freeswitch/src/switch_ivr.c

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/*
* FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application
* Copyright (C) 2005/2006, Anthony Minessale II <anthmct@yahoo.com>
*
* Version: MPL 1.1
*
* The contents of this file are subject to the Mozilla Public License Version
* 1.1 (the "License"); you may not use this file except in compliance with
* the License. You may obtain a copy of the License at
* http://www.mozilla.org/MPL/
*
* Software distributed under the License is distributed on an "AS IS" basis,
* WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License
* for the specific language governing rights and limitations under the
* License.
*
* The Original Code is FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application
*
* The Initial Developer of the Original Code is
* Anthony Minessale II <anthmct@yahoo.com>
* Portions created by the Initial Developer are Copyright (C)
* the Initial Developer. All Rights Reserved.
*
* Contributor(s):
*
* Anthony Minessale II <anthmct@yahoo.com>
*
*
* switch_ivr_api.c -- IVR Library
*
*/
#include <switch.h>
#include <switch_ivr.h>
static const switch_state_handler_table_t audio_bridge_peer_state_handlers;
SWITCH_DECLARE(switch_status_t) switch_ivr_sleep(switch_core_session_t *session, uint32_t ms)
{
switch_channel_t *channel;
switch_status_t status = SWITCH_STATUS_SUCCESS;
switch_time_t start, now, done = switch_time_now() + (ms * 1000);
switch_frame_t *read_frame;
int32_t left, elapsed;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
start = switch_time_now();
for(;;) {
now = switch_time_now();
elapsed = (int32_t)((now - start) / 1000);
left = ms - elapsed;
if (!switch_channel_ready(channel)) {
status = SWITCH_STATUS_FALSE;
break;
}
if (now > done || left <= 0) {
break;
}
if (switch_channel_test_flag(channel, CF_SERVICE)) {
switch_yield(1000);
} else {
status = switch_core_session_read_frame(session, &read_frame, left, 0);
if (!SWITCH_READ_ACCEPTABLE(status)) {
break;
}
}
}
return status;
}
static void switch_ivr_parse_event(switch_core_session_t *session, switch_event_t *event)
{
switch_channel_t *channel = switch_core_session_get_channel(session);
char *cmd = switch_event_get_header(event, "call-command");
unsigned long cmd_hash;
apr_ssize_t hlen = APR_HASH_KEY_STRING;
unsigned long CMD_EXECUTE = apr_hashfunc_default("execute", &hlen);
unsigned long CMD_HANGUP = apr_hashfunc_default("hangup", &hlen);
assert(channel != NULL);
hlen = (switch_size_t) strlen(cmd);
cmd_hash = apr_hashfunc_default(cmd, &hlen);
if (!switch_strlen_zero(cmd)) {
if (cmd_hash == CMD_EXECUTE) {
const switch_application_interface_t *application_interface;
char *app_name = switch_event_get_header(event, "execute-app-name");
char *app_arg = switch_event_get_header(event, "execute-app-arg");
if (app_name && app_arg) {
if ((application_interface = switch_loadable_module_get_application_interface(app_name))) {
if (application_interface->application_function) {
application_interface->application_function(session, app_arg);
}
}
}
} else if (cmd_hash == CMD_HANGUP) {
char *cause_name = switch_event_get_header(event, "hangup-cause");
switch_call_cause_t cause = SWITCH_CAUSE_NORMAL_CLEARING;
if (cause_name) {
cause = switch_channel_str2cause(cause_name);
}
switch_channel_hangup(channel, cause);
}
}
}
SWITCH_DECLARE(switch_status_t) switch_ivr_park(switch_core_session_t *session)
{
switch_status_t status = SWITCH_STATUS_SUCCESS;
switch_channel_t *channel;
switch_frame_t *frame;
int stream_id;
switch_event_t *event;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
switch_channel_answer(channel);
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_PARK) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(channel, event);
switch_event_fire(&event);
}
switch_channel_set_flag(channel, CF_CONTROLLED);
while (switch_channel_ready(channel)) {
for (stream_id = 0; stream_id < switch_core_session_get_stream_count(session); stream_id++) {
if ((status = switch_core_session_read_frame(session, &frame, -1, stream_id)) == SWITCH_STATUS_SUCCESS) {
if (!SWITCH_READ_ACCEPTABLE(status)) {
break;
}
if (switch_core_session_dequeue_private_event(session, &event) == SWITCH_STATUS_SUCCESS) {
switch_ivr_parse_event(session, event);
switch_event_destroy(&event);
}
}
}
}
switch_channel_clear_flag(channel, CF_CONTROLLED);
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_UNPARK) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(channel, event);
switch_event_fire(&event);
}
return status;
}
SWITCH_DECLARE(switch_status_t) switch_ivr_collect_digits_callback(switch_core_session_t *session,
switch_input_callback_function_t input_callback,
void *buf,
unsigned int buflen)
{
switch_channel_t *channel;
switch_status_t status = SWITCH_STATUS_SUCCESS;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
if (!input_callback) {
return SWITCH_STATUS_GENERR;
}
while(switch_channel_ready(channel)) {
switch_frame_t *read_frame;
switch_event_t *event;
char dtmf[128];
if (switch_core_session_dequeue_private_event(session, &event) == SWITCH_STATUS_SUCCESS) {
switch_ivr_parse_event(session, event);
switch_event_destroy(&event);
}
if (switch_channel_has_dtmf(channel)) {
switch_channel_dequeue_dtmf(channel, dtmf, sizeof(dtmf));
status = input_callback(session, dtmf, SWITCH_INPUT_TYPE_DTMF, buf, buflen);
}
if (switch_core_session_dequeue_event(session, &event) == SWITCH_STATUS_SUCCESS) {
status = input_callback(session, event, SWITCH_INPUT_TYPE_EVENT, buf, buflen);
switch_event_destroy(&event);
}
if (status != SWITCH_STATUS_SUCCESS) {
break;
}
if (switch_channel_test_flag(channel, CF_SERVICE)) {
switch_yield(1000);
} else {
status = switch_core_session_read_frame(session, &read_frame, -1, 0);
}
if (!SWITCH_READ_ACCEPTABLE(status)) {
break;
}
}
return status;
}
SWITCH_DECLARE(switch_status_t) switch_ivr_collect_digits_count(switch_core_session_t *session,
char *buf,
unsigned int buflen,
unsigned int maxdigits,
const char *terminators,
char *terminator,
unsigned int timeout)
{
unsigned int i = 0, x = (unsigned int) strlen(buf);
switch_channel_t *channel;
switch_status_t status = SWITCH_STATUS_SUCCESS;
switch_time_t started = 0;
unsigned int elapsed;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
*terminator = '\0';
if (terminators) {
for (i = 0 ; i < x; i++) {
if (strchr(terminators, buf[i])) {
*terminator = buf[i];
return SWITCH_STATUS_SUCCESS;
}
}
}
if (timeout) {
started = switch_time_now();
}
while(switch_channel_ready(channel)) {
switch_frame_t *read_frame;
switch_event_t *event;
if (timeout) {
elapsed = (unsigned int)((switch_time_now() - started) / 1000);
if (elapsed >= timeout) {
break;
}
}
if (switch_core_session_dequeue_private_event(session, &event) == SWITCH_STATUS_SUCCESS) {
switch_ivr_parse_event(session, event);
switch_event_destroy(&event);
}
if (switch_channel_has_dtmf(channel)) {
char dtmf[128];
switch_channel_dequeue_dtmf(channel, dtmf, sizeof(dtmf));
for(i =0 ; i < (unsigned int) strlen(dtmf); i++) {
if (strchr(terminators, dtmf[i])) {
*terminator = dtmf[i];
return SWITCH_STATUS_SUCCESS;
}
buf[x++] = dtmf[i];
buf[x] = '\0';
if (x >= buflen || x >= maxdigits) {
return SWITCH_STATUS_SUCCESS;
}
}
}
if (switch_channel_test_flag(channel, CF_SERVICE)) {
switch_yield(1000);
} else {
status = switch_core_session_read_frame(session, &read_frame, -1, 0);
if (!SWITCH_READ_ACCEPTABLE(status)) {
break;
}
}
}
return status;
}
SWITCH_DECLARE(switch_status_t) switch_ivr_record_file(switch_core_session_t *session,
switch_file_handle_t *fh,
char *file,
switch_input_callback_function_t input_callback,
void *buf,
unsigned int buflen)
{
switch_channel_t *channel;
char dtmf[128];
switch_file_handle_t lfh;
switch_frame_t *read_frame;
switch_codec_t codec, *read_codec;
char *codec_name;
switch_status_t status = SWITCH_STATUS_SUCCESS;
char *p;
const char *vval;
if (!fh) {
fh = &lfh;
}
memset(fh, 0, sizeof(*fh));
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
read_codec = switch_core_session_get_read_codec(session);
assert(read_codec != NULL);
fh->channels = read_codec->implementation->number_of_channels;
fh->samplerate = read_codec->implementation->samples_per_second;
if (switch_core_file_open(fh,
file,
SWITCH_FILE_FLAG_WRITE | SWITCH_FILE_DATA_SHORT,
switch_core_session_get_pool(session)) != SWITCH_STATUS_SUCCESS) {
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
switch_core_session_reset(session);
return SWITCH_STATUS_GENERR;
}
switch_channel_answer(channel);
if ((p = switch_channel_get_variable(channel, "RECORD_TITLE"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_TITLE, vval);
switch_channel_set_variable(channel, "RECORD_TITLE", NULL);
}
if ((p = switch_channel_get_variable(channel, "RECORD_COPYRIGHT"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_COPYRIGHT, vval);
switch_channel_set_variable(channel, "RECORD_COPYRIGHT", NULL);
}
if ((p = switch_channel_get_variable(channel, "RECORD_SOFTWARE"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_SOFTWARE, vval);
switch_channel_set_variable(channel, "RECORD_SOFTWARE", NULL);
}
if ((p = switch_channel_get_variable(channel, "RECORD_ARTIST"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_ARTIST, vval);
switch_channel_set_variable(channel, "RECORD_ARTIST", NULL);
}
if ((p = switch_channel_get_variable(channel, "RECORD_COMMENT"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_COMMENT, vval);
switch_channel_set_variable(channel, "RECORD_COMMENT", NULL);
}
if ((p = switch_channel_get_variable(channel, "RECORD_DATE"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_DATE, vval);
switch_channel_set_variable(channel, "RECORD_DATE", NULL);
}
codec_name = "L16";
if (switch_core_codec_init(&codec,
codec_name,
NULL,
read_codec->implementation->samples_per_second,
read_codec->implementation->microseconds_per_frame / 1000,
read_codec->implementation->number_of_channels,
SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE,
NULL, switch_core_session_get_pool(session)) == SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Raw Codec Activated\n");
switch_core_session_set_read_codec(session, &codec);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Raw Codec Activation Failed %s@%uhz %u channels %dms\n",
codec_name, fh->samplerate, fh->channels, read_codec->implementation->microseconds_per_frame / 1000);
switch_core_file_close(fh);
switch_core_session_reset(session);
return SWITCH_STATUS_GENERR;
}
while(switch_channel_ready(channel)) {
switch_size_t len;
switch_event_t *event;
if (switch_core_session_dequeue_private_event(session, &event) == SWITCH_STATUS_SUCCESS) {
switch_ivr_parse_event(session, event);
switch_event_destroy(&event);
}
if (input_callback || buf) {
/*
dtmf handler function you can hook up to be executed when a digit is dialed during playback
if you return anything but SWITCH_STATUS_SUCCESS the playback will stop.
*/
if (switch_channel_has_dtmf(channel)) {
switch_channel_dequeue_dtmf(channel, dtmf, sizeof(dtmf));
if (input_callback) {
status = input_callback(session, dtmf, SWITCH_INPUT_TYPE_DTMF, buf, buflen);
} else {
switch_copy_string((char *)buf, dtmf, buflen);
status = SWITCH_STATUS_BREAK;
}
}
if (input_callback) {
if (switch_core_session_dequeue_event(session, &event) == SWITCH_STATUS_SUCCESS) {
status = input_callback(session, event, SWITCH_INPUT_TYPE_EVENT, buf, buflen);
switch_event_destroy(&event);
}
}
if (status != SWITCH_STATUS_SUCCESS) {
break;
}
}
status = switch_core_session_read_frame(session, &read_frame, -1, 0);
if (!SWITCH_READ_ACCEPTABLE(status)) {
break;
}
if (!switch_test_flag(fh, SWITCH_FILE_PAUSE)) {
len = (switch_size_t) read_frame->datalen / 2;
switch_core_file_write(fh, read_frame->data, &len);
}
}
switch_core_session_set_read_codec(session, read_codec);
switch_core_file_close(fh);
switch_core_session_reset(session);
return status;
}
static void record_callback(switch_media_bug_t *bug, void *user_data, switch_abc_type_t type)
{
switch_file_handle_t *fh = (switch_file_handle_t *) user_data;
uint8_t data[SWITCH_RECCOMMENDED_BUFFER_SIZE];
switch_frame_t frame = {0};
frame.data = data;
frame.buflen = SWITCH_RECCOMMENDED_BUFFER_SIZE;
switch(type) {
case SWITCH_ABC_TYPE_INIT:
break;
case SWITCH_ABC_TYPE_CLOSE:
switch_core_file_close(fh);
case SWITCH_ABC_TYPE_READ:
if (fh) {
switch_size_t len;
if (switch_core_media_bug_read(bug, &frame) == SWITCH_STATUS_SUCCESS) {
len = (switch_size_t) frame.datalen / 2;
switch_core_file_write(fh, frame.data, &len);
}
}
break;
case SWITCH_ABC_TYPE_WRITE:
break;
}
}
SWITCH_DECLARE(switch_status_t) switch_ivr_stop_record_session(switch_core_session_t *session, char *file)
{
switch_media_bug_t *bug;
switch_channel_t *channel = switch_core_session_get_channel(session);
assert(channel != NULL);
if ((bug = switch_channel_get_private(channel, file))) {
switch_channel_set_private(channel, file, NULL);
switch_core_media_bug_remove(session, &bug);
return SWITCH_STATUS_SUCCESS;
}
return SWITCH_STATUS_FALSE;
}
SWITCH_DECLARE(switch_status_t) switch_ivr_record_session(switch_core_session_t *session, char *file, switch_file_handle_t *fh)
{
switch_channel_t *channel;
switch_codec_t *read_codec;
char *p;
const char *vval;
switch_media_bug_t *bug;
switch_status_t status;
if (!fh) {
if (!(fh = switch_core_session_alloc(session, sizeof(*fh)))) {
return SWITCH_STATUS_MEMERR;
}
}
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
read_codec = switch_core_session_get_read_codec(session);
assert(read_codec != NULL);
fh->channels = read_codec->implementation->number_of_channels;
fh->samplerate = read_codec->implementation->samples_per_second;
if (switch_core_file_open(fh,
file,
SWITCH_FILE_FLAG_WRITE | SWITCH_FILE_DATA_SHORT,
switch_core_session_get_pool(session)) != SWITCH_STATUS_SUCCESS) {
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
switch_core_session_reset(session);
return SWITCH_STATUS_GENERR;
}
switch_channel_answer(channel);
if ((p = switch_channel_get_variable(channel, "RECORD_TITLE"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_TITLE, vval);
switch_channel_set_variable(channel, "RECORD_TITLE", NULL);
}
if ((p = switch_channel_get_variable(channel, "RECORD_COPYRIGHT"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_COPYRIGHT, vval);
switch_channel_set_variable(channel, "RECORD_COPYRIGHT", NULL);
}
if ((p = switch_channel_get_variable(channel, "RECORD_SOFTWARE"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_SOFTWARE, vval);
switch_channel_set_variable(channel, "RECORD_SOFTWARE", NULL);
}
if ((p = switch_channel_get_variable(channel, "RECORD_ARTIST"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_ARTIST, vval);
switch_channel_set_variable(channel, "RECORD_ARTIST", NULL);
}
if ((p = switch_channel_get_variable(channel, "RECORD_COMMENT"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_COMMENT, vval);
switch_channel_set_variable(channel, "RECORD_COMMENT", NULL);
}
if ((p = switch_channel_get_variable(channel, "RECORD_DATE"))) {
vval = (const char *) switch_core_session_strdup(session, p);
switch_core_file_set_string(fh, SWITCH_AUDIO_COL_STR_DATE, vval);
switch_channel_set_variable(channel, "RECORD_DATE", NULL);
}
if ((status = switch_core_media_bug_add(session,
record_callback,
fh,
&bug)) != SWITCH_STATUS_SUCCESS) {
switch_core_file_close(fh);
return status;
}
switch_channel_set_private(channel, file, bug);
return SWITCH_STATUS_SUCCESS;
}
#define FILE_STARTSAMPLES 1024 * 32
#define FILE_BLOCKSIZE 1024 * 8
#define FILE_BUFSIZE 1024 * 64
SWITCH_DECLARE(switch_status_t) switch_ivr_play_file(switch_core_session_t *session,
switch_file_handle_t *fh,
char *file,
char *timer_name,
switch_input_callback_function_t input_callback,
void *buf,
unsigned int buflen)
{
switch_channel_t *channel;
int16_t abuf[FILE_STARTSAMPLES];
char dtmf[128];
uint32_t interval = 0, samples = 0, framelen;
uint32_t ilen = 0;
switch_size_t olen = 0;
switch_frame_t write_frame = {0};
switch_timer_t timer;
switch_core_thread_session_t thread_session;
switch_codec_t codec;
switch_memory_pool_t *pool = switch_core_session_get_pool(session);
char *codec_name;
int stream_id;
switch_status_t status = SWITCH_STATUS_SUCCESS;
switch_file_handle_t lfh;
switch_codec_t *read_codec = switch_core_session_get_read_codec(session);
const char *p;
char *title = "", *copyright = "", *software = "", *artist = "", *comment = "", *date = "";
uint8_t asis = 0;
char *ext;
if (file) {
if ((ext = strrchr(file, '.'))) {
ext++;
} else {
char *new_file;
uint32_t len;
ext = read_codec->implementation->iananame;
len = (uint32_t)strlen(file) + (uint32_t)strlen(ext) + 2;
new_file = switch_core_session_alloc(session, len);
snprintf(new_file, len, "%s.%s", file, ext);
file = new_file;
asis = 1;
}
}
if (!fh) {
fh = &lfh;
memset(fh, 0, sizeof(lfh));
}
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
if (switch_core_file_open(fh,
file,
SWITCH_FILE_FLAG_READ | SWITCH_FILE_DATA_SHORT,
switch_core_session_get_pool(session)) != SWITCH_STATUS_SUCCESS) {
switch_core_session_reset(session);
return SWITCH_STATUS_NOTFOUND;
}
write_frame.data = abuf;
write_frame.buflen = sizeof(abuf);
if (switch_core_file_get_string(fh, SWITCH_AUDIO_COL_STR_TITLE, &p) == SWITCH_STATUS_SUCCESS) {
title = (char *) switch_core_session_strdup(session, (char *)p);
switch_channel_set_variable(channel, "RECORD_TITLE", (char *)p);
}
if (switch_core_file_get_string(fh, SWITCH_AUDIO_COL_STR_COPYRIGHT, &p) == SWITCH_STATUS_SUCCESS) {
copyright = (char *) switch_core_session_strdup(session, (char *)p);
switch_channel_set_variable(channel, "RECORD_COPYRIGHT", (char *)p);
}
if (switch_core_file_get_string(fh, SWITCH_AUDIO_COL_STR_SOFTWARE, &p) == SWITCH_STATUS_SUCCESS) {
software = (char *) switch_core_session_strdup(session, (char *)p);
switch_channel_set_variable(channel, "RECORD_SOFTWARE", (char *)p);
}
if (switch_core_file_get_string(fh, SWITCH_AUDIO_COL_STR_ARTIST, &p) == SWITCH_STATUS_SUCCESS) {
artist = (char *) switch_core_session_strdup(session, (char *)p);
switch_channel_set_variable(channel, "RECORD_ARTIST", (char *)p);
}
if (switch_core_file_get_string(fh, SWITCH_AUDIO_COL_STR_COMMENT, &p) == SWITCH_STATUS_SUCCESS) {
comment = (char *) switch_core_session_strdup(session, (char *)p);
switch_channel_set_variable(channel, "RECORD_COMMENT", (char *)p);
}
if (switch_core_file_get_string(fh, SWITCH_AUDIO_COL_STR_DATE, &p) == SWITCH_STATUS_SUCCESS) {
date = (char *) switch_core_session_strdup(session, (char *)p);
switch_channel_set_variable(channel, "RECORD_DATE", (char *)p);
}
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
"OPEN FILE %s %uhz %u channels\n"
"TITLE=%s\n"
"COPYRIGHT=%s\n"
"SOFTWARE=%s\n"
"ARTIST=%s\n"
"COMMENT=%s\n"
"DATE=%s\n", file, fh->samplerate, fh->channels,
title,
copyright,
software,
artist,
comment,
date);
assert(read_codec != NULL);
interval = read_codec->implementation->microseconds_per_frame / 1000;
if (!fh->audio_buffer) {
switch_buffer_create_dynamic(&fh->audio_buffer, FILE_BLOCKSIZE, FILE_BUFSIZE, 0);
}
if (asis) {
write_frame.codec = read_codec;
samples = read_codec->implementation->samples_per_frame;
framelen = read_codec->implementation->encoded_bytes_per_frame;
} else {
codec_name = "L16";
if (switch_core_codec_init(&codec,
codec_name,
NULL,
fh->samplerate,
interval,
fh->channels,
SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE,
NULL, pool) == SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_DEBUG,
"Codec Activated %s@%uhz %u channels %dms\n",
codec_name,
fh->samplerate,
fh->channels,
interval);
write_frame.codec = &codec;
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Raw Codec Activation Failed %s@%uhz %u channels %dms\n",
codec_name, fh->samplerate, fh->channels, interval);
switch_core_file_close(fh);
switch_core_session_reset(session);
return SWITCH_STATUS_GENERR;
}
samples = codec.implementation->samples_per_frame;
framelen = codec.implementation->bytes_per_frame;
}
if (timer_name) {
uint32_t len;
len = samples * 2;
if (switch_core_timer_init(&timer, timer_name, interval, samples, pool) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "setup timer failed!\n");
switch_core_codec_destroy(&codec);
switch_core_file_close(fh);
switch_core_session_reset(session);
return SWITCH_STATUS_GENERR;
}
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "setup timer success %u bytes per %d ms!\n", len, interval);
}
write_frame.rate = fh->samplerate;
if (timer_name) {
/* start a thread to absorb incoming audio */
for (stream_id = 0; stream_id < switch_core_session_get_stream_count(session); stream_id++) {
switch_core_service_session(session, &thread_session, stream_id);
}
}
ilen = samples;
while(switch_channel_ready(channel)) {
int done = 0;
int do_speed = 1;
int last_speed = -1;
switch_event_t *event;
if (switch_core_session_dequeue_private_event(session, &event) == SWITCH_STATUS_SUCCESS) {
switch_ivr_parse_event(session, event);
switch_event_destroy(&event);
}
if (input_callback || buf) {
/*
dtmf handler function you can hook up to be executed when a digit is dialed during playback
if you return anything but SWITCH_STATUS_SUCCESS the playback will stop.
*/
if (switch_channel_has_dtmf(channel)) {
switch_channel_dequeue_dtmf(channel, dtmf, sizeof(dtmf));
if (input_callback) {
status = input_callback(session, dtmf, SWITCH_INPUT_TYPE_DTMF, buf, buflen);
} else {
switch_copy_string((char *)buf, dtmf, buflen);
status = SWITCH_STATUS_BREAK;
}
}
if (input_callback) {
if (switch_core_session_dequeue_event(session, &event) == SWITCH_STATUS_SUCCESS) {
status = input_callback(session, event, SWITCH_INPUT_TYPE_EVENT, buf, buflen);
switch_event_destroy(&event);
}
}
if (status != SWITCH_STATUS_SUCCESS) {
done = 1;
break;
}
}
if (switch_test_flag(fh, SWITCH_FILE_PAUSE)) {
memset(abuf, 0, framelen);
olen = ilen;
do_speed = 0;
} else if (fh->audio_buffer && (switch_buffer_inuse(fh->audio_buffer) > (switch_size_t)(framelen))) {
switch_buffer_read(fh->audio_buffer, abuf, framelen);
olen = asis ? framelen : ilen;
do_speed = 0;
} else {
olen = 32 * framelen;
switch_core_file_read(fh, abuf, &olen);
switch_buffer_write(fh->audio_buffer, abuf, asis ? olen : olen * 2);
olen = switch_buffer_read(fh->audio_buffer, abuf, framelen);
if (!asis) {
olen /= 2;
}
}
if (done || olen <= 0) {
break;
}
if (!asis) {
if (fh->speed > 2) {
fh->speed = 2;
} else if (fh->speed < -2) {
fh->speed = -2;
}
}
if (!asis && fh->audio_buffer && last_speed > -1 && last_speed != fh->speed) {
switch_buffer_zero(fh->audio_buffer);
}
if (!asis && fh->speed && do_speed) {
float factor = 0.25f * abs(fh->speed);
switch_size_t newlen, supplement, step;
short *bp = write_frame.data;
switch_size_t wrote = 0;
supplement = (int) (factor * olen);
newlen = (fh->speed > 0) ? olen - supplement : olen + supplement;
step = (fh->speed > 0) ? (newlen / supplement) : (olen / supplement);
while ((wrote + step) < newlen) {
switch_buffer_write(fh->audio_buffer, bp, step * 2);
wrote += step;
bp += step;
if (fh->speed > 0) {
bp++;
} else {
float f;
short s;
f = (float)(*bp + *(bp+1) + *(bp-1));
f /= 3;
s = (short) f;
switch_buffer_write(fh->audio_buffer, &s, 2);
wrote++;
}
}
if (wrote < newlen) {
switch_size_t r = newlen - wrote;
switch_buffer_write(fh->audio_buffer, bp, r*2);
wrote += r;
}
last_speed = fh->speed;
continue;
}
write_frame.datalen = (uint32_t)(olen * (asis ? 1 : 2));
write_frame.samples = (uint32_t)olen;
#ifndef WIN32
#if __BYTE_ORDER == __BIG_ENDIAN
if (!asis) {switch_swap_linear(write_frame.data, (int) write_frame.datalen / 2);}
#endif
#endif
for (stream_id = 0; stream_id < switch_core_session_get_stream_count(session); stream_id++) {
status = switch_core_session_write_frame(session, &write_frame, -1, stream_id);
if (status == SWITCH_STATUS_MORE_DATA) {
status = SWITCH_STATUS_SUCCESS;
continue;
} else if (status != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Bad Write\n");
done = 1;
break;
}
if (done) {
break;
}
}
if (timer_name) {
if (switch_core_timer_next(&timer) < 0) {
break;
}
} else { /* time off the channel (if you must) */
switch_frame_t *read_frame;
switch_status_t status;
while (switch_channel_test_flag(channel, CF_HOLD)) {
switch_yield(10000);
}
status = switch_core_session_read_frame(session, &read_frame, -1, 0);
if (!SWITCH_READ_ACCEPTABLE(status)) {
break;
}
}
}
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "done playing file\n");
switch_core_file_close(fh);
switch_buffer_destroy(&fh->audio_buffer);
if (!asis) {
switch_core_codec_destroy(&codec);
}
if (timer_name) {
/* End the audio absorbing thread */
switch_core_thread_session_end(&thread_session);
switch_core_timer_destroy(&timer);
}
switch_core_session_reset(session);
return status;
}
SWITCH_DECLARE(switch_status_t) switch_ivr_speak_text_handle(switch_core_session_t *session,
switch_speech_handle_t *sh,
switch_codec_t *codec,
switch_timer_t *timer,
switch_input_callback_function_t input_callback,
char *text,
void *buf,
unsigned int buflen)
{
switch_channel_t *channel;
short abuf[960];
char dtmf[128];
uint32_t len = 0;
switch_size_t ilen = 0;
switch_frame_t write_frame = {0};
int x;
int stream_id;
int done = 0;
int lead_in_out = 10;
switch_status_t status = SWITCH_STATUS_SUCCESS;
switch_speech_flag_t flags = SWITCH_SPEECH_FLAG_TTS;
uint32_t rate = 0, samples = 0;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
if (!sh) {
return SWITCH_STATUS_FALSE;
}
switch_channel_answer(channel);
write_frame.data = abuf;
write_frame.buflen = sizeof(abuf);
samples = (uint32_t)(sh->rate / 50);
len = samples * 2;
flags = 0;
switch_sleep(200000);
switch_core_speech_feed_tts(sh, text, &flags);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Speaking text: %s\n", text);
write_frame.rate = sh->rate;
memset(write_frame.data, 0, len);
write_frame.datalen = len;
write_frame.samples = len / 2;
write_frame.codec = codec;
for( x = 0; !done && x < lead_in_out; x++) {
for (stream_id = 0; stream_id < switch_core_session_get_stream_count(session); stream_id++) {
if (switch_core_session_write_frame(session, &write_frame, -1, stream_id) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Bad Write\n");
done = 1;
break;
}
}
}
ilen = len;
while(switch_channel_ready(channel)) {
switch_event_t *event;
if (switch_core_session_dequeue_private_event(session, &event) == SWITCH_STATUS_SUCCESS) {
switch_ivr_parse_event(session, event);
switch_event_destroy(&event);
}
if (input_callback || buf) {
/*
dtmf handler function you can hook up to be executed when a digit is dialed during playback
if you return anything but SWITCH_STATUS_SUCCESS the playback will stop.
*/
if (switch_channel_has_dtmf(channel)) {
if (buf && !strcasecmp(buf, "_break_")) {
status = SWITCH_STATUS_BREAK;
} else {
switch_channel_dequeue_dtmf(channel, dtmf, sizeof(dtmf));
if (input_callback) {
status = input_callback(session, dtmf, SWITCH_INPUT_TYPE_DTMF, buf, buflen);
} else {
switch_copy_string((char *)buf, dtmf, buflen);
status = SWITCH_STATUS_BREAK;
}
}
}
if (input_callback) {
if (switch_core_session_dequeue_event(session, &event) == SWITCH_STATUS_SUCCESS) {
status = input_callback(session, event, SWITCH_INPUT_TYPE_EVENT, buf, buflen);
switch_event_destroy(&event);
}
}
if (status != SWITCH_STATUS_SUCCESS) {
done = 1;
break;
}
}
if (switch_test_flag(sh, SWITCH_SPEECH_FLAG_PAUSE)) {
if (timer) {
if ((x = switch_core_timer_next(timer)) < 0) {
break;
}
} else {
switch_frame_t *read_frame;
switch_status_t status = switch_core_session_read_frame(session, &read_frame, -1, 0);
while (switch_channel_test_flag(channel, CF_HOLD)) {
switch_yield(10000);
}
if (!SWITCH_READ_ACCEPTABLE(status)) {
break;
}
}
continue;
}
flags = SWITCH_SPEECH_FLAG_BLOCKING;
status = switch_core_speech_read_tts(sh,
abuf,
&ilen,
&rate,
&flags);
if (status != SWITCH_STATUS_SUCCESS) {
for( x = 0; !done && x < lead_in_out; x++) {
for (stream_id = 0; stream_id < switch_core_session_get_stream_count(session); stream_id++) {
if (switch_core_session_write_frame(session, &write_frame, -1, stream_id) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Bad Write\n");
done = 1;
break;
}
}
}
if (status == SWITCH_STATUS_BREAK) {
status = SWITCH_STATUS_SUCCESS;
}
done = 1;
}
if (done) {
break;
}
write_frame.datalen = (uint32_t)ilen;
write_frame.samples = (uint32_t)(ilen / 2);
for (stream_id = 0; stream_id < switch_core_session_get_stream_count(session); stream_id++) {
if (switch_core_session_write_frame(session, &write_frame, -1, stream_id) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Bad Write\n");
done = 1;
break;
}
if (done) {
break;
}
}
if (timer) {
if ((x = switch_core_timer_next(timer)) < 0) {
break;
}
} else { /* time off the channel (if you must) */
switch_frame_t *read_frame;
switch_status_t status = switch_core_session_read_frame(session, &read_frame, -1, 0);
while (switch_channel_test_flag(channel, CF_HOLD)) {
switch_yield(10000);
}
if (!SWITCH_READ_ACCEPTABLE(status)) {
break;
}
}
}
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "done speaking text\n");
flags = 0;
switch_core_speech_flush_tts(sh);
return status;
}
SWITCH_DECLARE(switch_status_t) switch_ivr_speak_text(switch_core_session_t *session,
char *tts_name,
char *voice_name,
char *timer_name,
uint32_t rate,
switch_input_callback_function_t input_callback,
char *text,
void *buf,
unsigned int buflen)
{
switch_channel_t *channel;
int interval = 0;
uint32_t samples = 0;
uint32_t len = 0;
switch_frame_t write_frame = {0};
switch_timer_t timer;
switch_core_thread_session_t thread_session;
switch_codec_t codec;
switch_memory_pool_t *pool = switch_core_session_get_pool(session);
char *codec_name;
int stream_id;
switch_status_t status = SWITCH_STATUS_SUCCESS;
switch_speech_handle_t sh;
switch_speech_flag_t flags = SWITCH_SPEECH_FLAG_TTS;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
memset(&sh, 0, sizeof(sh));
if (switch_core_speech_open(&sh,
tts_name,
voice_name,
(unsigned int)rate,
&flags,
switch_core_session_get_pool(session)) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid TTS module!\n");
switch_core_session_reset(session);
return SWITCH_STATUS_FALSE;
}
switch_channel_answer(channel);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "OPEN TTS %s\n", tts_name);
interval = 20;
samples = (uint32_t)(rate / 50);
len = samples * 2;
codec_name = "L16";
if (switch_core_codec_init(&codec,
codec_name,
NULL,
(int)rate,
interval,
1,
SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE,
NULL, pool) == SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Raw Codec Activated\n");
write_frame.codec = &codec;
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Raw Codec Activation Failed %s@%uhz 1 channel %dms\n",
codec_name, rate, interval);
flags = 0;
switch_core_speech_close(&sh, &flags);
switch_core_session_reset(session);
return SWITCH_STATUS_GENERR;
}
if (timer_name) {
if (switch_core_timer_init(&timer, timer_name, interval, (int)samples, pool) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "setup timer failed!\n");
switch_core_codec_destroy(&codec);
flags = 0;
switch_core_speech_close(&sh, &flags);
switch_core_session_reset(session);
return SWITCH_STATUS_GENERR;
}
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "setup timer success %u bytes per %d ms!\n", len, interval);
/* start a thread to absorb incoming audio */
for (stream_id = 0; stream_id < switch_core_session_get_stream_count(session); stream_id++) {
switch_core_service_session(session, &thread_session, stream_id);
}
}
switch_ivr_speak_text_handle(session, &sh, &codec, timer_name ? &timer : NULL, input_callback, text, buf, buflen);
flags = 0;
switch_core_speech_close(&sh, &flags);
switch_core_codec_destroy(&codec);
if (timer_name) {
/* End the audio absorbing thread */
switch_core_thread_session_end(&thread_session);
switch_core_timer_destroy(&timer);
}
switch_core_session_reset(session);
return status;
}
/* Bridge Related Stuff*/
/*********************************************************************************/
struct audio_bridge_data {
switch_core_session_t *session_a;
switch_core_session_t *session_b;
int running;
};
static void *audio_bridge_thread(switch_thread_t *thread, void *obj)
{
switch_core_thread_session_t *his_thread, *data = obj;
int *stream_id_p;
int stream_id = 0, pre_b = 0, ans_a = 0, ans_b = 0, originator = 0;
switch_input_callback_function_t input_callback;
switch_core_session_message_t *message, msg = {0};
void *user_data;
switch_channel_t *chan_a, *chan_b;
switch_frame_t *read_frame;
switch_core_session_t *session_a, *session_b;
assert(!thread || thread);
session_a = data->objs[0];
session_b = data->objs[1];
stream_id_p = data->objs[2];
input_callback = (switch_input_callback_function_t) data->objs[3];
user_data = data->objs[4];
his_thread = data->objs[5];
if (stream_id_p) {
stream_id = *stream_id_p;
}
chan_a = switch_core_session_get_channel(session_a);
chan_b = switch_core_session_get_channel(session_b);
ans_a = switch_channel_test_flag(chan_a, CF_ANSWERED);
if ((originator = switch_channel_test_flag(chan_a, CF_ORIGINATOR))) {
pre_b = switch_channel_test_flag(chan_a, CF_EARLY_MEDIA);
ans_b = switch_channel_test_flag(chan_b, CF_ANSWERED);
}
switch_channel_set_flag(chan_a, CF_BRIDGED);
while (switch_channel_ready(chan_a) && data->running > 0 && his_thread->running > 0) {
switch_channel_state_t b_state = switch_channel_get_state(chan_b);
switch_status_t status;
switch_event_t *event;
switch (b_state) {
case CS_HANGUP:
case CS_DONE:
switch_mutex_lock(data->mutex);
data->running = -1;
switch_mutex_unlock(data->mutex);
continue;
default:
break;
}
if (switch_channel_test_flag(chan_a, CF_TRANSFER)) {
break;
}
if (switch_core_session_dequeue_private_event(session_a, &event) == SWITCH_STATUS_SUCCESS) {
switch_channel_set_flag(chan_b, CF_HOLD);
switch_ivr_parse_event(session_a, event);
switch_channel_clear_flag(chan_b, CF_HOLD);
switch_event_destroy(&event);
}
/* if 1 channel has DTMF pass it to the other */
if (switch_channel_has_dtmf(chan_a)) {
char dtmf[128];
switch_channel_dequeue_dtmf(chan_a, dtmf, sizeof(dtmf));
switch_core_session_send_dtmf(session_b, dtmf);
if (input_callback) {
if (input_callback(session_a, dtmf, SWITCH_INPUT_TYPE_DTMF, user_data, 0) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "%s ended call via DTMF\n", switch_channel_get_name(chan_a));
switch_mutex_lock(data->mutex);
data->running = -1;
switch_mutex_unlock(data->mutex);
break;
}
}
}
if (switch_core_session_dequeue_event(session_a, &event) == SWITCH_STATUS_SUCCESS) {
if (input_callback) {
status = input_callback(session_a, event, SWITCH_INPUT_TYPE_EVENT, user_data, 0);
}
if (switch_core_session_receive_event(session_b, &event) != SWITCH_STATUS_SUCCESS) {
switch_event_destroy(&event);
}
}
if (switch_core_session_dequeue_message(session_b, &message) == SWITCH_STATUS_SUCCESS) {
switch_core_session_receive_message(session_a, message);
if (switch_test_flag(message, SCSMF_DYNAMIC)) {
switch_safe_free(message);
}
}
if (!ans_a && originator) {
if (!ans_b && switch_channel_test_flag(chan_b, CF_ANSWERED)) {
switch_channel_answer(chan_a);
ans_a++;
} else if (!pre_b && switch_channel_test_flag(chan_b, CF_EARLY_MEDIA)) {
switch_channel_pre_answer(chan_a);
pre_b++;
}
switch_yield(10000);
continue;
}
/* read audio from 1 channel and write it to the other */
status = switch_core_session_read_frame(session_a, &read_frame, -1, stream_id);
if (SWITCH_READ_ACCEPTABLE(status)) {
if (status != SWITCH_STATUS_BREAK && !switch_channel_test_flag(chan_a, CF_HOLD)) {
if (switch_core_session_write_frame(session_b, read_frame, -1, stream_id) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "write: %s Bad Frame....[%u] Bubye!\n",
switch_channel_get_name(chan_b), read_frame->datalen);
switch_mutex_lock(data->mutex);
data->running = -1;
switch_mutex_unlock(data->mutex);
}
}
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "read: %s Bad Frame.... Bubye!\n", switch_channel_get_name(chan_a));
switch_mutex_lock(data->mutex);
data->running = -1;
switch_mutex_unlock(data->mutex);
}
}
msg.message_id = SWITCH_MESSAGE_INDICATE_UNBRIDGE;
msg.from = __FILE__;
switch_core_session_receive_message(session_a, &msg);
switch_channel_set_variable(chan_a, SWITCH_BRIDGE_VARIABLE, NULL);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "BRIDGE THREAD DONE [%s]\n", switch_channel_get_name(chan_a));
switch_channel_clear_flag(chan_a, CF_BRIDGED);
switch_mutex_lock(data->mutex);
data->running = 0;
switch_mutex_unlock(data->mutex);
return NULL;
}
static switch_status_t audio_bridge_on_loopback(switch_core_session_t *session)
{
switch_channel_t *channel = NULL;
void *arg;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
if ((arg = switch_channel_get_private(channel, "_bridge_"))) {
switch_channel_set_private(channel, "_bridge_", NULL);
audio_bridge_thread(NULL, (void *) arg);
} else {
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
}
switch_channel_clear_state_handler(channel, &audio_bridge_peer_state_handlers);
if (!switch_channel_test_flag(channel, CF_TRANSFER)) {
switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
}
return SWITCH_STATUS_FALSE;
}
static switch_status_t audio_bridge_on_ring(switch_core_session_t *session)
{
switch_channel_t *channel = NULL;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "CUSTOM RING\n");
/* put the channel in a passive state so we can loop audio to it */
switch_channel_set_state(channel, CS_HOLD);
return SWITCH_STATUS_FALSE;
}
static switch_status_t audio_bridge_on_hold(switch_core_session_t *session)
{
switch_channel_t *channel = NULL;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "CUSTOM HOLD\n");
/* put the channel in a passive state so we can loop audio to it */
return SWITCH_STATUS_FALSE;
}
static const switch_state_handler_table_t audio_bridge_peer_state_handlers = {
/*.on_init */ NULL,
/*.on_ring */ audio_bridge_on_ring,
/*.on_execute */ NULL,
/*.on_hangup */ NULL,
/*.on_loopback */ audio_bridge_on_loopback,
/*.on_transmit */ NULL,
/*.on_hold */ audio_bridge_on_hold,
};
static switch_status_t uuid_bridge_on_transmit(switch_core_session_t *session)
{
switch_channel_t *channel = NULL;
switch_core_session_t *other_session;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "CUSTOM TRANSMIT\n");
switch_channel_clear_state_handler(channel, NULL);
if (!switch_channel_test_flag(channel, CF_ORIGINATOR)) {
switch_channel_set_flag(channel, CF_TAGGED);
return SWITCH_STATUS_FALSE;
}
if ((other_session = switch_channel_get_private(channel, "_uuid_bridge_"))) {
switch_channel_t *other_channel = switch_core_session_get_channel(other_session);
switch_channel_state_t state = switch_channel_get_state(other_channel);
switch_channel_state_t mystate = switch_channel_get_state(channel);
switch_event_t *event;
uint8_t ready_a, ready_b;
switch_caller_profile_t *profile, *new_profile;
switch_channel_clear_flag(channel, CF_TRANSFER);
switch_channel_set_private(channel, "_uuid_bridge_", NULL);
while (mystate <= CS_HANGUP && state <= CS_HANGUP && !switch_channel_test_flag(other_channel, CF_TAGGED)) {
switch_yield(1000);
state = switch_channel_get_state(other_channel);
mystate = switch_channel_get_state(channel);
}
switch_channel_clear_flag(other_channel, CF_TRANSFER|CF_TAGGED);
switch_core_session_reset(session);
switch_core_session_reset(other_session);
ready_a = switch_channel_ready(channel);
ready_b = switch_channel_ready(other_channel);
if (!ready_a || !ready_b) {
if (!ready_a) {
switch_channel_hangup(other_channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
}
if (!ready_b) {
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
}
return SWITCH_STATUS_FALSE;
}
/* add another profile to both sessions for CDR's sake */
if ((profile = switch_channel_get_caller_profile(channel))) {
new_profile = switch_caller_profile_clone(session, profile);
new_profile->destination_number = switch_core_session_strdup(session, switch_core_session_get_uuid(other_session));
switch_channel_set_caller_profile(channel, new_profile);
}
if ((profile = switch_channel_get_caller_profile(other_channel))) {
new_profile = switch_caller_profile_clone(other_session, profile);
new_profile->destination_number = switch_core_session_strdup(other_session, switch_core_session_get_uuid(session));
switch_channel_set_caller_profile(other_channel, new_profile);
}
/* fire events that will change the data table from "show channels" */
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_EXECUTE) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(channel, event);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Application", "uuid_bridge");
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Application-Data", "%s", switch_core_session_get_uuid(other_session));
switch_event_fire(&event);
}
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_EXECUTE) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(other_channel, event);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Application", "uuid_bridge");
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Application-Data", "%s", switch_core_session_get_uuid(session));
switch_event_fire(&event);
}
switch_ivr_multi_threaded_bridge(session, other_session, NULL, NULL, NULL);
} else {
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
}
return SWITCH_STATUS_FALSE;
}
static const switch_state_handler_table_t uuid_bridge_state_handlers = {
/*.on_init */ NULL,
/*.on_ring */ NULL,
/*.on_execute */ NULL,
/*.on_hangup */ NULL,
/*.on_loopback */ NULL,
/*.on_transmit */ uuid_bridge_on_transmit,
/*.on_hold */ NULL
};
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
struct key_collect {
char *key;
char *file;
switch_core_session_t *session;
};
static void *SWITCH_THREAD_FUNC collect_thread_run(switch_thread_t *thread, void *obj)
{
struct key_collect *collect = (struct key_collect *) obj;
switch_channel_t *channel = switch_core_session_get_channel(collect->session);
char buf[10] = "";
char *p, term;
if (!strcasecmp(collect->key, "exec")) {
char *data;
const switch_application_interface_t *application_interface;
char *app_name, *app_data;
if (!(data = collect->file)) {
goto wbreak;
}
app_name = data;
if ((app_data = strchr(app_name, ' '))) {
*app_data++ = '\0';
}
if ((application_interface = switch_loadable_module_get_application_interface(app_name)) == 0) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid Application %s\n", app_name);
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
goto wbreak;
}
if (!application_interface->application_function) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "No Function for %s\n", app_name);
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
goto wbreak;
}
application_interface->application_function(collect->session, app_data);
if (switch_channel_get_state(channel) < CS_HANGUP) {
switch_channel_set_flag(channel, CF_WINNER);
}
goto wbreak;
}
if (!switch_channel_ready(channel)) {
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
goto wbreak;
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
while(switch_channel_ready(channel)) {
memset(buf, 0, sizeof(buf));
if (collect->file) {
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
switch_ivr_play_file(collect->session, NULL, collect->file, NULL, NULL, buf, sizeof(buf));
} else {
switch_ivr_collect_digits_count(collect->session, buf, sizeof(buf), 1, "", &term, 0);
}
for(p = buf; *p; p++) {
if (*collect->key == *p) {
switch_channel_set_flag(channel, CF_WINNER);
goto wbreak;
}
}
}
wbreak:
return NULL;
}
static void launch_collect_thread(struct key_collect *collect)
{
switch_thread_t *thread;
switch_threadattr_t *thd_attr = NULL;
switch_threadattr_create(&thd_attr, switch_core_session_get_pool(collect->session));
switch_threadattr_detach_set(thd_attr, 1);
switch_threadattr_stacksize_set(thd_attr, SWITCH_THREAD_STACKSIZE);
switch_thread_create(&thread, thd_attr, collect_thread_run, collect, switch_core_session_get_pool(collect->session));
}
static uint8_t check_channel_status(switch_channel_t **peer_channels,
switch_core_session_t **peer_sessions,
uint32_t len,
int32_t *idx,
char *file,
char *key)
{
uint32_t i;
uint32_t hups = 0;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
*idx = -1;
for (i = 0; i < len; i++) {
if (!peer_channels[i]) {
continue;
}
if (switch_channel_get_state(peer_channels[i]) >= CS_HANGUP) {
hups++;
} else if ((switch_channel_test_flag(peer_channels[i], CF_ANSWERED) || (len == 1 && switch_channel_test_flag(peer_channels[0], CF_EARLY_MEDIA))) &&
!switch_channel_test_flag(peer_channels[i], CF_TAGGED)) {
if (key) {
struct key_collect *collect;
if ((collect = switch_core_session_alloc(peer_sessions[i], sizeof(*collect)))) {
switch_channel_set_flag(peer_channels[i], CF_TAGGED);
collect->key = key;
if (file) {
collect->file = switch_core_session_strdup(peer_sessions[i], file);
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
collect->session = peer_sessions[i];
launch_collect_thread(collect);
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
} else {
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
*idx = i;
return 0;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
} else if (switch_channel_test_flag(peer_channels[i], CF_WINNER)) {
*idx = i;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
return 0;
}
}
if (hups == len) {
return 0;
} else {
return 1;
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
#define MAX_PEERS 256
SWITCH_DECLARE(switch_status_t) switch_ivr_originate(switch_core_session_t *session,
switch_core_session_t **bleg,
switch_call_cause_t *cause,
char *bridgeto,
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
uint32_t timelimit_sec,
const switch_state_handler_table_t *table,
char *cid_name_override,
char *cid_num_override,
switch_caller_profile_t *caller_profile_override
)
{
char *pipe_names[MAX_PEERS] = {0};
char *data = NULL;
switch_status_t status = SWITCH_STATUS_SUCCESS;
switch_channel_t *caller_channel = NULL;
char *peer_names[MAX_PEERS] = {0};
switch_core_session_t *peer_session, *peer_sessions[MAX_PEERS] = {0};
switch_caller_profile_t *caller_profiles[MAX_PEERS] = {0}, *caller_caller_profile;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
char *chan_type = NULL, *chan_data;
switch_channel_t *peer_channel = NULL, *peer_channels[MAX_PEERS] = {0};
time_t start;
switch_frame_t *read_frame = NULL;
switch_memory_pool_t *pool = NULL;
int r = 0, i, and_argc = 0, or_argc = 0;
int32_t idx = -1;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
switch_codec_t write_codec = {0};
switch_frame_t write_frame = {0};
uint8_t err = 0, fdata[1024], pass = 0;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
char *file = NULL, *key = NULL, *odata, *var;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
write_frame.data = fdata;
*bleg = NULL;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
odata = strdup(bridgeto);
data = odata;
if (!strncasecmp(data, "confirm=", 8)) {
data += 8;
file = data;
if ((data = strchr(file, ';'))) {
*data++ = '\0';
if ((key = strchr(file, ':'))) {
*key++ = '\0';
} else {
err++;
}
} else {
err++;
}
}
if (err) {
status = SWITCH_STATUS_GENERR;
goto done;
}
if (session) {
caller_channel = switch_core_session_get_channel(session);
assert(caller_channel != NULL);
switch_channel_set_variable(caller_channel, "originate_disposition", "failure");
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
if ((var = switch_channel_get_variable(caller_channel, "group_confirm_key"))) {
key = switch_core_session_strdup(session, var);
if ((var = switch_channel_get_variable(caller_channel, "group_confirm_file"))) {
file = switch_core_session_strdup(session, var);
}
}
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
if (file && !strcmp(file, "undef")) {
file = NULL;
}
or_argc = switch_separate_string(data, '|', pipe_names, (sizeof(pipe_names) / sizeof(pipe_names[0])));
for (r = 0; r < or_argc; r++) {
memset(peer_names, 0, sizeof(peer_names));
peer_session = NULL;
memset(peer_sessions, 0, sizeof(peer_sessions));
memset(peer_channels, 0, sizeof(peer_channels));
memset(caller_profiles, 0, sizeof(caller_profiles));
chan_type = NULL;
chan_data = NULL;
peer_channel = NULL;
start = 0;
read_frame = NULL;
pool = NULL;
pass = 0;
file = NULL;
key = NULL;
var = NULL;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
and_argc = switch_separate_string(pipe_names[r], '&', peer_names, (sizeof(peer_names) / sizeof(peer_names[0])));
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
for (i = 0; i < and_argc; i++) {
chan_type = peer_names[i];
if ((chan_data = strchr(chan_type, '/')) != 0) {
*chan_data = '\0';
chan_data++;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
if (session) {
if (!switch_channel_ready(caller_channel)) {
status = SWITCH_STATUS_FALSE;
goto done;
}
caller_caller_profile = caller_profile_override ? caller_profile_override : switch_channel_get_caller_profile(caller_channel);
if (!cid_name_override) {
cid_name_override = caller_caller_profile->caller_id_name;
}
if (!cid_num_override) {
cid_num_override = caller_caller_profile->caller_id_number;
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
caller_profiles[i] = switch_caller_profile_new(switch_core_session_get_pool(session),
caller_caller_profile->username,
caller_caller_profile->dialplan,
cid_name_override,
cid_num_override,
caller_caller_profile->network_addr,
NULL,
NULL,
caller_caller_profile->rdnis,
caller_caller_profile->source,
caller_caller_profile->context,
chan_data);
caller_profiles[i]->flags = caller_caller_profile->flags;
pool = NULL;
} else {
if (!cid_name_override) {
cid_name_override = "FreeSWITCH";
}
if (!cid_num_override) {
cid_num_override = "0000000000";
}
if (switch_core_new_memory_pool(&pool) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "OH OH no pool\n");
status = SWITCH_STATUS_TERM;
goto done;
}
if (caller_profile_override) {
caller_profiles[i] = switch_caller_profile_new(pool,
caller_profile_override->username,
caller_profile_override->dialplan,
caller_profile_override->caller_id_name,
caller_profile_override->caller_id_number,
caller_profile_override->network_addr,
caller_profile_override->ani,
caller_profile_override->aniii,
caller_profile_override->rdnis,
caller_profile_override->source,
caller_profile_override->context,
chan_data);
} else {
caller_profiles[i] = switch_caller_profile_new(pool,
NULL,
NULL,
cid_name_override,
cid_num_override,
NULL,
NULL,
NULL,
NULL,
__FILE__,
NULL,
chan_data);
}
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
if (switch_core_session_outgoing_channel(session, chan_type, caller_profiles[i], &peer_sessions[i], pool) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Cannot Create Outgoing Channel!\n");
if (pool) {
switch_core_destroy_memory_pool(&pool);
}
caller_profiles[i] = NULL;
peer_channels[i] = NULL;
peer_sessions[i] = NULL;
continue;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
switch_core_session_read_lock(peer_sessions[i]);
pool = NULL;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
peer_channels[i] = switch_core_session_get_channel(peer_sessions[i]);
assert(peer_channels[i] != NULL);
if (table == &noop_state_handler) {
table = NULL;
} else if (!table) {
table = &audio_bridge_peer_state_handlers;
}
if (table) {
switch_channel_add_state_handler(peer_channels[i], table);
}
if (switch_core_session_running(peer_sessions[i])) {
switch_channel_set_state(peer_channels[i], CS_RING);
} else {
switch_core_session_thread_launch(peer_sessions[i]);
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
time(&start);
for (;;) {
uint32_t valid_channels = 0;
for (i = 0; i < and_argc; i++) {
int state;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
if (!peer_channels[i]) {
continue;
}
valid_channels++;
state = switch_channel_get_state(peer_channels[i]);
if (state >= CS_RING) {
goto endfor1;
}
if (caller_channel && !switch_channel_ready(caller_channel)) {
break;
}
if ((time(NULL) - start) > (time_t)timelimit_sec) {
break;
}
switch_yield(1000);
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
if (valid_channels == 0) {
status = SWITCH_STATUS_GENERR;
goto done;
}
}
endfor1:
if (session && !switch_channel_test_flag(caller_channel, CF_NOMEDIA)) {
switch_codec_t *read_codec = NULL;
read_codec = switch_core_session_get_read_codec(session);
assert(read_codec != NULL);
if (!(pass = (uint8_t)switch_test_flag(read_codec, SWITCH_CODEC_FLAG_PASSTHROUGH))) {
if (switch_core_codec_init(&write_codec,
"L16",
NULL,
read_codec->implementation->samples_per_second,
read_codec->implementation->microseconds_per_frame / 1000,
1,
SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE,
NULL,
pool) == SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Raw Codec Activation Success L16@%uhz 1 channel %dms\n",
read_codec->implementation->samples_per_second,
read_codec->implementation->microseconds_per_frame / 1000);
write_frame.codec = &write_codec;
write_frame.datalen = read_codec->implementation->bytes_per_frame;
write_frame.samples = write_frame.datalen / 2;
memset(write_frame.data, 255, write_frame.datalen);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Codec Error!");
switch_channel_hangup(caller_channel, SWITCH_CAUSE_NORMAL_TEMPORARY_FAILURE);
}
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
while ((!caller_channel || switch_channel_ready(caller_channel)) &&
check_channel_status(peer_channels, peer_sessions, and_argc, &idx, file, key) && ((time(NULL) - start) < (time_t)timelimit_sec)) {
if (session && or_argc == 1 && and_argc == 1) { /* when there is only 1 channel to call and bridge */
switch_core_session_message_t *message = NULL;
if (switch_core_session_dequeue_message(peer_sessions[0], &message) == SWITCH_STATUS_SUCCESS) {
switch_core_session_receive_message(session, message);
if (switch_test_flag(message, SCSMF_DYNAMIC)) {
switch_safe_free(message);
}
}
}
/* read from the channel while we wait if the audio is up on it */
if (session && !switch_channel_test_flag(caller_channel, CF_NOMEDIA) &&
(switch_channel_test_flag(caller_channel, CF_ANSWERED) || switch_channel_test_flag(caller_channel, CF_EARLY_MEDIA))) {
switch_status_t status = switch_core_session_read_frame(session, &read_frame, 1000, 0);
if (!SWITCH_READ_ACCEPTABLE(status)) {
break;
}
if (read_frame && !pass) {
if (switch_core_session_write_frame(session, &write_frame, 1000, 0) != SWITCH_STATUS_SUCCESS) {
break;
}
}
} else {
switch_yield(1000);
}
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
if (session && !switch_channel_test_flag(caller_channel, CF_NOMEDIA)) {
switch_core_session_reset(session);
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
for (i = 0; i < and_argc; i++) {
if (!peer_channels[i]) {
continue;
}
if (i != idx) {
switch_channel_hangup(peer_channels[i], SWITCH_CAUSE_LOSE_RACE);
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
if (idx > -1) {
peer_session = peer_sessions[idx];
peer_channel = peer_channels[idx];
} else {
status = SWITCH_STATUS_FALSE;
goto done;
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
if (caller_channel) {
char *val;
if (switch_channel_test_flag(peer_channel, CF_NOMEDIA) && (val = switch_channel_get_variable(peer_channel, SWITCH_R_SDP_VARIABLE))) {
switch_channel_set_variable(caller_channel, SWITCH_L_SDP_VARIABLE, val);
}
if (switch_channel_test_flag(peer_channel, CF_ANSWERED)) {
switch_channel_answer(caller_channel);
} else if (switch_channel_test_flag(peer_channel, CF_EARLY_MEDIA)) {
switch_channel_pre_answer(caller_channel);
}
}
if (switch_channel_test_flag(peer_channel, CF_ANSWERED) || switch_channel_test_flag(peer_channel, CF_EARLY_MEDIA)) {
*bleg = peer_session;
status = SWITCH_STATUS_SUCCESS;
} else {
status = SWITCH_STATUS_FALSE;
}
done:
*cause = SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER;
if (status == SWITCH_STATUS_SUCCESS) {
if (caller_channel) {
switch_channel_set_variable(caller_channel, "originate_disposition", "call accepted");
}
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Originate Resulted in Success: [%s]\n", switch_channel_get_name(peer_channel));
} else {
if (peer_channel) {
*cause = switch_channel_get_cause(peer_channel);
} else {
for (i = 0; i < and_argc; i++) {
if (!peer_channels[i]) {
continue;
}
*cause = switch_channel_get_cause(peer_channels[i]);
break;
}
}
if (caller_channel) {
switch_channel_set_variable(caller_channel, "originate_disposition", switch_channel_cause2str(*cause));
}
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Originate Resulted in Error Cause: %d [%s]\n", *cause, switch_channel_cause2str(*cause));
}
if (!pass && write_codec.implementation) {
switch_core_codec_destroy(&write_codec);
}
for (i = 0; i < and_argc; i++) {
if (!peer_channels[i]) {
continue;
}
switch_core_session_rwunlock(peer_sessions[i]);
}
if (status == SWITCH_STATUS_SUCCESS) {
break;
}
}
switch_safe_free(odata);
return status;
}
static switch_status_t signal_bridge_on_hangup(switch_core_session_t *session)
{
char *uuid;
switch_channel_t *channel = NULL;
switch_core_session_t *other_session;
switch_event_t *event;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
if (switch_channel_test_flag(channel, CF_ORIGINATOR)) {
switch_channel_clear_flag(channel, CF_ORIGINATOR);
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_UNBRIDGE) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(channel, event);
switch_event_fire(&event);
}
}
if ((uuid = switch_channel_get_variable(channel, SWITCH_SIGNAL_BRIDGE_VARIABLE)) && (other_session = switch_core_session_locate(uuid))) {
switch_channel_t *other_channel = NULL;
other_channel = switch_core_session_get_channel(other_session);
assert(other_channel != NULL);
switch_channel_hangup(other_channel, switch_channel_get_cause(channel));
switch_core_session_rwunlock(other_session);
}
return SWITCH_STATUS_SUCCESS;
}
static const switch_state_handler_table_t signal_bridge_state_handlers = {
/*.on_init */ NULL,
/*.on_ring */ NULL,
/*.on_execute */ NULL,
/*.on_hangup */ signal_bridge_on_hangup,
/*.on_loopback */ NULL,
/*.on_transmit */ NULL,
/*.on_hold */ NULL
};
SWITCH_DECLARE(switch_status_t) switch_ivr_signal_bridge(switch_core_session_t *session, switch_core_session_t *peer_session)
{
switch_channel_t *caller_channel, *peer_channel;
switch_event_t *event;
caller_channel = switch_core_session_get_channel(session);
assert(caller_channel != NULL);
switch_channel_set_flag(caller_channel, CF_ORIGINATOR);
peer_channel = switch_core_session_get_channel(peer_session);
assert(peer_channel != NULL);
switch_channel_set_variable(caller_channel, SWITCH_SIGNAL_BRIDGE_VARIABLE, switch_core_session_get_uuid(peer_session));
switch_channel_set_variable(peer_channel, SWITCH_SIGNAL_BRIDGE_VARIABLE, switch_core_session_get_uuid(session));
switch_channel_clear_state_handler(caller_channel, NULL);
switch_channel_clear_state_handler(peer_channel, NULL);
switch_channel_add_state_handler(caller_channel, &signal_bridge_state_handlers);
switch_channel_add_state_handler(peer_channel, &signal_bridge_state_handlers);
/* fire events that will change the data table from "show channels" */
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_EXECUTE) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(caller_channel, event);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Application", "signal_bridge");
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Application-Data", "%s", switch_core_session_get_uuid(peer_session));
switch_event_fire(&event);
}
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_EXECUTE) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(peer_channel, event);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Application", "signal_bridge");
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Application-Data", "%s", switch_core_session_get_uuid(session));
switch_event_fire(&event);
}
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_BRIDGE) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(caller_channel, event);
switch_event_fire(&event);
}
switch_channel_set_state(caller_channel, CS_TRANSMIT);
switch_channel_set_state(peer_channel, CS_TRANSMIT);
return SWITCH_STATUS_SUCCESS;
}
SWITCH_DECLARE(switch_status_t) switch_ivr_multi_threaded_bridge(switch_core_session_t *session,
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
switch_core_session_t *peer_session,
switch_input_callback_function_t input_callback,
void *session_data,
void *peer_session_data)
{
switch_core_thread_session_t *this_audio_thread, *other_audio_thread;
switch_channel_t *caller_channel, *peer_channel;
int stream_id = 0;
switch_status_t status = SWITCH_STATUS_SUCCESS;
caller_channel = switch_core_session_get_channel(session);
assert(caller_channel != NULL);
switch_channel_set_flag(caller_channel, CF_ORIGINATOR);
peer_channel = switch_core_session_get_channel(peer_session);
assert(peer_channel != NULL);
other_audio_thread = switch_core_session_alloc(peer_session, sizeof(switch_core_thread_session_t));
this_audio_thread = switch_core_session_alloc(peer_session, sizeof(switch_core_thread_session_t));
other_audio_thread->objs[0] = session;
other_audio_thread->objs[1] = peer_session;
other_audio_thread->objs[2] = &stream_id;
other_audio_thread->objs[3] = (void *) input_callback;
other_audio_thread->objs[4] = session_data;
other_audio_thread->objs[5] = this_audio_thread;
other_audio_thread->running = 5;
switch_mutex_init(&other_audio_thread->mutex, SWITCH_MUTEX_NESTED, switch_core_session_get_pool(session));
this_audio_thread->objs[0] = peer_session;
this_audio_thread->objs[1] = session;
this_audio_thread->objs[2] = &stream_id;
this_audio_thread->objs[3] = (void *) input_callback;
this_audio_thread->objs[4] = peer_session_data;
this_audio_thread->objs[5] = other_audio_thread;
this_audio_thread->running = 2;
switch_mutex_init(&this_audio_thread->mutex, SWITCH_MUTEX_NESTED, switch_core_session_get_pool(peer_session));
switch_channel_add_state_handler(peer_channel, &audio_bridge_peer_state_handlers);
if (switch_channel_test_flag(peer_channel, CF_ANSWERED) && !switch_channel_test_flag(caller_channel, CF_ANSWERED)) {
switch_channel_answer(caller_channel);
}
if (switch_channel_test_flag(peer_channel, CF_ANSWERED) || switch_channel_test_flag(peer_channel, CF_EARLY_MEDIA)) {
switch_event_t *event;
switch_core_session_message_t msg = {0};
switch_channel_set_state(peer_channel, CS_HOLD);
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_BRIDGE) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(caller_channel, event);
switch_event_fire(&event);
}
if (switch_core_session_read_lock(peer_session) == SWITCH_STATUS_SUCCESS) {
switch_channel_set_variable(caller_channel, SWITCH_BRIDGE_VARIABLE, switch_core_session_get_uuid(peer_session));
switch_channel_set_variable(peer_channel, SWITCH_BRIDGE_VARIABLE, switch_core_session_get_uuid(session));
msg.message_id = SWITCH_MESSAGE_INDICATE_BRIDGE;
msg.from = __FILE__;
msg.pointer_arg = session;
switch_core_session_receive_message(peer_session, &msg);
if (!msg.pointer_arg) {
status = SWITCH_STATUS_FALSE;
switch_core_session_rwunlock(peer_session);
goto done;
}
msg.pointer_arg = peer_session;
switch_core_session_receive_message(session, &msg);
if (!msg.pointer_arg) {
status = SWITCH_STATUS_FALSE;
switch_core_session_rwunlock(peer_session);
goto done;
}
switch_channel_set_private(peer_channel, "_bridge_", other_audio_thread);
switch_channel_set_state(peer_channel, CS_LOOPBACK);
audio_bridge_thread(NULL, (void *) this_audio_thread);
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_UNBRIDGE) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(caller_channel, event);
switch_event_fire(&event);
}
if (switch_channel_get_state(caller_channel) != CS_EXECUTE && !switch_channel_test_flag(caller_channel, CF_TRANSFER)) {
switch_channel_hangup(caller_channel, SWITCH_CAUSE_NORMAL_CLEARING);
}
this_audio_thread->objs[0] = NULL;
this_audio_thread->objs[1] = NULL;
this_audio_thread->objs[2] = NULL;
this_audio_thread->objs[3] = NULL;
this_audio_thread->objs[4] = NULL;
this_audio_thread->objs[5] = NULL;
switch_mutex_lock(this_audio_thread->mutex);
this_audio_thread->running = 0;
switch_mutex_unlock(this_audio_thread->mutex);
switch_channel_clear_flag(caller_channel, CF_ORIGINATOR);
if (other_audio_thread->running > 0) {
switch_mutex_lock(other_audio_thread->mutex);
other_audio_thread->running = -1;
switch_mutex_unlock(other_audio_thread->mutex);
while (other_audio_thread->running) {
switch_yield(1000);
}
}
switch_core_session_rwunlock(peer_session);
} else {
status = SWITCH_STATUS_FALSE;
}
} else {
status = SWITCH_STATUS_FALSE;
}
if (status != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Bridge Failed %s->%s\n",
switch_channel_get_name(caller_channel),
switch_channel_get_name(peer_channel)
);
switch_channel_hangup(peer_channel, SWITCH_CAUSE_NO_ANSWER);
}
done:
return status;
}
SWITCH_DECLARE(switch_status_t) switch_ivr_uuid_bridge(char *originator_uuid, char *originatee_uuid)
{
switch_core_session_t *originator_session, *originatee_session;
switch_channel_t *originator_channel, *originatee_channel;
switch_status_t status = SWITCH_STATUS_FALSE;
if ((originator_session = switch_core_session_locate(originator_uuid))) {
if ((originatee_session = switch_core_session_locate(originatee_uuid))) {
originator_channel = switch_core_session_get_channel(originator_session);
originatee_channel = switch_core_session_get_channel(originatee_session);
/* override transmit state for originator_channel to bridge to originatee_channel
* install pointer to originatee_session into originator_channel
* set CF_TRANSFER on both channels and change state to CS_TRANSMIT to
* inturrupt anything they are already doing.
* originatee_session will fall asleep and originator_session will bridge to it
*/
switch_channel_clear_state_handler(originator_channel, NULL);
switch_channel_clear_state_handler(originatee_channel, NULL);
switch_channel_set_flag(originator_channel, CF_ORIGINATOR);
switch_channel_add_state_handler(originator_channel, &uuid_bridge_state_handlers);
switch_channel_add_state_handler(originatee_channel, &uuid_bridge_state_handlers);
switch_channel_set_private(originator_channel, "_uuid_bridge_", originatee_session);
/* switch_channel_set_state_flag sets flags you want to be set when the next stat change happens */
switch_channel_set_state_flag(originator_channel, CF_TRANSFER);
switch_channel_set_state_flag(originatee_channel, CF_TRANSFER);
/* release the read locks we have on the channels */
switch_core_session_rwunlock(originator_session);
switch_core_session_rwunlock(originatee_session);
/* change the states and let the chips fall where they may */
switch_channel_set_state(originator_channel, CS_TRANSMIT);
switch_channel_set_state(originatee_channel, CS_TRANSMIT);
status = SWITCH_STATUS_SUCCESS;
} else {
switch_core_session_rwunlock(originator_session);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "no channel for originatee uuid %s\n", originatee_uuid);
}
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "no channel for originator uuid %s\n", originator_uuid);
}
return status;
}
SWITCH_DECLARE(switch_status_t) switch_ivr_session_transfer(switch_core_session_t *session, char *extension, char *dialplan, char *context)
{
switch_channel_t *channel;
switch_caller_profile_t *profile, *new_profile;
switch_core_session_message_t msg = {0};
assert(session != NULL);
assert(extension != NULL);
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
if ((profile = switch_channel_get_caller_profile(channel))) {
new_profile = switch_caller_profile_clone(session, profile);
new_profile->destination_number = switch_core_session_strdup(session, extension);
if (dialplan) {
new_profile->dialplan = switch_core_session_strdup(session, dialplan);
} else {
dialplan = new_profile->dialplan;
}
if (context) {
new_profile->context = switch_core_session_strdup(session, context);
} else {
context = new_profile->context;
}
switch_channel_set_caller_profile(channel, new_profile);
switch_channel_set_flag(channel, CF_TRANSFER);
switch_channel_set_state(channel, CS_RING);
msg.message_id = SWITCH_MESSAGE_INDICATE_TRANSFER;
msg.from = __FILE__;
switch_core_session_receive_message(session, &msg);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Transfer %s to %s[%s@%s]\n",
switch_channel_get_name(channel), dialplan, extension, context);
return SWITCH_STATUS_SUCCESS;
}
return SWITCH_STATUS_FALSE;
}
SWITCH_DECLARE(switch_status_t) switch_ivr_transfer_variable(switch_core_session_t *sessa, switch_core_session_t *sessb, char *var)
{
switch_channel_t *chana = switch_core_session_get_channel(sessa);
switch_channel_t *chanb = switch_core_session_get_channel(sessb);
char *val = NULL;
if (var) {
if ((val = switch_channel_get_variable(chana, var))) {
switch_channel_set_variable(chanb, var, val);
}
} else {
switch_hash_index_t *hi;
void *vval;
const void *vvar;
for (hi = switch_channel_variable_first(chana, switch_core_session_get_pool(sessa)); hi; hi = switch_hash_next(hi)) {
switch_hash_this(hi, &vvar, NULL, &vval);
if (vvar && vval) {
switch_channel_set_variable(chanb, (char *) vvar, (char *) vval);
}
}
}
return SWITCH_STATUS_SUCCESS;
}