjsip_endpoint = NULL;
2114                 pj_caching_pool_destroy(&caching_pool);
2115                 return AST_MODULE_LOAD_DECLINE;
2116         }
2117
2118         if (ast_sip_initialize_distributor()) {
2119                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2120                 ast_res_pjsip_destroy_configuration();
2121                 ast_sip_destroy_global_headers();
2122                 stop_monitor_thread();
2123                 pj_pool_release(memory_pool);
2124                 memory_pool = NULL;
2125                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2126                 ast_pjsip_endpoint = NULL;
2127                 pj_caching_pool_destroy(&caching_pool);
2128                 return AST_MODULE_LOAD_DECLINE;
2129         }
2130
2131         if (ast_sip_initialize_outbound_authentication()) {
2132                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2133                 ast_sip_destroy_distributor();
2134                 ast_res_pjsip_destroy_configuration();
2135                 ast_sip_destroy_global_headers();
2136                 stop_monitor_thread();
2137                 pj_pool_release(memory_pool);
2138                 memory_pool = NULL;
2139                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2140                 ast_pjsip_endpoint = NULL;
2141                 pj_caching_pool_destroy(&caching_pool);
2142                 return AST_MODULE_LOAD_DECLINE;
2143         }
2144
2145         ast_res_pjsip_init_options_handling(0);
2146
2147         ast_module_ref(ast_module_info->self);
2148
2149         return AST_MODULE_LOAD_SUCCESS;
2150 }
2151
2152 static int reload_module(void)
2153 {
2154         if (ast_res_pjsip_reload_configuration()) {
2155                 return AST_MODULE_LOAD_DECLINE;
2156         }
2157         ast_res_pjsip_init_options_handling(1);
2158         return 0;
2159 }
2160
2161 static int unload_module(void)
2162 {
2163         /* This will never get called as this module can't be unloaded */
2164         return 0;
2165 }
2166
2167 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2168                 .load = load_module,
2169                 .unload = unload_module,
2170                 .reload = reload_module,
2171                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
2172 );