예제 #1
0
/*! \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;
}
예제 #2
0
파일: func_srv.c 프로젝트: cfhb/vlink-cti
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;
}
예제 #3
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;
}
예제 #4
0
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;
}
예제 #5
0
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;
}
예제 #6
0
/*! \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;
}
예제 #7
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;
}
예제 #8
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);
}
예제 #9
0
/*!
 * \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;
}
예제 #10
0
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;
}
예제 #11
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(&current);
			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;
}
예제 #12
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;
	}
}
예제 #13
0
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;
}
예제 #14
0
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);
}