res_pjsip_messaging.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2013, Digium, Inc.
 *
 * Kevin Harwell <kharwell@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.
 */
 
/*** MODULEINFO
    <depend>pjproject</depend>
    <depend>res_pjsip</depend>
    <depend>res_pjsip_session</depend>
    <support_level>core</support_level>
 ***/
 
/*** DOCUMENTATION
    <info name="MessageDestinationInfo" language="en_US" tech="PJSIP">
        <para>The <literal>destination</literal> parameter is used to construct
        the Request URI for an outgoing message.  It can be in one of the following
        formats, all prefixed with the <literal>pjsip:</literal> message tech.</para>
        <para>
        </para>
        <enumlist>
            <enum name="endpoint">
                <para>Request URI comes from the endpoint's default aor and contact.</para>
            </enum>
            <enum name="endpoint/aor">
                <para>Request URI comes from the specific aor/contact.</para>
            </enum>
            <enum name="endpoint@domain">
                <para>Request URI from the endpoint's default aor and contact.  The domain is discarded.</para>
            </enum>
        </enumlist>
        <para>
        </para>
        <para>These all use the endpoint to send the message with the specified URI:</para>
        <para>
        </para>
        <enumlist>
            <enum name="endpoint/&lt;sip[s]:host&gt;>"/>
            <enum name="endpoint/&lt;sip[s]:user@host&gt;"/>
            <enum name="endpoint/&quot;display name&quot; &lt;sip[s]:host&gt;"/>
            <enum name="endpoint/&quot;display name&quot; &lt;sip[s]:user@host&gt;"/>
            <enum name="endpoint/sip[s]:host"/>
            <enum name="endpoint/sip[s]:user@host"/>
            <enum name="endpoint/host"/>
            <enum name="endpoint/user@host"/>
        </enumlist>
        <para>
        </para>
        <para>These all use the default endpoint to send the message with the specified URI:</para>
        <para>
        </para>
        <enumlist>
            <enum name="&lt;sip[s]:host&gt;"/>
            <enum name="&lt;sip[s]:user@host&gt;"/>
            <enum name="&quot;display name&quot; &lt;sip[s]:host&gt;"/>
            <enum name="&quot;display name&quot; &lt;sip[s]:user@host&gt;"/>
            <enum name="sip[s]:host"/>
            <enum name="sip[s]:user@host"/>
        </enumlist>
        <para>
        </para>
        <para>These use the default endpoint to send the message with the specified host:</para>
        <para>
        </para>
        <enumlist>
            <enum name="host"/>
            <enum name="user@host"/>
        </enumlist>
        <para>
        </para>
        <para>This form is similar to a dialstring:</para>
        <para>
        </para>
        <enumlist>
            <enum name="PJSIP/user@endpoint"/>
        </enumlist>
        <para>
        </para>
        <para>You still need to prefix the destination with
        the <literal>pjsip:</literal> message technology prefix.  For example:
        <literal>pjsip:PJSIP/8005551212@myprovider</literal>.
        The endpoint contact's URI will have the <literal>user</literal> inserted
        into it and will become the Request URI.  If the contact URI already has
        a user specified, it will be replaced.
        </para>
        <para>
        </para>
    </info>
    <info name="MessageFromInfo" language="en_US" tech="PJSIP">
        <para>The <literal>from</literal> parameter is used to specity the <literal>From:</literal>
        header in the outgoing SIP MESSAGE.  It will override the value specified in
        MESSAGE(from) which itself will override any <literal>from</literal> value from
        an incoming SIP MESSAGE.
        </para>
        <para>
        </para>
    </info>
    <info name="MessageToInfo" language="en_US" tech="PJSIP">
        <para>The <literal>to</literal> parameter is used to specity the <literal>To:</literal>
        header in the outgoing SIP MESSAGE.  It will override the value specified in
        MESSAGE(to) which itself will override any <literal>to</literal> value from
        an incoming SIP MESSAGE.
        </para>
        <para>
        </para>
    </info>
 ***/
#include "asterisk.h"
 
#include <pjsip.h>
#include <pjsip_ua.h>
 
#include "asterisk/message.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/test.h"
#include "asterisk/uri.h"
 
const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
 
#define MAX_HDR_SIZE 512
#define MAX_BODY_SIZE 1024
#define MAX_USER_SIZE 128
 
static struct ast_taskprocessor *message_serializer;
 
/*!
 * \internal
 * \brief Checks to make sure the request has the correct content type.
 *
 * \details This module supports the following media types: "text/plain".
 * Return unsupported otherwise.
 *
 * \param rdata The SIP request
 */
static enum pjsip_status_code check_content_type(const pjsip_rx_data *rdata)
{
    int res;
    if (rdata->msg_info.msg->body && rdata->msg_info.msg->body->len) {
        res = ast_sip_is_content_type(
            &rdata->msg_info.msg->body->content_type, "text", "plain");
    } else {
        res = rdata->msg_info.ctype &&
            ast_sip_is_content_type(
                &rdata->msg_info.ctype->media, "text", "plain");
    }
 
    return res ? PJSIP_SC_OK : PJSIP_SC_UNSUPPORTED_MEDIA_TYPE;
}
 
/*!
 * \internal
 * \brief Checks to make sure the request has the correct content type.
 *
 * \details This module supports the following media types: "text/\*", "application/\*".
 * Return unsupported otherwise.
 *
 * \param rdata The SIP request
 */
static enum pjsip_status_code check_content_type_in_dialog(const pjsip_rx_data *rdata)
{
    int res = PJSIP_SC_UNSUPPORTED_MEDIA_TYPE;
    static const pj_str_t text = { "text", 4};
    static const pj_str_t application = { "application", 11};
 
    if (!(rdata->msg_info.msg->body && rdata->msg_info.msg->body->len > 0)) {
        return res;
    }
 
    /* We'll accept any text/ or application/ content type */
    if (pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &text) == 0
            || pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &application) == 0) {
        res = PJSIP_SC_OK;
    } else if (rdata->msg_info.ctype
        && (pj_stricmp(&rdata->msg_info.ctype->media.type, &text) == 0
        || pj_stricmp(&rdata->msg_info.ctype->media.type, &application) == 0)) {
        res = PJSIP_SC_OK;
    }
 
    return res;
}
 
/*!
 * \internal
 * \brief Update the display name in the To uri in the tdata with the one from the supplied uri
 *
 * \param tdata the outbound message data structure
 * \param to uri containing the display name to replace in the the To uri
 *
 * \return 0: success, -1: failure
 */
static int update_to_display_name(pjsip_tx_data *tdata, char *to)
{
    pjsip_name_addr *parsed_name_addr;
 
    parsed_name_addr = (pjsip_name_addr *) pjsip_parse_uri(tdata->pool, to, strlen(to),
        PJSIP_PARSE_URI_AS_NAMEADDR);
 
    if (parsed_name_addr) {
        if (pj_strlen(&parsed_name_addr->display)) {
            pjsip_name_addr *name_addr =
                (pjsip_name_addr *) PJSIP_MSG_TO_HDR(tdata->msg)->uri;
 
            pj_strdup(tdata->pool, &name_addr->display, &parsed_name_addr->display);
 
        }
        return 0;
    }
 
    return -1;
}
 
/*!
 * \internal
 * \brief Checks if the given msg var name should be blocked.
 *
 * \details Some headers are not allowed to be overriden by the user.
 *  Determine if the given var header name from the user is blocked for
 *  an outgoing MESSAGE.
 *
 * \param name name of header to see if it is blocked.
 *
 * \retval TRUE if the given header is blocked.
 */
static int is_msg_var_blocked(const char *name)
{
    int i;
 
    /* Don't block the Max-Forwards header because the user can override it */
    static const char *hdr[] = {
        "To",
        "From",
        "Via",
        "Route",
        "Contact",
        "Call-ID",
        "CSeq",
        "Allow",
        "Content-Length",
        "Content-Type",
        "Request-URI",
    };
 
    for (i = 0; i < ARRAY_LEN(hdr); ++i) {
        if (!strcasecmp(name, hdr[i])) {
            /* Block addition of this header. */
            return 1;
        }
    }
    return 0;
}
 
/*!
 * \internal
 * \brief Copies any other msg vars over to the request headers.
 *
 * \param msg The msg structure to copy headers from
 * \param tdata The SIP transmission data
 */
static enum pjsip_status_code vars_to_headers(const struct ast_msg *msg, pjsip_tx_data *tdata)
{
    const char *name;
    const char *value;
    int max_forwards;
    struct ast_msg_var_iterator *iter;
 
    for (iter = ast_msg_var_iterator_init(msg);
        ast_msg_var_iterator_next(msg, iter, &name, &value);
        ast_msg_var_unref_current(iter)) {
        if (!strcasecmp(name, "Max-Forwards")) {
            /* Decrement Max-Forwards for SIP loop prevention. */
            if (sscanf(value, "%30d", &max_forwards) != 1 || --max_forwards == 0) {
                ast_msg_var_iterator_destroy(iter);
                ast_log(LOG_NOTICE, "MESSAGE(Max-Forwards) reached zero.  MESSAGE not sent.\n");
                return -1;
            }
            sprintf((char *) value, "%d", max_forwards);
            ast_sip_add_header(tdata, name, value);
        } else if (!is_msg_var_blocked(name)) {
            ast_sip_add_header(tdata, name, value);
        }
    }
    ast_msg_var_iterator_destroy(iter);
 
    return PJSIP_SC_OK;
}
 
/*!
 * \internal
 * \brief Copies any other request header data over to ast_msg structure.
 *
 * \param rdata The SIP request
 * \param msg The msg structure to copy headers into
 */
static int headers_to_vars(const pjsip_rx_data *rdata, struct ast_msg *msg)
{
    char *c;
    char name[MAX_HDR_SIZE];
    char buf[MAX_HDR_SIZE];
    int res = 0;
    pjsip_hdr *h = rdata->msg_info.msg->hdr.next;
    pjsip_hdr *end= &rdata->msg_info.msg->hdr;
 
    while (h != end) {
        if ((res = pjsip_hdr_print_on(h, buf, sizeof(buf)-1)) > 0) {
            buf[res] = '\0';
            if ((c = strchr(buf, ':'))) {
                ast_copy_string(buf, ast_skip_blanks(c + 1), sizeof(buf));
            }
 
            ast_copy_pj_str(name, &h->name, sizeof(name));
            if ((res = ast_msg_set_var(msg, name, buf)) != 0) {
                break;
            }
        }
        h = h->next;
    }
    return 0;
}
 
/*!
 * \internal
 * \brief Prints the message body into the given char buffer.
 *
 * \details Copies body content from the received data into the given
 * character buffer removing any extra carriage return/line feeds.
 *
 * \param rdata The SIP request
 * \param buf Buffer to fill
 * \param len The length of the buffer
 */
static int print_body(pjsip_rx_data *rdata, char *buf, int len)
{
    int res;
 
    if (!rdata->msg_info.msg->body || !rdata->msg_info.msg->body->len) {
        return 0;
    }
 
    if ((res = rdata->msg_info.msg->body->print_body(
             rdata->msg_info.msg->body, buf, len)) < 0) {
        return res;
    }
 
    /* remove any trailing carriage return/line feeds */
    while (res > 0 && ((buf[--res] == '\r') || (buf[res] == '\n')));
 
    buf[++res] = '\0';
 
    return res;
}
 
/*!
 * \internal
 * \brief Converts a 'sip:' uri to a 'pjsip:' so it can be found by
 * the message tech.
 *
 * \param buf uri to insert 'pjsip' into
 * \param size length of the uri in buf
 * \param capacity total size of buf
 */
static char *sip_to_pjsip(char *buf, int size, int capacity)
{
    int count;
    const char *scheme;
    char *res = buf;
 
    /* remove any wrapping brackets */
    if (*buf == '<') {
        ++buf;
        --size;
    }
 
    scheme = strncmp(buf, "sip", 3) ? "pjsip:" : "pj";
    count = strlen(scheme);
    if (count + size >= capacity) {
        ast_log(LOG_WARNING, "Unable to handle MESSAGE- incoming uri "
            "too large for given buffer\n");
        return NULL;
    }
 
    memmove(res + count, buf, size);
    memcpy(res, scheme, count);
 
    buf += size - 1;
    if (*buf == '>') {
        *buf = '\0';
    }
 
    return res;
}
 
/*!
 * \internal
 * \brief Converts a pjsip_rx_data structure to an ast_msg structure.
 *
 * \details Attempts to fill in as much information as possible into the given
 * msg structure copied from the given request data.
 *
 * \param rdata The SIP request
 * \param msg The asterisk message structure to fill in.
 */
static enum pjsip_status_code rx_data_to_ast_msg(pjsip_rx_data *rdata, struct ast_msg *msg)
{
    RAII_VAR(struct ast_sip_endpoint *, endpt, NULL, ao2_cleanup);
    pjsip_uri *ruri = rdata->msg_info.msg->line.req.uri;
    pjsip_name_addr *name_addr;
    char buf[MAX_BODY_SIZE];
    const char *field;
    const char *context;
    char exten[AST_MAX_EXTENSION];
    int res = 0;
    int size;
 
    if (!ast_sip_is_allowed_uri(ruri)) {
        return PJSIP_SC_UNSUPPORTED_URI_SCHEME;
    }
 
    ast_copy_pj_str(exten, ast_sip_pjsip_uri_get_username(ruri), AST_MAX_EXTENSION);
 
    /*
     * We may want to match in the dialplan without any user
     * options getting in the way.
     */
    AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
 
    endpt = ast_pjsip_rdata_get_endpoint(rdata);
    ast_assert(endpt != NULL);
 
    context = S_OR(endpt->message_context, endpt->context);
    res |= ast_msg_set_context(msg, "%s", context);
    res |= ast_msg_set_exten(msg, "%s", exten);
 
    /* to header */
    name_addr = (pjsip_name_addr *)rdata->msg_info.to->uri;
    size = pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, name_addr, buf, sizeof(buf) - 1);
    if (size <= 0) {
        return PJSIP_SC_INTERNAL_SERVER_ERROR;
    }
    buf[size] = '\0';
    res |= ast_msg_set_to(msg, "%s", sip_to_pjsip(buf, ++size, sizeof(buf) - 1));
 
    /* from header */
    name_addr = (pjsip_name_addr *)rdata->msg_info.from->uri;
    size = pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, name_addr, buf, sizeof(buf) - 1);
    if (size <= 0) {
        return PJSIP_SC_INTERNAL_SERVER_ERROR;
    }
    buf[size] = '\0';
    res |= ast_msg_set_from(msg, "%s", buf);
 
    field = pj_sockaddr_print(&rdata->pkt_info.src_addr, buf, sizeof(buf) - 1, 3);
    res |= ast_msg_set_var(msg, "PJSIP_RECVADDR", field);
 
    switch (rdata->tp_info.transport->key.type) {
    case PJSIP_TRANSPORT_UDP:
    case PJSIP_TRANSPORT_UDP6:
        field = "udp";
        break;
    case PJSIP_TRANSPORT_TCP:
    case PJSIP_TRANSPORT_TCP6:
        field = "tcp";
        break;
    case PJSIP_TRANSPORT_TLS:
    case PJSIP_TRANSPORT_TLS6:
        field = "tls";
        break;
    default:
        field = rdata->tp_info.transport->type_name;
    }
    ast_msg_set_var(msg, "PJSIP_TRANSPORT", field);
 
    if (print_body(rdata, buf, sizeof(buf) - 1) > 0) {
        res |= ast_msg_set_body(msg, "%s", buf);
    }
 
    /* endpoint name */
    res |= ast_msg_set_tech(msg, "%s", "PJSIP");
    res |= ast_msg_set_endpoint(msg, "%s", ast_sorcery_object_get_id(endpt));
    if (endpt->id.self.name.valid) {
        res |= ast_msg_set_var(msg, "PJSIP_ENDPOINT", endpt->id.self.name.str);
    }
 
    res |= headers_to_vars(rdata, msg);
 
    return !res ? PJSIP_SC_OK : PJSIP_SC_INTERNAL_SERVER_ERROR;
}
 
struct msg_data {
    struct ast_msg *msg;
    char *destination;
    char *from;
};
 
static void msg_data_destroy(void *obj)
{
    struct msg_data *mdata = obj;
 
    ast_free(mdata->from);
    ast_free(mdata->destination);
 
    ast_msg_destroy(mdata->msg);
}
 
static struct msg_data *msg_data_create(const struct ast_msg *msg, const char *destination, const char *from)
{
    char *uri_params;
    struct msg_data *mdata = ao2_alloc(sizeof(*mdata), msg_data_destroy);
 
    if (!mdata) {
        return NULL;
    }
 
    /* typecast to suppress const warning */
    mdata->msg = ast_msg_ref((struct ast_msg *) msg);
 
    /* To starts with 'pjsip:' which needs to be removed. */
    if (!(destination = strchr(destination, ':'))) {
        ao2_ref(mdata, -1);
        return NULL;
    }
    ++destination;/* Now skip the ':' */
 
    mdata->destination = ast_strdup(destination);
    mdata->from = ast_strdup(from);
 
    /*
     * Sometimes from URI can contain URI parameters, so remove them.
     *
     * sip:user;user-options@domain;uri-parameters
     */
    uri_params = strchr(mdata->from, '@');
    if (uri_params && (uri_params = strchr(mdata->from, ';'))) {
        *uri_params = '\0';
    }
    return mdata;
}
 
static void update_content_type(pjsip_tx_data *tdata, struct ast_msg *msg, struct ast_sip_body *body)
{
    static const pj_str_t CONTENT_TYPE = { "Content-Type", sizeof("Content-Type") - 1 };
 
    const char *content_type = ast_msg_get_var(msg, pj_strbuf(&CONTENT_TYPE));
    if (content_type) {
        pj_str_t type, subtype;
        pjsip_ctype_hdr *parsed;
 
        /* Let pjsip do the parsing for us */
        parsed = pjsip_parse_hdr(tdata->pool, &CONTENT_TYPE,
            ast_strdupa(content_type), strlen(content_type),
            NULL);
 
        if (!parsed) {
            ast_log(LOG_WARNING, "Failed to parse '%s' as a content type. Using text/plain\n",
                content_type);
            return;
        }
 
        /* We need to turn type and subtype into zero-terminated strings */
        pj_strdup_with_null(tdata->pool, &type, &parsed->media.type);
        pj_strdup_with_null(tdata->pool, &subtype, &parsed->media.subtype);
 
        body->type = pj_strbuf(&type);
        body->subtype = pj_strbuf(&subtype);
    }
}
 
/*!
 * \internal
 * \brief Send a MESSAGE
 *
 * \param data The outbound message data structure
 *
 * \return 0: success, -1: failure
 *
 * mdata contains the To and From specified in the call to the MessageSend
 * dialplan app.  It also contains the ast_msg object that contains the
 * message body and may contain the To and From from the channel datastore,
 * usually set with the MESSAGE or MESSAGE_DATA dialplan functions but
 * could also come from an incoming sip MESSAGE.
 *
 * The mdata->to is always used as the basis for the Request URI
 * while the mdata->msg->to is used for the To header.  If
 * mdata->msg->to isn't available, mdata->to is used for the To header.
 *
 */
static int msg_send(void *data)
{
    struct msg_data *mdata = data; /* The caller holds a reference */
 
    struct ast_sip_body body = {
        .type = "text",
        .subtype = "plain",
        .body_text = ast_msg_get_body(mdata->msg)
    };
 
    pjsip_tx_data *tdata;
    RAII_VAR(char *, uri, NULL, ast_free);
    RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
 
    ast_debug(3, "mdata From: %s msg From: %s mdata Destination: %s msg To: %s\n",
        mdata->from, ast_msg_get_from(mdata->msg), mdata->destination, ast_msg_get_to(mdata->msg));
 
    endpoint = ast_sip_get_endpoint(mdata->destination, 1, &uri);
    if (!endpoint) {
        ast_log(LOG_ERROR,
            "PJSIP MESSAGE - Could not find endpoint '%s' and no default outbound endpoint configured\n",
            mdata->destination);
 
        ast_test_suite_event_notify("MSG_ENDPOINT_URI_FAIL",
            "MdataFrom: %s\r\n"
            "MsgFrom: %s\r\n"
            "MdataDestination: %s\r\n"
            "MsgTo: %s\r\n",
            mdata->from,
            ast_msg_get_from(mdata->msg),
            mdata->destination,
            ast_msg_get_to(mdata->msg));
 
        return -1;
    }
 
    ast_debug(3, "Request URI: %s\n", uri);
 
    if (ast_sip_create_request("MESSAGE", NULL, endpoint, uri, NULL, &tdata)) {
        ast_log(LOG_WARNING, "PJSIP MESSAGE - Could not create request\n");
        return -1;
    }
 
    /* If there was a To in the actual message, */
    if (!ast_strlen_zero(ast_msg_get_to(mdata->msg))) {
        char *msg_to = ast_strdupa(ast_msg_get_to(mdata->msg));
 
        /*
         * It's possible that the message To was copied from
         * an incoming MESSAGE in which case it'll have the
         * pjsip: tech prepended to it.  We need to remove it.
         */
        if (ast_begins_with(msg_to, "pjsip:")) {
            msg_to += 6;
        }
        ast_sip_update_to_uri(tdata, msg_to);
    } else {
        /*
         * If there was no To in the message, it's still possible
         * that there is a display name in the mdata To.  If so,
         * we'll copy the URI display name to the tdata To.
         */
        update_to_display_name(tdata, uri);
    }
 
    if (!ast_strlen_zero(mdata->from)) {
        ast_sip_update_from(tdata, mdata->from);
    } else if (!ast_strlen_zero(ast_msg_get_from(mdata->msg))) {
        ast_sip_update_from(tdata, (char *)ast_msg_get_from(mdata->msg));
    }
 
#ifdef TEST_FRAMEWORK
    {
        pjsip_name_addr *tdata_name_addr;
        pjsip_sip_uri *tdata_sip_uri;
        char touri[128];
        char fromuri[128];
 
        tdata_name_addr = (pjsip_name_addr *) PJSIP_MSG_TO_HDR(tdata->msg)->uri;
        tdata_sip_uri = pjsip_uri_get_uri(tdata_name_addr->uri);
        pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, tdata_sip_uri, touri, sizeof(touri));
        tdata_name_addr = (pjsip_name_addr *) PJSIP_MSG_FROM_HDR(tdata->msg)->uri;
        tdata_sip_uri = pjsip_uri_get_uri(tdata_name_addr->uri);
        pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, tdata_sip_uri, fromuri, sizeof(fromuri));
 
        ast_test_suite_event_notify("MSG_FROMTO_URI",
            "MdataFrom: %s\r\n"
            "MsgFrom: %s\r\n"
            "MdataDestination: %s\r\n"
            "MsgTo: %s\r\n"
            "Endpoint: %s\r\n"
            "RequestURI: %s\r\n"
            "ToURI: %s\r\n"
            "FromURI: %s\r\n",
            mdata->from,
            ast_msg_get_from(mdata->msg),
            mdata->destination,
            ast_msg_get_to(mdata->msg),
            ast_sorcery_object_get_id(endpoint),
            uri,
            touri,
            fromuri
            );
    }
#endif
 
    update_content_type(tdata, mdata->msg, &body);
 
    if (ast_sip_add_body(tdata, &body)) {
        pjsip_tx_data_dec_ref(tdata);
        ast_log(LOG_ERROR, "PJSIP MESSAGE - Could not add body to request\n");
        return -1;
    }
 
    /*
     * This copies any headers set with MESSAGE_DATA() to the
     * tdata.
     */
    vars_to_headers(mdata->msg, tdata);
 
    ast_debug(1, "Sending message to '%s' (via endpoint %s) from '%s'\n",
        uri, ast_sorcery_object_get_id(endpoint), mdata->from);
 
    if (ast_sip_send_request(tdata, NULL, endpoint, NULL, NULL)) {
        ast_log(LOG_ERROR, "PJSIP MESSAGE - Could not send request\n");
        return -1;
    }
 
    return 0;
}
 
static int sip_msg_send(const struct ast_msg *msg, const char *destination, const char *from)
{
    struct msg_data *mdata;
    int res;
 
    if (ast_strlen_zero(destination)) {
        ast_log(LOG_ERROR, "SIP MESSAGE - a 'To' URI  must be specified\n");
        return -1;
    }
 
    mdata = msg_data_create(msg, destination, from);
    if (!mdata) {
        return -1;
    }
 
    res = ast_sip_push_task_wait_serializer(message_serializer, msg_send, mdata);
    ao2_ref(mdata, -1);
 
    return res;
}
 
static const struct ast_msg_tech msg_tech = {
    .name = "pjsip",
    .msg_send = sip_msg_send,
};
 
static pj_status_t send_response(pjsip_rx_data *rdata, enum pjsip_status_code code,
                 pjsip_dialog *dlg, pjsip_transaction *tsx)
{
    pjsip_tx_data *tdata;
    pj_status_t status;
 
    status = ast_sip_create_response(rdata, code, NULL, &tdata);
    if (status != PJ_SUCCESS) {
        ast_log(LOG_ERROR, "Unable to create response (%d)\n", status);
        return status;
    }
 
    if (dlg && tsx) {
        status = pjsip_dlg_send_response(dlg, tsx, tdata);
    } else {
        struct ast_sip_endpoint *endpoint;
 
        endpoint = ast_pjsip_rdata_get_endpoint(rdata);
        status = ast_sip_send_stateful_response(rdata, tdata, endpoint);
        ao2_cleanup(endpoint);
    }
 
    if (status != PJ_SUCCESS) {
        ast_log(LOG_ERROR, "Unable to send response (%d)\n", status);
    }
 
    return status;
}
 
static pj_bool_t module_on_rx_request(pjsip_rx_data *rdata)
{
    enum pjsip_status_code code;
    struct ast_msg *msg;
 
    /* if not a MESSAGE, don't handle */
    if (pjsip_method_cmp(&rdata->msg_info.msg->line.req.method, &pjsip_message_method)) {
        return PJ_FALSE;
    }
 
    code = check_content_type(rdata);
    if (code != PJSIP_SC_OK) {
        send_response(rdata, code, NULL, NULL);
        return PJ_TRUE;
    }
 
    msg = ast_msg_alloc();
    if (!msg) {
        send_response(rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL);
        return PJ_TRUE;
    }
 
    code = rx_data_to_ast_msg(rdata, msg);
    if (code != PJSIP_SC_OK) {
        send_response(rdata, code, NULL, NULL);
        ast_msg_destroy(msg);
        return PJ_TRUE;
    }
 
    if (!ast_msg_has_destination(msg)) {
        ast_debug(1, "MESSAGE request received, but no handler wanted it\n");
        send_response(rdata, PJSIP_SC_NOT_FOUND, NULL, NULL);
        ast_msg_destroy(msg);
        return PJ_TRUE;
    }
 
    /* Send it to the messaging core.
     *
     * If we are unable to send a response, the most likely reason is that we
     * are handling a retransmission of an incoming MESSAGE and were unable to
     * create a transaction due to a duplicate key. If we are unable to send
     * a response, we should not queue the message to the dialplan
     */
    if (!send_response(rdata, PJSIP_SC_ACCEPTED, NULL, NULL)) {
        ast_msg_queue(msg);
    }
 
    return PJ_TRUE;
}
 
static int incoming_in_dialog_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
    enum pjsip_status_code code;
    int rc;
    pjsip_dialog *dlg = session->inv_session->dlg;
    pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
    struct ast_msg_data *msg;
    struct ast_party_caller *caller;
    pjsip_name_addr *name_addr;
    size_t from_len;
    size_t to_len;
    struct ast_msg_data_attribute attrs[4];
    int pos = 0;
    int body_pos;
 
    if (!session->channel) {
        send_response(rdata, PJSIP_SC_NOT_FOUND, dlg, tsx);
        return 0;
    }
 
    code = check_content_type_in_dialog(rdata);
    if (code != PJSIP_SC_OK) {
        send_response(rdata, code, dlg, tsx);
        return 0;
    }
 
    caller = ast_channel_caller(session->channel);
 
    name_addr = (pjsip_name_addr *) rdata->msg_info.from->uri;
    from_len = pj_strlen(&name_addr->display);
    if (from_len) {
        attrs[pos].type = AST_MSG_DATA_ATTR_FROM;
        from_len++;
        attrs[pos].value = ast_alloca(from_len);
        ast_copy_pj_str(attrs[pos].value, &name_addr->display, from_len);
        pos++;
    } else if (caller->id.name.valid && !ast_strlen_zero(caller->id.name.str)) {
        attrs[pos].type = AST_MSG_DATA_ATTR_FROM;
        attrs[pos].value = caller->id.name.str;
        pos++;
    }
 
    name_addr = (pjsip_name_addr *) rdata->msg_info.to->uri;
    to_len = pj_strlen(&name_addr->display);
    if (to_len) {
        attrs[pos].type = AST_MSG_DATA_ATTR_TO;
        to_len++;
        attrs[pos].value = ast_alloca(to_len);
        ast_copy_pj_str(attrs[pos].value, &name_addr->display, to_len);
        pos++;
    }
 
    attrs[pos].type = AST_MSG_DATA_ATTR_CONTENT_TYPE;
    attrs[pos].value = ast_alloca(rdata->msg_info.msg->body->content_type.type.slen
        + rdata->msg_info.msg->body->content_type.subtype.slen + 2);
    sprintf(attrs[pos].value, "%.*s/%.*s",
        (int)rdata->msg_info.msg->body->content_type.type.slen,
        rdata->msg_info.msg->body->content_type.type.ptr,
        (int)rdata->msg_info.msg->body->content_type.subtype.slen,
        rdata->msg_info.msg->body->content_type.subtype.ptr);
    pos++;
 
    body_pos = pos;
    attrs[pos].type = AST_MSG_DATA_ATTR_BODY;
    attrs[pos].value = ast_malloc(rdata->msg_info.msg->body->len + 1);
    if (!attrs[pos].value) {
        send_response(rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, dlg, tsx);
        return 0;
    }
    ast_copy_string(attrs[pos].value, rdata->msg_info.msg->body->data, rdata->msg_info.msg->body->len + 1);
    pos++;
 
    msg = ast_msg_data_alloc(AST_MSG_DATA_SOURCE_TYPE_IN_DIALOG, attrs, pos);
    if (!msg) {
        ast_free(attrs[body_pos].value);
        send_response(rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, dlg, tsx);
        return 0;
    }
 
    ast_debug(1, "Received in-dialog MESSAGE from '%s:%s': %s %s\n",
        ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_FROM),
        ast_channel_name(session->channel),
        ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_TO),
        ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_BODY));
 
    rc = ast_msg_data_queue_frame(session->channel, msg);
    ast_free(attrs[body_pos].value);
    ast_free(msg);
    if (rc != 0) {
        send_response(rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, dlg, tsx);
    } else {
        send_response(rdata, PJSIP_SC_ACCEPTED, dlg, tsx);
    }
 
    return 0;
}
 
static struct ast_sip_session_supplement messaging_supplement = {
    .method = "MESSAGE",
    .incoming_request = incoming_in_dialog_request
};
 
static pjsip_module messaging_module = {
    .name = {"Messaging Module", 16},
    .id = -1,
    .priority = PJSIP_MOD_PRIORITY_APPLICATION,
    .on_rx_request = module_on_rx_request,
};
 
static int load_module(void)
{
    if (ast_sip_register_service(&messaging_module) != PJ_SUCCESS) {
        return AST_MODULE_LOAD_DECLINE;
    }
 
    if (pjsip_endpt_add_capability(ast_sip_get_pjsip_endpoint(),
                       NULL, PJSIP_H_ALLOW, NULL, 1,
                       &pjsip_message_method.name) != PJ_SUCCESS) {
 
        ast_sip_unregister_service(&messaging_module);
        return AST_MODULE_LOAD_DECLINE;
    }
 
    if (ast_msg_tech_register(&msg_tech)) {
        ast_sip_unregister_service(&messaging_module);
        return AST_MODULE_LOAD_DECLINE;
    }
 
    message_serializer = ast_sip_create_serializer("pjsip/messaging");
    if (!message_serializer) {
        ast_sip_unregister_service(&messaging_module);
        ast_msg_tech_unregister(&msg_tech);
        return AST_MODULE_LOAD_DECLINE;
    }
 
    ast_sip_session_register_supplement(&messaging_supplement);
    return AST_MODULE_LOAD_SUCCESS;
}
 
static int unload_module(void)
{
    ast_sip_session_unregister_supplement(&messaging_supplement);
    ast_msg_tech_unregister(&msg_tech);
    ast_sip_unregister_service(&messaging_module);
    ast_taskprocessor_unreference(message_serializer);
    return 0;
}
 
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Messaging Support",
    .support_level = AST_MODULE_SUPPORT_CORE,
    .load = load_module,
    .unload = unload_module,
    .load_pri = AST_MODPRI_APP_DEPEND,
    .requires = "res_pjsip,res_pjsip_session",
);