2013-08-02 02:32:44 +00:00
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2013, Digium, Inc.
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* Copyright (C) 2012, Russell Bryant
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*
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* Matt Jordan <mjordan@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 Routines implementing call pickup
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*
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* \author Matt Jordan <mjordan@digium.com>
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*/
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/*!
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* \li Call pickup uses the configuration file \ref features.conf
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* \addtogroup configuration_file Configuration Files
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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/*** DOCUMENTATION
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<managerEvent language="en_US" name="Pickup">
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<managerEventInstance class="EVENT_FLAG_CALL">
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<synopsis>Raised when a call pickup occurs.</synopsis>
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<syntax>
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2013-08-02 13:29:01 +00:00
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<channel_snapshot/>
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<channel_snapshot prefix="Target"/>
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2013-08-02 02:32:44 +00:00
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</syntax>
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</managerEventInstance>
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</managerEvent>
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***/
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#include "asterisk.h"
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#include "asterisk/pickup.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/app.h"
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#include "asterisk/callerid.h"
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#include "asterisk/causes.h"
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#include "asterisk/stasis.h"
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#include "asterisk/stasis_channels.h"
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#include "asterisk/features_config.h"
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static struct ast_manager_event_blob *call_pickup_to_ami(struct stasis_message *message);
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STASIS_MESSAGE_TYPE_DEFN(
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ast_call_pickup_type,
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.to_ami = call_pickup_to_ami);
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/*!
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* The presence of this datastore on the channel indicates that
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* someone is attemting to pickup or has picked up the channel.
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* The purpose is to prevent a race between two channels
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* attempting to pickup the same channel.
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*/
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static const struct ast_datastore_info pickup_active = {
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.type = "pickup-active",
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};
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int ast_can_pickup(struct ast_channel *chan)
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{
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if (!ast_channel_pbx(chan) && !ast_channel_masq(chan) && !ast_test_flag(ast_channel_flags(chan), AST_FLAG_ZOMBIE)
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&& (ast_channel_state(chan) == AST_STATE_RINGING
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|| ast_channel_state(chan) == AST_STATE_RING
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/*
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* Check the down state as well because some SIP devices do not
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* give 180 ringing when they can just give 183 session progress
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* instead. Issue 14005. (Some ISDN switches as well for that
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* matter.)
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*/
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|| ast_channel_state(chan) == AST_STATE_DOWN)
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&& !ast_channel_datastore_find(chan, &pickup_active, NULL)) {
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return 1;
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}
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return 0;
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}
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static int find_channel_by_group(void *obj, void *arg, void *data, int flags)
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{
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struct ast_channel *target = obj; /*!< Potential pickup target */
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struct ast_channel *chan = arg; /*!< Channel wanting to pickup call */
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if (chan == target) {
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return 0;
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}
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ast_channel_lock(target);
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if (ast_can_pickup(target)) {
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/* Lock both channels. */
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while (ast_channel_trylock(chan)) {
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ast_channel_unlock(target);
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sched_yield();
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ast_channel_lock(target);
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}
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/*
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* Both callgroup and namedcallgroup pickup variants are
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* matched independently. Checking for named group match is
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* done last since it's a more expensive operation.
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*/
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if ((ast_channel_pickupgroup(chan) & ast_channel_callgroup(target))
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|| (ast_namedgroups_intersect(ast_channel_named_pickupgroups(chan),
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ast_channel_named_callgroups(target)))) {
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struct ao2_container *candidates = data;/*!< Candidate channels found. */
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/* This is a candidate to pickup */
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ao2_link(candidates, target);
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}
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ast_channel_unlock(chan);
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}
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ast_channel_unlock(target);
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return 0;
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}
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struct ast_channel *ast_pickup_find_by_group(struct ast_channel *chan)
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{
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struct ao2_container *candidates;/*!< Candidate channels found to pickup. */
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struct ast_channel *target;/*!< Potential pickup target */
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candidates = ao2_container_alloc_options(AO2_ALLOC_OPT_LOCK_NOLOCK, 1, NULL, NULL);
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if (!candidates) {
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return NULL;
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}
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/* Find all candidate targets by group. */
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ast_channel_callback(find_channel_by_group, chan, candidates, 0);
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/* Find the oldest pickup target candidate */
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target = NULL;
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for (;;) {
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struct ast_channel *candidate;/*!< Potential new older target */
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struct ao2_iterator iter;
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iter = ao2_iterator_init(candidates, 0);
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while ((candidate = ao2_iterator_next(&iter))) {
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if (!target) {
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/* First target. */
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target = candidate;
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continue;
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}
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if (ast_tvcmp(ast_channel_creationtime(candidate), ast_channel_creationtime(target)) < 0) {
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/* We have a new target. */
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ast_channel_unref(target);
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target = candidate;
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continue;
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}
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ast_channel_unref(candidate);
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}
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ao2_iterator_destroy(&iter);
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if (!target) {
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/* No candidates found. */
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break;
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}
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/* The found channel must be locked and ref'd. */
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ast_channel_lock(target);
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/* Recheck pickup ability */
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if (ast_can_pickup(target)) {
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/* This is the channel to pickup. */
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break;
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}
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/* Someone else picked it up or the call went away. */
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ast_channel_unlock(target);
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ao2_unlink(candidates, target);
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target = ast_channel_unref(target);
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}
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ao2_ref(candidates, -1);
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return target;
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}
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/*!
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* \brief Pickup a call
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* \param chan channel that initiated pickup.
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*
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* Walk list of channels, checking it is not itself, channel is pbx one,
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* check that the callgroup for both channels are the same and the channel is ringing.
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* Answer calling channel, flag channel as answered on queue, masq channels together.
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*/
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int ast_pickup_call(struct ast_channel *chan)
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{
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struct ast_channel *target;/*!< Potential pickup target */
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int res = -1;
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RAII_VAR(struct ast_features_pickup_config *, pickup_cfg, NULL, ao2_cleanup);
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const char *pickup_sound;
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const char *fail_sound;
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ast_debug(1, "Pickup attempt by %s\n", ast_channel_name(chan));
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ast_channel_lock(chan);
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pickup_cfg = ast_get_chan_features_pickup_config(chan);
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if (!pickup_cfg) {
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ast_log(LOG_ERROR, "Unable to retrieve pickup configuration. Unable to play pickup sounds\n");
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}
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pickup_sound = ast_strdupa(pickup_cfg ? pickup_cfg->pickupsound : "");
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fail_sound = ast_strdupa(pickup_cfg ? pickup_cfg->pickupfailsound : "");
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ast_channel_unlock(chan);
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/* The found channel is already locked. */
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target = ast_pickup_find_by_group(chan);
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if (target) {
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ast_log(LOG_NOTICE, "Pickup %s attempt by %s\n", ast_channel_name(target), ast_channel_name(chan));
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res = ast_do_pickup(chan, target);
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ast_channel_unlock(target);
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if (!res) {
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if (!ast_strlen_zero(pickup_sound)) {
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pbx_builtin_setvar_helper(target, "BRIDGE_PLAY_SOUND", pickup_sound);
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}
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} else {
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ast_log(LOG_WARNING, "Pickup %s failed by %s\n", ast_channel_name(target), ast_channel_name(chan));
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}
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target = ast_channel_unref(target);
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}
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if (res < 0) {
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ast_debug(1, "No call pickup possible... for %s\n", ast_channel_name(chan));
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if (!ast_strlen_zero(fail_sound)) {
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ast_answer(chan);
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ast_stream_and_wait(chan, fail_sound, "");
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}
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}
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return res;
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}
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static struct ast_manager_event_blob *call_pickup_to_ami(struct stasis_message *message)
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{
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struct ast_multi_channel_blob *contents = stasis_message_data(message);
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struct ast_channel_snapshot *chan;
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struct ast_channel_snapshot *target;
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struct ast_manager_event_blob *res;
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RAII_VAR(struct ast_str *, channel_str, NULL, ast_free);
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RAII_VAR(struct ast_str *, target_str, NULL, ast_free);
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chan = ast_multi_channel_blob_get_channel(contents, "channel");
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target = ast_multi_channel_blob_get_channel(contents, "target");
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ast_assert(chan != NULL && target != NULL);
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if (!(channel_str = ast_manager_build_channel_state_string(chan))) {
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return NULL;
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}
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if (!(target_str = ast_manager_build_channel_state_string_prefix(target, "Target"))) {
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return NULL;
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}
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res = ast_manager_event_blob_create(EVENT_FLAG_CALL, "Pickup",
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"%s"
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"%s",
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ast_str_buffer(channel_str),
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ast_str_buffer(target_str));
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return res;
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}
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static int send_call_pickup_stasis_message(struct ast_channel *picking_up, struct ast_channel_snapshot *chan, struct ast_channel_snapshot *target)
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{
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RAII_VAR(struct ast_multi_channel_blob *, pickup_payload, NULL, ao2_cleanup);
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RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
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2014-08-06 12:55:28 +00:00
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if (!ast_call_pickup_type()) {
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return -1;
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}
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2013-08-02 02:32:44 +00:00
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if (!(pickup_payload = ast_multi_channel_blob_create(ast_json_null()))) {
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return -1;
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}
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ast_multi_channel_blob_add_channel(pickup_payload, "channel", chan);
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ast_multi_channel_blob_add_channel(pickup_payload, "target", target);
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if (!(msg = stasis_message_create(ast_call_pickup_type(), pickup_payload))) {
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return -1;
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}
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stasis_publish(ast_channel_topic(picking_up), msg);
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return 0;
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}
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int ast_do_pickup(struct ast_channel *chan, struct ast_channel *target)
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{
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struct ast_party_connected_line connected_caller;
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struct ast_datastore *ds_pickup;
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const char *chan_name;/*!< A masquerade changes channel names. */
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const char *target_name;/*!< A masquerade changes channel names. */
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int res = -1;
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RAII_VAR(struct ast_channel_snapshot *, chan_snapshot, NULL, ao2_cleanup);
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RAII_VAR(struct ast_channel_snapshot *, target_snapshot, NULL, ao2_cleanup);
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target_name = ast_strdupa(ast_channel_name(target));
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ast_debug(1, "Call pickup on '%s' by '%s'\n", target_name, ast_channel_name(chan));
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/* Mark the target to block any call pickup race. */
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ds_pickup = ast_datastore_alloc(&pickup_active, NULL);
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if (!ds_pickup) {
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ast_log(LOG_WARNING,
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"Unable to create channel datastore on '%s' for call pickup\n", target_name);
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return -1;
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}
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ast_channel_datastore_add(target, ds_pickup);
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ast_party_connected_line_init(&connected_caller);
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ast_party_connected_line_copy(&connected_caller, ast_channel_connected(target));
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ast_channel_unlock(target);/* The pickup race is avoided so we do not need the lock anymore. */
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/* Reset any earlier private connected id representation */
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ast_party_id_reset(&connected_caller.priv);
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connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
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if (ast_channel_connected_line_sub(NULL, chan, &connected_caller, 0) &&
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ast_channel_connected_line_macro(NULL, chan, &connected_caller, 0, 0)) {
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ast_channel_update_connected_line(chan, &connected_caller, NULL);
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}
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ast_party_connected_line_free(&connected_caller);
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ast_channel_lock(chan);
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chan_name = ast_strdupa(ast_channel_name(chan));
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ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(chan));
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ast_channel_unlock(chan);
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connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
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if (ast_answer(chan)) {
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ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan_name);
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goto pickup_failed;
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}
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if (ast_queue_control(chan, AST_CONTROL_ANSWER)) {
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ast_log(LOG_WARNING, "Unable to queue answer on '%s'\n", chan_name);
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goto pickup_failed;
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}
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ast_channel_queue_connected_line_update(chan, &connected_caller, NULL);
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/* setting the HANGUPCAUSE so the ringing channel knows this call was not a missed call */
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ast_channel_hangupcause_set(chan, AST_CAUSE_ANSWERED_ELSEWHERE);
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2013-12-18 20:33:37 +00:00
|
|
|
ast_channel_lock(chan);
|
|
|
|
chan_snapshot = ast_channel_snapshot_create(chan);
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
if (!chan_snapshot) {
|
2013-08-02 02:32:44 +00:00
|
|
|
goto pickup_failed;
|
|
|
|
}
|
|
|
|
|
stasis: Reduce creation of channel snapshots to improve performance
During some performance testing of Asterisk with AGI, ARI, and lots of Local
channels, we noticed that there's quite a hit in performance during channel
creation and releasing to the dialplan (ARI continue). After investigating
the performance spike that occurs during channel creation, we discovered
that we create a lot of channel snapshots that are technically unnecessary.
This includes creating snapshots during:
* AGI execution
* Returning objects for ARI commands
* During some Local channel operations
* During some dialling operations
* During variable setting
* During some bridging operations
And more.
This patch does the following:
- It removes a number of fields from channel snapshots. These fields were
rarely used, were expensive to have on the snapshot, and hurt performance.
This included formats, translation paths, Log Call ID, callgroup, pickup
group, and all channel variables. As a result, AMI Status,
"core show channel", "core show channelvar", and "pjsip show channel" were
modified to either hit the live channel or not show certain pieces of data.
While this is unfortunate, the performance gain from this patch is worth
the loss in behaviour.
- It adds a mechanism to publish a cached snapshot + blob. A large number of
publications were changed to use this, including:
- During Dial begin
- During Variable assignment (if no AMI variables are emitted - if AMI
variables are set, we have to make snapshots when a variable is changed)
- During channel pickup
- When a channel is put on hold/unhold
- When a DTMF digit is begun/ended
- When creating a bridge snapshot
- When an AOC event is raised
- During Local channel optimization/Local bridging
- When endpoint snapshots are generated
- All AGI events
- All ARI responses that return a channel
- Events in the AgentPool, MeetMe, and some in Queue
- Additionally, some extraneous channel snapshots were being made that were
unnecessary. These were removed.
- The result of ast_hashtab_hash_string is now cached in stasis_cache. This
reduces a large number of calls to ast_hashtab_hash_string, which reduced
the amount of time spent in this function in gprof by around 50%.
#ASTERISK-23811 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3568/
........
Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
|
|
|
target_snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(target));
|
2013-12-18 20:33:37 +00:00
|
|
|
if (!target_snapshot) {
|
2013-08-02 02:32:44 +00:00
|
|
|
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;
|
|
|
|
}
|
|
|
|
|
2013-08-16 17:33:21 +00:00
|
|
|
/*!
|
|
|
|
* \internal
|
|
|
|
* \brief Clean up resources on Asterisk shutdown
|
|
|
|
*/
|
2013-08-02 02:32:44 +00:00
|
|
|
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);
|
2015-03-26 22:24:26 +00:00
|
|
|
ast_register_cleanup(pickup_shutdown);
|
2013-08-02 02:32:44 +00:00
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|