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",
);