asterisk/res/res_pjsip_diversion.c

356 lines
9.9 KiB
C
Raw Normal View History

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* Kevin Harwell <kharwell@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.
*/
/*** 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/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/callerid.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/strings.h"
static const pj_str_t diversion_name = { "Diversion", 9 };
/*! \brief Diversion header reasons
*
* The core defines a bunch of constants used to define
* redirecting reasons. This provides a translation table
* between those and the strings which may be present in
* a SIP Diversion header
*/
static const struct reasons {
enum AST_REDIRECTING_REASON code;
char *const text;
} reason_table[] = {
{ AST_REDIRECTING_REASON_UNKNOWN, "unknown" },
{ AST_REDIRECTING_REASON_USER_BUSY, "user-busy" },
{ AST_REDIRECTING_REASON_NO_ANSWER, "no-answer" },
{ AST_REDIRECTING_REASON_UNAVAILABLE, "unavailable" },
{ AST_REDIRECTING_REASON_UNCONDITIONAL, "unconditional" },
{ AST_REDIRECTING_REASON_TIME_OF_DAY, "time-of-day" },
{ AST_REDIRECTING_REASON_DO_NOT_DISTURB, "do-not-disturb" },
{ AST_REDIRECTING_REASON_DEFLECTION, "deflection" },
{ AST_REDIRECTING_REASON_FOLLOW_ME, "follow-me" },
{ AST_REDIRECTING_REASON_OUT_OF_ORDER, "out-of-service" },
{ AST_REDIRECTING_REASON_AWAY, "away" },
{ AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"},
{ AST_REDIRECTING_REASON_SEND_TO_VM, "send_to_vm"},
};
static const char *reason_code_to_str(const struct ast_party_redirecting_reason *reason)
{
int code = reason->code;
/* use specific string if given */
if (!ast_strlen_zero(reason->str)) {
return reason->str;
}
if (code >= 0 && code < ARRAY_LEN(reason_table)) {
return reason_table[code].text;
}
return "unknown";
}
static enum AST_REDIRECTING_REASON reason_str_to_code(const char *text)
{
enum AST_REDIRECTING_REASON code = AST_REDIRECTING_REASON_UNKNOWN;
int i;
for (i = 0; i < ARRAY_LEN(reason_table); ++i) {
if (!strcasecmp(text, reason_table[i].text)) {
code = reason_table[i].code;
break;
}
}
return code;
}
static pjsip_fromto_hdr *get_diversion_header(pjsip_rx_data *rdata)
{
static const pj_str_t from_name = { "From", 4 };
pjsip_generic_string_hdr *hdr;
pj_str_t value;
int size;
if (!(hdr = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &diversion_name, 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_name, value.ptr,
pj_strlen(&value), &size);
}
static void set_redirecting_value(char **dst, const pj_str_t *src)
{
ast_free(*dst);
*dst = ast_malloc(pj_strlen(src) + 1);
ast_copy_pj_str(*dst, src, pj_strlen(src) + 1);
}
static void set_redirecting_id(pjsip_name_addr *name_addr, struct ast_party_id *data,
struct ast_set_party_id *update)
{
pjsip_sip_uri *uri = pjsip_uri_get_uri(name_addr->uri);
if (pj_strlen(&uri->user)) {
update->number = 1;
data->number.valid = 1;
set_redirecting_value(&data->number.str, &uri->user);
}
if (pj_strlen(&name_addr->display)) {
update->name = 1;
data->name.valid = 1;
set_redirecting_value(&data->name.str, &name_addr->display);
}
}
static void copy_redirecting_id(struct ast_party_id *dst, const struct ast_party_id *src,
struct ast_set_party_id *update)
{
ast_party_id_copy(dst, src);
if (dst->number.valid) {
update->number = 1;
}
if (dst->name.valid) {
update->name = 1;
}
}
static void set_redirecting_reason(pjsip_fromto_hdr *hdr,
struct ast_party_redirecting_reason *data)
{
static const pj_str_t reason_name = { "reason", 6 };
pjsip_param *reason = pjsip_param_find(&hdr->other_param, &reason_name);
if (!reason) {
return;
}
set_redirecting_value(&data->str, &reason->value);
data->code = reason_str_to_code(data->str);
}
static void set_redirecting(struct ast_sip_session *session,
pjsip_fromto_hdr *from_info,
pjsip_name_addr *to_info)
{
struct ast_party_redirecting data;
struct ast_set_party_redirecting update;
if (!session->channel) {
return;
}
ast_party_redirecting_init(&data);
memset(&update, 0, sizeof(update));
if (from_info) {
set_redirecting_id((pjsip_name_addr*)from_info->uri,
&data.from, &update.from);
set_redirecting_reason(from_info, &data.reason);
} else {
copy_redirecting_id(&data.from, &session->id, &update.from);
}
set_redirecting_id(to_info, &data.to, &update.to);
ast_set_party_id_all(&update.priv_orig);
ast_set_party_id_all(&update.priv_from);
ast_set_party_id_all(&update.priv_to);
++data.count;
ast_channel_set_redirecting(session->channel, &data, &update);
ast_party_redirecting_free(&data);
}
static int diversion_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
pjsip_fromto_hdr *hdr = get_diversion_header(rdata);
if (hdr) {
set_redirecting(session, hdr, (pjsip_name_addr*)
PJSIP_MSG_TO_HDR(rdata->msg_info.msg)->uri);
}
return 0;
}
static void diversion_incoming_response(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
static const pj_str_t contact_name = { "Contact", 7 };
pjsip_status_line status = rdata->msg_info.msg->line.status;
pjsip_fromto_hdr *div_hdr;
pjsip_contact_hdr *contact_hdr;
if ((status.code != 302) && (status.code != 181)) {
return;
}
/* use the diversion header info if there is one. if not one then use the
session caller id info. if that doesn't exist use info from the To hdr*/
if (!(div_hdr = get_diversion_header(rdata)) && !session->id.number.valid) {
div_hdr = PJSIP_MSG_TO_HDR(rdata->msg_info.msg);
}
contact_hdr = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &contact_name, NULL);
set_redirecting(session, div_hdr, contact_hdr ? (pjsip_name_addr*)contact_hdr->uri :
(pjsip_name_addr*)PJSIP_MSG_FROM_HDR(rdata->msg_info.msg)->uri);
}
/*!
* \internal
* \brief Adds diversion header information to an outbound SIP message
*
* \param tdata The outbound message
* \param data The redirecting data used to fill parts of the diversion header
*/
static void add_diversion_header(pjsip_tx_data *tdata, struct ast_party_redirecting *data)
{
pjsip_fromto_hdr *hdr;
pjsip_name_addr *name_addr;
pjsip_sip_uri *uri;
pjsip_param *param;
Detect potential forwarding loops based on count. A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
2015-04-15 15:38:02 +00:00
pjsip_fromto_hdr *old_hdr;
struct ast_party_id *id = &data->from;
pjsip_uri *base = PJSIP_MSG_FROM_HDR(tdata->msg)->uri;
if (!id->number.valid || ast_strlen_zero(id->number.str)) {
return;
}
hdr = pjsip_from_hdr_create(tdata->pool);
hdr->type = PJSIP_H_OTHER;
pj_strdup(tdata->pool, &hdr->name, &diversion_name);
hdr->sname.slen = 0;
name_addr = pjsip_uri_clone(tdata->pool, base);
uri = pjsip_uri_get_uri(name_addr->uri);
pj_strdup2(tdata->pool, &name_addr->display, id->name.str);
pj_strdup2(tdata->pool, &uri->user, id->number.str);
param = PJ_POOL_ALLOC_T(tdata->pool, pjsip_param);
param->name = pj_str("reason");
param->value = pj_str((char*)reason_code_to_str(&data->reason));
pj_list_insert_before(&hdr->other_param, param);
hdr->uri = (pjsip_uri *) name_addr;
Detect potential forwarding loops based on count. A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
2015-04-15 15:38:02 +00:00
old_hdr = pjsip_msg_find_hdr_by_name(tdata->msg, &diversion_name, NULL);
if (old_hdr) {
pj_list_erase(old_hdr);
}
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)hdr);
}
static void get_redirecting_add_diversion(struct ast_sip_session *session, pjsip_tx_data *tdata)
{
struct ast_party_redirecting *data;
if (session->channel && session->endpoint->id.send_diversion &&
(data = ast_channel_redirecting(session->channel))->count) {
add_diversion_header(tdata, data);
}
}
/*!
* \internal
* \brief Adds a diversion header to an outgoing INVITE request if
* redirecting information is available.
*
* \param session The session on which the INVITE request is to be sent
* \param tdata The outbound INVITE request
*/
static void diversion_outgoing_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
{
get_redirecting_add_diversion(session, tdata);
}
/*!
* \internal
* \brief Adds a diversion header to an outgoing 3XX response
*
* \param session The session on which the INVITE response is to be sent
* \param tdata The outbound INVITE response
*/
static void diversion_outgoing_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
{
struct pjsip_status_line status = tdata->msg->line.status;
/* add to 302 and 181 */
if (PJSIP_IS_STATUS_IN_CLASS(status.code, 300) || (status.code == 181)) {
get_redirecting_add_diversion(session, tdata);
}
}
static struct ast_sip_session_supplement diversion_supplement = {
.method = "INVITE",
/* this supplement needs to be called after caller id
and after the channel has been created */
.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL + 100,
.incoming_request = diversion_incoming_request,
.incoming_response = diversion_incoming_response,
.outgoing_request = diversion_outgoing_request,
.outgoing_response = diversion_outgoing_response,
Resolve race condition where channels enter dialplan application before media has been negotiated. Testsuite tests will occasionally fail because on reception of a 200 OK SIP response, an AST_CONTROL_ANSWER frame is queued prior to when media has finished being negotiated. This is because session supplements are called into before PJSIP's inv_session code has told us that media has been updated. Sometimes the queued answer frame is handled by the PBX thread before the ensuing media negotiations occur, causing a test failure. As it turns out, there is another place that session supplements could be called into, which is after media has finished getting negotiated. What this commit introduces is a means for session supplements to indicate when they wish to be called into when handling an incoming SIP response. By default, all session supplements will be run at the same point that they were prior to this commit. However, session supplements may indicate that they wish to be handled earlier than normal on redirects, or they may indicate they wish to be handled after media has been negotiated. In this changeset, two session supplements have been updated to indicate a preference for when they should be run: res_pjsip_diversion executes before handling redirection in order to get information from the Diversion header, and chan_pjsip now handles responses to INVITEs after media negotiation to fix the race condition mentioned previously. ASTERISK-24212 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/3930 ........ Merged revisions 422536 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422542 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-02 20:29:58 +00:00
.response_priority = AST_SIP_SESSION_BEFORE_REDIRECTING,
};
static int load_module(void)
{
CHECK_PJSIP_SESSION_MODULE_LOADED();
ast_sip_session_register_supplement(&diversion_supplement);
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
ast_sip_session_unregister_supplement(&diversion_supplement);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Add Diversion Header Support",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
);