dialplan_functions.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2013, Digium, Inc.
 *
 * 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 \verbatim Joshua Colp <jcolp@digium.com> \endverbatim
 * \author \verbatim Matt Jordan <mjordan@digium.com> \endverbatim
 *
 * \ingroup functions
 *
 * \brief PJSIP channel dialplan functions
 */
 
/*** MODULEINFO
    <support_level>core</support_level>
 ***/
 
#include "asterisk.h"
 
#include <pjsip.h>
#include <pjlib.h>
#include <pjsip_ua.h>
 
#include "asterisk/astobj2.h"
#include "asterisk/module.h"
#include "asterisk/acl.h"
#include "asterisk/app.h"
#include "asterisk/conversions.h"
#include "asterisk/channel.h"
#include "asterisk/stream.h"
#include "asterisk/format.h"
#include "asterisk/dsp.h"
#include "asterisk/pbx.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "include/chan_pjsip.h"
#include "include/channel_info.h"
#include "include/dialplan_functions.h"
 
/*!
 * \brief String representations of the T.38 state enum
 */
static const char *t38state_to_string[T38_MAX_ENUM] = {
    [T38_DISABLED] = "DISABLED",
    [T38_LOCAL_REINVITE] = "LOCAL_REINVITE",
    [T38_PEER_REINVITE] = "REMOTE_REINVITE",
    [T38_ENABLED] = "ENABLED",
    [T38_REJECTED] = "REJECTED",
};
 
/*!
 * \internal \brief Escape a captured URI into the dialplan result buffer
 */
static int copy_escaped_uri(struct ast_channel *chan, const char *type,
    const struct pjsip_channel_uri *uri, char *buf, size_t size)
{
    if (uri->length < 0 || uri->length >= size) {
        ast_log(LOG_ERROR, "Channel %s: Unescaped %s too long for %zu byte buffer\n",
            ast_channel_name(chan), type, size);
        buf[0] = '\0';
        return -1;
    }
    ast_escape_quoted(uri->value, buf, size);
    return 0;
}
 
/*!
 * \internal \brief Handle reading RTP information
 */
static int channel_read_rtp(struct ast_channel *chan, struct ast_rtp_instance *rtp,
    const struct ast_sockaddr *direct_media_addr, unsigned int secure,
    unsigned int held, const char *type, const char *field, char *buf, size_t buflen)
{
    struct ast_sockaddr addr;
 
    if (ast_strlen_zero(type)) {
        ast_log(AST_LOG_WARNING, "You must supply a type field for 'rtp' information\n");
        return -1;
    }
 
    if (!ast_strlen_zero(field) && strcmp(field, "audio") && strcmp(field, "video")) {
        ast_log(AST_LOG_WARNING, "Unknown media type field '%s' for 'rtp' information\n", field);
        return -1;
    }
 
    if (!rtp) {
        ast_log(AST_LOG_WARNING, "Channel %s has no %s media/RTP session\n",
            ast_channel_name(chan), S_OR(field, "audio"));
        return -1;
    }
 
    if (!strcmp(type, "src")) {
        ast_rtp_instance_get_local_address(rtp, &addr);
        ast_copy_string(buf, ast_sockaddr_stringify(&addr), buflen);
    } else if (!strcmp(type, "dest")) {
        ast_rtp_instance_get_remote_address(rtp, &addr);
        ast_copy_string(buf, ast_sockaddr_stringify(&addr), buflen);
    } else if (!strcmp(type, "direct")) {
        ast_copy_string(buf, ast_sockaddr_stringify(direct_media_addr), buflen);
    } else if (!strcmp(type, "secure")) {
        snprintf(buf, buflen, "%u", secure);
    } else if (!strcmp(type, "hold")) {
        snprintf(buf, buflen, "%u", held);
    } else {
        ast_log(AST_LOG_WARNING, "Unknown type field '%s' specified for 'rtp' information\n", type);
        return -1;
    }
 
    return 0;
}
 
/*!
 * \internal \brief Handle reading RTCP information
 */
static int channel_read_rtcp(struct ast_channel *chan, struct ast_rtp_instance *rtp,
    const char *type, const char *field, char *buf, size_t buflen)
{
    if (ast_strlen_zero(type)) {
        ast_log(AST_LOG_WARNING, "You must supply a type field for 'rtcp' information\n");
        return -1;
    }
 
    if (!ast_strlen_zero(field) && strcmp(field, "audio") && strcmp(field, "video")) {
        ast_log(AST_LOG_WARNING, "Unknown media type field '%s' for 'rtcp' information\n", field);
        return -1;
    }
 
    if (!rtp) {
        ast_log(AST_LOG_WARNING, "Channel %s has no %s media/RTP session\n",
            ast_channel_name(chan), S_OR(field, "audio"));
        return -1;
    }
 
    if (!strncasecmp(type, "all", 3)) {
        enum ast_rtp_instance_stat_field stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY;
 
        if (!strcasecmp(type, "all_jitter")) {
            stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER;
        } else if (!strcasecmp(type, "all_rtt")) {
            stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT;
        } else if (!strcasecmp(type, "all_loss")) {
            stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS;
        } else if (!strcasecmp(type, "all_mes")) {
            stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_MES;
        }
 
        if (!ast_rtp_instance_get_quality(rtp, stat_field, buf, buflen)) {
            ast_log(AST_LOG_WARNING, "Unable to retrieve 'rtcp' statistics for %s\n", ast_channel_name(chan));
            return -1;
        }
    } else {
        struct ast_rtp_instance_stats stats;
        int i;
        struct {
            const char *name;
            enum { INT, DBL } type;
            union {
                unsigned int *i4;
                double *d8;
            };
        } lookup[] = {
            { "txcount",               INT, { .i4 = &stats.txcount, }, },
            { "rxcount",               INT, { .i4 = &stats.rxcount, }, },
            { "txjitter",              DBL, { .d8 = &stats.txjitter, }, },
            { "rxjitter",              DBL, { .d8 = &stats.rxjitter, }, },
            { "remote_maxjitter",      DBL, { .d8 = &stats.remote_maxjitter, }, },
            { "remote_minjitter",      DBL, { .d8 = &stats.remote_minjitter, }, },
            { "remote_normdevjitter",  DBL, { .d8 = &stats.remote_normdevjitter, }, },
            { "remote_stdevjitter",    DBL, { .d8 = &stats.remote_stdevjitter, }, },
            { "local_maxjitter",       DBL, { .d8 = &stats.local_maxjitter, }, },
            { "local_minjitter",       DBL, { .d8 = &stats.local_minjitter, }, },
            { "local_normdevjitter",   DBL, { .d8 = &stats.local_normdevjitter, }, },
            { "local_stdevjitter",     DBL, { .d8 = &stats.local_stdevjitter, }, },
            { "txploss",               INT, { .i4 = &stats.txploss, }, },
            { "rxploss",               INT, { .i4 = &stats.rxploss, }, },
            { "remote_maxrxploss",     DBL, { .d8 = &stats.remote_maxrxploss, }, },
            { "remote_minrxploss",     DBL, { .d8 = &stats.remote_minrxploss, }, },
            { "remote_normdevrxploss", DBL, { .d8 = &stats.remote_normdevrxploss, }, },
            { "remote_stdevrxploss",   DBL, { .d8 = &stats.remote_stdevrxploss, }, },
            { "local_maxrxploss",      DBL, { .d8 = &stats.local_maxrxploss, }, },
            { "local_minrxploss",      DBL, { .d8 = &stats.local_minrxploss, }, },
            { "local_normdevrxploss",  DBL, { .d8 = &stats.local_normdevrxploss, }, },
            { "local_stdevrxploss",    DBL, { .d8 = &stats.local_stdevrxploss, }, },
            { "rtt",                   DBL, { .d8 = &stats.rtt, }, },
            { "maxrtt",                DBL, { .d8 = &stats.maxrtt, }, },
            { "minrtt",                DBL, { .d8 = &stats.minrtt, }, },
            { "normdevrtt",            DBL, { .d8 = &stats.normdevrtt, }, },
            { "stdevrtt",              DBL, { .d8 = &stats.stdevrtt, }, },
            { "local_ssrc",            INT, { .i4 = &stats.local_ssrc, }, },
            { "remote_ssrc",           INT, { .i4 = &stats.remote_ssrc, }, },
            { "txmes",                 DBL, { .d8 = &stats.txmes, }, },
            { "rxmes",                 DBL, { .d8 = &stats.rxmes, }, },
            { "remote_maxmes",         DBL, { .d8 = &stats.remote_maxmes, }, },
            { "remote_minmes",         DBL, { .d8 = &stats.remote_minmes, }, },
            { "remote_normdevmes",     DBL, { .d8 = &stats.remote_normdevmes, }, },
            { "remote_stdevmes",       DBL, { .d8 = &stats.remote_stdevmes, }, },
            { "local_maxmes",          DBL, { .d8 = &stats.local_maxmes, }, },
            { "local_minmes",          DBL, { .d8 = &stats.local_minmes, }, },
            { "local_normdevmes",      DBL, { .d8 = &stats.local_normdevmes, }, },
            { "local_stdevmes",        DBL, { .d8 = &stats.local_stdevmes, }, },
            { NULL, },
        };
 
        if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
            ast_log(AST_LOG_WARNING, "Unable to retrieve 'rtcp' statistics for %s\n", ast_channel_name(chan));
            return -1;
        }
 
        for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
            if (!strcasecmp(type, lookup[i].name)) {
                if (lookup[i].type == INT) {
                    snprintf(buf, buflen, "%u", *lookup[i].i4);
                } else {
                    snprintf(buf, buflen, "%f", *lookup[i].d8);
                }
                return 0;
            }
        }
        ast_log(AST_LOG_WARNING, "Unrecognized argument '%s' for 'rtcp' information\n", type);
        return -1;
    }
 
    return 0;
}
 
/* The original Request-URI is cloned before channel creation and never changed. */
static int read_request_uri(struct ast_channel *chan, struct ast_sip_session *session,
    char *buf, size_t size)
{
    int length;
    char *copy;
 
    if (!session->request_uri) {
        return 0;
    }
    length = pjsip_uri_print(PJSIP_URI_IN_REQ_URI, session->request_uri, buf, size);
    if (length < 0 || length >= size) {
        ast_log(LOG_ERROR, "Channel %s: Request URI too long for %zu byte buffer\n",
            ast_channel_name(chan), size);
        buf[0] = '\0';
        return -1;
    }
    buf[length] = '\0';
    copy = ast_strdupa(buf);
    ast_escape_quoted(copy, buf, size);
    return 0;
}
 
/* The caller holds the channel lock. No dialog lock or serializer wait is allowed. */
static int channel_read_pjsip(struct ast_channel *chan, struct ast_sip_session *session,
    const char *type, char *buf, size_t len)
{
    const struct pjsip_channel_info *info;
 
    if (ast_strlen_zero(type)) {
        ast_log(LOG_WARNING, "You must supply a type field for 'pjsip' information\n");
        return -1;
    }
 
    if (!strcmp(type, "call-id") || !strcmp(type, "local_tag")) {
        pjsip_dialog *dlg = session->inv_session ? session->inv_session->dlg : NULL;
 
        if (!dlg) {
            return -1;
        }
        /* These identities are fixed for the lifetime of the retained dialog. */
        if (!strcmp(type, "call-id")) {
            ast_copy_pj_str(buf, &dlg->call_id->id, len);
        } else {
            char *copy;
 
            ast_copy_pj_str(buf, &dlg->local.info->tag, len);
            copy = ast_strdupa(buf);
            ast_escape_quoted(copy, buf, len);
        }
        return 0;
    }
    if (!strcmp(type, "request_uri")) {
        return read_request_uri(chan, session, buf, len);
    }
    if (!strcmp(type, "t38state")) {
        if (session->t38state < 0 || session->t38state >= T38_MAX_ENUM) {
            return -1;
        }
        ast_copy_string(buf, t38state_to_string[session->t38state], len);
        return 0;
    }
    if (!strcmp(type, "local_addr") || !strcmp(type, "remote_addr")) {
        RAII_VAR(struct ast_datastore *, datastore,
            ast_sip_session_get_datastore(session, "transport_info"), ao2_cleanup);
        struct transport_info_data *transport;
        const pj_sockaddr *addr;
 
        if (!datastore) {
            ast_log(LOG_WARNING, "No transport information for channel %s\n",
                ast_channel_name(chan));
            return -1;
        }
        /* The transport payload is fully initialized before publication and immutable. */
        transport = datastore->data;
        addr = !strcmp(type, "local_addr") ? &transport->local_addr : &transport->remote_addr;
        if (pj_sockaddr_has_addr(addr)) {
            pj_sockaddr_print(addr, buf, len, 3);
        }
        return 0;
    }
 
    info = pjsip_channel_info_get(chan);
    if (!info) {
        ast_log(LOG_WARNING, "Channel %s has no PJSIP channel information\n",
            ast_channel_name(chan));
        return -1;
    }
    if (!strcmp(type, "secure")) {
#ifdef HAVE_PJSIP_GET_DEST_INFO
        if (info->secure < 0) {
            return -1;
        }
        snprintf(buf, len, "%d", info->secure);
#else
        ast_log(LOG_WARNING, "Asterisk was built without support for CHANNEL(pjsip,secure)\n");
        return -1;
#endif
    } else if (!strcmp(type, "target_uri")) {
        return copy_escaped_uri(chan, type, &info->target_uri, buf, len);
    } else if (!strcmp(type, "local_uri")) {
        return copy_escaped_uri(chan, type, &info->local_uri, buf, len);
    } else if (!strcmp(type, "remote_uri")) {
        return copy_escaped_uri(chan, type, &info->remote_uri, buf, len);
    } else if (!strcmp(type, "remote_tag")) {
        if (!info->remote_tag) {
            return -1;
        }
        ast_escape_quoted(info->remote_tag, buf, len);
    } else {
        ast_log(LOG_WARNING, "Unrecognized argument '%s' for 'pjsip' information\n", type);
        return -1;
    }
    return 0;
}
 
int pjsip_acf_channel_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
    struct ast_sip_channel_pvt *channel;
    char *parse = ast_strdupa(data);
    int res = 0;
 
    AST_DECLARE_APP_ARGS(args,
        AST_APP_ARG(param);
        AST_APP_ARG(type);
        AST_APP_ARG(field);
    );
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    /* Check for zero arguments */
    if (ast_strlen_zero(parse)) {
        ast_log(LOG_ERROR, "Cannot call %s without arguments\n", cmd);
        return -1;
    }
 
    if (!len) {
        ast_log(LOG_ERROR, "Cannot call %s without an output buffer\n", cmd);
        return -1;
    }
 
    AST_STANDARD_APP_ARGS(args, parse);
 
    ast_channel_lock(chan);
 
    /* Sanity check */
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        ast_channel_unlock(chan);
        return 0;
    }
 
    channel = ast_channel_tech_pvt(chan);
    if (!channel) {
        ast_log(LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    if (!channel->session) {
        ast_log(LOG_WARNING, "Channel %s has no session\n", ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    memset(buf, 0, len);
 
    if (!strcmp(args.param, "endpoint")) {
        if (!channel->session->endpoint) {
            res = -1;
        } else {
            ast_copy_string(buf, ast_sorcery_object_get_id(channel->session->endpoint), len);
        }
    } else if (!strcmp(args.param, "contact")) {
        if (channel->session->contact) {
            ast_copy_string(buf, ast_sorcery_object_get_id(channel->session->contact), len);
        }
    } else if (!strcmp(args.param, "aor")) {
        if (channel->session->aor) {
            ast_copy_string(buf, ast_sorcery_object_get_id(channel->session->aor), len);
        }
    } else if (!strcmp(args.param, "pjsip")) {
        res = channel_read_pjsip(chan, channel->session, args.type, buf, len);
    } else if (!strcmp(args.param, "rtp") || !strcmp(args.param, "rtcp")) {
        const struct pjsip_channel_info *info = pjsip_channel_info_get(chan);
        struct ast_rtp_instance *rtp;
        struct ast_sip_session_media *media;
        struct ast_sockaddr no_address = { { 0, } };
        const struct ast_sockaddr *direct_media_addr;
        unsigned int secure;
        unsigned int held;
        int video = !strcmp(S_OR(args.field, "audio"), "video");
 
        if (!info) {
            res = -1;
        } else {
            /* The channel lock keeps these retained instances alive throughout the read. */
            rtp = video ? info->video_rtp : info->audio_rtp;
            media = channel->session->active_media_state
                ? channel->session->active_media_state->default_session[video
                    ? AST_MEDIA_TYPE_VIDEO : AST_MEDIA_TYPE_AUDIO] : NULL;
            direct_media_addr = media ? &media->direct_media_addr : &no_address;
            secure = video ? info->video_secure : info->audio_secure;
            held = video ? info->video_held : info->audio_held;
            if (!strcmp(args.param, "rtp")) {
                res = channel_read_rtp(chan, rtp, direct_media_addr, secure, held,
                    args.type, args.field, buf, len);
            } else {
                res = channel_read_rtcp(chan, rtp, args.type, args.field, buf, len);
            }
        }
    } else {
        res = -1;
    }
    ast_channel_unlock(chan);
    return res;
}
 
int pjsip_acf_dial_contacts_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
    RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
    RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
    const char *aor_name;
    char *rest;
 
    AST_DECLARE_APP_ARGS(args,
        AST_APP_ARG(endpoint_name);
        AST_APP_ARG(aor_name);
        AST_APP_ARG(request_user);
    );
 
    AST_STANDARD_APP_ARGS(args, data);
 
    if (ast_strlen_zero(args.endpoint_name)) {
        ast_log(LOG_WARNING, "An endpoint name must be specified when using the '%s' dialplan function\n", cmd);
        return -1;
    } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", args.endpoint_name))) {
        ast_log(LOG_WARNING, "Specified endpoint '%s' was not found\n", args.endpoint_name);
        return -1;
    }
 
    aor_name = S_OR(args.aor_name, endpoint->aors);
 
    if (ast_strlen_zero(aor_name)) {
        ast_log(LOG_WARNING, "No AOR has been provided and no AORs are configured on endpoint '%s'\n", args.endpoint_name);
        return -1;
    } else if (!(dial = ast_str_create(len))) {
        ast_log(LOG_WARNING, "Could not get enough buffer space for dialing contacts\n");
        return -1;
    } else if (!(rest = ast_strdupa(aor_name))) {
        ast_log(LOG_WARNING, "Could not duplicate provided AORs\n");
        return -1;
    }
 
    while ((aor_name = ast_strip(strsep(&rest, ",")))) {
        RAII_VAR(struct ast_sip_aor *, aor, ast_sip_location_retrieve_aor(aor_name), ao2_cleanup);
        RAII_VAR(struct ao2_container *, contacts, NULL, ao2_cleanup);
        struct ao2_iterator it_contacts;
        struct ast_sip_contact *contact;
 
        if (!aor) {
            /* If the AOR provided is not found skip it, there may be more */
            continue;
        } else if (!(contacts = ast_sip_location_retrieve_aor_contacts_filtered(aor, AST_SIP_CONTACT_FILTER_REACHABLE))) {
            /* No contacts are available, skip it as well */
            continue;
        } else if (!ao2_container_count(contacts)) {
            /* We were given a container but no contacts are in it... */
            continue;
        }
 
        it_contacts = ao2_iterator_init(contacts, 0);
        for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
            ast_str_append(&dial, -1, "PJSIP/");
 
            if (!ast_strlen_zero(args.request_user)) {
                ast_str_append(&dial, -1, "%s@", args.request_user);
            }
            ast_str_append(&dial, -1, "%s/%s&", args.endpoint_name, contact->uri);
        }
        ao2_iterator_destroy(&it_contacts);
    }
 
    /* Trim the '&' at the end off */
    ast_str_truncate(dial, ast_str_strlen(dial) - 1);
 
    ast_copy_string(buf, ast_str_buffer(dial), len);
 
    return 0;
}
 
/*! \brief Session refresh state information */
struct session_refresh_state {
    /*! \brief Created proposed media state */
    struct ast_sip_session_media_state *media_state;
};
 
/*! \brief Destructor for session refresh information */
static void session_refresh_state_destroy(void *obj)
{
    struct session_refresh_state *state = obj;
 
    ast_sip_session_media_state_free(state->media_state);
    ast_free(obj);
}
 
/*! \brief Datastore for attaching session refresh state information */
static const struct ast_datastore_info session_refresh_datastore = {
    .type = "pjsip_session_refresh",
    .destroy = session_refresh_state_destroy,
};
 
/*! \brief Helper function which retrieves or allocates a session refresh state information datastore */
static struct session_refresh_state *session_refresh_state_get_or_alloc(struct ast_sip_session *session)
{
    RAII_VAR(struct ast_datastore *, datastore, ast_sip_session_get_datastore(session, "pjsip_session_refresh"), ao2_cleanup);
    struct session_refresh_state *state;
 
    /* While the datastore refcount is decremented this is operating in the serializer so it will remain valid regardless */
    if (datastore) {
        return datastore->data;
    }
 
    if (!(datastore = ast_sip_session_alloc_datastore(&session_refresh_datastore, "pjsip_session_refresh"))
        || !(datastore->data = ast_calloc(1, sizeof(struct session_refresh_state)))
        || ast_sip_session_add_datastore(session, datastore)) {
        return NULL;
    }
 
    state = datastore->data;
    state->media_state = ast_sip_session_media_state_alloc();
    if (!state->media_state) {
        ast_sip_session_remove_datastore(session, "pjsip_session_refresh");
        return NULL;
    }
    state->media_state->topology = ast_stream_topology_clone(session->endpoint->media.topology);
    if (!state->media_state->topology) {
        ast_sip_session_remove_datastore(session, "pjsip_session_refresh");
        return NULL;
    }
 
    datastore->data = state;
 
    return state;
}
 
/*! \brief Struct used to push PJSIP_PARSE_URI function arguments to task processor */
struct parse_uri_args {
    const char *uri;
    const char *type;
    char *buf;
    size_t buflen;
    int ret;
};
 
/*! \internal \brief Taskprocessor callback that handles the PJSIP_PARSE_URI on a PJSIP thread */
static int parse_uri_cb(void *data)
{
    struct parse_uri_args *args = data;
    pj_pool_t *pool;
    pjsip_name_addr *uri;
    pjsip_sip_uri *sip_uri;
    pj_str_t tmp;
 
    args->ret = 0;
 
    pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "ParseUri", 128, 128);
    if (!pool) {
        ast_log(LOG_ERROR, "Failed to allocate ParseUri endpoint pool.\n");
        args->ret = -1;
        return 0;
    }
 
    pj_strdup2_with_null(pool, &tmp, args->uri);
    uri = (pjsip_name_addr *)pjsip_parse_uri(pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR);
    if (!uri || (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
        ast_log(LOG_WARNING, "Failed to parse URI '%s'\n", args->uri);
        pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
        args->ret = -1;
        return 0;
    }
 
    if (!strcmp(args->type, "scheme")) {
        ast_copy_pj_str(args->buf, pjsip_uri_get_scheme(uri), args->buflen);
        pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
        return 0;
    } else if (!strcmp(args->type, "display")) {
        ast_copy_pj_str(args->buf, &uri->display, args->buflen);
        pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
        return 0;
    }
 
    sip_uri = pjsip_uri_get_uri(uri);
    if (!sip_uri) {
        ast_log(LOG_ERROR, "Failed to get an URI object for '%s'\n", args->uri);
        pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
        args->ret = -1;
        return 0;
    }
 
    if (!strcmp(args->type, "user")) {
        ast_copy_pj_str(args->buf, &sip_uri->user, args->buflen);
    } else if (!strcmp(args->type, "passwd")) {
        ast_copy_pj_str(args->buf, &sip_uri->passwd, args->buflen);
    } else if (!strcmp(args->type, "host")) {
        ast_copy_pj_str(args->buf, &sip_uri->host, args->buflen);
    } else if (!strcmp(args->type, "port")) {
        snprintf(args->buf, args->buflen, "%d", sip_uri->port);
    } else if (!strcmp(args->type, "user_param")) {
        ast_copy_pj_str(args->buf, &sip_uri->user_param, args->buflen);
    } else if (!strcmp(args->type, "method_param")) {
        ast_copy_pj_str(args->buf, &sip_uri->method_param, args->buflen);
    } else if (!strcmp(args->type, "transport_param")) {
        ast_copy_pj_str(args->buf, &sip_uri->transport_param, args->buflen);
    } else if (!strcmp(args->type, "ttl_param")) {
        snprintf(args->buf, args->buflen, "%d", sip_uri->ttl_param);
    } else if (!strcmp(args->type, "lr_param")) {
        snprintf(args->buf, args->buflen, "%d", sip_uri->lr_param);
    } else if (!strcmp(args->type, "maddr_param")) {
        ast_copy_pj_str(args->buf, &sip_uri->maddr_param, args->buflen);
    } else {
        ast_log(AST_LOG_WARNING, "Unknown type part '%s' specified\n", args->type);
        pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
        args->ret = -1;
        return 0;
    }
 
    pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
 
    return 0;
}
 
int pjsip_acf_parse_uri_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t buflen)
{
    struct parse_uri_args func_args = { 0, };
    int reading_uri_from_var;
 
    AST_DECLARE_APP_ARGS(args,
        AST_APP_ARG(uri_str);
        AST_APP_ARG(type);
    );
 
    AST_STANDARD_APP_ARGS(args, data);
 
    reading_uri_from_var = !strcasecmp(cmd, "PJSIP_PARSE_URI_FROM");
 
    if (reading_uri_from_var) {
        const char *var;
 
        if (ast_strlen_zero(args.uri_str)) {
            ast_log(LOG_WARNING, "The name of a variable containing a URI must be specified when using the '%s' dialplan function\n", cmd);
            return -1;
        }
 
        ast_channel_lock(chan);
        if ((var = pbx_builtin_getvar_helper(chan, args.uri_str))) {
            args.uri_str = ast_strdupa(var);
        }
        ast_channel_unlock(chan);
    }
 
    if (ast_strlen_zero(args.uri_str)) {
        if (reading_uri_from_var) {
            ast_log(LOG_WARNING, "The variable provided to the '%s' dialplan function must contain a URI\n", cmd);
        } else {
            ast_log(LOG_WARNING, "A URI must be specified when using the '%s' dialplan function\n", cmd);
        }
        return -1;
    }
 
    if (ast_strlen_zero(args.type)) {
        ast_log(LOG_WARNING, "A type part of the URI must be specified when using the '%s' dialplan function\n", cmd);
        return -1;
    }
 
    memset(buf, 0, buflen);
 
    func_args.uri = args.uri_str;
    func_args.type = args.type;
    func_args.buf = buf;
    func_args.buflen = buflen;
    if (ast_sip_push_task_wait_serializer(NULL, parse_uri_cb, &func_args)) {
        ast_log(LOG_WARNING, "Unable to parse URI: failed to push task\n");
        return -1;
    }
 
    return func_args.ret;
}
 
static int media_offer_read_av(struct ast_sip_session *session, char *buf,
                   size_t len, enum ast_media_type media_type)
{
    struct ast_stream_topology *topology;
    int idx;
    struct ast_stream *stream = NULL;
    const struct ast_format_cap *caps;
    size_t accum = 0;
 
    if (session->inv_session->dlg->state == PJSIP_DIALOG_STATE_ESTABLISHED) {
        struct session_refresh_state *state;
 
        /* As we've already answered we need to store our media state until we are ready to send it */
        state = session_refresh_state_get_or_alloc(session);
        if (!state) {
            return -1;
        }
        topology = state->media_state->topology;
    } else {
        /* The session is not yet up so we are initially answering or offering */
        if (!session->pending_media_state->topology) {
            session->pending_media_state->topology = ast_stream_topology_clone(session->endpoint->media.topology);
            if (!session->pending_media_state->topology) {
                return -1;
            }
        }
        topology = session->pending_media_state->topology;
    }
 
    /* Find the first suitable stream */
    for (idx = 0; idx < ast_stream_topology_get_count(topology); ++idx) {
        stream = ast_stream_topology_get_stream(topology, idx);
 
        if (ast_stream_get_type(stream) != media_type ||
            ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED) {
            stream = NULL;
            continue;
        }
 
        break;
    }
 
    /* If no suitable stream then exit early */
    if (!stream) {
        buf[0] = '\0';
        return 0;
    }
 
    caps = ast_stream_get_formats(stream);
 
    /* Note: buf is not terminated while the string is being built. */
    for (idx = 0; idx < ast_format_cap_count(caps); ++idx) {
        struct ast_format *fmt;
        size_t size;
 
        fmt = ast_format_cap_get_format(caps, idx);
 
        /* Add one for a comma or terminator */
        size = strlen(ast_format_get_name(fmt)) + 1;
        if (len < size) {
            ao2_ref(fmt, -1);
            break;
        }
 
        /* Append the format name */
        strcpy(buf + accum, ast_format_get_name(fmt));/* Safe */
        ao2_ref(fmt, -1);
 
        accum += size;
        len -= size;
 
        /* The last comma on the built string will be set to the terminator. */
        buf[accum - 1] = ',';
    }
 
    /* Remove the trailing comma or terminate an empty buffer. */
    buf[accum ? accum - 1 : 0] = '\0';
    return 0;
}
 
struct media_offer_data {
    struct ast_sip_session *session;
    enum ast_media_type media_type;
    const char *value;
};
 
static int media_offer_write_av(void *obj)
{
    struct media_offer_data *data = obj;
    struct ast_stream_topology *topology;
    struct ast_stream *stream;
    struct ast_format_cap *caps;
 
    if (data->session->inv_session->dlg->state == PJSIP_DIALOG_STATE_ESTABLISHED) {
        struct session_refresh_state *state;
 
        /* As we've already answered we need to store our media state until we are ready to send it */
        state = session_refresh_state_get_or_alloc(data->session);
        if (!state) {
            return -1;
        }
        topology = state->media_state->topology;
    } else {
        /* The session is not yet up so we are initially answering or offering */
        if (!data->session->pending_media_state->topology) {
            data->session->pending_media_state->topology = ast_stream_topology_clone(data->session->endpoint->media.topology);
            if (!data->session->pending_media_state->topology) {
                return -1;
            }
        }
        topology = data->session->pending_media_state->topology;
    }
 
    /* XXX This method won't work when it comes time to do multistream support. The proper way to do this
     * will either be to
     * a) Alter all media streams of a particular type.
     * b) Change the dialplan function to be able to specify which stream to alter and alter only that
     * one stream
     */
    stream = ast_stream_topology_get_first_stream_by_type(topology, data->media_type);
    if (!stream) {
        return 0;
    }
 
    caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
    if (!caps) {
        return -1;
    }
 
    ast_format_cap_append_from_cap(caps, ast_stream_get_formats(stream),
        AST_MEDIA_TYPE_UNKNOWN);
    ast_format_cap_remove_by_type(caps, data->media_type);
    ast_format_cap_update_by_allow_disallow(caps, data->value, 1);
    ast_stream_set_formats(stream, caps);
    ast_stream_set_metadata(stream, "pjsip_session_refresh", "force");
    ao2_ref(caps, -1);
 
    return 0;
}
 
int pjsip_acf_media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
    struct ast_sip_channel_pvt *channel;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
 
    if (!strcmp(data, "audio")) {
        return media_offer_read_av(channel->session, buf, len, AST_MEDIA_TYPE_AUDIO);
    } else if (!strcmp(data, "video")) {
        return media_offer_read_av(channel->session, buf, len, AST_MEDIA_TYPE_VIDEO);
    } else {
        /* Ensure that the buffer is empty */
        buf[0] = '\0';
    }
 
    return 0;
}
 
int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
    struct ast_sip_channel_pvt *channel;
    struct media_offer_data mdata = {
        .value = value
    };
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
    mdata.session = channel->session;
 
    if (!strcmp(data, "audio")) {
        mdata.media_type = AST_MEDIA_TYPE_AUDIO;
    } else if (!strcmp(data, "video")) {
        mdata.media_type = AST_MEDIA_TYPE_VIDEO;
    }
 
    return ast_sip_push_task_wait_serializer(channel->session->serializer, media_offer_write_av, &mdata);
}
 
int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
    struct ast_sip_channel_pvt *channel;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        ast_channel_unlock(chan);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
 
    if (ast_sip_dtmf_to_str(channel->session->dtmf, buf, len) < 0) {
        ast_log(LOG_WARNING, "Unknown DTMF mode %d on PJSIP channel %s\n", channel->session->dtmf, ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    ast_channel_unlock(chan);
    return 0;
}
 
int pjsip_acf_moh_passthrough_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
    struct ast_sip_channel_pvt *channel;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    if (len < 3) {
        ast_log(LOG_WARNING, "%s: buffer too small\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        ast_channel_unlock(chan);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
    strncpy(buf, AST_YESNO(channel->session->moh_passthrough), len);
 
    ast_channel_unlock(chan);
    return 0;
}
 
struct refresh_data {
    struct ast_sip_session *session;
    enum ast_sip_session_refresh_method method;
};
 
static int sip_session_response_cb(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
    struct ast_format *fmt;
 
    if (!session->channel) {
        /* Egads! */
        return 0;
    }
 
    fmt = ast_format_cap_get_best_by_type(ast_channel_nativeformats(session->channel), AST_MEDIA_TYPE_AUDIO);
    if (!fmt) {
        /* No format? That's weird. */
        return 0;
    }
    ast_channel_set_writeformat(session->channel, fmt);
    ast_channel_set_rawwriteformat(session->channel, fmt);
    ast_channel_set_readformat(session->channel, fmt);
    ast_channel_set_rawreadformat(session->channel, fmt);
    ao2_ref(fmt, -1);
 
    return 0;
}
 
static int dtmf_mode_refresh_cb(void *obj)
{
    struct refresh_data *data = obj;
 
    if (data->session->inv_session->state == PJSIP_INV_STATE_CONFIRMED) {
        ast_debug(3, "Changing DTMF mode on channel %s after OFFER/ANSWER completion. Sending session refresh\n", ast_channel_name(data->session->channel));
 
        ast_sip_session_refresh(data->session, NULL, NULL,
            sip_session_response_cb, data->method, 1, NULL);
    } else if (data->session->inv_session->state == PJSIP_INV_STATE_INCOMING) {
        ast_debug(3, "Changing DTMF mode on channel %s during OFFER/ANSWER exchange. Updating SDP answer\n", ast_channel_name(data->session->channel));
        ast_sip_session_regenerate_answer(data->session, NULL);
    }
 
    return 0;
}
 
int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
    struct ast_sip_channel_pvt *channel;
    struct ast_sip_session_media *media;
    int dsp_features = 0;
    int dtmf = -1;
    struct refresh_data rdata = {
            .method = AST_SIP_SESSION_REFRESH_METHOD_INVITE,
        };
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        ast_channel_unlock(chan);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
    rdata.session = channel->session;
 
    dtmf = ast_sip_str_to_dtmf(value);
 
    if (dtmf == -1) {
        ast_log(LOG_WARNING, "Cannot set DTMF mode to '%s' on channel '%s' as value is invalid.\n", value,
            ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    if (channel->session->dtmf == dtmf) {
        /* DTMF mode unchanged, nothing to do! */
        ast_channel_unlock(chan);
        return 0;
    }
 
    channel->session->dtmf = dtmf;
 
    media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
 
    if (media && media->rtp) {
        if (channel->session->dtmf == AST_SIP_DTMF_RFC_4733) {
            ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 1);
            ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_RFC2833);
        } else if (channel->session->dtmf == AST_SIP_DTMF_INFO) {
            ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
            ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_NONE);
        } else if (channel->session->dtmf == AST_SIP_DTMF_INBAND) {
            ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
            ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_INBAND);
        } else if (channel->session->dtmf == AST_SIP_DTMF_NONE) {
            ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
            ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_NONE);
        } else if (channel->session->dtmf == AST_SIP_DTMF_AUTO) {
            if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_RFC2833) {
                /* no RFC4733 negotiated, enable inband */
                ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_INBAND);
            }
        } else if (channel->session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
            ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
            if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
                /* if inband, switch to INFO */
                ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_NONE);
            }
        }
    }
 
    if (channel->session->dsp) {
        dsp_features = ast_dsp_get_features(channel->session->dsp);
    }
    if (channel->session->dtmf == AST_SIP_DTMF_INBAND ||
        channel->session->dtmf == AST_SIP_DTMF_AUTO) {
        dsp_features |= DSP_FEATURE_DIGIT_DETECT;
    } else {
        dsp_features &= ~DSP_FEATURE_DIGIT_DETECT;
    }
    if (dsp_features) {
        if (!channel->session->dsp) {
            if (!(channel->session->dsp = ast_dsp_new())) {
                ast_channel_unlock(chan);
                return 0;
            }
        }
        ast_dsp_set_features(channel->session->dsp, dsp_features);
    } else if (channel->session->dsp) {
        ast_dsp_free(channel->session->dsp);
        channel->session->dsp = NULL;
    }
 
    ast_channel_unlock(chan);
 
    return ast_sip_push_task_wait_serializer(channel->session->serializer, dtmf_mode_refresh_cb, &rdata);
}
 
int pjsip_acf_moh_passthrough_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
    struct ast_sip_channel_pvt *channel;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        ast_channel_unlock(chan);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
    channel->session->moh_passthrough = ast_true(value);
 
    ast_channel_unlock(chan);
 
    return 0;
}
 
static int refresh_write_cb(void *obj)
{
    struct refresh_data *data = obj;
    struct session_refresh_state *state;
 
    state = session_refresh_state_get_or_alloc(data->session);
    if (!state) {
        return -1;
    }
 
    ast_sip_session_refresh(data->session, NULL, NULL,
        sip_session_response_cb, data->method, 1, state->media_state);
 
    state->media_state = NULL;
    ast_sip_session_remove_datastore(data->session, "pjsip_session_refresh");
 
    return 0;
}
 
int pjsip_acf_session_refresh_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
    struct ast_sip_channel_pvt *channel;
    struct refresh_data rdata = {
        .method = AST_SIP_SESSION_REFRESH_METHOD_INVITE,
    };
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    if (ast_channel_state(chan) != AST_STATE_UP) {
        ast_log(LOG_WARNING, "'%s' not allowed on unanswered channel '%s'.\n", cmd, ast_channel_name(chan));
        return -1;
    }
 
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
    rdata.session = channel->session;
 
    if (!strcmp(value, "invite")) {
        rdata.method = AST_SIP_SESSION_REFRESH_METHOD_INVITE;
    } else if (!strcmp(value, "update")) {
        rdata.method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
    }
 
    return ast_sip_push_task_wait_serializer(channel->session->serializer, refresh_write_cb, &rdata);
}
 
struct hangup_data {
    struct ast_sip_session *session;
    int response_code;
};
 
/*!
 * \brief Serializer task to hangup channel
 */
static int pjsip_hangup(void *obj)
{
    struct hangup_data *hdata = obj;
    pjsip_tx_data *packet = NULL;
 
    if ((hdata->session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED) &&
        (pjsip_inv_answer(hdata->session->inv_session, hdata->response_code, NULL, NULL, &packet) == PJ_SUCCESS)) {
        ast_sip_session_send_response(hdata->session, packet);
    }
 
    return 0;
}
 
/*!
 * \brief Callback that validates the response code
 */
static int response_code_validator(const char *channel_name,
    const char *response) {
    int response_code;
 
    int rc = ast_str_to_int(response, &response_code);
    if (rc != 0) {
        response_code = ast_sip_str2rc(response);
        if (response_code < 0) {
            ast_log(LOG_WARNING, "%s: Unrecognized response code parameter '%s'."
                " Defaulting to 603 DECLINE\n",
                channel_name, response);
            return PJSIP_SC_DECLINE;
        }
    }
 
    if (response_code < 400 || response_code > 699) {
        ast_log(LOG_WARNING, "%s: Response code %d is out of range 400 -> 699."
            " Defaulting to 603 DECLINE\n",
            channel_name, response_code);
        return PJSIP_SC_DECLINE;
    }
    return response_code;
}
 
/*!
 * \brief Called by pjsip_app_hangup and pjsip_action_hangup
 *        to actually perform the hangup
 */
static void pjsip_app_hangup_handler(struct ast_channel *chan, int response_code)
{
    struct ast_sip_channel_pvt *channel;
    struct hangup_data hdata = { NULL, -1 };
    const char *tag = ast_channel_name(chan);
 
    hdata.response_code = response_code;
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "%s: Not a PJSIP channel\n", tag);
        ast_channel_unlock(chan);
        return;
    }
 
    channel = ast_channel_tech_pvt(chan);
    hdata.session = channel->session;
 
    if (hdata.session->inv_session->role != PJSIP_ROLE_UAS || (
        hdata.session->inv_session->state != PJSIP_INV_STATE_INCOMING &&
        hdata.session->inv_session->state != PJSIP_INV_STATE_EARLY)) {
        ast_log(LOG_WARNING, "%s: Not an incoming channel or invalid state '%s'\n",
            tag, pjsip_inv_state_name(hdata.session->inv_session->state));
        ast_channel_unlock(chan);
        return;
    }
 
    ast_channel_unlock(chan);
 
    if (ast_sip_push_task_wait_serializer(channel->session->serializer,
        pjsip_hangup, &hdata) != 0) {
        ast_log(LOG_WARNING, "%s: failed to push hangup task to serializer\n", tag);
    }
 
    return;
}
 
/*!
 * \brief PJSIPHangup Dialplan App
 */
int pjsip_app_hangup(struct ast_channel *chan, const char *data)
{
    int response_code;
    const char *tag = ast_channel_name(chan);
 
    if (ast_strlen_zero(data)) {
        ast_log(LOG_WARNING, "%s: Missing response code parameter\n", tag);
        return -1;
    }
 
    response_code = response_code_validator(tag, data);
 
    pjsip_app_hangup_handler(chan, response_code);
 
    return -1;
}
 
/*!
 * \brief PJSIPHangup Manager Action
 */
int pjsip_action_hangup(struct mansession *s, const struct message *m)
{
    return ast_manager_hangup_helper(s, m,
        pjsip_app_hangup_handler, response_code_validator);
}
 
int pjsip_transfer_handling_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
    struct ast_sip_channel_pvt *channel;
    int ret = 0;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel %s\n", cmd, ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
 
    if (ast_strlen_zero(value) || !strcmp(value, "core")) {
        channel->session->transferhandling_ari = 0;
    } else if (!strcmp(value, "ari-only")) {
        channel->session->transferhandling_ari = 1;
    } else {
        ast_log(AST_LOG_WARNING, "Cannot set unknown transfer handling '%s' on channel '%s', transfer handling will remain unchanged.",
            value, ast_channel_name(chan));
        ret = -1;
    }
 
    ast_channel_unlock(chan);
    return ret;
}