Fix remnants of the pjsip renaming
[asterisk/asterisk.git] / channels / chan_pjsip.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2013, Digium, Inc.
5  *
6  * Joshua Colp <jcolp@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 /*! \file
20  *
21  * \author Joshua Colp <jcolp@digium.com>
22  *
23  * \brief PSJIP SIP Channel Driver
24  *
25  * \ingroup channel_drivers
26  */
27
28 /*** MODULEINFO
29         <depend>pjproject</depend>
30         <depend>res_pjsip</depend>
31         <depend>res_pjsip_session</depend>
32         <support_level>core</support_level>
33  ***/
34
35 #include "asterisk.h"
36
37 #include <pjsip.h>
38 #include <pjsip_ua.h>
39 #include <pjlib.h>
40
41 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
42
43 #include "asterisk/lock.h"
44 #include "asterisk/channel.h"
45 #include "asterisk/module.h"
46 #include "asterisk/pbx.h"
47 #include "asterisk/rtp_engine.h"
48 #include "asterisk/acl.h"
49 #include "asterisk/callerid.h"
50 #include "asterisk/file.h"
51 #include "asterisk/cli.h"
52 #include "asterisk/app.h"
53 #include "asterisk/musiconhold.h"
54 #include "asterisk/causes.h"
55 #include "asterisk/taskprocessor.h"
56 #include "asterisk/dsp.h"
57 #include "asterisk/stasis_endpoints.h"
58 #include "asterisk/stasis_channels.h"
59 #include "asterisk/indications.h"
60
61 #include "asterisk/res_pjsip.h"
62 #include "asterisk/res_pjsip_session.h"
63
64 /*** DOCUMENTATION
65         <function name="PJSIP_DIAL_CONTACTS" language="en_US">
66                 <synopsis>
67                         Return a dial string for dialing all contacts on an AOR.
68                 </synopsis>
69                 <syntax>
70                         <parameter name="endpoint" required="true">
71                                 <para>Name of the endpoint</para>
72                         </parameter>
73                         <parameter name="aor" required="false">
74                                 <para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
75                         </parameter>
76                         <parameter name="request_user" required="false">
77                                 <para>Optional request user to use in the request URI</para>
78                         </parameter>
79                 </syntax>
80                 <description>
81                         <para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
82                 </description>
83         </function>
84         <function name="PJSIP_MEDIA_OFFER" language="en_US">
85                 <synopsis>
86                         Media and codec offerings to be set on an outbound SIP channel prior to dialing.
87                 </synopsis>
88                 <syntax>
89                         <parameter name="media" required="true">
90                                 <para>types of media offered</para>
91                         </parameter>
92                 </syntax>
93                 <description>
94                         <para>Returns the codecs offered based upon the media choice</para>
95                 </description>
96         </function>
97  ***/
98
99 static const char desc[] = "PJSIP Channel";
100 static const char channel_type[] = "PJSIP";
101
102 static unsigned int chan_idx;
103
104 /*!
105  * \brief Positions of various media
106  */
107 enum sip_session_media_position {
108         /*! \brief First is audio */
109         SIP_MEDIA_AUDIO = 0,
110         /*! \brief Second is video */
111         SIP_MEDIA_VIDEO,
112         /*! \brief Last is the size for media details */
113         SIP_MEDIA_SIZE,
114 };
115
116 struct chan_pjsip_pvt {
117         struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
118 };
119
120 static void chan_pjsip_pvt_dtor(void *obj)
121 {
122         struct chan_pjsip_pvt *pvt = obj;
123         int i;
124
125         for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
126                 ao2_cleanup(pvt->media[i]);
127                 pvt->media[i] = NULL;
128         }
129 }
130
131 /* \brief Asterisk core interaction functions */
132 static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause);
133 static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text);
134 static int chan_pjsip_digit_begin(struct ast_channel *ast, char digit);
135 static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
136 static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout);
137 static int chan_pjsip_hangup(struct ast_channel *ast);
138 static int chan_pjsip_answer(struct ast_channel *ast);
139 static struct ast_frame *chan_pjsip_read(struct ast_channel *ast);
140 static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *f);
141 static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
142 static int chan_pjsip_transfer(struct ast_channel *ast, const char *target);
143 static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
144 static int chan_pjsip_devicestate(const char *data);
145 static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
146
147 /*! \brief PBX interface structure for channel registration */
148 static struct ast_channel_tech chan_pjsip_tech = {
149         .type = channel_type,
150         .description = "PJSIP Channel Driver",
151         .requester = chan_pjsip_request,
152         .send_text = chan_pjsip_sendtext,
153         .send_digit_begin = chan_pjsip_digit_begin,
154         .send_digit_end = chan_pjsip_digit_end,
155         .call = chan_pjsip_call,
156         .hangup = chan_pjsip_hangup,
157         .answer = chan_pjsip_answer,
158         .read = chan_pjsip_read,
159         .write = chan_pjsip_write,
160         .write_video = chan_pjsip_write,
161         .exception = chan_pjsip_read,
162         .indicate = chan_pjsip_indicate,
163         .transfer = chan_pjsip_transfer,
164         .fixup = chan_pjsip_fixup,
165         .devicestate = chan_pjsip_devicestate,
166         .queryoption = chan_pjsip_queryoption,
167         .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
168 };
169
170 /*! \brief SIP session interaction functions */
171 static void chan_pjsip_session_begin(struct ast_sip_session *session);
172 static void chan_pjsip_session_end(struct ast_sip_session *session);
173 static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
174 static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
175
176 /*! \brief SIP session supplement structure */
177 static struct ast_sip_session_supplement chan_pjsip_supplement = {
178         .method = "INVITE",
179         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
180         .session_begin = chan_pjsip_session_begin,
181         .session_end = chan_pjsip_session_end,
182         .incoming_request = chan_pjsip_incoming_request,
183         .incoming_response = chan_pjsip_incoming_response,
184 };
185
186 static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
187
188 static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
189         .method = "ACK",
190         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
191         .incoming_request = chan_pjsip_incoming_ack,
192 };
193
194 /*! \brief Dialplan function for constructing a dial string for calling all contacts */
195 static int chan_pjsip_dial_contacts(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
196 {
197         RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
198         RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
199         const char *aor_name;
200         char *rest;
201
202         AST_DECLARE_APP_ARGS(args,
203                 AST_APP_ARG(endpoint_name);
204                 AST_APP_ARG(aor_name);
205                 AST_APP_ARG(request_user);
206         );
207
208         AST_STANDARD_APP_ARGS(args, data);
209
210         if (ast_strlen_zero(args.endpoint_name)) {
211                 ast_log(LOG_WARNING, "An endpoint name must be specified when using the '%s' dialplan function\n", cmd);
212                 return -1;
213         } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", args.endpoint_name))) {
214                 ast_log(LOG_WARNING, "Specified endpoint '%s' was not found\n", args.endpoint_name);
215                 return -1;
216         }
217
218         aor_name = S_OR(args.aor_name, endpoint->aors);
219
220         if (ast_strlen_zero(aor_name)) {
221                 ast_log(LOG_WARNING, "No AOR has been provided and no AORs are configured on endpoint '%s'\n", args.endpoint_name);
222                 return -1;
223         } else if (!(dial = ast_str_create(len))) {
224                 ast_log(LOG_WARNING, "Could not get enough buffer space for dialing contacts\n");
225                 return -1;
226         } else if (!(rest = ast_strdupa(aor_name))) {
227                 ast_log(LOG_WARNING, "Could not duplicate provided AORs\n");
228                 return -1;
229         }
230
231         while ((aor_name = strsep(&rest, ","))) {
232                 RAII_VAR(struct ast_sip_aor *, aor, ast_sip_location_retrieve_aor(aor_name), ao2_cleanup);
233                 RAII_VAR(struct ao2_container *, contacts, NULL, ao2_cleanup);
234                 struct ao2_iterator it_contacts;
235                 struct ast_sip_contact *contact;
236
237                 if (!aor) {
238                         /* If the AOR provided is not found skip it, there may be more */
239                         continue;
240                 } else if (!(contacts = ast_sip_location_retrieve_aor_contacts(aor))) {
241                         /* No contacts are available, skip it as well */
242                         continue;
243                 } else if (!ao2_container_count(contacts)) {
244                         /* We were given a container but no contacts are in it... */
245                         continue;
246                 }
247
248                 it_contacts = ao2_iterator_init(contacts, 0);
249                 for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
250                         ast_str_append(&dial, -1, "PJSIP/");
251
252                         if (!ast_strlen_zero(args.request_user)) {
253                                 ast_str_append(&dial, -1, "%s@", args.request_user);
254                         }
255                         ast_str_append(&dial, -1, "%s/%s&", args.endpoint_name, contact->uri);
256                 }
257                 ao2_iterator_destroy(&it_contacts);
258         }
259
260         /* Trim the '&' at the end off */
261         ast_str_truncate(dial, ast_str_strlen(dial) - 1);
262
263         ast_copy_string(buf, ast_str_buffer(dial), len);
264
265         return 0;
266 }
267
268 static struct ast_custom_function chan_pjsip_dial_contacts_function = {
269         .name = "PJSIP_DIAL_CONTACTS",
270         .read = chan_pjsip_dial_contacts,
271 };
272
273 static int media_offer_read_av(struct ast_sip_session *session, char *buf,
274                                size_t len, enum ast_format_type media_type)
275 {
276         int i, size = 0;
277         struct ast_format fmt;
278         const char *name;
279
280         for (i = 0; ast_codec_pref_index(&session->override_prefs, i, &fmt); ++i) {
281                 if (AST_FORMAT_GET_TYPE(fmt.id) != media_type) {
282                         continue;
283                 }
284
285                 name = ast_getformatname(&fmt);
286
287                 if (ast_strlen_zero(name)) {
288                         ast_log(LOG_WARNING, "PJSIP_MEDIA_OFFER unrecognized format %s\n", name);
289                         continue;
290                 }
291
292                 /* add one since we'll include a comma */
293                 size = strlen(name) + 1;
294                 len -= size;
295                 if ((len) < 0) {
296                         break;
297                 }
298
299                 /* no reason to use strncat here since we have already ensured buf has
300                    enough space, so strcat can be safely used */
301                 strcat(buf, name);
302                 strcat(buf, ",");
303         }
304
305         if (size) {
306                 /* remove the extra comma */
307                 buf[strlen(buf) - 1] = '\0';
308         }
309         return 0;
310 }
311
312 struct media_offer_data {
313         struct ast_sip_session *session;
314         enum ast_format_type media_type;
315         const char *value;
316 };
317
318 static int media_offer_write_av(void *obj)
319 {
320         struct media_offer_data *data = obj;
321         int i;
322         struct ast_format fmt;
323         /* remove all of the given media type first */
324         for (i = 0; ast_codec_pref_index(&data->session->override_prefs, i, &fmt); ++i) {
325                 if (AST_FORMAT_GET_TYPE(fmt.id) == data->media_type) {
326                         ast_codec_pref_remove(&data->session->override_prefs, &fmt);
327                 }
328         }
329         ast_format_cap_remove_bytype(data->session->req_caps, data->media_type);
330         ast_parse_allow_disallow(&data->session->override_prefs, data->session->req_caps, data->value, 1);
331
332         return 0;
333 }
334
335 static int media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
336 {
337         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
338
339         if (!strcmp(data, "audio")) {
340                 return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_AUDIO);
341         } else if (!strcmp(data, "video")) {
342                 return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_VIDEO);
343         }
344
345         return 0;
346 }
347
348 static int media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
349 {
350         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
351
352         struct media_offer_data mdata = {
353                 .session = channel->session,
354                 .value = value
355         };
356
357         if (!strcmp(data, "audio")) {
358                 mdata.media_type = AST_FORMAT_TYPE_AUDIO;
359         } else if (!strcmp(data, "video")) {
360                 mdata.media_type = AST_FORMAT_TYPE_VIDEO;
361         }
362
363         return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
364 }
365
366 static struct ast_custom_function media_offer_function = {
367         .name = "PJSIP_MEDIA_OFFER",
368         .read = media_offer_read,
369         .write = media_offer_write
370 };
371
372 /*! \brief Function called by RTP engine to get local audio RTP peer */
373 static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
374 {
375         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
376         struct chan_pjsip_pvt *pvt = channel->pvt;
377         struct ast_sip_endpoint *endpoint;
378
379         if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
380                 return AST_RTP_GLUE_RESULT_FORBID;
381         }
382
383         endpoint = channel->session->endpoint;
384
385         *instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
386         ao2_ref(*instance, +1);
387
388         ast_assert(endpoint != NULL);
389         if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
390                 return AST_RTP_GLUE_RESULT_FORBID;
391         }
392
393         if (endpoint->media.direct_media.enabled) {
394                 return AST_RTP_GLUE_RESULT_REMOTE;
395         }
396
397         return AST_RTP_GLUE_RESULT_LOCAL;
398 }
399
400 /*! \brief Function called by RTP engine to get local video RTP peer */
401 static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
402 {
403         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
404         struct chan_pjsip_pvt *pvt = channel->pvt;
405         struct ast_sip_endpoint *endpoint;
406
407         if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
408                 return AST_RTP_GLUE_RESULT_FORBID;
409         }
410
411         endpoint = channel->session->endpoint;
412
413         *instance = pvt->media[SIP_MEDIA_VIDEO]->rtp;
414         ao2_ref(*instance, +1);
415
416         ast_assert(endpoint != NULL);
417         if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
418                 return AST_RTP_GLUE_RESULT_FORBID;
419         }
420
421         return AST_RTP_GLUE_RESULT_LOCAL;
422 }
423
424 /*! \brief Function called by RTP engine to get peer capabilities */
425 static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
426 {
427         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
428
429         ast_format_cap_copy(result, channel->session->endpoint->media.codecs);
430 }
431
432 static int send_direct_media_request(void *data)
433 {
434         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
435
436         return ast_sip_session_refresh(session, NULL, NULL, NULL,
437                         session->endpoint->media.direct_media.method, 1);
438 }
439
440 static struct ast_datastore_info direct_media_mitigation_info = { };
441
442 static int direct_media_mitigate_glare(struct ast_sip_session *session)
443 {
444         RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
445
446         if (session->endpoint->media.direct_media.glare_mitigation ==
447                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
448                 return 0;
449         }
450
451         datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation");
452         if (!datastore) {
453                 return 0;
454         }
455
456         /* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */
457         ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation");
458
459         if ((session->endpoint->media.direct_media.glare_mitigation ==
460                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING &&
461                         session->inv_session->role == PJSIP_ROLE_UAC) ||
462                         (session->endpoint->media.direct_media.glare_mitigation ==
463                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING &&
464                         session->inv_session->role == PJSIP_ROLE_UAS)) {
465                 return 1;
466         }
467
468         return 0;
469 }
470
471 static int check_for_rtp_changes(struct ast_channel *chan, struct ast_rtp_instance *rtp,
472                 struct ast_sip_session_media *media, int rtcp_fd)
473 {
474         int changed = 0;
475
476         if (rtp) {
477                 changed = ast_rtp_instance_get_and_cmp_remote_address(rtp, &media->direct_media_addr);
478                 if (media->rtp) {
479                         ast_channel_set_fd(chan, rtcp_fd, -1);
480                         ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 0);
481                 }
482         } else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
483                 ast_sockaddr_setnull(&media->direct_media_addr);
484                 changed = 1;
485                 if (media->rtp) {
486                         ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1);
487                         ast_channel_set_fd(chan, rtcp_fd, ast_rtp_instance_fd(media->rtp, 1));
488                 }
489         }
490
491         return changed;
492 }
493
494 /*! \brief Function called by RTP engine to change where the remote party should send media */
495 static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
496                 struct ast_rtp_instance *rtp,
497                 struct ast_rtp_instance *vrtp,
498                 struct ast_rtp_instance *tpeer,
499                 const struct ast_format_cap *cap,
500                 int nat_active)
501 {
502         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
503         struct chan_pjsip_pvt *pvt = channel->pvt;
504         struct ast_sip_session *session = channel->session;
505         int changed = 0;
506         struct ast_channel *bridge_peer;
507
508         /* Don't try to do any direct media shenanigans on early bridges */
509         bridge_peer = ast_channel_bridge_peer(chan);
510         if ((rtp || vrtp || tpeer) && !bridge_peer) {
511                 return 0;
512         }
513         ast_channel_cleanup(bridge_peer);
514
515         if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
516                 return 0;
517         }
518
519         if (pvt->media[SIP_MEDIA_AUDIO]) {
520                 changed |= check_for_rtp_changes(chan, rtp, pvt->media[SIP_MEDIA_AUDIO], 1);
521         }
522         if (pvt->media[SIP_MEDIA_VIDEO]) {
523                 changed |= check_for_rtp_changes(chan, vrtp, pvt->media[SIP_MEDIA_VIDEO], 3);
524         }
525
526         if (direct_media_mitigate_glare(session)) {
527                 return 0;
528         }
529
530         if (cap && !ast_format_cap_is_empty(cap) && !ast_format_cap_identical(session->direct_media_cap, cap)) {
531                 ast_format_cap_copy(session->direct_media_cap, cap);
532                 changed = 1;
533         }
534
535         if (changed) {
536                 ao2_ref(session, +1);
537
538
539                 if (ast_sip_push_task(session->serializer, send_direct_media_request, session)) {
540                         ao2_cleanup(session);
541                 }
542         }
543
544         return 0;
545 }
546
547 /*! \brief Local glue for interacting with the RTP engine core */
548 static struct ast_rtp_glue chan_pjsip_rtp_glue = {
549         .type = "PJSIP",
550         .get_rtp_info = chan_pjsip_get_rtp_peer,
551         .get_vrtp_info = chan_pjsip_get_vrtp_peer,
552         .get_codec = chan_pjsip_get_codec,
553         .update_peer = chan_pjsip_set_rtp_peer,
554 };
555
556 /*! \brief Function called to create a new PJSIP Asterisk channel */
557 static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const char *linkedid, const char *cid_name)
558 {
559         struct ast_channel *chan;
560         struct ast_format fmt;
561         RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
562         struct ast_sip_channel_pvt *channel;
563
564         if (!(pvt = ao2_alloc(sizeof(*pvt), chan_pjsip_pvt_dtor))) {
565                 return NULL;
566         }
567
568         if (!(chan = ast_channel_alloc(1, state, S_OR(session->id.number.str, ""), S_OR(session->id.name.str, ""), "", "", "", linkedid, 0, "PJSIP/%s-%08x", ast_sorcery_object_get_id(session->endpoint),
569                 ast_atomic_fetchadd_int((int *)&chan_idx, +1)))) {
570                 return NULL;
571         }
572
573         ast_channel_tech_set(chan, &chan_pjsip_tech);
574
575         if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
576                 ast_hangup(chan);
577                 return NULL;
578         }
579
580         /* If res_pjsip_session is ever updated to create/destroy ast_sip_session_media
581          * during a call such as if multiple same-type stream support is introduced,
582          * these will need to be recaptured as well */
583         pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
584         pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
585         ast_channel_tech_pvt_set(chan, channel);
586         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
587                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(chan));
588         }
589         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
590                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(chan));
591         }
592
593         if (ast_format_cap_is_empty(session->req_caps) || !ast_format_cap_has_joint(session->req_caps, session->endpoint->media.codecs)) {
594                 ast_format_cap_copy(ast_channel_nativeformats(chan), session->endpoint->media.codecs);
595         } else {
596                 ast_format_cap_copy(ast_channel_nativeformats(chan), session->req_caps);
597         }
598
599         ast_codec_choose(&session->endpoint->media.prefs, ast_channel_nativeformats(chan), 1, &fmt);
600         ast_format_copy(ast_channel_writeformat(chan), &fmt);
601         ast_format_copy(ast_channel_rawwriteformat(chan), &fmt);
602         ast_format_copy(ast_channel_readformat(chan), &fmt);
603         ast_format_copy(ast_channel_rawreadformat(chan), &fmt);
604
605         if (state == AST_STATE_RING) {
606                 ast_channel_rings_set(chan, 1);
607         }
608
609         ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
610
611         ast_channel_context_set(chan, session->endpoint->context);
612         ast_channel_exten_set(chan, S_OR(exten, "s"));
613         ast_channel_priority_set(chan, 1);
614
615         ast_channel_callgroup_set(chan, session->endpoint->pickup.callgroup);
616         ast_channel_pickupgroup_set(chan, session->endpoint->pickup.pickupgroup);
617
618         ast_channel_named_callgroups_set(chan, session->endpoint->pickup.named_callgroups);
619         ast_channel_named_pickupgroups_set(chan, session->endpoint->pickup.named_pickupgroups);
620
621         if (!ast_strlen_zero(session->endpoint->language)) {
622                 ast_channel_language_set(chan, session->endpoint->language);
623         }
624
625         if (!ast_strlen_zero(session->endpoint->zone)) {
626                 struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone);
627                 if (!zone) {
628                         ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone);
629                 }
630                 ast_channel_zone_set(chan, zone);
631         }
632
633         ast_endpoint_add_channel(session->endpoint->persistent, chan);
634
635         return chan;
636 }
637
638 static int answer(void *data)
639 {
640         pj_status_t status;
641         pjsip_tx_data *packet;
642         struct ast_sip_session *session = data;
643
644         if ((status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet)) == PJ_SUCCESS) {
645                 ast_sip_session_send_response(session, packet);
646         }
647
648         ao2_ref(session, -1);
649
650         return (status == PJ_SUCCESS) ? 0 : -1;
651 }
652
653 /*! \brief Function called by core when we should answer a PJSIP session */
654 static int chan_pjsip_answer(struct ast_channel *ast)
655 {
656         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
657
658         if (ast_channel_state(ast) == AST_STATE_UP) {
659                 return 0;
660         }
661
662         ast_setstate(ast, AST_STATE_UP);
663
664         ao2_ref(channel->session, +1);
665         if (ast_sip_push_task(channel->session->serializer, answer, channel->session)) {
666                 ast_log(LOG_WARNING, "Unable to push answer task to the threadpool. Cannot answer call\n");
667                 ao2_cleanup(channel->session);
668                 return -1;
669         }
670
671         return 0;
672 }
673
674 /*! \brief Internal helper function called when CNG tone is detected */
675 static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f)
676 {
677         const char *target_context;
678         int exists;
679
680         /* If we only needed this DSP for fax detection purposes we can just drop it now */
681         if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
682                 ast_dsp_set_features(session->dsp, DSP_FEATURE_DIGIT_DETECT);
683         } else {
684                 ast_dsp_free(session->dsp);
685                 session->dsp = NULL;
686         }
687
688         /* If already executing in the fax extension don't do anything */
689         if (!strcmp(ast_channel_exten(session->channel), "fax")) {
690                 return f;
691         }
692
693         target_context = S_OR(ast_channel_macrocontext(session->channel), ast_channel_context(session->channel));
694
695         /* We need to unlock the channel here because ast_exists_extension has the
696          * potential to start and stop an autoservice on the channel. Such action
697          * is prone to deadlock if the channel is locked.
698          */
699         ast_channel_unlock(session->channel);
700         exists = ast_exists_extension(session->channel, target_context, "fax", 1,
701                 S_COR(ast_channel_caller(session->channel)->id.number.valid,
702                         ast_channel_caller(session->channel)->id.number.str, NULL));
703         ast_channel_lock(session->channel);
704
705         if (exists) {
706                 ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
707                         ast_channel_name(session->channel));
708                 pbx_builtin_setvar_helper(session->channel, "FAXEXTEN", ast_channel_exten(session->channel));
709                 if (ast_async_goto(session->channel, target_context, "fax", 1)) {
710                         ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
711                                 ast_channel_name(session->channel), target_context);
712                 }
713                 ast_frfree(f);
714                 f = &ast_null_frame;
715         } else {
716                 ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
717                         ast_channel_name(session->channel), target_context);
718         }
719
720         return f;
721 }
722
723 /*! \brief Function called by core to read any waiting frames */
724 static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
725 {
726         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
727         struct chan_pjsip_pvt *pvt = channel->pvt;
728         struct ast_frame *f;
729         struct ast_sip_session_media *media = NULL;
730         int rtcp = 0;
731         int fdno = ast_channel_fdno(ast);
732
733         switch (fdno) {
734         case 0:
735                 media = pvt->media[SIP_MEDIA_AUDIO];
736                 break;
737         case 1:
738                 media = pvt->media[SIP_MEDIA_AUDIO];
739                 rtcp = 1;
740                 break;
741         case 2:
742                 media = pvt->media[SIP_MEDIA_VIDEO];
743                 break;
744         case 3:
745                 media = pvt->media[SIP_MEDIA_VIDEO];
746                 rtcp = 1;
747                 break;
748         }
749
750         if (!media || !media->rtp) {
751                 return &ast_null_frame;
752         }
753
754         if (!(f = ast_rtp_instance_read(media->rtp, rtcp))) {
755                 return f;
756         }
757
758         if (f->frametype != AST_FRAME_VOICE) {
759                 return f;
760         }
761
762         if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &f->subclass.format))) {
763                 ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(&f->subclass.format));
764                 ast_format_cap_set(ast_channel_nativeformats(ast), &f->subclass.format);
765                 ast_set_read_format(ast, ast_channel_readformat(ast));
766                 ast_set_write_format(ast, ast_channel_writeformat(ast));
767         }
768
769         if (channel->session->dsp) {
770                 f = ast_dsp_process(ast, channel->session->dsp, f);
771
772                 if (f && (f->frametype == AST_FRAME_DTMF)) {
773                         if (f->subclass.integer == 'f') {
774                                 ast_debug(3, "Fax CNG detected on %s\n", ast_channel_name(ast));
775                                 f = chan_pjsip_cng_tone_detected(channel->session, f);
776                         } else {
777                                 ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
778                                         ast_channel_name(ast));
779                         }
780                 }
781         }
782
783         return f;
784 }
785
786 /*! \brief Function called by core to write frames */
787 static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame)
788 {
789         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
790         struct chan_pjsip_pvt *pvt = channel->pvt;
791         struct ast_sip_session_media *media;
792         int res = 0;
793
794         switch (frame->frametype) {
795         case AST_FRAME_VOICE:
796                 media = pvt->media[SIP_MEDIA_AUDIO];
797
798                 if (!media) {
799                         return 0;
800                 }
801                 if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) {
802                         char buf[256];
803
804                         ast_log(LOG_WARNING,
805                                 "Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
806                                 ast_getformatname(&frame->subclass.format),
807                                 ast_getformatname_multiple(buf, sizeof(buf), ast_channel_nativeformats(ast)),
808                                 ast_getformatname(ast_channel_readformat(ast)),
809                                 ast_getformatname(ast_channel_writeformat(ast)));
810                         return 0;
811                 }
812                 if (media->rtp) {
813                         res = ast_rtp_instance_write(media->rtp, frame);
814                 }
815                 break;
816         case AST_FRAME_VIDEO:
817                 if ((media = pvt->media[SIP_MEDIA_VIDEO]) && media->rtp) {
818                         res = ast_rtp_instance_write(media->rtp, frame);
819                 }
820                 break;
821         case AST_FRAME_MODEM:
822                 break;
823         default:
824                 ast_log(LOG_WARNING, "Can't send %d type frames with PJSIP\n", frame->frametype);
825                 break;
826         }
827
828         return res;
829 }
830
831 struct fixup_data {
832         struct ast_sip_session *session;
833         struct ast_channel *chan;
834 };
835
836 static int fixup(void *data)
837 {
838         struct fixup_data *fix_data = data;
839         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(fix_data->chan);
840         struct chan_pjsip_pvt *pvt = channel->pvt;
841
842         channel->session->channel = fix_data->chan;
843         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
844                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(fix_data->chan));
845         }
846         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
847                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(fix_data->chan));
848         }
849
850         return 0;
851 }
852
853 /*! \brief Function called by core to change the underlying owner channel */
854 static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
855 {
856         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
857         struct fixup_data fix_data;
858
859         fix_data.session = channel->session;
860         fix_data.chan = newchan;
861
862         if (channel->session->channel != oldchan) {
863                 return -1;
864         }
865
866         if (ast_sip_push_task_synchronous(channel->session->serializer, fixup, &fix_data)) {
867                 ast_log(LOG_WARNING, "Unable to perform channel fixup\n");
868                 return -1;
869         }
870
871         return 0;
872 }
873
874 /*! \brief Function called to get the device state of an endpoint */
875 static int chan_pjsip_devicestate(const char *data)
876 {
877         RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
878         enum ast_device_state state = AST_DEVICE_UNKNOWN;
879         RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
880         RAII_VAR(struct stasis_caching_topic *, caching_topic, NULL, ao2_cleanup);
881         struct ast_devstate_aggregate aggregate;
882         int num, inuse = 0;
883
884         if (!endpoint) {
885                 return AST_DEVICE_INVALID;
886         }
887
888         endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent),
889                 ast_endpoint_get_resource(endpoint->persistent), 1);
890
891         if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) {
892                 state = AST_DEVICE_UNAVAILABLE;
893         } else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) {
894                 state = AST_DEVICE_NOT_INUSE;
895         }
896
897         if (!endpoint_snapshot->num_channels || !(caching_topic = ast_channel_topic_all_cached())) {
898                 return state;
899         }
900
901         ast_devstate_aggregate_init(&aggregate);
902
903         ao2_ref(caching_topic, +1);
904
905         for (num = 0; num < endpoint_snapshot->num_channels; num++) {
906                 RAII_VAR(struct stasis_message *, msg, stasis_cache_get_extended(caching_topic, ast_channel_snapshot_type(),
907                         endpoint_snapshot->channel_ids[num], 1), ao2_cleanup);
908                 struct ast_channel_snapshot *snapshot;
909
910                 if (!msg) {
911                         continue;
912                 }
913
914                 snapshot = stasis_message_data(msg);
915
916                 if (snapshot->state == AST_STATE_DOWN) {
917                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_NOT_INUSE);
918                 } else if (snapshot->state == AST_STATE_RINGING) {
919                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_RINGING);
920                 } else if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
921                         (snapshot->state == AST_STATE_BUSY)) {
922                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_INUSE);
923                         inuse++;
924                 }
925         }
926
927         if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) {
928                 state = AST_DEVICE_BUSY;
929         } else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) {
930                 state = ast_devstate_aggregate_result(&aggregate);
931         }
932
933         return state;
934 }
935
936 /*! \brief Function called to query options on a channel */
937 static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen)
938 {
939         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
940         struct ast_sip_session *session = channel->session;
941         int res = -1;
942         enum ast_sip_session_t38state state = T38_STATE_UNAVAILABLE;
943
944         switch (option) {
945         case AST_OPTION_T38_STATE:
946                 if (session->endpoint->media.t38.enabled) {
947                         switch (session->t38state) {
948                         case T38_LOCAL_REINVITE:
949                         case T38_PEER_REINVITE:
950                                 state = T38_STATE_NEGOTIATING;
951                                 break;
952                         case T38_ENABLED:
953                                 state = T38_STATE_NEGOTIATED;
954                                 break;
955                         case T38_REJECTED:
956                                 state = T38_STATE_REJECTED;
957                                 break;
958                         default:
959                                 state = T38_STATE_UNKNOWN;
960                                 break;
961                         }
962                 }
963
964                 *((enum ast_t38_state *) data) = state;
965                 res = 0;
966
967                 break;
968         default:
969                 break;
970         }
971
972         return res;
973 }
974
975 struct indicate_data {
976         struct ast_sip_session *session;
977         int condition;
978         int response_code;
979         void *frame_data;
980         size_t datalen;
981 };
982
983 static void indicate_data_destroy(void *obj)
984 {
985         struct indicate_data *ind_data = obj;
986
987         ast_free(ind_data->frame_data);
988         ao2_ref(ind_data->session, -1);
989 }
990
991 static struct indicate_data *indicate_data_alloc(struct ast_sip_session *session,
992                 int condition, int response_code, const void *frame_data, size_t datalen)
993 {
994         struct indicate_data *ind_data = ao2_alloc(sizeof(*ind_data), indicate_data_destroy);
995
996         if (!ind_data) {
997                 return NULL;
998         }
999
1000         ind_data->frame_data = ast_malloc(datalen);
1001         if (!ind_data->frame_data) {
1002                 ao2_ref(ind_data, -1);
1003                 return NULL;
1004         }
1005
1006         memcpy(ind_data->frame_data, frame_data, datalen);
1007         ind_data->datalen = datalen;
1008         ind_data->condition = condition;
1009         ind_data->response_code = response_code;
1010         ao2_ref(session, +1);
1011         ind_data->session = session;
1012
1013         return ind_data;
1014 }
1015
1016 static int indicate(void *data)
1017 {
1018         pjsip_tx_data *packet = NULL;
1019         struct indicate_data *ind_data = data;
1020         struct ast_sip_session *session = ind_data->session;
1021         int response_code = ind_data->response_code;
1022
1023         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
1024                 ast_sip_session_send_response(session, packet);
1025         }
1026
1027         ao2_ref(ind_data, -1);
1028
1029         return 0;
1030 }
1031
1032 /*! \brief Send SIP INFO with video update request */
1033 static int transmit_info_with_vidupdate(void *data)
1034 {
1035         const char * xml =
1036                 "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
1037                 " <media_control>\r\n"
1038                 "  <vc_primitive>\r\n"
1039                 "   <to_encoder>\r\n"
1040                 "    <picture_fast_update/>\r\n"
1041                 "   </to_encoder>\r\n"
1042                 "  </vc_primitive>\r\n"
1043                 " </media_control>\r\n";
1044
1045         const struct ast_sip_body body = {
1046                 .type = "application",
1047                 .subtype = "media_control+xml",
1048                 .body_text = xml
1049         };
1050
1051         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
1052         struct pjsip_tx_data *tdata;
1053
1054         if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
1055                 ast_log(LOG_ERROR, "Could not create text video update INFO request\n");
1056                 return -1;
1057         }
1058         if (ast_sip_add_body(tdata, &body)) {
1059                 ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n");
1060                 return -1;
1061         }
1062         ast_sip_session_send_request(session, tdata);
1063
1064         return 0;
1065 }
1066
1067 /*! \brief Update connected line information */
1068 static int update_connected_line_information(void *data)
1069 {
1070         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
1071
1072         if ((ast_channel_state(session->channel) != AST_STATE_UP) && (session->inv_session->role == PJSIP_UAS_ROLE)) {
1073                 int response_code = 0;
1074
1075                 if (ast_channel_state(session->channel) == AST_STATE_RING) {
1076                         response_code = !session->endpoint->inband_progress ? 180 : 183;
1077                 } else if (ast_channel_state(session->channel) == AST_STATE_RINGING) {
1078                         response_code = 183;
1079                 }
1080
1081                 if (response_code) {
1082                         struct pjsip_tx_data *packet = NULL;
1083
1084                         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
1085                                 ast_sip_session_send_response(session, packet);
1086                         }
1087                 }
1088         } else {
1089                 enum ast_sip_session_refresh_method method = session->endpoint->id.refresh_method;
1090
1091                 if (session->inv_session->invite_tsx && (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
1092                         method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
1093                 }
1094
1095                 ast_sip_session_refresh(session, NULL, NULL, NULL, method, 0);
1096         }
1097
1098         return 0;
1099 }
1100
1101 /*! \brief Function called by core to ask the channel to indicate some sort of condition */
1102 static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
1103 {
1104         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1105         struct chan_pjsip_pvt *pvt = channel->pvt;
1106         struct ast_sip_session_media *media;
1107         int response_code = 0;
1108         int res = 0;
1109
1110         switch (condition) {
1111         case AST_CONTROL_RINGING:
1112                 if (ast_channel_state(ast) == AST_STATE_RING) {
1113                         if (channel->session->endpoint->inband_progress) {
1114                                 response_code = 183;
1115                                 res = -1;
1116                         } else {
1117                                 response_code = 180;
1118                         }
1119                 } else {
1120                         res = -1;
1121                 }
1122                 ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
1123                 break;
1124         case AST_CONTROL_BUSY:
1125                 if (ast_channel_state(ast) != AST_STATE_UP) {
1126                         response_code = 486;
1127                 } else {
1128                         res = -1;
1129                 }
1130                 break;
1131         case AST_CONTROL_CONGESTION:
1132                 if (ast_channel_state(ast) != AST_STATE_UP) {
1133                         response_code = 503;
1134                 } else {
1135                         res = -1;
1136                 }
1137                 break;
1138         case AST_CONTROL_INCOMPLETE:
1139                 if (ast_channel_state(ast) != AST_STATE_UP) {
1140                         response_code = 484;
1141                 } else {
1142                         res = -1;
1143                 }
1144                 break;
1145         case AST_CONTROL_PROCEEDING:
1146                 if (ast_channel_state(ast) != AST_STATE_UP) {
1147                         response_code = 100;
1148                 } else {
1149                         res = -1;
1150                 }
1151                 break;
1152         case AST_CONTROL_PROGRESS:
1153                 if (ast_channel_state(ast) != AST_STATE_UP) {
1154                         response_code = 183;
1155                 } else {
1156                         res = -1;
1157                 }
1158                 break;
1159         case AST_CONTROL_VIDUPDATE:
1160                 media = pvt->media[SIP_MEDIA_VIDEO];
1161                 if (media && media->rtp) {
1162                         ao2_ref(channel->session, +1);
1163
1164                         if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
1165                                 ao2_cleanup(channel->session);
1166                         }
1167                 } else {
1168                         res = -1;
1169                 }
1170                 break;
1171         case AST_CONTROL_CONNECTED_LINE:
1172                 ao2_ref(channel->session, +1);
1173                 if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
1174                         ao2_cleanup(channel->session);
1175                 }
1176                 break;
1177         case AST_CONTROL_UPDATE_RTP_PEER:
1178                 break;
1179         case AST_CONTROL_PVT_CAUSE_CODE:
1180                 res = -1;
1181                 break;
1182         case AST_CONTROL_HOLD:
1183                 ast_moh_start(ast, data, NULL);
1184                 break;
1185         case AST_CONTROL_UNHOLD:
1186                 ast_moh_stop(ast);
1187                 break;
1188         case AST_CONTROL_SRCUPDATE:
1189                 break;
1190         case AST_CONTROL_SRCCHANGE:
1191                 break;
1192         case AST_CONTROL_REDIRECTING:
1193                 if (ast_channel_state(ast) != AST_STATE_UP) {
1194                         response_code = 181;
1195                 } else {
1196                         res = -1;
1197                 }
1198                 break;
1199         case AST_CONTROL_T38_PARAMETERS:
1200                 res = 0;
1201
1202                 if (channel->session->t38state == T38_PEER_REINVITE) {
1203                         const struct ast_control_t38_parameters *parameters = data;
1204
1205                         if (parameters->request_response == AST_T38_REQUEST_PARMS) {
1206                                 res = AST_T38_REQUEST_PARMS;
1207                         }
1208                 }
1209
1210                 break;
1211         case -1:
1212                 res = -1;
1213                 break;
1214         default:
1215                 ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
1216                 res = -1;
1217                 break;
1218         }
1219
1220         if (response_code) {
1221                 struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
1222                 if (!ind_data || ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
1223                         ast_log(LOG_NOTICE, "Cannot send response code %d to endpoint %s. Could not queue task properly\n",
1224                                         response_code, ast_sorcery_object_get_id(channel->session->endpoint));
1225                         ao2_cleanup(ind_data);
1226                         res = -1;
1227                 }
1228         }
1229
1230         return res;
1231 }
1232
1233 struct transfer_data {
1234         struct ast_sip_session *session;
1235         char *target;
1236 };
1237
1238 static void transfer_data_destroy(void *obj)
1239 {
1240         struct transfer_data *trnf_data = obj;
1241
1242         ast_free(trnf_data->target);
1243         ao2_cleanup(trnf_data->session);
1244 }
1245
1246 static struct transfer_data *transfer_data_alloc(struct ast_sip_session *session, const char *target)
1247 {
1248         struct transfer_data *trnf_data = ao2_alloc(sizeof(*trnf_data), transfer_data_destroy);
1249
1250         if (!trnf_data) {
1251                 return NULL;
1252         }
1253
1254         if (!(trnf_data->target = ast_strdup(target))) {
1255                 ao2_ref(trnf_data, -1);
1256                 return NULL;
1257         }
1258
1259         ao2_ref(session, +1);
1260         trnf_data->session = session;
1261
1262         return trnf_data;
1263 }
1264
1265 static void transfer_redirect(struct ast_sip_session *session, const char *target)
1266 {
1267         pjsip_tx_data *packet;
1268         enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1269         pjsip_contact_hdr *contact;
1270         pj_str_t tmp;
1271
1272         if (pjsip_inv_end_session(session->inv_session, 302, NULL, &packet) != PJ_SUCCESS) {
1273                 message = AST_TRANSFER_FAILED;
1274                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1275
1276                 return;
1277         }
1278
1279         if (!(contact = pjsip_msg_find_hdr(packet->msg, PJSIP_H_CONTACT, NULL))) {
1280                 contact = pjsip_contact_hdr_create(packet->pool);
1281         }
1282
1283         pj_strdup2_with_null(packet->pool, &tmp, target);
1284         if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) {
1285                 message = AST_TRANSFER_FAILED;
1286                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1287                 pjsip_tx_data_dec_ref(packet);
1288
1289                 return;
1290         }
1291         pjsip_msg_add_hdr(packet->msg, (pjsip_hdr *) contact);
1292
1293         ast_sip_session_send_response(session, packet);
1294         ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1295 }
1296
1297 static void transfer_refer(struct ast_sip_session *session, const char *target)
1298 {
1299         pjsip_evsub *sub;
1300         enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1301         pj_str_t tmp;
1302         pjsip_tx_data *packet;
1303
1304         if (pjsip_xfer_create_uac(session->inv_session->dlg, NULL, &sub) != PJ_SUCCESS) {
1305                 message = AST_TRANSFER_FAILED;
1306                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1307
1308                 return;
1309         }
1310
1311         if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
1312                 message = AST_TRANSFER_FAILED;
1313                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1314                 pjsip_evsub_terminate(sub, PJ_FALSE);
1315
1316                 return;
1317         }
1318
1319         pjsip_xfer_send_request(sub, packet);
1320         ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1321 }
1322
1323 static int transfer(void *data)
1324 {
1325         struct transfer_data *trnf_data = data;
1326
1327         if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
1328                 transfer_redirect(trnf_data->session, trnf_data->target);
1329         } else {
1330                 transfer_refer(trnf_data->session, trnf_data->target);
1331         }
1332
1333         ao2_ref(trnf_data, -1);
1334         return 0;
1335 }
1336
1337 /*! \brief Function called by core for Asterisk initiated transfer */
1338 static int chan_pjsip_transfer(struct ast_channel *chan, const char *target)
1339 {
1340         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
1341         struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
1342
1343         if (!trnf_data) {
1344                 return -1;
1345         }
1346
1347         if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
1348                 ast_log(LOG_WARNING, "Error requesting transfer\n");
1349                 ao2_cleanup(trnf_data);
1350                 return -1;
1351         }
1352
1353         return 0;
1354 }
1355
1356 /*! \brief Function called by core to start a DTMF digit */
1357 static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
1358 {
1359         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
1360         struct chan_pjsip_pvt *pvt = channel->pvt;
1361         struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
1362         int res = 0;
1363
1364         switch (channel->session->endpoint->dtmf) {
1365         case AST_SIP_DTMF_RFC_4733:
1366                 if (!media || !media->rtp) {
1367                         return -1;
1368                 }
1369
1370                 ast_rtp_instance_dtmf_begin(media->rtp, digit);
1371         case AST_SIP_DTMF_NONE:
1372                 break;
1373         case AST_SIP_DTMF_INBAND:
1374                 res = -1;
1375                 break;
1376         default:
1377                 break;
1378         }
1379
1380         return res;
1381 }
1382
1383 struct info_dtmf_data {
1384         struct ast_sip_session *session;
1385         char digit;
1386         unsigned int duration;
1387 };
1388
1389 static void info_dtmf_data_destroy(void *obj)
1390 {
1391         struct info_dtmf_data *dtmf_data = obj;
1392         ao2_ref(dtmf_data->session, -1);
1393 }
1394
1395 static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
1396 {
1397         struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy);
1398         if (!dtmf_data) {
1399                 return NULL;
1400         }
1401         ao2_ref(session, +1);
1402         dtmf_data->session = session;
1403         dtmf_data->digit = digit;
1404         dtmf_data->duration = duration;
1405         return dtmf_data;
1406 }
1407
1408 static int transmit_info_dtmf(void *data)
1409 {
1410         RAII_VAR(struct info_dtmf_data *, dtmf_data, data, ao2_cleanup);
1411
1412         struct ast_sip_session *session = dtmf_data->session;
1413         struct pjsip_tx_data *tdata;
1414
1415         RAII_VAR(struct ast_str *, body_text, NULL, ast_free_ptr);
1416
1417         struct ast_sip_body body = {
1418                 .type = "application",
1419                 .subtype = "dtmf-relay",
1420         };
1421
1422         if (!(body_text = ast_str_create(32))) {
1423                 ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n");
1424                 return -1;
1425         }
1426         ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration);
1427
1428         body.body_text = ast_str_buffer(body_text);
1429
1430         if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
1431                 ast_log(LOG_ERROR, "Could not create DTMF INFO request\n");
1432                 return -1;
1433         }
1434         if (ast_sip_add_body(tdata, &body)) {
1435                 ast_log(LOG_ERROR, "Could not add body to DTMF INFO request\n");
1436                 pjsip_tx_data_dec_ref(tdata);
1437                 return -1;
1438         }
1439         ast_sip_session_send_request(session, tdata);
1440
1441         return 0;
1442 }
1443
1444 /*! \brief Function called by core to stop a DTMF digit */
1445 static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
1446 {
1447         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1448         struct chan_pjsip_pvt *pvt = channel->pvt;
1449         struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
1450         int res = 0;
1451
1452         switch (channel->session->endpoint->dtmf) {
1453         case AST_SIP_DTMF_INFO:
1454         {
1455                 struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
1456
1457                 if (!dtmf_data) {
1458                         return -1;
1459                 }
1460
1461                 if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
1462                         ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
1463                         ao2_cleanup(dtmf_data);
1464                         return -1;
1465                 }
1466                 break;
1467         }
1468         case AST_SIP_DTMF_RFC_4733:
1469                 if (!media || !media->rtp) {
1470                         return -1;
1471                 }
1472
1473                 ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
1474         case AST_SIP_DTMF_NONE:
1475                 break;
1476         case AST_SIP_DTMF_INBAND:
1477                 res = -1;
1478                 break;
1479         }
1480
1481         return res;
1482 }
1483
1484 static int call(void *data)
1485 {
1486         struct ast_sip_session *session = data;
1487         pjsip_tx_data *tdata;
1488
1489         int res = ast_sip_session_create_invite(session, &tdata);
1490
1491         if (res) {
1492                 ast_queue_hangup(session->channel);
1493         } else {
1494                 ast_sip_session_send_request(session, tdata);
1495         }
1496         ao2_ref(session, -1);
1497         return res;
1498 }
1499
1500 /*! \brief Function called by core to actually start calling a remote party */
1501 static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout)
1502 {
1503         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1504
1505         ao2_ref(channel->session, +1);
1506         if (ast_sip_push_task(channel->session->serializer, call, channel->session)) {
1507                 ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
1508                 ao2_cleanup(channel->session);
1509                 return -1;
1510         }
1511
1512         return 0;
1513 }
1514
1515 /*! \brief Internal function which translates from Asterisk cause codes to SIP response codes */
1516 static int hangup_cause2sip(int cause)
1517 {
1518         switch (cause) {
1519         case AST_CAUSE_UNALLOCATED:             /* 1 */
1520         case AST_CAUSE_NO_ROUTE_DESTINATION:    /* 3 IAX2: Can't find extension in context */
1521         case AST_CAUSE_NO_ROUTE_TRANSIT_NET:    /* 2 */
1522                 return 404;
1523         case AST_CAUSE_CONGESTION:              /* 34 */
1524         case AST_CAUSE_SWITCH_CONGESTION:       /* 42 */
1525                 return 503;
1526         case AST_CAUSE_NO_USER_RESPONSE:        /* 18 */
1527                 return 408;
1528         case AST_CAUSE_NO_ANSWER:               /* 19 */
1529         case AST_CAUSE_UNREGISTERED:        /* 20 */
1530                 return 480;
1531         case AST_CAUSE_CALL_REJECTED:           /* 21 */
1532                 return 403;
1533         case AST_CAUSE_NUMBER_CHANGED:          /* 22 */
1534                 return 410;
1535         case AST_CAUSE_NORMAL_UNSPECIFIED:      /* 31 */
1536                 return 480;
1537         case AST_CAUSE_INVALID_NUMBER_FORMAT:
1538                 return 484;
1539         case AST_CAUSE_USER_BUSY:
1540                 return 486;
1541         case AST_CAUSE_FAILURE:
1542                 return 500;
1543         case AST_CAUSE_FACILITY_REJECTED:       /* 29 */
1544                 return 501;
1545         case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
1546                 return 503;
1547         case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
1548                 return 502;
1549         case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:       /* Can't find codec to connect to host */
1550                 return 488;
1551         case AST_CAUSE_INTERWORKING:    /* Unspecified Interworking issues */
1552                 return 500;
1553         case AST_CAUSE_NOTDEFINED:
1554         default:
1555                 ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
1556                 return 0;
1557         }
1558
1559         /* Never reached */
1560         return 0;
1561 }
1562
1563 struct hangup_data {
1564         int cause;
1565         struct ast_channel *chan;
1566 };
1567
1568 static void hangup_data_destroy(void *obj)
1569 {
1570         struct hangup_data *h_data = obj;
1571
1572         h_data->chan = ast_channel_unref(h_data->chan);
1573 }
1574
1575 static struct hangup_data *hangup_data_alloc(int cause, struct ast_channel *chan)
1576 {
1577         struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
1578
1579         if (!h_data) {
1580                 return NULL;
1581         }
1582
1583         h_data->cause = cause;
1584         h_data->chan = ast_channel_ref(chan);
1585
1586         return h_data;
1587 }
1588
1589 /*! \brief Clear a channel from a session along with its PVT */
1590 static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast, struct chan_pjsip_pvt *pvt)
1591 {
1592         session->channel = NULL;
1593         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
1594                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, "");
1595         }
1596         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
1597                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, "");
1598         }
1599         ast_channel_tech_pvt_set(ast, NULL);
1600 }
1601
1602 static int hangup(void *data)
1603 {
1604         pj_status_t status;
1605         pjsip_tx_data *packet = NULL;
1606         struct hangup_data *h_data = data;
1607         struct ast_channel *ast = h_data->chan;
1608         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1609         struct chan_pjsip_pvt *pvt = channel->pvt;
1610         struct ast_sip_session *session = channel->session;
1611         int cause = h_data->cause;
1612
1613         if (!session->defer_terminate &&
1614                 ((status = pjsip_inv_end_session(session->inv_session, cause ? cause : 603, NULL, &packet)) == PJ_SUCCESS) && packet) {
1615                 if (packet->msg->type == PJSIP_RESPONSE_MSG) {
1616                         ast_sip_session_send_response(session, packet);
1617                 } else {
1618                         ast_sip_session_send_request(session, packet);
1619                 }
1620         }
1621
1622         clear_session_and_channel(session, ast, pvt);
1623         ao2_cleanup(channel);
1624         ao2_cleanup(h_data);
1625
1626         return 0;
1627 }
1628
1629 /*! \brief Function called by core to hang up a PJSIP session */
1630 static int chan_pjsip_hangup(struct ast_channel *ast)
1631 {
1632         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1633         struct chan_pjsip_pvt *pvt = channel->pvt;
1634         int cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
1635         struct hangup_data *h_data = hangup_data_alloc(cause, ast);
1636
1637         if (!h_data) {
1638                 goto failure;
1639         }
1640
1641         if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
1642                 ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
1643                 goto failure;
1644         }
1645
1646         return 0;
1647
1648 failure:
1649         /* Go ahead and do our cleanup of the session and channel even if we're not going
1650          * to be able to send our SIP request/response
1651          */
1652         clear_session_and_channel(channel->session, ast, pvt);
1653         ao2_cleanup(channel);
1654         ao2_cleanup(h_data);
1655
1656         return -1;
1657 }
1658
1659 struct request_data {
1660         struct ast_sip_session *session;
1661         struct ast_format_cap *caps;
1662         const char *dest;
1663         int cause;
1664 };
1665
1666 static int request(void *obj)
1667 {
1668         struct request_data *req_data = obj;
1669         struct ast_sip_session *session = NULL;
1670         char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
1671         RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
1672
1673         AST_DECLARE_APP_ARGS(args,
1674                 AST_APP_ARG(endpoint);
1675                 AST_APP_ARG(aor);
1676         );
1677
1678         if (ast_strlen_zero(tmp)) {
1679                 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
1680                 req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
1681                 return -1;
1682         }
1683
1684         AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
1685
1686         /* If a request user has been specified extract it from the endpoint name portion */
1687         if ((endpoint_name = strchr(args.endpoint, '@'))) {
1688                 request_user = args.endpoint;
1689                 *endpoint_name++ = '\0';
1690         } else {
1691                 endpoint_name = args.endpoint;
1692         }
1693
1694         if (ast_strlen_zero(endpoint_name)) {
1695                 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
1696                 req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
1697         } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name))) {
1698                 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
1699                 req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
1700                 return -1;
1701         }
1702
1703         if (!(session = ast_sip_session_create_outgoing(endpoint, args.aor, request_user, req_data->caps))) {
1704                 req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
1705                 return -1;
1706         }
1707
1708         req_data->session = session;
1709
1710         return 0;
1711 }
1712
1713 /*! \brief Function called by core to create a new outgoing PJSIP session */
1714 static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause)
1715 {
1716         struct request_data req_data;
1717         RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
1718
1719         req_data.caps = cap;
1720         req_data.dest = data;
1721
1722         if (ast_sip_push_task_synchronous(NULL, request, &req_data)) {
1723                 *cause = req_data.cause;
1724                 return NULL;
1725         }
1726
1727         session = req_data.session;
1728
1729         if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, requestor ? ast_channel_linkedid(requestor) : NULL, NULL))) {
1730                 /* Session needs to be terminated prematurely */
1731                 return NULL;
1732         }
1733
1734         return session->channel;
1735 }
1736
1737 struct sendtext_data {
1738         struct ast_sip_session *session;
1739         char text[0];
1740 };
1741
1742 static void sendtext_data_destroy(void *obj)
1743 {
1744         struct sendtext_data *data = obj;
1745         ao2_ref(data->session, -1);
1746 }
1747
1748 static struct sendtext_data* sendtext_data_create(struct ast_sip_session *session, const char *text)
1749 {
1750         int size = strlen(text) + 1;
1751         struct sendtext_data *data = ao2_alloc(sizeof(*data)+size, sendtext_data_destroy);
1752
1753         if (!data) {
1754                 return NULL;
1755         }
1756
1757         data->session = session;
1758         ao2_ref(data->session, +1);
1759         ast_copy_string(data->text, text, size);
1760         return data;
1761 }
1762
1763 static int sendtext(void *obj)
1764 {
1765         RAII_VAR(struct sendtext_data *, data, obj, ao2_cleanup);
1766         pjsip_tx_data *tdata;
1767
1768         const struct ast_sip_body body = {
1769                 .type = "text",
1770                 .subtype = "plain",
1771                 .body_text = data->text
1772         };
1773
1774         /* NOT ast_strlen_zero, because a zero-length message is specifically
1775          * allowed by RFC 3428 (See section 10, Examples) */
1776         if (!data->text) {
1777                 return 0;
1778         }
1779
1780         ast_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, &tdata);
1781         ast_sip_add_body(tdata, &body);
1782         ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint);
1783
1784         return 0;
1785 }
1786
1787 /*! \brief Function called by core to send text on PJSIP session */
1788 static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text)
1789 {
1790         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1791         struct sendtext_data *data = sendtext_data_create(channel->session, text);
1792
1793         if (!data || ast_sip_push_task(channel->session->serializer, sendtext, data)) {
1794                 ao2_ref(data, -1);
1795                 return -1;
1796         }
1797         return 0;
1798 }
1799
1800 /*! \brief Convert SIP hangup causes to Asterisk hangup causes */
1801 static int hangup_sip2cause(int cause)
1802 {
1803         /* Possible values taken from causes.h */
1804
1805         switch(cause) {
1806         case 401:       /* Unauthorized */
1807                 return AST_CAUSE_CALL_REJECTED;
1808         case 403:       /* Not found */
1809                 return AST_CAUSE_CALL_REJECTED;
1810         case 404:       /* Not found */
1811                 return AST_CAUSE_UNALLOCATED;
1812         case 405:       /* Method not allowed */
1813                 return AST_CAUSE_INTERWORKING;
1814         case 407:       /* Proxy authentication required */
1815                 return AST_CAUSE_CALL_REJECTED;
1816         case 408:       /* No reaction */
1817                 return AST_CAUSE_NO_USER_RESPONSE;
1818         case 409:       /* Conflict */
1819                 return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
1820         case 410:       /* Gone */
1821                 return AST_CAUSE_NUMBER_CHANGED;
1822         case 411:       /* Length required */
1823                 return AST_CAUSE_INTERWORKING;
1824         case 413:       /* Request entity too large */
1825                 return AST_CAUSE_INTERWORKING;
1826         case 414:       /* Request URI too large */
1827                 return AST_CAUSE_INTERWORKING;
1828         case 415:       /* Unsupported media type */
1829                 return AST_CAUSE_INTERWORKING;
1830         case 420:       /* Bad extension */
1831                 return AST_CAUSE_NO_ROUTE_DESTINATION;
1832         case 480:       /* No answer */
1833                 return AST_CAUSE_NO_ANSWER;
1834         case 481:       /* No answer */
1835                 return AST_CAUSE_INTERWORKING;
1836         case 482:       /* Loop detected */
1837                 return AST_CAUSE_INTERWORKING;
1838         case 483:       /* Too many hops */
1839                 return AST_CAUSE_NO_ANSWER;
1840         case 484:       /* Address incomplete */
1841                 return AST_CAUSE_INVALID_NUMBER_FORMAT;
1842         case 485:       /* Ambiguous */
1843                 return AST_CAUSE_UNALLOCATED;
1844         case 486:       /* Busy everywhere */
1845                 return AST_CAUSE_BUSY;
1846         case 487:       /* Request terminated */
1847                 return AST_CAUSE_INTERWORKING;
1848         case 488:       /* No codecs approved */
1849                 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
1850         case 491:       /* Request pending */
1851                 return AST_CAUSE_INTERWORKING;
1852         case 493:       /* Undecipherable */
1853                 return AST_CAUSE_INTERWORKING;
1854         case 500:       /* Server internal failure */
1855                 return AST_CAUSE_FAILURE;
1856         case 501:       /* Call rejected */
1857                 return AST_CAUSE_FACILITY_REJECTED;
1858         case 502:
1859                 return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
1860         case 503:       /* Service unavailable */
1861                 return AST_CAUSE_CONGESTION;
1862         case 504:       /* Gateway timeout */
1863                 return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
1864         case 505:       /* SIP version not supported */
1865                 return AST_CAUSE_INTERWORKING;
1866         case 600:       /* Busy everywhere */
1867                 return AST_CAUSE_USER_BUSY;
1868         case 603:       /* Decline */
1869                 return AST_CAUSE_CALL_REJECTED;
1870         case 604:       /* Does not exist anywhere */
1871                 return AST_CAUSE_UNALLOCATED;
1872         case 606:       /* Not acceptable */
1873                 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
1874         default:
1875                 if (cause < 500 && cause >= 400) {
1876                         /* 4xx class error that is unknown - someting wrong with our request */
1877                         return AST_CAUSE_INTERWORKING;
1878                 } else if (cause < 600 && cause >= 500) {
1879                         /* 5xx class error - problem in the remote end */
1880                         return AST_CAUSE_CONGESTION;
1881                 } else if (cause < 700 && cause >= 600) {
1882                         /* 6xx - global errors in the 4xx class */
1883                         return AST_CAUSE_INTERWORKING;
1884                 }
1885                 return AST_CAUSE_NORMAL;
1886         }
1887         /* Never reached */
1888         return 0;
1889 }
1890
1891 static void chan_pjsip_session_begin(struct ast_sip_session *session)
1892 {
1893         RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
1894
1895         if (session->endpoint->media.direct_media.glare_mitigation ==
1896                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
1897                 return;
1898         }
1899
1900         datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info,
1901                         "direct_media_glare_mitigation");
1902
1903         if (!datastore) {
1904                 return;
1905         }
1906
1907         ast_sip_session_add_datastore(session, datastore);
1908 }
1909
1910 /*! \brief Function called when the session ends */
1911 static void chan_pjsip_session_end(struct ast_sip_session *session)
1912 {
1913         if (!session->channel) {
1914                 return;
1915         }
1916
1917         if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
1918                 int cause = hangup_sip2cause(session->inv_session->cause);
1919
1920                 ast_queue_hangup_with_cause(session->channel, cause);
1921         } else {
1922                 ast_queue_hangup(session->channel);
1923         }
1924 }
1925
1926 /*! \brief Function called when a request is received on the session */
1927 static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
1928 {
1929         pjsip_tx_data *packet = NULL;
1930
1931         if (session->channel) {
1932                 return 0;
1933         }
1934
1935         if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL))) {
1936                 if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
1937                         ast_sip_session_send_response(session, packet);
1938                 }
1939
1940                 ast_log(LOG_ERROR, "Failed to allocate new PJSIP channel on incoming SIP INVITE\n");
1941                 return -1;
1942         }
1943         /* channel gets created on incoming request, but we wait to call start
1944            so other supplements have a chance to run */
1945         return 0;
1946 }
1947
1948 static int pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
1949 {
1950         int res;
1951
1952         res = ast_pbx_start(session->channel);
1953
1954         switch (res) {
1955         case AST_PBX_FAILED:
1956                 ast_log(LOG_WARNING, "Failed to start PBX ;(\n");
1957                 ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
1958                 ast_hangup(session->channel);
1959                 break;
1960         case AST_PBX_CALL_LIMIT:
1961                 ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
1962                 ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
1963                 ast_hangup(session->channel);
1964                 break;
1965         case AST_PBX_SUCCESS:
1966         default:
1967                 break;
1968         }
1969
1970         ast_debug(3, "Started PBX on new PJSIP channel %s\n", ast_channel_name(session->channel));
1971
1972         return (res == AST_PBX_SUCCESS) ? 0 : -1;
1973 }
1974
1975 static struct ast_sip_session_supplement pbx_start_supplement = {
1976         .method = "INVITE",
1977         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_LAST,
1978         .incoming_request = pbx_start_incoming_request,
1979 };
1980
1981 /*! \brief Function called when a response is received on the session */
1982 static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
1983 {
1984         struct pjsip_status_line status = rdata->msg_info.msg->line.status;
1985
1986         if (!session->channel) {
1987                 return;
1988         }
1989
1990         switch (status.code) {
1991         case 180:
1992                 ast_queue_control(session->channel, AST_CONTROL_RINGING);
1993                 if (ast_channel_state(session->channel) != AST_STATE_UP) {
1994                         ast_setstate(session->channel, AST_STATE_RINGING);
1995                 }
1996                 break;
1997         case 183:
1998                 ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
1999                 break;
2000         case 200:
2001                 ast_queue_control(session->channel, AST_CONTROL_ANSWER);
2002                 break;
2003         default:
2004                 break;
2005         }
2006 }
2007
2008 static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
2009 {
2010         if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
2011                 if (session->endpoint->media.direct_media.enabled) {
2012                         ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
2013                 }
2014         }
2015         return 0;
2016 }
2017
2018 /*!
2019  * \brief Load the module
2020  *
2021  * Module loading including tests for configuration or dependencies.
2022  * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
2023  * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
2024  * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
2025  * configuration file or other non-critical problem return
2026  * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
2027  */
2028 static int load_module(void)
2029 {
2030         if (!(chan_pjsip_tech.capabilities = ast_format_cap_alloc())) {
2031                 return AST_MODULE_LOAD_DECLINE;
2032         }
2033
2034         ast_format_cap_add_all_by_type(chan_pjsip_tech.capabilities, AST_FORMAT_TYPE_AUDIO);
2035
2036         ast_rtp_glue_register(&chan_pjsip_rtp_glue);
2037
2038         if (ast_channel_register(&chan_pjsip_tech)) {
2039                 ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
2040                 goto end;
2041         }
2042
2043         if (ast_custom_function_register(&chan_pjsip_dial_contacts_function)) {
2044                 ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n");
2045                 goto end;
2046         }
2047
2048         if (ast_custom_function_register(&media_offer_function)) {
2049                 ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
2050         }
2051
2052         if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
2053                 ast_log(LOG_ERROR, "Unable to register PJSIP supplement\n");
2054                 goto end;
2055         }
2056
2057         if (ast_sip_session_register_supplement(&pbx_start_supplement)) {
2058                 ast_log(LOG_ERROR, "Unable to register PJSIP pbx start supplement\n");
2059                 ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2060                 goto end;
2061         }
2062
2063         if (ast_sip_session_register_supplement(&chan_pjsip_ack_supplement)) {
2064                 ast_log(LOG_ERROR, "Unable to register PJSIP ACK supplement\n");
2065                 ast_sip_session_unregister_supplement(&pbx_start_supplement);
2066                 ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2067                 goto end;
2068         }
2069
2070         return 0;
2071
2072 end:
2073         ast_custom_function_unregister(&media_offer_function);
2074         ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
2075         ast_channel_unregister(&chan_pjsip_tech);
2076         ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
2077
2078         return AST_MODULE_LOAD_FAILURE;
2079 }
2080
2081 /*! \brief Reload module */
2082 static int reload(void)
2083 {
2084         return -1;
2085 }
2086
2087 /*! \brief Unload the PJSIP channel from Asterisk */
2088 static int unload_module(void)
2089 {
2090         ast_custom_function_unregister(&media_offer_function);
2091
2092         ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2093         ast_sip_session_unregister_supplement(&pbx_start_supplement);
2094
2095         ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
2096         ast_channel_unregister(&chan_pjsip_tech);
2097         ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
2098
2099         return 0;
2100 }
2101
2102 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Channel Driver",
2103                 .load = load_module,
2104                 .unload = unload_module,
2105                 .reload = reload,
2106                 .load_pri = AST_MODPRI_CHANNEL_DRIVER,
2107                );