channels/chan_pjsip.c

   1 /*
   2  * Asterisk -- An open source telephony toolkit.
   3  *
   4  * Copyright (C) 2013, Digium, Inc.
   5  *
   6  * Joshua Colp <jcolp@digium.com>
   7  *
   8  * See http://www.asterisk.org for more information about
   9  * the Asterisk project. Please do not directly contact
  10  * any of the maintainers of this project for assistance;
  11  * the project provides a web site, mailing lists and IRC
  12  * channels for your use.
  13  *
  14  * This program is free software, distributed under the terms of
  15  * the GNU General Public License Version 2. See the LICENSE file
  16  * at the top of the source tree.
  17  */
  18
  19 /*! \file
  20  *
  21  * \author Joshua Colp <jcolp@digium.com>
  22  *
  23  * \brief PSJIP SIP Channel Driver
  24  *
  25  * \ingroup channel_drivers
  26  */
  27
  28 /*** MODULEINFO
  29         <depend>pjproject</depend>
  30         <depend>res_pjsip</depend>
  31         <depend>res_pjsip_session</depend>
  32         <support_level>core</support_level>
  33  ***/
  34
  35 #include "asterisk.h"
  36
  37 #include <pjsip.h>
  38 #include <pjsip_ua.h>
  39 #include <pjlib.h>
  40
  41 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
  42
  43 #include "asterisk/lock.h"
  44 #include "asterisk/channel.h"
  45 #include "asterisk/module.h"
  46 #include "asterisk/pbx.h"
  47 #include "asterisk/rtp_engine.h"
  48 #include "asterisk/acl.h"
  49 #include "asterisk/callerid.h"
  50 #include "asterisk/file.h"
  51 #include "asterisk/cli.h"
  52 #include "asterisk/app.h"
  53 #include "asterisk/musiconhold.h"
  54 #include "asterisk/causes.h"
  55 #include "asterisk/taskprocessor.h"
  56 #include "asterisk/dsp.h"
  57 #include "asterisk/stasis_endpoints.h"
  58 #include "asterisk/stasis_channels.h"
  59 #include "asterisk/indications.h"
  60
  61 #include "asterisk/res_pjsip.h"
  62 #include "asterisk/res_pjsip_session.h"
  63
  64 /*** DOCUMENTATION
  65         <function name="PJSIP_DIAL_CONTACTS" language="en_US">
  66                 <synopsis>
  67                         Return a dial string for dialing all contacts on an AOR.
  68                 </synopsis>
  69                 <syntax>
  70                         <parameter name="endpoint" required="true">
  71                                 <para>Name of the endpoint</para>
  72                         </parameter>
  73                         <parameter name="aor" required="false">
  74                                 <para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
  75                         </parameter>
  76                         <parameter name="request_user" required="false">
  77                                 <para>Optional request user to use in the request URI</para>
  78                         </parameter>
  79                 </syntax>
  80                 <description>
  81                         <para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
  82                 </description>
  83         </function>
  84         <function name="PJSIP_MEDIA_OFFER" language="en_US">
  85                 <synopsis>
  86                         Media and codec offerings to be set on an outbound SIP channel prior to dialing.
  87                 </synopsis>
  88                 <syntax>
  89                         <parameter name="media" required="true">
  90                                 <para>types of media offered</para>
  91                         </parameter>
  92                 </syntax>
  93                 <description>
  94                         <para>Returns the codecs offered based upon the media choice</para>
  95                 </description>
  96         </function>
  97  ***/
  98
  99 static const char desc[] = "PJSIP Channel";
 100 static const char channel_type[] = "PJSIP";
 101
 102 static unsigned int chan_idx;
 103
 104 /*!
 105  * \brief Positions of various media
 106  */
 107 enum sip_session_media_position {
 108         /*! \brief First is audio */
 109         SIP_MEDIA_AUDIO = 0,
 110         /*! \brief Second is video */
 111         SIP_MEDIA_VIDEO,
 112         /*! \brief Last is the size for media details */
 113         SIP_MEDIA_SIZE,
 114 };
 115
 116 struct chan_pjsip_pvt {
 117         struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
 118 };
 119
 120 static void chan_pjsip_pvt_dtor(void *obj)
 121 {
 122         struct chan_pjsip_pvt *pvt = obj;
 123         int i;
 124
 125         for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
 126                 ao2_cleanup(pvt->media[i]);
 127                 pvt->media[i] = NULL;
 128         }
 129 }
 130
 131 /* \brief Asterisk core interaction functions */
 132 static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause);
 133 static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text);
 134 static int chan_pjsip_digit_begin(struct ast_channel *ast, char digit);
 135 static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
 136 static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout);
 137 static int chan_pjsip_hangup(struct ast_channel *ast);
 138 static int chan_pjsip_answer(struct ast_channel *ast);
 139 static struct ast_frame *chan_pjsip_read(struct ast_channel *ast);
 140 static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *f);
 141 static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
 142 static int chan_pjsip_transfer(struct ast_channel *ast, const char *target);
 143 static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
 144 static int chan_pjsip_devicestate(const char *data);
 145 static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
 146
 147 /*! \brief PBX interface structure for channel registration */
 148 static struct ast_channel_tech chan_pjsip_tech = {
 149         .type = channel_type,
 150         .description = "PJSIP Channel Driver",
 151         .requester = chan_pjsip_request,
 152         .send_text = chan_pjsip_sendtext,
 153         .send_digit_begin = chan_pjsip_digit_begin,
 154         .send_digit_end = chan_pjsip_digit_end,
 155         .call = chan_pjsip_call,
 156         .hangup = chan_pjsip_hangup,
 157         .answer = chan_pjsip_answer,
 158         .read = chan_pjsip_read,
 159         .write = chan_pjsip_write,
 160         .write_video = chan_pjsip_write,
 161         .exception = chan_pjsip_read,
 162         .indicate = chan_pjsip_indicate,
 163         .transfer = chan_pjsip_transfer,
 164         .fixup = chan_pjsip_fixup,
 165         .devicestate = chan_pjsip_devicestate,
 166         .queryoption = chan_pjsip_queryoption,
 167         .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
 168 };
 169
 170 /*! \brief SIP session interaction functions */
 171 static void chan_pjsip_session_begin(struct ast_sip_session *session);
 172 static void chan_pjsip_session_end(struct ast_sip_session *session);
 173 static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
 174 static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
 175
 176 /*! \brief SIP session supplement structure */
 177 static struct ast_sip_session_supplement chan_pjsip_supplement = {
 178         .method = "INVITE",
 179         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
 180         .session_begin = chan_pjsip_session_begin,
 181         .session_end = chan_pjsip_session_end,
 182         .incoming_request = chan_pjsip_incoming_request,
 183         .incoming_response = chan_pjsip_incoming_response,
 184 };
 185
 186 static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
 187
 188 static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
 189         .method = "ACK",
 190         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
 191         .incoming_request = chan_pjsip_incoming_ack,
 192 };
 193
 194 /*! \brief Dialplan function for constructing a dial string for calling all contacts */
 195 static int chan_pjsip_dial_contacts(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
 196 {
 197         RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
 198         RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
 199         const char *aor_name;
 200         char *rest;
 201
 202         AST_DECLARE_APP_ARGS(args,
 203                 AST_APP_ARG(endpoint_name);
 204                 AST_APP_ARG(aor_name);
 205                 AST_APP_ARG(request_user);
 206         );
 207
 208         AST_STANDARD_APP_ARGS(args, data);
 209
 210         if (ast_strlen_zero(args.endpoint_name)) {
 211                 ast_log(LOG_WARNING, "An endpoint name must be specified when using the '%s' dialplan function\n", cmd);
 212                 return -1;
 213         } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", args.endpoint_name))) {
 214                 ast_log(LOG_WARNING, "Specified endpoint '%s' was not found\n", args.endpoint_name);
 215                 return -1;
 216         }
 217
 218         aor_name = S_OR(args.aor_name, endpoint->aors);
 219
 220         if (ast_strlen_zero(aor_name)) {
 221                 ast_log(LOG_WARNING, "No AOR has been provided and no AORs are configured on endpoint '%s'\n", args.endpoint_name);
 222                 return -1;
 223         } else if (!(dial = ast_str_create(len))) {
 224                 ast_log(LOG_WARNING, "Could not get enough buffer space for dialing contacts\n");
 225                 return -1;
 226         } else if (!(rest = ast_strdupa(aor_name))) {
 227                 ast_log(LOG_WARNING, "Could not duplicate provided AORs\n");
 228                 return -1;
 229         }
 230
 231         while ((aor_name = strsep(&rest, ","))) {
 232                 RAII_VAR(struct ast_sip_aor *, aor, ast_sip_location_retrieve_aor(aor_name), ao2_cleanup);
 233                 RAII_VAR(struct ao2_container *, contacts, NULL, ao2_cleanup);
 234                 struct ao2_iterator it_contacts;
 235                 struct ast_sip_contact *contact;
 236
 237                 if (!aor) {
 238                         /* If the AOR provided is not found skip it, there may be more */
 239                         continue;
 240                 } else if (!(contacts = ast_sip_location_retrieve_aor_contacts(aor))) {
 241                         /* No contacts are available, skip it as well */
 242                         continue;
 243                 } else if (!ao2_container_count(contacts)) {
 244                         /* We were given a container but no contacts are in it... */
 245                         continue;
 246                 }
 247
 248                 it_contacts = ao2_iterator_init(contacts, 0);
 249                 for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
 250                         ast_str_append(&dial, -1, "PJSIP/");
 251
 252                         if (!ast_strlen_zero(args.request_user)) {
 253                                 ast_str_append(&dial, -1, "%s@", args.request_user);
 254                         }
 255                         ast_str_append(&dial, -1, "%s/%s&", args.endpoint_name, contact->uri);
 256                 }
 257                 ao2_iterator_destroy(&it_contacts);
 258         }
 259
 260         /* Trim the '&' at the end off */
 261         ast_str_truncate(dial, ast_str_strlen(dial) - 1);
 262
 263         ast_copy_string(buf, ast_str_buffer(dial), len);
 264
 265         return 0;
 266 }
 267
 268 static struct ast_custom_function chan_pjsip_dial_contacts_function = {
 269         .name = "PJSIP_DIAL_CONTACTS",
 270         .read = chan_pjsip_dial_contacts,
 271 };
 272
 273 static int media_offer_read_av(struct ast_sip_session *session, char *buf,
 274                                size_t len, enum ast_format_type media_type)
 275 {
 276         int i, size = 0;
 277         struct ast_format fmt;
 278         const char *name;
 279
 280         for (i = 0; ast_codec_pref_index(&session->override_prefs, i, &fmt); ++i) {
 281                 if (AST_FORMAT_GET_TYPE(fmt.id) != media_type) {
 282                         continue;
 283                 }
 284
 285                 name = ast_getformatname(&fmt);
 286
 287                 if (ast_strlen_zero(name)) {
 288                         ast_log(LOG_WARNING, "PJSIP_MEDIA_OFFER unrecognized format %s\n", name);
 289                         continue;
 290                 }
 291
 292                 /* add one since we'll include a comma */
 293                 size = strlen(name) + 1;
 294                 len -= size;
 295                 if ((len) < 0) {
 296                         break;
 297                 }
 298
 299                 /* no reason to use strncat here since we have already ensured buf has
 300                    enough space, so strcat can be safely used */
 301                 strcat(buf, name);
 302                 strcat(buf, ",");
 303         }
 304
 305         if (size) {
 306                 /* remove the extra comma */
 307                 buf[strlen(buf) - 1] = '\0';
 308         }
 309         return 0;
 310 }
 311
 312 struct media_offer_data {
 313         struct ast_sip_session *session;
 314         enum ast_format_type media_type;
 315         const char *value;
 316 };
 317
 318 static int media_offer_write_av(void *obj)
 319 {
 320         struct media_offer_data *data = obj;
 321         int i;
 322         struct ast_format fmt;
 323         /* remove all of the given media type first */
 324         for (i = 0; ast_codec_pref_index(&data->session->override_prefs, i, &fmt); ++i) {
 325                 if (AST_FORMAT_GET_TYPE(fmt.id) == data->media_type) {
 326                         ast_codec_pref_remove(&data->session->override_prefs, &fmt);
 327                 }
 328         }
 329         ast_format_cap_remove_bytype(data->session->req_caps, data->media_type);
 330         ast_parse_allow_disallow(&data->session->override_prefs, data->session->req_caps, data->value, 1);
 331
 332         return 0;
 333 }
 334
 335 static int media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
 336 {
 337         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 338
 339         if (!strcmp(data, "audio")) {
 340                 return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_AUDIO);
 341         } else if (!strcmp(data, "video")) {
 342                 return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_VIDEO);
 343         }
 344
 345         return 0;
 346 }
 347
 348 static int media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
 349 {
 350         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 351
 352         struct media_offer_data mdata = {
 353                 .session = channel->session,
 354                 .value = value
 355         };
 356
 357         if (!strcmp(data, "audio")) {
 358                 mdata.media_type = AST_FORMAT_TYPE_AUDIO;
 359         } else if (!strcmp(data, "video")) {
 360                 mdata.media_type = AST_FORMAT_TYPE_VIDEO;
 361         }
 362
 363         return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
 364 }
 365
 366 static struct ast_custom_function media_offer_function = {
 367         .name = "PJSIP_MEDIA_OFFER",
 368         .read = media_offer_read,
 369         .write = media_offer_write
 370 };
 371
 372 /*! \brief Function called by RTP engine to get local audio RTP peer */
 373 static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
 374 {
 375         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 376         struct chan_pjsip_pvt *pvt = channel->pvt;
 377         struct ast_sip_endpoint *endpoint;
 378
 379         if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
 380                 return AST_RTP_GLUE_RESULT_FORBID;
 381         }
 382
 383         endpoint = channel->session->endpoint;
 384
 385         *instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
 386         ao2_ref(*instance, +1);
 387
 388         ast_assert(endpoint != NULL);
 389         if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
 390                 return AST_RTP_GLUE_RESULT_FORBID;
 391         }
 392
 393         if (endpoint->media.direct_media.enabled) {
 394                 return AST_RTP_GLUE_RESULT_REMOTE;
 395         }
 396
 397         return AST_RTP_GLUE_RESULT_LOCAL;
 398 }
 399
 400 /*! \brief Function called by RTP engine to get local video RTP peer */
 401 static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
 402 {
 403         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 404         struct chan_pjsip_pvt *pvt = channel->pvt;
 405         struct ast_sip_endpoint *endpoint;
 406
 407         if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
 408                 return AST_RTP_GLUE_RESULT_FORBID;
 409         }
 410
 411         endpoint = channel->session->endpoint;
 412
 413         *instance = pvt->media[SIP_MEDIA_VIDEO]->rtp;
 414         ao2_ref(*instance, +1);
 415
 416         ast_assert(endpoint != NULL);
 417         if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
 418                 return AST_RTP_GLUE_RESULT_FORBID;
 419         }
 420
 421         return AST_RTP_GLUE_RESULT_LOCAL;
 422 }
 423
 424 /*! \brief Function called by RTP engine to get peer capabilities */
 425 static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
 426 {
 427         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 428
 429         ast_format_cap_copy(result, channel->session->endpoint->media.codecs);
 430 }
 431
 432 static int send_direct_media_request(void *data)
 433 {
 434         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
 435
 436         return ast_sip_session_refresh(session, NULL, NULL, NULL,
 437                         session->endpoint->media.direct_media.method, 1);
 438 }
 439
 440 static struct ast_datastore_info direct_media_mitigation_info = { };
 441
 442 static int direct_media_mitigate_glare(struct ast_sip_session *session)
 443 {
 444         RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
 445
 446         if (session->endpoint->media.direct_media.glare_mitigation ==
 447                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
 448                 return 0;
 449         }
 450
 451         datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation");
 452         if (!datastore) {
 453                 return 0;
 454         }
 455
 456         /* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */
 457         ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation");
 458
 459         if ((session->endpoint->media.direct_media.glare_mitigation ==
 460                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING &&
 461                         session->inv_session->role == PJSIP_ROLE_UAC) ||
 462                         (session->endpoint->media.direct_media.glare_mitigation ==
 463                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING &&
 464                         session->inv_session->role == PJSIP_ROLE_UAS)) {
 465                 return 1;
 466         }
 467
 468         return 0;
 469 }
 470
 471 static int check_for_rtp_changes(struct ast_channel *chan, struct ast_rtp_instance *rtp,
 472                 struct ast_sip_session_media *media, int rtcp_fd)
 473 {
 474         int changed = 0;
 475
 476         if (rtp) {
 477                 changed = ast_rtp_instance_get_and_cmp_remote_address(rtp, &media->direct_media_addr);
 478                 if (media->rtp) {
 479                         ast_channel_set_fd(chan, rtcp_fd, -1);
 480                         ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 0);
 481                 }
 482         } else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
 483                 ast_sockaddr_setnull(&media->direct_media_addr);
 484                 changed = 1;
 485                 if (media->rtp) {
 486                         ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1);
 487                         ast_channel_set_fd(chan, rtcp_fd, ast_rtp_instance_fd(media->rtp, 1));
 488                 }
 489         }
 490
 491         return changed;
 492 }
 493
 494 /*! \brief Function called by RTP engine to change where the remote party should send media */
 495 static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
 496                 struct ast_rtp_instance *rtp,
 497                 struct ast_rtp_instance *vrtp,
 498                 struct ast_rtp_instance *tpeer,
 499                 const struct ast_format_cap *cap,
 500                 int nat_active)
 501 {
 502         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 503         struct chan_pjsip_pvt *pvt = channel->pvt;
 504         struct ast_sip_session *session = channel->session;
 505         int changed = 0;
 506         struct ast_channel *bridge_peer;
 507
 508         /* Don't try to do any direct media shenanigans on early bridges */
 509         bridge_peer = ast_channel_bridge_peer(chan);
 510         if ((rtp || vrtp || tpeer) && !bridge_peer) {
 511                 return 0;
 512         }
 513         ast_channel_cleanup(bridge_peer);
 514
 515         if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
 516                 return 0;
 517         }
 518
 519         if (pvt->media[SIP_MEDIA_AUDIO]) {
 520                 changed |= check_for_rtp_changes(chan, rtp, pvt->media[SIP_MEDIA_AUDIO], 1);
 521         }
 522         if (pvt->media[SIP_MEDIA_VIDEO]) {
 523                 changed |= check_for_rtp_changes(chan, vrtp, pvt->media[SIP_MEDIA_VIDEO], 3);
 524         }
 525
 526         if (direct_media_mitigate_glare(session)) {
 527                 return 0;
 528         }
 529
 530         if (cap && !ast_format_cap_is_empty(cap) && !ast_format_cap_identical(session->direct_media_cap, cap)) {
 531                 ast_format_cap_copy(session->direct_media_cap, cap);
 532                 changed = 1;
 533         }
 534
 535         if (changed) {
 536                 ao2_ref(session, +1);
 537
 538
 539                 if (ast_sip_push_task(session->serializer, send_direct_media_request, session)) {
 540                         ao2_cleanup(session);
 541                 }
 542         }
 543
 544         return 0;
 545 }
 546
 547 /*! \brief Local glue for interacting with the RTP engine core */
 548 static struct ast_rtp_glue chan_pjsip_rtp_glue = {
 549         .type = "PJSIP",
 550         .get_rtp_info = chan_pjsip_get_rtp_peer,
 551         .get_vrtp_info = chan_pjsip_get_vrtp_peer,
 552         .get_codec = chan_pjsip_get_codec,
 553         .update_peer = chan_pjsip_set_rtp_peer,
 554 };
 555
 556 /*! \brief Function called to create a new PJSIP Asterisk channel */
 557 static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const char *linkedid, const char *cid_name)
 558 {
 559         struct ast_channel *chan;
 560         struct ast_format fmt;
 561         RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
 562         struct ast_sip_channel_pvt *channel;
 563
 564         if (!(pvt = ao2_alloc(sizeof(*pvt), chan_pjsip_pvt_dtor))) {
 565                 return NULL;
 566         }
 567
 568         if (!(chan = ast_channel_alloc(1, state, S_OR(session->id.number.str, ""), S_OR(session->id.name.str, ""), "", "", "", linkedid, 0, "PJSIP/%s-%08x", ast_sorcery_object_get_id(session->endpoint),
 569                 ast_atomic_fetchadd_int((int *)&chan_idx, +1)))) {
 570                 return NULL;
 571         }
 572
 573         ast_channel_tech_set(chan, &chan_pjsip_tech);
 574
 575         if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
 576                 ast_hangup(chan);
 577                 return NULL;
 578         }
 579
 580         ast_channel_stage_snapshot(chan);
 581
 582         /* If res_pjsip_session is ever updated to create/destroy ast_sip_session_media
 583          * during a call such as if multiple same-type stream support is introduced,
 584          * these will need to be recaptured as well */
 585         pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
 586         pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
 587         ast_channel_tech_pvt_set(chan, channel);
 588         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
 589                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(chan));
 590         }
 591         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
 592                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(chan));
 593         }
 594
 595         if (ast_format_cap_is_empty(session->req_caps) || !ast_format_cap_has_joint(session->req_caps, session->endpoint->media.codecs)) {
 596                 ast_format_cap_copy(ast_channel_nativeformats(chan), session->endpoint->media.codecs);
 597         } else {
 598                 ast_format_cap_copy(ast_channel_nativeformats(chan), session->req_caps);
 599         }
 600
 601         ast_codec_choose(&session->endpoint->media.prefs, ast_channel_nativeformats(chan), 1, &fmt);
 602         ast_format_copy(ast_channel_writeformat(chan), &fmt);
 603         ast_format_copy(ast_channel_rawwriteformat(chan), &fmt);
 604         ast_format_copy(ast_channel_readformat(chan), &fmt);
 605         ast_format_copy(ast_channel_rawreadformat(chan), &fmt);
 606
 607         if (state == AST_STATE_RING) {
 608                 ast_channel_rings_set(chan, 1);
 609         }
 610
 611         ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
 612
 613         ast_channel_context_set(chan, session->endpoint->context);
 614         ast_channel_exten_set(chan, S_OR(exten, "s"));
 615         ast_channel_priority_set(chan, 1);
 616
 617         ast_channel_callgroup_set(chan, session->endpoint->pickup.callgroup);
 618         ast_channel_pickupgroup_set(chan, session->endpoint->pickup.pickupgroup);
 619
 620         ast_channel_named_callgroups_set(chan, session->endpoint->pickup.named_callgroups);
 621         ast_channel_named_pickupgroups_set(chan, session->endpoint->pickup.named_pickupgroups);
 622
 623         if (!ast_strlen_zero(session->endpoint->language)) {
 624                 ast_channel_language_set(chan, session->endpoint->language);
 625         }
 626
 627         if (!ast_strlen_zero(session->endpoint->zone)) {
 628                 struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone);
 629                 if (!zone) {
 630                         ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone);
 631                 }
 632                 ast_channel_zone_set(chan, zone);
 633         }
 634
 635         ast_endpoint_add_channel(session->endpoint->persistent, chan);
 636
 637         ast_channel_stage_snapshot_done(chan);
 638
 639         return chan;
 640 }
 641
 642 static int answer(void *data)
 643 {
 644         pj_status_t status = PJ_SUCCESS;
 645         pjsip_tx_data *packet;
 646         struct ast_sip_session *session = data;
 647
 648         pjsip_dlg_inc_lock(session->inv_session->dlg);
 649         if (session->inv_session->invite_tsx) {
 650                 status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet);
 651         }
 652         pjsip_dlg_dec_lock(session->inv_session->dlg);
 653
 654         if (status == PJ_SUCCESS && packet) {
 655                 ast_sip_session_send_response(session, packet);
 656         }
 657
 658         ao2_ref(session, -1);
 659
 660         return (status == PJ_SUCCESS) ? 0 : -1;
 661 }
 662
 663 /*! \brief Function called by core when we should answer a PJSIP session */
 664 static int chan_pjsip_answer(struct ast_channel *ast)
 665 {
 666         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 667
 668         if (ast_channel_state(ast) == AST_STATE_UP) {
 669                 return 0;
 670         }
 671
 672         ast_setstate(ast, AST_STATE_UP);
 673
 674         ao2_ref(channel->session, +1);
 675         if (ast_sip_push_task(channel->session->serializer, answer, channel->session)) {
 676                 ast_log(LOG_WARNING, "Unable to push answer task to the threadpool. Cannot answer call\n");
 677                 ao2_cleanup(channel->session);
 678                 return -1;
 679         }
 680
 681         return 0;
 682 }
 683
 684 /*! \brief Internal helper function called when CNG tone is detected */
 685 static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f)
 686 {
 687         const char *target_context;
 688         int exists;
 689
 690         /* If we only needed this DSP for fax detection purposes we can just drop it now */
 691         if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
 692                 ast_dsp_set_features(session->dsp, DSP_FEATURE_DIGIT_DETECT);
 693         } else {
 694                 ast_dsp_free(session->dsp);
 695                 session->dsp = NULL;
 696         }
 697
 698         /* If already executing in the fax extension don't do anything */
 699         if (!strcmp(ast_channel_exten(session->channel), "fax")) {
 700                 return f;
 701         }
 702
 703         target_context = S_OR(ast_channel_macrocontext(session->channel), ast_channel_context(session->channel));
 704
 705         /* We need to unlock the channel here because ast_exists_extension has the
 706          * potential to start and stop an autoservice on the channel. Such action
 707          * is prone to deadlock if the channel is locked.
 708          */
 709         ast_channel_unlock(session->channel);
 710         exists = ast_exists_extension(session->channel, target_context, "fax", 1,
 711                 S_COR(ast_channel_caller(session->channel)->id.number.valid,
 712                         ast_channel_caller(session->channel)->id.number.str, NULL));
 713         ast_channel_lock(session->channel);
 714
 715         if (exists) {
 716                 ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
 717                         ast_channel_name(session->channel));
 718                 pbx_builtin_setvar_helper(session->channel, "FAXEXTEN", ast_channel_exten(session->channel));
 719                 if (ast_async_goto(session->channel, target_context, "fax", 1)) {
 720                         ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
 721                                 ast_channel_name(session->channel), target_context);
 722                 }
 723                 ast_frfree(f);
 724                 f = &ast_null_frame;
 725         } else {
 726                 ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
 727                         ast_channel_name(session->channel), target_context);
 728         }
 729
 730         return f;
 731 }
 732
 733 /*! \brief Function called by core to read any waiting frames */
 734 static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
 735 {
 736         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 737         struct chan_pjsip_pvt *pvt = channel->pvt;
 738         struct ast_frame *f;
 739         struct ast_sip_session_media *media = NULL;
 740         int rtcp = 0;
 741         int fdno = ast_channel_fdno(ast);
 742
 743         switch (fdno) {
 744         case 0:
 745                 media = pvt->media[SIP_MEDIA_AUDIO];
 746                 break;
 747         case 1:
 748                 media = pvt->media[SIP_MEDIA_AUDIO];
 749                 rtcp = 1;
 750                 break;
 751         case 2:
 752                 media = pvt->media[SIP_MEDIA_VIDEO];
 753                 break;
 754         case 3:
 755                 media = pvt->media[SIP_MEDIA_VIDEO];
 756                 rtcp = 1;
 757                 break;
 758         }
 759
 760         if (!media || !media->rtp) {
 761                 return &ast_null_frame;
 762         }
 763
 764         if (!(f = ast_rtp_instance_read(media->rtp, rtcp))) {
 765                 return f;
 766         }
 767
 768         if (f->frametype != AST_FRAME_VOICE) {
 769                 return f;
 770         }
 771
 772         if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &f->subclass.format))) {
 773                 ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(&f->subclass.format));
 774                 ast_format_cap_set(ast_channel_nativeformats(ast), &f->subclass.format);
 775                 ast_set_read_format(ast, ast_channel_readformat(ast));
 776                 ast_set_write_format(ast, ast_channel_writeformat(ast));
 777         }
 778
 779         if (channel->session->dsp) {
 780                 f = ast_dsp_process(ast, channel->session->dsp, f);
 781
 782                 if (f && (f->frametype == AST_FRAME_DTMF)) {
 783                         if (f->subclass.integer == 'f') {
 784                                 ast_debug(3, "Fax CNG detected on %s\n", ast_channel_name(ast));
 785                                 f = chan_pjsip_cng_tone_detected(channel->session, f);
 786                         } else {
 787                                 ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
 788                                         ast_channel_name(ast));
 789                         }
 790                 }
 791         }
 792
 793         return f;
 794 }
 795
 796 /*! \brief Function called by core to write frames */
 797 static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame)
 798 {
 799         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 800         struct chan_pjsip_pvt *pvt = channel->pvt;
 801         struct ast_sip_session_media *media;
 802         int res = 0;
 803
 804         switch (frame->frametype) {
 805         case AST_FRAME_VOICE:
 806                 media = pvt->media[SIP_MEDIA_AUDIO];
 807
 808                 if (!media) {
 809                         return 0;
 810                 }
 811                 if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) {
 812                         char buf[256];
 813
 814                         ast_log(LOG_WARNING,
 815                                 "Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
 816                                 ast_getformatname(&frame->subclass.format),
 817                                 ast_getformatname_multiple(buf, sizeof(buf), ast_channel_nativeformats(ast)),
 818                                 ast_getformatname(ast_channel_readformat(ast)),
 819                                 ast_getformatname(ast_channel_writeformat(ast)));
 820                         return 0;
 821                 }
 822                 if (media->rtp) {
 823                         res = ast_rtp_instance_write(media->rtp, frame);
 824                 }
 825                 break;
 826         case AST_FRAME_VIDEO:
 827                 if ((media = pvt->media[SIP_MEDIA_VIDEO]) && media->rtp) {
 828                         res = ast_rtp_instance_write(media->rtp, frame);
 829                 }
 830                 break;
 831         case AST_FRAME_MODEM:
 832                 break;
 833         default:
 834                 ast_log(LOG_WARNING, "Can't send %d type frames with PJSIP\n", frame->frametype);
 835                 break;
 836         }
 837
 838         return res;
 839 }
 840
 841 struct fixup_data {
 842         struct ast_sip_session *session;
 843         struct ast_channel *chan;
 844 };
 845
 846 static int fixup(void *data)
 847 {
 848         struct fixup_data *fix_data = data;
 849         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(fix_data->chan);
 850         struct chan_pjsip_pvt *pvt = channel->pvt;
 851
 852         channel->session->channel = fix_data->chan;
 853         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
 854                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(fix_data->chan));
 855         }
 856         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
 857                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(fix_data->chan));
 858         }
 859
 860         return 0;
 861 }
 862
 863 /*! \brief Function called by core to change the underlying owner channel */
 864 static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
 865 {
 866         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
 867         struct fixup_data fix_data;
 868
 869         fix_data.session = channel->session;
 870         fix_data.chan = newchan;
 871
 872         if (channel->session->channel != oldchan) {
 873                 return -1;
 874         }
 875
 876         if (ast_sip_push_task_synchronous(channel->session->serializer, fixup, &fix_data)) {
 877                 ast_log(LOG_WARNING, "Unable to perform channel fixup\n");
 878                 return -1;
 879         }
 880
 881         return 0;
 882 }
 883
 884 /*! \brief Function called to get the device state of an endpoint */
 885 static int chan_pjsip_devicestate(const char *data)
 886 {
 887         RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
 888         enum ast_device_state state = AST_DEVICE_UNKNOWN;
 889         RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
 890         RAII_VAR(struct stasis_cache *, cache, NULL, ao2_cleanup);
 891         struct ast_devstate_aggregate aggregate;
 892         int num, inuse = 0;
 893
 894         if (!endpoint) {
 895                 return AST_DEVICE_INVALID;
 896         }
 897
 898         endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent),
 899                 ast_endpoint_get_resource(endpoint->persistent));
 900
 901         if (!endpoint_snapshot) {
 902                 return AST_DEVICE_INVALID;
 903         }
 904
 905         if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) {
 906                 state = AST_DEVICE_UNAVAILABLE;
 907         } else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) {
 908                 state = AST_DEVICE_NOT_INUSE;
 909         }
 910
 911         if (!endpoint_snapshot->num_channels || !(cache = ast_channel_cache())) {
 912                 return state;
 913         }
 914
 915         ast_devstate_aggregate_init(&aggregate);
 916
 917         ao2_ref(cache, +1);
 918
 919         for (num = 0; num < endpoint_snapshot->num_channels; num++) {
 920                 RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
 921                 struct ast_channel_snapshot *snapshot;
 922
 923                 msg = stasis_cache_get(cache, ast_channel_snapshot_type(),
 924                         endpoint_snapshot->channel_ids[num]);
 925
 926                 if (!msg) {
 927                         continue;
 928                 }
 929
 930                 snapshot = stasis_message_data(msg);
 931
 932                 if (snapshot->state == AST_STATE_DOWN) {
 933                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_NOT_INUSE);
 934                 } else if (snapshot->state == AST_STATE_RINGING) {
 935                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_RINGING);
 936                 } else if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
 937                         (snapshot->state == AST_STATE_BUSY)) {
 938                         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_INUSE);
 939                         inuse++;
 940                 }
 941         }
 942
 943         if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) {
 944                 state = AST_DEVICE_BUSY;
 945         } else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) {
 946                 state = ast_devstate_aggregate_result(&aggregate);
 947         }
 948
 949         return state;
 950 }
 951
 952 /*! \brief Function called to query options on a channel */
 953 static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen)
 954 {
 955         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 956         struct ast_sip_session *session = channel->session;
 957         int res = -1;
 958         enum ast_sip_session_t38state state = T38_STATE_UNAVAILABLE;
 959
 960         switch (option) {
 961         case AST_OPTION_T38_STATE:
 962                 if (session->endpoint->media.t38.enabled) {
 963                         switch (session->t38state) {
 964                         case T38_LOCAL_REINVITE:
 965                         case T38_PEER_REINVITE:
 966                                 state = T38_STATE_NEGOTIATING;
 967                                 break;
 968                         case T38_ENABLED:
 969                                 state = T38_STATE_NEGOTIATED;
 970                                 break;
 971                         case T38_REJECTED:
 972                                 state = T38_STATE_REJECTED;
 973                                 break;
 974                         default:
 975                                 state = T38_STATE_UNKNOWN;
 976                                 break;
 977                         }
 978                 }
 979
 980                 *((enum ast_t38_state *) data) = state;
 981                 res = 0;
 982
 983                 break;
 984         default:
 985                 break;
 986         }
 987
 988         return res;
 989 }
 990
 991 struct indicate_data {
 992         struct ast_sip_session *session;
 993         int condition;
 994         int response_code;
 995         void *frame_data;
 996         size_t datalen;
 997 };
 998
 999 static void indicate_data_destroy(void *obj)
1000 {
1001         struct indicate_data *ind_data = obj;
1002
1003         ast_free(ind_data->frame_data);
1004         ao2_ref(ind_data->session, -1);
1005 }
1006
1007 static struct indicate_data *indicate_data_alloc(struct ast_sip_session *session,
1008                 int condition, int response_code, const void *frame_data, size_t datalen)
1009 {
1010         struct indicate_data *ind_data = ao2_alloc(sizeof(*ind_data), indicate_data_destroy);
1011
1012         if (!ind_data) {
1013                 return NULL;
1014         }
1015
1016         ind_data->frame_data = ast_malloc(datalen);
1017         if (!ind_data->frame_data) {
1018                 ao2_ref(ind_data, -1);
1019                 return NULL;
1020         }
1021
1022         memcpy(ind_data->frame_data, frame_data, datalen);
1023         ind_data->datalen = datalen;
1024         ind_data->condition = condition;
1025         ind_data->response_code = response_code;
1026         ao2_ref(session, +1);
1027         ind_data->session = session;
1028
1029         return ind_data;
1030 }
1031
1032 static int indicate(void *data)
1033 {
1034         pjsip_tx_data *packet = NULL;
1035         struct indicate_data *ind_data = data;
1036         struct ast_sip_session *session = ind_data->session;
1037         int response_code = ind_data->response_code;
1038
1039         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
1040                 ast_sip_session_send_response(session, packet);
1041         }
1042
1043         ao2_ref(ind_data, -1);
1044
1045         return 0;
1046 }
1047
1048 /*! \brief Send SIP INFO with video update request */
1049 static int transmit_info_with_vidupdate(void *data)
1050 {
1051         const char * xml =
1052                 "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
1053                 " <media_control>\r\n"
1054                 "  <vc_primitive>\r\n"
1055                 "   <to_encoder>\r\n"
1056                 "    <picture_fast_update/>\r\n"
1057                 "   </to_encoder>\r\n"
1058                 "  </vc_primitive>\r\n"
1059                 " </media_control>\r\n";
1060
1061         const struct ast_sip_body body = {
1062                 .type = "application",
1063                 .subtype = "media_control+xml",
1064                 .body_text = xml
1065         };
1066
1067         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
1068         struct pjsip_tx_data *tdata;
1069
1070         if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
1071                 ast_log(LOG_ERROR, "Could not create text video update INFO request\n");
1072                 return -1;
1073         }
1074         if (ast_sip_add_body(tdata, &body)) {
1075                 ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n");
1076                 return -1;
1077         }
1078         ast_sip_session_send_request(session, tdata);
1079
1080         return 0;
1081 }
1082
1083 /*! \brief Update connected line information */
1084 static int update_connected_line_information(void *data)
1085 {
1086         RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
1087         struct ast_party_id connected_id;
1088
1089         if ((ast_channel_state(session->channel) != AST_STATE_UP) && (session->inv_session->role == PJSIP_UAS_ROLE)) {
1090                 int response_code = 0;
1091
1092                 if (ast_channel_state(session->channel) == AST_STATE_RING) {
1093                         response_code = !session->endpoint->inband_progress ? 180 : 183;
1094                 } else if (ast_channel_state(session->channel) == AST_STATE_RINGING) {
1095                         response_code = 183;
1096                 }
1097
1098                 if (response_code) {
1099                         struct pjsip_tx_data *packet = NULL;
1100
1101                         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
1102                                 ast_sip_session_send_response(session, packet);
1103                         }
1104                 }
1105         } else {
1106                 enum ast_sip_session_refresh_method method = session->endpoint->id.refresh_method;
1107
1108                 if (session->inv_session->invite_tsx && (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
1109                         method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
1110                 }
1111
1112                 connected_id = ast_channel_connected_effective_id(session->channel);
1113                 if ((session->endpoint->id.send_pai || session->endpoint->id.send_rpid) &&
1114                     (session->endpoint->id.trust_outbound ||
1115                      ((connected_id.name.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED &&
1116                       (connected_id.number.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED))) {
1117                         ast_sip_session_refresh(session, NULL, NULL, NULL, method, 1);
1118                 }
1119         }
1120
1121         return 0;
1122 }
1123
1124 /*! \brief Function called by core to ask the channel to indicate some sort of condition */
1125 static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
1126 {
1127         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1128         struct chan_pjsip_pvt *pvt = channel->pvt;
1129         struct ast_sip_session_media *media;
1130         int response_code = 0;
1131         int res = 0;
1132
1133         switch (condition) {
1134         case AST_CONTROL_RINGING:
1135                 if (ast_channel_state(ast) == AST_STATE_RING) {
1136                         if (channel->session->endpoint->inband_progress) {
1137                                 response_code = 183;
1138                                 res = -1;
1139                         } else {
1140                                 response_code = 180;
1141                         }
1142                 } else {
1143                         res = -1;
1144                 }
1145                 ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
1146                 break;
1147         case AST_CONTROL_BUSY:
1148                 if (ast_channel_state(ast) != AST_STATE_UP) {
1149                         response_code = 486;
1150                 } else {
1151                         res = -1;
1152                 }
1153                 break;
1154         case AST_CONTROL_CONGESTION:
1155                 if (ast_channel_state(ast) != AST_STATE_UP) {
1156                         response_code = 503;
1157                 } else {
1158                         res = -1;
1159                 }
1160                 break;
1161         case AST_CONTROL_INCOMPLETE:
1162                 if (ast_channel_state(ast) != AST_STATE_UP) {
1163                         response_code = 484;
1164                 } else {
1165                         res = -1;
1166                 }
1167                 break;
1168         case AST_CONTROL_PROCEEDING:
1169                 if (ast_channel_state(ast) != AST_STATE_UP) {
1170                         response_code = 100;
1171                 } else {
1172                         res = -1;
1173                 }
1174                 break;
1175         case AST_CONTROL_PROGRESS:
1176                 if (ast_channel_state(ast) != AST_STATE_UP) {
1177                         response_code = 183;
1178                 } else {
1179                         res = -1;
1180                 }
1181                 break;
1182         case AST_CONTROL_VIDUPDATE:
1183                 media = pvt->media[SIP_MEDIA_VIDEO];
1184                 if (media && media->rtp) {
1185                         /* FIXME: Only use this for VP8. Additional work would have to be done to
1186                          * fully support other video codecs */
1187                         struct ast_format_cap *fcap = ast_channel_nativeformats(ast);
1188                         struct ast_format vp8;
1189                         ast_format_set(&vp8, AST_FORMAT_VP8, 0);
1190                         if (ast_format_cap_iscompatible(fcap, &vp8)) {
1191                                 /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
1192                                  * RTP engine would provide a way to externally write/schedule RTCP
1193                                  * packets */
1194                                 struct ast_frame fr;
1195                                 fr.frametype = AST_FRAME_CONTROL;
1196                                 fr.subclass.integer = AST_CONTROL_VIDUPDATE;
1197                                 res = ast_rtp_instance_write(media->rtp, &fr);
1198                         } else {
1199                                 ao2_ref(channel->session, +1);
1200
1201                                 if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
1202                                         ao2_cleanup(channel->session);
1203                                 }
1204                         }
1205                 } else {
1206                         res = -1;
1207                 }
1208                 break;
1209         case AST_CONTROL_CONNECTED_LINE:
1210                 ao2_ref(channel->session, +1);
1211                 if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
1212                         ao2_cleanup(channel->session);
1213                 }
1214                 break;
1215         case AST_CONTROL_UPDATE_RTP_PEER:
1216                 break;
1217         case AST_CONTROL_PVT_CAUSE_CODE:
1218                 res = -1;
1219                 break;
1220         case AST_CONTROL_HOLD:
1221                 ast_moh_start(ast, data, NULL);
1222                 break;
1223         case AST_CONTROL_UNHOLD:
1224                 ast_moh_stop(ast);
1225                 break;
1226         case AST_CONTROL_SRCUPDATE:
1227                 break;
1228         case AST_CONTROL_SRCCHANGE:
1229                 break;
1230         case AST_CONTROL_REDIRECTING:
1231                 if (ast_channel_state(ast) != AST_STATE_UP) {
1232                         response_code = 181;
1233                 } else {
1234                         res = -1;
1235                 }
1236                 break;
1237         case AST_CONTROL_T38_PARAMETERS:
1238                 res = 0;
1239
1240                 if (channel->session->t38state == T38_PEER_REINVITE) {
1241                         const struct ast_control_t38_parameters *parameters = data;
1242
1243                         if (parameters->request_response == AST_T38_REQUEST_PARMS) {
1244                                 res = AST_T38_REQUEST_PARMS;
1245                         }
1246                 }
1247
1248                 break;
1249         case -1:
1250                 res = -1;
1251                 break;
1252         default:
1253                 ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
1254                 res = -1;
1255                 break;
1256         }
1257
1258         if (response_code) {
1259                 struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
1260                 if (!ind_data || ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
1261                         ast_log(LOG_NOTICE, "Cannot send response code %d to endpoint %s. Could not queue task properly\n",
1262                                         response_code, ast_sorcery_object_get_id(channel->session->endpoint));
1263                         ao2_cleanup(ind_data);
1264                         res = -1;
1265                 }
1266         }
1267
1268         return res;
1269 }
1270
1271 struct transfer_data {
1272         struct ast_sip_session *session;
1273         char *target;
1274 };
1275
1276 static void transfer_data_destroy(void *obj)
1277 {
1278         struct transfer_data *trnf_data = obj;
1279
1280         ast_free(trnf_data->target);
1281         ao2_cleanup(trnf_data->session);
1282 }
1283
1284 static struct transfer_data *transfer_data_alloc(struct ast_sip_session *session, const char *target)
1285 {
1286         struct transfer_data *trnf_data = ao2_alloc(sizeof(*trnf_data), transfer_data_destroy);
1287
1288         if (!trnf_data) {
1289                 return NULL;
1290         }
1291
1292         if (!(trnf_data->target = ast_strdup(target))) {
1293                 ao2_ref(trnf_data, -1);
1294                 return NULL;
1295         }
1296
1297         ao2_ref(session, +1);
1298         trnf_data->session = session;
1299
1300         return trnf_data;
1301 }
1302
1303 static void transfer_redirect(struct ast_sip_session *session, const char *target)
1304 {
1305         pjsip_tx_data *packet;
1306         enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1307         pjsip_contact_hdr *contact;
1308         pj_str_t tmp;
1309
1310         if (pjsip_inv_end_session(session->inv_session, 302, NULL, &packet) != PJ_SUCCESS) {
1311                 message = AST_TRANSFER_FAILED;
1312                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1313
1314                 return;
1315         }
1316
1317         if (!(contact = pjsip_msg_find_hdr(packet->msg, PJSIP_H_CONTACT, NULL))) {
1318                 contact = pjsip_contact_hdr_create(packet->pool);
1319         }
1320
1321         pj_strdup2_with_null(packet->pool, &tmp, target);
1322         if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) {
1323                 message = AST_TRANSFER_FAILED;
1324                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1325                 pjsip_tx_data_dec_ref(packet);
1326
1327                 return;
1328         }
1329         pjsip_msg_add_hdr(packet->msg, (pjsip_hdr *) contact);
1330
1331         ast_sip_session_send_response(session, packet);
1332         ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1333 }
1334
1335 static void transfer_refer(struct ast_sip_session *session, const char *target)
1336 {
1337         pjsip_evsub *sub;
1338         enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1339         pj_str_t tmp;
1340         pjsip_tx_data *packet;
1341
1342         if (pjsip_xfer_create_uac(session->inv_session->dlg, NULL, &sub) != PJ_SUCCESS) {
1343                 message = AST_TRANSFER_FAILED;
1344                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1345
1346                 return;
1347         }
1348
1349         if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
1350                 message = AST_TRANSFER_FAILED;
1351                 ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1352                 pjsip_evsub_terminate(sub, PJ_FALSE);
1353
1354                 return;
1355         }
1356
1357         pjsip_xfer_send_request(sub, packet);
1358         ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
1359 }
1360
1361 static int transfer(void *data)
1362 {
1363         struct transfer_data *trnf_data = data;
1364
1365         if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
1366                 transfer_redirect(trnf_data->session, trnf_data->target);
1367         } else {
1368                 transfer_refer(trnf_data->session, trnf_data->target);
1369         }
1370
1371         ao2_ref(trnf_data, -1);
1372         return 0;
1373 }
1374
1375 /*! \brief Function called by core for Asterisk initiated transfer */
1376 static int chan_pjsip_transfer(struct ast_channel *chan, const char *target)
1377 {
1378         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
1379         struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
1380
1381         if (!trnf_data) {
1382                 return -1;
1383         }
1384
1385         if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
1386                 ast_log(LOG_WARNING, "Error requesting transfer\n");
1387                 ao2_cleanup(trnf_data);
1388                 return -1;
1389         }
1390
1391         return 0;
1392 }
1393
1394 /*! \brief Function called by core to start a DTMF digit */
1395 static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
1396 {
1397         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
1398         struct chan_pjsip_pvt *pvt = channel->pvt;
1399         struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
1400         int res = 0;
1401
1402         switch (channel->session->endpoint->dtmf) {
1403         case AST_SIP_DTMF_RFC_4733:
1404                 if (!media || !media->rtp) {
1405                         return -1;
1406                 }
1407
1408                 ast_rtp_instance_dtmf_begin(media->rtp, digit);
1409         case AST_SIP_DTMF_NONE:
1410                 break;
1411         case AST_SIP_DTMF_INBAND:
1412                 res = -1;
1413                 break;
1414         default:
1415                 break;
1416         }
1417
1418         return res;
1419 }
1420
1421 struct info_dtmf_data {
1422         struct ast_sip_session *session;
1423         char digit;
1424         unsigned int duration;
1425 };
1426
1427 static void info_dtmf_data_destroy(void *obj)
1428 {
1429         struct info_dtmf_data *dtmf_data = obj;
1430         ao2_ref(dtmf_data->session, -1);
1431 }
1432
1433 static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
1434 {
1435         struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy);
1436         if (!dtmf_data) {
1437                 return NULL;
1438         }
1439         ao2_ref(session, +1);
1440         dtmf_data->session = session;
1441         dtmf_data->digit = digit;
1442         dtmf_data->duration = duration;
1443         return dtmf_data;
1444 }
1445
1446 static int transmit_info_dtmf(void *data)
1447 {
1448         RAII_VAR(struct info_dtmf_data *, dtmf_data, data, ao2_cleanup);
1449
1450         struct ast_sip_session *session = dtmf_data->session;
1451         struct pjsip_tx_data *tdata;
1452
1453         RAII_VAR(struct ast_str *, body_text, NULL, ast_free_ptr);
1454
1455         struct ast_sip_body body = {
1456                 .type = "application",
1457                 .subtype = "dtmf-relay",
1458         };
1459
1460         if (!(body_text = ast_str_create(32))) {
1461                 ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n");
1462                 return -1;
1463         }
1464         ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration);
1465
1466         body.body_text = ast_str_buffer(body_text);
1467
1468         if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
1469                 ast_log(LOG_ERROR, "Could not create DTMF INFO request\n");
1470                 return -1;
1471         }
1472         if (ast_sip_add_body(tdata, &body)) {
1473                 ast_log(LOG_ERROR, "Could not add body to DTMF INFO request\n");
1474                 pjsip_tx_data_dec_ref(tdata);
1475                 return -1;
1476         }
1477         ast_sip_session_send_request(session, tdata);
1478
1479         return 0;
1480 }
1481
1482 /*! \brief Function called by core to stop a DTMF digit */
1483 static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
1484 {
1485         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1486         struct chan_pjsip_pvt *pvt = channel->pvt;
1487         struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
1488         int res = 0;
1489
1490         switch (channel->session->endpoint->dtmf) {
1491         case AST_SIP_DTMF_INFO:
1492         {
1493                 struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
1494
1495                 if (!dtmf_data) {
1496                         return -1;
1497                 }
1498
1499                 if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
1500                         ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
1501                         ao2_cleanup(dtmf_data);
1502                         return -1;
1503                 }
1504                 break;
1505         }
1506         case AST_SIP_DTMF_RFC_4733:
1507                 if (!media || !media->rtp) {
1508                         return -1;
1509                 }
1510
1511                 ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
1512         case AST_SIP_DTMF_NONE:
1513                 break;
1514         case AST_SIP_DTMF_INBAND:
1515                 res = -1;
1516                 break;
1517         }
1518
1519         return res;
1520 }
1521
1522 static int call(void *data)
1523 {
1524         struct ast_sip_session *session = data;
1525         pjsip_tx_data *tdata;
1526
1527         int res = ast_sip_session_create_invite(session, &tdata);
1528
1529         if (res) {
1530                 ast_queue_hangup(session->channel);
1531         } else {
1532                 ast_sip_session_send_request(session, tdata);
1533         }
1534         ao2_ref(session, -1);
1535         return res;
1536 }
1537
1538 /*! \brief Function called by core to actually start calling a remote party */
1539 static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout)
1540 {
1541         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1542
1543         ao2_ref(channel->session, +1);
1544         if (ast_sip_push_task(channel->session->serializer, call, channel->session)) {
1545                 ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
1546                 ao2_cleanup(channel->session);
1547                 return -1;
1548         }
1549
1550         return 0;
1551 }
1552
1553 /*! \brief Internal function which translates from Asterisk cause codes to SIP response codes */
1554 static int hangup_cause2sip(int cause)
1555 {
1556         switch (cause) {
1557         case AST_CAUSE_UNALLOCATED:             /* 1 */
1558         case AST_CAUSE_NO_ROUTE_DESTINATION:    /* 3 IAX2: Can't find extension in context */
1559         case AST_CAUSE_NO_ROUTE_TRANSIT_NET:    /* 2 */
1560                 return 404;
1561         case AST_CAUSE_CONGESTION:              /* 34 */
1562         case AST_CAUSE_SWITCH_CONGESTION:       /* 42 */
1563                 return 503;
1564         case AST_CAUSE_NO_USER_RESPONSE:        /* 18 */
1565                 return 408;
1566         case AST_CAUSE_NO_ANSWER:               /* 19 */
1567         case AST_CAUSE_UNREGISTERED:        /* 20 */
1568                 return 480;
1569         case AST_CAUSE_CALL_REJECTED:           /* 21 */
1570                 return 403;
1571         case AST_CAUSE_NUMBER_CHANGED:          /* 22 */
1572                 return 410;
1573         case AST_CAUSE_NORMAL_UNSPECIFIED:      /* 31 */
1574                 return 480;
1575         case AST_CAUSE_INVALID_NUMBER_FORMAT:
1576                 return 484;
1577         case AST_CAUSE_USER_BUSY:
1578                 return 486;
1579         case AST_CAUSE_FAILURE:
1580                 return 500;
1581         case AST_CAUSE_FACILITY_REJECTED:       /* 29 */
1582                 return 501;
1583         case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
1584                 return 503;
1585         case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
1586                 return 502;
1587         case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:       /* Can't find codec to connect to host */
1588                 return 488;
1589         case AST_CAUSE_INTERWORKING:    /* Unspecified Interworking issues */
1590                 return 500;
1591         case AST_CAUSE_NOTDEFINED:
1592         default:
1593                 ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
1594                 return 0;
1595         }
1596
1597         /* Never reached */
1598         return 0;
1599 }
1600
1601 struct hangup_data {
1602         int cause;
1603         struct ast_channel *chan;
1604 };
1605
1606 static void hangup_data_destroy(void *obj)
1607 {
1608         struct hangup_data *h_data = obj;
1609
1610         h_data->chan = ast_channel_unref(h_data->chan);
1611 }
1612
1613 static struct hangup_data *hangup_data_alloc(int cause, struct ast_channel *chan)
1614 {
1615         struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
1616
1617         if (!h_data) {
1618                 return NULL;
1619         }
1620
1621         h_data->cause = cause;
1622         h_data->chan = ast_channel_ref(chan);
1623
1624         return h_data;
1625 }
1626
1627 /*! \brief Clear a channel from a session along with its PVT */
1628 static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast, struct chan_pjsip_pvt *pvt)
1629 {
1630         session->channel = NULL;
1631         if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
1632                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, "");
1633         }
1634         if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
1635                 ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, "");
1636         }
1637         ast_channel_tech_pvt_set(ast, NULL);
1638 }
1639
1640 static int hangup(void *data)
1641 {
1642         pj_status_t status;
1643         pjsip_tx_data *packet = NULL;
1644         struct hangup_data *h_data = data;
1645         struct ast_channel *ast = h_data->chan;
1646         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1647         struct chan_pjsip_pvt *pvt = channel->pvt;
1648         struct ast_sip_session *session = channel->session;
1649         int cause = h_data->cause;
1650
1651         if (!session->defer_terminate &&
1652                 ((status = pjsip_inv_end_session(session->inv_session, cause ? cause : 603, NULL, &packet)) == PJ_SUCCESS) && packet) {
1653                 if (packet->msg->type == PJSIP_RESPONSE_MSG) {
1654                         ast_sip_session_send_response(session, packet);
1655                 } else {
1656                         ast_sip_session_send_request(session, packet);
1657                 }
1658         }
1659
1660         clear_session_and_channel(session, ast, pvt);
1661         ao2_cleanup(channel);
1662         ao2_cleanup(h_data);
1663
1664         return 0;
1665 }
1666
1667 /*! \brief Function called by core to hang up a PJSIP session */
1668 static int chan_pjsip_hangup(struct ast_channel *ast)
1669 {
1670         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1671         struct chan_pjsip_pvt *pvt = channel->pvt;
1672         int cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
1673         struct hangup_data *h_data = hangup_data_alloc(cause, ast);
1674
1675         if (!h_data) {
1676                 goto failure;
1677         }
1678
1679         if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
1680                 ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
1681                 goto failure;
1682         }
1683
1684         return 0;
1685
1686 failure:
1687         /* Go ahead and do our cleanup of the session and channel even if we're not going
1688          * to be able to send our SIP request/response
1689          */
1690         clear_session_and_channel(channel->session, ast, pvt);
1691         ao2_cleanup(channel);
1692         ao2_cleanup(h_data);
1693
1694         return -1;
1695 }
1696
1697 struct request_data {
1698         struct ast_sip_session *session;
1699         struct ast_format_cap *caps;
1700         const char *dest;
1701         int cause;
1702 };
1703
1704 static int request(void *obj)
1705 {
1706         struct request_data *req_data = obj;
1707         struct ast_sip_session *session = NULL;
1708         char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
1709         RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
1710
1711         AST_DECLARE_APP_ARGS(args,
1712                 AST_APP_ARG(endpoint);
1713                 AST_APP_ARG(aor);
1714         );
1715
1716         if (ast_strlen_zero(tmp)) {
1717                 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
1718                 req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
1719                 return -1;
1720         }
1721
1722         AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
1723
1724         /* If a request user has been specified extract it from the endpoint name portion */
1725         if ((endpoint_name = strchr(args.endpoint, '@'))) {
1726                 request_user = args.endpoint;
1727                 *endpoint_name++ = '\0';
1728         } else {
1729                 endpoint_name = args.endpoint;
1730         }
1731
1732         if (ast_strlen_zero(endpoint_name)) {
1733                 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
1734                 req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
1735         } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name))) {
1736                 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
1737                 req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
1738                 return -1;
1739         }
1740
1741         if (!(session = ast_sip_session_create_outgoing(endpoint, args.aor, request_user, req_data->caps))) {
1742                 req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
1743                 return -1;
1744         }
1745
1746         req_data->session = session;
1747
1748         return 0;
1749 }
1750
1751 /*! \brief Function called by core to create a new outgoing PJSIP session */
1752 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)
1753 {
1754         struct request_data req_data;
1755         RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
1756
1757         req_data.caps = cap;
1758         req_data.dest = data;
1759
1760         if (ast_sip_push_task_synchronous(NULL, request, &req_data)) {
1761                 *cause = req_data.cause;
1762                 return NULL;
1763         }
1764
1765         session = req_data.session;
1766
1767         if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, requestor ? ast_channel_linkedid(requestor) : NULL, NULL))) {
1768                 /* Session needs to be terminated prematurely */
1769                 return NULL;
1770         }
1771
1772         return session->channel;
1773 }
1774
1775 struct sendtext_data {
1776         struct ast_sip_session *session;
1777         char text[0];
1778 };
1779
1780 static void sendtext_data_destroy(void *obj)
1781 {
1782         struct sendtext_data *data = obj;
1783         ao2_ref(data->session, -1);
1784 }
1785
1786 static struct sendtext_data* sendtext_data_create(struct ast_sip_session *session, const char *text)
1787 {
1788         int size = strlen(text) + 1;
1789         struct sendtext_data *data = ao2_alloc(sizeof(*data)+size, sendtext_data_destroy);
1790
1791         if (!data) {
1792                 return NULL;
1793         }
1794
1795         data->session = session;
1796         ao2_ref(data->session, +1);
1797         ast_copy_string(data->text, text, size);
1798         return data;
1799 }
1800
1801 static int sendtext(void *obj)
1802 {
1803         RAII_VAR(struct sendtext_data *, data, obj, ao2_cleanup);
1804         pjsip_tx_data *tdata;
1805
1806         const struct ast_sip_body body = {
1807                 .type = "text",
1808                 .subtype = "plain",
1809                 .body_text = data->text
1810         };
1811
1812         /* NOT ast_strlen_zero, because a zero-length message is specifically
1813          * allowed by RFC 3428 (See section 10, Examples) */
1814         if (!data->text) {
1815                 return 0;
1816         }
1817
1818         ast_debug(3, "Sending in dialog SIP message\n");
1819
1820         ast_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, &tdata);
1821         ast_sip_add_body(tdata, &body);
1822         ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint);
1823
1824         return 0;
1825 }
1826
1827 /*! \brief Function called by core to send text on PJSIP session */
1828 static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text)
1829 {
1830         struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1831         struct sendtext_data *data = sendtext_data_create(channel->session, text);
1832
1833         if (!data || ast_sip_push_task(channel->session->serializer, sendtext, data)) {
1834                 ao2_ref(data, -1);
1835                 return -1;
1836         }
1837         return 0;
1838 }
1839
1840 /*! \brief Convert SIP hangup causes to Asterisk hangup causes */
1841 static int hangup_sip2cause(int cause)
1842 {
1843         /* Possible values taken from causes.h */
1844
1845         switch(cause) {
1846         case 401:       /* Unauthorized */
1847                 return AST_CAUSE_CALL_REJECTED;
1848         case 403:       /* Not found */
1849                 return AST_CAUSE_CALL_REJECTED;
1850         case 404:       /* Not found */
1851                 return AST_CAUSE_UNALLOCATED;
1852         case 405:       /* Method not allowed */
1853                 return AST_CAUSE_INTERWORKING;
1854         case 407:       /* Proxy authentication required */
1855                 return AST_CAUSE_CALL_REJECTED;
1856         case 408:       /* No reaction */
1857                 return AST_CAUSE_NO_USER_RESPONSE;
1858         case 409:       /* Conflict */
1859                 return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
1860         case 410:       /* Gone */
1861                 return AST_CAUSE_NUMBER_CHANGED;
1862         case 411:       /* Length required */
1863                 return AST_CAUSE_INTERWORKING;
1864         case 413:       /* Request entity too large */
1865                 return AST_CAUSE_INTERWORKING;
1866         case 414:       /* Request URI too large */
1867                 return AST_CAUSE_INTERWORKING;
1868         case 415:       /* Unsupported media type */
1869                 return AST_CAUSE_INTERWORKING;
1870         case 420:       /* Bad extension */
1871                 return AST_CAUSE_NO_ROUTE_DESTINATION;
1872         case 480:       /* No answer */
1873                 return AST_CAUSE_NO_ANSWER;
1874         case 481:       /* No answer */
1875                 return AST_CAUSE_INTERWORKING;
1876         case 482:       /* Loop detected */
1877                 return AST_CAUSE_INTERWORKING;
1878         case 483:       /* Too many hops */
1879                 return AST_CAUSE_NO_ANSWER;
1880         case 484:       /* Address incomplete */
1881                 return AST_CAUSE_INVALID_NUMBER_FORMAT;
1882         case 485:       /* Ambiguous */
1883                 return AST_CAUSE_UNALLOCATED;
1884         case 486:       /* Busy everywhere */
1885                 return AST_CAUSE_BUSY;
1886         case 487:       /* Request terminated */
1887                 return AST_CAUSE_INTERWORKING;
1888         case 488:       /* No codecs approved */
1889                 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
1890         case 491:       /* Request pending */
1891                 return AST_CAUSE_INTERWORKING;
1892         case 493:       /* Undecipherable */
1893                 return AST_CAUSE_INTERWORKING;
1894         case 500:       /* Server internal failure */
1895                 return AST_CAUSE_FAILURE;
1896         case 501:       /* Call rejected */
1897                 return AST_CAUSE_FACILITY_REJECTED;
1898         case 502:
1899                 return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
1900         case 503:       /* Service unavailable */
1901                 return AST_CAUSE_CONGESTION;
1902         case 504:       /* Gateway timeout */
1903                 return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
1904         case 505:       /* SIP version not supported */
1905                 return AST_CAUSE_INTERWORKING;
1906         case 600:       /* Busy everywhere */
1907                 return AST_CAUSE_USER_BUSY;
1908         case 603:       /* Decline */
1909                 return AST_CAUSE_CALL_REJECTED;
1910         case 604:       /* Does not exist anywhere */
1911                 return AST_CAUSE_UNALLOCATED;
1912         case 606:       /* Not acceptable */
1913                 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
1914         default:
1915                 if (cause < 500 && cause >= 400) {
1916                         /* 4xx class error that is unknown - someting wrong with our request */
1917                         return AST_CAUSE_INTERWORKING;
1918                 } else if (cause < 600 && cause >= 500) {
1919                         /* 5xx class error - problem in the remote end */
1920                         return AST_CAUSE_CONGESTION;
1921                 } else if (cause < 700 && cause >= 600) {
1922                         /* 6xx - global errors in the 4xx class */
1923                         return AST_CAUSE_INTERWORKING;
1924                 }
1925                 return AST_CAUSE_NORMAL;
1926         }
1927         /* Never reached */
1928         return 0;
1929 }
1930
1931 static void chan_pjsip_session_begin(struct ast_sip_session *session)
1932 {
1933         RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
1934
1935         if (session->endpoint->media.direct_media.glare_mitigation ==
1936                         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
1937                 return;
1938         }
1939
1940         datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info,
1941                         "direct_media_glare_mitigation");
1942
1943         if (!datastore) {
1944                 return;
1945         }
1946
1947         ast_sip_session_add_datastore(session, datastore);
1948 }
1949
1950 /*! \brief Function called when the session ends */
1951 static void chan_pjsip_session_end(struct ast_sip_session *session)
1952 {
1953         if (!session->channel) {
1954                 return;
1955         }
1956
1957         if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
1958                 int cause = hangup_sip2cause(session->inv_session->cause);
1959
1960                 ast_queue_hangup_with_cause(session->channel, cause);
1961         } else {
1962                 ast_queue_hangup(session->channel);
1963         }
1964 }
1965
1966 /*! \brief Function called when a request is received on the session */
1967 static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
1968 {
1969         pjsip_tx_data *packet = NULL;
1970
1971         if (session->channel) {
1972                 return 0;
1973         }
1974
1975         if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL))) {
1976                 if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
1977                         ast_sip_session_send_response(session, packet);
1978                 }
1979
1980                 ast_log(LOG_ERROR, "Failed to allocate new PJSIP channel on incoming SIP INVITE\n");
1981                 return -1;
1982         }
1983         /* channel gets created on incoming request, but we wait to call start
1984            so other supplements have a chance to run */
1985         return 0;
1986 }
1987
1988 static int pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
1989 {
1990         int res;
1991
1992         res = ast_pbx_start(session->channel);
1993
1994         switch (res) {
1995         case AST_PBX_FAILED:
1996                 ast_log(LOG_WARNING, "Failed to start PBX ;(\n");
1997                 ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
1998                 ast_hangup(session->channel);
1999                 break;
2000         case AST_PBX_CALL_LIMIT:
2001                 ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
2002                 ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
2003                 ast_hangup(session->channel);
2004                 break;
2005         case AST_PBX_SUCCESS:
2006         default:
2007                 break;
2008         }
2009
2010         ast_debug(3, "Started PBX on new PJSIP channel %s\n", ast_channel_name(session->channel));
2011
2012         return (res == AST_PBX_SUCCESS) ? 0 : -1;
2013 }
2014
2015 static struct ast_sip_session_supplement pbx_start_supplement = {
2016         .method = "INVITE",
2017         .priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_LAST,
2018         .incoming_request = pbx_start_incoming_request,
2019 };
2020
2021 /*! \brief Function called when a response is received on the session */
2022 static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
2023 {
2024         struct pjsip_status_line status = rdata->msg_info.msg->line.status;
2025
2026         if (!session->channel) {
2027                 return;
2028         }
2029
2030         switch (status.code) {
2031         case 180:
2032                 ast_queue_control(session->channel, AST_CONTROL_RINGING);
2033                 if (ast_channel_state(session->channel) != AST_STATE_UP) {
2034                         ast_setstate(session->channel, AST_STATE_RINGING);
2035                 }
2036                 break;
2037         case 183:
2038                 ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
2039                 break;
2040         case 200:
2041                 ast_queue_control(session->channel, AST_CONTROL_ANSWER);
2042                 break;
2043         default:
2044                 break;
2045         }
2046 }
2047
2048 static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
2049 {
2050         if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
2051                 if (session->endpoint->media.direct_media.enabled && session->channel) {
2052                         ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
2053                 }
2054         }
2055         return 0;
2056 }
2057
2058 /*!
2059  * \brief Load the module
2060  *
2061  * Module loading including tests for configuration or dependencies.
2062  * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
2063  * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
2064  * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
2065  * configuration file or other non-critical problem return
2066  * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
2067  */
2068 static int load_module(void)
2069 {
2070         if (!(chan_pjsip_tech.capabilities = ast_format_cap_alloc(0))) {
2071                 return AST_MODULE_LOAD_DECLINE;
2072         }
2073
2074         ast_format_cap_add_all_by_type(chan_pjsip_tech.capabilities, AST_FORMAT_TYPE_AUDIO);
2075
2076         ast_rtp_glue_register(&chan_pjsip_rtp_glue);
2077
2078         if (ast_channel_register(&chan_pjsip_tech)) {
2079                 ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
2080                 goto end;
2081         }
2082
2083         if (ast_custom_function_register(&chan_pjsip_dial_contacts_function)) {
2084                 ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n");
2085                 goto end;
2086         }
2087
2088         if (ast_custom_function_register(&media_offer_function)) {
2089                 ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
2090         }
2091
2092         if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
2093                 ast_log(LOG_ERROR, "Unable to register PJSIP supplement\n");
2094                 goto end;
2095         }
2096
2097         if (ast_sip_session_register_supplement(&pbx_start_supplement)) {
2098                 ast_log(LOG_ERROR, "Unable to register PJSIP pbx start supplement\n");
2099                 ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2100                 goto end;
2101         }
2102
2103         if (ast_sip_session_register_supplement(&chan_pjsip_ack_supplement)) {
2104                 ast_log(LOG_ERROR, "Unable to register PJSIP ACK supplement\n");
2105                 ast_sip_session_unregister_supplement(&pbx_start_supplement);
2106                 ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2107                 goto end;
2108         }
2109
2110         return 0;
2111
2112 end:
2113         ast_custom_function_unregister(&media_offer_function);
2114         ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
2115         ast_channel_unregister(&chan_pjsip_tech);
2116         ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
2117
2118         return AST_MODULE_LOAD_FAILURE;
2119 }
2120
2121 /*! \brief Reload module */
2122 static int reload(void)
2123 {
2124         return -1;
2125 }
2126
2127 /*! \brief Unload the PJSIP channel from Asterisk */
2128 static int unload_module(void)
2129 {
2130         ast_custom_function_unregister(&media_offer_function);
2131
2132         ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
2133         ast_sip_session_unregister_supplement(&pbx_start_supplement);
2134         ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
2135
2136         ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
2137         ast_channel_unregister(&chan_pjsip_tech);
2138         ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
2139
2140         return 0;
2141 }
2142
2143 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Channel Driver",
2144                 .load = load_module,
2145                 .unload = unload_module,
2146                 .reload = reload,
2147                 .load_pri = AST_MODPRI_CHANNEL_DRIVER,
2148                );