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/channel_info.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,
};
 
static int chan_pjsip_incoming_prack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
 
static struct ast_sip_session_supplement chan_pjsip_prack_supplement = {
    .method = "PRACK",
    .priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
    .incoming_request = chan_pjsip_incoming_prack,
};
 
/*! \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,
};
 
 
/*!
 * \brief Determine the local signaling address used for an outgoing request.
 *
 * For unreliable transports, the transport local address may be a wildcard
 * address. Ask the PJSIP transport manager for the local interface selected
 * for the request destination, then fall back to the transport local address
 * if needed.
 */
static int set_transport_info_local_addr(struct pjsip_tx_data *tdata, pj_sockaddr *local_addr)
{
    if (!(tdata->tp_info.transport->flag & PJSIP_TRANSPORT_RELIABLE)) {
        pjsip_tpmgr_fla2_param prm;
 
        pjsip_tpmgr_fla2_param_default(&prm);
        prm.tp_type = tdata->tp_info.transport->key.type;
        pj_strset2(&prm.dst_host, tdata->tp_info.dst_name);
        prm.local_if = PJ_TRUE;
 
        if (pjsip_tpmgr_find_local_addr2(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()),
            tdata->pool, &prm) == PJ_SUCCESS) {
            int af = prm.tp_type & PJSIP_TRANSPORT_IPV6 ? pj_AF_INET6() : pj_AF_INET();
 
            if (prm.tp_type == PJSIP_TRANSPORT_UDP || prm.tp_type == PJSIP_TRANSPORT_UDP6) {
                prm.ret_port = tdata->tp_info.transport->local_name.port;
            }
 
            if (pj_sockaddr_init(af, local_addr, &prm.ret_addr, prm.ret_port) == PJ_SUCCESS) {
                return 0;
            }
        }
    }
 
    pj_sockaddr_cp(local_addr, &tdata->tp_info.transport->local_addr);
    return pj_sockaddr_has_addr(local_addr) ? 0 : -1;
}
 
/*!
 * \brief Store transport information for outgoing PJSIP session requests.
 *
 * Incoming PJSIP channels get their transport information from the received
 * request. Outgoing channels may not have this information available through
 * the channel datastore, so store the selected transport addresses once the
 * outgoing request has been created and a destination is known.
 */
static pj_status_t transport_info_on_tx_request(pjsip_tx_data *tdata)
{
    RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
    RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
    struct transport_info_data *transport_data;
    pjsip_dialog *dlg;
 
    if (!tdata->tp_info.transport
        || !pj_sockaddr_has_addr(&tdata->tp_info.dst_addr)) {
        return PJ_SUCCESS;
    }
 
    dlg = pjsip_tdata_get_dlg(tdata);
    if (!dlg) {
        return PJ_SUCCESS;
    }
 
    session = ast_sip_dialog_get_session(dlg);
    if (!session
        || !session->inv_session
        || session->inv_session->role != PJSIP_ROLE_UAC) {
        return PJ_SUCCESS;
    }
 
    datastore = ast_sip_session_get_datastore(session, "transport_info");
    if (datastore) {
        return PJ_SUCCESS;
    }
 
    datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
    if (!datastore) {
        return PJ_SUCCESS;
    }
 
    transport_data = ast_calloc(1, sizeof(*transport_data));
    if (!transport_data) {
        return PJ_SUCCESS;
    }
 
    if (set_transport_info_local_addr(tdata, &transport_data->local_addr)) {
        ast_free(transport_data);
        return PJ_SUCCESS;
    }
    pj_sockaddr_cp(&transport_data->remote_addr, &tdata->tp_info.dst_addr);
 
    datastore->data = transport_data;
    ast_sip_session_add_datastore(session, datastore);
 
    return PJ_SUCCESS;
}
 
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) {
            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);
    pjsip_channel_info_update(session);
 
    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;
    struct ast_channel_initializers initializers = {
        .version = AST_CHANNEL_INITIALIZERS_VERSION,
        .tenantid = session->endpoint->tenantid,
    };
    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_initializers(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, &initializers, "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 (pjsip_channel_info_create(chan, session)) {
        /* Keep technology hangup from accessing the not yet attached session. */
        ast_channel_tech_pvt_set(chan, NULL);
        ao2_cleanup(channel);
        ast_channel_unlock(chan);
        ast_hangup(chan);
        SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create PJSIP channel state\n");
    }
 
    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);
    }
 
    ast_channel_stage_snapshot_done(chan);
    ast_channel_unlock(chan);
 
    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)));
    }
 
    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 taskpool 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 taskpool.\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 =  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, cur->subclass.format, ast_channel_readformat(ast));
 
        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_get_snapshot(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_DOWN && snapshot->state != AST_STATE_RESERVED) {
            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 (!channel || !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 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)
{
    struct ast_sip_session_media *session_media = session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
    if (session_media) {
        session_media->locally_held = 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;
    enum ast_channel_state state = ast_channel_state(session->channel);
    enum ast_sip_session_refresh_method method = AST_SIP_SESSION_REFRESH_METHOD_INVITE;
    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)));
 
    /* See RFC 6337, especially section 3.2: If the early media SDP was sent reliably, we are allowed
     * to send UPDATEs. Only relevant for AST_STATE_RINGING and AST_STATE_RING - if the channel is UP,
     * re-INVITES can be sent.
     */
    if (session->early_confirmed && (state == AST_STATE_RINGING || state == AST_STATE_RING)) {
        method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
    }
 
    ret = ast_sip_session_refresh(session, NULL, NULL, on_topology_change_response,
        method, 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);
}
 
/* Forward declarations */
static int transmit_info_dtmf(void *data);
static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration);
 
/*! \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];
    unsigned int duration;
    char digit;
    struct info_dtmf_data *dtmf_data;
 
    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)) {
                res = -1;
                if (ast_sip_get_allow_sending_180_after_183()) {
                    response_code = 180;
                } else {
                    response_code = 183;
                }
            } 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_FLASH:
        duration = 300;
        digit = '!';
        dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
 
        if (!dtmf_data) {
            res = -1;
            break;
        }
 
        if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
            ast_log(LOG_WARNING, "Error sending FLASH via INFO on channel %s\n", ast_channel_name(ast));
            ao2_ref(dtmf_data, -1); /* dtmf_data can't be null here */
            res = -1;
        }
        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 ||
                    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_UNKNOWN, AST_DEVSTATE_CACHABLE, 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, AST_DEVSTATE_CACHABLE, 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:
        if (!channel->session->endpoint->media.bundle) {
            /* Generate a new SSRC due to media source change and RTP timestamp reset.
               Ensures RFC 3550 compliance and avoids SBC interoperability issues (Sonus/Ribbon)*/
            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->rtp) {
                    ast_rtp_instance_change_source(media->rtp);
                }
            }
        }
        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 AST_CONTROL_TRANSFER:
        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 General PJSIP module for chan_pjsip, currently used for REFER subscription data and outgoing request callbacks */
static pjsip_module chan_pjsip_module = {
    .name = { "chan_pjsip", 10 },
    .id = -1,
    .on_tx_request = transport_info_on_tx_request,
};
 
/*!
 * \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, chan_pjsip_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, chan_pjsip_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, chan_pjsip_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;
    }
 
    /* chan_pjsip_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, chan_pjsip_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, chan_pjsip_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_session *session = data;
    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(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);
    }
    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);
    struct ast_sip_session *session = ao2_bump(channel->session);
 
    SCOPE_ENTER(1, "%s Topology: %s\n", ast_sip_session_get_name(session),
        ast_str_tmp(256, ast_stream_topology_to_str(session->pending_media_state->topology, &STR_TMP)));
 
    ast_channel_unlock(ast);
 
    /* The creation of the INVITE needs to be pushed synchronously to prevent a race condition
       with bridging on attended transfers that can result in a loss of set Caller ID. */
    if (ast_sip_push_task_wait_serializer(session->serializer, call, session)) {
        ast_log(LOG_WARNING, "Error attempting to place outbound call to '%s'\n", dest);
        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, "'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_REDIRECTED_TO_NEW_DESTINATION:   /* 23 */
        return 302;
    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 (channel lock held) */
static void clear_session_and_channel_locked(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);
}
 
/*! \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)
{
    /* Serialize detachment with channel-locked readers of the session. */
    ast_channel_lock(ast);
    clear_session_and_channel_locked(session, ast);
    ast_channel_unlock(ast);
}
 
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;
 
            if (channel->session->active_media_state &&
                channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
                struct ast_sip_session_media *media =
                    channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
                if (media->rtp) {
                    ast_rtp_instance_set_stats_vars(ast, media->rtp);
                }
            }
 
            /*
            * 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;
    int tech_cause;
    int original_tech_cause;
    struct hangup_data *h_data;
    SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
 
    if (!channel || !channel->session) {
        SCOPE_EXIT_RTN_VALUE(-1, "%s: No channel or session\n", ast_channel_name(ast));
    }
 
    cause = ast_channel_hangupcause(channel->session->channel);
    tech_cause = hangup_cause2sip(cause);
    original_tech_cause = ast_channel_tech_hangupcause(channel->session->channel);
    if (!original_tech_cause) {
        ast_channel_tech_hangupcause_set(channel->session->channel, tech_cause);
    }
 
    h_data = hangup_data_alloc(tech_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 taskpool. Expect bad things\n");
        goto failure;
    }
 
    SCOPE_EXIT_RTN_VALUE(0, "%s: Cause: %d  Tech Cause: %d\n", ast_channel_name(ast),
        cause, tech_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_locked(channel->session, ast);
    ao2_cleanup(channel);
    ao2_cleanup(h_data);
 
    SCOPE_EXIT_RTN_VALUE(-1, "%s: Cause: %d\n", ast_channel_name(ast), 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)) {
        char *content_type_copy = ast_strdupa(content_type);
        sep = strchr(content_type_copy, '/');
        if (sep) {
            *sep = '\0';
            body.type = content_type_copy;
            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;
}
 
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)
{
    int existing_cause = 0;
    int existing_tech_cause = 0;
    int new_cause = 0;
    int new_tech_cause = 0;
    SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
    if (session->inv_session) {
        ast_trace(-1, "%s: inv_session->cause: %d\n", ast_sip_session_get_name(session),
            session->inv_session->cause);
    }
 
    if (!session->channel) {
        SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
    }
 
    if (session->active_media_state &&
        session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
        struct ast_sip_session_media *media =
            session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
        if (media->rtp) {
            ast_rtp_instance_set_stats_vars(session->channel, media->rtp);
        }
    }
 
    chan_pjsip_remove_hold(ast_channel_uniqueid(session->channel));
    ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
 
    existing_cause = ast_channel_hangupcause(session->channel);
    new_cause = existing_cause;
    existing_tech_cause = ast_channel_tech_hangupcause(session->channel);
    new_tech_cause = existing_tech_cause;
 
    ast_trace(-1, "%s: existing ast_cause: %d  tech_cause: %d\n", ast_sip_session_get_name(session),
        existing_cause, existing_tech_cause);
 
    if (session->inv_session) {
        struct ao2_iterator dialed_causes_iterator = ast_channel_dialed_causes_iterator(session->channel);
        struct ast_control_pvt_cause_code *pvt_cause;
        int tech_cause = 0;
 
        /*
         * The dialed causes represent the results of each channel dialed but we are only
         * interested in the cause codes for _this_ channel. For example, in a redirect
         * scenario, if this channel is the redirecting channel (it responded with a 302),
         * then chan_pjsip_incoming_response_update_cause() will have been called with the
         * 302 and it would have been added to the dialed causes for this channel.  When this
         * session ends however, the inv_session->cause will probably be something like
         * "487 Request terminated" which is just a generic "the session ended" code and
         * not really informative.
         *
         * We'll iterate over the dialed causes to find the one for this channel and if we
         * find one with a non-2xx SIP response code, we'll use that cause code for the
         * channel hangup cause instead of the generic 487 cause code that results from
         * the session termination.
         */
        while ((pvt_cause = ao2_iterator_next(&dialed_causes_iterator))) {
            if (ast_strings_equal(pvt_cause->chan_name, ast_channel_name(session->channel))) {
                /*
                 * The "SIP <code> <reason>" format for pvt_cause->code is set in
                 * chan_pjsip_incoming_response_update_cause but only for PJSIP channels.
                 * However, since we've just checked that the chan_name matches our channel,
                 * we can be confident that this format is correct and attempt to parse out the
                 * tech cause from it. If for some reason it doesn't match this format,
                 * we'll just log a trace/debug and skip using this cause code.
                 */
                int count = sscanf(pvt_cause->code, "SIP %d ", &tech_cause);
                if (count != 1) {
                    ast_trace(-1, "%s: Unable to parse tech_cause from code '%s'\n",
                        ast_sip_session_get_name(session), pvt_cause->code);
                    continue;
                }
                /* We only want to use non-2XX SIP response codes */
                if (tech_cause / 100 > 2) {
                    new_tech_cause = tech_cause;
                    if (!existing_cause) {
                        new_cause = pvt_cause->ast_cause;
                    }
                    ast_trace(-1, "%s: %s dialed ast_cause: %d tech_cause: %d used\n", ast_sip_session_get_name(session),
                        pvt_cause->chan_name, new_cause, new_tech_cause);
                } else {
                    ast_trace(-1, "%s: %s dialed ast_cause: %d tech_cause: %s. Skipped 2XX code.\n",
                        ast_sip_session_get_name(session), pvt_cause->chan_name,
                        pvt_cause->ast_cause, pvt_cause->code);
                }
            } else {
                ast_trace(-1, "%s: %s dialed ast_cause: %d tech_cause: %s. Skipped other channel.\n",
                    ast_sip_session_get_name(session), pvt_cause->chan_name,
                    pvt_cause->ast_cause, pvt_cause->code);
            }
            ao2_cleanup(pvt_cause);
        }
        ao2_iterator_destroy(&dialed_causes_iterator);
 
        /*
         * We have a chicken and egg thing going here.  We can derive the tech_cause
         * from the ast cause but we can also derive the ast cause from the tech cause.
         */
 
        /* We only want to use non 2XX response codes for tech cause. */
        if (new_tech_cause == 0 && session->inv_session->cause / 100 > 2) {
            new_tech_cause = session->inv_session->cause;
            ast_trace(-1, "%s: Using tech_cause %d from invite session\n",
                ast_sip_session_get_name(session), session->inv_session->cause);
        }
 
        if (new_cause == 0 && new_tech_cause > 0) {
            new_cause = ast_sip_hangup_sip2cause(new_tech_cause);
            ast_trace(-1, "%s: Using ast_cause %d derived from tech_cause %d\n",
                ast_sip_session_get_name(session), new_cause, new_tech_cause);
        }
 
        if (new_cause != existing_cause) {
            ast_trace(-1, "%s: Setting ast_cause %d\n",
                ast_sip_session_get_name(session), new_cause);
            ast_channel_hangupcause_set(session->channel, new_cause);
        }
 
        if (new_tech_cause == 0) {
            new_tech_cause = hangup_cause2sip(new_cause);
            ast_trace(-1, "%s: Using tech_cause cause %d derived from ast_cause %d\n",
                ast_sip_session_get_name(session), new_tech_cause, new_cause);
        }
 
        if (new_tech_cause != existing_tech_cause && new_tech_cause / 100 > 2) {
            ast_trace(-1, "%s: Setting tech_cause %d\n",
                ast_sip_session_get_name(session), new_tech_cause);
            ast_channel_tech_hangupcause_set(session->channel, new_tech_cause);
        }
    }
 
    ast_queue_hangup(session->channel);
 
    SCOPE_EXIT_RTN("%s: ast_cause: %d  tech_cause: %d\n", ast_sip_session_get_name(session),
        new_cause, new_tech_cause);
}
 
static void set_sipdomain_variable(struct ast_sip_session *session)
{
    const pj_str_t *host = ast_sip_pjsip_uri_get_hostname(session->request_uri);
    size_t size = pj_strlen(host) + 1;
    char *domain = ast_alloca(size);
 
    ast_copy_pj_str(domain, 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);
 
    /*
     * The cause code string is built in the format "SIP <status code> <reason phrase>".
     * Unfortunately, the ast_control_pvt_cause_code structure doesn't have a separate
     * field for the numeric code and adding it would break ABI so we'll have to parse
     * this string in chan_pjsip_session_end() later.  If the string needs to be
     * changed, make sure the parsing in chan_pjsip_session_end() is adjusted as well.
     */
    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 = ast_sip_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: ast_cause: %d tech_cause: %s\n", ast_sip_session_get_name(session),
        cause_code->ast_cause, cause_code->code);
}
 
/*! \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));
            session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
            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);
                session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
                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 chan_pjsip_incoming_prack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
    SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
 
    if (pj_strcmp2(&rdata->msg_info.msg->line.req.method.name, "PRACK") == 0 &&
        pjmedia_sdp_neg_get_state(session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE) {
 
        session->early_confirmed = 1;
    }
    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 chan_pjsip_parse_uri_from_function = {
    .name = "PJSIP_PARSE_URI_FROM",
    .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,
};
 
static struct ast_custom_function transfer_handling_function = {
    .name = "PJSIP_TRANSFER_HANDLING",
    .write = pjsip_transfer_handling_write,
};
 
static char *app_pjsip_hangup = "PJSIPHangup";
 
/*!
 * \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);
    pjsip_channel_info_register();
 
    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(&chan_pjsip_parse_uri_from_function)) {
        ast_log(LOG_ERROR, "Unable to register PJSIP_PARSE_URI_FROM 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;
    }
 
    if (ast_custom_function_register(&transfer_handling_function)) {
        ast_log(LOG_WARNING, "Unable to register PJSIP_TRANSFER_HANDLING dialplan function\n");
        goto end;
    }
 
    if (ast_register_application_xml(app_pjsip_hangup, pjsip_app_hangup)) {
        ast_log(LOG_WARNING, "Unable to register PJSIPHangup dialplan application\n");
        goto end;
    }
    ast_manager_register_xml(app_pjsip_hangup, EVENT_FLAG_SYSTEM | EVENT_FLAG_CALL, pjsip_action_hangup);
 
 
    ast_sip_register_service(&chan_pjsip_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);
    ast_sip_session_register_supplement(&chan_pjsip_prack_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:
    pjsip_channel_info_unregister();
    ao2_cleanup(pjsip_uids_onhold);
    pjsip_uids_onhold = NULL;
    ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
    ast_sip_session_unregister_supplement(&chan_pjsip_prack_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(&chan_pjsip_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(&chan_pjsip_parse_uri_from_function);
    ast_custom_function_unregister(&session_refresh_function);
    ast_custom_function_unregister(&transfer_handling_function);
    ast_unregister_application(app_pjsip_hangup);
    ast_manager_unregister(app_pjsip_hangup);
 
    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)
{
    pjsip_channel_info_unregister();
    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(&chan_pjsip_prack_supplement);
    ast_sip_session_unregister_supplement(&call_pickup_supplement);
 
    ast_sip_unregister_service(&chan_pjsip_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(&chan_pjsip_parse_uri_from_function);
    ast_custom_function_unregister(&session_refresh_function);
    ast_custom_function_unregister(&transfer_handling_function);
    ast_unregister_application(app_pjsip_hangup);
    ast_manager_unregister(app_pjsip_hangup);
 
    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",
);