res_pjsip_send_to_voicemail.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2013, Digium, Inc.
 *
 * Jonathan Rose <[email protected]>
 *
 * 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 <[email protected]>
 */

/*** 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"
#define SEND_TO_VM_REDIRECT_QUOTED_VALUE "\"" SEND_TO_VM_REDIRECT_VALUE "\""

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);

 if (!hdr) {
 return 0;
 }
 reason = get_diversion_reason(hdr);
 return reason
 && (!pj_stricmp2(&reason->value, SEND_TO_VM_REDIRECT_QUOTED_VALUE)
 || !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;
 int has_reason;

 if (!session->channel) {
 return 0;
 }

 has_feature = has_call_feature(rdata);
 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)) {
 ao2_ref(sip_session_datastore, -1);
 send_response(session, 500, rdata);
 return -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);
 }

 ao2_ref(sip_session_datastore, -1);
 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)
{
 ast_sip_session_register_supplement(&refer_supplement);

 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",
 .support_level = AST_MODULE_SUPPORT_CORE,
 .load = load_module,
 .unload = unload_module,
 .load_pri = AST_MODPRI_APP_DEPEND,
 .requires = "res_pjsip,res_pjsip_session",
);