pjsip: reinvite for connected line updates occurs when it should not
[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 = PJ_SUCCESS;
641         pjsip_tx_data *packet;
642         struct ast_sip_session *session = data;
643
644         pjsip_dlg_inc_lock(session->inv_session->dlg);
645         if (session->inv_session->invite_tsx) {
646                 status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet);
647         }
648         pjsip_dlg_dec_lock(session->inv_session->dlg);
649
650         if (status == PJ_SUCCESS && packet) {
651                 ast_sip_session_send_response(session, packet);
652         }
653
654         ao2_ref(session, -1);
655
656         return (status == PJ_SUCCESS) ? 0 : -1;
657 }
658
659 /*! \brief Function called by core when we should answer a PJSIP session */
660 static int chan_pjsip_answer(struct ast_channel *ast)
661 {
662         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
663
664         if (ast_channel_state(ast) == AST_STATE_UP) {
665                 return 0;
666         }
667
668         ast_setstate(ast, AST_STATE_UP);
669
670         ao2_ref(channel->session, +1);
671         if (ast_sip_push_task(channel->session->serializer, answer, channel->session)) {
672                 ast_log(LOG_WARNING, "Unable to push answer task to the threadpool. Cannot answer call\n");
673                 ao2_cleanup(channel->session);
674                 return -1;
675         }
676
677         return 0;
678 }
679
680 /*! \brief Internal helper function called when CNG tone is detected */
681 static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f)
682 {
683         const char *target_context;
684         int exists;
685
686         /* If we only needed this DSP for fax detection purposes we can just drop it now */
687         if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
688                 ast_dsp_set_features(session->dsp, DSP_FEATURE_DIGIT_DETECT);
689         } else {
690                 ast_dsp_free(session->dsp);
691                 session->dsp = NULL;
692         }
693
694         /* If already executing in the fax extension don't do anything */
695         if (!strcmp(ast_channel_exten(session->channel), "fax")) {
696                 return f;
697         }
698
699         target_context = S_OR(ast_channel_macrocontext(session->channel), ast_channel_context(session->channel));
700
701         /* We need to unlock the channel here because ast_exists_extension has the
702          * potential to start and stop an autoservice on the channel. Such action
703          * is prone to deadlock if the channel is locked.
704          */
705         ast_channel_unlock(session->channel);
706         exists = ast_exists_extension(session->channel, target_context, "fax", 1,
707                 S_COR(ast_channel_caller(session->channel)->id.number.valid,
708                         ast_channel_caller(session->channel)->id.number.str, NULL));
709         ast_channel_lock(session->channel);
710
711         if (exists) {
712                 ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
713                         ast_channel_name(session->channel));
714                 pbx_builtin_setvar_helper(session->channel, "FAXEXTEN", ast_channel_exten(session->channel));
715                 if (ast_async_goto(session->channel, target_context, "fax", 1)) {
716                         ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
717                                 ast_channel_name(session->channel), target_context);
718                 }
719                 ast_frfree(f);
720                 f = &ast_null_frame;
721         } else {
722                 ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
723                         ast_channel_name(session->channel), target_context);
724         }
725
726         return f;
727 }
728
729 /*! \brief Function called by core to read any waiting frames */
730 static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
731 {
732         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
733         struct chan_pjsip_pvt *pvt = channel->pvt;
734         struct ast_frame *f;
735         struct ast_sip_session_media *media = NULL;
736         int rtcp = 0;
737         int fdno = ast_channel_fdno(ast);
738
739         switch (fdno) {
740         case 0:
741                 media = pvt->media[SIP_MEDIA_AUDIO];
742                 break;
743         case 1:
744                 media = pvt->media[SIP_MEDIA_AUDIO];
745                 rtcp = 1;
746                 break;
747         case 2:
748                 media = pvt->media[SIP_MEDIA_VIDEO];
749                 break;
750         case 3:
751                 media = pvt->media[SIP_MEDIA_VIDEO];
752                 rtcp = 1;
753                 break;
754         }
755
756         if (!media || !media->rtp) {
757                 return &ast_null_frame;
758         }
759
760         if (!(f = ast_rtp_instance_read(media->rtp, rtcp))) {
761                 return f;
762         }
763
764         if (f->frametype != AST_FRAME_VOICE) {
765                 return f;
766         }
767
768         if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &f->subclass.format))) {
769                 ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(&f->subclass.format));
770                 ast_format_cap_set(ast_channel_nativeformats(ast), &f->subclass.format);
771                 ast_set_read_format(ast, ast_channel_readformat(ast));
772                 ast_set_write_format(ast, ast_channel_writeformat(ast));
773         }
774
775         if (channel->session->dsp) {
776                 f = ast_dsp_process(ast, channel->session->dsp, f);
777
778                 if (f && (f->frametype == AST_FRAME_DTMF)) {
779                         if (f->subclass.integer == 'f') {
780                                 ast_debug(3, "Fax CNG detected on %s\n", ast_channel_name(ast));
781                                 f = chan_pjsip_cng_tone_detected(channel->session, f);
782                         } else {
783                                 ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
784                                         ast_channel_name(ast));
785                         }
786                 }
787         }
788
789         return f;
790 }
791
792 /*! \brief Function called by core to write frames */
793 static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame)
794 {
795         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
796         struct chan_pjsip_pvt *pvt = channel->pvt;
797         struct ast_sip_session_media *media;
798         int res = 0;
799
800         switch (frame->frametype) {
801         case AST_FRAME_VOICE:
802                 media = pvt->media[SIP_MEDIA_AUDIO];
803
804                 if (!media) {
805                         return 0;
806                 }
807                 if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) {
808                         char buf[256];
809
810                         ast_log(LOG_WARNING,
811                                 "Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
812                                 ast_getformatname(&frame->subclass.format),
813                                 ast_getformatname_multiple(buf, sizeof(buf), ast_channel_nativeformats(ast)),
814                                 ast_getformatname(ast_channel_readformat(ast)),
815                                 ast_getformatname(ast_channel_writeformat(ast)));
816                         return 0;
817                 }
818                 if (media->rtp) {
819                         res = ast_rtp_instance_write(media->rtp, frame);
820                 }
821                 break;
822         case AST_FRAME_VIDEO:
823                 if ((media = pvt->media[SIP_MEDIA_VIDEO]) && media->rtp) {
824                         res = ast_rtp_instance_write(media->rtp, frame);
825                 }
826                 break;
827         case AST_FRAME_MODEM:
828                 break;
829         default:
830                 ast_log(LOG_WARNING, "Can't send %d type frames with PJSIP\n", frame->frametype);
831                 break;
832         }
833
834         return res;
835 }
836
837 struct fixup_data {
838         struct ast_sip_session *session;
839         struct ast_channel *chan;
840 };
841
842 static int fixup(void *data)
843 {
844         struct fixup_data *fix_data = data;
845         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(fix_data->chan);
846         struct chan_pjsip_pvt *pvt = channel->pvt;
847
848         channel->session->channel = fix_data->chan;
849         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
850                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(fix_data->chan));
851         }
852         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
853                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(fix_data->chan));
854         }
855
856         return 0;
857 }
858
859 /*! \brief Function called by core to change the underlying owner channel */
860 static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
861 {
862         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
863         struct fixup_data fix_data;
864
865         fix_data.session = channel->session;
866         fix_data.chan = newchan;
867
868         if (channel->session->channel != oldchan) {
869                 return -1;
870         }
871
872         if (ast_sip_push_task_synchronous(channel->session->serializer, fixup, &fix_data)) {
873                 ast_log(LOG_WARNING, "Unable to perform channel fixup\n");
874                 return -1;
875         }
876
877         return 0;
878 }
879
880 /*! \brief Function called to get the device state of an endpoint */
881 static int chan_pjsip_devicestate(const char *data)
882 {
883         RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
884         enum ast_device_state state = AST_DEVICE_UNKNOWN;
885         RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
886         RAII_VAR(struct stasis_cache *, cache, NULL, ao2_cleanup);
887         struct ast_devstate_aggregate aggregate;
888         int num, inuse = 0;
889
890         if (!endpoint) {
891                 return AST_DEVICE_INVALID;
892         }
893
894         endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent),
895                 ast_endpoint_get_resource(endpoint->persistent), 1);
896
897         if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) {
898                 state = AST_DEVICE_UNAVAILABLE;
899         } else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) {
900                 state = AST_DEVICE_NOT_INUSE;
901         }
902
903         if (!endpoint_snapshot->num_channels || !(cache = ast_channel_cache())) {
904                 return state;
905         }
906
907         ast_devstate_aggregate_init(&aggregate);
908
909         ao2_ref(cache, +1);
910
911         for (num = 0; num < endpoint_snapshot->num_channels; num++) {
912                 RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
913                 struct ast_channel_snapshot *snapshot;
914
915                 stasis_topic_wait(ast_channel_topic_all_cached());
916                 msg = stasis_cache_get(cache, ast_channel_snapshot_type(),
917                         endpoint_snapshot->channel_ids[num]);
918
919                 if (!msg) {
920                         continue;
921                 }
922
923                 snapshot = stasis_message_data(msg);
924
925                 if (snapshot->state == AST_STATE_DOWN) {
926                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_NOT_INUSE);
927                 } else if (snapshot->state == AST_STATE_RINGING) {
928                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_RINGING);
929                 } else if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
930                         (snapshot->state == AST_STATE_BUSY)) {
931                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_INUSE);
932                         inuse++;
933                 }
934         }
935
936         if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) {
937                 state = AST_DEVICE_BUSY;
938         } else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) {
939                 state = ast_devstate_aggregate_result(&aggregate);
940         }
941
942         return state;
943 }
944
945 /*! \brief Function called to query options on a channel */
946 static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen)
947 {
948         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
949         struct ast_sip_session *session = channel->session;
950         int res = -1;
951         enum ast_sip_session_t38state state = T38_STATE_UNAVAILABLE;
952
953         switch (option) {
954         case AST_OPTION_T38_STATE:
955                 if (session->endpoint->media.t38.enabled) {
956                         switch (session->t38state) {
957                         case T38_LOCAL_REINVITE:
958                         case T38_PEER_REINVITE:
959                                 state = T38_STATE_NEGOTIATING;
960                                 break;
961                         case T38_ENABLED:
962                                 state = T38_STATE_NEGOTIATED;
963                                 break;
964                         case T38_REJECTED:
965                                 state = T38_STATE_REJECTED;
966                                 break;
967                         default:
968                                 state = T38_STATE_UNKNOWN;
969                                 break;
970                         }
971                 }
972
973                 *((enum ast_t38_state *) data) = state;
974                 res = 0;
975
976                 break;
977         default:
978                 break;
979         }
980
981         return res;
982 }
983
984 struct indicate_data {
985         struct ast_sip_session *session;
986         int condition;
987         int response_code;
988         void *frame_data;
989         size_t datalen;
990 };
991
992 static void indicate_data_destroy(void *obj)
993 {
994         struct indicate_data *ind_data = obj;
995
996         ast_free(ind_data->frame_data);
997         ao2_ref(ind_data->session, -1);
998 }
999
1000 static struct indicate_data *indicate_data_alloc(struct ast_sip_session *session,
1001                 int condition, int response_code, const void *frame_data, size_t datalen)
1002 {
1003         struct indicate_data *ind_data = ao2_alloc(sizeof(*ind_data), indicate_data_destroy);
1004
1005         if (!ind_data) {
1006                 return NULL;
1007         }
1008
1009         ind_data->frame_data = ast_malloc(datalen);
1010         if (!ind_data->frame_data) {
1011                 ao2_ref(ind_data, -1);
1012                 return NULL;
1013         }
1014
1015         memcpy(ind_data->frame_data, frame_data, datalen);
1016         ind_data->datalen = datalen;
1017         ind_data->condition = condition;
1018         ind_data->response_code = response_code;
1019         ao2_ref(session, +1);
1020         ind_data->session = session;
1021
1022         return ind_data;
1023 }
1024
1025 static int indicate(void *data)
1026 {
1027         pjsip_tx_data *packet = NULL;
1028         struct indicate_data *ind_data = data;
1029         struct ast_sip_session *session = ind_data->session;
1030         int response_code = ind_data->response_code;
1031
1032         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
1033                 ast_sip_session_send_response(session, packet);
1034         }
1035
1036         ao2_ref(ind_data, -1);
1037
1038         return 0;
1039 }
1040
1041 /*! \brief Send SIP INFO with video update request */
1042 static int transmit_info_with_vidupdate(void *data)
1043 {
1044         const char * xml =
1045                 "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
1046                 " <media_control>\r\n"
1047                 "  <vc_primitive>\r\n"
1048                 "   <to_encoder>\r\n"
1049                 "    <picture_fast_update/>\r\n"
1050                 "   </to_encoder>\r\n"
1051                 "  </vc_primitive>\r\n"
1052                 " </media_control>\r\n";
1053
1054         const struct ast_sip_body body = {
1055                 .type = "application",
1056                 .subtype = "media_control+xml",
1057                 .body_text = xml
1058         };
1059
1060         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
1061         struct pjsip_tx_data *tdata;
1062
1063         if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
1064                 ast_log(LOG_ERROR, "Could not create text video update INFO request\n");
1065                 return -1;
1066         }
1067         if (ast_sip_add_body(tdata, &body)) {
1068                 ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n");
1069                 return -1;
1070         }
1071         ast_sip_session_send_request(session, tdata);
1072
1073         return 0;
1074 }
1075
1076 /*! \brief Update connected line information */
1077 static int update_connected_line_information(void *data)
1078 {
1079         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
1080         struct ast_party_id connected_id;
1081
1082         if ((ast_channel_state(session->channel) != AST_STATE_UP) && (session->inv_session->role == PJSIP_UAS_ROLE)) {
1083                 int response_code = 0;
1084
1085                 if (ast_channel_state(session->channel) == AST_STATE_RING) {
1086                         response_code = !session->endpoint->inband_progress ? 180 : 183;
1087                 } else if (ast_channel_state(session->channel) == AST_STATE_RINGING) {
1088                         response_code = 183;
1089                 }
1090
1091                 if (response_code) {
1092                         struct pjsip_tx_data *packet = NULL;
1093
1094                         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
1095                                 ast_sip_session_send_response(session, packet);
1096                         }
1097                 }
1098         } else {
1099                 enum ast_sip_session_refresh_method method = session->endpoint->id.refresh_method;
1100
1101                 if (session->inv_session->invite_tsx && (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
1102                         method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
1103                 }
1104
1105                 connected_id = ast_channel_connected_effective_id(session->channel);
1106                 if ((session->endpoint->id.send_pai || session->endpoint->id.send_rpid) &&
1107                     (session->endpoint->id.trust_outbound ||
1108                      ((connected_id.name.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED &&
1109                       (connected_id.number.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED))) {
1110                         ast_sip_session_refresh(session, NULL, NULL, NULL, method, 1);
1111                 }
1112         }
1113
1114         return 0;
1115 }
1116
1117 /*! \brief Function called by core to ask the channel to indicate some sort of condition */
1118 static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
1119 {
1120         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1121         struct chan_pjsip_pvt *pvt = channel->pvt;
1122         struct ast_sip_session_media *media;
1123         int response_code = 0;
1124         int res = 0;
1125
1126         switch (condition) {
1127         case AST_CONTROL_RINGING:
1128                 if (ast_channel_state(ast) == AST_STATE_RING) {
1129                         if (channel->session->endpoint->inband_progress) {
1130                                 response_code = 183;
1131                                 res = -1;
1132                         } else {
1133                                 response_code = 180;
1134                         }
1135                 } else {
1136                         res = -1;
1137                 }
1138                 ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
1139                 break;
1140         case AST_CONTROL_BUSY:
1141                 if (ast_channel_state(ast) != AST_STATE_UP) {
1142                         response_code = 486;
1143                 } else {
1144                         res = -1;
1145                 }
1146                 break;
1147         case AST_CONTROL_CONGESTION:
1148                 if (ast_channel_state(ast) != AST_STATE_UP) {
1149                         response_code = 503;
1150                 } else {
1151                         res = -1;
1152                 }
1153                 break;
1154         case AST_CONTROL_INCOMPLETE:
1155                 if (ast_channel_state(ast) != AST_STATE_UP) {
1156                         response_code = 484;
1157                 } else {
1158                         res = -1;
1159                 }
1160                 break;
1161         case AST_CONTROL_PROCEEDING:
1162                 if (ast_channel_state(ast) != AST_STATE_UP) {
1163                         response_code = 100;
1164                 } else {
1165                         res = -1;
1166                 }
1167                 break;
1168         case AST_CONTROL_PROGRESS:
1169                 if (ast_channel_state(ast) != AST_STATE_UP) {
1170                         response_code = 183;
1171                 } else {
1172                         res = -1;
1173                 }
1174                 break;
1175         case AST_CONTROL_VIDUPDATE:
1176                 media = pvt->media[SIP_MEDIA_VIDEO];
1177                 if (media && media->rtp) {
1178                         /* FIXME: Only use this for VP8. Additional work would have to be done to
1179                          * fully support other video codecs */
1180                         struct ast_format_cap *fcap = ast_channel_nativeformats(ast);
1181                         struct ast_format vp8;
1182                         ast_format_set(&vp8, AST_FORMAT_VP8, 0);
1183                         if (ast_format_cap_iscompatible(fcap, &vp8)) {
1184                                 /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
1185                                  * RTP engine would provide a way to externally write/schedule RTCP
1186                                  * packets */
1187                                 struct ast_frame fr;
1188                                 fr.frametype = AST_FRAME_CONTROL;
1189                                 fr.subclass.integer = AST_CONTROL_VIDUPDATE;
1190                                 res = ast_rtp_instance_write(media->rtp, &fr);
1191                         } else {
1192                                 ao2_ref(channel->session, +1);
1193
1194                                 if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
1195                                         ao2_cleanup(channel->session);
1196                                 }
1197                         }
1198                 } else {
1199                         res = -1;
1200                 }
1201                 break;
1202         case AST_CONTROL_CONNECTED_LINE:
1203                 ao2_ref(channel->session, +1);
1204                 if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
1205                         ao2_cleanup(channel->session);
1206                 }
1207                 break;
1208         case AST_CONTROL_UPDATE_RTP_PEER:
1209                 break;
1210         case AST_CONTROL_PVT_CAUSE_CODE:
1211                 res = -1;
1212                 break;
1213         case AST_CONTROL_HOLD:
1214                 ast_moh_start(ast, data, NULL);
1215                 break;
1216         case AST_CONTROL_UNHOLD:
1217                 ast_moh_stop(ast);
1218                 break;
1219         case AST_CONTROL_SRCUPDATE:
1220                 break;
1221         case AST_CONTROL_SRCCHANGE:
1222                 break;
1223         case AST_CONTROL_REDIRECTING:
1224                 if (ast_channel_state(ast) != AST_STATE_UP) {
1225                         response_code = 181;
1226                 } else {
1227                         res = -1;
1228                 }
1229                 break;
1230         case AST_CONTROL_T38_PARAMETERS:
1231                 res = 0;
1232
1233                 if (channel->session->t38state == T38_PEER_REINVITE) {
1234                         const struct ast_control_t38_parameters *parameters = data;
1235
1236                         if (parameters->request_response == AST_T38_REQUEST_PARMS) {
1237                                 res = AST_T38_REQUEST_PARMS;
1238                         }
1239                 }
1240
1241                 break;
1242         case -1:
1243                 res = -1;
1244                 break;
1245         default:
1246                 ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
1247                 res = -1;
1248                 break;
1249         }
1250
1251         if (response_code) {
1252                 struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
1253                 if (!ind_data || ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
1254                         ast_log(LOG_NOTICE, "Cannot send response code %d to endpoint %s. Could not queue task properly\n",
1255                                         response_code, ast_sorcery_object_get_id(channel->session->endpoint));
1256                         ao2_cleanup(ind_data);
1257                         res = -1;
1258                 }
1259         }
1260
1261         return res;
1262 }
1263
1264 struct transfer_data {
1265         struct ast_sip_session *session;
1266         char *target;
1267 };
1268
1269 static void transfer_data_destroy(void *obj)
1270 {
1271         struct transfer_data *trnf_data = obj;
1272
1273         ast_free(trnf_data->target);
1274         ao2_cleanup(trnf_data->session);
1275 }
1276
1277 static struct transfer_data *transfer_data_alloc(struct ast_sip_session *session, const char *target)
1278 {
1279         struct transfer_data *trnf_data = ao2_alloc(sizeof(*trnf_data), transfer_data_destroy);
1280
1281         if (!trnf_data) {
1282                 return NULL;
1283         }
1284
1285         if (!(trnf_data->target = ast_strdup(target))) {
1286                 ao2_ref(trnf_data, -1);
1287                 return NULL;
1288         }
1289
1290         ao2_ref(session, +1);
1291         trnf_data->session = session;
1292
1293         return trnf_data;
1294 }
1295
1296 static void transfer_redirect(struct ast_sip_session *session, const char *target)
1297 {
1298         pjsip_tx_data *packet;
1299         enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1300         pjsip_contact_hdr *contact;
1301         pj_str_t tmp;
1302
1303         if (pjsip_inv_end_session(session->inv_session, 302, NULL, &packet) != PJ_SUCCESS) {
1304                 message = AST_TRANSFER_FAILED;
1305                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1306
1307                 return;
1308         }
1309
1310         if (!(contact = pjsip_msg_find_hdr(packet->msg, PJSIP_H_CONTACT, NULL))) {
1311                 contact = pjsip_contact_hdr_create(packet->pool);
1312         }
1313
1314         pj_strdup2_with_null(packet->pool, &tmp, target);
1315         if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) {
1316                 message = AST_TRANSFER_FAILED;
1317                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1318                 pjsip_tx_data_dec_ref(packet);
1319
1320                 return;
1321         }
1322         pjsip_msg_add_hdr(packet->msg, (pjsip_hdr *) contact);
1323
1324         ast_sip_session_send_response(session, packet);
1325         ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1326 }
1327
1328 static void transfer_refer(struct ast_sip_session *session, const char *target)
1329 {
1330         pjsip_evsub *sub;
1331         enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1332         pj_str_t tmp;
1333         pjsip_tx_data *packet;
1334
1335         if (pjsip_xfer_create_uac(session->inv_session->dlg, NULL, &sub) != PJ_SUCCESS) {
1336                 message = AST_TRANSFER_FAILED;
1337                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1338
1339                 return;
1340         }
1341
1342         if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
1343                 message = AST_TRANSFER_FAILED;
1344                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1345                 pjsip_evsub_terminate(sub, PJ_FALSE);
1346
1347                 return;
1348         }
1349
1350         pjsip_xfer_send_request(sub, packet);
1351         ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1352 }
1353
1354 static int transfer(void *data)
1355 {
1356         struct transfer_data *trnf_data = data;
1357
1358         if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
1359                 transfer_redirect(trnf_data->session, trnf_data->target);
1360         } else {
1361                 transfer_refer(trnf_data->session, trnf_data->target);
1362         }
1363
1364         ao2_ref(trnf_data, -1);
1365         return 0;
1366 }
1367
1368 /*! \brief Function called by core for Asterisk initiated transfer */
1369 static int chan_pjsip_transfer(struct ast_channel *chan, const char *target)
1370 {
1371         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
1372         struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
1373
1374         if (!trnf_data) {
1375                 return -1;
1376         }
1377
1378         if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
1379                 ast_log(LOG_WARNING, "Error requesting transfer\n");
1380                 ao2_cleanup(trnf_data);
1381                 return -1;
1382         }
1383
1384         return 0;
1385 }
1386
1387 /*! \brief Function called by core to start a DTMF digit */
1388 static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
1389 {
1390         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
1391         struct chan_pjsip_pvt *pvt = channel->pvt;
1392         struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
1393         int res = 0;
1394
1395         switch (channel->session->endpoint->dtmf) {
1396         case AST_SIP_DTMF_RFC_4733:
1397                 if (!media || !media->rtp) {
1398                         return -1;
1399                 }
1400
1401                 ast_rtp_instance_dtmf_begin(media->rtp, digit);
1402         case AST_SIP_DTMF_NONE:
1403                 break;
1404         case AST_SIP_DTMF_INBAND:
1405                 res = -1;
1406                 break;
1407         default:
1408                 break;
1409         }
1410
1411         return res;
1412 }
1413
1414 struct info_dtmf_data {
1415         struct ast_sip_session *session;
1416         char digit;
1417         unsigned int duration;
1418 };
1419
1420 static void info_dtmf_data_destroy(void *obj)
1421 {
1422         struct info_dtmf_data *dtmf_data = obj;
1423         ao2_ref(dtmf_data->session, -1);
1424 }
1425
1426 static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
1427 {
1428         struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy);
1429         if (!dtmf_data) {
1430                 return NULL;
1431         }
1432         ao2_ref(session, +1);
1433         dtmf_data->session = session;
1434         dtmf_data->digit = digit;
1435         dtmf_data->duration = duration;
1436         return dtmf_data;
1437 }
1438
1439 static int transmit_info_dtmf(void *data)
1440 {
1441         RAII_VAR(struct info_dtmf_data *, dtmf_data, data, ao2_cleanup);
1442
1443         struct ast_sip_session *session = dtmf_data->session;
1444         struct pjsip_tx_data *tdata;
1445
1446         RAII_VAR(struct ast_str *, body_text, NULL, ast_free_ptr);
1447
1448         struct ast_sip_body body = {
1449                 .type = "application",
1450                 .subtype = "dtmf-relay",
1451         };
1452
1453         if (!(body_text = ast_str_create(32))) {
1454                 ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n");
1455                 return -1;
1456         }
1457         ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration);
1458
1459         body.body_text = ast_str_buffer(body_text);
1460
1461         if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
1462                 ast_log(LOG_ERROR, "Could not create DTMF INFO request\n");
1463                 return -1;
1464         }
1465         if (ast_sip_add_body(tdata, &body)) {
1466                 ast_log(LOG_ERROR, "Could not add body to DTMF INFO request\n");
1467                 pjsip_tx_data_dec_ref(tdata);
1468                 return -1;
1469         }
1470         ast_sip_session_send_request(session, tdata);
1471
1472         return 0;
1473 }
1474
1475 /*! \brief Function called by core to stop a DTMF digit */
1476 static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
1477 {
1478         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1479         struct chan_pjsip_pvt *pvt = channel->pvt;
1480         struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
1481         int res = 0;
1482
1483         switch (channel->session->endpoint->dtmf) {
1484         case AST_SIP_DTMF_INFO:
1485         {
1486                 struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
1487
1488                 if (!dtmf_data) {
1489                         return -1;
1490                 }
1491
1492                 if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
1493                         ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
1494                         ao2_cleanup(dtmf_data);
1495                         return -1;
1496                 }
1497                 break;
1498         }
1499         case AST_SIP_DTMF_RFC_4733:
1500                 if (!media || !media->rtp) {
1501                         return -1;
1502                 }
1503
1504                 ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
1505         case AST_SIP_DTMF_NONE:
1506                 break;
1507         case AST_SIP_DTMF_INBAND:
1508                 res = -1;
1509                 break;
1510         }
1511
1512         return res;
1513 }
1514
1515 static int call(void *data)
1516 {
1517         struct ast_sip_session *session = data;
1518         pjsip_tx_data *tdata;
1519
1520         int res = ast_sip_session_create_invite(session, &tdata);
1521
1522         if (res) {
1523                 ast_queue_hangup(session->channel);
1524         } else {
1525                 ast_sip_session_send_request(session, tdata);
1526         }
1527         ao2_ref(session, -1);
1528         return res;
1529 }
1530
1531 /*! \brief Function called by core to actually start calling a remote party */
1532 static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout)
1533 {
1534         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1535
1536         ao2_ref(channel->session, +1);
1537         if (ast_sip_push_task(channel->session->serializer, call, channel->session)) {
1538                 ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
1539                 ao2_cleanup(channel->session);
1540                 return -1;
1541         }
1542
1543         return 0;
1544 }
1545
1546 /*! \brief Internal function which translates from Asterisk cause codes to SIP response codes */
1547 static int hangup_cause2sip(int cause)
1548 {
1549         switch (cause) {
1550         case AST_CAUSE_UNALLOCATED:             /* 1 */
1551         case AST_CAUSE_NO_ROUTE_DESTINATION:    /* 3 IAX2: Can't find extension in context */
1552         case AST_CAUSE_NO_ROUTE_TRANSIT_NET:    /* 2 */
1553                 return 404;
1554         case AST_CAUSE_CONGESTION:              /* 34 */
1555         case AST_CAUSE_SWITCH_CONGESTION:       /* 42 */
1556                 return 503;
1557         case AST_CAUSE_NO_USER_RESPONSE:        /* 18 */
1558                 return 408;
1559         case AST_CAUSE_NO_ANSWER:               /* 19 */
1560         case AST_CAUSE_UNREGISTERED:        /* 20 */
1561                 return 480;
1562         case AST_CAUSE_CALL_REJECTED:           /* 21 */
1563                 return 403;
1564         case AST_CAUSE_NUMBER_CHANGED:          /* 22 */
1565                 return 410;
1566         case AST_CAUSE_NORMAL_UNSPECIFIED:      /* 31 */
1567                 return 480;
1568         case AST_CAUSE_INVALID_NUMBER_FORMAT:
1569                 return 484;
1570         case AST_CAUSE_USER_BUSY:
1571                 return 486;
1572         case AST_CAUSE_FAILURE:
1573                 return 500;
1574         case AST_CAUSE_FACILITY_REJECTED:       /* 29 */
1575                 return 501;
1576         case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
1577                 return 503;
1578         case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
1579                 return 502;
1580         case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:       /* Can't find codec to connect to host */
1581                 return 488;
1582         case AST_CAUSE_INTERWORKING:    /* Unspecified Interworking issues */
1583                 return 500;
1584         case AST_CAUSE_NOTDEFINED:
1585         default:
1586                 ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
1587                 return 0;
1588         }
1589
1590         /* Never reached */
1591         return 0;
1592 }
1593
1594 struct hangup_data {
1595         int cause;
1596         struct ast_channel *chan;
1597 };
1598
1599 static void hangup_data_destroy(void *obj)
1600 {
1601         struct hangup_data *h_data = obj;
1602
1603         h_data->chan = ast_channel_unref(h_data->chan);
1604 }
1605
1606 static struct hangup_data *hangup_data_alloc(int cause, struct ast_channel *chan)
1607 {
1608         struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
1609
1610         if (!h_data) {
1611                 return NULL;
1612         }
1613
1614         h_data->cause = cause;
1615         h_data->chan = ast_channel_ref(chan);
1616
1617         return h_data;
1618 }
1619
1620 /*! \brief Clear a channel from a session along with its PVT */
1621 static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast, struct chan_pjsip_pvt *pvt)
1622 {
1623         session->channel = NULL;
1624         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
1625                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, "");
1626         }
1627         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
1628                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, "");
1629         }
1630         ast_channel_tech_pvt_set(ast, NULL);
1631 }
1632
1633 static int hangup(void *data)
1634 {
1635         pj_status_t status;
1636         pjsip_tx_data *packet = NULL;
1637         struct hangup_data *h_data = data;
1638         struct ast_channel *ast = h_data->chan;
1639         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1640         struct chan_pjsip_pvt *pvt = channel->pvt;
1641         struct ast_sip_session *session = channel->session;
1642         int cause = h_data->cause;
1643
1644         if (!session->defer_terminate &&
1645                 ((status = pjsip_inv_end_session(session->inv_session, cause ? cause : 603, NULL, &packet)) == PJ_SUCCESS) && packet) {
1646                 if (packet->msg->type == PJSIP_RESPONSE_MSG) {
1647                         ast_sip_session_send_response(session, packet);
1648                 } else {
1649                         ast_sip_session_send_request(session, packet);
1650                 }
1651         }
1652
1653         clear_session_and_channel(session, ast, pvt);
1654         ao2_cleanup(channel);
1655         ao2_cleanup(h_data);
1656
1657         return 0;
1658 }
1659
1660 /*! \brief Function called by core to hang up a PJSIP session */
1661 static int chan_pjsip_hangup(struct ast_channel *ast)
1662 {
1663         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1664         struct chan_pjsip_pvt *pvt = channel->pvt;
1665         int cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
1666         struct hangup_data *h_data = hangup_data_alloc(cause, ast);
1667
1668         if (!h_data) {
1669                 goto failure;
1670         }
1671
1672         if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
1673                 ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
1674                 goto failure;
1675         }
1676
1677         return 0;
1678
1679 failure:
1680         /* Go ahead and do our cleanup of the session and channel even if we're not going
1681          * to be able to send our SIP request/response
1682          */
1683         clear_session_and_channel(channel->session, ast, pvt);
1684         ao2_cleanup(channel);
1685         ao2_cleanup(h_data);
1686
1687         return -1;
1688 }
1689
1690 struct request_data {
1691         struct ast_sip_session *session;
1692         struct ast_format_cap *caps;
1693         const char *dest;
1694         int cause;
1695 };
1696
1697 static int request(void *obj)
1698 {
1699         struct request_data *req_data = obj;
1700         struct ast_sip_session *session = NULL;
1701         char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
1702         RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
1703
1704         AST_DECLARE_APP_ARGS(args,
1705                 AST_APP_ARG(endpoint);
1706                 AST_APP_ARG(aor);
1707         );
1708
1709         if (ast_strlen_zero(tmp)) {
1710                 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
1711                 req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
1712                 return -1;
1713         }
1714
1715         AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
1716
1717         /* If a request user has been specified extract it from the endpoint name portion */
1718         if ((endpoint_name = strchr(args.endpoint, '@'))) {
1719                 request_user = args.endpoint;
1720                 *endpoint_name++ = '\0';
1721         } else {
1722                 endpoint_name = args.endpoint;
1723         }
1724
1725         if (ast_strlen_zero(endpoint_name)) {
1726                 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
1727                 req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
1728         } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name))) {
1729                 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
1730                 req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
1731                 return -1;
1732         }
1733
1734         if (!(session = ast_sip_session_create_outgoing(endpoint, args.aor, request_user, req_data->caps))) {
1735                 req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
1736                 return -1;
1737         }
1738
1739         req_data->session = session;
1740
1741         return 0;
1742 }
1743
1744 /*! \brief Function called by core to create a new outgoing PJSIP session */
1745 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)
1746 {
1747         struct request_data req_data;
1748         RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
1749
1750         req_data.caps = cap;
1751         req_data.dest = data;
1752
1753         if (ast_sip_push_task_synchronous(NULL, request, &req_data)) {
1754                 *cause = req_data.cause;
1755                 return NULL;
1756         }
1757
1758         session = req_data.session;
1759
1760         if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, requestor ? ast_channel_linkedid(requestor) : NULL, NULL))) {
1761                 /* Session needs to be terminated prematurely */
1762                 return NULL;
1763         }
1764
1765         return session->channel;
1766 }
1767
1768 struct sendtext_data {
1769         struct ast_sip_session *session;
1770         char text[0];
1771 };
1772
1773 static void sendtext_data_destroy(void *obj)
1774 {
1775         struct sendtext_data *data = obj;
1776         ao2_ref(data->session, -1);
1777 }
1778
1779 static struct sendtext_data* sendtext_data_create(struct ast_sip_session *session, const char *text)
1780 {
1781         int size = strlen(text) + 1;
1782         struct sendtext_data *data = ao2_alloc(sizeof(*data)+size, sendtext_data_destroy);
1783
1784         if (!data) {
1785                 return NULL;
1786         }
1787
1788         data->session = session;
1789         ao2_ref(data->session, +1);
1790         ast_copy_string(data->text, text, size);
1791         return data;
1792 }
1793
1794 static int sendtext(void *obj)
1795 {
1796         RAII_VAR(struct sendtext_data *, data, obj, ao2_cleanup);
1797         pjsip_tx_data *tdata;
1798
1799         const struct ast_sip_body body = {
1800                 .type = "text",
1801                 .subtype = "plain",
1802                 .body_text = data->text
1803         };
1804
1805         /* NOT ast_strlen_zero, because a zero-length message is specifically
1806          * allowed by RFC 3428 (See section 10, Examples) */
1807         if (!data->text) {
1808                 return 0;
1809         }
1810
1811         ast_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, &tdata);
1812         ast_sip_add_body(tdata, &body);
1813         ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint);
1814
1815         return 0;
1816 }
1817
1818 /*! \brief Function called by core to send text on PJSIP session */
1819 static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text)
1820 {
1821         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1822         struct sendtext_data *data = sendtext_data_create(channel->session, text);
1823
1824         if (!data || ast_sip_push_task(channel->session->serializer, sendtext, data)) {
1825                 ao2_ref(data, -1);
1826                 return -1;
1827         }
1828         return 0;
1829 }
1830
1831 /*! \brief Convert SIP hangup causes to Asterisk hangup causes */
1832 static int hangup_sip2cause(int cause)
1833 {
1834         /* Possible values taken from causes.h */
1835
1836         switch(cause) {
1837         case 401:       /* Unauthorized */
1838                 return AST_CAUSE_CALL_REJECTED;
1839         case 403:       /* Not found */
1840                 return AST_CAUSE_CALL_REJECTED;
1841         case 404:       /* Not found */
1842                 return AST_CAUSE_UNALLOCATED;
1843         case 405:       /* Method not allowed */
1844                 return AST_CAUSE_INTERWORKING;
1845         case 407:       /* Proxy authentication required */
1846                 return AST_CAUSE_CALL_REJECTED;
1847         case 408:       /* No reaction */
1848                 return AST_CAUSE_NO_USER_RESPONSE;
1849         case 409:       /* Conflict */
1850                 return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
1851         case 410:       /* Gone */
1852                 return AST_CAUSE_NUMBER_CHANGED;
1853         case 411:       /* Length required */
1854                 return AST_CAUSE_INTERWORKING;
1855         case 413:       /* Request entity too large */
1856                 return AST_CAUSE_INTERWORKING;
1857         case 414:       /* Request URI too large */
1858                 return AST_CAUSE_INTERWORKING;
1859         case 415:       /* Unsupported media type */
1860                 return AST_CAUSE_INTERWORKING;
1861         case 420:       /* Bad extension */
1862                 return AST_CAUSE_NO_ROUTE_DESTINATION;
1863         case 480:       /* No answer */
1864                 return AST_CAUSE_NO_ANSWER;
1865         case 481:       /* No answer */
1866                 return AST_CAUSE_INTERWORKING;
1867         case 482:       /* Loop detected */
1868                 return AST_CAUSE_INTERWORKING;
1869         case 483:       /* Too many hops */
1870                 return AST_CAUSE_NO_ANSWER;
1871         case 484:       /* Address incomplete */
1872                 return AST_CAUSE_INVALID_NUMBER_FORMAT;
1873         case 485:       /* Ambiguous */
1874                 return AST_CAUSE_UNALLOCATED;
1875         case 486:       /* Busy everywhere */
1876                 return AST_CAUSE_BUSY;
1877         case 487:       /* Request terminated */
1878                 return AST_CAUSE_INTERWORKING;
1879         case 488:       /* No codecs approved */
1880                 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
1881         case 491:       /* Request pending */
1882                 return AST_CAUSE_INTERWORKING;
1883         case 493:       /* Undecipherable */
1884                 return AST_CAUSE_INTERWORKING;
1885         case 500:       /* Server internal failure */
1886                 return AST_CAUSE_FAILURE;
1887         case 501:       /* Call rejected */
1888                 return AST_CAUSE_FACILITY_REJECTED;
1889         case 502:
1890                 return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
1891         case 503:       /* Service unavailable */
1892                 return AST_CAUSE_CONGESTION;
1893         case 504:       /* Gateway timeout */
1894                 return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
1895         case 505:       /* SIP version not supported */
1896                 return AST_CAUSE_INTERWORKING;
1897         case 600:       /* Busy everywhere */
1898                 return AST_CAUSE_USER_BUSY;
1899         case 603:       /* Decline */
1900                 return AST_CAUSE_CALL_REJECTED;
1901         case 604:       /* Does not exist anywhere */
1902                 return AST_CAUSE_UNALLOCATED;
1903         case 606:       /* Not acceptable */
1904                 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
1905         default:
1906                 if (cause < 500 && cause >= 400) {
1907                         /* 4xx class error that is unknown - someting wrong with our request */
1908                         return AST_CAUSE_INTERWORKING;
1909                 } else if (cause < 600 && cause >= 500) {
1910                         /* 5xx class error - problem in the remote end */
1911                         return AST_CAUSE_CONGESTION;
1912                 } else if (cause < 700 && cause >= 600) {
1913                         /* 6xx - global errors in the 4xx class */
1914                         return AST_CAUSE_INTERWORKING;
1915                 }
1916                 return AST_CAUSE_NORMAL;
1917         }
1918         /* Never reached */
1919         return 0;
1920 }
1921
1922 static void chan_pjsip_session_begin(struct ast_sip_session *session)
1923 {
1924         RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
1925
1926         if (session->endpoint->media.direct_media.glare_mitigation ==
1927                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
1928                 return;
1929         }
1930
1931         datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info,
1932                         "direct_media_glare_mitigation");
1933
1934         if (!datastore) {
1935                 return;
1936         }
1937
1938         ast_sip_session_add_datastore(session, datastore);
1939 }
1940
1941 /*! \brief Function called when the session ends */
1942 static void chan_pjsip_session_end(struct ast_sip_session *session)
1943 {
1944         if (!session->channel) {
1945                 return;
1946         }
1947
1948         if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
1949                 int cause = hangup_sip2cause(session->inv_session->cause);
1950
1951                 ast_queue_hangup_with_cause(session->channel, cause);
1952         } else {
1953                 ast_queue_hangup(session->channel);
1954         }
1955 }
1956
1957 /*! \brief Function called when a request is received on the session */
1958 static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
1959 {
1960         pjsip_tx_data *packet = NULL;
1961
1962         if (session->channel) {
1963                 return 0;
1964         }
1965
1966         if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL))) {
1967                 if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
1968                         ast_sip_session_send_response(session, packet);
1969                 }
1970
1971                 ast_log(LOG_ERROR, "Failed to allocate new PJSIP channel on incoming SIP INVITE\n");
1972                 return -1;
1973         }
1974         /* channel gets created on incoming request, but we wait to call start
1975            so other supplements have a chance to run */
1976         return 0;
1977 }
1978
1979 static int pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
1980 {
1981         int res;
1982
1983         res = ast_pbx_start(session->channel);
1984
1985         switch (res) {
1986         case AST_PBX_FAILED:
1987                 ast_log(LOG_WARNING, "Failed to start PBX ;(\n");
1988                 ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
1989                 ast_hangup(session->channel);
1990                 break;
1991         case AST_PBX_CALL_LIMIT:
1992                 ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
1993                 ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
1994                 ast_hangup(session->channel);
1995                 break;
1996         case AST_PBX_SUCCESS:
1997         default:
1998                 break;
1999         }
2000
2001         ast_debug(3, "Started PBX on new PJSIP channel %s\n", ast_channel_name(session->channel));
2002
2003         return (res == AST_PBX_SUCCESS) ? 0 : -1;
2004 }
2005
2006 static struct ast_sip_session_supplement pbx_start_supplement = {
2007         .method = "INVITE",
2008         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_LAST,
2009         .incoming_request = pbx_start_incoming_request,
2010 };
2011
2012 /*! \brief Function called when a response is received on the session */
2013 static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
2014 {
2015         struct pjsip_status_line status = rdata->msg_info.msg->line.status;
2016
2017         if (!session->channel) {
2018                 return;
2019         }
2020
2021         switch (status.code) {
2022         case 180:
2023                 ast_queue_control(session->channel, AST_CONTROL_RINGING);
2024                 if (ast_channel_state(session->channel) != AST_STATE_UP) {
2025                         ast_setstate(session->channel, AST_STATE_RINGING);
2026                 }
2027                 break;
2028         case 183:
2029                 ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
2030                 break;
2031         case 200:
2032                 ast_queue_control(session->channel, AST_CONTROL_ANSWER);
2033                 break;
2034         default:
2035                 break;
2036         }
2037 }
2038
2039 static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
2040 {
2041         if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
2042                 if (session->endpoint->media.direct_media.enabled) {
2043                         ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
2044                 }
2045         }
2046         return 0;
2047 }
2048
2049 /*!
2050  * \brief Load the module
2051  *
2052  * Module loading including tests for configuration or dependencies.
2053  * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
2054  * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
2055  * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
2056  * configuration file or other non-critical problem return
2057  * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
2058  */
2059 static int load_module(void)
2060 {
2061         if (!(chan_pjsip_tech.capabilities = ast_format_cap_alloc())) {
2062                 return AST_MODULE_LOAD_DECLINE;
2063         }
2064
2065         ast_format_cap_add_all_by_type(chan_pjsip_tech.capabilities, AST_FORMAT_TYPE_AUDIO);
2066
2067         ast_rtp_glue_register(&chan_pjsip_rtp_glue);
2068
2069         if (ast_channel_register(&chan_pjsip_tech)) {
2070                 ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
2071                 goto end;
2072         }
2073
2074         if (ast_custom_function_register(&chan_pjsip_dial_contacts_function)) {
2075                 ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n");
2076                 goto end;
2077         }
2078
2079         if (ast_custom_function_register(&media_offer_function)) {
2080                 ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
2081         }
2082
2083         if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
2084                 ast_log(LOG_ERROR, "Unable to register PJSIP supplement\n");
2085                 goto end;
2086         }
2087
2088         if (ast_sip_session_register_supplement(&pbx_start_supplement)) {
2089                 ast_log(LOG_ERROR, "Unable to register PJSIP pbx start supplement\n");
2090                 ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2091                 goto end;
2092         }
2093
2094         if (ast_sip_session_register_supplement(&chan_pjsip_ack_supplement)) {
2095                 ast_log(LOG_ERROR, "Unable to register PJSIP ACK supplement\n");
2096                 ast_sip_session_unregister_supplement(&pbx_start_supplement);
2097                 ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2098                 goto end;
2099         }
2100
2101         return 0;
2102
2103 end:
2104         ast_custom_function_unregister(&media_offer_function);
2105         ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
2106         ast_channel_unregister(&chan_pjsip_tech);
2107         ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
2108
2109         return AST_MODULE_LOAD_FAILURE;
2110 }
2111
2112 /*! \brief Reload module */
2113 static int reload(void)
2114 {
2115         return -1;
2116 }
2117
2118 /*! \brief Unload the PJSIP channel from Asterisk */
2119 static int unload_module(void)
2120 {
2121         ast_custom_function_unregister(&media_offer_function);
2122
2123         ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2124         ast_sip_session_unregister_supplement(&pbx_start_supplement);
2125
2126         ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
2127         ast_channel_unregister(&chan_pjsip_tech);
2128         ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
2129
2130         return 0;
2131 }
2132
2133 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Channel Driver",
2134                 .load = load_module,
2135                 .unload = unload_module,
2136                 .reload = reload,
2137                 .load_pri = AST_MODPRI_CHANNEL_DRIVER,
2138                );