ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
230 lines
6.3 KiB
C
230 lines
6.3 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2013, Digium, Inc.
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*
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* Jonathan Rose <jrose@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Module for managing send to voicemail requests in SIP
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* REFER messages against PJSIP channels
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*
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* \author Jonathan Rose <jrose@digium.com>
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*/
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/*** MODULEINFO
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<depend>pjproject</depend>
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<depend>res_pjsip</depend>
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<depend>res_pjsip_session</depend>
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include <pjsip.h>
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#include <pjsip_ua.h>
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#include "asterisk/pbx.h"
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#include "asterisk/res_pjsip.h"
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#include "asterisk/res_pjsip_session.h"
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#include "asterisk/module.h"
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#define DATASTORE_NAME "call_feature_send_to_vm_datastore"
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#define SEND_TO_VM_HEADER "PJSIP_HEADER(add,X-Digium-Call-Feature)"
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#define SEND_TO_VM_HEADER_VALUE "feature_send_to_vm"
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#define SEND_TO_VM_REDIRECT "REDIRECTING(reason)"
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#define SEND_TO_VM_REDIRECT_VALUE "\"send_to_vm\""
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static void send_response(struct ast_sip_session *session, int code, struct pjsip_rx_data *rdata)
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{
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pjsip_tx_data *tdata;
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if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, code, NULL, &tdata) == PJ_SUCCESS) {
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struct pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
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pjsip_dlg_send_response(session->inv_session->dlg, tsx, tdata);
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}
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}
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static void channel_cleanup_wrapper(void *data)
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{
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struct ast_channel *chan = data;
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ast_channel_cleanup(chan);
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}
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static struct ast_datastore_info call_feature_info = {
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.type = "REFER call feature info",
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.destroy = channel_cleanup_wrapper,
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};
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static pjsip_param *get_diversion_reason(pjsip_fromto_hdr *hdr)
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{
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static const pj_str_t reason_str = { "reason", 6 };
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return pjsip_param_find(&hdr->other_param, &reason_str);
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}
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static pjsip_fromto_hdr *get_diversion_header(pjsip_rx_data *rdata)
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{
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static const pj_str_t from_str = { "From", 4 };
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static const pj_str_t diversion_str = { "Diversion", 9 };
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pjsip_generic_string_hdr *hdr;
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pj_str_t value;
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if (!(hdr = pjsip_msg_find_hdr_by_name(
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rdata->msg_info.msg, &diversion_str, NULL))) {
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return NULL;
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}
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pj_strdup_with_null(rdata->tp_info.pool, &value, &hdr->hvalue);
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/* parse as a fromto header */
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return pjsip_parse_hdr(rdata->tp_info.pool, &from_str, value.ptr,
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pj_strlen(&value), NULL);
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}
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static int has_diversion_reason(pjsip_rx_data *rdata)
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{
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pjsip_param *reason;
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pjsip_fromto_hdr *hdr = get_diversion_header(rdata);
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return hdr &&
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(reason = get_diversion_reason(hdr)) &&
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!pj_stricmp2(&reason->value, SEND_TO_VM_REDIRECT_VALUE);
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}
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static int has_call_feature(pjsip_rx_data *rdata)
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{
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static const pj_str_t call_feature_str = { "X-Digium-Call-Feature", 21 };
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pjsip_generic_string_hdr *hdr = pjsip_msg_find_hdr_by_name(
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rdata->msg_info.msg, &call_feature_str, NULL);
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return hdr && !pj_stricmp2(&hdr->hvalue, SEND_TO_VM_HEADER_VALUE);
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}
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static int handle_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
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{
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struct ast_datastore *sip_session_datastore;
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struct ast_channel *other_party;
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int has_feature = has_call_feature(rdata);
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int has_reason = has_diversion_reason(rdata);
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if (!has_feature && !has_reason) {
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/* If we don't have a call feature or diversion reason or if
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it's not a feature this module is related to then there
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is nothing to do. */
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return 0;
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}
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/* Check bridge status... */
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other_party = ast_channel_bridge_peer(session->channel);
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if (!other_party) {
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/* The channel wasn't in a two party bridge */
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ast_log(LOG_WARNING, "%s (%s) attempted to transfer to voicemail, "
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"but was not in a two party bridge.\n",
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ast_sorcery_object_get_id(session->endpoint),
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ast_channel_name(session->channel));
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send_response(session, 400, rdata);
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return -1;
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}
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sip_session_datastore = ast_sip_session_alloc_datastore(
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&call_feature_info, DATASTORE_NAME);
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if (!sip_session_datastore) {
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ast_channel_unref(other_party);
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send_response(session, 500, rdata);
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return -1;
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}
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sip_session_datastore->data = other_party;
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if (ast_sip_session_add_datastore(session, sip_session_datastore)) {
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ast_channel_unref(other_party);
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ao2_ref(sip_session_datastore, -1);
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send_response(session, 500, rdata);
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return -1;
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}
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ao2_ref(sip_session_datastore, -1);
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if (has_feature) {
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pbx_builtin_setvar_helper(other_party, SEND_TO_VM_HEADER,
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SEND_TO_VM_HEADER_VALUE);
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}
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if (has_reason) {
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pbx_builtin_setvar_helper(other_party, SEND_TO_VM_REDIRECT,
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SEND_TO_VM_REDIRECT_VALUE);
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}
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return 0;
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}
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static void handle_outgoing_response(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
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{
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pjsip_status_line status = tdata->msg->line.status;
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struct ast_datastore *feature_datastore =
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ast_sip_session_get_datastore(session, DATASTORE_NAME);
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struct ast_channel *target_chan;
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if (!feature_datastore) {
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return;
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}
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/* Since we are handling the response, there is no need to keep the datastore in the session anymore. */
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ast_sip_session_remove_datastore(session, DATASTORE_NAME);
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/* If the response >= 300, the refer failed and we need to clear the feature. */
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if (status.code >= 300) {
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target_chan = feature_datastore->data;
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pbx_builtin_setvar_helper(target_chan, SEND_TO_VM_HEADER, NULL);
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pbx_builtin_setvar_helper(target_chan, SEND_TO_VM_REDIRECT, NULL);
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}
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ao2_ref(feature_datastore, -1);
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}
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static struct ast_sip_session_supplement refer_supplement = {
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.method = "REFER",
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.incoming_request = handle_incoming_request,
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.outgoing_response = handle_outgoing_response,
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};
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static int load_module(void)
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{
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if (ast_sip_session_register_supplement(&refer_supplement)) {
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ast_log(LOG_ERROR, "Unable to register Send to Voicemail supplement\n");
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return AST_MODULE_LOAD_FAILURE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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static int unload_module(void)
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{
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ast_sip_session_unregister_supplement(&refer_supplement);
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return 0;
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP REFER Send to Voicemail Support",
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.support_level = AST_MODULE_SUPPORT_CORE,
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.load = load_module,
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.unload = unload_module,
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.load_pri = AST_MODPRI_APP_DEPEND,
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);
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