8db3f094fce1596114b1edcbb345802ed0909ddb
[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 #include "pjsip/include/chan_pjsip.h"
65 #include "pjsip/include/dialplan_functions.h"
66
67 static const char desc[] = "PJSIP Channel";
68 static const char channel_type[] = "PJSIP";
69
70 static unsigned int chan_idx;
71
72 static void chan_pjsip_pvt_dtor(void *obj)
73 {
74         struct chan_pjsip_pvt *pvt = obj;
75         int i;
76
77         for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
78                 ao2_cleanup(pvt->media[i]);
79                 pvt->media[i] = NULL;
80         }
81 }
82
83 /* \brief Asterisk core interaction functions */
84 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);
85 static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text);
86 static int chan_pjsip_digit_begin(struct ast_channel *ast, char digit);
87 static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
88 static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout);
89 static int chan_pjsip_hangup(struct ast_channel *ast);
90 static int chan_pjsip_answer(struct ast_channel *ast);
91 static struct ast_frame *chan_pjsip_read(struct ast_channel *ast);
92 static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *f);
93 static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
94 static int chan_pjsip_transfer(struct ast_channel *ast, const char *target);
95 static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
96 static int chan_pjsip_devicestate(const char *data);
97 static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
98
99 /*! \brief PBX interface structure for channel registration */
100 struct ast_channel_tech chan_pjsip_tech = {
101         .type = channel_type,
102         .description = "PJSIP Channel Driver",
103         .requester = chan_pjsip_request,
104         .send_text = chan_pjsip_sendtext,
105         .send_digit_begin = chan_pjsip_digit_begin,
106         .send_digit_end = chan_pjsip_digit_end,
107         .call = chan_pjsip_call,
108         .hangup = chan_pjsip_hangup,
109         .answer = chan_pjsip_answer,
110         .read = chan_pjsip_read,
111         .write = chan_pjsip_write,
112         .write_video = chan_pjsip_write,
113         .exception = chan_pjsip_read,
114         .indicate = chan_pjsip_indicate,
115         .transfer = chan_pjsip_transfer,
116         .fixup = chan_pjsip_fixup,
117         .devicestate = chan_pjsip_devicestate,
118         .queryoption = chan_pjsip_queryoption,
119         .func_channel_read = pjsip_acf_channel_read,
120         .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
121 };
122
123 /*! \brief SIP session interaction functions */
124 static void chan_pjsip_session_begin(struct ast_sip_session *session);
125 static void chan_pjsip_session_end(struct ast_sip_session *session);
126 static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
127 static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
128
129 /*! \brief SIP session supplement structure */
130 static struct ast_sip_session_supplement chan_pjsip_supplement = {
131         .method = "INVITE",
132         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
133         .session_begin = chan_pjsip_session_begin,
134         .session_end = chan_pjsip_session_end,
135         .incoming_request = chan_pjsip_incoming_request,
136         .incoming_response = chan_pjsip_incoming_response,
137 };
138
139 static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
140
141 static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
142         .method = "ACK",
143         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
144         .incoming_request = chan_pjsip_incoming_ack,
145 };
146
147 /*! \brief Function called by RTP engine to get local audio RTP peer */
148 static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
149 {
150         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
151         struct chan_pjsip_pvt *pvt = channel->pvt;
152         struct ast_sip_endpoint *endpoint;
153
154         if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
155                 return AST_RTP_GLUE_RESULT_FORBID;
156         }
157
158         endpoint = channel->session->endpoint;
159
160         *instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
161         ao2_ref(*instance, +1);
162
163         ast_assert(endpoint != NULL);
164         if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
165                 return AST_RTP_GLUE_RESULT_FORBID;
166         }
167
168         if (endpoint->media.direct_media.enabled) {
169                 return AST_RTP_GLUE_RESULT_REMOTE;
170         }
171
172         return AST_RTP_GLUE_RESULT_LOCAL;
173 }
174
175 /*! \brief Function called by RTP engine to get local video RTP peer */
176 static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
177 {
178         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
179         struct chan_pjsip_pvt *pvt = channel->pvt;
180         struct ast_sip_endpoint *endpoint;
181
182         if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
183                 return AST_RTP_GLUE_RESULT_FORBID;
184         }
185
186         endpoint = channel->session->endpoint;
187
188         *instance = pvt->media[SIP_MEDIA_VIDEO]->rtp;
189         ao2_ref(*instance, +1);
190
191         ast_assert(endpoint != NULL);
192         if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
193                 return AST_RTP_GLUE_RESULT_FORBID;
194         }
195
196         return AST_RTP_GLUE_RESULT_LOCAL;
197 }
198
199 /*! \brief Function called by RTP engine to get peer capabilities */
200 static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
201 {
202         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
203
204         ast_format_cap_copy(result, channel->session->endpoint->media.codecs);
205 }
206
207 static int send_direct_media_request(void *data)
208 {
209         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
210
211         return ast_sip_session_refresh(session, NULL, NULL, NULL,
212                         session->endpoint->media.direct_media.method, 1);
213 }
214
215 /*! \brief Destructor function for \ref transport_info_data */
216 static void transport_info_destroy(void *obj)
217 {
218         struct transport_info_data *data = obj;
219         ast_free(data);
220 }
221
222 /*! \brief Datastore used to store local/remote addresses for the
223  * INVITE request that created the PJSIP channel */
224 static struct ast_datastore_info transport_info = {
225         .type = "chan_pjsip_transport_info",
226         .destroy = transport_info_destroy,
227 };
228
229 static struct ast_datastore_info direct_media_mitigation_info = { };
230
231 static int direct_media_mitigate_glare(struct ast_sip_session *session)
232 {
233         RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
234
235         if (session->endpoint->media.direct_media.glare_mitigation ==
236                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
237                 return 0;
238         }
239
240         datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation");
241         if (!datastore) {
242                 return 0;
243         }
244
245         /* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */
246         ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation");
247
248         if ((session->endpoint->media.direct_media.glare_mitigation ==
249                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING &&
250                         session->inv_session->role == PJSIP_ROLE_UAC) ||
251                         (session->endpoint->media.direct_media.glare_mitigation ==
252                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING &&
253                         session->inv_session->role == PJSIP_ROLE_UAS)) {
254                 return 1;
255         }
256
257         return 0;
258 }
259
260 static int check_for_rtp_changes(struct ast_channel *chan, struct ast_rtp_instance *rtp,
261                 struct ast_sip_session_media *media, int rtcp_fd)
262 {
263         int changed = 0;
264
265         if (rtp) {
266                 changed = ast_rtp_instance_get_and_cmp_remote_address(rtp, &media->direct_media_addr);
267                 if (media->rtp) {
268                         ast_channel_set_fd(chan, rtcp_fd, -1);
269                         ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 0);
270                 }
271         } else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
272                 ast_sockaddr_setnull(&media->direct_media_addr);
273                 changed = 1;
274                 if (media->rtp) {
275                         ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1);
276                         ast_channel_set_fd(chan, rtcp_fd, ast_rtp_instance_fd(media->rtp, 1));
277                 }
278         }
279
280         return changed;
281 }
282
283 /*! \brief Function called by RTP engine to change where the remote party should send media */
284 static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
285                 struct ast_rtp_instance *rtp,
286                 struct ast_rtp_instance *vrtp,
287                 struct ast_rtp_instance *tpeer,
288                 const struct ast_format_cap *cap,
289                 int nat_active)
290 {
291         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
292         struct chan_pjsip_pvt *pvt = channel->pvt;
293         struct ast_sip_session *session = channel->session;
294         int changed = 0;
295         struct ast_channel *bridge_peer;
296
297         /* Don't try to do any direct media shenanigans on early bridges */
298         bridge_peer = ast_channel_bridge_peer(chan);
299         if ((rtp || vrtp || tpeer) && !bridge_peer) {
300                 return 0;
301         }
302         ast_channel_cleanup(bridge_peer);
303
304         if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
305                 return 0;
306         }
307
308         if (pvt->media[SIP_MEDIA_AUDIO]) {
309                 changed |= check_for_rtp_changes(chan, rtp, pvt->media[SIP_MEDIA_AUDIO], 1);
310         }
311         if (pvt->media[SIP_MEDIA_VIDEO]) {
312                 changed |= check_for_rtp_changes(chan, vrtp, pvt->media[SIP_MEDIA_VIDEO], 3);
313         }
314
315         if (direct_media_mitigate_glare(session)) {
316                 return 0;
317         }
318
319         if (cap && !ast_format_cap_is_empty(cap) && !ast_format_cap_identical(session->direct_media_cap, cap)) {
320                 ast_format_cap_copy(session->direct_media_cap, cap);
321                 changed = 1;
322         }
323
324         if (changed) {
325                 ao2_ref(session, +1);
326
327
328                 if (ast_sip_push_task(session->serializer, send_direct_media_request, session)) {
329                         ao2_cleanup(session);
330                 }
331         }
332
333         return 0;
334 }
335
336 /*! \brief Local glue for interacting with the RTP engine core */
337 static struct ast_rtp_glue chan_pjsip_rtp_glue = {
338         .type = "PJSIP",
339         .get_rtp_info = chan_pjsip_get_rtp_peer,
340         .get_vrtp_info = chan_pjsip_get_vrtp_peer,
341         .get_codec = chan_pjsip_get_codec,
342         .update_peer = chan_pjsip_set_rtp_peer,
343 };
344
345 /*! \brief Function called to create a new PJSIP Asterisk channel */
346 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)
347 {
348         struct ast_channel *chan;
349         struct ast_format fmt;
350         RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
351         struct ast_sip_channel_pvt *channel;
352
353         if (!(pvt = ao2_alloc(sizeof(*pvt), chan_pjsip_pvt_dtor))) {
354                 return NULL;
355         }
356
357         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),
358                 ast_atomic_fetchadd_int((int *)&chan_idx, +1)))) {
359                 return NULL;
360         }
361
362         ast_channel_tech_set(chan, &chan_pjsip_tech);
363
364         if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
365                 ast_hangup(chan);
366                 return NULL;
367         }
368
369         ast_channel_stage_snapshot(chan);
370
371         /* If res_pjsip_session is ever updated to create/destroy ast_sip_session_media
372          * during a call such as if multiple same-type stream support is introduced,
373          * these will need to be recaptured as well */
374         pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
375         pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
376         ast_channel_tech_pvt_set(chan, channel);
377         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
378                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(chan));
379         }
380         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
381                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(chan));
382         }
383
384         if (ast_format_cap_is_empty(session->req_caps) || !ast_format_cap_has_joint(session->req_caps, session->endpoint->media.codecs)) {
385                 ast_format_cap_copy(ast_channel_nativeformats(chan), session->endpoint->media.codecs);
386         } else {
387                 ast_format_cap_copy(ast_channel_nativeformats(chan), session->req_caps);
388         }
389
390         ast_codec_choose(&session->endpoint->media.prefs, ast_channel_nativeformats(chan), 1, &fmt);
391         ast_format_copy(ast_channel_writeformat(chan), &fmt);
392         ast_format_copy(ast_channel_rawwriteformat(chan), &fmt);
393         ast_format_copy(ast_channel_readformat(chan), &fmt);
394         ast_format_copy(ast_channel_rawreadformat(chan), &fmt);
395
396         if (state == AST_STATE_RING) {
397                 ast_channel_rings_set(chan, 1);
398         }
399
400         ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
401
402         ast_channel_context_set(chan, session->endpoint->context);
403         ast_channel_exten_set(chan, S_OR(exten, "s"));
404         ast_channel_priority_set(chan, 1);
405
406         ast_channel_callgroup_set(chan, session->endpoint->pickup.callgroup);
407         ast_channel_pickupgroup_set(chan, session->endpoint->pickup.pickupgroup);
408
409         ast_channel_named_callgroups_set(chan, session->endpoint->pickup.named_callgroups);
410         ast_channel_named_pickupgroups_set(chan, session->endpoint->pickup.named_pickupgroups);
411
412         if (!ast_strlen_zero(session->endpoint->language)) {
413                 ast_channel_language_set(chan, session->endpoint->language);
414         }
415
416         if (!ast_strlen_zero(session->endpoint->zone)) {
417                 struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone);
418                 if (!zone) {
419                         ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone);
420                 }
421                 ast_channel_zone_set(chan, zone);
422         }
423
424         ast_endpoint_add_channel(session->endpoint->persistent, chan);
425
426         ast_channel_stage_snapshot_done(chan);
427
428         return chan;
429 }
430
431 static int answer(void *data)
432 {
433         pj_status_t status = PJ_SUCCESS;
434         pjsip_tx_data *packet;
435         struct ast_sip_session *session = data;
436
437         pjsip_dlg_inc_lock(session->inv_session->dlg);
438         if (session->inv_session->invite_tsx) {
439                 status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet);
440         }
441         pjsip_dlg_dec_lock(session->inv_session->dlg);
442
443         if (status == PJ_SUCCESS && packet) {
444                 ast_sip_session_send_response(session, packet);
445         }
446
447         ao2_ref(session, -1);
448
449         return (status == PJ_SUCCESS) ? 0 : -1;
450 }
451
452 /*! \brief Function called by core when we should answer a PJSIP session */
453 static int chan_pjsip_answer(struct ast_channel *ast)
454 {
455         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
456
457         if (ast_channel_state(ast) == AST_STATE_UP) {
458                 return 0;
459         }
460
461         ast_setstate(ast, AST_STATE_UP);
462
463         ao2_ref(channel->session, +1);
464         if (ast_sip_push_task(channel->session->serializer, answer, channel->session)) {
465                 ast_log(LOG_WARNING, "Unable to push answer task to the threadpool. Cannot answer call\n");
466                 ao2_cleanup(channel->session);
467                 return -1;
468         }
469
470         return 0;
471 }
472
473 /*! \brief Internal helper function called when CNG tone is detected */
474 static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f)
475 {
476         const char *target_context;
477         int exists;
478
479         /* If we only needed this DSP for fax detection purposes we can just drop it now */
480         if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
481                 ast_dsp_set_features(session->dsp, DSP_FEATURE_DIGIT_DETECT);
482         } else {
483                 ast_dsp_free(session->dsp);
484                 session->dsp = NULL;
485         }
486
487         /* If already executing in the fax extension don't do anything */
488         if (!strcmp(ast_channel_exten(session->channel), "fax")) {
489                 return f;
490         }
491
492         target_context = S_OR(ast_channel_macrocontext(session->channel), ast_channel_context(session->channel));
493
494         /* We need to unlock the channel here because ast_exists_extension has the
495          * potential to start and stop an autoservice on the channel. Such action
496          * is prone to deadlock if the channel is locked.
497          */
498         ast_channel_unlock(session->channel);
499         exists = ast_exists_extension(session->channel, target_context, "fax", 1,
500                 S_COR(ast_channel_caller(session->channel)->id.number.valid,
501                         ast_channel_caller(session->channel)->id.number.str, NULL));
502         ast_channel_lock(session->channel);
503
504         if (exists) {
505                 ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
506                         ast_channel_name(session->channel));
507                 pbx_builtin_setvar_helper(session->channel, "FAXEXTEN", ast_channel_exten(session->channel));
508                 if (ast_async_goto(session->channel, target_context, "fax", 1)) {
509                         ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
510                                 ast_channel_name(session->channel), target_context);
511                 }
512                 ast_frfree(f);
513                 f = &ast_null_frame;
514         } else {
515                 ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
516                         ast_channel_name(session->channel), target_context);
517         }
518
519         return f;
520 }
521
522 /*! \brief Function called by core to read any waiting frames */
523 static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
524 {
525         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
526         struct chan_pjsip_pvt *pvt = channel->pvt;
527         struct ast_frame *f;
528         struct ast_sip_session_media *media = NULL;
529         int rtcp = 0;
530         int fdno = ast_channel_fdno(ast);
531
532         switch (fdno) {
533         case 0:
534                 media = pvt->media[SIP_MEDIA_AUDIO];
535                 break;
536         case 1:
537                 media = pvt->media[SIP_MEDIA_AUDIO];
538                 rtcp = 1;
539                 break;
540         case 2:
541                 media = pvt->media[SIP_MEDIA_VIDEO];
542                 break;
543         case 3:
544                 media = pvt->media[SIP_MEDIA_VIDEO];
545                 rtcp = 1;
546                 break;
547         }
548
549         if (!media || !media->rtp) {
550                 return &ast_null_frame;
551         }
552
553         if (!(f = ast_rtp_instance_read(media->rtp, rtcp))) {
554                 return f;
555         }
556
557         if (f->frametype != AST_FRAME_VOICE) {
558                 return f;
559         }
560
561         if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &f->subclass.format))) {
562                 ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(&f->subclass.format));
563                 ast_format_cap_set(ast_channel_nativeformats(ast), &f->subclass.format);
564                 ast_set_read_format(ast, ast_channel_readformat(ast));
565                 ast_set_write_format(ast, ast_channel_writeformat(ast));
566         }
567
568         if (channel->session->dsp) {
569                 f = ast_dsp_process(ast, channel->session->dsp, f);
570
571                 if (f && (f->frametype == AST_FRAME_DTMF)) {
572                         if (f->subclass.integer == 'f') {
573                                 ast_debug(3, "Fax CNG detected on %s\n", ast_channel_name(ast));
574                                 f = chan_pjsip_cng_tone_detected(channel->session, f);
575                         } else {
576                                 ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
577                                         ast_channel_name(ast));
578                         }
579                 }
580         }
581
582         return f;
583 }
584
585 /*! \brief Function called by core to write frames */
586 static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame)
587 {
588         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
589         struct chan_pjsip_pvt *pvt = channel->pvt;
590         struct ast_sip_session_media *media;
591         int res = 0;
592
593         switch (frame->frametype) {
594         case AST_FRAME_VOICE:
595                 media = pvt->media[SIP_MEDIA_AUDIO];
596
597                 if (!media) {
598                         return 0;
599                 }
600                 if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) {
601                         char buf[256];
602
603                         ast_log(LOG_WARNING,
604                                 "Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
605                                 ast_getformatname(&frame->subclass.format),
606                                 ast_getformatname_multiple(buf, sizeof(buf), ast_channel_nativeformats(ast)),
607                                 ast_getformatname(ast_channel_readformat(ast)),
608                                 ast_getformatname(ast_channel_writeformat(ast)));
609                         return 0;
610                 }
611                 if (media->rtp) {
612                         res = ast_rtp_instance_write(media->rtp, frame);
613                 }
614                 break;
615         case AST_FRAME_VIDEO:
616                 if ((media = pvt->media[SIP_MEDIA_VIDEO]) && media->rtp) {
617                         res = ast_rtp_instance_write(media->rtp, frame);
618                 }
619                 break;
620         case AST_FRAME_MODEM:
621                 break;
622         default:
623                 ast_log(LOG_WARNING, "Can't send %d type frames with PJSIP\n", frame->frametype);
624                 break;
625         }
626
627         return res;
628 }
629
630 struct fixup_data {
631         struct ast_sip_session *session;
632         struct ast_channel *chan;
633 };
634
635 static int fixup(void *data)
636 {
637         struct fixup_data *fix_data = data;
638         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(fix_data->chan);
639         struct chan_pjsip_pvt *pvt = channel->pvt;
640
641         channel->session->channel = fix_data->chan;
642         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
643                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(fix_data->chan));
644         }
645         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
646                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(fix_data->chan));
647         }
648
649         return 0;
650 }
651
652 /*! \brief Function called by core to change the underlying owner channel */
653 static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
654 {
655         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
656         struct fixup_data fix_data;
657
658         fix_data.session = channel->session;
659         fix_data.chan = newchan;
660
661         if (channel->session->channel != oldchan) {
662                 return -1;
663         }
664
665         if (ast_sip_push_task_synchronous(channel->session->serializer, fixup, &fix_data)) {
666                 ast_log(LOG_WARNING, "Unable to perform channel fixup\n");
667                 return -1;
668         }
669
670         return 0;
671 }
672
673 /*! \brief Function called to get the device state of an endpoint */
674 static int chan_pjsip_devicestate(const char *data)
675 {
676         RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
677         enum ast_device_state state = AST_DEVICE_UNKNOWN;
678         RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
679         RAII_VAR(struct stasis_cache *, cache, NULL, ao2_cleanup);
680         struct ast_devstate_aggregate aggregate;
681         int num, inuse = 0;
682
683         if (!endpoint) {
684                 return AST_DEVICE_INVALID;
685         }
686
687         endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent),
688                 ast_endpoint_get_resource(endpoint->persistent));
689
690         if (!endpoint_snapshot) {
691                 return AST_DEVICE_INVALID;
692         }
693
694         if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) {
695                 state = AST_DEVICE_UNAVAILABLE;
696         } else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) {
697                 state = AST_DEVICE_NOT_INUSE;
698         }
699
700         if (!endpoint_snapshot->num_channels || !(cache = ast_channel_cache())) {
701                 return state;
702         }
703
704         ast_devstate_aggregate_init(&aggregate);
705
706         ao2_ref(cache, +1);
707
708         for (num = 0; num < endpoint_snapshot->num_channels; num++) {
709                 RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
710                 struct ast_channel_snapshot *snapshot;
711
712                 msg = stasis_cache_get(cache, ast_channel_snapshot_type(),
713                         endpoint_snapshot->channel_ids[num]);
714
715                 if (!msg) {
716                         continue;
717                 }
718
719                 snapshot = stasis_message_data(msg);
720
721                 if (snapshot->state == AST_STATE_DOWN) {
722                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_NOT_INUSE);
723                 } else if (snapshot->state == AST_STATE_RINGING) {
724                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_RINGING);
725                 } else if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
726                         (snapshot->state == AST_STATE_BUSY)) {
727                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_INUSE);
728                         inuse++;
729                 }
730         }
731
732         if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) {
733                 state = AST_DEVICE_BUSY;
734         } else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) {
735                 state = ast_devstate_aggregate_result(&aggregate);
736         }
737
738         return state;
739 }
740
741 /*! \brief Function called to query options on a channel */
742 static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen)
743 {
744         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
745         struct ast_sip_session *session = channel->session;
746         int res = -1;
747         enum ast_sip_session_t38state state = T38_STATE_UNAVAILABLE;
748
749         switch (option) {
750         case AST_OPTION_T38_STATE:
751                 if (session->endpoint->media.t38.enabled) {
752                         switch (session->t38state) {
753                         case T38_LOCAL_REINVITE:
754                         case T38_PEER_REINVITE:
755                                 state = T38_STATE_NEGOTIATING;
756                                 break;
757                         case T38_ENABLED:
758                                 state = T38_STATE_NEGOTIATED;
759                                 break;
760                         case T38_REJECTED:
761                                 state = T38_STATE_REJECTED;
762                                 break;
763                         default:
764                                 state = T38_STATE_UNKNOWN;
765                                 break;
766                         }
767                 }
768
769                 *((enum ast_t38_state *) data) = state;
770                 res = 0;
771
772                 break;
773         default:
774                 break;
775         }
776
777         return res;
778 }
779
780 struct indicate_data {
781         struct ast_sip_session *session;
782         int condition;
783         int response_code;
784         void *frame_data;
785         size_t datalen;
786 };
787
788 static void indicate_data_destroy(void *obj)
789 {
790         struct indicate_data *ind_data = obj;
791
792         ast_free(ind_data->frame_data);
793         ao2_ref(ind_data->session, -1);
794 }
795
796 static struct indicate_data *indicate_data_alloc(struct ast_sip_session *session,
797                 int condition, int response_code, const void *frame_data, size_t datalen)
798 {
799         struct indicate_data *ind_data = ao2_alloc(sizeof(*ind_data), indicate_data_destroy);
800
801         if (!ind_data) {
802                 return NULL;
803         }
804
805         ind_data->frame_data = ast_malloc(datalen);
806         if (!ind_data->frame_data) {
807                 ao2_ref(ind_data, -1);
808                 return NULL;
809         }
810
811         memcpy(ind_data->frame_data, frame_data, datalen);
812         ind_data->datalen = datalen;
813         ind_data->condition = condition;
814         ind_data->response_code = response_code;
815         ao2_ref(session, +1);
816         ind_data->session = session;
817
818         return ind_data;
819 }
820
821 static int indicate(void *data)
822 {
823         pjsip_tx_data *packet = NULL;
824         struct indicate_data *ind_data = data;
825         struct ast_sip_session *session = ind_data->session;
826         int response_code = ind_data->response_code;
827
828         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
829                 ast_sip_session_send_response(session, packet);
830         }
831
832         ao2_ref(ind_data, -1);
833
834         return 0;
835 }
836
837 /*! \brief Send SIP INFO with video update request */
838 static int transmit_info_with_vidupdate(void *data)
839 {
840         const char * xml =
841                 "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
842                 " <media_control>\r\n"
843                 "  <vc_primitive>\r\n"
844                 "   <to_encoder>\r\n"
845                 "    <picture_fast_update/>\r\n"
846                 "   </to_encoder>\r\n"
847                 "  </vc_primitive>\r\n"
848                 " </media_control>\r\n";
849
850         const struct ast_sip_body body = {
851                 .type = "application",
852                 .subtype = "media_control+xml",
853                 .body_text = xml
854         };
855
856         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
857         struct pjsip_tx_data *tdata;
858
859         if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
860                 ast_log(LOG_ERROR, "Could not create text video update INFO request\n");
861                 return -1;
862         }
863         if (ast_sip_add_body(tdata, &body)) {
864                 ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n");
865                 return -1;
866         }
867         ast_sip_session_send_request(session, tdata);
868
869         return 0;
870 }
871
872 /*! \brief Update connected line information */
873 static int update_connected_line_information(void *data)
874 {
875         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
876         struct ast_party_id connected_id;
877
878         if ((ast_channel_state(session->channel) != AST_STATE_UP) && (session->inv_session->role == PJSIP_UAS_ROLE)) {
879                 int response_code = 0;
880
881                 if (ast_channel_state(session->channel) == AST_STATE_RING) {
882                         response_code = !session->endpoint->inband_progress ? 180 : 183;
883                 } else if (ast_channel_state(session->channel) == AST_STATE_RINGING) {
884                         response_code = 183;
885                 }
886
887                 if (response_code) {
888                         struct pjsip_tx_data *packet = NULL;
889
890                         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
891                                 ast_sip_session_send_response(session, packet);
892                         }
893                 }
894         } else {
895                 enum ast_sip_session_refresh_method method = session->endpoint->id.refresh_method;
896
897                 if (session->inv_session->invite_tsx && (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
898                         method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
899                 }
900
901                 connected_id = ast_channel_connected_effective_id(session->channel);
902                 if ((session->endpoint->id.send_pai || session->endpoint->id.send_rpid) &&
903                     (session->endpoint->id.trust_outbound ||
904                      ((connected_id.name.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED &&
905                       (connected_id.number.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED))) {
906                         ast_sip_session_refresh(session, NULL, NULL, NULL, method, 1);
907                 }
908         }
909
910         return 0;
911 }
912
913 /*! \brief Function called by core to ask the channel to indicate some sort of condition */
914 static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
915 {
916         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
917         struct chan_pjsip_pvt *pvt = channel->pvt;
918         struct ast_sip_session_media *media;
919         int response_code = 0;
920         int res = 0;
921
922         switch (condition) {
923         case AST_CONTROL_RINGING:
924                 if (ast_channel_state(ast) == AST_STATE_RING) {
925                         if (channel->session->endpoint->inband_progress) {
926                                 response_code = 183;
927                                 res = -1;
928                         } else {
929                                 response_code = 180;
930                         }
931                 } else {
932                         res = -1;
933                 }
934                 ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
935                 break;
936         case AST_CONTROL_BUSY:
937                 if (ast_channel_state(ast) != AST_STATE_UP) {
938                         response_code = 486;
939                 } else {
940                         res = -1;
941                 }
942                 break;
943         case AST_CONTROL_CONGESTION:
944                 if (ast_channel_state(ast) != AST_STATE_UP) {
945                         response_code = 503;
946                 } else {
947                         res = -1;
948                 }
949                 break;
950         case AST_CONTROL_INCOMPLETE:
951                 if (ast_channel_state(ast) != AST_STATE_UP) {
952                         response_code = 484;
953                 } else {
954                         res = -1;
955                 }
956                 break;
957         case AST_CONTROL_PROCEEDING:
958                 if (ast_channel_state(ast) != AST_STATE_UP) {
959                         response_code = 100;
960                 } else {
961                         res = -1;
962                 }
963                 break;
964         case AST_CONTROL_PROGRESS:
965                 if (ast_channel_state(ast) != AST_STATE_UP) {
966                         response_code = 183;
967                 } else {
968                         res = -1;
969                 }
970                 break;
971         case AST_CONTROL_VIDUPDATE:
972                 media = pvt->media[SIP_MEDIA_VIDEO];
973                 if (media && media->rtp) {
974                         /* FIXME: Only use this for VP8. Additional work would have to be done to
975                          * fully support other video codecs */
976                         struct ast_format_cap *fcap = ast_channel_nativeformats(ast);
977                         struct ast_format vp8;
978                         ast_format_set(&vp8, AST_FORMAT_VP8, 0);
979                         if (ast_format_cap_iscompatible(fcap, &vp8)) {
980                                 /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
981                                  * RTP engine would provide a way to externally write/schedule RTCP
982                                  * packets */
983                                 struct ast_frame fr;
984                                 fr.frametype = AST_FRAME_CONTROL;
985                                 fr.subclass.integer = AST_CONTROL_VIDUPDATE;
986                                 res = ast_rtp_instance_write(media->rtp, &fr);
987                         } else {
988                                 ao2_ref(channel->session, +1);
989
990                                 if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
991                                         ao2_cleanup(channel->session);
992                                 }
993                         }
994                 } else {
995                         res = -1;
996                 }
997                 break;
998         case AST_CONTROL_CONNECTED_LINE:
999                 ao2_ref(channel->session, +1);
1000                 if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
1001                         ao2_cleanup(channel->session);
1002                 }
1003                 break;
1004         case AST_CONTROL_UPDATE_RTP_PEER:
1005                 break;
1006         case AST_CONTROL_PVT_CAUSE_CODE:
1007                 res = -1;
1008                 break;
1009         case AST_CONTROL_HOLD:
1010                 ast_moh_start(ast, data, NULL);
1011                 break;
1012         case AST_CONTROL_UNHOLD:
1013                 ast_moh_stop(ast);
1014                 break;
1015         case AST_CONTROL_SRCUPDATE:
1016                 break;
1017         case AST_CONTROL_SRCCHANGE:
1018                 break;
1019         case AST_CONTROL_REDIRECTING:
1020                 if (ast_channel_state(ast) != AST_STATE_UP) {
1021                         response_code = 181;
1022                 } else {
1023                         res = -1;
1024                 }
1025                 break;
1026         case AST_CONTROL_T38_PARAMETERS:
1027                 res = 0;
1028
1029                 if (channel->session->t38state == T38_PEER_REINVITE) {
1030                         const struct ast_control_t38_parameters *parameters = data;
1031
1032                         if (parameters->request_response == AST_T38_REQUEST_PARMS) {
1033                                 res = AST_T38_REQUEST_PARMS;
1034                         }
1035                 }
1036
1037                 break;
1038         case -1:
1039                 res = -1;
1040                 break;
1041         default:
1042                 ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
1043                 res = -1;
1044                 break;
1045         }
1046
1047         if (response_code) {
1048                 struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
1049                 if (!ind_data || ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
1050                         ast_log(LOG_NOTICE, "Cannot send response code %d to endpoint %s. Could not queue task properly\n",
1051                                         response_code, ast_sorcery_object_get_id(channel->session->endpoint));
1052                         ao2_cleanup(ind_data);
1053                         res = -1;
1054                 }
1055         }
1056
1057         return res;
1058 }
1059
1060 struct transfer_data {
1061         struct ast_sip_session *session;
1062         char *target;
1063 };
1064
1065 static void transfer_data_destroy(void *obj)
1066 {
1067         struct transfer_data *trnf_data = obj;
1068
1069         ast_free(trnf_data->target);
1070         ao2_cleanup(trnf_data->session);
1071 }
1072
1073 static struct transfer_data *transfer_data_alloc(struct ast_sip_session *session, const char *target)
1074 {
1075         struct transfer_data *trnf_data = ao2_alloc(sizeof(*trnf_data), transfer_data_destroy);
1076
1077         if (!trnf_data) {
1078                 return NULL;
1079         }
1080
1081         if (!(trnf_data->target = ast_strdup(target))) {
1082                 ao2_ref(trnf_data, -1);
1083                 return NULL;
1084         }
1085
1086         ao2_ref(session, +1);
1087         trnf_data->session = session;
1088
1089         return trnf_data;
1090 }
1091
1092 static void transfer_redirect(struct ast_sip_session *session, const char *target)
1093 {
1094         pjsip_tx_data *packet;
1095         enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1096         pjsip_contact_hdr *contact;
1097         pj_str_t tmp;
1098
1099         if (pjsip_inv_end_session(session->inv_session, 302, NULL, &packet) != PJ_SUCCESS) {
1100                 message = AST_TRANSFER_FAILED;
1101                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1102
1103                 return;
1104         }
1105
1106         if (!(contact = pjsip_msg_find_hdr(packet->msg, PJSIP_H_CONTACT, NULL))) {
1107                 contact = pjsip_contact_hdr_create(packet->pool);
1108         }
1109
1110         pj_strdup2_with_null(packet->pool, &tmp, target);
1111         if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) {
1112                 message = AST_TRANSFER_FAILED;
1113                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1114                 pjsip_tx_data_dec_ref(packet);
1115
1116                 return;
1117         }
1118         pjsip_msg_add_hdr(packet->msg, (pjsip_hdr *) contact);
1119
1120         ast_sip_session_send_response(session, packet);
1121         ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1122 }
1123
1124 static void transfer_refer(struct ast_sip_session *session, const char *target)
1125 {
1126         pjsip_evsub *sub;
1127         enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1128         pj_str_t tmp;
1129         pjsip_tx_data *packet;
1130
1131         if (pjsip_xfer_create_uac(session->inv_session->dlg, NULL, &sub) != PJ_SUCCESS) {
1132                 message = AST_TRANSFER_FAILED;
1133                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1134
1135                 return;
1136         }
1137
1138         if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
1139                 message = AST_TRANSFER_FAILED;
1140                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1141                 pjsip_evsub_terminate(sub, PJ_FALSE);
1142
1143                 return;
1144         }
1145
1146         pjsip_xfer_send_request(sub, packet);
1147         ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1148 }
1149
1150 static int transfer(void *data)
1151 {
1152         struct transfer_data *trnf_data = data;
1153
1154         if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
1155                 transfer_redirect(trnf_data->session, trnf_data->target);
1156         } else {
1157                 transfer_refer(trnf_data->session, trnf_data->target);
1158         }
1159
1160         ao2_ref(trnf_data, -1);
1161         return 0;
1162 }
1163
1164 /*! \brief Function called by core for Asterisk initiated transfer */
1165 static int chan_pjsip_transfer(struct ast_channel *chan, const char *target)
1166 {
1167         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
1168         struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
1169
1170         if (!trnf_data) {
1171                 return -1;
1172         }
1173
1174         if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
1175                 ast_log(LOG_WARNING, "Error requesting transfer\n");
1176                 ao2_cleanup(trnf_data);
1177                 return -1;
1178         }
1179
1180         return 0;
1181 }
1182
1183 /*! \brief Function called by core to start a DTMF digit */
1184 static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
1185 {
1186         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
1187         struct chan_pjsip_pvt *pvt = channel->pvt;
1188         struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
1189         int res = 0;
1190
1191         switch (channel->session->endpoint->dtmf) {
1192         case AST_SIP_DTMF_RFC_4733:
1193                 if (!media || !media->rtp) {
1194                         return -1;
1195                 }
1196
1197                 ast_rtp_instance_dtmf_begin(media->rtp, digit);
1198         case AST_SIP_DTMF_NONE:
1199                 break;
1200         case AST_SIP_DTMF_INBAND:
1201                 res = -1;
1202                 break;
1203         default:
1204                 break;
1205         }
1206
1207         return res;
1208 }
1209
1210 struct info_dtmf_data {
1211         struct ast_sip_session *session;
1212         char digit;
1213         unsigned int duration;
1214 };
1215
1216 static void info_dtmf_data_destroy(void *obj)
1217 {
1218         struct info_dtmf_data *dtmf_data = obj;
1219         ao2_ref(dtmf_data->session, -1);
1220 }
1221
1222 static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
1223 {
1224         struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy);
1225         if (!dtmf_data) {
1226                 return NULL;
1227         }
1228         ao2_ref(session, +1);
1229         dtmf_data->session = session;
1230         dtmf_data->digit = digit;
1231         dtmf_data->duration = duration;
1232         return dtmf_data;
1233 }
1234
1235 static int transmit_info_dtmf(void *data)
1236 {
1237         RAII_VAR(struct info_dtmf_data *, dtmf_data, data, ao2_cleanup);
1238
1239         struct ast_sip_session *session = dtmf_data->session;
1240         struct pjsip_tx_data *tdata;
1241
1242         RAII_VAR(struct ast_str *, body_text, NULL, ast_free_ptr);
1243
1244         struct ast_sip_body body = {
1245                 .type = "application",
1246                 .subtype = "dtmf-relay",
1247         };
1248
1249         if (!(body_text = ast_str_create(32))) {
1250                 ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n");
1251                 return -1;
1252         }
1253         ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration);
1254
1255         body.body_text = ast_str_buffer(body_text);
1256
1257         if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
1258                 ast_log(LOG_ERROR, "Could not create DTMF INFO request\n");
1259                 return -1;
1260         }
1261         if (ast_sip_add_body(tdata, &body)) {
1262                 ast_log(LOG_ERROR, "Could not add body to DTMF INFO request\n");
1263                 pjsip_tx_data_dec_ref(tdata);
1264                 return -1;
1265         }
1266         ast_sip_session_send_request(session, tdata);
1267
1268         return 0;
1269 }
1270
1271 /*! \brief Function called by core to stop a DTMF digit */
1272 static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
1273 {
1274         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1275         struct chan_pjsip_pvt *pvt = channel->pvt;
1276         struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
1277         int res = 0;
1278
1279         switch (channel->session->endpoint->dtmf) {
1280         case AST_SIP_DTMF_INFO:
1281         {
1282                 struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
1283
1284                 if (!dtmf_data) {
1285                         return -1;
1286                 }
1287
1288                 if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
1289                         ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
1290                         ao2_cleanup(dtmf_data);
1291                         return -1;
1292                 }
1293                 break;
1294         }
1295         case AST_SIP_DTMF_RFC_4733:
1296                 if (!media || !media->rtp) {
1297                         return -1;
1298                 }
1299
1300                 ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
1301         case AST_SIP_DTMF_NONE:
1302                 break;
1303         case AST_SIP_DTMF_INBAND:
1304                 res = -1;
1305                 break;
1306         }
1307
1308         return res;
1309 }
1310
1311 static int call(void *data)
1312 {
1313         struct ast_sip_session *session = data;
1314         pjsip_tx_data *tdata;
1315
1316         int res = ast_sip_session_create_invite(session, &tdata);
1317
1318         if (res) {
1319                 ast_queue_hangup(session->channel);
1320         } else {
1321                 ast_sip_session_send_request(session, tdata);
1322         }
1323         ao2_ref(session, -1);
1324         return res;
1325 }
1326
1327 /*! \brief Function called by core to actually start calling a remote party */
1328 static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout)
1329 {
1330         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1331
1332         ao2_ref(channel->session, +1);
1333         if (ast_sip_push_task(channel->session->serializer, call, channel->session)) {
1334                 ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
1335                 ao2_cleanup(channel->session);
1336                 return -1;
1337         }
1338
1339         return 0;
1340 }
1341
1342 /*! \brief Internal function which translates from Asterisk cause codes to SIP response codes */
1343 static int hangup_cause2sip(int cause)
1344 {
1345         switch (cause) {
1346         case AST_CAUSE_UNALLOCATED:             /* 1 */
1347         case AST_CAUSE_NO_ROUTE_DESTINATION:    /* 3 IAX2: Can't find extension in context */
1348         case AST_CAUSE_NO_ROUTE_TRANSIT_NET:    /* 2 */
1349                 return 404;
1350         case AST_CAUSE_CONGESTION:              /* 34 */
1351         case AST_CAUSE_SWITCH_CONGESTION:       /* 42 */
1352                 return 503;
1353         case AST_CAUSE_NO_USER_RESPONSE:        /* 18 */
1354                 return 408;
1355         case AST_CAUSE_NO_ANSWER:               /* 19 */
1356         case AST_CAUSE_UNREGISTERED:        /* 20 */
1357                 return 480;
1358         case AST_CAUSE_CALL_REJECTED:           /* 21 */
1359                 return 403;
1360         case AST_CAUSE_NUMBER_CHANGED:          /* 22 */
1361                 return 410;
1362         case AST_CAUSE_NORMAL_UNSPECIFIED:      /* 31 */
1363                 return 480;
1364         case AST_CAUSE_INVALID_NUMBER_FORMAT:
1365                 return 484;
1366         case AST_CAUSE_USER_BUSY:
1367                 return 486;
1368         case AST_CAUSE_FAILURE:
1369                 return 500;
1370         case AST_CAUSE_FACILITY_REJECTED:       /* 29 */
1371                 return 501;
1372         case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
1373                 return 503;
1374         case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
1375                 return 502;
1376         case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:       /* Can't find codec to connect to host */
1377                 return 488;
1378         case AST_CAUSE_INTERWORKING:    /* Unspecified Interworking issues */
1379                 return 500;
1380         case AST_CAUSE_NOTDEFINED:
1381         default:
1382                 ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
1383                 return 0;
1384         }
1385
1386         /* Never reached */
1387         return 0;
1388 }
1389
1390 struct hangup_data {
1391         int cause;
1392         struct ast_channel *chan;
1393 };
1394
1395 static void hangup_data_destroy(void *obj)
1396 {
1397         struct hangup_data *h_data = obj;
1398
1399         h_data->chan = ast_channel_unref(h_data->chan);
1400 }
1401
1402 static struct hangup_data *hangup_data_alloc(int cause, struct ast_channel *chan)
1403 {
1404         struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
1405
1406         if (!h_data) {
1407                 return NULL;
1408         }
1409
1410         h_data->cause = cause;
1411         h_data->chan = ast_channel_ref(chan);
1412
1413         return h_data;
1414 }
1415
1416 /*! \brief Clear a channel from a session along with its PVT */
1417 static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast, struct chan_pjsip_pvt *pvt)
1418 {
1419         session->channel = NULL;
1420         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
1421                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, "");
1422         }
1423         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
1424                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, "");
1425         }
1426         ast_channel_tech_pvt_set(ast, NULL);
1427 }
1428
1429 static int hangup(void *data)
1430 {
1431         pj_status_t status;
1432         pjsip_tx_data *packet = NULL;
1433         struct hangup_data *h_data = data;
1434         struct ast_channel *ast = h_data->chan;
1435         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1436         struct chan_pjsip_pvt *pvt = channel->pvt;
1437         struct ast_sip_session *session = channel->session;
1438         int cause = h_data->cause;
1439
1440         if (!session->defer_terminate &&
1441                 ((status = pjsip_inv_end_session(session->inv_session, cause ? cause : 603, NULL, &packet)) == PJ_SUCCESS) && packet) {
1442                 if (packet->msg->type == PJSIP_RESPONSE_MSG) {
1443                         ast_sip_session_send_response(session, packet);
1444                 } else {
1445                         ast_sip_session_send_request(session, packet);
1446                 }
1447         }
1448
1449         clear_session_and_channel(session, ast, pvt);
1450         ao2_cleanup(channel);
1451         ao2_cleanup(h_data);
1452
1453         return 0;
1454 }
1455
1456 /*! \brief Function called by core to hang up a PJSIP session */
1457 static int chan_pjsip_hangup(struct ast_channel *ast)
1458 {
1459         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1460         struct chan_pjsip_pvt *pvt = channel->pvt;
1461         int cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
1462         struct hangup_data *h_data = hangup_data_alloc(cause, ast);
1463
1464         if (!h_data) {
1465                 goto failure;
1466         }
1467
1468         if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
1469                 ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
1470                 goto failure;
1471         }
1472
1473         return 0;
1474
1475 failure:
1476         /* Go ahead and do our cleanup of the session and channel even if we're not going
1477          * to be able to send our SIP request/response
1478          */
1479         clear_session_and_channel(channel->session, ast, pvt);
1480         ao2_cleanup(channel);
1481         ao2_cleanup(h_data);
1482
1483         return -1;
1484 }
1485
1486 struct request_data {
1487         struct ast_sip_session *session;
1488         struct ast_format_cap *caps;
1489         const char *dest;
1490         int cause;
1491 };
1492
1493 static int request(void *obj)
1494 {
1495         struct request_data *req_data = obj;
1496         struct ast_sip_session *session = NULL;
1497         char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
1498         RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
1499
1500         AST_DECLARE_APP_ARGS(args,
1501                 AST_APP_ARG(endpoint);
1502                 AST_APP_ARG(aor);
1503         );
1504
1505         if (ast_strlen_zero(tmp)) {
1506                 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
1507                 req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
1508                 return -1;
1509         }
1510
1511         AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
1512
1513         /* If a request user has been specified extract it from the endpoint name portion */
1514         if ((endpoint_name = strchr(args.endpoint, '@'))) {
1515                 request_user = args.endpoint;
1516                 *endpoint_name++ = '\0';
1517         } else {
1518                 endpoint_name = args.endpoint;
1519         }
1520
1521         if (ast_strlen_zero(endpoint_name)) {
1522                 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
1523                 req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
1524         } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name))) {
1525                 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
1526                 req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
1527                 return -1;
1528         }
1529
1530         if (!(session = ast_sip_session_create_outgoing(endpoint, args.aor, request_user, req_data->caps))) {
1531                 req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
1532                 return -1;
1533         }
1534
1535         req_data->session = session;
1536
1537         return 0;
1538 }
1539
1540 /*! \brief Function called by core to create a new outgoing PJSIP session */
1541 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)
1542 {
1543         struct request_data req_data;
1544         RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
1545
1546         req_data.caps = cap;
1547         req_data.dest = data;
1548
1549         if (ast_sip_push_task_synchronous(NULL, request, &req_data)) {
1550                 *cause = req_data.cause;
1551                 return NULL;
1552         }
1553
1554         session = req_data.session;
1555
1556         if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, requestor ? ast_channel_linkedid(requestor) : NULL, NULL))) {
1557                 /* Session needs to be terminated prematurely */
1558                 return NULL;
1559         }
1560
1561         return session->channel;
1562 }
1563
1564 struct sendtext_data {
1565         struct ast_sip_session *session;
1566         char text[0];
1567 };
1568
1569 static void sendtext_data_destroy(void *obj)
1570 {
1571         struct sendtext_data *data = obj;
1572         ao2_ref(data->session, -1);
1573 }
1574
1575 static struct sendtext_data* sendtext_data_create(struct ast_sip_session *session, const char *text)
1576 {
1577         int size = strlen(text) + 1;
1578         struct sendtext_data *data = ao2_alloc(sizeof(*data)+size, sendtext_data_destroy);
1579
1580         if (!data) {
1581                 return NULL;
1582         }
1583
1584         data->session = session;
1585         ao2_ref(data->session, +1);
1586         ast_copy_string(data->text, text, size);
1587         return data;
1588 }
1589
1590 static int sendtext(void *obj)
1591 {
1592         RAII_VAR(struct sendtext_data *, data, obj, ao2_cleanup);
1593         pjsip_tx_data *tdata;
1594
1595         const struct ast_sip_body body = {
1596                 .type = "text",
1597                 .subtype = "plain",
1598                 .body_text = data->text
1599         };
1600
1601         /* NOT ast_strlen_zero, because a zero-length message is specifically
1602          * allowed by RFC 3428 (See section 10, Examples) */
1603         if (!data->text) {
1604                 return 0;
1605         }
1606
1607         ast_debug(3, "Sending in dialog SIP message\n");
1608
1609         ast_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, &tdata);
1610         ast_sip_add_body(tdata, &body);
1611         ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint);
1612
1613         return 0;
1614 }
1615
1616 /*! \brief Function called by core to send text on PJSIP session */
1617 static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text)
1618 {
1619         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1620         struct sendtext_data *data = sendtext_data_create(channel->session, text);
1621
1622         if (!data || ast_sip_push_task(channel->session->serializer, sendtext, data)) {
1623                 ao2_ref(data, -1);
1624                 return -1;
1625         }
1626         return 0;
1627 }
1628
1629 /*! \brief Convert SIP hangup causes to Asterisk hangup causes */
1630 static int hangup_sip2cause(int cause)
1631 {
1632         /* Possible values taken from causes.h */
1633
1634         switch(cause) {
1635         case 401:       /* Unauthorized */
1636                 return AST_CAUSE_CALL_REJECTED;
1637         case 403:       /* Not found */
1638                 return AST_CAUSE_CALL_REJECTED;
1639         case 404:       /* Not found */
1640                 return AST_CAUSE_UNALLOCATED;
1641         case 405:       /* Method not allowed */
1642                 return AST_CAUSE_INTERWORKING;
1643         case 407:       /* Proxy authentication required */
1644                 return AST_CAUSE_CALL_REJECTED;
1645         case 408:       /* No reaction */
1646                 return AST_CAUSE_NO_USER_RESPONSE;
1647         case 409:       /* Conflict */
1648                 return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
1649         case 410:       /* Gone */
1650                 return AST_CAUSE_NUMBER_CHANGED;
1651         case 411:       /* Length required */
1652                 return AST_CAUSE_INTERWORKING;
1653         case 413:       /* Request entity too large */
1654                 return AST_CAUSE_INTERWORKING;
1655         case 414:       /* Request URI too large */
1656                 return AST_CAUSE_INTERWORKING;
1657         case 415:       /* Unsupported media type */
1658                 return AST_CAUSE_INTERWORKING;
1659         case 420:       /* Bad extension */
1660                 return AST_CAUSE_NO_ROUTE_DESTINATION;
1661         case 480:       /* No answer */
1662                 return AST_CAUSE_NO_ANSWER;
1663         case 481:       /* No answer */
1664                 return AST_CAUSE_INTERWORKING;
1665         case 482:       /* Loop detected */
1666                 return AST_CAUSE_INTERWORKING;
1667         case 483:       /* Too many hops */
1668                 return AST_CAUSE_NO_ANSWER;
1669         case 484:       /* Address incomplete */
1670                 return AST_CAUSE_INVALID_NUMBER_FORMAT;
1671         case 485:       /* Ambiguous */
1672                 return AST_CAUSE_UNALLOCATED;
1673         case 486:       /* Busy everywhere */
1674                 return AST_CAUSE_BUSY;
1675         case 487:       /* Request terminated */
1676                 return AST_CAUSE_INTERWORKING;
1677         case 488:       /* No codecs approved */
1678                 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
1679         case 491:       /* Request pending */
1680                 return AST_CAUSE_INTERWORKING;
1681         case 493:       /* Undecipherable */
1682                 return AST_CAUSE_INTERWORKING;
1683         case 500:       /* Server internal failure */
1684                 return AST_CAUSE_FAILURE;
1685         case 501:       /* Call rejected */
1686                 return AST_CAUSE_FACILITY_REJECTED;
1687         case 502:
1688                 return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
1689         case 503:       /* Service unavailable */
1690                 return AST_CAUSE_CONGESTION;
1691         case 504:       /* Gateway timeout */
1692                 return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
1693         case 505:       /* SIP version not supported */
1694                 return AST_CAUSE_INTERWORKING;
1695         case 600:       /* Busy everywhere */
1696                 return AST_CAUSE_USER_BUSY;
1697         case 603:       /* Decline */
1698                 return AST_CAUSE_CALL_REJECTED;
1699         case 604:       /* Does not exist anywhere */
1700                 return AST_CAUSE_UNALLOCATED;
1701         case 606:       /* Not acceptable */
1702                 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
1703         default:
1704                 if (cause < 500 && cause >= 400) {
1705                         /* 4xx class error that is unknown - someting wrong with our request */
1706                         return AST_CAUSE_INTERWORKING;
1707                 } else if (cause < 600 && cause >= 500) {
1708                         /* 5xx class error - problem in the remote end */
1709                         return AST_CAUSE_CONGESTION;
1710                 } else if (cause < 700 && cause >= 600) {
1711                         /* 6xx - global errors in the 4xx class */
1712                         return AST_CAUSE_INTERWORKING;
1713                 }
1714                 return AST_CAUSE_NORMAL;
1715         }
1716         /* Never reached */
1717         return 0;
1718 }
1719
1720 static void chan_pjsip_session_begin(struct ast_sip_session *session)
1721 {
1722         RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
1723
1724         if (session->endpoint->media.direct_media.glare_mitigation ==
1725                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
1726                 return;
1727         }
1728
1729         datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info,
1730                         "direct_media_glare_mitigation");
1731
1732         if (!datastore) {
1733                 return;
1734         }
1735
1736         ast_sip_session_add_datastore(session, datastore);
1737 }
1738
1739 /*! \brief Function called when the session ends */
1740 static void chan_pjsip_session_end(struct ast_sip_session *session)
1741 {
1742         if (!session->channel) {
1743                 return;
1744         }
1745
1746         if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
1747                 int cause = hangup_sip2cause(session->inv_session->cause);
1748
1749                 ast_queue_hangup_with_cause(session->channel, cause);
1750         } else {
1751                 ast_queue_hangup(session->channel);
1752         }
1753 }
1754
1755 /*! \brief Function called when a request is received on the session */
1756 static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
1757 {
1758         RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
1759         struct transport_info_data *transport_data;
1760         pjsip_tx_data *packet = NULL;
1761
1762         if (session->channel) {
1763                 return 0;
1764         }
1765
1766         datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
1767         if (!datastore) {
1768                 return -1;
1769         }
1770
1771         transport_data = ast_calloc(1, sizeof(*transport_data));
1772         if (!transport_data) {
1773                 return -1;
1774         }
1775         pj_sockaddr_cp(&transport_data->local_addr, &rdata->tp_info.transport->local_addr);
1776         pj_sockaddr_cp(&transport_data->remote_addr, &rdata->pkt_info.src_addr);
1777         datastore->data = transport_data;
1778         ast_sip_session_add_datastore(session, datastore);
1779
1780         if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL))) {
1781                 if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
1782                         ast_sip_session_send_response(session, packet);
1783                 }
1784
1785                 ast_log(LOG_ERROR, "Failed to allocate new PJSIP channel on incoming SIP INVITE\n");
1786                 return -1;
1787         }
1788         /* channel gets created on incoming request, but we wait to call start
1789            so other supplements have a chance to run */
1790         return 0;
1791 }
1792
1793 static int pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
1794 {
1795         int res;
1796
1797         res = ast_pbx_start(session->channel);
1798
1799         switch (res) {
1800         case AST_PBX_FAILED:
1801                 ast_log(LOG_WARNING, "Failed to start PBX ;(\n");
1802                 ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
1803                 ast_hangup(session->channel);
1804                 break;
1805         case AST_PBX_CALL_LIMIT:
1806                 ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
1807                 ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
1808                 ast_hangup(session->channel);
1809                 break;
1810         case AST_PBX_SUCCESS:
1811         default:
1812                 break;
1813         }
1814
1815         ast_debug(3, "Started PBX on new PJSIP channel %s\n", ast_channel_name(session->channel));
1816
1817         return (res == AST_PBX_SUCCESS) ? 0 : -1;
1818 }
1819
1820 static struct ast_sip_session_supplement pbx_start_supplement = {
1821         .method = "INVITE",
1822         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_LAST,
1823         .incoming_request = pbx_start_incoming_request,
1824 };
1825
1826 /*! \brief Function called when a response is received on the session */
1827 static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
1828 {
1829         struct pjsip_status_line status = rdata->msg_info.msg->line.status;
1830
1831         if (!session->channel) {
1832                 return;
1833         }
1834
1835         switch (status.code) {
1836         case 180:
1837                 ast_queue_control(session->channel, AST_CONTROL_RINGING);
1838                 if (ast_channel_state(session->channel) != AST_STATE_UP) {
1839                         ast_setstate(session->channel, AST_STATE_RINGING);
1840                 }
1841                 break;
1842         case 183:
1843                 ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
1844                 break;
1845         case 200:
1846                 ast_queue_control(session->channel, AST_CONTROL_ANSWER);
1847                 break;
1848         default:
1849                 break;
1850         }
1851 }
1852
1853 static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
1854 {
1855         if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
1856                 if (session->endpoint->media.direct_media.enabled && session->channel) {
1857                         ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
1858                 }
1859         }
1860         return 0;
1861 }
1862
1863 static struct ast_custom_function chan_pjsip_dial_contacts_function = {
1864         .name = "PJSIP_DIAL_CONTACTS",
1865         .read = pjsip_acf_dial_contacts_read,
1866 };
1867
1868 static struct ast_custom_function media_offer_function = {
1869         .name = "PJSIP_MEDIA_OFFER",
1870         .read = pjsip_acf_media_offer_read,
1871         .write = pjsip_acf_media_offer_write
1872 };
1873
1874 /*!
1875  * \brief Load the module
1876  *
1877  * Module loading including tests for configuration or dependencies.
1878  * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
1879  * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
1880  * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
1881  * configuration file or other non-critical problem return
1882  * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
1883  */
1884 static int load_module(void)
1885 {
1886         if (!(chan_pjsip_tech.capabilities = ast_format_cap_alloc(0))) {
1887                 return AST_MODULE_LOAD_DECLINE;
1888         }
1889
1890         ast_format_cap_add_all_by_type(chan_pjsip_tech.capabilities, AST_FORMAT_TYPE_AUDIO);
1891
1892         ast_rtp_glue_register(&chan_pjsip_rtp_glue);
1893
1894         if (ast_channel_register(&chan_pjsip_tech)) {
1895                 ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
1896                 goto end;
1897         }
1898
1899         if (ast_custom_function_register(&chan_pjsip_dial_contacts_function)) {
1900                 ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n");
1901                 goto end;
1902         }
1903
1904         if (ast_custom_function_register(&media_offer_function)) {
1905                 ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
1906                 goto end;
1907         }
1908
1909         if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
1910                 ast_log(LOG_ERROR, "Unable to register PJSIP supplement\n");
1911                 goto end;
1912         }
1913
1914         if (ast_sip_session_register_supplement(&pbx_start_supplement)) {
1915                 ast_log(LOG_ERROR, "Unable to register PJSIP pbx start supplement\n");
1916                 ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
1917                 goto end;
1918         }
1919
1920         if (ast_sip_session_register_supplement(&chan_pjsip_ack_supplement)) {
1921                 ast_log(LOG_ERROR, "Unable to register PJSIP ACK supplement\n");
1922                 ast_sip_session_unregister_supplement(&pbx_start_supplement);
1923                 ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
1924                 goto end;
1925         }
1926
1927         return 0;
1928
1929 end:
1930         ast_custom_function_unregister(&media_offer_function);
1931         ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
1932         ast_channel_unregister(&chan_pjsip_tech);
1933         ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
1934
1935         return AST_MODULE_LOAD_FAILURE;
1936 }
1937
1938 /*! \brief Reload module */
1939 static int reload(void)
1940 {
1941         return -1;
1942 }
1943
1944 /*! \brief Unload the PJSIP channel from Asterisk */
1945 static int unload_module(void)
1946 {
1947         ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
1948         ast_sip_session_unregister_supplement(&pbx_start_supplement);
1949         ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
1950
1951         ast_custom_function_unregister(&media_offer_function);
1952         ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
1953
1954         ast_channel_unregister(&chan_pjsip_tech);
1955         ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
1956
1957         return 0;
1958 }
1959
1960 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Channel Driver",
1961                 .load = load_module,
1962                 .unload = unload_module,
1963                 .reload = reload,
1964                 .load_pri = AST_MODPRI_CHANNEL_DRIVER,
1965                );