--- /dev/null
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * Jonathan Rose <jrose@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Module for managing send to voicemail requests in SIP
+ * REFER messages against PJSIP channels
+ *
+ * \author Jonathan Rose <jrose@digium.com>
+ */
+
+/*** MODULEINFO
+ <depend>pjproject</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
+ <support_level>core</support_level>
+***/
+
+#include "asterisk.h"
+
+#include <pjsip.h>
+#include <pjsip_ua.h>
+
+#include "asterisk/pbx.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
+#include "asterisk/module.h"
+
+#define DATASTORE_NAME "call_feature_send_to_vm_datastore"
+
+#define SEND_TO_VM_HEADER "PJSIP_HEADER(add,X-Digium-Call-Feature)"
+#define SEND_TO_VM_HEADER_VALUE "feature_send_to_vm"
+
+#define SEND_TO_VM_REDIRECT "REDIRECTING(reason)"
+#define SEND_TO_VM_REDIRECT_VALUE "\"send_to_vm\""
+
+static void send_response(struct ast_sip_session *session, int code, struct pjsip_rx_data *rdata)
+{
+ pjsip_tx_data *tdata;
+
+ if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, code, NULL, &tdata) == PJ_SUCCESS) {
+ struct pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
+
+ pjsip_dlg_send_response(session->inv_session->dlg, tsx, tdata);
+ }
+}
+
+static void channel_cleanup_wrapper(void *data)
+{
+ struct ast_channel *chan = data;
+ ast_channel_cleanup(chan);
+}
+
+static struct ast_datastore_info call_feature_info = {
+ .type = "REFER call feature info",
+ .destroy = channel_cleanup_wrapper,
+};
+
+static pjsip_param *get_diversion_reason(pjsip_fromto_hdr *hdr)
+{
+ static const pj_str_t reason_str = { "reason", 6 };
+ return pjsip_param_find(&hdr->other_param, &reason_str);
+}
+
+static pjsip_fromto_hdr *get_diversion_header(pjsip_rx_data *rdata)
+{
+ static const pj_str_t from_str = { "From", 4 };
+ static const pj_str_t diversion_str = { "Diversion", 9 };
+
+ pjsip_generic_string_hdr *hdr;
+ pj_str_t value;
+
+ if (!(hdr = pjsip_msg_find_hdr_by_name(
+ rdata->msg_info.msg, &diversion_str, NULL))) {
+ return NULL;
+ }
+
+ pj_strdup_with_null(rdata->tp_info.pool, &value, &hdr->hvalue);
+
+ /* parse as a fromto header */
+ return pjsip_parse_hdr(rdata->tp_info.pool, &from_str, value.ptr,
+ pj_strlen(&value), NULL);
+}
+
+static int has_diversion_reason(pjsip_rx_data *rdata)
+{
+ pjsip_param *reason;
+ pjsip_fromto_hdr *hdr = get_diversion_header(rdata);
+
+ return hdr &&
+ (reason = get_diversion_reason(hdr)) &&
+ !pj_stricmp2(&reason->value, SEND_TO_VM_REDIRECT_VALUE);
+}
+
+static int has_call_feature(pjsip_rx_data *rdata)
+{
+ static const pj_str_t call_feature_str = { "X-Digium-Call-Feature", 21 };
+
+ pjsip_generic_string_hdr *hdr = pjsip_msg_find_hdr_by_name(
+ rdata->msg_info.msg, &call_feature_str, NULL);
+
+ return hdr && !pj_stricmp2(&hdr->hvalue, SEND_TO_VM_HEADER_VALUE);
+}
+
+static int handle_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
+
+ struct ast_datastore *sip_session_datastore;
+ struct ast_channel *other_party;
+
+ int has_feature = has_call_feature(rdata);
+ int has_reason = has_diversion_reason(rdata);
+
+ if (!has_feature && !has_reason) {
+ /* If we don't have a call feature or diversion reason or if
+ it's not a feature this module is related to then there
+ is nothing to do. */
+ return 0;
+ }
+
+ /* Check bridge status... */
+ other_party = ast_channel_bridge_peer(session->channel);
+ if (!other_party) {
+ /* The channel wasn't in a two party bridge */
+ ast_log(LOG_WARNING, "%s (%s) attempted to transfer to voicemail, "
+ "but was not in a two party bridge.\n",
+ ast_sorcery_object_get_id(session->endpoint),
+ ast_channel_name(session->channel));
+ send_response(session, 400, rdata);
+ return -1;
+ }
+
+ sip_session_datastore = ast_sip_session_alloc_datastore(
+ &call_feature_info, DATASTORE_NAME);
+ if (!sip_session_datastore) {
+ ast_channel_unref(other_party);
+ send_response(session, 500, rdata);
+ return -1;
+ }
+
+ sip_session_datastore->data = other_party;
+
+ if (ast_sip_session_add_datastore(session, sip_session_datastore)) {
+ ast_channel_unref(other_party);
+ ao2_ref(sip_session_datastore, -1);
+ send_response(session, 500, rdata);
+ return -1;
+ }
+ ao2_ref(sip_session_datastore, -1);
+
+ if (has_feature) {
+ pbx_builtin_setvar_helper(other_party, SEND_TO_VM_HEADER,
+ SEND_TO_VM_HEADER_VALUE);
+ }
+
+ if (has_reason) {
+ pbx_builtin_setvar_helper(other_party, SEND_TO_VM_REDIRECT,
+ SEND_TO_VM_REDIRECT_VALUE);
+ }
+
+ return 0;
+}
+
+static void handle_outgoing_response(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
+{
+ pjsip_status_line status = tdata->msg->line.status;
+ struct ast_datastore *feature_datastore =
+ ast_sip_session_get_datastore(session, DATASTORE_NAME);
+ struct ast_channel *target_chan;
+
+ if (!feature_datastore) {
+ return;
+ }
+
+ /* Since we are handling the response, there is no need to keep the datastore in the session anymore. */
+ ast_sip_session_remove_datastore(session, DATASTORE_NAME);
+
+ /* If the response >= 300, the refer failed and we need to clear the feature. */
+ if (status.code >= 300) {
+ target_chan = feature_datastore->data;
+ pbx_builtin_setvar_helper(target_chan, SEND_TO_VM_HEADER, NULL);
+ pbx_builtin_setvar_helper(target_chan, SEND_TO_VM_REDIRECT, NULL);
+ }
+ ao2_ref(feature_datastore, -1);
+}
+
+static struct ast_sip_session_supplement refer_supplement = {
+ .method = "REFER",
+ .incoming_request = handle_incoming_request,
+ .outgoing_response = handle_outgoing_response,
+};
+
+static int load_module(void)
+{
+ if (ast_sip_session_register_supplement(&refer_supplement)) {
+ ast_log(LOG_ERROR, "Unable to register Send to Voicemail supplement\n");
+ return AST_MODULE_LOAD_FAILURE;
+ }
+
+ return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+ ast_sip_session_unregister_supplement(&refer_supplement);
+ return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP REFER Send to Voicemail Support",
+ .load = load_module,
+ .unload = unload_module,
+ .load_pri = AST_MODPRI_APP_DEPEND,
+ );
projects / thirdparty / asterisk.git / commitdiff
