3ef203f85faa82247ad43f320d39e054886f8753
[asterisk/asterisk.git] / res / res_pjsip.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2013, Digium, Inc.
5  *
6  * Mark Michelson <mmichelson@digium.com>
7  *
8  * See http://www.asterisk.org for more information about
9  * the Asterisk project. Please do not directly contact
10  * any of the maintainers of this project for assistance;
11  * the project provides a web site, mailing lists and IRC
12  * channels for your use.
13  *
14  * This program is free software, distributed under the terms of
15  * the GNU General Public License Version 2. See the LICENSE file
16  * at the top of the source tree.
17  */
18
19 #include "asterisk.h"
20
21 #include <pjsip.h>
22 /* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
23 #include <pjsip_simple.h>
24 #include <pjlib.h>
25
26 #include "asterisk/res_pjsip.h"
27 #include "res_pjsip/include/res_pjsip_private.h"
28 #include "asterisk/linkedlists.h"
29 #include "asterisk/logger.h"
30 #include "asterisk/lock.h"
31 #include "asterisk/utils.h"
32 #include "asterisk/astobj2.h"
33 #include "asterisk/module.h"
34 #include "asterisk/threadpool.h"
35 #include "asterisk/taskprocessor.h"
36 #include "asterisk/uuid.h"
37 #include "asterisk/sorcery.h"
38
39 /*** MODULEINFO
40         <depend>pjproject</depend>
41         <depend>res_sorcery_config</depend>
42         <support_level>core</support_level>
43  ***/
44
45 /*** DOCUMENTATION
46         <configInfo name="res_pjsip" language="en_US">
47                 <synopsis>SIP Resource using PJProject</synopsis>
48                 <configFile name="pjsip.conf">
49                         <configObject name="endpoint">
50                                 <synopsis>Endpoint</synopsis>
51                                 <description><para>
52                                         The <emphasis>Endpoint</emphasis> is the primary configuration object.
53                                         It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
54                                         dialable entries of their own. Communication with another SIP device is
55                                         accomplished via Addresses of Record (AoRs) which have one or more
56                                         contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
57                                         use a <literal>transport</literal> will default to first transport found
58                                         in <filename>pjsip.conf</filename> that matches its type.
59                                         </para>
60                                         <para>Example: An Endpoint has been configured with no transport.
61                                         When it comes time to call an AoR, PJSIP will find the
62                                         first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
63                                         will use the first IPv6 transport and try to send the request.
64                                         </para>
65                                         <para>If the anonymous endpoint identifier is in use an endpoint with the name
66                                         "anonymous@domain" will be searched for as a last resort. If this is not found
67                                         it will fall back to searching for "anonymous". If neither endpoints are found
68                                         the anonymous endpoint identifier will not return an endpoint and anonymous
69                                         calling will not be possible.
70                                         </para>
71                                 </description>
72                                 <configOption name="100rel" default="yes">
73                                         <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
74                                         <description>
75                                                 <enumlist>
76                                                         <enum name="no" />
77                                                         <enum name="required" />
78                                                         <enum name="yes" />
79                                                 </enumlist>
80                                         </description>
81                                 </configOption>
82                                 <configOption name="aggregate_mwi" default="yes">
83                                         <synopsis></synopsis>
84                                         <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
85                                         waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
86                                         individual NOTIFYs are sent for each mailbox.</para></description>
87                                 </configOption>
88                                 <configOption name="allow">
89                                         <synopsis>Media Codec(s) to allow</synopsis>
90                                 </configOption>
91                                 <configOption name="aors">
92                                         <synopsis>AoR(s) to be used with the endpoint</synopsis>
93                                         <description><para>
94                                                 List of comma separated AoRs that the endpoint should be associated with.
95                                         </para></description>
96                                 </configOption>
97                                 <configOption name="auth">
98                                         <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
99                                         <description><para>
100                                                 This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
101                                                 in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
102                                                 </para><para>
103                                                 Endpoints without an <literal>authentication</literal> object
104                                                 configured will allow connections without vertification.
105                                         </para></description>
106                                 </configOption>
107                                 <configOption name="callerid">
108                                         <synopsis>CallerID information for the endpoint</synopsis>
109                                         <description><para>
110                                                 Must be in the format <literal>Name &lt;Number&gt;</literal>,
111                                                 or only <literal>&lt;Number&gt;</literal>.
112                                         </para></description>
113                                 </configOption>
114                                 <configOption name="callerid_privacy">
115                                         <synopsis>Default privacy level</synopsis>
116                                         <description>
117                                                 <enumlist>
118                                                         <enum name="allowed_not_screened" />
119                                                         <enum name="allowed_passed_screened" />
120                                                         <enum name="allowed_failed_screened" />
121                                                         <enum name="allowed" />
122                                                         <enum name="prohib_not_screened" />
123                                                         <enum name="prohib_passed_screened" />
124                                                         <enum name="prohib_failed_screened" />
125                                                         <enum name="prohib" />
126                                                         <enum name="unavailable" />
127                                                 </enumlist>
128                                         </description>
129                                 </configOption>
130                                 <configOption name="callerid_tag">
131                                         <synopsis>Internal id_tag for the endpoint</synopsis>
132                                 </configOption>
133                                 <configOption name="context">
134                                         <synopsis>Dialplan context for inbound sessions</synopsis>
135                                 </configOption>
136                                 <configOption name="direct_media_glare_mitigation" default="none">
137                                         <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
138                                         <description>
139                                                 <para>
140                                                 This setting attempts to avoid creating INVITE glare scenarios
141                                                 by disabling direct media reINVITEs in one direction thereby allowing
142                                                 designated servers (according to this option) to initiate direct
143                                                 media reINVITEs without contention and significantly reducing call
144                                                 setup time.
145                                                 </para>
146                                                 <para>
147                                                 A more detailed description of how this option functions can be found on
148                                                 the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
149                                                 </para>
150                                                 <enumlist>
151                                                         <enum name="none" />
152                                                         <enum name="outgoing" />
153                                                         <enum name="incoming" />
154                                                 </enumlist>
155                                         </description>
156                                 </configOption>
157                                 <configOption name="direct_media_method" default="invite">
158                                         <synopsis>Direct Media method type</synopsis>
159                                         <description>
160                                                 <para>Method for setting up Direct Media between endpoints.</para>
161                                                 <enumlist>
162                                                         <enum name="invite" />
163                                                         <enum name="reinvite">
164                                                                 <para>Alias for the <literal>invite</literal> value.</para>
165                                                         </enum>
166                                                         <enum name="update" />
167                                                 </enumlist>
168                                         </description>
169                                 </configOption>
170                                 <configOption name="connected_line_method" default="invite">
171                                         <synopsis>Connected line method type</synopsis>
172                                         <description>
173                                                 <para>Method used when updating connected line information.</para>
174                                                 <enumlist>
175                                                         <enum name="invite" />
176                                                         <enum name="reinvite">
177                                                                 <para>Alias for the <literal>invite</literal> value.</para>
178                                                         </enum>
179                                                         <enum name="update" />
180                                                 </enumlist>
181                                         </description>
182                                 </configOption>
183                                 <configOption name="direct_media" default="yes">
184                                         <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
185                                 </configOption>
186                                 <configOption name="disable_direct_media_on_nat" default="no">
187                                         <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
188                                 </configOption>
189                                 <configOption name="disallow">
190                                         <synopsis>Media Codec(s) to disallow</synopsis>
191                                 </configOption>
192                                 <configOption name="dtmf_mode" default="rfc4733">
193                                         <synopsis>DTMF mode</synopsis>
194                                         <description>
195                                                 <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
196                                                 <enumlist>
197                                                         <enum name="rfc4733">
198                                                                 <para>DTMF is sent out of band of the main audio stream.This
199                                                                 supercedes the older <emphasis>RFC-2833</emphasis> used within
200                                                                 the older <literal>chan_sip</literal>.</para>
201                                                         </enum>
202                                                         <enum name="inband">
203                                                                 <para>DTMF is sent as part of audio stream.</para>
204                                                         </enum>
205                                                         <enum name="info">
206                                                                 <para>DTMF is sent as SIP INFO packets.</para>
207                                                         </enum>
208                                                 </enumlist>
209                                         </description>
210                                 </configOption>
211                                 <configOption name="media_address">
212                                         <synopsis>IP address used in SDP for media handling</synopsis>
213                                         <description><para>
214                                                 At the time of SDP creation, the IP address defined here will be used as
215                                                 the media address for individual streams in the SDP.
216                                         </para>
217                                         <note><para>
218                                                 Be aware that the <literal>external_media_address</literal> option, set in Transport
219                                                 configuration, can also affect the final media address used in the SDP.
220                                         </para></note>
221                                         </description>
222                                 </configOption>
223                                 <configOption name="force_rport" default="yes">
224                                         <synopsis>Force use of return port</synopsis>
225                                 </configOption>
226                                 <configOption name="ice_support" default="no">
227                                         <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
228                                 </configOption>
229                                 <configOption name="identify_by" default="username,location">
230                                         <synopsis>Way(s) for Endpoint to be identified</synopsis>
231                                         <description><para>
232                                                 An endpoint can be identified in multiple ways. Currently, the only supported
233                                                 option is <literal>username</literal>, which matches the endpoint based on the
234                                                 username in the From header.
235                                                 </para>
236                                                 <note><para>Endpoints can also be identified by IP address; however, that method
237                                                 of identification is not handled by this configuration option. See the documentation
238                                                 for the <literal>identify</literal> configuration section for more details on that
239                                                 method of endpoint identification. If this option is set to <literal>username</literal>
240                                                 and an <literal>identify</literal> configuration section exists for the endpoint, then
241                                                 the endpoint can be identified in multiple ways.</para></note>
242                                                 <enumlist>
243                                                         <enum name="username" />
244                                                 </enumlist>
245                                         </description>
246                                 </configOption>
247                                 <configOption name="redirect_method">
248                                         <synopsis>How redirects received from an endpoint are handled</synopsis>
249                                         <description><para>
250                                                 When a redirect is received from an endpoint there are multiple ways it can be handled.
251                                                 If this option is set to <literal>user</literal> the user portion of the redirect target
252                                                 is treated as an extension within the dialplan and dialed using a Local channel. If this option
253                                                 is set to <literal>uri_core</literal> the target URI is returned to the dialing application
254                                                 which dials it using the PJSIP channel driver and endpoint originally used. If this option is
255                                                 set to <literal>uri_pjsip</literal> the redirect occurs within chan_pjsip itself and is not exposed
256                                                 to the core at all. The <literal>uri_pjsip</literal> option has the benefit of being more efficient
257                                                 and also supporting multiple potential redirect targets. The con is that since redirection occurs
258                                                 within chan_pjsip redirecting information is not forwarded and redirection can not be
259                                                 prevented.
260                                                 </para>
261                                                 <enumlist>
262                                                         <enum name="user" />
263                                                         <enum name="uri_core" />
264                                                         <enum name="uri_pjsip" />
265                                                 </enumlist>
266                                         </description>
267                                 </configOption>
268                                 <configOption name="mailboxes">
269                                         <synopsis>NOTIFY the endpoint when state changes for any of the specified mailboxes</synopsis>
270                                         <description><para>
271                                                 Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state
272                                                 changes happen for any of the specified mailboxes. More than one mailbox can be
273                                                 specified with a comma-delimited string. app_voicemail mailboxes must be specified
274                                                 as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by
275                                                 external sources, such as through the res_external_mwi module, you must specify
276                                                 strings supported by the external system.
277                                         </para><para>
278                                                 For endpoints that SUBSCRIBE for MWI, use the <literal>mailboxes</literal> option in your AOR
279                                                 configuration.
280                                         </para></description>
281                                 </configOption>
282                                 <configOption name="moh_suggest" default="default">
283                                         <synopsis>Default Music On Hold class</synopsis>
284                                 </configOption>
285                                 <configOption name="outbound_auth">
286                                         <synopsis>Authentication object used for outbound requests</synopsis>
287                                 </configOption>
288                                 <configOption name="outbound_proxy">
289                                         <synopsis>Proxy through which to send requests, a full SIP URI must be provided</synopsis>
290                                 </configOption>
291                                 <configOption name="rewrite_contact">
292                                         <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
293                                         <description><para>
294                                                 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
295                                                 source IP address and port. This option does not affect outbound messages send to this
296                                                 endpoint.
297                                         </para></description>
298                                 </configOption>
299                                 <configOption name="rtp_ipv6" default="no">
300                                         <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
301                                 </configOption>
302                                 <configOption name="rtp_symmetric" default="no">
303                                         <synopsis>Enforce that RTP must be symmetric</synopsis>
304                                 </configOption>
305                                 <configOption name="send_diversion" default="yes">
306                                         <synopsis>Send the Diversion header, conveying the diversion
307                                         information to the called user agent</synopsis>
308                                 </configOption>
309                                 <configOption name="send_pai" default="no">
310                                         <synopsis>Send the P-Asserted-Identity header</synopsis>
311                                 </configOption>
312                                 <configOption name="send_rpid" default="no">
313                                         <synopsis>Send the Remote-Party-ID header</synopsis>
314                                 </configOption>
315                                 <configOption name="timers_min_se" default="90">
316                                         <synopsis>Minimum session timers expiration period</synopsis>
317                                         <description><para>
318                                                 Minimium session timer expiration period. Time in seconds.
319                                         </para></description>
320                                 </configOption>
321                                 <configOption name="timers" default="yes">
322                                         <synopsis>Session timers for SIP packets</synopsis>
323                                         <description>
324                                                 <enumlist>
325                                                         <enum name="forced" />
326                                                         <enum name="no" />
327                                                         <enum name="required" />
328                                                         <enum name="yes" />
329                                                 </enumlist>
330                                         </description>
331                                 </configOption>
332                                 <configOption name="timers_sess_expires" default="1800">
333                                         <synopsis>Maximum session timer expiration period</synopsis>
334                                         <description><para>
335                                                 Maximium session timer expiration period. Time in seconds.
336                                         </para></description>
337                                 </configOption>
338                                 <configOption name="transport">
339                                         <synopsis>Desired transport configuration</synopsis>
340                                         <description><para>
341                                                 This will set the desired transport configuration to send SIP data through.
342                                                 </para>
343                                                 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
344                                                 to the first configured transport in <filename>pjsip.conf</filename> which is
345                                                 valid for the URI we are trying to contact.
346                                                 </para></warning>
347                                                 <warning><para>Transport configuration is not affected by reloads. In order to
348                                                 change transports, a full Asterisk restart is required</para></warning>
349                                         </description>
350                                 </configOption>
351                                 <configOption name="trust_id_inbound" default="no">
352                                         <synopsis>Accept identification information received from this endpoint</synopsis>
353                                         <description><para>This option determines whether Asterisk will accept
354                                         identification from the endpoint from headers such as P-Asserted-Identity
355                                         or Remote-Party-ID header. This option applies both to calls originating from the
356                                         endpoint and calls originating from Asterisk. If <literal>no</literal>, the
357                                         configured Caller-ID from pjsip.conf will always be used as the identity for
358                                         the endpoint.</para></description>
359                                 </configOption>
360                                 <configOption name="trust_id_outbound" default="no">
361                                         <synopsis>Send private identification details to the endpoint.</synopsis>
362                                         <description><para>This option determines whether res_pjsip will send private
363                                         identification information to the endpoint. If <literal>no</literal>,
364                                         private Caller-ID information will not be forwarded to the endpoint.
365                                         "Private" in this case refers to any method of restricting identification.
366                                         Example: setting <replaceable>callerid_privacy</replaceable> to any
367                                         <literal>prohib</literal> variation.
368                                         Example: If <replaceable>trust_id_inbound</replaceable> is set to
369                                         <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
370                                         header in a SIP request or response would indicate the identification
371                                         provided in the request is private.</para></description>
372                                 </configOption>
373                                 <configOption name="type">
374                                         <synopsis>Must be of type 'endpoint'.</synopsis>
375                                 </configOption>
376                                 <configOption name="use_ptime" default="no">
377                                         <synopsis>Use Endpoint's requested packetisation interval</synopsis>
378                                 </configOption>
379                                 <configOption name="use_avpf" default="no">
380                                         <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
381                                         endpoint.</synopsis>
382                                         <description><para>
383                                                 If set to <literal>yes</literal>, res_pjsip will use the AVPF or SAVPF RTP
384                                                 profile for all media offers on outbound calls and media updates and will
385                                                 decline media offers not using the AVPF or SAVPF profile.
386                                         </para><para>
387                                                 If set to <literal>no</literal>, res_pjsip will use the AVP or SAVP RTP
388                                                 profile for all media offers on outbound calls and media updates, but will
389                                                 accept either the AVP/AVPF or SAVP/SAVPF RTP profile for all inbound
390                                                 media offers.
391                                         </para></description>
392                                 </configOption>
393                                 <configOption name="media_encryption" default="no">
394                                         <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
395                                         for this endpoint.</synopsis>
396                                         <description>
397                                                 <enumlist>
398                                                         <enum name="no"><para>
399                                                                 res_pjsip will offer no encryption and allow no encryption to be setup.
400                                                         </para></enum>
401                                                         <enum name="sdes"><para>
402                                                                 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
403                                                                 transport should be used in conjunction with this option to prevent
404                                                                 exposure of media encryption keys.
405                                                         </para></enum>
406                                                         <enum name="dtls"><para>
407                                                                 res_pjsip will offer DTLS-SRTP setup.
408                                                         </para></enum>
409                                                 </enumlist>
410                                         </description>
411                                 </configOption>
412                                 <configOption name="inband_progress" default="no">
413                                         <synopsis>Determines whether chan_pjsip will indicate ringing using inband
414                                             progress.</synopsis>
415                                         <description><para>
416                                                 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
417                                                 when told to indicate ringing and will immediately start sending ringing
418                                                 as audio.
419                                         </para><para>
420                                                 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
421                                                 to indicate ringing and will NOT send it as audio.
422                                         </para></description>
423                                 </configOption>
424                                 <configOption name="call_group">
425                                         <synopsis>The numeric pickup groups for a channel.</synopsis>
426                                         <description><para>
427                                                 Can be set to a comma separated list of numbers or ranges between the values
428                                                 of 0-63 (maximum of 64 groups).
429                                         </para></description>
430                                 </configOption>
431                                 <configOption name="pickup_group">
432                                         <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
433                                         <description><para>
434                                                 Can be set to a comma separated list of numbers or ranges between the values
435                                                 of 0-63 (maximum of 64 groups).
436                                         </para></description>
437                                 </configOption>
438                                 <configOption name="named_call_group">
439                                         <synopsis>The named pickup groups for a channel.</synopsis>
440                                         <description><para>
441                                                 Can be set to a comma separated list of case sensitive strings limited by
442                                                 supported line length.
443                                         </para></description>
444                                 </configOption>
445                                 <configOption name="named_pickup_group">
446                                         <synopsis>The named pickup groups that a channel can pickup.</synopsis>
447                                         <description><para>
448                                                 Can be set to a comma separated list of case sensitive strings limited by
449                                                 supported line length.
450                                         </para></description>
451                                 </configOption>
452                                 <configOption name="device_state_busy_at" default="0">
453                                         <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
454                                         <description><para>
455                                                 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
456                                                 PJSIP channel driver will return busy as the device state instead of in use.
457                                         </para></description>
458                                 </configOption>
459                                 <configOption name="t38_udptl" default="no">
460                                         <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
461                                         <description><para>
462                                                 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
463                                                 and relayed.
464                                         </para></description>
465                                 </configOption>
466                                 <configOption name="t38_udptl_ec" default="none">
467                                         <synopsis>T.38 UDPTL error correction method</synopsis>
468                                         <description>
469                                                 <enumlist>
470                                                         <enum name="none"><para>
471                                                                 No error correction should be used.
472                                                         </para></enum>
473                                                         <enum name="fec"><para>
474                                                                 Forward error correction should be used.
475                                                         </para></enum>
476                                                         <enum name="redundancy"><para>
477                                                                 Redundacy error correction should be used.
478                                                         </para></enum>
479                                                 </enumlist>
480                                         </description>
481                                 </configOption>
482                                 <configOption name="t38_udptl_maxdatagram" default="0">
483                                         <synopsis>T.38 UDPTL maximum datagram size</synopsis>
484                                         <description><para>
485                                                 This option can be set to override the maximum datagram of a remote endpoint for broken
486                                                 endpoints.
487                                         </para></description>
488                                 </configOption>
489                                 <configOption name="fax_detect" default="no">
490                                         <synopsis>Whether CNG tone detection is enabled</synopsis>
491                                         <description><para>
492                                                 This option can be set to send the session to the fax extension when a CNG tone is
493                                                 detected.
494                                         </para></description>
495                                 </configOption>
496                                 <configOption name="t38_udptl_nat" default="no">
497                                         <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
498                                         <description><para>
499                                                 When enabled the UDPTL stack will send UDPTL packets to the source address of
500                                                 received packets.
501                                         </para></description>
502                                 </configOption>
503                                 <configOption name="t38_udptl_ipv6" default="no">
504                                         <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
505                                         <description><para>
506                                                 When enabled the UDPTL stack will use IPv6.
507                                         </para></description>
508                                 </configOption>
509                                 <configOption name="tone_zone">
510                                         <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
511                                 </configOption>
512                                 <configOption name="language">
513                                         <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
514                                 </configOption>
515                                 <configOption name="one_touch_recording" default="no">
516                                         <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
517                                         <see-also>
518                                                 <ref type="configOption">record_on_feature</ref>
519                                                 <ref type="configOption">record_off_feature</ref>
520                                         </see-also>
521                                 </configOption>
522                                 <configOption name="record_on_feature" default="automixmon">
523                                         <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
524                                         <description>
525                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
526                                                 feature will be enabled for the channel. The feature designated here can be any built-in
527                                                 or dynamic feature defined in features.conf.</para>
528                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
529                                         </description>
530                                         <see-also>
531                                                 <ref type="configOption">one_touch_recording</ref>
532                                                 <ref type="configOption">record_off_feature</ref>
533                                         </see-also>
534                                 </configOption>
535                                 <configOption name="record_off_feature" default="automixmon">
536                                         <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
537                                         <description>
538                                                 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
539                                                 feature will be enabled for the channel. The feature designated here can be any built-in
540                                                 or dynamic feature defined in features.conf.</para>
541                                                 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
542                                         </description>
543                                         <see-also>
544                                                 <ref type="configOption">one_touch_recording</ref>
545                                                 <ref type="configOption">record_on_feature</ref>
546                                         </see-also>
547                                 </configOption>
548                                 <configOption name="rtp_engine" default="asterisk">
549                                         <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
550                                 </configOption>
551                                 <configOption name="allow_transfer" default="yes">
552                                         <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
553                                 </configOption>
554                                 <configOption name="sdp_owner" default="-">
555                                         <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
556                                 </configOption>
557                                 <configOption name="sdp_session" default="Asterisk">
558                                         <synopsis>String used for the SDP session (s=) line.</synopsis>
559                                 </configOption>
560                                 <configOption name="tos_audio">
561                                         <synopsis>DSCP TOS bits 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="tos_video">
567                                         <synopsis>DSCP TOS bits 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="cos_audio">
573                                         <synopsis>Priority for audio streams</synopsis>
574                                         <description><para>
575                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
576                                         </para></description>
577                                 </configOption>
578                                 <configOption name="cos_video">
579                                         <synopsis>Priority for video streams</synopsis>
580                                         <description><para>
581                                                 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
582                                         </para></description>
583                                 </configOption>
584                                 <configOption name="allow_subscribe" default="yes">
585                                         <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
586                                 </configOption>
587                                 <configOption name="sub_min_expiry" default="60">
588                                         <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
589                                 </configOption>
590                                 <configOption name="from_user">
591                                         <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
592                                 </configOption>
593                                 <configOption name="mwi_from_user">
594                                         <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
595                                 </configOption>
596                                 <configOption name="from_domain">
597                                         <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
598                                 </configOption>
599                                 <configOption name="dtls_verify">
600                                         <synopsis>Verify that the provided peer certificate is valid</synopsis>
601                                         <description><para>
602                                                 This option only applies if <replaceable>media_encryption</replaceable> is
603                                                 set to <literal>dtls</literal>.
604                                         </para></description>
605                                 </configOption>
606                                 <configOption name="dtls_rekey">
607                                         <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
608                                         <description><para>
609                                                 This option only applies if <replaceable>media_encryption</replaceable> is
610                                                 set to <literal>dtls</literal>.
611                                         </para><para>
612                                                 If this is not set or the value provided is 0 rekeying will be disabled.
613                                         </para></description>
614                                 </configOption>
615                                 <configOption name="dtls_cert_file">
616                                         <synopsis>Path to certificate file to present to peer</synopsis>
617                                         <description><para>
618                                                 This option only applies if <replaceable>media_encryption</replaceable> is
619                                                 set to <literal>dtls</literal>.
620                                         </para></description>
621                                 </configOption>
622                                 <configOption name="dtls_private_key">
623                                         <synopsis>Path to private key for certificate file</synopsis>
624                                         <description><para>
625                                                 This option only applies if <replaceable>media_encryption</replaceable> is
626                                                 set to <literal>dtls</literal>.
627                                         </para></description>
628                                 </configOption>
629                                 <configOption name="dtls_cipher">
630                                         <synopsis>Cipher to use for DTLS negotiation</synopsis>
631                                         <description><para>
632                                                 This option only applies if <replaceable>media_encryption</replaceable> is
633                                                 set to <literal>dtls</literal>.
634                                         </para><para>
635                                                 Many options for acceptable ciphers. See link for more:
636                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
637                                         </para></description>
638                                 </configOption>
639                                 <configOption name="dtls_ca_file">
640                                         <synopsis>Path to certificate authority certificate</synopsis>
641                                         <description><para>
642                                                 This option only applies if <replaceable>media_encryption</replaceable> is
643                                                 set to <literal>dtls</literal>.
644                                         </para></description>
645                                 </configOption>
646                                 <configOption name="dtls_ca_path">
647                                         <synopsis>Path to a directory containing certificate authority certificates</synopsis>
648                                         <description><para>
649                                                 This option only applies if <replaceable>media_encryption</replaceable> is
650                                                 set to <literal>dtls</literal>.
651                                         </para></description>
652                                 </configOption>
653                                 <configOption name="dtls_setup">
654                                         <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
655                                         <description>
656                                                 <para>
657                                                         This option only applies if <replaceable>media_encryption</replaceable> is
658                                                         set to <literal>dtls</literal>.
659                                                 </para>
660                                                 <enumlist>
661                                                         <enum name="active"><para>
662                                                                 res_pjsip will make a connection to the peer.
663                                                         </para></enum>
664                                                         <enum name="passive"><para>
665                                                                 res_pjsip will accept connections from the peer.
666                                                         </para></enum>
667                                                         <enum name="actpass"><para>
668                                                                 res_pjsip will offer and accept connections from the peer.
669                                                         </para></enum>
670                                                 </enumlist>
671                                         </description>
672                                 </configOption>
673                                 <configOption name="srtp_tag_32">
674                                         <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
675                                         <description><para>
676                                                 This option only applies if <replaceable>media_encryption</replaceable> is
677                                                 set to <literal>sdes</literal> or <literal>dtls</literal>.
678                                         </para></description>
679                                 </configOption>
680                                 <configOption name="set_var">
681                                         <synopsis>Variable set on a channel involving the endpoint.</synopsis>
682                                         <description><para>
683                                                 When a new channel is created using the endpoint set the specified
684                                                 variable(s) on that channel. For multiple channel variables specify
685                                                 multiple 'set_var'(s).
686                                         </para></description>
687                                 </configOption>
688                         </configObject>
689                         <configObject name="auth">
690                                 <synopsis>Authentication type</synopsis>
691                                 <description><para>
692                                         Authentication objects hold the authentication information for use
693                                         by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
694                                         This also allows for multiple objects to use a single auth object. See
695                                         the <literal>auth_type</literal> config option for password style choices.
696                                 </para></description>
697                                 <configOption name="auth_type" default="userpass">
698                                         <synopsis>Authentication type</synopsis>
699                                         <description><para>
700                                                 This option specifies which of the password style config options should be read
701                                                 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
702                                                 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
703                                                 from 'md5_cred'.
704                                                 </para>
705                                                 <enumlist>
706                                                         <enum name="md5"/>
707                                                         <enum name="userpass"/>
708                                                 </enumlist>
709                                         </description>
710                                 </configOption>
711                                 <configOption name="nonce_lifetime" default="32">
712                                         <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
713                                 </configOption>
714                                 <configOption name="md5_cred">
715                                         <synopsis>MD5 Hash used for authentication.</synopsis>
716                                         <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
717                                 </configOption>
718                                 <configOption name="password">
719                                         <synopsis>PlainText password used for authentication.</synopsis>
720                                         <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
721                                 </configOption>
722                                 <configOption name="realm" default="asterisk">
723                                         <synopsis>SIP realm for endpoint</synopsis>
724                                 </configOption>
725                                 <configOption name="type">
726                                         <synopsis>Must be 'auth'</synopsis>
727                                 </configOption>
728                                 <configOption name="username">
729                                         <synopsis>Username to use for account</synopsis>
730                                 </configOption>
731                         </configObject>
732                         <configObject name="domain_alias">
733                                 <synopsis>Domain Alias</synopsis>
734                                 <description><para>
735                                         Signifies that a domain is an alias. If the domain on a session is
736                                         not found to match an AoR then this object is used to see if we have
737                                         an alias for the AoR to which the endpoint is binding. This objects
738                                         name as defined in configuration should be the domain alias and a
739                                         config option is provided to specify the domain to be aliased.
740                                 </para></description>
741                                 <configOption name="type">
742                                         <synopsis>Must be of type 'domain_alias'.</synopsis>
743                                 </configOption>
744                                 <configOption name="domain">
745                                         <synopsis>Domain to be aliased</synopsis>
746                                 </configOption>
747                         </configObject>
748                         <configObject name="transport">
749                                 <synopsis>SIP Transport</synopsis>
750                                 <description><para>
751                                         <emphasis>Transports</emphasis>
752                                         </para>
753                                         <para>There are different transports and protocol derivatives
754                                                 supported by <literal>res_pjsip</literal>. They are in order of
755                                                 preference: UDP, TCP, and WebSocket (WS).</para>
756                                         <note><para>Changes to transport configuration in pjsip.conf will only be
757                                                 effected on a complete restart of Asterisk. A module reload
758                                                 will not suffice.</para></note>
759                                 </description>
760                                 <configOption name="async_operations" default="1">
761                                         <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
762                                 </configOption>
763                                 <configOption name="bind">
764                                         <synopsis>IP Address and optional port to bind to for this transport</synopsis>
765                                 </configOption>
766                                 <configOption name="ca_list_file">
767                                         <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
768                                 </configOption>
769                                 <configOption name="cert_file">
770                                         <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
771                                 </configOption>
772                                 <configOption name="cipher">
773                                         <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
774                                         <description><para>
775                                                 Many options for acceptable ciphers see link for more:
776                                                 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
777                                         </para></description>
778                                 </configOption>
779                                 <configOption name="domain">
780                                         <synopsis>Domain the transport comes from</synopsis>
781                                 </configOption>
782                                 <configOption name="external_media_address">
783                                         <synopsis>External IP address to use in RTP handling</synopsis>
784                                         <description><para>
785                                                 When a request or response is sent out, if the destination of the
786                                                 message is outside the IP network defined in the option <literal>localnet</literal>,
787                                                 and the media address in the SDP is within the localnet network, then the
788                                                 media address in the SDP will be rewritten to the value defined for
789                                                 <literal>external_media_address</literal>.
790                                         </para></description>
791                                 </configOption>
792                                 <configOption name="external_signaling_address">
793                                         <synopsis>External address for SIP signalling</synopsis>
794                                 </configOption>
795                                 <configOption name="external_signaling_port" default="0">
796                                         <synopsis>External port for SIP signalling</synopsis>
797                                 </configOption>
798                                 <configOption name="method">
799                                         <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
800                                         <description>
801                                                 <enumlist>
802                                                         <enum name="default" />
803                                                         <enum name="unspecified" />
804                                                         <enum name="tlsv1" />
805                                                         <enum name="sslv2" />
806                                                         <enum name="sslv3" />
807                                                         <enum name="sslv23" />
808                                                 </enumlist>
809                                         </description>
810                                 </configOption>
811                                 <configOption name="local_net">
812                                         <synopsis>Network to consider local (used for NAT purposes).</synopsis>
813                                         <description><para>This must be in CIDR or dotted decimal format with the IP
814                                         and mask separated with a slash ('/').</para></description>
815                                 </configOption>
816                                 <configOption name="password">
817                                         <synopsis>Password required for transport</synopsis>
818                                 </configOption>
819                                 <configOption name="priv_key_file">
820                                         <synopsis>Private key file (TLS ONLY)</synopsis>
821                                 </configOption>
822                                 <configOption name="protocol" default="udp">
823                                         <synopsis>Protocol to use for SIP traffic</synopsis>
824                                         <description>
825                                                 <enumlist>
826                                                         <enum name="udp" />
827                                                         <enum name="tcp" />
828                                                         <enum name="tls" />
829                                                         <enum name="ws" />
830                                                         <enum name="wss" />
831                                                 </enumlist>
832                                         </description>
833                                 </configOption>
834                                 <configOption name="require_client_cert" default="false">
835                                         <synopsis>Require client certificate (TLS ONLY)</synopsis>
836                                 </configOption>
837                                 <configOption name="type">
838                                         <synopsis>Must be of type 'transport'.</synopsis>
839                                 </configOption>
840                                 <configOption name="verify_client" default="false">
841                                         <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
842                                 </configOption>
843                                 <configOption name="verify_server" default="false">
844                                         <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
845                                 </configOption>
846                                 <configOption name="tos" default="false">
847                                         <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
848                                         <description>
849                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
850                                         for more information on this parameter.</para>
851                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
852                                         or the <replaceable>wss</replaceable> protocols.</para></note>
853                                         </description>
854                                 </configOption>
855                                 <configOption name="cos" default="false">
856                                         <synopsis>Enable COS for the signalling sent over this transport</synopsis>
857                                         <description>
858                                         <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
859                                         for more information on this parameter.</para>
860                                         <note><para>This option does not apply to the <replaceable>ws</replaceable>
861                                         or the <replaceable>wss</replaceable> protocols.</para></note>
862                                         </description>
863                                 </configOption>
864                         </configObject>
865                         <configObject name="contact">
866                                 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
867                                 <description><para>
868                                         Contacts are a way to hide SIP URIs from the dialplan directly.
869                                         They are also used to make a group of contactable parties when
870                                         in use with <literal>AoR</literal> lists.
871                                 </para></description>
872                                 <configOption name="type">
873                                         <synopsis>Must be of type 'contact'.</synopsis>
874                                 </configOption>
875                                 <configOption name="uri">
876                                         <synopsis>SIP URI to contact peer</synopsis>
877                                 </configOption>
878                                 <configOption name="expiration_time">
879                                         <synopsis>Time to keep alive a contact</synopsis>
880                                         <description><para>
881                                                 Time to keep alive a contact. String style specification.
882                                         </para></description>
883                                 </configOption>
884                                 <configOption name="qualify_frequency" default="0">
885                                         <synopsis>Interval at which to qualify a contact</synopsis>
886                                         <description><para>
887                                                 Interval between attempts to qualify the contact for reachability.
888                                                 If <literal>0</literal> never qualify. Time in seconds.
889                                         </para></description>
890                                 </configOption>
891                                 <configOption name="outbound_proxy">
892                                         <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
893                                         <description><para>
894                                                 If set the provided URI will be used as the outbound proxy when an
895                                                 OPTIONS request is sent to a contact for qualify purposes.
896                                         </para></description>
897                                 </configOption>
898                                 <configOption name="path">
899                                         <synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
900                                 </configOption>
901                                 <configOption name="user_agent">
902                                         <synopsis>User-Agent header from registration.</synopsis>
903                                         <description><para>
904                                                 The User-Agent is automatically stored based on data present in incoming SIP
905                                                 REGISTER requests and is not intended to be configured manually.
906                                         </para></description>
907                                 </configOption>
908                         </configObject>
909                         <configObject name="aor">
910                                 <synopsis>The configuration for a location of an endpoint</synopsis>
911                                 <description><para>
912                                         An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
913                                         AoRs are specified, an endpoint will not be reachable by Asterisk.
914                                         Beyond that, an AoR has other uses within Asterisk, such as inbound
915                                         registration.
916                                         </para><para>
917                                         An <literal>AoR</literal> is a way to allow dialing a group
918                                         of <literal>Contacts</literal> that all use the same
919                                         <literal>endpoint</literal> for calls.
920                                         </para><para>
921                                         This can be used as another way of grouping a list of contacts to dial
922                                         rather than specifing them each directly when dialing via the dialplan.
923                                         This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
924                                         </para><para>
925                                         Registrations: For Asterisk to match an inbound registration to an endpoint,
926                                         the AoR object name must match the user portion of the SIP URI in the "To:"
927                                         header of the inbound SIP registration. That will usually be equivalent
928                                         to the "user name" set in your hard or soft phones configuration.
929                                 </para></description>
930                                 <configOption name="contact">
931                                         <synopsis>Permanent contacts assigned to AoR</synopsis>
932                                         <description><para>
933                                                 Contacts specified will be called whenever referenced
934                                                 by <literal>chan_pjsip</literal>.
935                                                 </para><para>
936                                                 Use a separate "contact=" entry for each contact required. Contacts
937                                                 are specified using a SIP URI.
938                                         </para></description>
939                                 </configOption>
940                                 <configOption name="default_expiration" default="3600">
941                                         <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
942                                 </configOption>
943                                 <configOption name="mailboxes">
944                                         <synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
945                                         <description><para>This option applies when an external entity subscribes to an AoR
946                                                 for Message Waiting Indications. The mailboxes specified will be subscribed to.
947                                                 More than one mailbox can be specified with a comma-delimited string.
948                                                 app_voicemail mailboxes must be specified as mailbox@context;
949                                                 for example: mailboxes=6001@default. For mailboxes provided by external sources,
950                                                 such as through the res_external_mwi module, you must specify strings supported by
951                                                 the external system.
952                                         </para><para>
953                                                 For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
954                                                 endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
955                                         </para></description>
956                                 </configOption>
957                                 <configOption name="maximum_expiration" default="7200">
958                                         <synopsis>Maximum time to keep an AoR</synopsis>
959                                         <description><para>
960                                                 Maximium time to keep a peer with explicit expiration. Time in seconds.
961                                         </para></description>
962                                 </configOption>
963                                 <configOption name="max_contacts" default="0">
964                                         <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
965                                         <description><para>
966                                                 Maximum number of contacts that can associate with this AoR. This value does
967                                                 not affect the number of contacts that can be added with the "contact" option.
968                                                 It only limits contacts added through external interaction, such as
969                                                 registration.
970                                                 </para>
971                                                 <note><para>This should be set to <literal>1</literal> and
972                                                 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
973                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
974                                                 </para></note>
975                                         </description>
976                                 </configOption>
977                                 <configOption name="minimum_expiration" default="60">
978                                         <synopsis>Minimum keep alive time for an AoR</synopsis>
979                                         <description><para>
980                                                 Minimum time to keep a peer with an explict expiration. Time in seconds.
981                                         </para></description>
982                                 </configOption>
983                                 <configOption name="remove_existing" default="no">
984                                         <synopsis>Determines whether new contacts replace existing ones.</synopsis>
985                                         <description><para>
986                                                 On receiving a new registration to the AoR should it remove
987                                                 the existing contact that was registered against it?
988                                                 </para>
989                                                 <note><para>This should be set to <literal>yes</literal> and
990                                                 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
991                                                 wish to stick with the older <literal>chan_sip</literal> behaviour.
992                                                 </para></note>
993                                         </description>
994                                 </configOption>
995                                 <configOption name="type">
996                                         <synopsis>Must be of type 'aor'.</synopsis>
997                                 </configOption>
998                                 <configOption name="qualify_frequency" default="0">
999                                         <synopsis>Interval at which to qualify an AoR</synopsis>
1000                                         <description><para>
1001                                                 Interval between attempts to qualify the AoR for reachability.
1002                                                 If <literal>0</literal> never qualify. Time in seconds.
1003                                         </para></description>
1004                                 </configOption>
1005                                 <configOption name="authenticate_qualify" default="no">
1006                                         <synopsis>Authenticates a qualify request if needed</synopsis>
1007                                         <description><para>
1008                                                 If true and a qualify request receives a challenge or authenticate response
1009                                                 authentication is attempted before declaring the contact available.
1010                                         </para></description>
1011                                 </configOption>
1012                                 <configOption name="outbound_proxy">
1013                                         <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
1014                                         <description><para>
1015                                                 If set the provided URI will be used as the outbound proxy when an
1016                                                 OPTIONS request is sent to a contact for qualify purposes.
1017                                         </para></description>
1018                                 </configOption>
1019                                 <configOption name="support_path">
1020                                         <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
1021                                         <description><para>
1022                                                 When this option is enabled, the Path headers in register requests will be saved
1023                                                 and its contents will be used in Route headers for outbound out-of-dialog requests
1024                                                 and in Path headers for outbound 200 responses. Path support will also be indicated
1025                                                 in the Supported header.
1026                                         </para></description>
1027                                 </configOption>
1028                         </configObject>
1029                         <configObject name="system">
1030                                 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
1031                                 <description><para>
1032                                         The settings in this section are global. In addition to being global, the values will
1033                                         not be re-evaluated when a reload is performed. This is because the values must be set
1034                                         before the SIP stack is initialized. The only way to reset these values is to either
1035                                         restart Asterisk, or unload res_pjsip.so and then load it again.
1036                                 </para></description>
1037                                 <configOption name="timer_t1" default="500">
1038                                         <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
1039                                         <description><para>
1040                                                 Timer T1 is the base for determining how long to wait before retransmitting
1041                                                 requests that receive no response when using an unreliable transport (e.g. UDP).
1042                                                 For more information on this timer, see RFC 3261, Section 17.1.1.1.
1043                                         </para></description>
1044                                 </configOption>
1045                                 <configOption name="timer_b" default="32000">
1046                                         <synopsis>Set transaction timer B value (milliseconds).</synopsis>
1047                                         <description><para>
1048                                                 Timer B determines the maximum amount of time to wait after sending an INVITE
1049                                                 request before terminating the transaction. It is recommended that this be set
1050                                                 to 64 * Timer T1, but it may be set higher if desired. For more information on
1051                                                 this timer, see RFC 3261, Section 17.1.1.1.
1052                                         </para></description>
1053                                 </configOption>
1054                                 <configOption name="compact_headers" default="no">
1055                                         <synopsis>Use the short forms of common SIP header names.</synopsis>
1056                                 </configOption>
1057                                 <configOption name="threadpool_initial_size" default="0">
1058                                         <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
1059                                 </configOption>
1060                                 <configOption name="threadpool_auto_increment" default="5">
1061                                         <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1062                                 </configOption>
1063                                 <configOption name="threadpool_idle_timeout" default="60">
1064                                         <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1065                                 </configOption>
1066                                 <configOption name="threadpool_max_size" default="0">
1067                                         <synopsis>Maximum number of threads in the res_pjsip threadpool.
1068                                         A value of 0 indicates no maximum.</synopsis>
1069                                 </configOption>
1070                                 <configOption name="type">
1071                                         <synopsis>Must be of type 'system'.</synopsis>
1072                                 </configOption>
1073                         </configObject>
1074                         <configObject name="global">
1075                                 <synopsis>Options that apply globally to all SIP communications</synopsis>
1076                                 <description><para>
1077                                         The settings in this section are global. Unlike options in the <literal>system</literal>
1078                                         section, these options can be refreshed by performing a reload.
1079                                 </para></description>
1080                                 <configOption name="max_forwards" default="70">
1081                                         <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1082                                 </configOption>
1083                                 <configOption name="type">
1084                                         <synopsis>Must be of type 'global'.</synopsis>
1085                                 </configOption>
1086                                 <configOption name="user_agent" default="Asterisk &lt;Asterisk Version&gt;">
1087                                         <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1088                                 </configOption>
1089                                 <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
1090                                         <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
1091                                 </configOption>
1092                                 <configOption name="debug" default="no">
1093                                         <synopsis>Enable/Disable SIP debug logging.  Valid options include yes|no or
1094                                         a host address</synopsis>
1095                                 </configOption>
1096                         </configObject>
1097                 </configFile>
1098         </configInfo>
1099         <manager name="PJSIPQualify" language="en_US">
1100                 <synopsis>
1101                         Qualify a chan_pjsip endpoint.
1102                 </synopsis>
1103                 <syntax>
1104                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1105                         <parameter name="Endpoint" required="true">
1106                                 <para>The endpoint you want to qualify.</para>
1107                         </parameter>
1108                 </syntax>
1109                 <description>
1110                         <para>Qualify a chan_pjsip endpoint.</para>
1111                 </description>
1112         </manager>
1113         <manager name="PJSIPShowEndpoints" language="en_US">
1114                 <synopsis>
1115                         Lists PJSIP endpoints.
1116                 </synopsis>
1117                 <syntax />
1118                 <description>
1119                         <para>
1120                         Provides a listing of all endpoints.  For each endpoint an <literal>EndpointList</literal> event
1121                         is raised that contains relevant attributes and status information.  Once all
1122                         endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1123                         </para>
1124                 </description>
1125         </manager>
1126         <manager name="PJSIPShowEndpoint" language="en_US">
1127                 <synopsis>
1128                         Detail listing of an endpoint and its objects.
1129                 </synopsis>
1130                 <syntax>
1131                         <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1132                         <parameter name="Endpoint" required="true">
1133                                 <para>The endpoint to list.</para>
1134                         </parameter>
1135                 </syntax>
1136                 <description>
1137                         <para>
1138                         Provides a detailed listing of options for a given endpoint.  Events are issued
1139                         showing the configuration and status of the endpoint and associated objects.  These
1140                         events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1141                         <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1142                         <literal>IdentifyDetail</literal>.  Some events may be listed multiple times if multiple objects are
1143                         associated (for instance AoRs).  Once all detail events have been raised a final
1144                         <literal>EndpointDetailComplete</literal> event is issued.
1145                         </para>
1146                 </description>
1147         </manager>
1148  ***/
1149
1150 #define MOD_DATA_CONTACT "contact"
1151
1152 static pjsip_endpoint *ast_pjsip_endpoint;
1153
1154 static struct ast_threadpool *sip_threadpool;
1155
1156 static int register_service(void *data)
1157 {
1158         pjsip_module **module = data;
1159         if (!ast_pjsip_endpoint) {
1160                 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1161                 return -1;
1162         }
1163         if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1164                 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1165                 return -1;
1166         }
1167         ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1168         ast_module_ref(ast_module_info->self);
1169         return 0;
1170 }
1171
1172 int ast_sip_register_service(pjsip_module *module)
1173 {
1174         return ast_sip_push_task_synchronous(NULL, register_service, &module);
1175 }
1176
1177 static int unregister_service(void *data)
1178 {
1179         pjsip_module **module = data;
1180         ast_module_unref(ast_module_info->self);
1181         if (!ast_pjsip_endpoint) {
1182                 return -1;
1183         }
1184         pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1185         ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1186         return 0;
1187 }
1188
1189 void ast_sip_unregister_service(pjsip_module *module)
1190 {
1191         ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1192 }
1193
1194 static struct ast_sip_authenticator *registered_authenticator;
1195
1196 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1197 {
1198         if (registered_authenticator) {
1199                 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1200                 return -1;
1201         }
1202         registered_authenticator = auth;
1203         ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1204         ast_module_ref(ast_module_info->self);
1205         return 0;
1206 }
1207
1208 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1209 {
1210         if (registered_authenticator != auth) {
1211                 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1212                                 auth, registered_authenticator);
1213                 return;
1214         }
1215         registered_authenticator = NULL;
1216         ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1217         ast_module_unref(ast_module_info->self);
1218 }
1219
1220 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1221 {
1222         if (!registered_authenticator) {
1223                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1224                 return 0;
1225         }
1226
1227         return registered_authenticator->requires_authentication(endpoint, rdata);
1228 }
1229
1230 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1231                 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1232 {
1233         if (!registered_authenticator) {
1234                 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1235                 return 0;
1236         }
1237         return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1238 }
1239
1240 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1241
1242 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1243 {
1244         if (registered_outbound_authenticator) {
1245                 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1246                 return -1;
1247         }
1248         registered_outbound_authenticator = auth;
1249         ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1250         ast_module_ref(ast_module_info->self);
1251         return 0;
1252 }
1253
1254 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1255 {
1256         if (registered_outbound_authenticator != auth) {
1257                 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1258                                 auth, registered_outbound_authenticator);
1259                 return;
1260         }
1261         registered_outbound_authenticator = NULL;
1262         ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1263         ast_module_unref(ast_module_info->self);
1264 }
1265
1266 int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
1267                 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1268 {
1269         if (!registered_outbound_authenticator) {
1270                 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1271                 return -1;
1272         }
1273         return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1274 }
1275
1276 struct endpoint_identifier_list {
1277         struct ast_sip_endpoint_identifier *identifier;
1278         AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1279 };
1280
1281 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1282
1283 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1284 {
1285         struct endpoint_identifier_list *id_list_item;
1286         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1287
1288         id_list_item = ast_calloc(1, sizeof(*id_list_item));
1289         if (!id_list_item) {
1290                 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1291                 return -1;
1292         }
1293         id_list_item->identifier = identifier;
1294
1295         AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1296         ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1297
1298         ast_module_ref(ast_module_info->self);
1299         return 0;
1300 }
1301
1302 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1303 {
1304         struct endpoint_identifier_list *iter;
1305         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1306         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1307                 if (iter->identifier == identifier) {
1308                         AST_RWLIST_REMOVE_CURRENT(list);
1309                         ast_free(iter);
1310                         ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1311                         ast_module_unref(ast_module_info->self);
1312                         break;
1313                 }
1314         }
1315         AST_RWLIST_TRAVERSE_SAFE_END;
1316 }
1317
1318 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1319 {
1320         struct endpoint_identifier_list *iter;
1321         struct ast_sip_endpoint *endpoint = NULL;
1322         SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1323         AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1324                 ast_assert(iter->identifier->identify_endpoint != NULL);
1325                 endpoint = iter->identifier->identify_endpoint(rdata);
1326                 if (endpoint) {
1327                         break;
1328                 }
1329         }
1330         return endpoint;
1331 }
1332
1333 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1334
1335 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1336 {
1337         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1338         AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1339         ast_module_ref(ast_module_info->self);
1340         return 0;
1341 }
1342
1343 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1344 {
1345         struct ast_sip_endpoint_formatter *i;
1346         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1347         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1348                 if (i == obj) {
1349                         AST_RWLIST_REMOVE_CURRENT(next);
1350                         ast_module_unref(ast_module_info->self);
1351                         break;
1352                 }
1353         }
1354         AST_RWLIST_TRAVERSE_SAFE_END;
1355 }
1356
1357 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1358                                 struct ast_sip_ami *ami, int *count)
1359 {
1360         int res = 0;
1361         struct ast_sip_endpoint_formatter *i;
1362         SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1363         *count = 0;
1364         AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1365                 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1366                         return res;
1367                 }
1368
1369                 if (!res) {
1370                         (*count)++;
1371                 }
1372         }
1373         return 0;
1374 }
1375
1376 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1377 {
1378         return ast_pjsip_endpoint;
1379 }
1380
1381 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)
1382 {
1383         pj_str_t tmp, local_addr;
1384         pjsip_uri *uri;
1385         pjsip_sip_uri *sip_uri;
1386         pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1387         int local_port;
1388         char uuid_str[AST_UUID_STR_LEN];
1389
1390         if (ast_strlen_zero(user)) {
1391                 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1392                 if (!uuid) {
1393                         return -1;
1394                 }
1395                 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1396         }
1397
1398         /* Parse the provided target URI so we can determine what transport it will end up using */
1399         pj_strdup_with_null(pool, &tmp, target);
1400
1401         if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1402             (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1403                 return -1;
1404         }
1405
1406         sip_uri = pjsip_uri_get_uri(uri);
1407
1408         /* Determine the transport type to use */
1409         if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1410                 type = PJSIP_TRANSPORT_TLS;
1411         } else if (!sip_uri->transport_param.slen) {
1412                 type = PJSIP_TRANSPORT_UDP;
1413         } else {
1414                 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1415         }
1416
1417         if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1418                 return -1;
1419         }
1420
1421         /* If the host is IPv6 turn the transport into an IPv6 version */
1422         if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1423                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1424         }
1425
1426         if (!ast_strlen_zero(domain)) {
1427                 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1428                 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1429                                 "<sip:%s@%s%s%s>",
1430                                 user,
1431                                 domain,
1432                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1433                                 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1434                 return 0;
1435         }
1436
1437         /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1438         if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1439                                                               &local_addr, &local_port) != PJ_SUCCESS) {
1440
1441                 /* If no local address can be retrieved using the transport manager use the host one */
1442                 pj_strdup(pool, &local_addr, pj_gethostname());
1443                 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1444         }
1445
1446         /* If IPv6 was specified in the transport, set the proper type */
1447         if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1448                 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1449         }
1450
1451         from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1452         from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1453                                       "<sip:%s@%s%.*s%s:%d%s%s>",
1454                                       user,
1455                                       (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1456                                       (int)local_addr.slen,
1457                                       local_addr.ptr,
1458                                       (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1459                                       local_port,
1460                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1461                                       (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1462
1463         return 0;
1464 }
1465
1466 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1467 {
1468         RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1469         const char *transport_name = endpoint->transport;
1470
1471         if (ast_strlen_zero(transport_name)) {
1472                 return 0;
1473         }
1474
1475         transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1476
1477         if (!transport || !transport->state) {
1478                 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1479                         transport_name, ast_sorcery_object_get_id(endpoint));
1480                 return -1;
1481         }
1482
1483         if (transport->state->transport) {
1484                 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1485                 selector->u.transport = transport->state->transport;
1486         } else if (transport->state->factory) {
1487                 selector->type = PJSIP_TPSELECTOR_LISTENER;
1488                 selector->u.listener = transport->state->factory;
1489         } else {
1490                 return -1;
1491         }
1492
1493         return 0;
1494 }
1495
1496 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1497 {
1498         char enclosed_uri[PJSIP_MAX_URL_SIZE];
1499         pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1500         pjsip_dialog *dlg = NULL;
1501         const char *outbound_proxy = endpoint->outbound_proxy;
1502         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1503         static const pj_str_t HCONTACT = { "Contact", 7 };
1504
1505         snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1506         pj_cstr(&remote_uri, enclosed_uri);
1507
1508         pj_cstr(&target_uri, uri);
1509
1510         if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1511                 return NULL;
1512         }
1513
1514         if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1515                 pjsip_dlg_terminate(dlg);
1516                 return NULL;
1517         }
1518
1519         if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1520                 pjsip_dlg_terminate(dlg);
1521                 return NULL;
1522         }
1523
1524         /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1525         pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1526         dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1527         dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1528
1529         /* If a request user has been specified and we are permitted to change it, do so */
1530         if (!ast_strlen_zero(request_user)) {
1531                 pjsip_sip_uri *sip_uri;
1532
1533                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1534                         sip_uri = pjsip_uri_get_uri(dlg->target);
1535                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1536                 }
1537                 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1538                         sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1539                         pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1540                 }
1541         }
1542
1543         /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1544         dlg->sess_count++;
1545
1546         pjsip_dlg_set_transport(dlg, &selector);
1547
1548         if (!ast_strlen_zero(outbound_proxy)) {
1549                 pjsip_route_hdr route_set, *route;
1550                 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1551                 pj_str_t tmp;
1552
1553                 pj_list_init(&route_set);
1554
1555                 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1556                 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1557                         dlg->sess_count--;
1558                         pjsip_dlg_terminate(dlg);
1559                         return NULL;
1560                 }
1561                 pj_list_insert_nodes_before(&route_set, route);
1562
1563                 pjsip_dlg_set_route_set(dlg, &route_set);
1564         }
1565
1566         dlg->sess_count--;
1567
1568         return dlg;
1569 }
1570
1571 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1572 {
1573         pjsip_dialog *dlg;
1574         pj_str_t contact;
1575         pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1576         pj_status_t status;
1577
1578         contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1579         contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1580                         "<sip:%s%.*s%s:%d%s%s>",
1581                         (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1582                         (int)rdata->tp_info.transport->local_name.host.slen,
1583                         rdata->tp_info.transport->local_name.host.ptr,
1584                         (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1585                         rdata->tp_info.transport->local_name.port,
1586                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1587                         (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1588
1589         status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1590         if (status != PJ_SUCCESS) {
1591                 char err[PJ_ERR_MSG_SIZE];
1592
1593                 pj_strerror(status, err, sizeof(err));
1594                 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1595                                 ast_sorcery_object_get_id(endpoint), err);
1596                 return NULL;
1597         }
1598
1599         return dlg;
1600 }
1601
1602 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1603 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1604 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1605
1606 static struct {
1607         const char *method;
1608         const pjsip_method *pmethod;
1609 } methods [] = {
1610         { "INVITE", &pjsip_invite_method },
1611         { "CANCEL", &pjsip_cancel_method },
1612         { "ACK", &pjsip_ack_method },
1613         { "BYE", &pjsip_bye_method },
1614         { "REGISTER", &pjsip_register_method },
1615         { "OPTIONS", &pjsip_options_method },
1616         { "SUBSCRIBE", &pjsip_subscribe_method },
1617         { "NOTIFY", &pjsip_notify_method },
1618         { "PUBLISH", &pjsip_publish_method },
1619         { "INFO", &info_method },
1620         { "MESSAGE", &message_method },
1621 };
1622
1623 static const pjsip_method *get_pjsip_method(const char *method)
1624 {
1625         int i;
1626         for (i = 0; i < ARRAY_LEN(methods); ++i) {
1627                 if (!strcmp(method, methods[i].method)) {
1628                         return methods[i].pmethod;
1629                 }
1630         }
1631         return NULL;
1632 }
1633
1634 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1635 {
1636         if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1637                 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1638                 return -1;
1639         }
1640
1641         return 0;
1642 }
1643
1644 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata);
1645 static pjsip_module supplement_module = {
1646         .name = { "Out of dialog supplement hook", 29 },
1647         .id = -1,
1648         .priority = PJSIP_MOD_PRIORITY_APPLICATION - 1,
1649         .on_rx_request = supplement_on_rx_request,
1650 };
1651
1652 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1653                 const char *uri, struct ast_sip_contact *provided_contact, pjsip_tx_data **tdata)
1654 {
1655         RAII_VAR(struct ast_sip_contact *, contact, ao2_bump(provided_contact), ao2_cleanup);
1656         pj_str_t remote_uri;
1657         pj_str_t from;
1658         pj_pool_t *pool;
1659         pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1660
1661         if (ast_strlen_zero(uri)) {
1662                 if (!endpoint && !contact) {
1663                         ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1664                         return -1;
1665                 }
1666
1667                 if (!contact) {
1668                         contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1669                 }
1670                 if (!contact || ast_strlen_zero(contact->uri)) {
1671                         ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1672                                         ast_sorcery_object_get_id(endpoint));
1673                         return -1;
1674                 }
1675
1676                 pj_cstr(&remote_uri, contact->uri);
1677         } else {
1678                 pj_cstr(&remote_uri, uri);
1679         }
1680
1681         if (endpoint) {
1682                 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1683                         ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1684                                 ast_sorcery_object_get_id(endpoint));
1685                         return -1;
1686                 }
1687         }
1688
1689         pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1690
1691         if (!pool) {
1692                 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1693                 return -1;
1694         }
1695
1696         if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1697                                 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1698                 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1699                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1700                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1701                 return -1;
1702         }
1703
1704         if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1705                         &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1706                 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1707                                 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1708                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1709                 return -1;
1710         }
1711
1712         /* If an outbound proxy is specified on the endpoint apply it to this request */
1713         if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
1714                 ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
1715                 ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
1716                         (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1717                 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1718                 return -1;
1719         }
1720
1721         ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
1722
1723         /* We can release this pool since request creation copied all the necessary
1724          * data into the outbound request's pool
1725          */
1726         pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1727         return 0;
1728 }
1729
1730 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1731                 struct ast_sip_endpoint *endpoint, const char *uri,
1732                 struct ast_sip_contact *contact, pjsip_tx_data **tdata)
1733 {
1734         const pjsip_method *pmethod = get_pjsip_method(method);
1735
1736         if (!pmethod) {
1737                 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1738                 return -1;
1739         }
1740
1741         if (dlg) {
1742                 return create_in_dialog_request(pmethod, dlg, tdata);
1743         } else {
1744                 return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
1745         }
1746 }
1747
1748 AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
1749
1750 int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
1751 {
1752         struct ast_sip_supplement *iter;
1753         int inserted = 0;
1754         SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1755
1756         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1757                 if (iter->priority > supplement->priority) {
1758                         AST_RWLIST_INSERT_BEFORE_CURRENT(supplement, next);
1759                         inserted = 1;
1760                         break;
1761                 }
1762         }
1763         AST_RWLIST_TRAVERSE_SAFE_END;
1764
1765         if (!inserted) {
1766                 AST_RWLIST_INSERT_TAIL(&supplements, supplement, next);
1767         }
1768         ast_module_ref(ast_module_info->self);
1769         return 0;
1770 }
1771
1772 void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
1773 {
1774         struct ast_sip_supplement *iter;
1775         SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1776         AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
1777                 if (supplement == iter) {
1778                         AST_RWLIST_REMOVE_CURRENT(next);
1779                         ast_module_unref(ast_module_info->self);
1780                         break;
1781                 }
1782         }
1783         AST_RWLIST_TRAVERSE_SAFE_END;
1784 }
1785
1786 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1787 {
1788         if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1789                 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1790                 return -1;
1791         }
1792         return 0;
1793 }
1794
1795 static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
1796 {
1797         pj_str_t method;
1798
1799         if (ast_strlen_zero(supplement_method)) {
1800                 return PJ_TRUE;
1801         }
1802
1803         pj_cstr(&method, supplement_method);
1804
1805         return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
1806 }
1807
1808 /*! \brief Structure to hold information about an outbound request */
1809 struct send_request_data {
1810         struct ast_sip_endpoint *endpoint;              /*! The endpoint associated with this request */
1811         void *token;                                    /*! Information to be provided to the callback upon receipt of a response */
1812         void (*callback)(void *token, pjsip_event *e);  /*! The callback to be called upon receipt of a response */
1813 };
1814
1815 static void send_request_data_destroy(void *obj)
1816 {
1817         struct send_request_data *req_data = obj;
1818         ao2_cleanup(req_data->endpoint);
1819 }
1820
1821 static struct send_request_data *send_request_data_alloc(struct ast_sip_endpoint *endpoint,
1822         void *token, void (*callback)(void *token, pjsip_event *e))
1823 {
1824         struct send_request_data *req_data = ao2_alloc(sizeof(*req_data), send_request_data_destroy);
1825
1826         if (!req_data) {
1827                 return NULL;
1828         }
1829
1830         req_data->endpoint = ao2_bump(endpoint);
1831         req_data->token = token;
1832         req_data->callback = callback;
1833
1834         return req_data;
1835 }
1836
1837 static void send_request_cb(void *token, pjsip_event *e)
1838 {
1839         RAII_VAR(struct send_request_data *, req_data, token, ao2_cleanup);
1840         pjsip_transaction *tsx = e->body.tsx_state.tsx;
1841         pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1842         pjsip_tx_data *tdata;
1843         struct ast_sip_supplement *supplement;
1844
1845         AST_RWLIST_RDLOCK(&supplements);
1846         AST_LIST_TRAVERSE(&supplements, supplement, next) {
1847                 if (supplement->incoming_response && does_method_match(&challenge->msg_info.cseq->method.name, supplement->method)) {
1848                         supplement->incoming_response(req_data->endpoint, challenge);
1849                 }
1850         }
1851         AST_RWLIST_UNLOCK(&supplements);
1852
1853         if ((tsx->status_code == 401 || tsx->status_code == 407)
1854                 && req_data->endpoint
1855                 && !ast_sip_create_request_with_auth(&req_data->endpoint->outbound_auths, challenge, tsx, &tdata)
1856                 && pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data->token, req_data->callback)
1857                         == PJ_SUCCESS) {
1858                 return;
1859         }
1860
1861         if (req_data->callback) {
1862                 req_data->callback(req_data->token, e);
1863         }
1864 }
1865
1866 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint,
1867         void *token, void (*callback)(void *token, pjsip_event *e))
1868 {
1869         struct ast_sip_supplement *supplement;
1870         struct send_request_data *req_data = send_request_data_alloc(endpoint, token, callback);
1871         struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
1872
1873         if (!req_data) {
1874                 pjsip_tx_data_dec_ref(tdata);
1875                 return -1;
1876         }
1877
1878         AST_RWLIST_RDLOCK(&supplements);
1879         AST_LIST_TRAVERSE(&supplements, supplement, next) {
1880                 if (supplement->outgoing_request && does_method_match(&tdata->msg->line.req.method.name, supplement->method)) {
1881                         supplement->outgoing_request(endpoint, contact, tdata);
1882                 }
1883         }
1884         AST_RWLIST_UNLOCK(&supplements);
1885
1886         ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
1887         ao2_cleanup(contact);
1888
1889         if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, req_data, send_request_cb) != PJ_SUCCESS) {
1890                 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1891                                 (int) pj_strlen(&tdata->msg->line.req.method.name),
1892                                 pj_strbuf(&tdata->msg->line.req.method.name),
1893                                 endpoint ? ast_sorcery_object_get_id(endpoint) : "<unknown>");
1894                 ao2_cleanup(req_data);
1895                 return -1;
1896         }
1897
1898         return 0;
1899 }
1900
1901 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
1902         struct ast_sip_endpoint *endpoint, void *token,
1903         void (*callback)(void *token, pjsip_event *e))
1904 {
1905         ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1906
1907         if (dlg) {
1908                 return send_in_dialog_request(tdata, dlg);
1909         } else {
1910                 return send_out_of_dialog_request(tdata, endpoint, token, callback);
1911         }
1912 }
1913
1914 int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
1915 {
1916         pjsip_route_hdr *route;
1917         static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1918         pj_str_t tmp;
1919
1920         pj_strdup2_with_null(tdata->pool, &tmp, proxy);
1921         if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1922                 return -1;
1923         }
1924
1925         pj_list_insert_nodes_before(&tdata->msg->hdr, (pjsip_hdr*)route);
1926
1927         return 0;
1928 }
1929
1930 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1931 {
1932         pj_str_t hdr_name;
1933         pj_str_t hdr_value;
1934         pjsip_generic_string_hdr *hdr;
1935
1936         pj_cstr(&hdr_name, name);
1937         pj_cstr(&hdr_value, value);
1938
1939         hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1940
1941         pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1942         return 0;
1943 }
1944
1945 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1946 {
1947         pj_str_t type;
1948         pj_str_t subtype;
1949         pj_str_t body_text;
1950
1951         pj_cstr(&type, body->type);
1952         pj_cstr(&subtype, body->subtype);
1953         pj_cstr(&body_text, body->body_text);
1954
1955         return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1956 }
1957
1958 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1959 {
1960         pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1961         tdata->msg->body = pjsip_body;
1962         return 0;
1963 }
1964
1965 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1966 {
1967         int i;
1968         /* NULL for type and subtype automatically creates "multipart/mixed" */
1969         pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1970
1971         for (i = 0; i < num_bodies; ++i) {
1972                 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1973                 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1974                 pjsip_multipart_add_part(tdata->pool, body, part);
1975         }
1976
1977         tdata->msg->body = body;
1978         return 0;
1979 }
1980
1981 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1982 {
1983         size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1984         struct ast_str *body_buffer = ast_str_alloca(combined_size);
1985
1986         ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1987
1988         tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1989         pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1990         tdata->msg->body->len = combined_size;
1991
1992         return 0;
1993 }
1994
1995 struct ast_taskprocessor *ast_sip_create_serializer(void)
1996 {
1997         struct ast_taskprocessor *serializer;
1998         RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1999         char name[AST_UUID_STR_LEN];
2000
2001         if (!uuid) {
2002                 return NULL;
2003         }
2004
2005         ast_uuid_to_str(uuid, name, sizeof(name));
2006
2007         serializer = ast_threadpool_serializer(name, sip_threadpool);
2008         if (!serializer) {
2009                 return NULL;
2010         }
2011         return serializer;
2012 }
2013
2014 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2015 {
2016         if (serializer) {
2017                 return ast_taskprocessor_push(serializer, sip_task, task_data);
2018         } else {
2019                 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
2020         }
2021 }
2022
2023 struct sync_task_data {
2024         ast_mutex_t lock;
2025         ast_cond_t cond;
2026         int complete;
2027         int fail;
2028         int (*task)(void *);
2029         void *task_data;
2030 };
2031
2032 static int sync_task(void *data)
2033 {
2034         struct sync_task_data *std = data;
2035         std->fail = std->task(std->task_data);
2036
2037         ast_mutex_lock(&std->lock);
2038         std->complete = 1;
2039         ast_cond_signal(&std->cond);
2040         ast_mutex_unlock(&std->lock);
2041         return std->fail;
2042 }
2043
2044 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
2045 {
2046         /* This method is an onion */
2047         struct sync_task_data std;
2048
2049         if (ast_sip_thread_is_servant()) {
2050                 return sip_task(task_data);
2051         }
2052
2053         ast_mutex_init(&std.lock);
2054         ast_cond_init(&std.cond, NULL);
2055         std.fail = std.complete = 0;
2056         std.task = sip_task;
2057         std.task_data = task_data;
2058
2059         if (serializer) {
2060                 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
2061                         return -1;
2062                 }
2063         } else {
2064                 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
2065                         return -1;
2066                 }
2067         }
2068
2069         ast_mutex_lock(&std.lock);
2070         while (!std.complete) {
2071                 ast_cond_wait(&std.cond, &std.lock);
2072         }
2073         ast_mutex_unlock(&std.lock);
2074
2075         ast_mutex_destroy(&std.lock);
2076         ast_cond_destroy(&std.cond);
2077         return std.fail;
2078 }
2079
2080 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
2081 {
2082         size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
2083         memcpy(dest, pj_strbuf(src), chars_to_copy);
2084         dest[chars_to_copy] = '\0';
2085 }
2086
2087 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
2088 {
2089         pjsip_media_type compare;
2090
2091         if (!content_type) {
2092                 return 0;
2093         }
2094
2095         pjsip_media_type_init2(&compare, type, subtype);
2096
2097         return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
2098 }
2099
2100 pj_caching_pool caching_pool;
2101 pj_pool_t *memory_pool;
2102 pj_thread_t *monitor_thread;
2103 static int monitor_continue;
2104
2105 static void *monitor_thread_exec(void *endpt)
2106 {
2107         while (monitor_continue) {
2108                 const pj_time_val delay = {0, 10};
2109                 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
2110         }
2111         return NULL;
2112 }
2113
2114 static void stop_monitor_thread(void)
2115 {
2116         monitor_continue = 0;
2117         pj_thread_join(monitor_thread);
2118 }
2119
2120 AST_THREADSTORAGE(pj_thread_storage);
2121 AST_THREADSTORAGE(servant_id_storage);
2122 #define SIP_SERVANT_ID 0x5E2F1D
2123
2124 static void sip_thread_start(void)
2125 {
2126         pj_thread_desc *desc;
2127         pj_thread_t *thread;
2128         uint32_t *servant_id;
2129
2130         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2131         if (!servant_id) {
2132                 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
2133                 return;
2134         }
2135         *servant_id = SIP_SERVANT_ID;
2136
2137         desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
2138         if (!desc) {
2139                 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
2140                 return;
2141         }
2142         pj_bzero(*desc, sizeof(*desc));
2143
2144         if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
2145                 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
2146         }
2147 }
2148
2149 int ast_sip_thread_is_servant(void)
2150 {
2151         uint32_t *servant_id;
2152
2153         servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
2154         if (!servant_id) {
2155                 return 0;
2156         }
2157
2158         return *servant_id == SIP_SERVANT_ID;
2159 }
2160
2161 void *ast_sip_dict_get(void *ht, const char *key)
2162 {
2163         unsigned int hval = 0;
2164
2165         if (!ht) {
2166                 return NULL;
2167         }
2168
2169         return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
2170 }
2171
2172 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
2173                        const char *key, void *val)
2174 {
2175         if (!ht) {
2176                 ht = pj_hash_create(pool, 11);
2177         }
2178
2179         pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
2180
2181         return ht;
2182 }
2183
2184 static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata)
2185 {
2186         struct ast_sip_supplement *supplement;
2187
2188         if (pjsip_rdata_get_dlg(rdata)) {
2189                 return PJ_FALSE;
2190         }
2191
2192         AST_RWLIST_RDLOCK(&supplements);
2193         AST_LIST_TRAVERSE(&supplements, supplement, next) {
2194                 if (supplement->incoming_request && does_method_match(&rdata->msg_info.msg->line.req.method.name, supplement->method)) {
2195                         supplement->incoming_request(ast_pjsip_rdata_get_endpoint(rdata), rdata);
2196                 }
2197         }
2198         AST_RWLIST_UNLOCK(&supplements);
2199
2200         return PJ_FALSE;
2201 }
2202
2203 int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
2204 {
2205         struct ast_sip_supplement *supplement;
2206         pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
2207         struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
2208
2209         AST_RWLIST_RDLOCK(&supplements);
2210         AST_LIST_TRAVERSE(&supplements, supplement, next) {
2211                 if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
2212                         supplement->outgoing_response(sip_endpoint, contact, tdata);
2213                 }
2214         }
2215         AST_RWLIST_UNLOCK(&supplements);
2216
2217         ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
2218         ao2_cleanup(contact);
2219
2220         return pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
2221 }
2222
2223 int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
2224         struct ast_sip_contact *contact, pjsip_tx_data **tdata)
2225 {
2226         int res = pjsip_endpt_create_response(ast_sip_get_pjsip_endpoint(), rdata, st_code, NULL, tdata);
2227
2228         if (!res) {
2229                 ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
2230         }
2231
2232         return res;
2233 }
2234
2235 static void remove_request_headers(pjsip_endpoint *endpt)
2236 {
2237         const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
2238         pjsip_hdr *iter = request_headers->next;
2239
2240         while (iter != request_headers) {
2241                 pjsip_hdr *to_erase = iter;
2242                 iter = iter->next;
2243                 pj_list_erase(to_erase);
2244         }
2245 }
2246
2247 static int load_module(void)
2248 {
2249         /* The third parameter is just copied from
2250          * example code from PJLIB. This can be adjusted
2251          * if necessary.
2252          */
2253         pj_status_t status;
2254         struct ast_threadpool_options options;
2255
2256         if (pj_init() != PJ_SUCCESS) {
2257                 return AST_MODULE_LOAD_DECLINE;
2258         }
2259
2260         if (pjlib_util_init() != PJ_SUCCESS) {
2261                 pj_shutdown();
2262                 return AST_MODULE_LOAD_DECLINE;
2263         }
2264
2265         pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
2266         if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
2267                 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
2268                 pj_caching_pool_destroy(&caching_pool);
2269                 return AST_MODULE_LOAD_DECLINE;
2270         }
2271
2272         /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
2273          * we need to stop PJSIP from doing it automatically
2274          */
2275         remove_request_headers(ast_pjsip_endpoint);
2276
2277         memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2278         if (!memory_pool) {
2279                 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2280                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2281                 ast_pjsip_endpoint = NULL;
2282                 pj_caching_pool_destroy(&caching_pool);
2283                 return AST_MODULE_LOAD_DECLINE;
2284         }
2285
2286         if (ast_sip_initialize_system()) {
2287                 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2288                 pj_pool_release(memory_pool);
2289                 memory_pool = NULL;
2290                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2291                 ast_pjsip_endpoint = NULL;
2292                 pj_caching_pool_destroy(&caching_pool);
2293                 return AST_MODULE_LOAD_DECLINE;
2294         }
2295
2296         sip_get_threadpool_options(&options);
2297         options.thread_start = sip_thread_start;
2298         sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2299         if (!sip_threadpool) {
2300                 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2301                 ast_sip_destroy_system();
2302                 pj_pool_release(memory_pool);
2303                 memory_pool = NULL;
2304                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2305                 ast_pjsip_endpoint = NULL;
2306                 pj_caching_pool_destroy(&caching_pool);
2307                 return AST_MODULE_LOAD_DECLINE;
2308         }
2309
2310         ast_sip_initialize_dns();
2311
2312         pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2313         pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2314
2315         monitor_continue = 1;
2316         status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2317                         NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2318         if (status != PJ_SUCCESS) {
2319                 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2320                 ast_sip_destroy_system();
2321                 pj_pool_release(memory_pool);
2322                 memory_pool = NULL;
2323                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2324                 ast_pjsip_endpoint = NULL;
2325                 pj_caching_pool_destroy(&caching_pool);
2326                 return AST_MODULE_LOAD_DECLINE;
2327         }
2328
2329         ast_sip_initialize_global_headers();
2330
2331         if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2332                 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2333                 ast_sip_destroy_global_headers();
2334                 stop_monitor_thread();
2335                 ast_sip_destroy_system();
2336                 pj_pool_release(memory_pool);
2337                 memory_pool = NULL;
2338                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2339                 ast_pjsip_endpoint = NULL;
2340                 pj_caching_pool_destroy(&caching_pool);
2341                 return AST_MODULE_LOAD_DECLINE;
2342         }
2343
2344         if (ast_sip_initialize_distributor()) {
2345                 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2346                 ast_res_pjsip_destroy_configuration();
2347                 ast_sip_destroy_global_headers();
2348                 stop_monitor_thread();
2349                 ast_sip_destroy_system();
2350                 pj_pool_release(memory_pool);
2351                 memory_pool = NULL;
2352                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2353                 ast_pjsip_endpoint = NULL;
2354                 pj_caching_pool_destroy(&caching_pool);
2355                 return AST_MODULE_LOAD_DECLINE;
2356         }
2357
2358         if (ast_sip_register_service(&supplement_module)) {
2359                 ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
2360                 ast_sip_destroy_distributor();
2361                 ast_res_pjsip_destroy_configuration();
2362                 ast_sip_destroy_global_headers();
2363                 stop_monitor_thread();
2364                 ast_sip_destroy_system();
2365                 pj_pool_release(memory_pool);
2366                 memory_pool = NULL;
2367                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2368                 ast_pjsip_endpoint = NULL;
2369                 pj_caching_pool_destroy(&caching_pool);
2370                 return AST_MODULE_LOAD_DECLINE;
2371         }
2372
2373         if (ast_sip_initialize_outbound_authentication()) {
2374                 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2375                 ast_sip_unregister_service(&supplement_module);
2376                 ast_sip_destroy_distributor();
2377                 ast_res_pjsip_destroy_configuration();
2378                 ast_sip_destroy_global_headers();
2379                 stop_monitor_thread();
2380                 ast_sip_destroy_system();
2381                 pj_pool_release(memory_pool);
2382                 memory_pool = NULL;
2383                 pjsip_endpt_destroy(ast_pjsip_endpoint);
2384                 ast_pjsip_endpoint = NULL;
2385                 pj_caching_pool_destroy(&caching_pool);
2386                 return AST_MODULE_LOAD_DECLINE;
2387         }
2388
2389         ast_res_pjsip_init_options_handling(0);
2390
2391         ast_module_ref(ast_module_info->self);
2392
2393         return AST_MODULE_LOAD_SUCCESS;
2394 }
2395
2396 static int reload_module(void)
2397 {
2398         if (ast_res_pjsip_reload_configuration()) {
2399                 return AST_MODULE_LOAD_DECLINE;
2400         }
2401         ast_res_pjsip_init_options_handling(1);
2402         ast_sip_initialize_dns();
2403         return 0;
2404 }
2405
2406 static int unload_module(void)
2407 {
2408         /* This will never get called as this module can't be unloaded */
2409         return 0;
2410 }
2411
2412 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2413                 .load = load_module,
2414                 .unload = unload_module,
2415                 .reload = reload_module,
2416                 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
2417 );