2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 2013, Digium, Inc.
6 * Mark Michelson <mmichelson@digium.com>
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.
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.
22 /* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
23 #include <pjsip_simple.h>
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"
40 <depend>pjproject</depend>
41 <depend>res_sorcery_config</depend>
42 <support_level>core</support_level>
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>
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.
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.
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.
72 <configOption name="100rel" default="yes">
73 <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
77 <enum name="required" />
82 <configOption name="aggregate_mwi" default="yes">
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>
88 <configOption name="allow">
89 <synopsis>Media Codec(s) to allow</synopsis>
91 <configOption name="aors">
92 <synopsis>AoR(s) to be used with the endpoint</synopsis>
94 List of comma separated AoRs that the endpoint should be associated with.
97 <configOption name="auth">
98 <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
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.
103 Endpoints without an <literal>authentication</literal> object
104 configured will allow connections without vertification.
105 </para></description>
107 <configOption name="callerid">
108 <synopsis>CallerID information for the endpoint</synopsis>
110 Must be in the format <literal>Name <Number></literal>,
111 or only <literal><Number></literal>.
112 </para></description>
114 <configOption name="callerid_privacy">
115 <synopsis>Default privacy level</synopsis>
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" />
130 <configOption name="callerid_tag">
131 <synopsis>Internal id_tag for the endpoint</synopsis>
133 <configOption name="context">
134 <synopsis>Dialplan context for inbound sessions</synopsis>
136 <configOption name="direct_media_glare_mitigation" default="none">
137 <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
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
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
152 <enum name="outgoing" />
153 <enum name="incoming" />
157 <configOption name="direct_media_method" default="invite">
158 <synopsis>Direct Media method type</synopsis>
160 <para>Method for setting up Direct Media between endpoints.</para>
162 <enum name="invite" />
163 <enum name="reinvite">
164 <para>Alias for the <literal>invite</literal> value.</para>
166 <enum name="update" />
170 <configOption name="connected_line_method" default="invite">
171 <synopsis>Connected line method type</synopsis>
173 <para>Method used when updating connected line information.</para>
175 <enum name="invite" />
176 <enum name="reinvite">
177 <para>Alias for the <literal>invite</literal> value.</para>
179 <enum name="update" />
183 <configOption name="direct_media" default="yes">
184 <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
186 <configOption name="disable_direct_media_on_nat" default="no">
187 <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
189 <configOption name="disallow">
190 <synopsis>Media Codec(s) to disallow</synopsis>
192 <configOption name="dtmf_mode" default="rfc4733">
193 <synopsis>DTMF mode</synopsis>
195 <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
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>
203 <para>DTMF is sent as part of audio stream.</para>
206 <para>DTMF is sent as SIP INFO packets.</para>
211 <configOption name="media_address">
212 <synopsis>IP address used in SDP for media handling</synopsis>
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.
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.
223 <configOption name="force_rport" default="yes">
224 <synopsis>Force use of return port</synopsis>
226 <configOption name="ice_support" default="no">
227 <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
229 <configOption name="identify_by" default="username,location">
230 <synopsis>Way(s) for Endpoint to be identified</synopsis>
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.
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>
243 <enum name="username" />
247 <configOption name="redirect_method">
248 <synopsis>How redirects received from an endpoint are handled</synopsis>
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
263 <enum name="uri_core" />
264 <enum name="uri_pjsip" />
268 <configOption name="mailboxes">
269 <synopsis>Mailbox(es) to be associated with</synopsis>
271 <configOption name="moh_suggest" default="default">
272 <synopsis>Default Music On Hold class</synopsis>
274 <configOption name="outbound_auth">
275 <synopsis>Authentication object used for outbound requests</synopsis>
277 <configOption name="outbound_proxy">
278 <synopsis>Proxy through which to send requests, a full SIP URI must be provided</synopsis>
280 <configOption name="rewrite_contact">
281 <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
283 On inbound SIP messages from this endpoint, the Contact header will be changed to have the
284 source IP address and port. This option does not affect outbound messages send to this
286 </para></description>
288 <configOption name="rtp_ipv6" default="no">
289 <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
291 <configOption name="rtp_symmetric" default="no">
292 <synopsis>Enforce that RTP must be symmetric</synopsis>
294 <configOption name="send_diversion" default="yes">
295 <synopsis>Send the Diversion header, conveying the diversion
296 information to the called user agent</synopsis>
298 <configOption name="send_pai" default="no">
299 <synopsis>Send the P-Asserted-Identity header</synopsis>
301 <configOption name="send_rpid" default="no">
302 <synopsis>Send the Remote-Party-ID header</synopsis>
304 <configOption name="timers_min_se" default="90">
305 <synopsis>Minimum session timers expiration period</synopsis>
307 Minimium session timer expiration period. Time in seconds.
308 </para></description>
310 <configOption name="timers" default="yes">
311 <synopsis>Session timers for SIP packets</synopsis>
314 <enum name="forced" />
316 <enum name="required" />
321 <configOption name="timers_sess_expires" default="1800">
322 <synopsis>Maximum session timer expiration period</synopsis>
324 Maximium session timer expiration period. Time in seconds.
325 </para></description>
327 <configOption name="transport">
328 <synopsis>Desired transport configuration</synopsis>
330 This will set the desired transport configuration to send SIP data through.
332 <warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
333 to the first configured transport in <filename>pjsip.conf</filename> which is
334 valid for the URI we are trying to contact.
336 <warning><para>Transport configuration is not affected by reloads. In order to
337 change transports, a full Asterisk restart is required</para></warning>
340 <configOption name="trust_id_inbound" default="no">
341 <synopsis>Accept identification information received from this endpoint</synopsis>
342 <description><para>This option determines whether Asterisk will accept
343 identification from the endpoint from headers such as P-Asserted-Identity
344 or Remote-Party-ID header. This option applies both to calls originating from the
345 endpoint and calls originating from Asterisk. If <literal>no</literal>, the
346 configured Caller-ID from pjsip.conf will always be used as the identity for
347 the endpoint.</para></description>
349 <configOption name="trust_id_outbound" default="no">
350 <synopsis>Send private identification details to the endpoint.</synopsis>
351 <description><para>This option determines whether res_pjsip will send private
352 identification information to the endpoint. If <literal>no</literal>,
353 private Caller-ID information will not be forwarded to the endpoint.
354 "Private" in this case refers to any method of restricting identification.
355 Example: setting <replaceable>callerid_privacy</replaceable> to any
356 <literal>prohib</literal> variation.
357 Example: If <replaceable>trust_id_inbound</replaceable> is set to
358 <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
359 header in a SIP request or response would indicate the identification
360 provided in the request is private.</para></description>
362 <configOption name="type">
363 <synopsis>Must be of type 'endpoint'.</synopsis>
365 <configOption name="use_ptime" default="no">
366 <synopsis>Use Endpoint's requested packetisation interval</synopsis>
368 <configOption name="use_avpf" default="no">
369 <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
372 If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
373 profile for all media offers on outbound calls and media updates and will
374 decline media offers not using the AVPF or SAVPF profile.
376 If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
377 profile for all media offers on outbound calls and media updates and will
378 decline media offers not using the AVP or SAVP profile.
379 </para></description>
381 <configOption name="media_encryption" default="no">
382 <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
383 for this endpoint.</synopsis>
386 <enum name="no"><para>
387 res_pjsip will offer no encryption and allow no encryption to be setup.
389 <enum name="sdes"><para>
390 res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
391 transport should be used in conjunction with this option to prevent
392 exposure of media encryption keys.
394 <enum name="dtls"><para>
395 res_pjsip will offer DTLS-SRTP setup.
400 <configOption name="inband_progress" default="no">
401 <synopsis>Determines whether chan_pjsip will indicate ringing using inband
404 If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
405 when told to indicate ringing and will immediately start sending ringing
408 If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
409 to indicate ringing and will NOT send it as audio.
410 </para></description>
412 <configOption name="call_group">
413 <synopsis>The numeric pickup groups for a channel.</synopsis>
415 Can be set to a comma separated list of numbers or ranges between the values
416 of 0-63 (maximum of 64 groups).
417 </para></description>
419 <configOption name="pickup_group">
420 <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
422 Can be set to a comma separated list of numbers or ranges between the values
423 of 0-63 (maximum of 64 groups).
424 </para></description>
426 <configOption name="named_call_group">
427 <synopsis>The named pickup groups for a channel.</synopsis>
429 Can be set to a comma separated list of case sensitive strings limited by
430 supported line length.
431 </para></description>
433 <configOption name="named_pickup_group">
434 <synopsis>The named pickup groups that a channel can pickup.</synopsis>
436 Can be set to a comma separated list of case sensitive strings limited by
437 supported line length.
438 </para></description>
440 <configOption name="device_state_busy_at" default="0">
441 <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
443 When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
444 PJSIP channel driver will return busy as the device state instead of in use.
445 </para></description>
447 <configOption name="t38_udptl" default="no">
448 <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
450 If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
452 </para></description>
454 <configOption name="t38_udptl_ec" default="none">
455 <synopsis>T.38 UDPTL error correction method</synopsis>
458 <enum name="none"><para>
459 No error correction should be used.
461 <enum name="fec"><para>
462 Forward error correction should be used.
464 <enum name="redundancy"><para>
465 Redundacy error correction should be used.
470 <configOption name="t38_udptl_maxdatagram" default="0">
471 <synopsis>T.38 UDPTL maximum datagram size</synopsis>
473 This option can be set to override the maximum datagram of a remote endpoint for broken
475 </para></description>
477 <configOption name="fax_detect" default="no">
478 <synopsis>Whether CNG tone detection is enabled</synopsis>
480 This option can be set to send the session to the fax extension when a CNG tone is
482 </para></description>
484 <configOption name="t38_udptl_nat" default="no">
485 <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
487 When enabled the UDPTL stack will send UDPTL packets to the source address of
489 </para></description>
491 <configOption name="t38_udptl_ipv6" default="no">
492 <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
494 When enabled the UDPTL stack will use IPv6.
495 </para></description>
497 <configOption name="tone_zone">
498 <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
500 <configOption name="language">
501 <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
503 <configOption name="one_touch_recording" default="no">
504 <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
506 <ref type="configOption">recordonfeature</ref>
507 <ref type="configOption">recordofffeature</ref>
510 <configOption name="record_on_feature" default="automixmon">
511 <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
513 <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
514 feature will be enabled for the channel. The feature designated here can be any built-in
515 or dynamic feature defined in features.conf.</para>
516 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
519 <ref type="configOption">one_touch_recording</ref>
520 <ref type="configOption">recordofffeature</ref>
523 <configOption name="record_off_feature" default="automixmon">
524 <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
526 <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
527 feature will be enabled for the channel. The feature designated here can be any built-in
528 or dynamic feature defined in features.conf.</para>
529 <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
532 <ref type="configOption">one_touch_recording</ref>
533 <ref type="configOption">recordonfeature</ref>
536 <configOption name="rtp_engine" default="asterisk">
537 <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
539 <configOption name="allow_transfer" default="yes">
540 <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
542 <configOption name="sdp_owner" default="-">
543 <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
545 <configOption name="sdp_session" default="Asterisk">
546 <synopsis>String used for the SDP session (s=) line.</synopsis>
548 <configOption name="tos_audio">
549 <synopsis>DSCP TOS bits for audio streams</synopsis>
551 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
552 </para></description>
554 <configOption name="tos_video">
555 <synopsis>DSCP TOS bits for video streams</synopsis>
557 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
558 </para></description>
560 <configOption name="cos_audio">
561 <synopsis>Priority for audio streams</synopsis>
563 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
564 </para></description>
566 <configOption name="cos_video">
567 <synopsis>Priority for video streams</synopsis>
569 See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
570 </para></description>
572 <configOption name="allow_subscribe" default="yes">
573 <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
575 <configOption name="sub_min_expiry" default="60">
576 <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
578 <configOption name="from_user">
579 <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
581 <configOption name="mwi_from_user">
582 <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
584 <configOption name="from_domain">
585 <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
587 <configOption name="dtls_verify">
588 <synopsis>Verify that the provided peer certificate is valid</synopsis>
590 This option only applies if <replaceable>media_encryption</replaceable> is
591 set to <literal>dtls</literal>.
592 </para></description>
594 <configOption name="dtls_rekey">
595 <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
597 This option only applies if <replaceable>media_encryption</replaceable> is
598 set to <literal>dtls</literal>.
600 If this is not set or the value provided is 0 rekeying will be disabled.
601 </para></description>
603 <configOption name="dtls_cert_file">
604 <synopsis>Path to certificate file to present to peer</synopsis>
606 This option only applies if <replaceable>media_encryption</replaceable> is
607 set to <literal>dtls</literal>.
608 </para></description>
610 <configOption name="dtls_private_key">
611 <synopsis>Path to private key for certificate file</synopsis>
613 This option only applies if <replaceable>media_encryption</replaceable> is
614 set to <literal>dtls</literal>.
615 </para></description>
617 <configOption name="dtls_cipher">
618 <synopsis>Cipher to use for DTLS negotiation</synopsis>
620 This option only applies if <replaceable>media_encryption</replaceable> is
621 set to <literal>dtls</literal>.
623 Many options for acceptable ciphers. See link for more:
624 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
625 </para></description>
627 <configOption name="dtls_ca_file">
628 <synopsis>Path to certificate authority certificate</synopsis>
630 This option only applies if <replaceable>media_encryption</replaceable> is
631 set to <literal>dtls</literal>.
632 </para></description>
634 <configOption name="dtls_ca_path">
635 <synopsis>Path to a directory containing certificate authority certificates</synopsis>
637 This option only applies if <replaceable>media_encryption</replaceable> is
638 set to <literal>dtls</literal>.
639 </para></description>
641 <configOption name="dtls_setup">
642 <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
645 This option only applies if <replaceable>media_encryption</replaceable> is
646 set to <literal>dtls</literal>.
649 <enum name="active"><para>
650 res_pjsip will make a connection to the peer.
652 <enum name="passive"><para>
653 res_pjsip will accept connections from the peer.
655 <enum name="actpass"><para>
656 res_pjsip will offer and accept connections from the peer.
661 <configOption name="srtp_tag_32">
662 <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
664 This option only applies if <replaceable>media_encryption</replaceable> is
665 set to <literal>sdes</literal> or <literal>dtls</literal>.
666 </para></description>
669 <configObject name="auth">
670 <synopsis>Authentication type</synopsis>
672 Authentication objects hold the authentication information for use
673 by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
674 This also allows for multiple objects to use a single auth object. See
675 the <literal>auth_type</literal> config option for password style choices.
676 </para></description>
677 <configOption name="auth_type" default="userpass">
678 <synopsis>Authentication type</synopsis>
680 This option specifies which of the password style config options should be read
681 when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
682 then we'll read from the 'password' option. For <literal>md5</literal> we'll read
687 <enum name="userpass"/>
691 <configOption name="nonce_lifetime" default="32">
692 <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
694 <configOption name="md5_cred">
695 <synopsis>MD5 Hash used for authentication.</synopsis>
696 <description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
698 <configOption name="password">
699 <synopsis>PlainText password used for authentication.</synopsis>
700 <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
702 <configOption name="realm" default="asterisk">
703 <synopsis>SIP realm for endpoint</synopsis>
705 <configOption name="type">
706 <synopsis>Must be 'auth'</synopsis>
708 <configOption name="username">
709 <synopsis>Username to use for account</synopsis>
712 <configObject name="domain_alias">
713 <synopsis>Domain Alias</synopsis>
715 Signifies that a domain is an alias. If the domain on a session is
716 not found to match an AoR then this object is used to see if we have
717 an alias for the AoR to which the endpoint is binding. This objects
718 name as defined in configuration should be the domain alias and a
719 config option is provided to specify the domain to be aliased.
720 </para></description>
721 <configOption name="type">
722 <synopsis>Must be of type 'domain_alias'.</synopsis>
724 <configOption name="domain">
725 <synopsis>Domain to be aliased</synopsis>
728 <configObject name="transport">
729 <synopsis>SIP Transport</synopsis>
731 <emphasis>Transports</emphasis>
733 <para>There are different transports and protocol derivatives
734 supported by <literal>res_pjsip</literal>. They are in order of
735 preference: UDP, TCP, and WebSocket (WS).</para>
736 <note><para>Changes to transport configuration in pjsip.conf will only be
737 effected on a complete restart of Asterisk. A module reload
738 will not suffice.</para></note>
740 <configOption name="async_operations" default="1">
741 <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
743 <configOption name="bind">
744 <synopsis>IP Address and optional port to bind to for this transport</synopsis>
746 <configOption name="ca_list_file">
747 <synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
749 <configOption name="cert_file">
750 <synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
752 <configOption name="cipher">
753 <synopsis>Preferred Cryptography Cipher (TLS ONLY)</synopsis>
755 Many options for acceptable ciphers see link for more:
756 http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
757 </para></description>
759 <configOption name="domain">
760 <synopsis>Domain the transport comes from</synopsis>
762 <configOption name="external_media_address">
763 <synopsis>External IP address to use in RTP handling</synopsis>
765 When a request or response is sent out, if the destination of the
766 message is outside the IP network defined in the option <literal>localnet</literal>,
767 and the media address in the SDP is within the localnet network, then the
768 media address in the SDP will be rewritten to the value defined for
769 <literal>external_media_address</literal>.
770 </para></description>
772 <configOption name="external_signaling_address">
773 <synopsis>External address for SIP signalling</synopsis>
775 <configOption name="external_signaling_port" default="0">
776 <synopsis>External port for SIP signalling</synopsis>
778 <configOption name="method">
779 <synopsis>Method of SSL transport (TLS ONLY)</synopsis>
782 <enum name="default" />
783 <enum name="unspecified" />
784 <enum name="tlsv1" />
785 <enum name="sslv2" />
786 <enum name="sslv3" />
787 <enum name="sslv23" />
791 <configOption name="local_net">
792 <synopsis>Network to consider local (used for NAT purposes).</synopsis>
793 <description><para>This must be in CIDR or dotted decimal format with the IP
794 and mask separated with a slash ('/').</para></description>
796 <configOption name="password">
797 <synopsis>Password required for transport</synopsis>
799 <configOption name="priv_key_file">
800 <synopsis>Private key file (TLS ONLY)</synopsis>
802 <configOption name="protocol" default="udp">
803 <synopsis>Protocol to use for SIP traffic</synopsis>
814 <configOption name="require_client_cert" default="false">
815 <synopsis>Require client certificate (TLS ONLY)</synopsis>
817 <configOption name="type">
818 <synopsis>Must be of type 'transport'.</synopsis>
820 <configOption name="verify_client" default="false">
821 <synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
823 <configOption name="verify_server" default="false">
824 <synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
826 <configOption name="tos" default="false">
827 <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
829 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
830 for more information on this parameter.</para>
831 <note><para>This option does not apply to the <replaceable>ws</replaceable>
832 or the <replaceable>wss</replaceable> protocols.</para></note>
835 <configOption name="cos" default="false">
836 <synopsis>Enable COS for the signalling sent over this transport</synopsis>
838 <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
839 for more information on this parameter.</para>
840 <note><para>This option does not apply to the <replaceable>ws</replaceable>
841 or the <replaceable>wss</replaceable> protocols.</para></note>
845 <configObject name="contact">
846 <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
848 Contacts are a way to hide SIP URIs from the dialplan directly.
849 They are also used to make a group of contactable parties when
850 in use with <literal>AoR</literal> lists.
851 </para></description>
852 <configOption name="type">
853 <synopsis>Must be of type 'contact'.</synopsis>
855 <configOption name="uri">
856 <synopsis>SIP URI to contact peer</synopsis>
858 <configOption name="expiration_time">
859 <synopsis>Time to keep alive a contact</synopsis>
861 Time to keep alive a contact. String style specification.
862 </para></description>
864 <configOption name="qualify_frequency" default="0">
865 <synopsis>Interval at which to qualify a contact</synopsis>
867 Interval between attempts to qualify the contact for reachability.
868 If <literal>0</literal> never qualify. Time in seconds.
869 </para></description>
872 <configObject name="aor">
873 <synopsis>The configuration for a location of an endpoint</synopsis>
875 An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
876 AoRs are specified, an endpoint will not be reachable by Asterisk.
877 Beyond that, an AoR has other uses within Asterisk, such as inbound
880 An <literal>AoR</literal> is a way to allow dialing a group
881 of <literal>Contacts</literal> that all use the same
882 <literal>endpoint</literal> for calls.
884 This can be used as another way of grouping a list of contacts to dial
885 rather than specifing them each directly when dialing via the dialplan.
886 This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
888 Registrations: For Asterisk to match an inbound registration to an endpoint,
889 the AoR object name must match the user portion of the SIP URI in the "To:"
890 header of the inbound SIP registration. That will usually be equivalent
891 to the "user name" set in your hard or soft phones configuration.
892 </para></description>
893 <configOption name="contact">
894 <synopsis>Permanent contacts assigned to AoR</synopsis>
896 Contacts specified will be called whenever referenced
897 by <literal>chan_pjsip</literal>.
899 Use a separate "contact=" entry for each contact required. Contacts
900 are specified using a SIP URI.
901 </para></description>
903 <configOption name="default_expiration" default="3600">
904 <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
906 <configOption name="mailboxes">
907 <synopsis>Mailbox(es) to be associated with</synopsis>
908 <description><para>This option applies when an external entity subscribes to an AoR
909 for message waiting indications. The mailboxes specified will be subscribed to.
910 More than one mailbox can be specified with a comma-delimited string.</para></description>
912 <configOption name="maximum_expiration" default="7200">
913 <synopsis>Maximum time to keep an AoR</synopsis>
915 Maximium time to keep a peer with explicit expiration. Time in seconds.
916 </para></description>
918 <configOption name="max_contacts" default="0">
919 <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
921 Maximum number of contacts that can associate with this AoR. This value does
922 not affect the number of contacts that can be added with the "contact" option.
923 It only limits contacts added through external interaction, such as
926 <note><para>This should be set to <literal>1</literal> and
927 <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
928 wish to stick with the older <literal>chan_sip</literal> behaviour.
932 <configOption name="minimum_expiration" default="60">
933 <synopsis>Minimum keep alive time for an AoR</synopsis>
935 Minimum time to keep a peer with an explict expiration. Time in seconds.
936 </para></description>
938 <configOption name="remove_existing" default="no">
939 <synopsis>Determines whether new contacts replace existing ones.</synopsis>
941 On receiving a new registration to the AoR should it remove
942 the existing contact that was registered against it?
944 <note><para>This should be set to <literal>yes</literal> and
945 <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
946 wish to stick with the older <literal>chan_sip</literal> behaviour.
950 <configOption name="type">
951 <synopsis>Must be of type 'aor'.</synopsis>
953 <configOption name="qualify_frequency" default="0">
954 <synopsis>Interval at which to qualify an AoR</synopsis>
956 Interval between attempts to qualify the AoR for reachability.
957 If <literal>0</literal> never qualify. Time in seconds.
958 </para></description>
960 <configOption name="authenticate_qualify" default="no">
961 <synopsis>Authenticates a qualify request if needed</synopsis>
963 If true and a qualify request receives a challenge or authenticate response
964 authentication is attempted before declaring the contact available.
965 </para></description>
968 <configObject name="system">
969 <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
971 The settings in this section are global. In addition to being global, the values will
972 not be re-evaluated when a reload is performed. This is because the values must be set
973 before the SIP stack is initialized. The only way to reset these values is to either
974 restart Asterisk, or unload res_pjsip.so and then load it again.
975 </para></description>
976 <configOption name="timer_t1" default="500">
977 <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
979 Timer T1 is the base for determining how long to wait before retransmitting
980 requests that receive no response when using an unreliable transport (e.g. UDP).
981 For more information on this timer, see RFC 3261, Section 17.1.1.1.
982 </para></description>
984 <configOption name="timer_b" default="32000">
985 <synopsis>Set transaction timer B value (milliseconds).</synopsis>
987 Timer B determines the maximum amount of time to wait after sending an INVITE
988 request before terminating the transaction. It is recommended that this be set
989 to 64 * Timer T1, but it may be set higher if desired. For more information on
990 this timer, see RFC 3261, Section 17.1.1.1.
991 </para></description>
993 <configOption name="compact_headers" default="no">
994 <synopsis>Use the short forms of common SIP header names.</synopsis>
996 <configOption name="threadpool_initial_size" default="0">
997 <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
999 <configOption name="threadpool_auto_increment" default="5">
1000 <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
1002 <configOption name="threadpool_idle_timeout" default="60">
1003 <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
1005 <configOption name="threadpool_max_size" default="0">
1006 <synopsis>Maximum number of threads in the res_pjsip threadpool.
1007 A value of 0 indicates no maximum.</synopsis>
1009 <configOption name="type">
1010 <synopsis>Must be of type 'system'.</synopsis>
1013 <configObject name="global">
1014 <synopsis>Options that apply globally to all SIP communications</synopsis>
1016 The settings in this section are global. Unlike options in the <literal>system</literal>
1017 section, these options can be refreshed by performing a reload.
1018 </para></description>
1019 <configOption name="max_forwards" default="70">
1020 <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
1022 <configOption name="type">
1023 <synopsis>Must be of type 'global'.</synopsis>
1025 <configOption name="user_agent" default="Asterisk <Asterisk Version>">
1026 <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
1031 <manager name="PJSIPQualify" language="en_US">
1033 Qualify a chan_pjsip endpoint.
1036 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1037 <parameter name="Endpoint" required="true">
1038 <para>The endpoint you want to qualify.</para>
1042 <para>Qualify a chan_pjsip endpoint.</para>
1045 <manager name="PJSIPShowEndpoints" language="en_US">
1047 Lists PJSIP endpoints.
1052 Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
1053 is raised that contains relevant attributes and status information. Once all
1054 endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
1058 <manager name="PJSIPShowEndpoint" language="en_US">
1060 Detail listing of an endpoint and its objects.
1063 <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
1064 <parameter name="Endpoint" required="true">
1065 <para>The endpoint to list.</para>
1070 Provides a detailed listing of options for a given endpoint. Events are issued
1071 showing the configuration and status of the endpoint and associated objects. These
1072 events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
1073 <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
1074 <literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
1075 associated (for instance AoRs). Once all detail events have been raised a final
1076 <literal>EndpointDetailComplete</literal> event is issued.
1083 static pjsip_endpoint *ast_pjsip_endpoint;
1085 static struct ast_threadpool *sip_threadpool;
1087 static int register_service(void *data)
1089 pjsip_module **module = data;
1090 if (!ast_pjsip_endpoint) {
1091 ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
1094 if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
1095 ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1098 ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
1099 ast_module_ref(ast_module_info->self);
1103 int ast_sip_register_service(pjsip_module *module)
1105 return ast_sip_push_task_synchronous(NULL, register_service, &module);
1108 static int unregister_service(void *data)
1110 pjsip_module **module = data;
1111 ast_module_unref(ast_module_info->self);
1112 if (!ast_pjsip_endpoint) {
1115 pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
1116 ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
1120 void ast_sip_unregister_service(pjsip_module *module)
1122 ast_sip_push_task_synchronous(NULL, unregister_service, &module);
1125 static struct ast_sip_authenticator *registered_authenticator;
1127 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
1129 if (registered_authenticator) {
1130 ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
1133 registered_authenticator = auth;
1134 ast_debug(1, "Registered SIP authenticator module %p\n", auth);
1135 ast_module_ref(ast_module_info->self);
1139 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
1141 if (registered_authenticator != auth) {
1142 ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
1143 auth, registered_authenticator);
1146 registered_authenticator = NULL;
1147 ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
1148 ast_module_unref(ast_module_info->self);
1151 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1153 if (!registered_authenticator) {
1154 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
1158 return registered_authenticator->requires_authentication(endpoint, rdata);
1161 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1162 pjsip_rx_data *rdata, pjsip_tx_data *tdata)
1164 if (!registered_authenticator) {
1165 ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
1168 return registered_authenticator->check_authentication(endpoint, rdata, tdata);
1171 static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
1173 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1175 if (registered_outbound_authenticator) {
1176 ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
1179 registered_outbound_authenticator = auth;
1180 ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
1181 ast_module_ref(ast_module_info->self);
1185 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
1187 if (registered_outbound_authenticator != auth) {
1188 ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
1189 auth, registered_outbound_authenticator);
1192 registered_outbound_authenticator = NULL;
1193 ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
1194 ast_module_unref(ast_module_info->self);
1197 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1198 pjsip_transaction *tsx, pjsip_tx_data **new_request)
1200 if (!registered_outbound_authenticator) {
1201 ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
1204 return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
1207 struct endpoint_identifier_list {
1208 struct ast_sip_endpoint_identifier *identifier;
1209 AST_RWLIST_ENTRY(endpoint_identifier_list) list;
1212 static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
1214 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1216 struct endpoint_identifier_list *id_list_item;
1217 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1219 id_list_item = ast_calloc(1, sizeof(*id_list_item));
1220 if (!id_list_item) {
1221 ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
1224 id_list_item->identifier = identifier;
1226 AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
1227 ast_debug(1, "Registered endpoint identifier %p\n", identifier);
1229 ast_module_ref(ast_module_info->self);
1233 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
1235 struct endpoint_identifier_list *iter;
1236 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1237 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
1238 if (iter->identifier == identifier) {
1239 AST_RWLIST_REMOVE_CURRENT(list);
1241 ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
1242 ast_module_unref(ast_module_info->self);
1246 AST_RWLIST_TRAVERSE_SAFE_END;
1249 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
1251 struct endpoint_identifier_list *iter;
1252 struct ast_sip_endpoint *endpoint = NULL;
1253 SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1254 AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
1255 ast_assert(iter->identifier->identify_endpoint != NULL);
1256 endpoint = iter->identifier->identify_endpoint(rdata);
1264 AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
1266 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1268 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1269 AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
1270 ast_module_ref(ast_module_info->self);
1274 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
1276 struct ast_sip_endpoint_formatter *i;
1277 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
1278 AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
1280 AST_RWLIST_REMOVE_CURRENT(next);
1281 ast_module_unref(ast_module_info->self);
1285 AST_RWLIST_TRAVERSE_SAFE_END;
1288 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1289 struct ast_sip_ami *ami, int *count)
1292 struct ast_sip_endpoint_formatter *i;
1293 SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
1295 AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
1296 if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
1307 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
1309 return ast_pjsip_endpoint;
1312 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)
1314 pj_str_t tmp, local_addr;
1316 pjsip_sip_uri *sip_uri;
1317 pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
1319 char uuid_str[AST_UUID_STR_LEN];
1321 if (ast_strlen_zero(user)) {
1322 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1326 user = ast_uuid_to_str(uuid, uuid_str, sizeof(uuid_str));
1329 /* Parse the provided target URI so we can determine what transport it will end up using */
1330 pj_strdup_with_null(pool, &tmp, target);
1332 if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
1333 (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
1337 sip_uri = pjsip_uri_get_uri(uri);
1339 /* Determine the transport type to use */
1340 if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
1341 type = PJSIP_TRANSPORT_TLS;
1342 } else if (!sip_uri->transport_param.slen) {
1343 type = PJSIP_TRANSPORT_UDP;
1345 type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
1348 if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
1352 /* If the host is IPv6 turn the transport into an IPv6 version */
1353 if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1354 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1357 if (!ast_strlen_zero(domain)) {
1358 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1359 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1361 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1364 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1365 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1369 /* Get the local bound address for the transport that will be used when communicating with the provided URI */
1370 if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
1371 &local_addr, &local_port) != PJ_SUCCESS) {
1373 /* If no local address can be retrieved using the transport manager use the host one */
1374 pj_strdup(pool, &local_addr, pj_gethostname());
1375 local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
1378 /* If IPv6 was specified in the transport, set the proper type */
1379 if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
1380 type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
1383 from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
1384 from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
1385 "<%s:%s@%s%.*s%s:%d%s%s>",
1386 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1388 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1389 (int)local_addr.slen,
1391 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1393 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1394 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1399 static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
1401 RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
1402 const char *transport_name = endpoint->transport;
1404 if (ast_strlen_zero(transport_name)) {
1408 transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
1410 if (!transport || !transport->state) {
1411 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
1412 transport_name, ast_sorcery_object_get_id(endpoint));
1416 if (transport->state->transport) {
1417 selector->type = PJSIP_TPSELECTOR_TRANSPORT;
1418 selector->u.transport = transport->state->transport;
1419 } else if (transport->state->factory) {
1420 selector->type = PJSIP_TPSELECTOR_LISTENER;
1421 selector->u.listener = transport->state->factory;
1429 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
1431 char enclosed_uri[PJSIP_MAX_URL_SIZE];
1432 pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
1433 pjsip_dialog *dlg = NULL;
1434 const char *outbound_proxy = endpoint->outbound_proxy;
1435 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1436 static const pj_str_t HCONTACT = { "Contact", 7 };
1438 snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
1439 pj_cstr(&remote_uri, enclosed_uri);
1441 pj_cstr(&target_uri, uri);
1443 if (pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg) != PJ_SUCCESS) {
1447 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1448 pjsip_dlg_terminate(dlg);
1452 if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
1453 pjsip_dlg_terminate(dlg);
1457 /* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
1458 pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
1459 dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
1460 dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
1462 /* If a request user has been specified and we are permitted to change it, do so */
1463 if (!ast_strlen_zero(request_user)) {
1464 pjsip_sip_uri *sip_uri;
1466 if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
1467 sip_uri = pjsip_uri_get_uri(dlg->target);
1468 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1470 if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
1471 sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
1472 pj_strdup2(dlg->pool, &sip_uri->user, request_user);
1476 /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
1479 pjsip_dlg_set_transport(dlg, &selector);
1481 if (!ast_strlen_zero(outbound_proxy)) {
1482 pjsip_route_hdr route_set, *route;
1483 static const pj_str_t ROUTE_HNAME = { "Route", 5 };
1486 pj_list_init(&route_set);
1488 pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
1489 if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
1491 pjsip_dlg_terminate(dlg);
1494 pj_list_push_back(&route_set, route);
1496 pjsip_dlg_set_route_set(dlg, &route_set);
1504 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
1508 pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
1511 contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
1512 contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
1513 "<%s:%s%.*s%s:%d%s%s>",
1514 (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
1515 (type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
1516 (int)rdata->tp_info.transport->local_name.host.slen,
1517 rdata->tp_info.transport->local_name.host.ptr,
1518 (type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
1519 rdata->tp_info.transport->local_name.port,
1520 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
1521 (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
1523 status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
1524 if (status != PJ_SUCCESS) {
1525 char err[PJ_ERR_MSG_SIZE];
1527 pj_strerror(status, err, sizeof(err));
1528 ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
1529 ast_sorcery_object_get_id(endpoint), err);
1536 /* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
1537 static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
1538 static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
1542 const pjsip_method *pmethod;
1544 { "INVITE", &pjsip_invite_method },
1545 { "CANCEL", &pjsip_cancel_method },
1546 { "ACK", &pjsip_ack_method },
1547 { "BYE", &pjsip_bye_method },
1548 { "REGISTER", &pjsip_register_method },
1549 { "OPTIONS", &pjsip_options_method },
1550 { "SUBSCRIBE", &pjsip_subscribe_method },
1551 { "NOTIFY", &pjsip_notify_method },
1552 { "PUBLISH", &pjsip_publish_method },
1553 { "INFO", &info_method },
1554 { "MESSAGE", &message_method },
1557 static const pjsip_method *get_pjsip_method(const char *method)
1560 for (i = 0; i < ARRAY_LEN(methods); ++i) {
1561 if (!strcmp(method, methods[i].method)) {
1562 return methods[i].pmethod;
1568 static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
1570 if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
1571 ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
1578 static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
1579 const char *uri, pjsip_tx_data **tdata)
1581 RAII_VAR(struct ast_sip_contact *, contact, NULL, ao2_cleanup);
1582 pj_str_t remote_uri;
1585 pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
1587 if (ast_strlen_zero(uri)) {
1589 ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
1593 contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
1594 if (!contact || ast_strlen_zero(contact->uri)) {
1595 ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
1596 ast_sorcery_object_get_id(endpoint));
1600 pj_cstr(&remote_uri, contact->uri);
1602 pj_cstr(&remote_uri, uri);
1606 if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
1607 ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
1608 ast_sorcery_object_get_id(endpoint));
1613 pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
1616 ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
1620 if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
1621 endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
1622 ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
1623 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1624 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1628 if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
1629 &from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
1630 ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
1631 (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
1632 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1636 /* We can release this pool since request creation copied all the necessary
1637 * data into the outbound request's pool
1639 pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
1643 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1644 struct ast_sip_endpoint *endpoint, const char *uri,
1645 pjsip_tx_data **tdata)
1647 const pjsip_method *pmethod = get_pjsip_method(method);
1650 ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
1655 return create_in_dialog_request(pmethod, dlg, tdata);
1657 return create_out_of_dialog_request(pmethod, endpoint, uri, tdata);
1661 static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
1663 if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
1664 ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
1670 static void send_request_cb(void *token, pjsip_event *e)
1672 RAII_VAR(struct ast_sip_endpoint *, endpoint, token, ao2_cleanup);
1673 pjsip_transaction *tsx = e->body.tsx_state.tsx;
1674 pjsip_rx_data *challenge = e->body.tsx_state.src.rdata;
1675 pjsip_tx_data *tdata;
1677 if (tsx->status_code != 401 && tsx->status_code != 407) {
1681 if (!ast_sip_create_request_with_auth(&endpoint->outbound_auths, challenge, tsx, &tdata)) {
1682 pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, NULL, NULL);
1686 static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint)
1688 ao2_ref(endpoint, +1);
1689 if (pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, -1, endpoint, send_request_cb) != PJ_SUCCESS) {
1690 ast_log(LOG_ERROR, "Error attempting to send outbound %.*s request to endpoint %s\n",
1691 (int) pj_strlen(&tdata->msg->line.req.method.name),
1692 pj_strbuf(&tdata->msg->line.req.method.name),
1693 ast_sorcery_object_get_id(endpoint));
1694 ao2_ref(endpoint, -1);
1701 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint)
1703 ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
1706 return send_in_dialog_request(tdata, dlg);
1708 return send_out_of_dialog_request(tdata, endpoint);
1712 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
1716 pjsip_generic_string_hdr *hdr;
1718 pj_cstr(&hdr_name, name);
1719 pj_cstr(&hdr_value, value);
1721 hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
1723 pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
1727 static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
1733 pj_cstr(&type, body->type);
1734 pj_cstr(&subtype, body->subtype);
1735 pj_cstr(&body_text, body->body_text);
1737 return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
1740 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
1742 pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
1743 tdata->msg->body = pjsip_body;
1747 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
1750 /* NULL for type and subtype automatically creates "multipart/mixed" */
1751 pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
1753 for (i = 0; i < num_bodies; ++i) {
1754 pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
1755 part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
1756 pjsip_multipart_add_part(tdata->pool, body, part);
1759 tdata->msg->body = body;
1763 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
1765 size_t combined_size = strlen(body_text) + tdata->msg->body->len;
1766 struct ast_str *body_buffer = ast_str_alloca(combined_size);
1768 ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
1770 tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
1771 pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
1772 tdata->msg->body->len = combined_size;
1777 struct ast_taskprocessor *ast_sip_create_serializer(void)
1779 struct ast_taskprocessor *serializer;
1780 RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
1781 char name[AST_UUID_STR_LEN];
1787 ast_uuid_to_str(uuid, name, sizeof(name));
1789 serializer = ast_threadpool_serializer(name, sip_threadpool);
1796 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1799 return ast_taskprocessor_push(serializer, sip_task, task_data);
1801 return ast_threadpool_push(sip_threadpool, sip_task, task_data);
1805 struct sync_task_data {
1810 int (*task)(void *);
1814 static int sync_task(void *data)
1816 struct sync_task_data *std = data;
1817 std->fail = std->task(std->task_data);
1819 ast_mutex_lock(&std->lock);
1821 ast_cond_signal(&std->cond);
1822 ast_mutex_unlock(&std->lock);
1826 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
1828 /* This method is an onion */
1829 struct sync_task_data std;
1830 ast_mutex_init(&std.lock);
1831 ast_cond_init(&std.cond, NULL);
1832 std.fail = std.complete = 0;
1833 std.task = sip_task;
1834 std.task_data = task_data;
1837 if (ast_taskprocessor_push(serializer, sync_task, &std)) {
1841 if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
1846 ast_mutex_lock(&std.lock);
1847 while (!std.complete) {
1848 ast_cond_wait(&std.cond, &std.lock);
1850 ast_mutex_unlock(&std.lock);
1852 ast_mutex_destroy(&std.lock);
1853 ast_cond_destroy(&std.cond);
1857 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
1859 size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
1860 memcpy(dest, pj_strbuf(src), chars_to_copy);
1861 dest[chars_to_copy] = '\0';
1864 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
1866 pjsip_media_type compare;
1868 if (!content_type) {
1872 pjsip_media_type_init2(&compare, type, subtype);
1874 return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
1877 pj_caching_pool caching_pool;
1878 pj_pool_t *memory_pool;
1879 pj_thread_t *monitor_thread;
1880 static int monitor_continue;
1882 static void *monitor_thread_exec(void *endpt)
1884 while (monitor_continue) {
1885 const pj_time_val delay = {0, 10};
1886 pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
1891 static void stop_monitor_thread(void)
1893 monitor_continue = 0;
1894 pj_thread_join(monitor_thread);
1897 AST_THREADSTORAGE(pj_thread_storage);
1898 AST_THREADSTORAGE(servant_id_storage);
1899 #define SIP_SERVANT_ID 0x5E2F1D
1901 static void sip_thread_start(void)
1903 pj_thread_desc *desc;
1904 pj_thread_t *thread;
1905 uint32_t *servant_id;
1907 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1909 ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
1912 *servant_id = SIP_SERVANT_ID;
1914 desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
1916 ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
1919 pj_bzero(*desc, sizeof(*desc));
1921 if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
1922 ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
1926 int ast_sip_thread_is_servant(void)
1928 uint32_t *servant_id;
1930 servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
1935 return *servant_id == SIP_SERVANT_ID;
1938 void *ast_sip_dict_get(void *ht, const char *key)
1946 return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
1949 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
1950 const char *key, void *val)
1953 ht = pj_hash_create(pool, 11);
1956 pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
1961 static void remove_request_headers(pjsip_endpoint *endpt)
1963 const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
1964 pjsip_hdr *iter = request_headers->next;
1966 while (iter != request_headers) {
1967 pjsip_hdr *to_erase = iter;
1969 pj_list_erase(to_erase);
1973 static int load_module(void)
1975 /* The third parameter is just copied from
1976 * example code from PJLIB. This can be adjusted
1980 struct ast_threadpool_options options;
1982 if (pj_init() != PJ_SUCCESS) {
1983 return AST_MODULE_LOAD_DECLINE;
1986 if (pjlib_util_init() != PJ_SUCCESS) {
1988 return AST_MODULE_LOAD_DECLINE;
1991 pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
1992 if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
1993 ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
1994 pj_caching_pool_destroy(&caching_pool);
1995 return AST_MODULE_LOAD_DECLINE;
1998 /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
1999 * we need to stop PJSIP from doing it automatically
2001 remove_request_headers(ast_pjsip_endpoint);
2003 memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
2005 ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
2006 pjsip_endpt_destroy(ast_pjsip_endpoint);
2007 ast_pjsip_endpoint = NULL;
2008 pj_caching_pool_destroy(&caching_pool);
2009 return AST_MODULE_LOAD_DECLINE;
2012 if (ast_sip_initialize_system()) {
2013 ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
2014 pj_pool_release(memory_pool);
2016 pjsip_endpt_destroy(ast_pjsip_endpoint);
2017 ast_pjsip_endpoint = NULL;
2018 pj_caching_pool_destroy(&caching_pool);
2019 return AST_MODULE_LOAD_DECLINE;
2022 sip_get_threadpool_options(&options);
2023 options.thread_start = sip_thread_start;
2024 sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
2025 if (!sip_threadpool) {
2026 ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
2027 pj_pool_release(memory_pool);
2029 pjsip_endpt_destroy(ast_pjsip_endpoint);
2030 ast_pjsip_endpoint = NULL;
2031 pj_caching_pool_destroy(&caching_pool);
2032 return AST_MODULE_LOAD_DECLINE;
2035 pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
2036 pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
2038 monitor_continue = 1;
2039 status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
2040 NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
2041 if (status != PJ_SUCCESS) {
2042 ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
2043 pj_pool_release(memory_pool);
2045 pjsip_endpt_destroy(ast_pjsip_endpoint);
2046 ast_pjsip_endpoint = NULL;
2047 pj_caching_pool_destroy(&caching_pool);
2048 return AST_MODULE_LOAD_DECLINE;
2051 ast_sip_initialize_global_headers();
2053 if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
2054 ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
2055 ast_sip_destroy_global_headers();
2056 stop_monitor_thread();
2057 pj_pool_release(memory_pool);
2059 pjsip_endpt_destroy(ast_pjsip_endpoint);
2060 ast_pjsip_endpoint = NULL;
2061 pj_caching_pool_destroy(&caching_pool);
2062 return AST_MODULE_LOAD_DECLINE;
2065 if (ast_sip_initialize_distributor()) {
2066 ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
2067 ast_res_pjsip_destroy_configuration();
2068 ast_sip_destroy_global_headers();
2069 stop_monitor_thread();
2070 pj_pool_release(memory_pool);
2072 pjsip_endpt_destroy(ast_pjsip_endpoint);
2073 ast_pjsip_endpoint = NULL;
2074 pj_caching_pool_destroy(&caching_pool);
2075 return AST_MODULE_LOAD_DECLINE;
2078 if (ast_sip_initialize_outbound_authentication()) {
2079 ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
2080 ast_sip_destroy_distributor();
2081 ast_res_pjsip_destroy_configuration();
2082 ast_sip_destroy_global_headers();
2083 stop_monitor_thread();
2084 pj_pool_release(memory_pool);
2086 pjsip_endpt_destroy(ast_pjsip_endpoint);
2087 ast_pjsip_endpoint = NULL;
2088 pj_caching_pool_destroy(&caching_pool);
2089 return AST_MODULE_LOAD_DECLINE;
2092 ast_res_pjsip_init_options_handling(0);
2094 ast_module_ref(ast_module_info->self);
2096 return AST_MODULE_LOAD_SUCCESS;
2099 static int reload_module(void)
2101 if (ast_res_pjsip_reload_configuration()) {
2102 return AST_MODULE_LOAD_DECLINE;
2104 ast_res_pjsip_init_options_handling(1);
2108 static int unload_module(void)
2110 /* This will never get called as this module can't be unloaded */
2114 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
2115 .load = load_module,
2116 .unload = unload_module,
2117 .reload = reload_module,
2118 .load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
projects / asterisk/asterisk.git / blob