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