/*! \brief SpeechDestroy() Dialplan Application */
static int speech_destroy(struct ast_channel *chan, void *data)
{
int res = 0;
struct ast_module_user *u = NULL;
struct ast_speech *speech = find_speech(chan);
struct ast_datastore *datastore = NULL;
u = ast_module_user_add(chan);
if (speech == NULL) {
ast_module_user_remove(u);
return -1;
}
/* Destroy speech structure */
ast_speech_destroy(speech);
datastore = ast_channel_datastore_find(chan, &speech_datastore, NULL);
if (datastore != NULL) {
ast_channel_datastore_remove(chan, datastore);
}
ast_module_user_remove(u);
return res;
}
static int srv_query_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct ast_datastore *datastore;
if (!chan) {
ast_log(LOG_WARNING, "%s cannot be used without a channel\n", cmd);
return -1;
}
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "%s requires a service as an argument\n", cmd);
return -1;
}
/* If they already called SRVQUERY for this service once,
* we need to kill the old datastore.
*/
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &srv_result_datastore_info, data);
ast_channel_unlock(chan);
if (datastore) {
ast_channel_datastore_remove(chan, datastore);
ast_datastore_free(datastore);
}
if (!srv_datastore_setup(data, chan)) {
return -1;
}
ast_copy_string(buf, data, len);
return 0;
}
/*!
* \brief Dial timeout
*
* This is a bridge interval hook callback. The interval hook triggering
* means that the dial timeout has been reached. If the channel has not
* been answered by the time this callback is called, then the channel
* is hung up
*
* \param bridge_channel Bridge channel on which interval hook has been called
* \param ignore Ignored
* \return -1 (i.e. remove the interval hook)
*/
static int bridge_timeout(struct ast_bridge_channel *bridge_channel, void *ignore)
{
struct ast_datastore *datastore;
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = stasis_app_control_find_by_channel(bridge_channel->chan);
ast_channel_lock(bridge_channel->chan);
if (ast_channel_state(bridge_channel->chan) != AST_STATE_UP) {
/* Don't bother removing the datastore because it will happen when the channel is hung up */
ast_channel_unlock(bridge_channel->chan);
stasis_app_send_command_async(control, hangup_channel, NULL, NULL);
return -1;
}
datastore = ast_channel_datastore_find(bridge_channel->chan, &timeout_datastore, NULL);
if (!datastore) {
ast_channel_unlock(bridge_channel->chan);
return -1;
}
ast_channel_datastore_remove(bridge_channel->chan, datastore);
ast_channel_unlock(bridge_channel->chan);
ast_datastore_free(datastore);
return -1;
}
static struct chanspy_ds *chanspy_ds_free(struct chanspy_ds *chanspy_ds)
{
if (!chanspy_ds)
return NULL;
ast_mutex_lock(&chanspy_ds->lock);
if (chanspy_ds->chan) {
struct ast_datastore *datastore;
struct ast_channel *chan;
chan = chanspy_ds->chan;
ast_channel_lock(chan);
if ((datastore = ast_channel_datastore_find(chan, &chanspy_ds_info, chanspy_ds->unique_id))) {
ast_channel_datastore_remove(chan, datastore);
/* chanspy_ds->chan is NULL after this call */
chanspy_ds_destroy(datastore->data);
datastore->data = NULL;
ast_channel_datastore_free(datastore);
}
ast_channel_unlock(chan);
}
ast_mutex_unlock(&chanspy_ds->lock);
return NULL;
}
static struct chanspy_ds *chanspy_ds_free(struct chanspy_ds *chanspy_ds)
{
struct ast_channel *chan;
if (!chanspy_ds) {
return NULL;
}
ast_mutex_lock(&chanspy_ds->lock);
while ((chan = chanspy_ds->chan)) {
struct ast_datastore *datastore;
if (ast_channel_trylock(chan)) {
DEADLOCK_AVOIDANCE(&chanspy_ds->lock);
continue;
}
if ((datastore = ast_channel_datastore_find(chan, &chanspy_ds_info, chanspy_ds->unique_id))) {
ast_channel_datastore_remove(chan, datastore);
/* chanspy_ds->chan is NULL after this call */
chanspy_ds_destroy(datastore->data);
datastore->data = NULL;
ast_channel_datastore_free(datastore);
}
ast_channel_unlock(chan);
break;
}
ast_mutex_unlock(&chanspy_ds->lock);
return NULL;
}
/*! \internal \brief Disable talk detection on the channel */
static int remove_talk_detect(struct ast_channel *chan)
{
struct ast_datastore *datastore = NULL;
struct talk_detect_params *td_params;
SCOPED_CHANNELLOCK(chan_lock, chan);
datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL);
if (!datastore) {
ast_log(AST_LOG_WARNING, "Cannot remove TALK_DETECT from %s: TALK_DETECT not currently enabled\n",
ast_channel_name(chan));
return -1;
}
td_params = datastore->data;
if (ast_audiohook_remove(chan, &td_params->audiohook)) {
ast_log(AST_LOG_WARNING, "Failed to remove TALK_DETECT audiohook from channel %s\n",
ast_channel_name(chan));
return -1;
}
if (ast_channel_datastore_remove(chan, datastore)) {
ast_log(AST_LOG_WARNING, "Failed to remove TALK_DETECT datastore from channel %s\n",
ast_channel_name(chan));
return -1;
}
ast_datastore_free(datastore);
return 0;
}
static int frame_trace_helper(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct frame_trace_data *framedata;
struct ast_datastore *datastore = NULL;
struct ast_framehook_interface interface = {
.version = AST_FRAMEHOOK_INTERFACE_VERSION,
.event_cb = hook_event_cb,
.destroy_cb = hook_destroy_cb,
};
int i = 0;
if (!(framedata = ast_calloc(1, sizeof(*framedata)))) {
return 0;
}
interface.data = framedata;
if (!strcasecmp(data, "black")) {
framedata->list_type = 1;
}
for (i = 0; i < ARRAY_LEN(frametype2str); i++) {
if (strcasestr(value, frametype2str[i].str)) {
framedata->values[i] = 1;
}
}
ast_channel_lock(chan);
i = ast_framehook_attach(chan, &interface);
if (i >= 0) {
int *id;
if ((datastore = ast_channel_datastore_find(chan, &frame_trace_datastore, NULL))) {
id = datastore->data;
ast_framehook_detach(chan, *id);
ast_channel_datastore_remove(chan, datastore);
}
if (!(datastore = ast_datastore_alloc(&frame_trace_datastore, NULL))) {
ast_framehook_detach(chan, i);
ast_channel_unlock(chan);
return 0;
}
if (!(id = ast_calloc(1, sizeof(int)))) {
ast_datastore_free(datastore);
ast_framehook_detach(chan, i);
ast_channel_unlock(chan);
return 0;
}
*id = i; /* Store off the id. The channel is still locked so it is safe to access this ptr. */
datastore->data = id;
ast_channel_datastore_add(chan, datastore);
}
ast_channel_unlock(chan);
return 0;
}
static void wipe_subscription_datastore(struct ast_channel *chan)
{
struct ast_datastore *datastore;
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &parked_subscription_info, NULL);
if (datastore) {
ast_channel_datastore_remove(chan, datastore);
ast_datastore_free(datastore);
}
ast_channel_unlock(chan);
}
/*!
* \internal
* \brief Destroy the speech datastore on the given channel.
*
* \param chan Channel to destroy speech datastore.
*
* \retval 0 on success.
* \retval -1 not found.
*/
static int speech_datastore_destroy(struct ast_channel *chan)
{
struct ast_datastore *datastore;
int res;
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &speech_datastore, NULL);
if (datastore) {
ast_channel_datastore_remove(chan, datastore);
}
ast_channel_unlock(chan);
if (datastore) {
ast_datastore_free(datastore);
res = 0;
} else {
res = -1;
}
return res;
}
static int speex_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_datastore *datastore = NULL;
struct speex_info *si = NULL;
struct speex_direction_info **sdi = NULL;
int is_new = 0;
ast_channel_lock(chan);
if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
ast_channel_unlock(chan);
if (!(datastore = ast_datastore_alloc(&speex_datastore, NULL))) {
return 0;
}
if (!(si = ast_calloc(1, sizeof(*si)))) {
ast_datastore_free(datastore);
return 0;
}
ast_audiohook_init(&si->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "speex", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
si->audiohook.manipulate_callback = speex_callback;
si->lastrate = 8000;
is_new = 1;
} else {
ast_channel_unlock(chan);
si = datastore->data;
}
if (!strcasecmp(data, "rx")) {
sdi = &si->rx;
} else if (!strcasecmp(data, "tx")) {
sdi = &si->tx;
} else {
ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);
if (is_new) {
ast_datastore_free(datastore);
return -1;
}
}
if (!*sdi) {
if (!(*sdi = ast_calloc(1, sizeof(**sdi)))) {
return 0;
}
/* Right now, the audiohooks API will _only_ provide us 8 kHz slinear
* audio. When it supports 16 kHz (or any other sample rates, we will
* have to take that into account here. */
(*sdi)->samples = -1;
}
if (!strcasecmp(cmd, "agc")) {
if (!sscanf(value, "%30f", &(*sdi)->agclevel))
(*sdi)->agclevel = ast_true(value) ? DEFAULT_AGC_LEVEL : 0.0;
if ((*sdi)->agclevel > 32768.0) {
ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n",
((*sdi == si->rx) ? "rx" : "tx"), (*sdi)->agclevel);
(*sdi)->agclevel = 32768.0;
}
(*sdi)->agc = !!((*sdi)->agclevel);
if ((*sdi)->state) {
speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC, &(*sdi)->agc);
if ((*sdi)->agc) {
speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &(*sdi)->agclevel);
}
}
} else if (!strcasecmp(cmd, "denoise")) {
(*sdi)->denoise = (ast_true(value) != 0);
if ((*sdi)->state) {
speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_DENOISE, &(*sdi)->denoise);
}
}
if (!(*sdi)->agc && !(*sdi)->denoise) {
if ((*sdi)->state)
speex_preprocess_state_destroy((*sdi)->state);
ast_free(*sdi);
*sdi = NULL;
}
if (!si->rx && !si->tx) {
if (is_new) {
is_new = 0;
} else {
ast_channel_lock(chan);
ast_channel_datastore_remove(chan, datastore);
ast_channel_unlock(chan);
ast_audiohook_remove(chan, &si->audiohook);
ast_audiohook_detach(&si->audiohook);
}
ast_datastore_free(datastore);
}
if (is_new) {
datastore->data = si;
ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
ast_audiohook_attach(chan, &si->audiohook);
}
return 0;
}
/*! \brief SpeechBackground(Sound File|Timeout) Dialplan Application */
static int speech_background(struct ast_channel *chan, void *data)
{
unsigned int timeout = 0;
int res = 0, done = 0, argc = 0, started = 0, quieted = 0, max_dtmf_len = 0;
struct ast_module_user *u = NULL;
struct ast_speech *speech = find_speech(chan);
struct ast_frame *f = NULL;
int oldreadformat = AST_FORMAT_SLINEAR;
char dtmf[AST_MAX_EXTENSION] = "";
time_t start, current;
struct ast_datastore *datastore = NULL;
char *argv[2], *args = NULL, *filename_tmp = NULL, *filename = NULL, tmp[2] = "", dtmf_terminator = '#';
const char *tmp2 = NULL;
args = ast_strdupa(data);
u = ast_module_user_add(chan);
if (speech == NULL) {
ast_module_user_remove(u);
return -1;
}
/* If channel is not already answered, then answer it */
if (chan->_state != AST_STATE_UP && ast_answer(chan)) {
ast_module_user_remove(u);
return -1;
}
/* Record old read format */
oldreadformat = chan->readformat;
/* Change read format to be signed linear */
if (ast_set_read_format(chan, AST_FORMAT_SLINEAR)) {
ast_module_user_remove(u);
return -1;
}
/* Parse out options */
argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0]));
if (argc > 0) {
/* Yay sound file */
filename_tmp = ast_strdupa(argv[0]);
if (!ast_strlen_zero(argv[1])) {
if ((timeout = atoi(argv[1])) == 0)
timeout = -1;
} else
timeout = 0;
}
/* See if the maximum DTMF length variable is set... we use a variable in case they want to carry it through their entire dialplan */
if ((tmp2 = pbx_builtin_getvar_helper(chan, "SPEECH_DTMF_MAXLEN")) && !ast_strlen_zero(tmp2))
max_dtmf_len = atoi(tmp2);
/* See if a terminator is specified */
if ((tmp2 = pbx_builtin_getvar_helper(chan, "SPEECH_DTMF_TERMINATOR"))) {
if (ast_strlen_zero(tmp2))
dtmf_terminator = '\0';
else
dtmf_terminator = tmp2[0];
}
/* Before we go into waiting for stuff... make sure the structure is ready, if not - start it again */
if (speech->state == AST_SPEECH_STATE_NOT_READY || speech->state == AST_SPEECH_STATE_DONE) {
ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
ast_speech_start(speech);
}
/* Ensure no streams are currently running */
ast_stopstream(chan);
/* Okay it's streaming so go into a loop grabbing frames! */
while (done == 0) {
/* If the filename is null and stream is not running, start up a new sound file */
if (!quieted && (chan->streamid == -1 && chan->timingfunc == NULL) && (filename = strsep(&filename_tmp, "&"))) {
/* Discard old stream information */
ast_stopstream(chan);
/* Start new stream */
speech_streamfile(chan, filename, chan->language);
}
/* Run scheduled stuff */
ast_sched_runq(chan->sched);
/* Yay scheduling */
res = ast_sched_wait(chan->sched);
if (res < 0) {
res = 1000;
}
/* If there is a frame waiting, get it - if not - oh well */
if (ast_waitfor(chan, res) > 0) {
f = ast_read(chan);
if (f == NULL) {
/* The channel has hung up most likely */
done = 3;
break;
}
}
/* Do timeout check (shared between audio/dtmf) */
if ((!quieted || strlen(dtmf)) && started == 1) {
time(¤t);
if ((current-start) >= timeout) {
done = 1;
if (f)
ast_frfree(f);
break;
}
}
/* Do checks on speech structure to see if it's changed */
ast_mutex_lock(&speech->lock);
if (ast_test_flag(speech, AST_SPEECH_QUIET)) {
if (chan->stream)
ast_stopstream(chan);
ast_clear_flag(speech, AST_SPEECH_QUIET);
quieted = 1;
}
/* Check state so we can see what to do */
switch (speech->state) {
case AST_SPEECH_STATE_READY:
/* If audio playback has stopped do a check for timeout purposes */
if (chan->streamid == -1 && chan->timingfunc == NULL)
ast_stopstream(chan);
if (!quieted && chan->stream == NULL && timeout && started == 0 && !filename_tmp) {
if (timeout == -1) {
done = 1;
if (f)
ast_frfree(f);
break;
}
time(&start);
started = 1;
}
/* Write audio frame out to speech engine if no DTMF has been received */
if (!strlen(dtmf) && f != NULL && f->frametype == AST_FRAME_VOICE) {
ast_speech_write(speech, f->data, f->datalen);
}
break;
case AST_SPEECH_STATE_WAIT:
/* Cue up waiting sound if not already playing */
if (!strlen(dtmf)) {
if (chan->stream == NULL) {
if (speech->processing_sound != NULL) {
if (strlen(speech->processing_sound) > 0 && strcasecmp(speech->processing_sound,"none")) {
speech_streamfile(chan, speech->processing_sound, chan->language);
}
}
} else if (chan->streamid == -1 && chan->timingfunc == NULL) {
ast_stopstream(chan);
if (speech->processing_sound != NULL) {
if (strlen(speech->processing_sound) > 0 && strcasecmp(speech->processing_sound,"none")) {
speech_streamfile(chan, speech->processing_sound, chan->language);
}
}
}
}
break;
case AST_SPEECH_STATE_DONE:
/* Now that we are done... let's switch back to not ready state */
ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
if (!strlen(dtmf)) {
/* Copy to speech structure the results, if available */
speech->results = ast_speech_results_get(speech);
/* Break out of our background too */
done = 1;
/* Stop audio playback */
if (chan->stream != NULL) {
ast_stopstream(chan);
}
}
break;
default:
break;
}
ast_mutex_unlock(&speech->lock);
/* Deal with other frame types */
if (f != NULL) {
/* Free the frame we received */
switch (f->frametype) {
case AST_FRAME_DTMF:
if (dtmf_terminator != '\0' && f->subclass == dtmf_terminator) {
done = 1;
} else {
if (chan->stream != NULL) {
ast_stopstream(chan);
}
if (!started) {
/* Change timeout to be 5 seconds for DTMF input */
timeout = (chan->pbx && chan->pbx->dtimeout) ? chan->pbx->dtimeout : 5;
started = 1;
}
time(&start);
snprintf(tmp, sizeof(tmp), "%c", f->subclass);
strncat(dtmf, tmp, sizeof(dtmf) - strlen(dtmf) - 1);
/* If the maximum length of the DTMF has been reached, stop now */
if (max_dtmf_len && strlen(dtmf) == max_dtmf_len)
done = 1;
}
break;
case AST_FRAME_CONTROL:
switch (f->subclass) {
case AST_CONTROL_HANGUP:
/* Since they hung up we should destroy the speech structure */
done = 3;
default:
break;
}
default:
break;
}
ast_frfree(f);
f = NULL;
}
}
if (strlen(dtmf)) {
/* We sort of make a results entry */
speech->results = ast_calloc(1, sizeof(*speech->results));
if (speech->results != NULL) {
ast_speech_dtmf(speech, dtmf);
speech->results->score = 1000;
speech->results->text = strdup(dtmf);
speech->results->grammar = strdup("dtmf");
}
ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
}
/* See if it was because they hung up */
if (done == 3) {
/* Destroy speech structure */
ast_speech_destroy(speech);
datastore = ast_channel_datastore_find(chan, &speech_datastore, NULL);
if (datastore != NULL) {
ast_channel_datastore_remove(chan, datastore);
}
} else {
/* Channel is okay so restore read format */
ast_set_read_format(chan, oldreadformat);
}
ast_module_user_remove(u);
return 0;
}
/*!
* \brief Get the lua_State for this channel
*
* If no channel is passed then a new state is allocated. States with no
* channel assocatied with them should only be used for matching extensions.
* If the channel does not yet have a lua state associated with it, one will be
* created.
*
* \note If no channel was passed then the caller is expected to free the state
* using lua_close().
*
* \return a lua_State
*/
static lua_State *lua_get_state(struct ast_channel *chan)
{
struct ast_datastore *datastore = NULL;
lua_State *L;
if (!chan) {
L = luaL_newstate();
if (!L) {
ast_log(LOG_ERROR, "Error allocating lua_State, no memory\n");
return NULL;
}
if (lua_load_extensions(L, NULL)) {
const char *error = lua_tostring(L, -1);
ast_log(LOG_ERROR, "Error loading extensions.lua: %s\n", error);
lua_close(L);
return NULL;
}
return L;
} else {
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &lua_datastore, NULL);
ast_channel_unlock(chan);
if (!datastore) {
/* nothing found, allocate a new lua state */
datastore = ast_datastore_alloc(&lua_datastore, NULL);
if (!datastore) {
ast_log(LOG_ERROR, "Error allocation channel datastore for lua_State\n");
return NULL;
}
datastore->data = luaL_newstate();
if (!datastore->data) {
ast_datastore_free(datastore);
ast_log(LOG_ERROR, "Error allocating lua_State, no memory\n");
return NULL;
}
ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
L = datastore->data;
if (lua_load_extensions(L, chan)) {
const char *error = lua_tostring(L, -1);
ast_log(LOG_ERROR, "Error loading extensions.lua for %s: %s\n", ast_channel_name(chan), error);
ast_channel_lock(chan);
ast_channel_datastore_remove(chan, datastore);
ast_channel_unlock(chan);
ast_datastore_free(datastore);
return NULL;
}
}
return datastore->data;
}
}
int ast_do_pickup(struct ast_channel *chan, struct ast_channel *target)
{
struct ast_party_connected_line connected_caller;
struct ast_datastore *ds_pickup;
const char *chan_name;/*!< A masquerade changes channel names. */
const char *target_name;/*!< A masquerade changes channel names. */
int res = -1;
RAII_VAR(struct ast_channel_snapshot *, chan_snapshot, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, target_snapshot, NULL, ao2_cleanup);
target_name = ast_strdupa(ast_channel_name(target));
ast_debug(1, "Call pickup on '%s' by '%s'\n", target_name, ast_channel_name(chan));
/* Mark the target to block any call pickup race. */
ds_pickup = ast_datastore_alloc(&pickup_active, NULL);
if (!ds_pickup) {
ast_log(LOG_WARNING,
"Unable to create channel datastore on '%s' for call pickup\n", target_name);
return -1;
}
ast_channel_datastore_add(target, ds_pickup);
ast_party_connected_line_init(&connected_caller);
ast_party_connected_line_copy(&connected_caller, ast_channel_connected(target));
ast_channel_unlock(target);/* The pickup race is avoided so we do not need the lock anymore. */
/* Reset any earlier private connected id representation */
ast_party_id_reset(&connected_caller.priv);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
if (ast_channel_connected_line_sub(NULL, chan, &connected_caller, 0) &&
ast_channel_connected_line_macro(NULL, chan, &connected_caller, 0, 0)) {
ast_channel_update_connected_line(chan, &connected_caller, NULL);
}
ast_party_connected_line_free(&connected_caller);
ast_channel_lock(chan);
chan_name = ast_strdupa(ast_channel_name(chan));
ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(chan));
ast_channel_unlock(chan);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
if (ast_answer(chan)) {
ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan_name);
goto pickup_failed;
}
if (ast_queue_control(chan, AST_CONTROL_ANSWER)) {
ast_log(LOG_WARNING, "Unable to queue answer on '%s'\n", chan_name);
goto pickup_failed;
}
ast_channel_queue_connected_line_update(chan, &connected_caller, NULL);
/* setting the HANGUPCAUSE so the ringing channel knows this call was not a missed call */
ast_channel_hangupcause_set(chan, AST_CAUSE_ANSWERED_ELSEWHERE);
ast_channel_lock(chan);
chan_snapshot = ast_channel_snapshot_create(chan);
ast_channel_unlock(chan);
if (!chan_snapshot) {
goto pickup_failed;
}
target_snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(target));
if (!target_snapshot) {
goto pickup_failed;
}
if (ast_channel_move(target, chan)) {
ast_log(LOG_WARNING, "Unable to complete call pickup of '%s' with '%s'\n",
chan_name, target_name);
goto pickup_failed;
}
/* target points to the channel that did the pickup at this point, so use that channel's topic instead of chan */
send_call_pickup_stasis_message(target, chan_snapshot, target_snapshot);
res = 0;
pickup_failed:
ast_channel_lock(target);
if (!ast_channel_datastore_remove(target, ds_pickup)) {
ast_datastore_free(ds_pickup);
}
ast_party_connected_line_free(&connected_caller);
return res;
}
void ast_jb_create_framehook(struct ast_channel *chan, struct ast_jb_conf *jb_conf, int prefer_existing)
{
struct jb_framedata *framedata;
struct ast_datastore *datastore = NULL;
struct ast_framehook_interface interface = {
.version = AST_FRAMEHOOK_INTERFACE_VERSION,
.event_cb = hook_event_cb,
.destroy_cb = hook_destroy_cb,
};
int i = 0;
/* If disabled, strip any existing jitterbuffer and don't replace it. */
if (!strcasecmp(jb_conf->impl, "disabled")) {
int *id;
ast_channel_lock(chan);
if ((datastore = ast_channel_datastore_find(chan, &jb_datastore, NULL))) {
id = datastore->data;
ast_framehook_detach(chan, *id);
ast_channel_datastore_remove(chan, datastore);
ast_datastore_free(datastore);
}
ast_channel_unlock(chan);
return;
}
if (!(framedata = ast_calloc(1, sizeof(*framedata)))) {
return;
}
if (jb_framedata_init(framedata, jb_conf)) {
jb_framedata_destroy(framedata);
return;
}
interface.data = framedata;
ast_channel_lock(chan);
i = ast_framehook_attach(chan, &interface);
if (i >= 0) {
int *id;
if ((datastore = ast_channel_datastore_find(chan, &jb_datastore, NULL))) {
/* There is already a jitterbuffer on the channel. */
if (prefer_existing) {
/* We prefer the existing jitterbuffer, so remove the new one and keep the old one. */
ast_framehook_detach(chan, i);
ast_channel_unlock(chan);
return;
}
/* We prefer the new jitterbuffer, so strip the old one. */
id = datastore->data;
ast_framehook_detach(chan, *id);
ast_channel_datastore_remove(chan, datastore);
ast_datastore_free(datastore);
}
if (!(datastore = ast_datastore_alloc(&jb_datastore, NULL))) {
ast_framehook_detach(chan, i);
ast_channel_unlock(chan);
return;
}
if (!(id = ast_calloc(1, sizeof(int)))) {
ast_datastore_free(datastore);
ast_framehook_detach(chan, i);
ast_channel_unlock(chan);
return;
}
*id = i; /* Store off the id. The channel is still locked so it is safe to access this ptr. */
datastore->data = id;
ast_channel_datastore_add(chan, datastore);
ast_channel_set_fd(chan, AST_JITTERBUFFER_FD, framedata->timer_fd);
} else {
jb_framedata_destroy(framedata);
framedata = NULL;
}
ast_channel_unlock(chan);
}
