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/*
* 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"
#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",
);
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