chan_pjsip.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2013, Digium, Inc.
 *
 * Joshua Colp <jcolp@digium.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */
 
/*! \file
 *
 * \author Joshua Colp <jcolp@digium.com>
 *
 * \brief PSJIP SIP Channel Driver
 *
 * \ingroup channel_drivers
 */
 
/*** MODULEINFO
   <depend>pjproject</depend>
   <depend>res_pjsip</depend>
   <depend>res_pjsip_pubsub</depend>
   <depend>res_pjsip_session</depend>
   <support_level>core</support_level>
 ***/
 
#include "asterisk.h"
 
#include <pjsip.h>
#include <pjsip_ua.h>
#include <pjlib.h>
 
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/causes.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/dsp.h"
#include "asterisk/stasis_endpoints.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/indications.h"
#include "asterisk/format_cache.h"
#include "asterisk/translate.h"
#include "asterisk/threadstorage.h"
#include "asterisk/features_config.h"
#include "asterisk/pickup.h"
#include "asterisk/test.h"
#include "asterisk/message.h"
 
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/stream.h"
 
#include "pjsip/include/chan_pjsip.h"
#include "pjsip/include/dialplan_functions.h"
#include "pjsip/include/cli_functions.h"
 
AST_THREADSTORAGE(uniqueid_threadbuf);
#define UNIQUEID_BUFSIZE 256
 
static const char channel_type[] = "PJSIP";
 
static unsigned int chan_idx;
 
static void chan_pjsip_pvt_dtor(void *obj)
{
}
 
/*! \brief Asterisk core interaction functions */
static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
static struct ast_channel *chan_pjsip_request_with_stream_topology(const char *type,
   struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids,
   const struct ast_channel *requestor, const char *data, int *cause);
static int chan_pjsip_sendtext_data(struct ast_channel *ast, struct ast_msg_data *msg);
static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text);
static int chan_pjsip_digit_begin(struct ast_channel *ast, char digit);
static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout);
static int chan_pjsip_hangup(struct ast_channel *ast);
static int chan_pjsip_answer(struct ast_channel *ast);
static struct ast_frame *chan_pjsip_read_stream(struct ast_channel *ast);
static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *f);
static int chan_pjsip_write_stream(struct ast_channel *ast, int stream_num, struct ast_frame *f);
static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
static int chan_pjsip_transfer(struct ast_channel *ast, const char *target);
static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static int chan_pjsip_devicestate(const char *data);
static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
static const char *chan_pjsip_get_uniqueid(struct ast_channel *ast);
 
/*! \brief PBX interface structure for channel registration */
struct ast_channel_tech chan_pjsip_tech = {
   .type = channel_type,
   .description = "PJSIP Channel Driver",
   .requester = chan_pjsip_request,
   .requester_with_stream_topology = chan_pjsip_request_with_stream_topology,
   .send_text = chan_pjsip_sendtext,
   .send_text_data = chan_pjsip_sendtext_data,
   .send_digit_begin = chan_pjsip_digit_begin,
   .send_digit_end = chan_pjsip_digit_end,
   .call = chan_pjsip_call,
   .hangup = chan_pjsip_hangup,
   .answer = chan_pjsip_answer,
   .read_stream = chan_pjsip_read_stream,
   .write = chan_pjsip_write,
   .write_stream = chan_pjsip_write_stream,
   .exception = chan_pjsip_read_stream,
   .indicate = chan_pjsip_indicate,
   .transfer = chan_pjsip_transfer,
   .fixup = chan_pjsip_fixup,
   .devicestate = chan_pjsip_devicestate,
   .queryoption = chan_pjsip_queryoption,
   .func_channel_read = pjsip_acf_channel_read,
   .get_pvt_uniqueid = chan_pjsip_get_uniqueid,
   .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER | AST_CHAN_TP_SEND_TEXT_DATA
};
 
/*! \brief SIP session interaction functions */
static void chan_pjsip_session_begin(struct ast_sip_session *session);
static void chan_pjsip_session_end(struct ast_sip_session *session);
static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
static void chan_pjsip_incoming_response_update_cause(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
 
/*! \brief SIP session supplement structure */
static struct ast_sip_session_supplement chan_pjsip_supplement = {
   .method = "INVITE",
   .priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
   .session_begin = chan_pjsip_session_begin,
   .session_end = chan_pjsip_session_end,
   .incoming_request = chan_pjsip_incoming_request,
   .incoming_response = chan_pjsip_incoming_response,
   /* It is important that this supplement runs after media has been negotiated */
   .response_priority = AST_SIP_SESSION_AFTER_MEDIA,
};
 
/*! \brief SIP session supplement structure just for responses */
static struct ast_sip_session_supplement chan_pjsip_supplement_response = {
   .method = "INVITE",
   .priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
   .incoming_response = chan_pjsip_incoming_response_update_cause,
   .response_priority = AST_SIP_SESSION_BEFORE_MEDIA | AST_SIP_SESSION_AFTER_MEDIA,
};
 
static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
 
static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
   .method = "ACK",
   .priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
   .incoming_request = chan_pjsip_incoming_ack,
};
 
/*! \brief Function called by RTP engine to get local audio RTP peer */
static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
   struct ast_sip_endpoint *endpoint;
   struct ast_datastore *datastore;
   struct ast_sip_session_media *media;
 
   if (!channel || !channel->session) {
      return AST_RTP_GLUE_RESULT_FORBID;
   }
 
   /* XXX Getting the first RTP instance for direct media related stuff seems just
    * absolutely wrong. But the native RTP bridge knows no other method than single-stream
    * for direct media. So this is the best we can do.
    */
   media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
   if (!media || !media->rtp) {
      return AST_RTP_GLUE_RESULT_FORBID;
   }
 
   datastore = ast_sip_session_get_datastore(channel->session, "t38");
   if (datastore) {
      ao2_ref(datastore, -1);
      return AST_RTP_GLUE_RESULT_FORBID;
   }
 
   endpoint = channel->session->endpoint;
 
   *instance = media->rtp;
   ao2_ref(*instance, +1);
 
   ast_assert(endpoint != NULL);
   if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
      return AST_RTP_GLUE_RESULT_FORBID;
   }
 
   if (endpoint->media.direct_media.enabled) {
      return AST_RTP_GLUE_RESULT_REMOTE;
   }
 
   return AST_RTP_GLUE_RESULT_LOCAL;
}
 
/*! \brief Function called by RTP engine to get local video RTP peer */
static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
   struct ast_sip_endpoint *endpoint;
   struct ast_sip_session_media *media;
 
   if (!channel || !channel->session) {
      return AST_RTP_GLUE_RESULT_FORBID;
   }
 
   media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO];
   if (!media || !media->rtp) {
      return AST_RTP_GLUE_RESULT_FORBID;
   }
 
   endpoint = channel->session->endpoint;
 
   *instance = media->rtp;
   ao2_ref(*instance, +1);
 
   ast_assert(endpoint != NULL);
   if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
      return AST_RTP_GLUE_RESULT_FORBID;
   }
 
   return AST_RTP_GLUE_RESULT_LOCAL;
}
 
/*! \brief Function called by RTP engine to get peer capabilities */
static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
{
   SCOPE_ENTER(1, "%s Native formats %s\n", ast_channel_name(chan),
      ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(ast_channel_nativeformats(chan), &STR_TMP)));
   ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN);
   SCOPE_EXIT_RTN();
}
 
/*! \brief Destructor function for \ref transport_info_data */
static void transport_info_destroy(void *obj)
{
   struct transport_info_data *data = obj;
   ast_free(data);
}
 
/*! \brief Datastore used to store local/remote addresses for the
 * INVITE request that created the PJSIP channel */
static struct ast_datastore_info transport_info = {
   .type = "chan_pjsip_transport_info",
   .destroy = transport_info_destroy,
};
 
static struct ast_datastore_info direct_media_mitigation_info = { };
 
static int direct_media_mitigate_glare(struct ast_sip_session *session)
{
   RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
 
   if (session->endpoint->media.direct_media.glare_mitigation ==
         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
      return 0;
   }
 
   datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation");
   if (!datastore) {
      return 0;
   }
 
   /* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */
   ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation");
 
   if ((session->endpoint->media.direct_media.glare_mitigation ==
         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING &&
         session->inv_session->role == PJSIP_ROLE_UAC) ||
         (session->endpoint->media.direct_media.glare_mitigation ==
         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING &&
         session->inv_session->role == PJSIP_ROLE_UAS)) {
      return 1;
   }
 
   return 0;
}
 
/*! \brief Helper function to find the position for RTCP */
static int rtp_find_rtcp_fd_position(struct ast_sip_session *session, struct ast_rtp_instance *rtp)
{
   int index;
 
   for (index = 0; index < AST_VECTOR_SIZE(&session->active_media_state->read_callbacks); ++index) {
      struct ast_sip_session_media_read_callback_state *callback_state =
         AST_VECTOR_GET_ADDR(&session->active_media_state->read_callbacks, index);
 
      if (callback_state->fd != ast_rtp_instance_fd(rtp, 1)) {
         continue;
      }
 
      return index;
   }
 
   return -1;
}
 
/*!
 * \pre chan is locked
 */
static int check_for_rtp_changes(struct ast_channel *chan, struct ast_rtp_instance *rtp,
      struct ast_sip_session_media *media, struct ast_sip_session *session)
{
   int changed = 0, position = -1;
 
   if (media->rtp) {
      position = rtp_find_rtcp_fd_position(session, media->rtp);
   }
 
   if (rtp) {
      changed = ast_rtp_instance_get_and_cmp_remote_address(rtp, &media->direct_media_addr);
      if (media->rtp) {
         if (position != -1) {
            ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, -1);
         }
         ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 0);
      }
   } else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
      ast_sockaddr_setnull(&media->direct_media_addr);
      changed = 1;
      if (media->rtp) {
         /* Direct media has ended - reset time of last received RTP packet
          * to avoid premature RTP timeout. Synchronisation between the
          * modification of direct_mdedia_addr+last_rx here and reading the
          * values in res_pjsip_sdp_rtp.c:rtp_check_timeout() is provided
          * by the channel's lock (which is held while this function is
          * executed).
          */
         ast_rtp_instance_set_last_rx(media->rtp, time(NULL));
         ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1);
         if (position != -1) {
            ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, ast_rtp_instance_fd(media->rtp, 1));
         }
      }
   }
 
   return changed;
}
 
struct rtp_direct_media_data {
   struct ast_channel *chan;
   struct ast_rtp_instance *rtp;
   struct ast_rtp_instance *vrtp;
   struct ast_format_cap *cap;
   struct ast_sip_session *session;
};
 
static void rtp_direct_media_data_destroy(void *data)
{
   struct rtp_direct_media_data *cdata = data;
 
   ao2_cleanup(cdata->session);
   ao2_cleanup(cdata->cap);
   ao2_cleanup(cdata->vrtp);
   ao2_cleanup(cdata->rtp);
   ao2_cleanup(cdata->chan);
}
 
static struct rtp_direct_media_data *rtp_direct_media_data_create(
   struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp,
   const struct ast_format_cap *cap, struct ast_sip_session *session)
{
   struct rtp_direct_media_data *cdata = ao2_alloc(sizeof(*cdata), rtp_direct_media_data_destroy);
 
   if (!cdata) {
      return NULL;
   }
 
   cdata->chan = ao2_bump(chan);
   cdata->rtp = ao2_bump(rtp);
   cdata->vrtp = ao2_bump(vrtp);
   cdata->cap = ao2_bump((struct ast_format_cap *)cap);
   cdata->session = ao2_bump(session);
 
   return cdata;
}
 
static int send_direct_media_request(void *data)
{
   struct rtp_direct_media_data *cdata = data;
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(cdata->chan);
   struct ast_sip_session *session;
   int changed = 0;
   int res = 0;
 
   /* XXX In an ideal world each media stream would be direct, but for now preserve behavior
    * and connect only the default media sessions for audio and video.
    */
 
   /* The channel needs to be locked when checking for RTP changes.
    * Otherwise, we could end up destroying an underlying RTCP structure
    * at the same time that the channel thread is attempting to read RTCP
    */
   ast_channel_lock(cdata->chan);
   session = channel->session;
   if (session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
      changed |= check_for_rtp_changes(
         cdata->chan, cdata->rtp, session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO], session);
   }
   if (session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO]) {
      changed |= check_for_rtp_changes(
         cdata->chan, cdata->vrtp, session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO], session);
   }
   ast_channel_unlock(cdata->chan);
 
   if (direct_media_mitigate_glare(cdata->session)) {
      ast_debug(4, "Disregarding setting RTP on %s: mitigating re-INVITE glare\n", ast_channel_name(cdata->chan));
      ao2_ref(cdata, -1);
      return 0;
   }
 
   if (cdata->cap && ast_format_cap_count(cdata->cap) &&
       !ast_format_cap_identical(cdata->session->direct_media_cap, cdata->cap)) {
      ast_format_cap_remove_by_type(cdata->session->direct_media_cap, AST_MEDIA_TYPE_UNKNOWN);
      ast_format_cap_append_from_cap(cdata->session->direct_media_cap, cdata->cap, AST_MEDIA_TYPE_UNKNOWN);
      changed = 1;
   }
 
   if (changed) {
      ast_debug(4, "RTP changed on %s; initiating direct media update\n", ast_channel_name(cdata->chan));
      res = ast_sip_session_refresh(cdata->session, NULL, NULL, NULL,
         cdata->session->endpoint->media.direct_media.method, 1, NULL);
   }
 
   ao2_ref(cdata, -1);
   return res;
}
 
/*! \brief Function called by RTP engine to change where the remote party should send media */
static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
      struct ast_rtp_instance *rtp,
      struct ast_rtp_instance *vrtp,
      struct ast_rtp_instance *tpeer,
      const struct ast_format_cap *cap,
      int nat_active)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
   struct ast_sip_session *session = channel->session;
   struct rtp_direct_media_data *cdata;
   SCOPE_ENTER(1, "%s %s\n", ast_channel_name(chan),
      ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(cap, &STR_TMP)));
 
   /* Don't try to do any direct media shenanigans on early bridges */
   if ((rtp || vrtp || tpeer) && !ast_channel_is_bridged(chan)) {
      ast_debug(4, "Disregarding setting RTP on %s: channel is not bridged\n", ast_channel_name(chan));
      SCOPE_EXIT_RTN_VALUE(0, "Channel not bridged\n");
   }
 
   if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
      ast_debug(4, "Disregarding setting RTP on %s: NAT is active\n", ast_channel_name(chan));
      SCOPE_EXIT_RTN_VALUE(0, "NAT is active\n");
   }
 
   cdata = rtp_direct_media_data_create(chan, rtp, vrtp, cap, session);
   if (!cdata) {
      SCOPE_EXIT_RTN_VALUE(0);
   }
 
   if (ast_sip_push_task(session->serializer, send_direct_media_request, cdata)) {
      ast_log(LOG_ERROR, "Unable to send direct media request for channel %s\n", ast_channel_name(chan));
      ao2_ref(cdata, -1);
   }
 
   SCOPE_EXIT_RTN_VALUE(0);
}
 
/*! \brief Local glue for interacting with the RTP engine core */
static struct ast_rtp_glue chan_pjsip_rtp_glue = {
   .type = "PJSIP",
   .get_rtp_info = chan_pjsip_get_rtp_peer,
   .get_vrtp_info = chan_pjsip_get_vrtp_peer,
   .get_codec = chan_pjsip_get_codec,
   .update_peer = chan_pjsip_set_rtp_peer,
};
 
static void set_channel_on_rtp_instance(const struct ast_sip_session *session,
   const char *channel_id)
{
   int i;
 
   for (i = 0; i < AST_VECTOR_SIZE(&session->active_media_state->sessions); ++i) {
      struct ast_sip_session_media *session_media;
 
      session_media = AST_VECTOR_GET(&session->active_media_state->sessions, i);
      if (!session_media || !session_media->rtp) {
         continue;
      }
 
      ast_rtp_instance_set_channel_id(session_media->rtp, channel_id);
   }
}
 
/*!
 * \brief Determine if a topology is compatible with format capabilities
 *
 * This will return true if ANY formats in the topology are compatible with the format
 * capabilities.
 *
 * XXX When supporting true multistream, we will need to be sure to mark which streams from
 * top1 are compatible with which streams from top2. Then the ones that are not compatible
 * will need to be marked as "removed" so that they are negotiated as expected.
 *
 * \param top Topology
 * \param cap Format capabilities
 * \retval 1 The topology has at least one compatible format
 * \retval 0 The topology has no compatible formats or an error occurred.
 */
static int compatible_formats_exist(struct ast_stream_topology *top, struct ast_format_cap *cap)
{
   struct ast_format_cap *cap_from_top;
   int res;
   SCOPE_ENTER(1, "Topology: %s Formats: %s\n",
      ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_stream_topology_to_str(top, &STR_TMP)),
      ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(cap, &STR_TMP)));
 
   cap_from_top = ast_stream_topology_get_formats(top);
 
   if (!cap_from_top) {
      SCOPE_EXIT_RTN_VALUE(0, "Topology had no formats\n");
   }
 
   res = ast_format_cap_iscompatible(cap_from_top, cap);
   ao2_ref(cap_from_top, -1);
 
   SCOPE_EXIT_RTN_VALUE(res, "Compatible? %s\n", res ? "yes" : "no");
}
 
/*! \brief Function called to create a new PJSIP Asterisk channel */
static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *cid_name)
{
   struct ast_channel *chan;
   struct ast_format_cap *caps;
   RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
   struct ast_sip_channel_pvt *channel;
   struct ast_variable *var;
   struct ast_stream_topology *topology;
   SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
   if (!(pvt = ao2_alloc_options(sizeof(*pvt), chan_pjsip_pvt_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK))) {
      SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create pvt\n");
   }
 
   chan = ast_channel_alloc_with_endpoint(1, state,
      S_COR(session->id.number.valid, session->id.number.str, ""),
      S_COR(session->id.name.valid, session->id.name.str, ""),
      session->endpoint->accountcode,
      exten, session->endpoint->context,
      assignedids, requestor, 0,
      session->endpoint->persistent, "PJSIP/%s-%08x",
      ast_sorcery_object_get_id(session->endpoint),
      (unsigned) ast_atomic_fetchadd_int((int *) &chan_idx, +1));
   if (!chan) {
      SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
   }
 
   ast_channel_tech_set(chan, &chan_pjsip_tech);
 
   if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
      ast_channel_unlock(chan);
      ast_hangup(chan);
      SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create pvt channel\n");
   }
 
   ast_channel_tech_pvt_set(chan, channel);
 
   if (!ast_stream_topology_get_count(session->pending_media_state->topology) ||
      !compatible_formats_exist(session->pending_media_state->topology, session->endpoint->media.codecs)) {
      caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
      if (!caps) {
         ast_channel_unlock(chan);
         ast_hangup(chan);
         SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create caps\n");
      }
      ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN);
      topology = ast_stream_topology_clone(session->endpoint->media.topology);
   } else {
      caps = ast_stream_topology_get_formats(session->pending_media_state->topology);
      topology = ast_stream_topology_clone(session->pending_media_state->topology);
   }
 
   if (!topology || !caps) {
      ao2_cleanup(caps);
      ast_stream_topology_free(topology);
      ast_channel_unlock(chan);
      ast_hangup(chan);
      SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't get caps or clone topology\n");
   }
 
   ast_channel_stage_snapshot(chan);
 
   ast_channel_nativeformats_set(chan, caps);
   ast_channel_set_stream_topology(chan, topology);
 
   if (!ast_format_cap_empty(caps)) {
      struct ast_format *fmt;
 
      fmt = ast_format_cap_get_best_by_type(caps, AST_MEDIA_TYPE_AUDIO);
      if (!fmt) {
         /* Since our capabilities aren't empty, this will succeed */
         fmt = ast_format_cap_get_format(caps, 0);
      }
      ast_channel_set_writeformat(chan, fmt);
      ast_channel_set_rawwriteformat(chan, fmt);
      ast_channel_set_readformat(chan, fmt);
      ast_channel_set_rawreadformat(chan, fmt);
      ao2_ref(fmt, -1);
   }
 
   ao2_ref(caps, -1);
 
   if (state == AST_STATE_RING) {
      ast_channel_rings_set(chan, 1);
   }
 
   ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
 
   ast_party_id_copy(&ast_channel_caller(chan)->id, &session->id);
   ast_party_id_copy(&ast_channel_caller(chan)->ani, &session->id);
   ast_channel_caller(chan)->ani2 = session->ani2;
 
   if (!ast_strlen_zero(exten)) {
      /* Set provided DNID on the new channel. */
      ast_channel_dialed(chan)->number.str = ast_strdup(exten);
   }
 
   ast_channel_priority_set(chan, 1);
 
   ast_channel_callgroup_set(chan, session->endpoint->pickup.callgroup);
   ast_channel_pickupgroup_set(chan, session->endpoint->pickup.pickupgroup);
 
   ast_channel_named_callgroups_set(chan, session->endpoint->pickup.named_callgroups);
   ast_channel_named_pickupgroups_set(chan, session->endpoint->pickup.named_pickupgroups);
 
   if (!ast_strlen_zero(session->endpoint->language)) {
      ast_channel_language_set(chan, session->endpoint->language);
   }
 
   if (!ast_strlen_zero(session->endpoint->zone)) {
      struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone);
      if (!zone) {
         ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone);
      }
      ast_channel_zone_set(chan, zone);
   }
 
   for (var = session->endpoint->channel_vars; var; var = var->next) {
      char buf[512];
      pbx_builtin_setvar_helper(chan, var->name, ast_get_encoded_str(
                    var->value, buf, sizeof(buf)));
   }
 
   ast_channel_stage_snapshot_done(chan);
   ast_channel_unlock(chan);
 
   set_channel_on_rtp_instance(session, ast_channel_uniqueid(chan));
 
   SCOPE_EXIT_RTN_VALUE(chan);
}
 
struct answer_data {
   struct ast_sip_session *session;
   unsigned long indent;
};
 
static int answer(void *data)
{
   struct answer_data *ans_data = data;
   pj_status_t status = PJ_SUCCESS;
   pjsip_tx_data *packet = NULL;
   struct ast_sip_session *session = ans_data->session;
   SCOPE_ENTER_TASK(1, ans_data->indent, "%s\n", ast_sip_session_get_name(session));
 
   if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
      ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
         session->inv_session->cause,
         pjsip_get_status_text(session->inv_session->cause)->ptr);
      SCOPE_EXIT_RTN_VALUE(0, "Disconnected\n");
   }
 
   pjsip_dlg_inc_lock(session->inv_session->dlg);
   if (session->inv_session->invite_tsx) {
      status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet);
   } else {
      ast_log(LOG_ERROR,"Cannot answer '%s' because there is no associated SIP transaction\n",
         ast_channel_name(session->channel));
   }
   pjsip_dlg_dec_lock(session->inv_session->dlg);
 
   if (status == PJ_SUCCESS && packet) {
      ast_sip_session_send_response(session, packet);
   }
 
   if (status != PJ_SUCCESS) {
      char err[PJ_ERR_MSG_SIZE];
 
      pj_strerror(status, err, sizeof(err));
      ast_log(LOG_WARNING,"Cannot answer '%s': %s\n",
         ast_channel_name(session->channel), err);
      /*
       * Return this value so we can distinguish between this
       * failure and the threadpool synchronous push failing.
       */
      SCOPE_EXIT_RTN_VALUE(-2, "pjproject failure\n");
   }
   SCOPE_EXIT_RTN_VALUE(0);
}
 
/*! \brief Function called by core when we should answer a PJSIP session */
static int chan_pjsip_answer(struct ast_channel *ast)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   struct ast_sip_session *session;
   struct answer_data ans_data = { 0, };
   int res;
   SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
 
   if (ast_channel_state(ast) == AST_STATE_UP) {
      SCOPE_EXIT_RTN_VALUE(0, "Already up\n");
      return 0;
   }
 
   ast_setstate(ast, AST_STATE_UP);
   session = ao2_bump(channel->session);
 
   /* the answer task needs to be pushed synchronously otherwise a race condition
      can occur between this thread and bridging (specifically when native bridging
      attempts to do direct media) */
   ast_channel_unlock(ast);
   ans_data.session = session;
   ans_data.indent = ast_trace_get_indent();
   res = ast_sip_push_task_wait_serializer(session->serializer, answer, &ans_data);
   if (res) {
      if (res == -1) {
         ast_log(LOG_ERROR,"Cannot answer '%s': Unable to push answer task to the threadpool.\n",
            ast_channel_name(session->channel));
      }
      ao2_ref(session, -1);
      ast_channel_lock(ast);
      SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
   }
   ao2_ref(session, -1);
   ast_channel_lock(ast);
 
   SCOPE_EXIT_RTN_VALUE(0);
}
 
/*! \brief Internal helper function called when CNG tone is detected */
static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_channel *ast, struct ast_sip_session *session,
   struct ast_frame *f)
{
   const char *target_context;
   int exists;
   int dsp_features;
 
   dsp_features = ast_dsp_get_features(session->dsp);
   dsp_features &= ~DSP_FEATURE_FAX_DETECT;
   if (dsp_features) {
      ast_dsp_set_features(session->dsp, dsp_features);
   } else {
      ast_dsp_free(session->dsp);
      session->dsp = NULL;
   }
 
   /* If already executing in the fax extension don't do anything */
   if (!strcmp(ast_channel_exten(ast), "fax")) {
      return f;
   }
 
   target_context = S_OR(ast_channel_macrocontext(ast), ast_channel_context(ast));
 
   /*
    * We need to unlock the channel here because ast_exists_extension has the
    * potential to start and stop an autoservice on the channel. Such action
    * is prone to deadlock if the channel is locked.
    *
    * ast_async_goto() has its own restriction on not holding the channel lock.
    */
   ast_channel_unlock(ast);
   ast_frfree(f);
   f = &ast_null_frame;
   exists = ast_exists_extension(ast, target_context, "fax", 1,
      S_COR(ast_channel_caller(ast)->id.number.valid,
         ast_channel_caller(ast)->id.number.str, NULL));
   if (exists) {
      ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
         ast_channel_name(ast));
      pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast_channel_exten(ast));
      if (ast_async_goto(ast, target_context, "fax", 1)) {
         ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
            ast_channel_name(ast), target_context);
      }
   } else {
      ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
         ast_channel_name(ast), target_context);
   }
 
   /* It's possible for a masquerade to have occurred when doing the ast_async_goto resulting in
    * the channel on the session having changed. Since we need to return with the original channel
    * locked we lock the channel that was passed in and not session->channel.
    */
   ast_channel_lock(ast);
 
   return f;
}
 
/*! \brief Determine if the given frame is in a format we've negotiated */
static int is_compatible_format(struct ast_sip_session *session, struct ast_frame *f)
{
   struct ast_stream_topology *topology = session->active_media_state->topology;
   struct ast_stream *stream = ast_stream_topology_get_stream(topology, f->stream_num);
   const struct ast_format_cap *cap = ast_stream_get_formats(stream);
 
   return ast_format_cap_iscompatible_format(cap, f->subclass.format) != AST_FORMAT_CMP_NOT_EQUAL;
}
 
/*!
 * \brief Function called by core to read any waiting frames
 *
 * \note The channel is already locked.
 */
static struct ast_frame *chan_pjsip_read_stream(struct ast_channel *ast)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   struct ast_sip_session *session = channel->session;
   struct ast_sip_session_media_read_callback_state *callback_state;
   struct ast_frame *f;
   int fdno = ast_channel_fdno(ast) - AST_EXTENDED_FDS;
   struct ast_frame *cur;
 
   if (fdno >= AST_VECTOR_SIZE(&session->active_media_state->read_callbacks)) {
      return &ast_null_frame;
   }
 
   callback_state = AST_VECTOR_GET_ADDR(&session->active_media_state->read_callbacks, fdno);
   f = callback_state->read_callback(session, callback_state->session);
 
   if (!f) {
      return f;
   }
 
   for (cur = f; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
      if (cur->frametype == AST_FRAME_VOICE) {
         break;
      }
   }
 
   if (!cur || callback_state->session != session->active_media_state->default_session[callback_state->session->type]) {
      return f;
   }
 
   session = channel->session;
 
   /*
    * Asymmetric RTP only has one native format set at a time.
    * Therefore we need to update the native format to the current
    * raw read format BEFORE the native format check
    */
   if (!session->endpoint->asymmetric_rtp_codec &&
      ast_format_cmp(ast_channel_rawwriteformat(ast), cur->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL &&
      is_compatible_format(session, cur)) {
      struct ast_format_cap *caps;
 
      /* For maximum compatibility we ensure that the formats match that of the received media */
      ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n",
         ast_format_get_name(cur->subclass.format), ast_channel_name(ast),
         ast_format_get_name(ast_channel_rawwriteformat(ast)));
 
      caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
      if (caps) {
         ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(ast), AST_MEDIA_TYPE_UNKNOWN);
         ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO);
         ast_format_cap_append(caps, cur->subclass.format, 0);
         ast_channel_nativeformats_set(ast, caps);
         ao2_ref(caps, -1);
      }
 
      ast_set_write_format_path(ast, ast_channel_writeformat(ast), cur->subclass.format);
      ast_set_read_format_path(ast, ast_channel_readformat(ast), cur->subclass.format);
 
      if (ast_channel_is_bridged(ast)) {
         ast_channel_set_unbridged_nolock(ast, 1);
      }
   }
 
   if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), cur->subclass.format)
         == AST_FORMAT_CMP_NOT_EQUAL) {
      ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when it has not been negotiated\n",
            ast_format_get_name(cur->subclass.format), ast_channel_name(ast));
      ast_frfree(f);
      return &ast_null_frame;
   }
 
   if (session->dsp) {
      int dsp_features;
 
      dsp_features = ast_dsp_get_features(session->dsp);
      if ((dsp_features & DSP_FEATURE_FAX_DETECT)
         && session->endpoint->faxdetect_timeout
         && session->endpoint->faxdetect_timeout <= ast_channel_get_up_time(ast)) {
         dsp_features &= ~DSP_FEATURE_FAX_DETECT;
         if (dsp_features) {
            ast_dsp_set_features(session->dsp, dsp_features);
         } else {
            ast_dsp_free(session->dsp);
            session->dsp = NULL;
         }
         ast_debug(3, "Channel driver fax CNG detection timeout on %s\n",
            ast_channel_name(ast));
      }
   }
   if (session->dsp) {
      f = ast_dsp_process(ast, session->dsp, f);
      if (f && (f->frametype == AST_FRAME_DTMF)) {
         if (f->subclass.integer == 'f') {
            ast_debug(3, "Channel driver fax CNG detected on %s\n",
               ast_channel_name(ast));
            f = chan_pjsip_cng_tone_detected(ast, session, f);
            /* When chan_pjsip_cng_tone_detected returns it is possible for the
             * channel pointed to by ast and by session->channel to differ due to a
             * masquerade. It's best not to touch things after this.
             */
         } else {
            ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
               ast_channel_name(ast));
         }
      }
   }
 
   return f;
}
 
static int chan_pjsip_write_stream(struct ast_channel *ast, int stream_num, struct ast_frame *frame)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   struct ast_sip_session *session = channel->session;
   struct ast_sip_session_media *media = NULL;
   int res = 0;
 
   /* The core provides a guarantee that the stream will exist when we are called if stream_num is provided */
   if (stream_num >= 0) {
      /* What is not guaranteed is that a media session will exist */
      if (stream_num < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions)) {
         media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, stream_num);
      }
   }
 
   switch (frame->frametype) {
   case AST_FRAME_VOICE:
      if (!media) {
         return 0;
      } else if (media->type != AST_MEDIA_TYPE_AUDIO) {
         ast_debug(3, "Channel %s stream %d is of type '%s', not audio!\n",
            ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
         return 0;
      } else if (media == channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO] &&
         ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
         struct ast_str *cap_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
         struct ast_str *write_transpath = ast_str_alloca(256);
         struct ast_str *read_transpath = ast_str_alloca(256);
 
         ast_log(LOG_WARNING,
            "Channel %s asked to send %s frame when native formats are %s (rd:%s->%s;%s wr:%s->%s;%s)\n",
            ast_channel_name(ast),
            ast_format_get_name(frame->subclass.format),
            ast_format_cap_get_names(ast_channel_nativeformats(ast), &cap_buf),
            ast_format_get_name(ast_channel_rawreadformat(ast)),
            ast_format_get_name(ast_channel_readformat(ast)),
            ast_translate_path_to_str(ast_channel_readtrans(ast), &read_transpath),
            ast_format_get_name(ast_channel_writeformat(ast)),
            ast_format_get_name(ast_channel_rawwriteformat(ast)),
            ast_translate_path_to_str(ast_channel_writetrans(ast), &write_transpath));
         return 0;
      } else if (media->write_callback) {
         res = media->write_callback(session, media, frame);
 
      }
      break;
   case AST_FRAME_VIDEO:
      if (!media) {
         return 0;
      } else if (media->type != AST_MEDIA_TYPE_VIDEO) {
         ast_debug(3, "Channel %s stream %d is of type '%s', not video!\n",
            ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
         return 0;
      } else if (media->write_callback) {
         res = media->write_callback(session, media, frame);
      }
      break;
   case AST_FRAME_MODEM:
      if (!media) {
         return 0;
      } else if (media->type != AST_MEDIA_TYPE_IMAGE) {
         ast_debug(3, "Channel %s stream %d is of type '%s', not image!\n",
            ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
         return 0;
      } else if (media->write_callback) {
         res = media->write_callback(session, media, frame);
      }
      break;
   case AST_FRAME_CNG:
      break;
   case AST_FRAME_RTCP:
      /* We only support writing out feedback */
      if (frame->subclass.integer != AST_RTP_RTCP_PSFB || !media) {
         return 0;
      } else if (media->type != AST_MEDIA_TYPE_VIDEO) {
         ast_debug(3, "Channel %s stream %d is of type '%s', not video! Unable to write RTCP feedback.\n",
            ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
         return 0;
      } else if (media->write_callback) {
         res = media->write_callback(session, media, frame);
      }
      break;
   default:
      ast_log(LOG_WARNING, "Can't send %u type frames with PJSIP\n", frame->frametype);
      break;
   }
 
   return res;
}
 
static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame)
{
   return chan_pjsip_write_stream(ast, -1, frame);
}
 
/*! \brief Function called by core to change the underlying owner channel */
static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
 
   if (channel->session->channel != oldchan) {
      return -1;
   }
 
   /*
    * The masquerade has suspended the channel's session
    * serializer so we can safely change it outside of
    * the serializer thread.
    */
   channel->session->channel = newchan;
 
   set_channel_on_rtp_instance(channel->session, ast_channel_uniqueid(newchan));
 
   return 0;
}
 
/*! AO2 hash function for on hold UIDs */
static int uid_hold_hash_fn(const void *obj, const int flags)
{
   const char *key = obj;
 
   switch (flags & OBJ_SEARCH_MASK) {
   case OBJ_SEARCH_KEY:
      break;
   case OBJ_SEARCH_OBJECT:
      break;
   default:
      /* Hash can only work on something with a full key. */
      ast_assert(0);
      return 0;
   }
   return ast_str_hash(key);
}
 
/*! AO2 sort function for on hold UIDs */
static int uid_hold_sort_fn(const void *obj_left, const void *obj_right, const int flags)
{
   const char *left = obj_left;
   const char *right = obj_right;
   int cmp;
 
   switch (flags & OBJ_SEARCH_MASK) {
   case OBJ_SEARCH_OBJECT:
   case OBJ_SEARCH_KEY:
      cmp = strcmp(left, right);
      break;
   case OBJ_SEARCH_PARTIAL_KEY:
      cmp = strncmp(left, right, strlen(right));
      break;
   default:
      /* Sort can only work on something with a full or partial key. */
      ast_assert(0);
      cmp = 0;
      break;
   }
   return cmp;
}
 
static struct ao2_container *pjsip_uids_onhold;
 
/*!
 * \brief Add a channel ID to the list of PJSIP channels on hold
 *
 * \param chan_uid - Unique ID of the channel being put into the hold list
 *
 * \retval 0 Channel has been added to or was already in the hold list
 * \retval -1 Failed to add channel to the hold list
 */
static int chan_pjsip_add_hold(const char *chan_uid)
{
   RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);
 
   hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
   if (hold_uid) {
      /* Device is already on hold. Nothing to do. */
      return 0;
   }
 
   /* Device wasn't in hold list already. Create a new one. */
   hold_uid = ao2_alloc_options(strlen(chan_uid) + 1, NULL,
      AO2_ALLOC_OPT_LOCK_NOLOCK);
   if (!hold_uid) {
      return -1;
   }
 
   ast_copy_string(hold_uid, chan_uid, strlen(chan_uid) + 1);
 
   if (ao2_link(pjsip_uids_onhold, hold_uid) == 0) {
      return -1;
   }
 
   return 0;
}
 
/*!
 * \brief Remove a channel ID from the list of PJSIP channels on hold
 *
 * \param chan_uid - Unique ID of the channel being taken out of the hold list
 */
static void chan_pjsip_remove_hold(const char *chan_uid)
{
   ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY | OBJ_UNLINK | OBJ_NODATA);
}
 
/*!
 * \brief Determine whether a channel ID is in the list of PJSIP channels on hold
 *
 * \param chan_uid - Channel being checked
 *
 * \retval 0 The channel is not in the hold list
 * \retval 1 The channel is in the hold list
 */
static int chan_pjsip_get_hold(const char *chan_uid)
{
   RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);
 
   hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
   if (!hold_uid) {
      return 0;
   }
 
   return 1;
}
 
/*! \brief Function called to get the device state of an endpoint */
static int chan_pjsip_devicestate(const char *data)
{
   RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
   enum ast_device_state state = AST_DEVICE_UNKNOWN;
   RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
   struct ast_devstate_aggregate aggregate;
   int num, inuse = 0;
 
   if (!endpoint) {
      return AST_DEVICE_INVALID;
   }
 
   endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent),
      ast_endpoint_get_resource(endpoint->persistent));
 
   if (!endpoint_snapshot) {
      return AST_DEVICE_INVALID;
   }
 
   if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) {
      state = AST_DEVICE_UNAVAILABLE;
   } else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) {
      state = AST_DEVICE_NOT_INUSE;
   }
 
   if (!endpoint_snapshot->num_channels) {
      return state;
   }
 
   ast_devstate_aggregate_init(&aggregate);
 
   for (num = 0; num < endpoint_snapshot->num_channels; num++) {
      struct ast_channel_snapshot *snapshot;
 
      snapshot = ast_channel_snapshot_get_latest(endpoint_snapshot->channel_ids[num]);
      if (!snapshot) {
         continue;
      }
 
      if (chan_pjsip_get_hold(snapshot->base->uniqueid)) {
         ast_devstate_aggregate_add(&aggregate, AST_DEVICE_ONHOLD);
      } else {
         ast_devstate_aggregate_add(&aggregate, ast_state_chan2dev(snapshot->state));
      }
 
      if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
         (snapshot->state == AST_STATE_BUSY)) {
         inuse++;
      }
 
      ao2_ref(snapshot, -1);
   }
 
   if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) {
      state = AST_DEVICE_BUSY;
   } else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) {
      state = ast_devstate_aggregate_result(&aggregate);
   }
 
   return state;
}
 
/*! \brief Function called to query options on a channel */
static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   int res = -1;
   enum ast_t38_state state = T38_STATE_UNAVAILABLE;
 
   if (!channel) {
      return -1;
   }
 
   switch (option) {
   case AST_OPTION_T38_STATE:
      if (channel->session->endpoint->media.t38.enabled) {
         switch (channel->session->t38state) {
         case T38_LOCAL_REINVITE:
         case T38_PEER_REINVITE:
            state = T38_STATE_NEGOTIATING;
            break;
         case T38_ENABLED:
            state = T38_STATE_NEGOTIATED;
            break;
         case T38_REJECTED:
            state = T38_STATE_REJECTED;
            break;
         default:
            state = T38_STATE_UNKNOWN;
            break;
         }
      }
 
      *((enum ast_t38_state *) data) = state;
      res = 0;
 
      break;
   default:
      break;
   }
 
   return res;
}
 
static const char *chan_pjsip_get_uniqueid(struct ast_channel *ast)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   char *uniqueid = ast_threadstorage_get(&uniqueid_threadbuf, UNIQUEID_BUFSIZE);
 
   if (!uniqueid) {
      return "";
   }
 
   ast_copy_pj_str(uniqueid, &channel->session->inv_session->dlg->call_id->id, UNIQUEID_BUFSIZE);
 
   return uniqueid;
}
 
struct indicate_data {
   struct ast_sip_session *session;
   int condition;
   int response_code;
   void *frame_data;
   size_t datalen;
};
 
static void indicate_data_destroy(void *obj)
{
   struct indicate_data *ind_data = obj;
 
   ast_free(ind_data->frame_data);
   ao2_ref(ind_data->session, -1);
}
 
static struct indicate_data *indicate_data_alloc(struct ast_sip_session *session,
      int condition, int response_code, const void *frame_data, size_t datalen)
{
   struct indicate_data *ind_data = ao2_alloc(sizeof(*ind_data), indicate_data_destroy);
 
   if (!ind_data) {
      return NULL;
   }
 
   ind_data->frame_data = ast_malloc(datalen);
   if (!ind_data->frame_data) {
      ao2_ref(ind_data, -1);
      return NULL;
   }
 
   memcpy(ind_data->frame_data, frame_data, datalen);
   ind_data->datalen = datalen;
   ind_data->condition = condition;
   ind_data->response_code = response_code;
   ao2_ref(session, +1);
   ind_data->session = session;
 
   return ind_data;
}
 
static int indicate(void *data)
{
   pjsip_tx_data *packet = NULL;
   struct indicate_data *ind_data = data;
   struct ast_sip_session *session = ind_data->session;
   int response_code = ind_data->response_code;
 
   if ((session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED) &&
      (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS)) {
      ast_sip_session_send_response(session, packet);
   }
 
   ao2_ref(ind_data, -1);
 
   return 0;
}
 
/*! \brief Send SIP INFO with video update request */
static int transmit_info_with_vidupdate(void *data)
{
   const char * xml =
      "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
      " <media_control>\r\n"
      "  <vc_primitive>\r\n"
      "   <to_encoder>\r\n"
      "    <picture_fast_update/>\r\n"
      "   </to_encoder>\r\n"
      "  </vc_primitive>\r\n"
      " </media_control>\r\n";
 
   const struct ast_sip_body body = {
      .type = "application",
      .subtype = "media_control+xml",
      .body_text = xml
   };
 
   RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
   struct pjsip_tx_data *tdata;
 
   if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
      ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
         session->inv_session->cause,
         pjsip_get_status_text(session->inv_session->cause)->ptr);
      return -1;
   }
 
   if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, NULL, &tdata)) {
      ast_log(LOG_ERROR, "Could not create text video update INFO request\n");
      return -1;
   }
   if (ast_sip_add_body(tdata, &body)) {
      ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n");
      return -1;
   }
   ast_sip_session_send_request(session, tdata);
 
   return 0;
}
 
/*!
 * \internal
 * \brief TRUE if a COLP update can be sent to the peer.
 * \since 13.3.0
 *
 * \param session The session to see if the COLP update is allowed.
 *
 * \retval 0 Update is not allowed.
 * \retval 1 Update is allowed.
 */
static int is_colp_update_allowed(struct ast_sip_session *session)
{
   struct ast_party_id connected_id;
   int update_allowed = 0;
 
   if (!session->endpoint->id.send_connected_line
      || (!session->endpoint->id.send_pai && !session->endpoint->id.send_rpid)) {
      return 0;
   }
 
   /*
    * Check if privacy allows the update.  Check while the channel
    * is locked so we can work with the shallow connected_id copy.
    */
   ast_channel_lock(session->channel);
   connected_id = ast_channel_connected_effective_id(session->channel);
   if (connected_id.number.valid
      && (session->endpoint->id.trust_outbound
         || (ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED)) {
      update_allowed = 1;
   }
   ast_channel_unlock(session->channel);
 
   return update_allowed;
}
 
/*! \brief Update connected line information */
static int update_connected_line_information(void *data)
{
   struct ast_sip_session *session = data;
 
   if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
      ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
         session->inv_session->cause,
         pjsip_get_status_text(session->inv_session->cause)->ptr);
      ao2_ref(session, -1);
      return -1;
   }
 
   if (ast_channel_state(session->channel) == AST_STATE_UP
      || session->inv_session->role == PJSIP_ROLE_UAC) {
      if (is_colp_update_allowed(session)) {
         enum ast_sip_session_refresh_method method;
         int generate_new_sdp;
 
         method = session->endpoint->id.refresh_method;
         if (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE) {
            method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
         }
 
         /* Only the INVITE method actually needs SDP, UPDATE can do without */
         generate_new_sdp = (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE);
 
         ast_sip_session_refresh(session, NULL, NULL, NULL, method, generate_new_sdp, NULL);
      }
   } else if (session->endpoint->id.rpid_immediate
      && session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED
      && is_colp_update_allowed(session)) {
      int response_code = 0;
 
      if (ast_channel_state(session->channel) == AST_STATE_RING) {
         response_code = !session->endpoint->inband_progress ? 180 : 183;
      } else if (ast_channel_state(session->channel) == AST_STATE_RINGING) {
         response_code = 183;
      }
 
      if (response_code) {
         struct pjsip_tx_data *packet = NULL;
 
         if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
            ast_sip_session_send_response(session, packet);
         }
      }
   }
 
   ao2_ref(session, -1);
   return 0;
}
 
/*! \brief Callback which changes the value of locally held on the media stream */
static void local_hold_set_state(struct ast_sip_session_media *session_media, unsigned int held)
{
   if (session_media) {
      session_media->locally_held = held;
   }
}
 
/*! \brief Update local hold state and send a re-INVITE with the new SDP */
static int remote_send_hold_refresh(struct ast_sip_session *session, unsigned int held)
{
   AST_VECTOR_CALLBACK_VOID(&session->active_media_state->sessions, local_hold_set_state, held);
   ast_sip_session_refresh(session, NULL, NULL, NULL, AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1, NULL);
   ao2_ref(session, -1);
 
   return 0;
}
 
/*! \brief Update local hold state to be held */
static int remote_send_hold(void *data)
{
   return remote_send_hold_refresh(data, 1);
}
 
/*! \brief Update local hold state to be unheld */
static int remote_send_unhold(void *data)
{
   return remote_send_hold_refresh(data, 0);
}
 
struct topology_change_refresh_data {
   struct ast_sip_session *session;
   struct ast_sip_session_media_state *media_state;
};
 
static void topology_change_refresh_data_free(struct topology_change_refresh_data *refresh_data)
{
   ao2_cleanup(refresh_data->session);
 
   ast_sip_session_media_state_free(refresh_data->media_state);
   ast_free(refresh_data);
}
 
static struct topology_change_refresh_data *topology_change_refresh_data_alloc(
   struct ast_sip_session *session, const struct ast_stream_topology *topology)
{
   struct topology_change_refresh_data *refresh_data;
 
   refresh_data = ast_calloc(1, sizeof(*refresh_data));
   if (!refresh_data) {
      return NULL;
   }
 
   refresh_data->session = ao2_bump(session);
   refresh_data->media_state = ast_sip_session_media_state_alloc();
   if (!refresh_data->media_state) {
      topology_change_refresh_data_free(refresh_data);
      return NULL;
   }
   refresh_data->media_state->topology = ast_stream_topology_clone(topology);
   if (!refresh_data->media_state->topology) {
      topology_change_refresh_data_free(refresh_data);
      return NULL;
   }
 
   return refresh_data;
}
 
static int on_topology_change_response(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
   SCOPE_ENTER(3, "%s: Received response code %d.  PT: %s  AT: %s\n", ast_sip_session_get_name(session),
      rdata->msg_info.msg->line.status.code,
      ast_str_tmp(256, ast_stream_topology_to_str(session->pending_media_state->topology, &STR_TMP)),
      ast_str_tmp(256, ast_stream_topology_to_str(session->active_media_state->topology, &STR_TMP)));
 
 
   if (PJSIP_IS_STATUS_IN_CLASS(rdata->msg_info.msg->line.status.code, 200)) {
      /* The topology was changed to something new so give notice to what requested
       * it so it queries the channel and updates accordingly.
       */
      if (session->channel) {
         ast_queue_control(session->channel, AST_CONTROL_STREAM_TOPOLOGY_CHANGED);
         SCOPE_EXIT_RTN_VALUE(0, "%s: Queued topology change frame\n", ast_sip_session_get_name(session));
      }
      SCOPE_EXIT_RTN_VALUE(0, "%s: No channel?  Can't queue topology change frame\n", ast_sip_session_get_name(session));
   } else if (300 <= rdata->msg_info.msg->line.status.code) {
      /* The topology change failed, so drop the current pending media state */
      ast_sip_session_media_state_reset(session->pending_media_state);
      SCOPE_EXIT_RTN_VALUE(0, "%s: response code > 300.  Resetting pending media state\n", ast_sip_session_get_name(session));
   }
 
   SCOPE_EXIT_RTN_VALUE(0, "%s: Nothing to do\n", ast_sip_session_get_name(session));
}
 
static int send_topology_change_refresh(void *data)
{
   struct topology_change_refresh_data *refresh_data = data;
   struct ast_sip_session *session = refresh_data->session;
   int ret;
   SCOPE_ENTER(3, "%s: %s\n", ast_sip_session_get_name(session),
      ast_str_tmp(256, ast_stream_topology_to_str(refresh_data->media_state->topology, &STR_TMP)));
 
 
   ret = ast_sip_session_refresh(session, NULL, NULL, on_topology_change_response,
      AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1, refresh_data->media_state);
   refresh_data->media_state = NULL;
   topology_change_refresh_data_free(refresh_data);
 
   SCOPE_EXIT_RTN_VALUE(ret, "%s\n", ast_sip_session_get_name(session));
}
 
static int handle_topology_request_change(struct ast_sip_session *session,
   const struct ast_stream_topology *proposed)
{
   struct topology_change_refresh_data *refresh_data;
   int res;
   SCOPE_ENTER(1);
 
   refresh_data = topology_change_refresh_data_alloc(session, proposed);
   if (!refresh_data) {
      SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create refresh_data\n");
   }
 
   res = ast_sip_push_task(session->serializer, send_topology_change_refresh, refresh_data);
   if (res) {
      topology_change_refresh_data_free(refresh_data);
   }
   SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
}
 
/*! \brief Function called by core to ask the channel to indicate some sort of condition */
static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   struct ast_sip_session_media *media;
   int response_code = 0;
   int res = 0;
   char *device_buf;
   size_t device_buf_size;
   int i;
   const struct ast_stream_topology *topology;
   struct ast_frame f = {
      .frametype = AST_FRAME_CONTROL,
      .subclass = {
         .integer = condition
      },
      .datalen = datalen,
      .data.ptr = (void *)data,
   };
   char condition_name[256];
   SCOPE_ENTER(3, "%s: Indicated %s\n", ast_channel_name(ast),
      ast_frame_subclass2str(&f, condition_name, sizeof(condition_name), NULL, 0));
 
   switch (condition) {
   case AST_CONTROL_RINGING:
      if (ast_channel_state(ast) == AST_STATE_RING) {
         if (channel->session->endpoint->inband_progress ||
            (channel->session->inv_session && channel->session->inv_session->neg &&
            pjmedia_sdp_neg_get_state(channel->session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE)) {
            response_code = 183;
            res = -1;
         } else {
            response_code = 180;
         }
      } else {
         res = -1;
      }
      ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
      break;
   case AST_CONTROL_BUSY:
      if (ast_channel_state(ast) != AST_STATE_UP) {
         response_code = 486;
      } else {
         res = -1;
      }
      break;
   case AST_CONTROL_CONGESTION:
      if (ast_channel_state(ast) != AST_STATE_UP) {
         response_code = 503;
      } else {
         res = -1;
      }
      break;
   case AST_CONTROL_INCOMPLETE:
      if (ast_channel_state(ast) != AST_STATE_UP) {
         response_code = 484;
      } else {
         res = -1;
      }
      break;
   case AST_CONTROL_PROCEEDING:
      if (ast_channel_state(ast) != AST_STATE_UP) {
         response_code = 100;
      } else {
         res = -1;
      }
      break;
   case AST_CONTROL_PROGRESS:
      if (ast_channel_state(ast) != AST_STATE_UP) {
         response_code = 183;
      } else {
         res = -1;
      }
      ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
      break;
   case AST_CONTROL_VIDUPDATE:
      for (i = 0; i < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions); ++i) {
         media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, i);
         if (!media || media->type != AST_MEDIA_TYPE_VIDEO) {
            continue;
         }
         if (media->rtp) {
            /* FIXME: Only use this for VP8. Additional work would have to be done to
             * fully support other video codecs */
 
            if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL ||
               ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp9) != AST_FORMAT_CMP_NOT_EQUAL ||
               ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h265) != AST_FORMAT_CMP_NOT_EQUAL ||
               (channel->session->endpoint->media.webrtc &&
                ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h264) != AST_FORMAT_CMP_NOT_EQUAL)) {
               /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
                * RTP engine would provide a way to externally write/schedule RTCP
                * packets */
               struct ast_frame fr;
               fr.frametype = AST_FRAME_CONTROL;
               fr.subclass.integer = AST_CONTROL_VIDUPDATE;
               res = ast_rtp_instance_write(media->rtp, &fr);
            } else {
               ao2_ref(channel->session, +1);
               if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
                  ao2_cleanup(channel->session);
               }
            }
            ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Success");
         } else {
            ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Failure");
            res = -1;
         }
      }
      /* XXX If there were no video streams, then this should set
       * res to -1
       */
      break;
   case AST_CONTROL_CONNECTED_LINE:
      ao2_ref(channel->session, +1);
      if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
         ao2_cleanup(channel->session);
      }
      break;
   case AST_CONTROL_UPDATE_RTP_PEER:
      break;
   case AST_CONTROL_PVT_CAUSE_CODE:
      res = -1;
      break;
   case AST_CONTROL_MASQUERADE_NOTIFY:
      ast_assert(datalen == sizeof(int));
      if (*(int *) data) {
         /*
          * Masquerade is beginning:
          * Wait for session serializer to get suspended.
          */
         ast_channel_unlock(ast);
         ast_sip_session_suspend(channel->session);
         ast_channel_lock(ast);
      } else {
         /*
          * Masquerade is complete:
          * Unsuspend the session serializer.
          */
         ast_sip_session_unsuspend(channel->session);
      }
      break;
   case AST_CONTROL_HOLD:
      chan_pjsip_add_hold(ast_channel_uniqueid(ast));
      device_buf_size = strlen(ast_channel_name(ast)) + 1;
      device_buf = alloca(device_buf_size);
      ast_channel_get_device_name(ast, device_buf, device_buf_size);
      ast_devstate_changed_literal(AST_DEVICE_ONHOLD, 1, device_buf);
      if (!channel->session->moh_passthrough) {
         ast_moh_start(ast, data, NULL);
      } else {
         if (ast_sip_push_task(channel->session->serializer, remote_send_hold, ao2_bump(channel->session))) {
            ast_log(LOG_WARNING, "Could not queue task to remotely put session '%s' on hold with endpoint '%s'\n",
               ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint));
            ao2_ref(channel->session, -1);
         }
      }
      break;
   case AST_CONTROL_UNHOLD:
      chan_pjsip_remove_hold(ast_channel_uniqueid(ast));
      device_buf_size = strlen(ast_channel_name(ast)) + 1;
      device_buf = alloca(device_buf_size);
      ast_channel_get_device_name(ast, device_buf, device_buf_size);
      ast_devstate_changed_literal(AST_DEVICE_UNKNOWN, 1, device_buf);
      if (!channel->session->moh_passthrough) {
         ast_moh_stop(ast);
      } else {
         if (ast_sip_push_task(channel->session->serializer, remote_send_unhold, ao2_bump(channel->session))) {
            ast_log(LOG_WARNING, "Could not queue task to remotely take session '%s' off hold with endpoint '%s'\n",
               ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint));
            ao2_ref(channel->session, -1);
         }
      }
      break;
   case AST_CONTROL_SRCUPDATE:
      break;
   case AST_CONTROL_SRCCHANGE:
      break;
   case AST_CONTROL_REDIRECTING:
      if (ast_channel_state(ast) != AST_STATE_UP) {
         response_code = 181;
      } else {
         res = -1;
      }
      break;
   case AST_CONTROL_T38_PARAMETERS:
      res = 0;
 
      if (channel->session->t38state == T38_PEER_REINVITE) {
         const struct ast_control_t38_parameters *parameters = data;
 
         if (parameters->request_response == AST_T38_REQUEST_PARMS) {
            res = AST_T38_REQUEST_PARMS;
         }
      }
 
      break;
   case AST_CONTROL_STREAM_TOPOLOGY_REQUEST_CHANGE:
      topology = data;
      ast_trace(-1, "%s: New topology: %s\n", ast_channel_name(ast),
         ast_str_tmp(256, ast_stream_topology_to_str(topology, &STR_TMP)));
      res = handle_topology_request_change(channel->session, topology);
      break;
   case AST_CONTROL_STREAM_TOPOLOGY_CHANGED:
      break;
   case AST_CONTROL_STREAM_TOPOLOGY_SOURCE_CHANGED:
      break;
   case -1:
      res = -1;
      break;
   default:
      ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
      res = -1;
      break;
   }
 
   if (response_code) {
      struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
 
      if (!ind_data) {
         SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc indicate data\n", ast_channel_name(ast));
      }
 
      if (ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
         ast_log(LOG_ERROR, "%s: Cannot send response code %d to endpoint %s. Could not queue task properly\n",
               ast_channel_name(ast), response_code, ast_sorcery_object_get_id(channel->session->endpoint));
         ao2_cleanup(ind_data);
         res = -1;
      }
   }
 
   SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_channel_name(ast));
}
 
struct transfer_data {
   struct ast_sip_session *session;
   char *target;
};
 
static void transfer_data_destroy(void *obj)
{
   struct transfer_data *trnf_data = obj;
 
   ast_free(trnf_data->target);
   ao2_cleanup(trnf_data->session);
}
 
static struct transfer_data *transfer_data_alloc(struct ast_sip_session *session, const char *target)
{
   struct transfer_data *trnf_data = ao2_alloc(sizeof(*trnf_data), transfer_data_destroy);
 
   if (!trnf_data) {
      return NULL;
   }
 
   if (!(trnf_data->target = ast_strdup(target))) {
      ao2_ref(trnf_data, -1);
      return NULL;
   }
 
   ao2_ref(session, +1);
   trnf_data->session = session;
 
   return trnf_data;
}
 
static void transfer_redirect(struct ast_sip_session *session, const char *target)
{
   pjsip_tx_data *packet;
   enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
   pjsip_contact_hdr *contact;
   pj_str_t tmp;
 
   if (pjsip_inv_end_session(session->inv_session, 302, NULL, &packet) != PJ_SUCCESS
      || !packet) {
      ast_log(LOG_WARNING, "Failed to redirect PJSIP session for channel %s\n",
         ast_channel_name(session->channel));
      message = AST_TRANSFER_FAILED;
      ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
 
      return;
   }
 
   if (!(contact = pjsip_msg_find_hdr(packet->msg, PJSIP_H_CONTACT, NULL))) {
      contact = pjsip_contact_hdr_create(packet->pool);
   }
 
   pj_strdup2_with_null(packet->pool, &tmp, target);
   if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) {
      ast_log(LOG_WARNING, "Failed to parse destination URI '%s' for channel %s\n",
         target, ast_channel_name(session->channel));
      message = AST_TRANSFER_FAILED;
      ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
      pjsip_tx_data_dec_ref(packet);
 
      return;
   }
   pjsip_msg_add_hdr(packet->msg, (pjsip_hdr *) contact);
 
   ast_sip_session_send_response(session, packet);
   ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
}
 
/*! \brief REFER Callback module, used to attach session data structure to subscription */
static pjsip_module refer_callback_module = {
   .name = { "REFER Callback", 14 },
   .id = -1,
};
 
/*!
 * \brief Callback function to report status of implicit REFER-NOTIFY subscription.
 *
 * This function will be called on any state change in the REFER-NOTIFY subscription.
 * Its primary purpose is to report SUCCESS/FAILURE of a transfer initiated via
 * \ref transfer_refer as well as to terminate the subscription, if necessary.
 */
static void xfer_client_on_evsub_state(pjsip_evsub *sub, pjsip_event *event)
{
   struct ast_channel *chan;
   enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
   int res = 0;
 
   if (!event) {
      return;
   }
 
   chan = pjsip_evsub_get_mod_data(sub, refer_callback_module.id);
   if (!chan) {
      return;
   }
 
   if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACCEPTED) {
      /* Check if subscription is suppressed and terminate and send completion code, if so. */
      pjsip_rx_data *rdata;
      pjsip_generic_string_hdr *refer_sub;
      const pj_str_t REFER_SUB = { "Refer-Sub", 9 };
 
      ast_debug(3, "Transfer accepted on channel %s\n", ast_channel_name(chan));
 
      /* Check if response message */
      if (event->type == PJSIP_EVENT_TSX_STATE && event->body.tsx_state.type == PJSIP_EVENT_RX_MSG) {
         rdata = event->body.tsx_state.src.rdata;
 
         /* Find Refer-Sub header */
         refer_sub = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &REFER_SUB, NULL);
 
         /* Check if subscription is suppressed. If it is, the far end will not terminate it,
          * and the subscription will remain active until it times out.  Terminating it here
          * eliminates the unnecessary timeout.
          */
         if (refer_sub && !pj_stricmp2(&refer_sub->hvalue, "false")) {
            /* Since no subscription is desired, assume that call has been transferred successfully. */
            /* Channel reference will be released at end of function */
            /* Terminate subscription. */
            pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
            pjsip_evsub_terminate(sub, PJ_TRUE);
            res = -1;
         }
      }
   } else if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACTIVE ||
         pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
      /* Check for NOTIFY complete or error. */
      pjsip_msg *msg;
      pjsip_msg_body *body;
      pjsip_status_line status_line = { .code = 0 };
      pj_bool_t is_last;
      pj_status_t status;
 
      if (event->type == PJSIP_EVENT_TSX_STATE && event->body.tsx_state.type == PJSIP_EVENT_RX_MSG) {
         pjsip_rx_data *rdata;
 
         rdata = event->body.tsx_state.src.rdata;
         msg = rdata->msg_info.msg;
 
         if (msg->type == PJSIP_REQUEST_MSG) {
            if (!pjsip_method_cmp(&msg->line.req.method, pjsip_get_notify_method())) {
               body = msg->body;
               if (body && !pj_stricmp2(&body->content_type.type, "message")
                  && !pj_stricmp2(&body->content_type.subtype, "sipfrag")) {
                  pjsip_parse_status_line((char *)body->data, body->len, &status_line);
               }
            }
         } else {
            status_line.code = msg->line.status.code;
            status_line.reason = msg->line.status.reason;
         }
      } else {
         status_line.code = 500;
         status_line.reason = *pjsip_get_status_text(500);
      }
 
      is_last = (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED);
      /* If the status code is >= 200, the subscription is finished. */
      if (status_line.code >= 200 || is_last) {
         res = -1;
 
         /* If the subscription has terminated, return AST_TRANSFER_SUCCESS for 2XX.
          * Return AST_TRANSFER_FAILED for any code < 200.
          * Otherwise, return the status code.
          * The subscription should not terminate for any code < 200,
          * but if it does, that constitutes a failure. */
         if (status_line.code < 200) {
            message = AST_TRANSFER_FAILED;
         } else if (status_line.code >= 300) {
            message = status_line.code;
         }
 
         /* If subscription not terminated and subscription is finished (status code >= 200)
          * terminate it */
         if (!is_last) {
            pjsip_tx_data *tdata;
 
            status = pjsip_evsub_initiate(sub, pjsip_get_subscribe_method(), 0, &tdata);
            if (status == PJ_SUCCESS) {
               pjsip_evsub_send_request(sub, tdata);
            }
         }
         /* Finished. Remove session from subscription */
         pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
         ast_debug(3, "Transfer channel %s completed: %d %.*s (%s)\n",
               ast_channel_name(chan),
               status_line.code,
               (int)status_line.reason.slen, status_line.reason.ptr,
               (message == AST_TRANSFER_SUCCESS) ? "Success" : "Failure");
      }
   }
 
   if (res) {
      ast_queue_control_data(chan, AST_CONTROL_TRANSFER, &message, sizeof(message));
      ao2_ref(chan, -1);
   }
}
 
static void transfer_refer(struct ast_sip_session *session, const char *target)
{
   pjsip_evsub *sub;
   enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
   pj_str_t tmp;
   pjsip_tx_data *packet;
   const char *ref_by_val;
   char local_info[pj_strlen(&session->inv_session->dlg->local.info_str) + 1];
   struct pjsip_evsub_user xfer_cb;
   struct ast_channel *chan = session->channel;
 
   pj_bzero(&xfer_cb, sizeof(xfer_cb));
   xfer_cb.on_evsub_state = &xfer_client_on_evsub_state;
 
   if (pjsip_xfer_create_uac(session->inv_session->dlg, &xfer_cb, &sub) != PJ_SUCCESS) {
      message = AST_TRANSFER_FAILED;
      ast_queue_control_data(chan, AST_CONTROL_TRANSFER, &message, sizeof(message));
 
      return;
   }
 
   /* refer_callback_module requires a reference to chan
    * which will be released in xfer_client_on_evsub_state()
    * when the implicit REFER subscription terminates */
   pjsip_evsub_set_mod_data(sub, refer_callback_module.id, chan);
   ao2_ref(chan, +1);
 
   if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
      goto failure;
   }
 
   ref_by_val = pbx_builtin_getvar_helper(chan, "SIPREFERREDBYHDR");
   if (!ast_strlen_zero(ref_by_val)) {
      ast_sip_add_header(packet, "Referred-By", ref_by_val);
   } else {
      ast_copy_pj_str(local_info, &session->inv_session->dlg->local.info_str, sizeof(local_info));
      ast_sip_add_header(packet, "Referred-By", local_info);
   }
 
   if (pjsip_xfer_send_request(sub, packet) == PJ_SUCCESS) {
      return;
   }
 
failure:
   message = AST_TRANSFER_FAILED;
   ast_queue_control_data(chan, AST_CONTROL_TRANSFER, &message, sizeof(message));
   pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
   pjsip_evsub_terminate(sub, PJ_FALSE);
 
   ao2_ref(chan, -1);
}
 
static int transfer(void *data)
{
   struct transfer_data *trnf_data = data;
   struct ast_sip_endpoint *endpoint = NULL;
   struct ast_sip_contact *contact = NULL;
   const char *target = trnf_data->target;
 
   if (trnf_data->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
      ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
         trnf_data->session->inv_session->cause,
         pjsip_get_status_text(trnf_data->session->inv_session->cause)->ptr);
   } else {
      /* See if we have an endpoint; if so, use its contact */
      endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", target);
      if (endpoint) {
         contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
         if (contact && !ast_strlen_zero(contact->uri)) {
            target = contact->uri;
         }
      }
 
      if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
         transfer_redirect(trnf_data->session, target);
      } else {
         transfer_refer(trnf_data->session, target);
      }
   }
 
   ao2_ref(trnf_data, -1);
   ao2_cleanup(endpoint);
   ao2_cleanup(contact);
   return 0;
}
 
/*! \brief Function called by core for Asterisk initiated transfer */
static int chan_pjsip_transfer(struct ast_channel *chan, const char *target)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
   struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
 
   if (!trnf_data) {
      return -1;
   }
 
   if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
      ast_log(LOG_WARNING, "Error requesting transfer\n");
      ao2_cleanup(trnf_data);
      return -1;
   }
 
   return 0;
}
 
/*! \brief Function called by core to start a DTMF digit */
static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
   struct ast_sip_session_media *media;
 
   media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
 
   switch (channel->session->dtmf) {
   case AST_SIP_DTMF_RFC_4733:
      if (!media || !media->rtp) {
         return 0;
      }
 
      ast_rtp_instance_dtmf_begin(media->rtp, digit);
      break;
   case AST_SIP_DTMF_AUTO:
      if (!media || !media->rtp) {
         return 0;
      }
 
      if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
         return -1;
      }
 
      ast_rtp_instance_dtmf_begin(media->rtp, digit);
      break;
   case AST_SIP_DTMF_AUTO_INFO:
      if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_NONE)) {
         return 0;
      }
      ast_rtp_instance_dtmf_begin(media->rtp, digit);
      break;
   case AST_SIP_DTMF_NONE:
      break;
   case AST_SIP_DTMF_INBAND:
      return -1;
   default:
      break;
   }
 
   return 0;
}
 
struct info_dtmf_data {
   struct ast_sip_session *session;
   char digit;
   unsigned int duration;
};
 
static void info_dtmf_data_destroy(void *obj)
{
   struct info_dtmf_data *dtmf_data = obj;
   ao2_ref(dtmf_data->session, -1);
}
 
static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
{
   struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy);
   if (!dtmf_data) {
      return NULL;
   }
   ao2_ref(session, +1);
   dtmf_data->session = session;
   dtmf_data->digit = digit;
   dtmf_data->duration = duration;
   return dtmf_data;
}
 
static int transmit_info_dtmf(void *data)
{
   RAII_VAR(struct info_dtmf_data *, dtmf_data, data, ao2_cleanup);
 
   struct ast_sip_session *session = dtmf_data->session;
   struct pjsip_tx_data *tdata;
 
   RAII_VAR(struct ast_str *, body_text, NULL, ast_free_ptr);
 
   struct ast_sip_body body = {
      .type = "application",
      .subtype = "dtmf-relay",
   };
 
   if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
      ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
         session->inv_session->cause,
         pjsip_get_status_text(session->inv_session->cause)->ptr);
      return -1;
   }
 
   if (!(body_text = ast_str_create(32))) {
      ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n");
      return -1;
   }
   ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration);
 
   body.body_text = ast_str_buffer(body_text);
 
   if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, NULL, &tdata)) {
      ast_log(LOG_ERROR, "Could not create DTMF INFO request\n");
      return -1;
   }
   if (ast_sip_add_body(tdata, &body)) {
      ast_log(LOG_ERROR, "Could not add body to DTMF INFO request\n");
      pjsip_tx_data_dec_ref(tdata);
      return -1;
   }
   ast_sip_session_send_request(session, tdata);
 
   return 0;
}
 
/*! \brief Function called by core to stop a DTMF digit */
static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   struct ast_sip_session_media *media;
 
   if (!channel || !channel->session) {
      /* This happens when the channel is hungup while a DTMF digit is playing. See ASTERISK-28086 */
      ast_debug(3, "Channel %s disappeared while calling digit_end\n", ast_channel_name(ast));
      return -1;
   }
 
   media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
 
   switch (channel->session->dtmf) {
   case AST_SIP_DTMF_AUTO_INFO:
   {
      if (!media || !media->rtp) {
         return 0;
      }
 
      if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_NONE) {
         ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast));
         ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
         break;
      }
      /* If RFC_4733 was not negotiated, fail through to the DTMF_INFO processing */
      ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 NOT negotiated using INFO instead.\n", ast_channel_name(ast));
   }
 
   case AST_SIP_DTMF_INFO:
   {
      struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
 
      if (!dtmf_data) {
         return -1;
      }
 
      if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
         ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
         ao2_cleanup(dtmf_data);
         return -1;
      }
      break;
   }
   case AST_SIP_DTMF_RFC_4733:
      if (!media || !media->rtp) {
         return 0;
      }
 
      ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
      break;
   case AST_SIP_DTMF_AUTO:
      if (!media || !media->rtp) {
         return 0;
      }
 
      if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
          return -1;
      }
 
      ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
      break;
   case AST_SIP_DTMF_NONE:
      break;
   case AST_SIP_DTMF_INBAND:
      return -1;
   }
 
   return 0;
}
 
static void update_initial_connected_line(struct ast_sip_session *session)
{
   struct ast_party_connected_line connected;
 
   /*
    * Use the channel CALLERID() as the initial connected line data.
    * The core or a predial handler may have supplied missing values
    * from the session->endpoint->id.self about who we are calling.
    */
   ast_channel_lock(session->channel);
   ast_party_id_copy(&session->id, &ast_channel_caller(session->channel)->id);
   ast_channel_unlock(session->channel);
 
   /* Supply initial connected line information if available. */
   if (!session->id.number.valid && !session->id.name.valid) {
      return;
   }
 
   ast_party_connected_line_init(&connected);
   connected.id = session->id;
   connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
 
   ast_channel_queue_connected_line_update(session->channel, &connected, NULL);
}
 
static int call(void *data)
{
   struct ast_sip_channel_pvt *channel = data;
   struct ast_sip_session *session = channel->session;
   pjsip_tx_data *tdata;
   int res = 0;
   SCOPE_ENTER(1, "%s Topology: %s\n",
      ast_sip_session_get_name(session),
      ast_str_tmp(256, ast_stream_topology_to_str(channel->session->pending_media_state->topology, &STR_TMP))
      );
 
 
   res = ast_sip_session_create_invite(session, &tdata);
 
   if (res) {
      ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
      ast_queue_hangup(session->channel);
   } else {
      set_channel_on_rtp_instance(session, ast_channel_uniqueid(session->channel));
      update_initial_connected_line(session);
      ast_sip_session_send_request(session, tdata);
   }
   ao2_ref(channel, -1);
   SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
}
 
/*! \brief Function called by core to actually start calling a remote party */
static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   SCOPE_ENTER(1, "%s Topology: %s\n", ast_sip_session_get_name(channel->session),
      ast_str_tmp(256, ast_stream_topology_to_str(channel->session->pending_media_state->topology, &STR_TMP)));
 
   ao2_ref(channel, +1);
   if (ast_sip_push_task(channel->session->serializer, call, channel)) {
      ast_log(LOG_WARNING, "Error attempting to place outbound call to '%s'\n", dest);
      ao2_cleanup(channel);
      SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
   }
 
   SCOPE_EXIT_RTN_VALUE(0, "'call' task pushed\n");
}
 
/*! \brief Internal function which translates from Asterisk cause codes to SIP response codes */
static int hangup_cause2sip(int cause)
{
   switch (cause) {
   case AST_CAUSE_UNALLOCATED:             /* 1 */
   case AST_CAUSE_NO_ROUTE_DESTINATION:    /* 3 IAX2: Can't find extension in context */
   case AST_CAUSE_NO_ROUTE_TRANSIT_NET:    /* 2 */
      return 404;
   case AST_CAUSE_CONGESTION:              /* 34 */
   case AST_CAUSE_SWITCH_CONGESTION:       /* 42 */
      return 503;
   case AST_CAUSE_NO_USER_RESPONSE:        /* 18 */
      return 408;
   case AST_CAUSE_NO_ANSWER:               /* 19 */
   case AST_CAUSE_UNREGISTERED:        /* 20 */
      return 480;
   case AST_CAUSE_CALL_REJECTED:           /* 21 */
      return 403;
   case AST_CAUSE_NUMBER_CHANGED:          /* 22 */
      return 410;
   case AST_CAUSE_NORMAL_UNSPECIFIED:      /* 31 */
      return 480;
   case AST_CAUSE_INVALID_NUMBER_FORMAT:
      return 484;
   case AST_CAUSE_USER_BUSY:
      return 486;
   case AST_CAUSE_FAILURE:
      return 500;
   case AST_CAUSE_FACILITY_REJECTED:       /* 29 */
      return 501;
   case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
      return 503;
   case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
      return 502;
   case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:       /* Can't find codec to connect to host */
      return 488;
   case AST_CAUSE_INTERWORKING:    /* Unspecified Interworking issues */
      return 500;
   case AST_CAUSE_NOTDEFINED:
   default:
      ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
      return 0;
   }
 
   /* Never reached */
   return 0;
}
 
struct hangup_data {
   int cause;
   struct ast_channel *chan;
};
 
static void hangup_data_destroy(void *obj)
{
   struct hangup_data *h_data = obj;
 
   h_data->chan = ast_channel_unref(h_data->chan);
}
 
static struct hangup_data *hangup_data_alloc(int cause, struct ast_channel *chan)
{
   struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
 
   if (!h_data) {
      return NULL;
   }
 
   h_data->cause = cause;
   h_data->chan = ast_channel_ref(chan);
 
   return h_data;
}
 
/*! \brief Clear a channel from a session along with its PVT */
static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast)
{
   session->channel = NULL;
   set_channel_on_rtp_instance(session, "");
   ast_channel_tech_pvt_set(ast, NULL);
}
 
static int hangup(void *data)
{
   struct hangup_data *h_data = data;
   struct ast_channel *ast = h_data->chan;
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
 
   /*
    * Before cleaning we have to ensure that channel or its session is not NULL
    * we have seen rare case when taskprocessor calls hangup but channel is NULL
    * due to SIP session timeout and answer happening at the same time
    */
   if (channel) {
      struct ast_sip_session *session = channel->session;
      if (session) {
         int cause = h_data->cause;
 
         /*
         * It's possible that session_terminate might cause the session to be destroyed
         * immediately so we need to keep a reference to it so we can NULL session->channel
         * afterwards.
         */
         ast_sip_session_terminate(ao2_bump(session), cause);
         clear_session_and_channel(session, ast);
         ao2_cleanup(session);
      }
      ao2_cleanup(channel);
   }
   ao2_cleanup(h_data);
   SCOPE_EXIT_RTN_VALUE(0);
}
 
/*! \brief Function called by core to hang up a PJSIP session */
static int chan_pjsip_hangup(struct ast_channel *ast)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   int cause;
   struct hangup_data *h_data;
   SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
 
   if (!channel || !channel->session) {
      SCOPE_EXIT_RTN_VALUE(-1, "No channel or session\n");
   }
 
   cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
   h_data = hangup_data_alloc(cause, ast);
 
   if (!h_data) {
      goto failure;
   }
 
   if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
      ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
      goto failure;
   }
 
   SCOPE_EXIT_RTN_VALUE(0, "Cause: %d\n", cause);
 
failure:
   /* Go ahead and do our cleanup of the session and channel even if we're not going
    * to be able to send our SIP request/response
    */
   clear_session_and_channel(channel->session, ast);
   ao2_cleanup(channel);
   ao2_cleanup(h_data);
 
   SCOPE_EXIT_RTN_VALUE(-1, "Cause: %d\n", cause);
}
 
struct request_data {
   struct ast_sip_session *session;
   struct ast_stream_topology *topology;
   const char *dest;
   int cause;
};
 
static int request(void *obj)
{
   struct request_data *req_data = obj;
   struct ast_sip_session *session = NULL;
   char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
   struct ast_sip_endpoint *endpoint;
 
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(endpoint);
      AST_APP_ARG(aor);
   );
   SCOPE_ENTER(1, "%s\n",tmp);
 
   if (ast_strlen_zero(tmp)) {
      ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
      req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
      SCOPE_EXIT_RTN_VALUE(-1, "Empty destination\n");
   }
 
   AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
 
   if (ast_sip_get_disable_multi_domain()) {
      /* If a request user has been specified extract it from the endpoint name portion */
      if ((endpoint_name = strchr(args.endpoint, '@'))) {
         request_user = args.endpoint;
         *endpoint_name++ = '\0';
      } else {
         endpoint_name = args.endpoint;
      }
 
      if (ast_strlen_zero(endpoint_name)) {
         if (request_user) {
            ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name: %s@<endpoint-name>\n",
               request_user);
         } else {
            ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
         }
         req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
         SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
      }
      endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
         endpoint_name);
      if (!endpoint) {
         ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
         req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
         SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
      }
   } else {
      /* First try to find an exact endpoint match, for single (user) or multi-domain (user@domain) */
      endpoint_name = args.endpoint;
      if (ast_strlen_zero(endpoint_name)) {
         ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
         req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
         SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
      }
      endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
         endpoint_name);
      if (!endpoint) {
         /* It seems it's not a multi-domain endpoint or single endpoint exact match,
          * it's possible that it's a SIP trunk with a specified user (user@trunkname),
          * so extract the user before @ sign.
          */
         endpoint_name = strchr(args.endpoint, '@');
         if (!endpoint_name) {
            /*
             * Couldn't find an '@' so it had to be an endpoint
             * name that doesn't exist.
             */
            ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n",
               args.endpoint);
            req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
            SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
         }
         request_user = args.endpoint;
         *endpoint_name++ = '\0';
 
         if (ast_strlen_zero(endpoint_name)) {
            ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name: %s@<endpoint-name>\n",
               request_user);
            req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
            SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
         }
 
         endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
            endpoint_name);
         if (!endpoint) {
            ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
            req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
            SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
         }
      }
   }
 
   session = ast_sip_session_create_outgoing(endpoint, NULL, args.aor, request_user,
      req_data->topology);
   ao2_ref(endpoint, -1);
   if (!session) {
      ast_log(LOG_ERROR, "Failed to create outgoing session to endpoint '%s'\n", endpoint_name);
      req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
      SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create session\n");
   }
 
   req_data->session = session;
 
   SCOPE_EXIT_RTN_VALUE(0);
}
 
/*! \brief Function called by core to create a new outgoing PJSIP session */
static struct ast_channel *chan_pjsip_request_with_stream_topology(const char *type, struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
{
   struct request_data req_data;
   RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
   SCOPE_ENTER(1, "%s Topology: %s\n", data,
      ast_str_tmp(256, ast_stream_topology_to_str(topology, &STR_TMP)));
 
   req_data.topology = topology;
   req_data.dest = data;
   /* Default failure value in case ast_sip_push_task_wait_servant() itself fails. */
   req_data.cause = AST_CAUSE_FAILURE;
 
   if (ast_sip_push_task_wait_servant(NULL, request, &req_data)) {
      *cause = req_data.cause;
      SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't push task\n");
   }
 
   session = req_data.session;
 
   if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, assignedids, requestor, NULL))) {
      /* Session needs to be terminated prematurely */
      SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
   }
 
   SCOPE_EXIT_RTN_VALUE(session->channel, "Channel: %s\n", ast_channel_name(session->channel));
}
 
static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
{
   struct ast_stream_topology *topology;
   struct ast_channel *chan;
 
   topology = ast_stream_topology_create_from_format_cap(cap);
   if (!topology) {
      return NULL;
   }
 
   chan = chan_pjsip_request_with_stream_topology(type, topology, assignedids, requestor, data, cause);
 
   ast_stream_topology_free(topology);
 
   return chan;
}
 
struct sendtext_data {
   struct ast_sip_session *session;
   struct ast_msg_data *msg;
};
 
static void sendtext_data_destroy(void *obj)
{
   struct sendtext_data *data = obj;
   ao2_cleanup(data->session);
   ast_free(data->msg);
}
 
static struct sendtext_data* sendtext_data_create(struct ast_channel *chan,
   struct ast_msg_data *msg)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
   struct sendtext_data *data = ao2_alloc(sizeof(*data), sendtext_data_destroy);
 
   if (!data) {
      return NULL;
   }
 
   data->msg = ast_msg_data_dup(msg);
   if (!data->msg) {
      ao2_cleanup(data);
      return NULL;
   }
   data->session = channel->session;
   ao2_ref(data->session, +1);
 
   return data;
}
 
static int sendtext(void *obj)
{
   struct sendtext_data *data = obj;
   pjsip_tx_data *tdata;
   const char *body_text = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_BODY);
   const char *content_type = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_CONTENT_TYPE);
   char *sep;
   struct ast_sip_body body = {
      .type = "text",
      .subtype = "plain",
      .body_text = body_text,
   };
 
   if (!ast_strlen_zero(content_type)) {
      sep = strchr(content_type, '/');
      if (sep) {
         *sep = '\0';
         body.type = content_type;
         body.subtype = ++sep;
      }
   }
 
   if (data->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
      ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
         data->session->inv_session->cause,
         pjsip_get_status_text(data->session->inv_session->cause)->ptr);
   } else {
      pjsip_from_hdr *hdr;
      pjsip_name_addr *name_addr;
      const char *from = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_FROM);
      const char *to = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_TO);
      int invalidate_tdata = 0;
 
      ast_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, NULL, &tdata);
      ast_sip_add_body(tdata, &body);
 
      /*
       * If we have a 'from' in the msg, set the display name in the From
       * header to it.
       */
      if (!ast_strlen_zero(from)) {
         hdr = PJSIP_MSG_FROM_HDR(tdata->msg);
         name_addr = (pjsip_name_addr *) hdr->uri;
         pj_strdup2(tdata->pool, &name_addr->display, from);
         invalidate_tdata = 1;
      }
 
      /*
       * If we have a 'to' in the msg, set the display name in the To
       * header to it.
       */
      if (!ast_strlen_zero(to)) {
         hdr = PJSIP_MSG_TO_HDR(tdata->msg);
         name_addr = (pjsip_name_addr *) hdr->uri;
         pj_strdup2(tdata->pool, &name_addr->display, to);
         invalidate_tdata = 1;
      }
 
      if (invalidate_tdata) {
         pjsip_tx_data_invalidate_msg(tdata);
      }
 
      ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint, NULL, NULL);
   }
 
   ao2_cleanup(data);
 
   return 0;
}
 
/*! \brief Function called by core to send text on PJSIP session */
static int chan_pjsip_sendtext_data(struct ast_channel *ast, struct ast_msg_data *msg)
{
   struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
   struct sendtext_data *data = sendtext_data_create(ast, msg);
 
   ast_debug(1, "Sending MESSAGE from '%s' to '%s:%s': %s\n",
      ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_FROM),
      ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_TO),
      ast_channel_name(ast),
      ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_BODY));
 
   if (!data) {
      return -1;
   }
 
   if (ast_sip_push_task(channel->session->serializer, sendtext, data)) {
      ao2_ref(data, -1);
      return -1;
   }
   return 0;
}
 
static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text)
{
   struct ast_msg_data *msg;
   int rc;
   struct ast_msg_data_attribute attrs[] =
   {
      {
         .type = AST_MSG_DATA_ATTR_BODY,
         .value = (char *)text,
      }
   };
 
   msg = ast_msg_data_alloc(AST_MSG_DATA_SOURCE_TYPE_UNKNOWN, attrs, ARRAY_LEN(attrs));
   if (!msg) {
      return -1;
   }
   rc = chan_pjsip_sendtext_data(ast, msg);
   ast_free(msg);
 
   return rc;
}
 
/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
static int hangup_sip2cause(int cause)
{
   /* Possible values taken from causes.h */
 
   switch(cause) {
   case 401:       /* Unauthorized */
      return AST_CAUSE_CALL_REJECTED;
   case 403:       /* Not found */
      return AST_CAUSE_CALL_REJECTED;
   case 404:       /* Not found */
      return AST_CAUSE_UNALLOCATED;
   case 405:       /* Method not allowed */
      return AST_CAUSE_INTERWORKING;
   case 407:       /* Proxy authentication required */
      return AST_CAUSE_CALL_REJECTED;
   case 408:       /* No reaction */
      return AST_CAUSE_NO_USER_RESPONSE;
   case 409:       /* Conflict */
      return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
   case 410:       /* Gone */
      return AST_CAUSE_NUMBER_CHANGED;
   case 411:       /* Length required */
      return AST_CAUSE_INTERWORKING;
   case 413:       /* Request entity too large */
      return AST_CAUSE_INTERWORKING;
   case 414:       /* Request URI too large */
      return AST_CAUSE_INTERWORKING;
   case 415:       /* Unsupported media type */
      return AST_CAUSE_INTERWORKING;
   case 420:       /* Bad extension */
      return AST_CAUSE_NO_ROUTE_DESTINATION;
   case 480:       /* No answer */
      return AST_CAUSE_NO_ANSWER;
   case 481:       /* No answer */
      return AST_CAUSE_INTERWORKING;
   case 482:       /* Loop detected */
      return AST_CAUSE_INTERWORKING;
   case 483:       /* Too many hops */
      return AST_CAUSE_NO_ANSWER;
   case 484:       /* Address incomplete */
      return AST_CAUSE_INVALID_NUMBER_FORMAT;
   case 485:       /* Ambiguous */
      return AST_CAUSE_UNALLOCATED;
   case 486:       /* Busy everywhere */
      return AST_CAUSE_BUSY;
   case 487:       /* Request terminated */
      return AST_CAUSE_INTERWORKING;
   case 488:       /* No codecs approved */
      return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
   case 491:       /* Request pending */
      return AST_CAUSE_INTERWORKING;
   case 493:       /* Undecipherable */
      return AST_CAUSE_INTERWORKING;
   case 500:       /* Server internal failure */
      return AST_CAUSE_FAILURE;
   case 501:       /* Call rejected */
      return AST_CAUSE_FACILITY_REJECTED;
   case 502:
      return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
   case 503:       /* Service unavailable */
      return AST_CAUSE_CONGESTION;
   case 504:       /* Gateway timeout */
      return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
   case 505:       /* SIP version not supported */
      return AST_CAUSE_INTERWORKING;
   case 600:       /* Busy everywhere */
      return AST_CAUSE_USER_BUSY;
   case 603:       /* Decline */
      return AST_CAUSE_CALL_REJECTED;
   case 604:       /* Does not exist anywhere */
      return AST_CAUSE_UNALLOCATED;
   case 606:       /* Not acceptable */
      return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
   default:
      if (cause < 500 && cause >= 400) {
         /* 4xx class error that is unknown - someting wrong with our request */
         return AST_CAUSE_INTERWORKING;
      } else if (cause < 600 && cause >= 500) {
         /* 5xx class error - problem in the remote end */
         return AST_CAUSE_CONGESTION;
      } else if (cause < 700 && cause >= 600) {
         /* 6xx - global errors in the 4xx class */
         return AST_CAUSE_INTERWORKING;
      }
      return AST_CAUSE_NORMAL;
   }
   /* Never reached */
   return 0;
}
 
static void chan_pjsip_session_begin(struct ast_sip_session *session)
{
   RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
   SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
   if (session->endpoint->media.direct_media.glare_mitigation ==
         AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
      SCOPE_EXIT_RTN("Direct media no glare mitigation\n");
   }
 
   datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info,
         "direct_media_glare_mitigation");
 
   if (!datastore) {
      SCOPE_EXIT_RTN("Couldn't create datastore\n");
   }
 
   ast_sip_session_add_datastore(session, datastore);
   SCOPE_EXIT_RTN();
}
 
/*! \brief Function called when the session ends */
static void chan_pjsip_session_end(struct ast_sip_session *session)
{
   SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
   if (!session->channel) {
      SCOPE_EXIT_RTN("No channel\n");
   }
 
   chan_pjsip_remove_hold(ast_channel_uniqueid(session->channel));
 
   ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
   if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
      int cause = hangup_sip2cause(session->inv_session->cause);
 
      ast_queue_hangup_with_cause(session->channel, cause);
   } else {
      ast_queue_hangup(session->channel);
   }
 
   SCOPE_EXIT_RTN();
}
 
static void set_sipdomain_variable(struct ast_sip_session *session)
{
   pjsip_sip_uri *sip_ruri = pjsip_uri_get_uri(session->request_uri);
   size_t size = pj_strlen(&sip_ruri->host) + 1;
   char *domain = ast_alloca(size);
 
   ast_copy_pj_str(domain, &sip_ruri->host, size);
 
   pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
   return;
}
 
/*! \brief Function called when a request is received on the session */
static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
   RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
   struct transport_info_data *transport_data;
   pjsip_tx_data *packet = NULL;
   SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
 
   if (session->channel) {
      SCOPE_EXIT_RTN_VALUE(0, "%s: No channel\n", ast_sip_session_get_name(session));
   }
 
   /* Check for a to-tag to determine if this is a reinvite */
   if (rdata->msg_info.to->tag.slen) {
      /* Weird case. We've received a reinvite but we don't have a channel. The most
       * typical case for this happening is that a blind transfer fails, and so the
       * transferer attempts to reinvite himself back into the call. We already got
       * rid of that channel, and the other side of the call is unrecoverable.
       *
       * We treat this as a failure, so our best bet is to just hang this call
       * up and not create a new channel. Clearing defer_terminate here ensures that
       * calling ast_sip_session_terminate() can result in a BYE being sent ASAP.
       */
      session->defer_terminate = 0;
      ast_sip_session_terminate(session, 400);
      SCOPE_EXIT_RTN_VALUE(-1, "%s: We have a To tag but no channel.  Terminating session\n", ast_sip_session_get_name(session));
   }
 
   datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
   if (!datastore) {
      SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info datastore\n", ast_sip_session_get_name(session));
   }
 
   transport_data = ast_calloc(1, sizeof(*transport_data));
   if (!transport_data) {
      SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info\n", ast_sip_session_get_name(session));
   }
   pj_sockaddr_cp(&transport_data->local_addr, &rdata->tp_info.transport->local_addr);
   pj_sockaddr_cp(&transport_data->remote_addr, &rdata->pkt_info.src_addr);
   datastore->data = transport_data;
   ast_sip_session_add_datastore(session, datastore);
 
   if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL, NULL))) {
      if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS
         && packet) {
         ast_sip_session_send_response(session, packet);
      }
 
      SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to allocate new PJSIP channel on incoming SIP INVITE\n",
          ast_sip_session_get_name(session));
   }
 
   set_sipdomain_variable(session);
 
   /* channel gets created on incoming request, but we wait to call start
           so other supplements have a chance to run */
   SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
}
 
static int call_pickup_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
   struct ast_features_pickup_config *pickup_cfg;
   struct ast_channel *chan;
 
   /* Check for a to-tag to determine if this is a reinvite */
   if (rdata->msg_info.to->tag.slen) {
      /* We don't care about reinvites */
      return 0;
   }
 
   pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
   if (!pickup_cfg) {
      ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension.\n");
      return 0;
   }
 
   if (strcmp(session->exten, pickup_cfg->pickupexten)) {
      ao2_ref(pickup_cfg, -1);
      return 0;
   }
   ao2_ref(pickup_cfg, -1);
 
   /* We can't directly use session->channel because the pickup operation will cause a masquerade to occur,
    * changing the channel pointer in session to a different channel. To ensure we work on the right channel
    * we store a pointer locally before we begin and keep a reference so it remains valid no matter what.
    */
   chan = ast_channel_ref(session->channel);
   if (ast_pickup_call(chan)) {
      ast_channel_hangupcause_set(chan, AST_CAUSE_CALL_REJECTED);
   } else {
      ast_channel_hangupcause_set(chan, AST_CAUSE_NORMAL_CLEARING);
   }
   /* A hangup always occurs because the pickup operation will have either failed resulting in the call
    * needing to be hung up OR the pickup operation was a success and the channel we now have is actually
    * the channel that was replaced, which should be hung up since it is literally in limbo not connected
    * to anything at all.
    */
   ast_hangup(chan);
   ast_channel_unref(chan);
 
   return 1;
}
 
static struct ast_sip_session_supplement call_pickup_supplement = {
   .method = "INVITE",
   .priority = AST_SIP_SUPPLEMENT_PRIORITY_LAST - 1,
   .incoming_request = call_pickup_incoming_request,
};
 
static int pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
   int res;
   SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
   /* Check for a to-tag to determine if this is a reinvite */
   if (rdata->msg_info.to->tag.slen) {
      /* We don't care about reinvites */
      SCOPE_EXIT_RTN_VALUE(0, "Reinvite\n");
   }
 
   res = ast_pbx_start(session->channel);
 
   switch (res) {
   case AST_PBX_FAILED:
      ast_log(LOG_WARNING, "Failed to start PBX ;(\n");
      ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
      ast_hangup(session->channel);
      break;
   case AST_PBX_CALL_LIMIT:
      ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
      ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
      ast_hangup(session->channel);
      break;
   case AST_PBX_SUCCESS:
   default:
      break;
   }
 
   ast_debug(3, "Started PBX on new PJSIP channel %s\n", ast_channel_name(session->channel));
 
   SCOPE_EXIT_RTN_VALUE((res == AST_PBX_SUCCESS) ? 0 : -1, "RC: %d\n", res);
}
 
static struct ast_sip_session_supplement pbx_start_supplement = {
   .method = "INVITE",
   .priority = AST_SIP_SUPPLEMENT_PRIORITY_LAST,
   .incoming_request = pbx_start_incoming_request,
};
 
/*! \brief Function called when a response is received on the session */
static void chan_pjsip_incoming_response_update_cause(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
   struct pjsip_status_line status = rdata->msg_info.msg->line.status;
   struct ast_control_pvt_cause_code *cause_code;
   int data_size = sizeof(*cause_code);
   SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
 
   if (!session->channel) {
      SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
   }
 
   /* Build and send the tech-specific cause information */
   /* size of the string making up the cause code is "SIP " number + " " + reason length */
   data_size += 4 + 4 + pj_strlen(&status.reason);
   cause_code = ast_alloca(data_size);
   memset(cause_code, 0, data_size);
 
   ast_copy_string(cause_code->chan_name, ast_channel_name(session->channel), AST_CHANNEL_NAME);
 
   snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "SIP %d %.*s", status.code,
   (int) pj_strlen(&status.reason), pj_strbuf(&status.reason));
 
   cause_code->ast_cause = hangup_sip2cause(status.code);
   ast_queue_control_data(session->channel, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
   ast_channel_hangupcause_hash_set(session->channel, cause_code, data_size);
 
   SCOPE_EXIT_RTN("%s\n", ast_sip_session_get_name(session));
}
 
/*! \brief Function called when a response is received on the session */
static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
   struct pjsip_status_line status = rdata->msg_info.msg->line.status;
   SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
 
   if (!session->channel) {
      SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
   }
 
   switch (status.code) {
   case 180: {
      pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
      if (sdp && sdp->body.ptr) {
         ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
         ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
      } else {
         ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
         ast_queue_control(session->channel, AST_CONTROL_RINGING);
      }
 
      ast_channel_lock(session->channel);
      if (ast_channel_state(session->channel) != AST_STATE_UP) {
         ast_setstate(session->channel, AST_STATE_RINGING);
      }
      ast_channel_unlock(session->channel);
      break;
   }
   case 183:
      if (session->endpoint->ignore_183_without_sdp) {
         pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
         if (sdp && sdp->body.ptr) {
            ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
            ast_trace(1, "%s Method: %.*s Status: %d  Queueing PROGRESS with SDP\n", ast_sip_session_get_name(session),
               (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
            ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
         }
      } else {
         ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
         ast_trace(1, "%s Method: %.*s Status: %d  Queueing PROGRESS without SDP\n", ast_sip_session_get_name(session),
            (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
         ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
      }
      break;
   case 200:
      ast_trace(-1, "%s: Queueing ANSWER\n", ast_sip_session_get_name(session));
      ast_queue_control(session->channel, AST_CONTROL_ANSWER);
      break;
   default:
      ast_trace(-1, "%s: Not queueing anything\n", ast_sip_session_get_name(session));
      break;
   }
 
   SCOPE_EXIT_RTN("%s\n", ast_sip_session_get_name(session));
}
 
static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
   SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
 
   if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
      if (session->endpoint->media.direct_media.enabled && session->channel) {
         ast_trace(-1, "%s: Queueing SRCCHANGE\n", ast_sip_session_get_name(session));
         ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
      }
   }
   SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
}
 
static int update_devstate(void *obj, void *arg, int flags)
{
   ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE,
              "PJSIP/%s", ast_sorcery_object_get_id(obj));
   return 0;
}
 
static struct ast_custom_function chan_pjsip_dial_contacts_function = {
   .name = "PJSIP_DIAL_CONTACTS",
   .read = pjsip_acf_dial_contacts_read,
};
 
static struct ast_custom_function chan_pjsip_parse_uri_function = {
   .name = "PJSIP_PARSE_URI",
   .read = pjsip_acf_parse_uri_read,
};
 
static struct ast_custom_function media_offer_function = {
   .name = "PJSIP_MEDIA_OFFER",
   .read = pjsip_acf_media_offer_read,
   .write = pjsip_acf_media_offer_write
};
 
static struct ast_custom_function dtmf_mode_function = {
   .name = "PJSIP_DTMF_MODE",
   .read = pjsip_acf_dtmf_mode_read,
   .write = pjsip_acf_dtmf_mode_write
};
 
static struct ast_custom_function moh_passthrough_function = {
   .name = "PJSIP_MOH_PASSTHROUGH",
   .read = pjsip_acf_moh_passthrough_read,
   .write = pjsip_acf_moh_passthrough_write
};
 
static struct ast_custom_function session_refresh_function = {
   .name = "PJSIP_SEND_SESSION_REFRESH",
   .write = pjsip_acf_session_refresh_write,
};
 
/*!
 * \brief Load the module
 *
 * Module loading including tests for configuration or dependencies.
 * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
 * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
 * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
 * configuration file or other non-critical problem return
 * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
 */
static int load_module(void)
{
   struct ao2_container *endpoints;
 
   if (!(chan_pjsip_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
      return AST_MODULE_LOAD_DECLINE;
   }
 
   ast_format_cap_append_by_type(chan_pjsip_tech.capabilities, AST_MEDIA_TYPE_AUDIO);
 
   ast_rtp_glue_register(&chan_pjsip_rtp_glue);
 
   if (ast_channel_register(&chan_pjsip_tech)) {
      ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
      goto end;
   }
 
   if (ast_custom_function_register(&chan_pjsip_dial_contacts_function)) {
      ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n");
      goto end;
   }
 
   if (ast_custom_function_register(&chan_pjsip_parse_uri_function)) {
      ast_log(LOG_ERROR, "Unable to register PJSIP_PARSE_URI dialplan function\n");
      goto end;
   }
 
   if (ast_custom_function_register(&media_offer_function)) {
      ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
      goto end;
   }
 
   if (ast_custom_function_register(&dtmf_mode_function)) {
      ast_log(LOG_WARNING, "Unable to register PJSIP_DTMF_MODE dialplan function\n");
      goto end;
   }
 
   if (ast_custom_function_register(&moh_passthrough_function)) {
      ast_log(LOG_WARNING, "Unable to register PJSIP_MOH_PASSTHROUGH dialplan function\n");
      goto end;
   }
 
   if (ast_custom_function_register(&session_refresh_function)) {
      ast_log(LOG_WARNING, "Unable to register PJSIP_SEND_SESSION_REFRESH dialplan function\n");
      goto end;
   }
 
   ast_sip_register_service(&refer_callback_module);
 
   ast_sip_session_register_supplement(&chan_pjsip_supplement);
   ast_sip_session_register_supplement(&chan_pjsip_supplement_response);
 
   if (!(pjsip_uids_onhold = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_RWLOCK,
         AO2_CONTAINER_ALLOC_OPT_DUPS_REJECT, 37, uid_hold_hash_fn,
         uid_hold_sort_fn, NULL))) {
      ast_log(LOG_ERROR, "Unable to create held channels container\n");
      goto end;
   }
 
   ast_sip_session_register_supplement(&call_pickup_supplement);
   ast_sip_session_register_supplement(&pbx_start_supplement);
   ast_sip_session_register_supplement(&chan_pjsip_ack_supplement);
 
   if (pjsip_channel_cli_register()) {
      ast_log(LOG_ERROR, "Unable to register PJSIP Channel CLI\n");
      goto end;
   }
 
   /* since endpoints are loaded before the channel driver their device
      states get set to 'invalid', so they need to be updated */
   if ((endpoints = ast_sip_get_endpoints())) {
      ao2_callback(endpoints, OBJ_NODATA, update_devstate, NULL);
      ao2_ref(endpoints, -1);
   }
 
   return 0;
 
end:
   ao2_cleanup(pjsip_uids_onhold);
   pjsip_uids_onhold = NULL;
   ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
   ast_sip_session_unregister_supplement(&pbx_start_supplement);
   ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
   ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
   ast_sip_session_unregister_supplement(&call_pickup_supplement);
   ast_sip_unregister_service(&refer_callback_module);
   ast_custom_function_unregister(&dtmf_mode_function);
   ast_custom_function_unregister(&moh_passthrough_function);
   ast_custom_function_unregister(&media_offer_function);
   ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
   ast_custom_function_unregister(&chan_pjsip_parse_uri_function);
   ast_custom_function_unregister(&session_refresh_function);
   ast_channel_unregister(&chan_pjsip_tech);
   ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
 
   return AST_MODULE_LOAD_DECLINE;
}
 
/*! \brief Unload the PJSIP channel from Asterisk */
static int unload_module(void)
{
   ao2_cleanup(pjsip_uids_onhold);
   pjsip_uids_onhold = NULL;
 
   pjsip_channel_cli_unregister();
 
   ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
   ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
   ast_sip_session_unregister_supplement(&pbx_start_supplement);
   ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
   ast_sip_session_unregister_supplement(&call_pickup_supplement);
 
   ast_sip_unregister_service(&refer_callback_module);
 
   ast_custom_function_unregister(&dtmf_mode_function);
   ast_custom_function_unregister(&moh_passthrough_function);
   ast_custom_function_unregister(&media_offer_function);
   ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
   ast_custom_function_unregister(&chan_pjsip_parse_uri_function);
   ast_custom_function_unregister(&session_refresh_function);
 
   ast_channel_unregister(&chan_pjsip_tech);
   ao2_ref(chan_pjsip_tech.capabilities, -1);
   ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
 
   return 0;
}
 
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Channel Driver",
   .support_level = AST_MODULE_SUPPORT_CORE,
   .load = load_module,
   .unload = unload_module,
   .load_pri = AST_MODPRI_CHANNEL_DRIVER,
   .requires = "res_pjsip,res_pjsip_session,res_pjsip_pubsub",
);