file:///srv/subversion/repos/asterisk/trunk ................ r376918 | mmichelson | 2012-11-30 10:56:53 -0600 (Fri, 30 Nov 2012) | 29 lines Fix potential crashes during SIP attended transfers. The principal behind this patch is simple. During a transfer, we manipulate channels that are owned by a separate thread than the one we currently are running in, so it makes sense that we need to grab a reference to the channels so that they cannot disappear out from under us. In the wild, crashes were sometimes seen when the transferring party would hang up the call before the transfer target answered the call. The most common place to see the crash occur was when attempting to send a connected line update to the transferer channel. (closes issue ASTERISK-20226) Reported by Jared Smith Patches: ASTERISK-20226.patch uploaded by Mark Michelson (License #5049) Tested by: Jared Smith ........ Merged revisions 376901 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376916 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376917 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r376922 | seanbright | 2012-11-30 11:08:41 -0600 (Fri, 30 Nov 2012) | 11 lines Minor spelling fix to the VOLUME documentation. ........ Merged revisions 376919 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376920 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376921 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376929 65c4cc65-6c06-0410-ace0-fbb531ad65f3
237 lines
6.1 KiB
C
237 lines
6.1 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2011, Digium, Inc.
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*
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* Joshua Colp <jcolp@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 Technology independent volume control
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*
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* \author Joshua Colp <jcolp@digium.com>
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*
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* \ingroup functions
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*
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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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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/module.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/utils.h"
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#include "asterisk/audiohook.h"
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#include "asterisk/app.h"
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/*** DOCUMENTATION
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<function name="VOLUME" language="en_US">
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<synopsis>
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Set the TX or RX volume of a channel.
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</synopsis>
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<syntax>
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<parameter name="direction" required="true">
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<para>Must be <literal>TX</literal> or <literal>RX</literal>.</para>
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</parameter>
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<parameter name="options">
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<optionlist>
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<option name="p">
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<para>Enable DTMF volume control</para>
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</option>
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</optionlist>
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</parameter>
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</syntax>
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<description>
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<para>The VOLUME function can be used to increase or decrease the <literal>tx</literal> or
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<literal>rx</literal> gain of any channel.</para>
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<para>For example:</para>
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<para>Set(VOLUME(TX)=3)</para>
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<para>Set(VOLUME(RX)=2)</para>
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<para>Set(VOLUME(TX,p)=3)</para>
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<para>Set(VOLUME(RX,p)=3)</para>
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</description>
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</function>
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***/
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struct volume_information {
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struct ast_audiohook audiohook;
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int tx_gain;
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int rx_gain;
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unsigned int flags;
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};
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enum volume_flags {
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VOLUMEFLAG_CHANGE = (1 << 1),
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};
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AST_APP_OPTIONS(volume_opts, {
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AST_APP_OPTION('p', VOLUMEFLAG_CHANGE),
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});
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static void destroy_callback(void *data)
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{
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struct volume_information *vi = data;
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/* Destroy the audiohook, and destroy ourselves */
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ast_audiohook_lock(&vi->audiohook);
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ast_audiohook_detach(&vi->audiohook);
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ast_audiohook_unlock(&vi->audiohook);
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ast_audiohook_destroy(&vi->audiohook);
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ast_free(vi);
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return;
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}
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/*! \brief Static structure for datastore information */
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static const struct ast_datastore_info volume_datastore = {
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.type = "volume",
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.destroy = destroy_callback
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};
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static int volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
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{
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struct ast_datastore *datastore = NULL;
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struct volume_information *vi = NULL;
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int *gain = NULL;
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/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
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if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
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return 0;
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/* Grab datastore which contains our gain information */
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if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL)))
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return 0;
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vi = datastore->data;
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/* If this is DTMF then allow them to increase/decrease the gains */
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if (ast_test_flag(vi, VOLUMEFLAG_CHANGE)) {
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if (frame->frametype == AST_FRAME_DTMF) {
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/* Only use DTMF coming from the source... not going to it */
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if (direction != AST_AUDIOHOOK_DIRECTION_READ)
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return 0;
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if (frame->subclass.integer == '*') {
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vi->tx_gain += 1;
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vi->rx_gain += 1;
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} else if (frame->subclass.integer == '#') {
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vi->tx_gain -= 1;
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vi->rx_gain -= 1;
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}
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}
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}
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if (frame->frametype == AST_FRAME_VOICE) {
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/* Based on direction of frame grab the gain, and confirm it is applicable */
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if (!(gain = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? &vi->rx_gain : &vi->tx_gain) || !*gain)
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return 0;
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/* Apply gain to frame... easy as pi */
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ast_frame_adjust_volume(frame, *gain);
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}
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return 0;
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}
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static int volume_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
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{
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struct ast_datastore *datastore = NULL;
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struct volume_information *vi = NULL;
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int is_new = 0;
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/* Separate options from argument */
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(direction);
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AST_APP_ARG(options);
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);
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AST_STANDARD_APP_ARGS(args, data);
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ast_channel_lock(chan);
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if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL))) {
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ast_channel_unlock(chan);
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/* Allocate a new datastore to hold the reference to this volume and audiohook information */
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if (!(datastore = ast_datastore_alloc(&volume_datastore, NULL)))
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return 0;
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if (!(vi = ast_calloc(1, sizeof(*vi)))) {
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ast_datastore_free(datastore);
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return 0;
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}
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ast_audiohook_init(&vi->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
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vi->audiohook.manipulate_callback = volume_callback;
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ast_set_flag(&vi->audiohook, AST_AUDIOHOOK_WANTS_DTMF);
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is_new = 1;
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} else {
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ast_channel_unlock(chan);
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vi = datastore->data;
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}
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/* Adjust gain on volume information structure */
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if (ast_strlen_zero(args.direction)) {
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ast_log(LOG_ERROR, "Direction must be specified for VOLUME function\n");
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return -1;
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}
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if (!strcasecmp(args.direction, "tx")) {
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vi->tx_gain = atoi(value);
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} else if (!strcasecmp(args.direction, "rx")) {
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vi->rx_gain = atoi(value);
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} else {
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ast_log(LOG_ERROR, "Direction must be either RX or TX\n");
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}
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if (is_new) {
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datastore->data = vi;
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ast_channel_lock(chan);
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ast_channel_datastore_add(chan, datastore);
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ast_channel_unlock(chan);
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ast_audiohook_attach(chan, &vi->audiohook);
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}
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/* Add Option data to struct */
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if (!ast_strlen_zero(args.options)) {
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struct ast_flags flags = { 0 };
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ast_app_parse_options(volume_opts, &flags, NULL, args.options);
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vi->flags = flags.flags;
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} else {
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vi->flags = 0;
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}
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return 0;
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}
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static struct ast_custom_function volume_function = {
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.name = "VOLUME",
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.write = volume_write,
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};
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static int unload_module(void)
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{
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return ast_custom_function_unregister(&volume_function);
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}
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static int load_module(void)
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{
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return ast_custom_function_register(&volume_function);
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Technology independent volume control");
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