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authorMark Michelson <mmichelson@digium.com>2013-07-30 18:14:50 +0000
committerMark Michelson <mmichelson@digium.com>2013-07-30 18:14:50 +0000
commit735b30ad71110c2a51404cb8686bbe3cf14b630c (patch)
tree76b1f10135c1b7f210e576be1359539de7e3476c /res/res_pjsip_messaging.c
parent895c8e0d2c97cd04299f3f179e99d8a3873c06c6 (diff)
The large GULP->PJSIP renaming effort.
The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*" * All files that were "res_sip*" are now "res_pjsip*" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip_messaging.c')
-rw-r--r--res/res_pjsip_messaging.c660
1 files changed, 660 insertions, 0 deletions
diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
new file mode 100644
index 000000000..8f5151292
--- /dev/null
+++ b/res/res_pjsip_messaging.c
@@ -0,0 +1,660 @@
+/*
+ * 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>
+ ***/
+
+#include "asterisk.h"
+
+#include "pjsua-lib/pjsua.h"
+
+#include "asterisk/message.h"
+#include "asterisk/module.h"
+#include "asterisk/pbx.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
+
+const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
+
+#define MAX_HDR_SIZE 512
+#define MAX_BODY_SIZE 1024
+#define MAX_EXTEN_SIZE 256
+#define MAX_USER_SIZE 128
+
+/*!
+ * \internal
+ * \brief Determine where in the dialplan a call should go
+ *
+ * \details This uses the username in the request URI to try to match
+ * an extension in an endpoint's context in order to route the call.
+ *
+ * \param rdata The SIP request
+ * \param context The context to use
+ * \param exten The extension to use
+ */
+static enum pjsip_status_code get_destination(const pjsip_rx_data *rdata, const char *context, char *exten)
+{
+ pjsip_uri *ruri = rdata->msg_info.msg->line.req.uri;
+ pjsip_sip_uri *sip_ruri;
+
+ if (!PJSIP_URI_SCHEME_IS_SIP(ruri) && !PJSIP_URI_SCHEME_IS_SIPS(ruri)) {
+ return PJSIP_SC_UNSUPPORTED_URI_SCHEME;
+ }
+
+ sip_ruri = pjsip_uri_get_uri(ruri);
+ ast_copy_pj_str(exten, &sip_ruri->user, MAX_EXTEN_SIZE);
+
+ if (ast_exists_extension(NULL, context, exten, 1, NULL)) {
+ return PJSIP_SC_OK;
+ }
+ return PJSIP_SC_NOT_FOUND;
+}
+
+/*!
+ * \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)
+{
+ if (ast_sip_is_content_type(&rdata->msg_info.msg->body->content_type,
+ "text",
+ "plain")) {
+ return PJSIP_SC_OK;
+ } else {
+ return PJSIP_SC_UNSUPPORTED_MEDIA_TYPE;
+ }
+}
+
+/*!
+ * \internal
+ * \brief Puts pointer past 'sip[s]:' string that should be at the
+ * front of the given 'fromto' parameter
+ *
+ * \param fromto 'From' or 'To' field containing 'sip:'
+ */
+static const char* skip_sip(const char *fromto)
+{
+ const char *p;
+
+ /* need to be one past 'sip:' or 'sips:' */
+ if (!(p = strstr(fromto, "sip"))) {
+ return fromto;
+ }
+
+ p += 3;
+ if (*p == 's') {
+ ++p;
+ }
+ return ++p;
+}
+
+/*!
+ * \internal
+ * \brief Retrieves an endpoint if specified in the given 'fromto'
+ *
+ * Expects the given 'fromto' to be in one of the following formats:
+ * sip[s]:endpoint[/aor]
+ * sip[s]:endpoint[/uri]
+ *
+ * If an optional aor is given it will try to find an associated uri
+ * to return. If an optional uri is given then that will be returned,
+ * otherwise uri will be NULL.
+ *
+ * \param fromto 'From' or 'To' field with possible endpoint
+ * \param uri Optional uri to return
+ */
+static struct ast_sip_endpoint* get_endpoint(const char *fromto, char **uri)
+{
+ const char *name = skip_sip(fromto);
+ struct ast_sip_endpoint* endpoint;
+ struct ast_sip_aor *aor;
+
+ if ((*uri = strchr(name, '/'))) {
+ *(*uri)++ = '\0';
+ }
+
+ /* endpoint is required */
+ if (ast_strlen_zero(name)) {
+ return NULL;
+ }
+
+ if (!(endpoint = ast_sorcery_retrieve_by_id(
+ ast_sip_get_sorcery(), "endpoint", name))) {
+ return NULL;
+ }
+
+ if (*uri && (aor = ast_sip_location_retrieve_aor(*uri))) {
+ *uri = (char*)ast_sip_location_retrieve_first_aor_contact(aor)->uri;
+ }
+
+ return endpoint;
+}
+
+/*!
+ * \internal
+ * \brief Updates fields in an outgoing 'From' header.
+ *
+ * \param tdata The outgoing message data structure
+ * \param from Info to potentially copy into the 'From' header
+ */
+static void update_from(pjsip_tx_data *tdata, const char *from)
+{
+ /* static const pj_str_t hname = { "From", 4 }; */
+ pjsip_name_addr *from_name_addr;
+ pjsip_sip_uri *from_uri;
+ pjsip_uri *parsed;
+ char *uri;
+
+ RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
+
+ if (ast_strlen_zero(from)) {
+ return;
+ }
+
+ if (!(endpoint = get_endpoint(from, &uri))) {
+ return;
+ }
+
+ if (ast_strlen_zero(uri)) {
+ /* if no aor/uri was specified get one from the endpoint */
+ uri = (char*)ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors)->uri;
+ }
+
+ /* get current 'from' hdr & uri - going to overwrite some fields */
+ from_name_addr = (pjsip_name_addr *)PJSIP_MSG_FROM_HDR(tdata->msg)->uri;
+ from_uri = pjsip_uri_get_uri(from_name_addr);
+
+ /* check to see if uri is in 'name <sip:user@domain>' format */
+ if ((parsed = pjsip_parse_uri(tdata->pool, uri, strlen(uri), PJSIP_PARSE_URI_AS_NAMEADDR))) {
+ pjsip_name_addr *name_addr = (pjsip_name_addr *)parsed;
+ pjsip_sip_uri *sip_uri = pjsip_uri_get_uri(name_addr->uri);
+
+ pj_strdup(tdata->pool, &from_name_addr->display, &name_addr->display);
+ pj_strdup(tdata->pool, &from_uri->user, &sip_uri->user);
+ pj_strdup(tdata->pool, &from_uri->host, &sip_uri->host);
+ from_uri->port = sip_uri->port;
+ } else {
+ /* assume it is 'user[@domain]' format */
+ char *domain = strchr(uri, '@');
+ if (domain) {
+ *domain++ = '\0';
+ pj_strdup2(tdata->pool, &from_uri->host, domain);
+ }
+ pj_strdup2(tdata->pool, &from_uri->user, uri);
+ }
+}
+
+/*!
+ * \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 Content-Type or Max-Forwards headers because the
+ * user can override them.
+ */
+ static const char *hdr[] = {
+ "To",
+ "From",
+ "Via",
+ "Route",
+ "Contact",
+ "Call-ID",
+ "CSeq",
+ "Allow",
+ "Content-Length",
+ "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;
+
+ RAII_VAR(struct ast_msg_var_iterator *, i, ast_msg_var_iterator_init(msg), ast_msg_var_iterator_destroy);
+ while (ast_msg_var_iterator_next(msg, i, &name, &value)) {
+ if (!strcasecmp(name, "Max-Forwards")) {
+ /* Decrement Max-Forwards for SIP loop prevention. */
+ if (sscanf(value, "%30d", &max_forwards) != 1 || --max_forwards == 0) {
+ 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_unref_current(i);
+ }
+ 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 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)-(c-buf));
+ }
+
+ if ((res = ast_msg_set_var(msg, pj_strbuf(&h->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 = rdata->msg_info.msg->body->print_body(
+ rdata->msg_info.msg->body, buf, len);
+
+ if (res < 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 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)
+{
+
+#define CHECK_RES(z_) do { if (z_) { ast_msg_destroy(msg); \
+ return PJSIP_SC_INTERNAL_SERVER_ERROR; } } while (0)
+
+ int size;
+ char buf[MAX_BODY_SIZE];
+ pjsip_name_addr *name_addr;
+ const char *field;
+ pjsip_status_code code;
+ struct ast_sip_endpoint *endpt = ast_pjsip_rdata_get_endpoint(rdata);
+
+ /* make sure there is an appropriate context and extension*/
+ if ((code = get_destination(rdata, endpt->context, buf)) != PJSIP_SC_OK) {
+ return code;
+ }
+
+ CHECK_RES(ast_msg_set_context(msg, "%s", endpt->context));
+ CHECK_RES(ast_msg_set_exten(msg, "%s", buf));
+
+ /* to header */
+ name_addr = (pjsip_name_addr *)rdata->msg_info.to->uri;
+ if ((size = pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, name_addr, buf, sizeof(buf)-1)) > 0) {
+ buf[size] = '\0';
+ CHECK_RES(ast_msg_set_to(msg, "%s", buf));
+ }
+
+ /* from header */
+ name_addr = (pjsip_name_addr *)rdata->msg_info.from->uri;
+ if ((size = pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, name_addr, buf, sizeof(buf)-1)) > 0) {
+ buf[size] = '\0';
+ CHECK_RES(ast_msg_set_from(msg, "%s", buf));
+ }
+
+ /* contact header */
+ if ((size = pjsip_hdr_print_on(pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL), buf, sizeof(buf)-1)) > 0) {
+ buf[size] = '\0';
+ CHECK_RES(ast_msg_set_var(msg, "SIP_FULLCONTACT", buf));
+ }
+
+ /* receive address */
+ field = pj_sockaddr_print(&rdata->pkt_info.src_addr, buf, sizeof(buf)-1, 1);
+ CHECK_RES(ast_msg_set_var(msg, "SIP_RECVADDR", field));
+
+ /* body */
+ if (print_body(rdata, buf, sizeof(buf) - 1) > 0) {
+ CHECK_RES(ast_msg_set_body(msg, "%s", buf));
+ }
+
+ /* endpoint name */
+ if (endpt->id.self.name.valid) {
+ CHECK_RES(ast_msg_set_var(msg, "SIP_PEERNAME", endpt->id.self.name.str));
+ }
+
+ CHECK_RES(headers_to_vars(rdata, msg));
+
+ return PJSIP_SC_OK;
+}
+
+struct msg_data {
+ struct ast_msg *msg;
+ char *to;
+ char *from;
+};
+
+static void msg_data_destroy(void *obj)
+{
+ struct msg_data *mdata = obj;
+
+ ast_free(mdata->from);
+ ast_free(mdata->to);
+
+ ast_msg_destroy(mdata->msg);
+}
+
+static struct msg_data* msg_data_create(const struct ast_msg *msg, const char *to, const char *from)
+{
+ char *tag;
+ 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);
+
+ mdata->to = ast_strdup(to);
+ mdata->from = ast_strdup(from);
+
+ /* sometimes from can still contain the tag at this point, so remove it */
+ if ((tag = strchr(mdata->from, ';'))) {
+ *tag = '\0';
+ }
+
+ return mdata;
+}
+
+static int msg_send(void *data)
+{
+ RAII_VAR(struct msg_data *, mdata, data, ao2_cleanup);
+
+ const struct ast_sip_body body = {
+ .type = "text",
+ .subtype = "plain",
+ .body_text = ast_msg_get_body(mdata->msg)
+ };
+
+ pjsip_tx_data *tdata;
+ char *uri;
+
+ RAII_VAR(struct ast_sip_endpoint *, endpoint, get_endpoint(
+ mdata->to, &uri), ao2_cleanup);
+ if (!endpoint) {
+ ast_log(LOG_ERROR, "SIP MESSAGE - Endpoint not found in %s\n", mdata->to);
+ return -1;
+ }
+
+ if (ast_sip_create_request("MESSAGE", NULL, endpoint, uri, &tdata)) {
+ ast_log(LOG_ERROR, "SIP MESSAGE - Could not create request\n");
+ return -1;
+ }
+
+ if (ast_sip_add_body(tdata, &body)) {
+ pjsip_tx_data_dec_ref(tdata);
+ ast_log(LOG_ERROR, "SIP MESSAGE - Could not add body to request\n");
+ return -1;
+ }
+
+ update_from(tdata, mdata->from);
+ vars_to_headers(mdata->msg, tdata);
+ if (ast_sip_send_request(tdata, NULL, endpoint)) {
+ pjsip_tx_data_dec_ref(tdata);
+ ast_log(LOG_ERROR, "SIP MESSAGE - Could not send request\n");
+ return -1;
+ }
+
+ return PJ_SUCCESS;
+}
+
+static int sip_msg_send(const struct ast_msg *msg, const char *to, const char *from)
+{
+ struct msg_data *mdata;
+
+ if (ast_strlen_zero(to)) {
+ ast_log(LOG_ERROR, "SIP MESSAGE - a 'To' URI must be specified\n");
+ return -1;
+ }
+
+ if (!(mdata = msg_data_create(msg, to, from)) ||
+ ast_sip_push_task(NULL, msg_send, mdata)) {
+ ao2_ref(mdata, -1);
+ return -1;
+ }
+ return 0;
+}
+
+static const struct ast_msg_tech msg_tech = {
+ .name = "sip",
+ .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;
+ pjsip_response_addr res_addr;
+
+ pjsip_endpoint *endpt = ast_sip_get_pjsip_endpoint();
+
+ status = pjsip_endpt_create_response(endpt, 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 {
+ /* Get where to send request. */
+ status = pjsip_get_response_addr(tdata->pool, rdata, &res_addr);
+ if (status != PJ_SUCCESS) {
+ ast_log(LOG_ERROR, "Unable to get response address (%d)\n", status);
+ return status;
+ }
+ status = pjsip_endpt_send_response(endpt, &res_addr, tdata, NULL, NULL);
+ }
+
+ 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;
+ }
+
+ msg = ast_msg_alloc();
+ if (!msg) {
+ send_response(rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL);
+ return PJ_TRUE;
+ }
+
+ if ((code = check_content_type(rdata)) != PJSIP_SC_OK) {
+ send_response(rdata, code, NULL, NULL);
+ return PJ_TRUE;
+ }
+
+ if ((code = rx_data_to_ast_msg(rdata, msg)) == PJSIP_SC_OK) {
+ /* send it to the dialplan */
+ ast_msg_queue(msg);
+ code = PJSIP_SC_ACCEPTED;
+ }
+
+ send_response(rdata, code, NULL, NULL);
+ return PJ_TRUE;
+}
+
+static int incoming_in_dialog_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
+ char buf[MAX_BODY_SIZE];
+ enum pjsip_status_code code;
+ struct ast_frame f;
+
+ pjsip_dialog *dlg = session->inv_session->dlg;
+ pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
+
+ if ((code = check_content_type(rdata)) != PJSIP_SC_OK) {
+ send_response(rdata, code, dlg, tsx);
+ return 0;
+ }
+
+ if (print_body(rdata, buf, sizeof(buf)-1) < 1) {
+ /* invalid body size */
+ return 0;
+ }
+
+ memset(&f, 0, sizeof(f));
+ f.frametype = AST_FRAME_TEXT;
+ f.subclass.integer = 0;
+ f.offset = 0;
+ f.data.ptr = buf;
+ f.datalen = strlen(buf) + 1;
+ ast_queue_frame(session->channel, &f);
+
+ 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;
+ }
+
+ 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);
+ return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Messaging Support",
+ .load = load_module,
+ .unload = unload_module,
+ .load_pri = AST_MODPRI_APP_DEPEND,
+ );