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* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel driver specific. If the channel variable is set on the transferrer channel, the sound will be played to the target of an attended transfer. * The channel variable BRIDGEPEER becomes a comma separated list of peers in a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers listed. Any more peers in the bridge will not be included in the list. BRIDGEPEER is not valid in holding bridges like parking since those channels do not talk to each other even though they are in a bridge. * The channel variable BRIDGEPVTCALLID is only valid for two party bridges and will contain a value if the BRIDGEPEER's channel driver supports it. * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that activated the dynamic feature. * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set only on the channel executing the dynamic feature. Executing a dynamic feature on the bridge peer in a multi-party bridge will execute it on all peers of the activating channel. (closes issue ASTERISK-21555) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2582/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
496 lines
16 KiB
C
496 lines
16 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2009, 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 Built in bridging features
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*
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* \author Joshua Colp <jcolp@digium.com>
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*
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* \ingroup bridges
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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 <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <sys/types.h>
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#include <sys/stat.h>
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#include "asterisk/module.h"
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#include "asterisk/channel.h"
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#include "asterisk/bridging.h"
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#include "asterisk/bridging_technology.h"
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#include "asterisk/frame.h"
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#include "asterisk/file.h"
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#include "asterisk/app.h"
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#include "asterisk/astobj2.h"
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#include "asterisk/pbx.h"
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#include "asterisk/parking.h"
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#include "asterisk/features_config.h"
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/*!
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* \brief Helper function that presents dialtone and grabs extension
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*
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* \retval 0 on success
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* \retval -1 on failure
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*/
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static int grab_transfer(struct ast_channel *chan, char *exten, size_t exten_len, const char *context)
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{
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int res;
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int digit_timeout;
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RAII_VAR(struct ast_features_xfer_config *, xfer_cfg, NULL, ao2_cleanup);
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ast_channel_lock(chan);
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xfer_cfg = ast_get_chan_features_xfer_config(chan);
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if (!xfer_cfg) {
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ast_log(LOG_ERROR, "Unable to get transfer configuration\n");
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ast_channel_unlock(chan);
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return -1;
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}
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digit_timeout = xfer_cfg->transferdigittimeout;
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ast_channel_unlock(chan);
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/* Play the simple "transfer" prompt out and wait */
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res = ast_stream_and_wait(chan, "pbx-transfer", AST_DIGIT_ANY);
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ast_stopstream(chan);
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if (res < 0) {
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/* Hangup or error */
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return -1;
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}
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if (res) {
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/* Store the DTMF digit that interrupted playback of the file. */
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exten[0] = res;
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}
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/* Drop to dialtone so they can enter the extension they want to transfer to */
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res = ast_app_dtget(chan, context, exten, exten_len, exten_len - 1, digit_timeout);
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if (res < 0) {
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/* Hangup or error */
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res = -1;
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} else if (!res) {
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/* 0 for invalid extension dialed. */
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if (ast_strlen_zero(exten)) {
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ast_debug(1, "%s dialed no digits.\n", ast_channel_name(chan));
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} else {
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ast_debug(1, "%s dialed '%s@%s' does not exist.\n",
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ast_channel_name(chan), exten, context);
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}
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ast_stream_and_wait(chan, "pbx-invalid", AST_DIGIT_NONE);
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res = -1;
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} else {
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/* Dialed extension is valid. */
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res = 0;
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}
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return res;
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}
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static void copy_caller_data(struct ast_channel *dest, struct ast_channel *caller)
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{
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ast_channel_lock_both(caller, dest);
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ast_connected_line_copy_from_caller(ast_channel_connected(dest), ast_channel_caller(caller));
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ast_channel_inherit_variables(caller, dest);
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ast_channel_datastore_inherit(caller, dest);
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ast_channel_unlock(dest);
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ast_channel_unlock(caller);
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}
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/*! \brief Helper function that creates an outgoing channel and returns it immediately */
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static struct ast_channel *dial_transfer(struct ast_channel *caller, const char *exten, const char *context)
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{
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char destination[AST_MAX_EXTENSION + AST_MAX_CONTEXT + 1];
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struct ast_channel *chan;
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int cause;
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/* Fill the variable with the extension and context we want to call */
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snprintf(destination, sizeof(destination), "%s@%s", exten, context);
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/* Now we request a local channel to prepare to call the destination */
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chan = ast_request("Local", ast_channel_nativeformats(caller), caller, destination,
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&cause);
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if (!chan) {
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return NULL;
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}
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/* Who is transferring the call. */
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pbx_builtin_setvar_helper(chan, "TRANSFERERNAME", ast_channel_name(caller));
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/* Before we actually dial out let's inherit appropriate information. */
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copy_caller_data(chan, caller);
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/* Since the above worked fine now we actually call it and return the channel */
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if (ast_call(chan, destination, 0)) {
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ast_hangup(chan);
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return NULL;
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}
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return chan;
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}
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/*!
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* \internal
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* \brief Determine the transfer context to use.
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* \since 12.0.0
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*
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* \param transferer Channel initiating the transfer.
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* \param context User supplied context if available. May be NULL.
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*
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* \return The context to use for the transfer.
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*/
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static const char *get_transfer_context(struct ast_channel *transferer, const char *context)
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{
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if (!ast_strlen_zero(context)) {
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return context;
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}
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context = pbx_builtin_getvar_helper(transferer, "TRANSFER_CONTEXT");
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if (!ast_strlen_zero(context)) {
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return context;
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}
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context = ast_channel_macrocontext(transferer);
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if (!ast_strlen_zero(context)) {
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return context;
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}
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context = ast_channel_context(transferer);
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if (!ast_strlen_zero(context)) {
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return context;
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}
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return "default";
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}
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static void blind_transfer_cb(struct ast_channel *new_channel, void *user_data,
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enum ast_transfer_type transfer_type)
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{
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struct ast_channel *transferer_channel = user_data;
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if (transfer_type == AST_BRIDGE_TRANSFER_MULTI_PARTY) {
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copy_caller_data(new_channel, transferer_channel);
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}
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}
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/*! \brief Internal built in feature for blind transfers */
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static int feature_blind_transfer(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, void *hook_pvt)
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{
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char exten[AST_MAX_EXTENSION] = "";
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struct ast_bridge_features_blind_transfer *blind_transfer = hook_pvt;
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const char *context;
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char *goto_on_blindxfr;
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ast_bridge_channel_write_hold(bridge_channel, NULL);
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ast_channel_lock(bridge_channel->chan);
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context = ast_strdupa(get_transfer_context(bridge_channel->chan,
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blind_transfer ? blind_transfer->context : NULL));
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goto_on_blindxfr = ast_strdupa(S_OR(pbx_builtin_getvar_helper(bridge_channel->chan,
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"GOTO_ON_BLINDXFR"), ""));
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ast_channel_unlock(bridge_channel->chan);
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/* Grab the extension to transfer to */
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if (grab_transfer(bridge_channel->chan, exten, sizeof(exten), context)) {
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ast_bridge_channel_write_unhold(bridge_channel);
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return 0;
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}
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if (!ast_strlen_zero(goto_on_blindxfr)) {
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ast_debug(1, "After transfer, transferer %s goes to %s\n",
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ast_channel_name(bridge_channel->chan), goto_on_blindxfr);
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ast_after_bridge_set_go_on(bridge_channel->chan, NULL, NULL, 0, goto_on_blindxfr);
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}
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if (ast_bridge_transfer_blind(bridge_channel->chan, exten, context, blind_transfer_cb,
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bridge_channel->chan) != AST_BRIDGE_TRANSFER_SUCCESS &&
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!ast_strlen_zero(goto_on_blindxfr)) {
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ast_after_bridge_goto_discard(bridge_channel->chan);
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}
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return 0;
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}
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/*! Attended transfer code */
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enum atxfer_code {
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/*! Party C hungup or other reason to abandon the transfer. */
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ATXFER_INCOMPLETE,
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/*! Transfer party C to party A. */
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ATXFER_COMPLETE,
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/*! Turn the transfer into a threeway call. */
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ATXFER_THREEWAY,
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/*! Hangup party C and return party B to the bridge. */
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ATXFER_ABORT,
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};
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/*! \brief Attended transfer feature to complete transfer */
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static int attended_transfer_complete(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, void *hook_pvt)
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{
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enum atxfer_code *transfer_code = hook_pvt;
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*transfer_code = ATXFER_COMPLETE;
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ast_bridge_change_state(bridge_channel, AST_BRIDGE_CHANNEL_STATE_HANGUP);
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return 0;
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}
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/*! \brief Attended transfer feature to turn it into a threeway call */
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static int attended_transfer_threeway(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, void *hook_pvt)
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{
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enum atxfer_code *transfer_code = hook_pvt;
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*transfer_code = ATXFER_THREEWAY;
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ast_bridge_change_state(bridge_channel, AST_BRIDGE_CHANNEL_STATE_HANGUP);
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return 0;
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}
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/*! \brief Attended transfer feature to abort transfer */
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static int attended_transfer_abort(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, void *hook_pvt)
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{
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enum atxfer_code *transfer_code = hook_pvt;
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*transfer_code = ATXFER_ABORT;
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ast_bridge_change_state(bridge_channel, AST_BRIDGE_CHANNEL_STATE_HANGUP);
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return 0;
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}
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/*! \brief Internal built in feature for attended transfers */
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static int feature_attended_transfer(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, void *hook_pvt)
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{
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char exten[AST_MAX_EXTENSION] = "";
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struct ast_channel *peer;
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struct ast_bridge *attended_bridge;
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struct ast_bridge_features caller_features;
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int xfer_failed;
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struct ast_bridge_features_attended_transfer *attended_transfer = hook_pvt;
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const char *complete_sound;
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const char *context;
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enum atxfer_code transfer_code = ATXFER_INCOMPLETE;
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const char *atxfer_abort;
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const char *atxfer_threeway;
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const char *atxfer_complete;
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const char *fail_sound;
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RAII_VAR(struct ast_features_xfer_config *, xfer_cfg, NULL, ao2_cleanup);
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ast_bridge_channel_write_hold(bridge_channel, NULL);
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bridge = ast_bridge_channel_merge_inhibit(bridge_channel, +1);
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ast_channel_lock(bridge_channel->chan);
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context = ast_strdupa(get_transfer_context(bridge_channel->chan,
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attended_transfer ? attended_transfer->context : NULL));
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xfer_cfg = ast_get_chan_features_xfer_config(bridge_channel->chan);
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if (!xfer_cfg) {
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ast_log(LOG_ERROR, "Unable to get transfer configuration options\n");
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ast_channel_unlock(bridge_channel->chan);
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return 0;
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}
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if (attended_transfer) {
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atxfer_abort = ast_strdupa(S_OR(attended_transfer->abort, xfer_cfg->atxferabort));
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atxfer_threeway = ast_strdupa(S_OR(attended_transfer->threeway, xfer_cfg->atxferthreeway));
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atxfer_complete = ast_strdupa(S_OR(attended_transfer->complete, xfer_cfg->atxfercomplete));
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} else {
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atxfer_abort = ast_strdupa(xfer_cfg->atxferabort);
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atxfer_threeway = ast_strdupa(xfer_cfg->atxferthreeway);
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atxfer_complete = ast_strdupa(xfer_cfg->atxfercomplete);
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}
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fail_sound = ast_strdupa(xfer_cfg->xferfailsound);
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ast_channel_unlock(bridge_channel->chan);
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/* Grab the extension to transfer to */
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if (grab_transfer(bridge_channel->chan, exten, sizeof(exten), context)) {
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ast_bridge_merge_inhibit(bridge, -1);
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ao2_ref(bridge, -1);
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ast_bridge_channel_write_unhold(bridge_channel);
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return 0;
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}
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/* Get a channel that is the destination we wish to call */
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peer = dial_transfer(bridge_channel->chan, exten, context);
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if (!peer) {
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ast_bridge_merge_inhibit(bridge, -1);
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ao2_ref(bridge, -1);
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ast_stream_and_wait(bridge_channel->chan, fail_sound, AST_DIGIT_NONE);
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ast_bridge_channel_write_unhold(bridge_channel);
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return 0;
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}
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/* BUGBUG bridging API features does not support the features.conf atxfer bounce between C & B channels */
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/* Setup a DTMF menu to control the transfer. */
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if (ast_bridge_features_init(&caller_features)
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|| ast_bridge_hangup_hook(&caller_features,
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attended_transfer_complete, &transfer_code, NULL, 0)
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|| ast_bridge_dtmf_hook(&caller_features, atxfer_abort,
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attended_transfer_abort, &transfer_code, NULL, 0)
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|| ast_bridge_dtmf_hook(&caller_features, atxfer_complete,
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attended_transfer_complete, &transfer_code, NULL, 0)
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|| ast_bridge_dtmf_hook(&caller_features, atxfer_threeway,
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attended_transfer_threeway, &transfer_code, NULL, 0)) {
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ast_bridge_features_cleanup(&caller_features);
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ast_hangup(peer);
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ast_bridge_merge_inhibit(bridge, -1);
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ao2_ref(bridge, -1);
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ast_stream_and_wait(bridge_channel->chan, fail_sound, AST_DIGIT_NONE);
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ast_bridge_channel_write_unhold(bridge_channel);
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return 0;
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}
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/* Create a bridge to use to talk to the person we are calling */
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attended_bridge = ast_bridge_base_new(AST_BRIDGE_CAPABILITY_1TO1MIX,
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AST_BRIDGE_FLAG_DISSOLVE_HANGUP);
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if (!attended_bridge) {
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ast_bridge_features_cleanup(&caller_features);
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ast_hangup(peer);
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ast_bridge_merge_inhibit(bridge, -1);
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ao2_ref(bridge, -1);
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ast_stream_and_wait(bridge_channel->chan, fail_sound, AST_DIGIT_NONE);
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ast_bridge_channel_write_unhold(bridge_channel);
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return 0;
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}
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ast_bridge_merge_inhibit(attended_bridge, +1);
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/* This is how this is going down, we are imparting the channel we called above into this bridge first */
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/* BUGBUG we should impart the peer as an independent and move it to the original bridge. */
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if (ast_bridge_impart(attended_bridge, peer, NULL, NULL, 0)) {
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ast_bridge_destroy(attended_bridge);
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ast_bridge_features_cleanup(&caller_features);
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ast_hangup(peer);
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ast_bridge_merge_inhibit(bridge, -1);
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ao2_ref(bridge, -1);
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ast_stream_and_wait(bridge_channel->chan, fail_sound, AST_DIGIT_NONE);
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ast_bridge_channel_write_unhold(bridge_channel);
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return 0;
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}
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/*
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* For the caller we want to join the bridge in a blocking
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* fashion so we don't spin around in this function doing
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* nothing while waiting.
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*/
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ast_bridge_join(attended_bridge, bridge_channel->chan, NULL, &caller_features, NULL, 0);
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/*
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* BUGBUG there is a small window where the channel does not point to the bridge_channel.
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*
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* This window is expected to go away when atxfer is redesigned
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* to fully support existing functionality. There will be one
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* and only one ast_bridge_channel structure per channel.
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*/
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/* Point the channel back to the original bridge and bridge_channel. */
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ast_bridge_channel_lock(bridge_channel);
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ast_channel_lock(bridge_channel->chan);
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ast_channel_internal_bridge_channel_set(bridge_channel->chan, bridge_channel);
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ast_channel_internal_bridge_set(bridge_channel->chan, bridge_channel->bridge);
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ast_channel_unlock(bridge_channel->chan);
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ast_bridge_channel_unlock(bridge_channel);
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/* Wait for peer thread to exit bridge and die. */
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if (!ast_autoservice_start(bridge_channel->chan)) {
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ast_bridge_depart(peer);
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ast_autoservice_stop(bridge_channel->chan);
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} else {
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ast_bridge_depart(peer);
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}
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/* Now that all channels are out of it we can destroy the bridge and the feature structures */
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ast_bridge_destroy(attended_bridge);
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ast_bridge_features_cleanup(&caller_features);
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/* Is there a courtesy sound to play to the peer? */
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ast_channel_lock(bridge_channel->chan);
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complete_sound = pbx_builtin_getvar_helper(bridge_channel->chan,
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"ATTENDED_TRANSFER_COMPLETE_SOUND");
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if (!ast_strlen_zero(complete_sound)) {
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complete_sound = ast_strdupa(complete_sound);
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} else {
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complete_sound = NULL;
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}
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ast_channel_unlock(bridge_channel->chan);
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if (complete_sound) {
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pbx_builtin_setvar_helper(peer, "BRIDGE_PLAY_SOUND", complete_sound);
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}
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xfer_failed = -1;
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switch (transfer_code) {
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case ATXFER_INCOMPLETE:
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/* Peer hungup */
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break;
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case ATXFER_COMPLETE:
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/* The peer takes our place in the bridge. */
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ast_bridge_channel_write_unhold(bridge_channel);
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ast_bridge_change_state(bridge_channel, AST_BRIDGE_CHANNEL_STATE_HANGUP);
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xfer_failed = ast_bridge_impart(bridge_channel->bridge, peer, bridge_channel->chan, NULL, 1);
|
|
break;
|
|
case ATXFER_THREEWAY:
|
|
/*
|
|
* Transferer wants to convert to a threeway call.
|
|
*
|
|
* Just impart the peer onto the bridge and have us return to it
|
|
* as normal.
|
|
*/
|
|
ast_bridge_channel_write_unhold(bridge_channel);
|
|
xfer_failed = ast_bridge_impart(bridge_channel->bridge, peer, NULL, NULL, 1);
|
|
break;
|
|
case ATXFER_ABORT:
|
|
/* Transferer decided not to transfer the call after all. */
|
|
break;
|
|
}
|
|
ast_bridge_merge_inhibit(bridge, -1);
|
|
ao2_ref(bridge, -1);
|
|
if (xfer_failed) {
|
|
ast_hangup(peer);
|
|
if (!ast_check_hangup_locked(bridge_channel->chan)) {
|
|
ast_stream_and_wait(bridge_channel->chan, fail_sound, AST_DIGIT_NONE);
|
|
}
|
|
ast_bridge_channel_write_unhold(bridge_channel);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Internal built in feature for hangup */
|
|
static int feature_hangup(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, void *hook_pvt)
|
|
{
|
|
/*
|
|
* This is very simple, we simply change the state on the
|
|
* bridge_channel to force the channel out of the bridge and the
|
|
* core takes care of the rest.
|
|
*/
|
|
ast_bridge_change_state(bridge_channel, AST_BRIDGE_CHANNEL_STATE_END);
|
|
return 0;
|
|
}
|
|
|
|
static int unload_module(void)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
ast_bridge_features_register(AST_BRIDGE_BUILTIN_BLINDTRANSFER, feature_blind_transfer, NULL);
|
|
ast_bridge_features_register(AST_BRIDGE_BUILTIN_ATTENDEDTRANSFER, feature_attended_transfer, NULL);
|
|
ast_bridge_features_register(AST_BRIDGE_BUILTIN_HANGUP, feature_hangup, NULL);
|
|
|
|
/* Bump up our reference count so we can't be unloaded */
|
|
ast_module_ref(ast_module_info->self);
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Built in bridging features");
|