channels/chan_pjsip.c

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