asterisk/main/pickup.c
Matt Jordan 4a58261694 git migration: Refactor the ASTERISK_FILE_VERSION macro
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-13 03:48:57 -04:00

410 lines
12 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2013, Digium, Inc.
* Copyright (C) 2012, Russell Bryant
*
* Matt Jordan <mjordan@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 Routines implementing call pickup
*
* \author Matt Jordan <mjordan@digium.com>
*/
/*!
* \li Call pickup uses the configuration file \ref features.conf
* \addtogroup configuration_file Configuration Files
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
/*** DOCUMENTATION
<managerEvent language="en_US" name="Pickup">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a call pickup occurs.</synopsis>
<syntax>
<channel_snapshot/>
<channel_snapshot prefix="Target"/>
</syntax>
</managerEventInstance>
</managerEvent>
***/
#include "asterisk.h"
ASTERISK_REGISTER_FILE()
#include "asterisk/pickup.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/app.h"
#include "asterisk/callerid.h"
#include "asterisk/causes.h"
#include "asterisk/stasis.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/features_config.h"
static struct ast_manager_event_blob *call_pickup_to_ami(struct stasis_message *message);
STASIS_MESSAGE_TYPE_DEFN(
ast_call_pickup_type,
.to_ami = call_pickup_to_ami);
/*!
* The presence of this datastore on the channel indicates that
* someone is attemting to pickup or has picked up the channel.
* The purpose is to prevent a race between two channels
* attempting to pickup the same channel.
*/
static const struct ast_datastore_info pickup_active = {
.type = "pickup-active",
};
int ast_can_pickup(struct ast_channel *chan)
{
if (!ast_channel_pbx(chan) && !ast_channel_masq(chan) && !ast_test_flag(ast_channel_flags(chan), AST_FLAG_ZOMBIE)
&& (ast_channel_state(chan) == AST_STATE_RINGING
|| ast_channel_state(chan) == AST_STATE_RING
/*
* Check the down state as well because some SIP devices do not
* give 180 ringing when they can just give 183 session progress
* instead. Issue 14005. (Some ISDN switches as well for that
* matter.)
*/
|| ast_channel_state(chan) == AST_STATE_DOWN)
&& !ast_channel_datastore_find(chan, &pickup_active, NULL)) {
return 1;
}
return 0;
}
static int find_channel_by_group(void *obj, void *arg, void *data, int flags)
{
struct ast_channel *target = obj; /*!< Potential pickup target */
struct ast_channel *chan = arg; /*!< Channel wanting to pickup call */
if (chan == target) {
return 0;
}
ast_channel_lock(target);
if (ast_can_pickup(target)) {
/* Lock both channels. */
while (ast_channel_trylock(chan)) {
ast_channel_unlock(target);
sched_yield();
ast_channel_lock(target);
}
/*
* Both callgroup and namedcallgroup pickup variants are
* matched independently. Checking for named group match is
* done last since it's a more expensive operation.
*/
if ((ast_channel_pickupgroup(chan) & ast_channel_callgroup(target))
|| (ast_namedgroups_intersect(ast_channel_named_pickupgroups(chan),
ast_channel_named_callgroups(target)))) {
struct ao2_container *candidates = data;/*!< Candidate channels found. */
/* This is a candidate to pickup */
ao2_link(candidates, target);
}
ast_channel_unlock(chan);
}
ast_channel_unlock(target);
return 0;
}
struct ast_channel *ast_pickup_find_by_group(struct ast_channel *chan)
{
struct ao2_container *candidates;/*!< Candidate channels found to pickup. */
struct ast_channel *target;/*!< Potential pickup target */
candidates = ao2_container_alloc_options(AO2_ALLOC_OPT_LOCK_NOLOCK, 1, NULL, NULL);
if (!candidates) {
return NULL;
}
/* Find all candidate targets by group. */
ast_channel_callback(find_channel_by_group, chan, candidates, 0);
/* Find the oldest pickup target candidate */
target = NULL;
for (;;) {
struct ast_channel *candidate;/*!< Potential new older target */
struct ao2_iterator iter;
iter = ao2_iterator_init(candidates, 0);
while ((candidate = ao2_iterator_next(&iter))) {
if (!target) {
/* First target. */
target = candidate;
continue;
}
if (ast_tvcmp(ast_channel_creationtime(candidate), ast_channel_creationtime(target)) < 0) {
/* We have a new target. */
ast_channel_unref(target);
target = candidate;
continue;
}
ast_channel_unref(candidate);
}
ao2_iterator_destroy(&iter);
if (!target) {
/* No candidates found. */
break;
}
/* The found channel must be locked and ref'd. */
ast_channel_lock(target);
/* Recheck pickup ability */
if (ast_can_pickup(target)) {
/* This is the channel to pickup. */
break;
}
/* Someone else picked it up or the call went away. */
ast_channel_unlock(target);
ao2_unlink(candidates, target);
target = ast_channel_unref(target);
}
ao2_ref(candidates, -1);
return target;
}
/*!
* \brief Pickup a call
* \param chan channel that initiated pickup.
*
* Walk list of channels, checking it is not itself, channel is pbx one,
* check that the callgroup for both channels are the same and the channel is ringing.
* Answer calling channel, flag channel as answered on queue, masq channels together.
*/
int ast_pickup_call(struct ast_channel *chan)
{
struct ast_channel *target;/*!< Potential pickup target */
int res = -1;
RAII_VAR(struct ast_features_pickup_config *, pickup_cfg, NULL, ao2_cleanup);
const char *pickup_sound;
const char *fail_sound;
ast_debug(1, "Pickup attempt by %s\n", ast_channel_name(chan));
ast_channel_lock(chan);
pickup_cfg = ast_get_chan_features_pickup_config(chan);
if (!pickup_cfg) {
ast_log(LOG_ERROR, "Unable to retrieve pickup configuration. Unable to play pickup sounds\n");
}
pickup_sound = ast_strdupa(pickup_cfg ? pickup_cfg->pickupsound : "");
fail_sound = ast_strdupa(pickup_cfg ? pickup_cfg->pickupfailsound : "");
ast_channel_unlock(chan);
/* The found channel is already locked. */
target = ast_pickup_find_by_group(chan);
if (target) {
ast_log(LOG_NOTICE, "Pickup %s attempt by %s\n", ast_channel_name(target), ast_channel_name(chan));
res = ast_do_pickup(chan, target);
ast_channel_unlock(target);
if (!res) {
if (!ast_strlen_zero(pickup_sound)) {
pbx_builtin_setvar_helper(target, "BRIDGE_PLAY_SOUND", pickup_sound);
}
} else {
ast_log(LOG_WARNING, "Pickup %s failed by %s\n", ast_channel_name(target), ast_channel_name(chan));
}
target = ast_channel_unref(target);
}
if (res < 0) {
ast_debug(1, "No call pickup possible... for %s\n", ast_channel_name(chan));
if (!ast_strlen_zero(fail_sound)) {
ast_answer(chan);
ast_stream_and_wait(chan, fail_sound, "");
}
}
return res;
}
static struct ast_manager_event_blob *call_pickup_to_ami(struct stasis_message *message)
{
struct ast_multi_channel_blob *contents = stasis_message_data(message);
struct ast_channel_snapshot *chan;
struct ast_channel_snapshot *target;
struct ast_manager_event_blob *res;
RAII_VAR(struct ast_str *, channel_str, NULL, ast_free);
RAII_VAR(struct ast_str *, target_str, NULL, ast_free);
chan = ast_multi_channel_blob_get_channel(contents, "channel");
target = ast_multi_channel_blob_get_channel(contents, "target");
ast_assert(chan != NULL && target != NULL);
if (!(channel_str = ast_manager_build_channel_state_string(chan))) {
return NULL;
}
if (!(target_str = ast_manager_build_channel_state_string_prefix(target, "Target"))) {
return NULL;
}
res = ast_manager_event_blob_create(EVENT_FLAG_CALL, "Pickup",
"%s"
"%s",
ast_str_buffer(channel_str),
ast_str_buffer(target_str));
return res;
}
static int send_call_pickup_stasis_message(struct ast_channel *picking_up, struct ast_channel_snapshot *chan, struct ast_channel_snapshot *target)
{
RAII_VAR(struct ast_multi_channel_blob *, pickup_payload, NULL, ao2_cleanup);
RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
if (!ast_call_pickup_type()) {
return -1;
}
if (!(pickup_payload = ast_multi_channel_blob_create(ast_json_null()))) {
return -1;
}
ast_multi_channel_blob_add_channel(pickup_payload, "channel", chan);
ast_multi_channel_blob_add_channel(pickup_payload, "target", target);
if (!(msg = stasis_message_create(ast_call_pickup_type(), pickup_payload))) {
return -1;
}
stasis_publish(ast_channel_topic(picking_up), msg);
return 0;
}
int ast_do_pickup(struct ast_channel *chan, struct ast_channel *target)
{
struct ast_party_connected_line connected_caller;
struct ast_datastore *ds_pickup;
const char *chan_name;/*!< A masquerade changes channel names. */
const char *target_name;/*!< A masquerade changes channel names. */
int res = -1;
RAII_VAR(struct ast_channel_snapshot *, chan_snapshot, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, target_snapshot, NULL, ao2_cleanup);
target_name = ast_strdupa(ast_channel_name(target));
ast_debug(1, "Call pickup on '%s' by '%s'\n", target_name, ast_channel_name(chan));
/* Mark the target to block any call pickup race. */
ds_pickup = ast_datastore_alloc(&pickup_active, NULL);
if (!ds_pickup) {
ast_log(LOG_WARNING,
"Unable to create channel datastore on '%s' for call pickup\n", target_name);
return -1;
}
ast_channel_datastore_add(target, ds_pickup);
ast_party_connected_line_init(&connected_caller);
ast_party_connected_line_copy(&connected_caller, ast_channel_connected(target));
ast_channel_unlock(target);/* The pickup race is avoided so we do not need the lock anymore. */
/* Reset any earlier private connected id representation */
ast_party_id_reset(&connected_caller.priv);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
if (ast_channel_connected_line_sub(NULL, chan, &connected_caller, 0) &&
ast_channel_connected_line_macro(NULL, chan, &connected_caller, 0, 0)) {
ast_channel_update_connected_line(chan, &connected_caller, NULL);
}
ast_party_connected_line_free(&connected_caller);
ast_channel_lock(chan);
chan_name = ast_strdupa(ast_channel_name(chan));
ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(chan));
ast_channel_unlock(chan);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
if (ast_answer(chan)) {
ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan_name);
goto pickup_failed;
}
if (ast_queue_control(chan, AST_CONTROL_ANSWER)) {
ast_log(LOG_WARNING, "Unable to queue answer on '%s'\n", chan_name);
goto pickup_failed;
}
ast_channel_queue_connected_line_update(chan, &connected_caller, NULL);
/* setting the HANGUPCAUSE so the ringing channel knows this call was not a missed call */
ast_channel_hangupcause_set(chan, AST_CAUSE_ANSWERED_ELSEWHERE);
ast_channel_lock(chan);
chan_snapshot = ast_channel_snapshot_create(chan);
ast_channel_unlock(chan);
if (!chan_snapshot) {
goto pickup_failed;
}
target_snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(target));
if (!target_snapshot) {
goto pickup_failed;
}
if (ast_channel_move(target, chan)) {
ast_log(LOG_WARNING, "Unable to complete call pickup of '%s' with '%s'\n",
chan_name, target_name);
goto pickup_failed;
}
/* target points to the channel that did the pickup at this point, so use that channel's topic instead of chan */
send_call_pickup_stasis_message(target, chan_snapshot, target_snapshot);
res = 0;
pickup_failed:
ast_channel_lock(target);
if (!ast_channel_datastore_remove(target, ds_pickup)) {
ast_datastore_free(ds_pickup);
}
ast_party_connected_line_free(&connected_caller);
return res;
}
/*!
* \internal
* \brief Clean up resources on Asterisk shutdown
*/
static void pickup_shutdown(void)
{
STASIS_MESSAGE_TYPE_CLEANUP(ast_call_pickup_type);
}
int ast_pickup_init(void)
{
STASIS_MESSAGE_TYPE_INIT(ast_call_pickup_type);
ast_register_cleanup(pickup_shutdown);
return 0;
}