コード例 #1
0
ファイル: format_pref.c プロジェクト: Djcd/asterisk-opus
/*! \brief Pick a codec */
struct ast_format *ast_codec_choose(struct ast_codec_pref *pref, struct ast_format_cap *cap, int find_best, struct ast_format *result)
{
	int x, slot, found;
	size_t f_len = 0;
	const struct ast_format_list *f_list = ast_format_list_get(&f_len);

	for (x = 0; x < f_len; x++) {
		slot = pref->order[x];

		if (!slot)
			break;
		if (ast_format_cap_get_compatible_format(cap, &f_list[slot-1].format, result)) {
			found = 1; /*format is found and stored in result */
			break;
		}
	}
	ast_format_list_destroy(f_list);
	if (found && (AST_FORMAT_GET_TYPE(result->id) == AST_FORMAT_TYPE_AUDIO)) {
		return result;
	}
	ast_format_clear(result);
	ast_debug(4, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");

	return find_best ? ast_best_codec(cap, result) : NULL;
}
コード例 #2
0
ファイル: app_mrcpsynth.c プロジェクト: Jared-Prime/UniMRCP
/* The entry point of the application. */
static int app_synth_exec(struct ast_channel *chan, ast_app_data data)
{
	ast_format_compat nwriteformat;
	ast_format_clear(&nwriteformat);
	get_synth_format(chan, &nwriteformat);

	int samplerate = 8000;
	/* int framesize = DEFAULT_FRAMESIZE; */
	int framesize = format_to_bytes_per_sample(&nwriteformat) * (DEFAULT_FRAMESIZE / 2);
	struct ast_frame *f;
	struct ast_frame fr;
	struct timeval next;
	int ms;
	apr_size_t len;
	int rres;
	ast_mrcp_profile_t *profile;
	apr_uint32_t speech_channel_number = get_next_speech_channel_number();
	const char *name;
	char buffer[framesize];
	speech_channel_status_t status;
	char *parse;
	int i;
	mrcpsynth_options_t mrcpsynth_options;
	mrcpsynth_session_t mrcpsynth_session;

	AST_DECLARE_APP_ARGS(args,
		AST_APP_ARG(prompt);
		AST_APP_ARG(options);
	);
コード例 #3
0
/** \brief Version dependent prototypes of uni_recog_create() function */
#if AST_VERSION_AT_LEAST(10,0,0)
static int uni_recog_create(struct ast_speech *speech, ast_format_compat *format)
{
	return uni_recog_create_internal(speech,format);
}
#elif AST_VERSION_AT_LEAST(1,6,0)
static int uni_recog_create(struct ast_speech *speech, int format_id)
{
	ast_format_compat format;
	ast_format_clear(&format);
	format.id = format_id;
	return uni_recog_create_internal(speech,&format);
}
#else /* 1.4 */
static int uni_recog_create(struct ast_speech *speech)
{
	ast_format_compat format;
	ast_format_clear(&format);
	format.id = 0;
	return uni_recog_create_internal(speech,&format);
}
コード例 #4
0
ファイル: format_pref.c プロジェクト: Djcd/asterisk-opus
struct ast_format *ast_codec_pref_index(struct ast_codec_pref *pref, int idx, struct ast_format *result)
{
	if ((idx >= 0) && (idx < sizeof(pref->order)) && pref->formats[idx].id) {
		ast_format_copy(result, &pref->formats[idx]);
	} else {
		ast_format_clear(result);
		return NULL;
	}

	return result;
}
コード例 #5
0
int ast_format_cap_best_byid(const struct ast_format_cap *cap, enum ast_format_id id, struct ast_format *result)
{
	struct byid_data data;
	data.result = result;
	data.id = id;

	ast_format_clear(result);
	ao2_callback(cap->formats,
		OBJ_MULTIPLE | OBJ_NODATA | cap->nolock,
		find_best_byid_cb,
		&data);
	return result->id ? 1 : 0;
}
コード例 #6
0
int ast_format_cap_get_compatible_format(const struct ast_format_cap *cap, const struct ast_format *format, struct ast_format *result)
{
	struct ast_format *f;
	struct ast_format_cap *tmp_cap = (struct ast_format_cap *) cap;
	f = ao2_find(tmp_cap->formats, (struct ast_format *) format, OBJ_POINTER | tmp_cap->nolock);

	if (f) {
		ast_format_copy(result, f);
		ao2_ref(f, -1);
		return 1;
	}
	ast_format_clear(result);
	return 0;
}
コード例 #7
0
struct ast_format *ast_format_set(struct ast_format *format, enum ast_format_id id, int set_attributes, ... )
{
	/* initialize the structure before setting it. */
	ast_format_clear(format);

	format->id = id;

	if (set_attributes) {
		va_list ap;
		va_start(ap, set_attributes);
		format_set_helper(format, ap);
		va_end(ap);
	}

	return format;
}
コード例 #8
0
ファイル: app_chanspy.c プロジェクト: xhook/asterisk-v11
static int dahdiscan_exec(struct ast_channel *chan, const char *data)
{
	const char *spec = "DAHDI";
	struct ast_flags flags;
	struct spy_dtmf_options user_options = {
		.cycle = '#',
		.volume = '\0',
		.exit = '*',
	};
	struct ast_format oldwf;
	int res;
	char *mygroup = NULL;

	/* Coverity - This uninit_use should be ignored since this macro initializes the flags */
	ast_clear_flag(&flags, AST_FLAGS_ALL);
	ast_format_clear(&oldwf);
	if (!ast_strlen_zero(data)) {
		mygroup = ast_strdupa(data);
	}
	ast_set_flag(&flags, OPTION_DTMF_EXIT);
	ast_set_flag(&flags, OPTION_DTMF_CYCLE);
	ast_set_flag(&flags, OPTION_DAHDI_SCAN);

	ast_format_copy(&oldwf, ast_channel_writeformat(chan));
	if (ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR) < 0) {
		ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
		return -1;
	}

	res = common_exec(chan, &flags, 0, 0, &user_options, mygroup, NULL, spec, NULL, NULL, NULL, NULL);

	if (oldwf.id && ast_set_write_format(chan, &oldwf) < 0)
		ast_log(LOG_ERROR, "Could Not Set Write Format.\n");

	return res;
}

static int unload_module(void)
{
	int res = 0;

	res |= ast_unregister_application(app_chan);
	res |= ast_unregister_application(app_ext);
	res |= ast_unregister_application(app_dahdiscan);

	return res;
}
コード例 #9
0
ファイル: ooh323cDriver.c プロジェクト: Djcd/asterisk-opus
struct ast_format *convertH323CapToAsteriskCap(int cap, struct ast_format *result)
{
   ast_format_clear(result);
   switch(cap)
   {
      case OO_G711ULAW64K:
         return ast_format_set(result, AST_FORMAT_ULAW, 0);
      case OO_G711ALAW64K:
         return ast_format_set(result, AST_FORMAT_ALAW, 0);
      case OO_GSMFULLRATE:
         return ast_format_set(result, AST_FORMAT_GSM, 0);

#ifdef AST_FORMAT_AMRNB
      case OO_AMRNB:
         return ast_format_set(result, AST_FORMAT_AMRNB, 0);
#endif
#ifdef AST_FORMAT_SPEEX
      case OO_SPEEX:
         return ast_format_set(result, AST_FORMAT_SPEEX, 0);
#endif

      case OO_G729:
         return ast_format_set(result, AST_FORMAT_G729A, 0);
      case OO_G729A:
         return ast_format_set(result, AST_FORMAT_G729A, 0);
      case OO_G729B:
         return ast_format_set(result, AST_FORMAT_G729A, 0);
      case OO_G7231:
         return ast_format_set(result, AST_FORMAT_G723_1, 0);
      case OO_G726:
         return ast_format_set(result, AST_FORMAT_G726, 0);
      case OO_G726AAL2:
         return ast_format_set(result, AST_FORMAT_G726_AAL2, 0);
      case OO_H263VIDEO:
         return ast_format_set(result, AST_FORMAT_H263, 0);
      default:
         ast_debug(1, "Cap %d is not supported by driver yet\n", cap);
         return NULL;
   }

   return NULL;
}
コード例 #10
0
ファイル: app_fax.c プロジェクト: bugrahantopall/asterisk
static int transmit_audio(fax_session *s)
{
	int res = -1;
	struct ast_format original_read_fmt;
	struct ast_format original_write_fmt;
	fax_state_t fax;
	t30_state_t *t30state;
	struct ast_frame *inf = NULL;
	int last_state = 0;
	struct timeval now, start, state_change;
	enum ast_t38_state t38_state;
	struct ast_control_t38_parameters t38_parameters = { .version = 0,
							     .max_ifp = 800,
							     .rate = AST_T38_RATE_14400,
							     .rate_management = AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF,
							     .fill_bit_removal = 1,
/*
 * spandsp has API calls to support MMR and JBIG transcoding, but they aren't
 * implemented quite yet... so don't offer them to the remote endpoint
 *							     .transcoding_mmr = 1,
 *							     .transcoding_jbig = 1,
*/
	};

	ast_format_clear(&original_read_fmt);
	ast_format_clear(&original_write_fmt);

	/* if in called party mode, try to use T.38 */
	if (s->caller_mode == FALSE) {
		/* check if we are already in T.38 mode (unlikely), or if we can request
		 * a switch... if so, request it now and wait for the result, rather
		 * than starting an audio FAX session that will have to be cancelled
		 */
		if ((t38_state = ast_channel_get_t38_state(s->chan)) == T38_STATE_NEGOTIATED) {
			return 1;
		} else if ((t38_state != T38_STATE_UNAVAILABLE) &&
			   (t38_parameters.request_response = AST_T38_REQUEST_NEGOTIATE,
			    (ast_indicate_data(s->chan, AST_CONTROL_T38_PARAMETERS, &t38_parameters, sizeof(t38_parameters)) == 0))) {
			/* wait up to five seconds for negotiation to complete */
			unsigned int timeout = 5000;
			int ms;

			ast_debug(1, "Negotiating T.38 for receive on %s\n", ast_channel_name(s->chan));
			while (timeout > 0) {
				ms = ast_waitfor(s->chan, 1000);
				if (ms < 0) {
					ast_log(LOG_WARNING, "something bad happened while channel '%s' was polling.\n", ast_channel_name(s->chan));
					return -1;
				}
				if (!ms) {
					/* nothing happened */
					if (timeout > 0) {
						timeout -= 1000;
						continue;
					} else {
						ast_log(LOG_WARNING, "channel '%s' timed-out during the T.38 negotiation.\n", ast_channel_name(s->chan));
						break;
					}
				}
				if (!(inf = ast_read(s->chan))) {
					return -1;
				}
				if ((inf->frametype == AST_FRAME_CONTROL) &&
				    (inf->subclass.integer == AST_CONTROL_T38_PARAMETERS) &&
				    (inf->datalen == sizeof(t38_parameters))) {
					struct ast_control_t38_parameters *parameters = inf->data.ptr;

					switch (parameters->request_response) {
					case AST_T38_NEGOTIATED:
						ast_debug(1, "Negotiated T.38 for receive on %s\n", ast_channel_name(s->chan));
						res = 1;
						break;
					case AST_T38_REFUSED:
						ast_log(LOG_WARNING, "channel '%s' refused to negotiate T.38\n", ast_channel_name(s->chan));
						break;
					default:
						ast_log(LOG_ERROR, "channel '%s' failed to negotiate T.38\n", ast_channel_name(s->chan));
						break;
					}
					ast_frfree(inf);
					if (res == 1) {
						return 1;
					} else {
						break;
					}
				}
				ast_frfree(inf);
			}
		}
	}

#if SPANDSP_RELEASE_DATE >= 20080725
        /* for spandsp shaphots 0.0.6 and higher */
        t30state = &fax.t30;
#else
        /* for spandsp release 0.0.5 */
        t30state = &fax.t30_state;
#endif

	ast_format_copy(&original_read_fmt, ast_channel_readformat(s->chan));
	if (original_read_fmt.id != AST_FORMAT_SLINEAR) {
		res = ast_set_read_format_by_id(s->chan, AST_FORMAT_SLINEAR);
		if (res < 0) {
			ast_log(LOG_WARNING, "Unable to set to linear read mode, giving up\n");
			goto done;
		}
	}

	ast_format_copy(&original_write_fmt, ast_channel_writeformat(s->chan));
	if (original_write_fmt.id != AST_FORMAT_SLINEAR) {
		res = ast_set_write_format_by_id(s->chan, AST_FORMAT_SLINEAR);
		if (res < 0) {
			ast_log(LOG_WARNING, "Unable to set to linear write mode, giving up\n");
			goto done;
		}
	}

	/* Initialize T30 terminal */
	fax_init(&fax, s->caller_mode);

	/* Setup logging */
	set_logging(&fax.logging);
	set_logging(&t30state->logging);

	/* Configure terminal */
	set_local_info(t30state, s);
	set_file(t30state, s);
	set_ecm(t30state, TRUE);

	fax_set_transmit_on_idle(&fax, TRUE);

	t30_set_phase_e_handler(t30state, phase_e_handler, s);

	start = state_change = ast_tvnow();

	ast_activate_generator(s->chan, &generator, &fax);

	while (!s->finished) {
		inf = NULL;

		if ((res = ast_waitfor(s->chan, 25)) < 0) {
			ast_debug(1, "Error waiting for a frame\n");
			break;
		}

		/* Watchdog */
		now = ast_tvnow();
		if (ast_tvdiff_sec(now, start) > WATCHDOG_TOTAL_TIMEOUT || ast_tvdiff_sec(now, state_change) > WATCHDOG_STATE_TIMEOUT) {
			ast_log(LOG_WARNING, "It looks like we hung. Aborting.\n");
			res = -1;
			break;
		}

		if (!res) {
			/* There was timeout waiting for a frame. Loop around and wait again */
			continue;
		}

		/* There is a frame available. Get it */
		res = 0;

		if (!(inf = ast_read(s->chan))) {
			ast_debug(1, "Channel hangup\n");
			res = -1;
			break;
		}

		ast_debug(10, "frame %d/%u, len=%d\n", inf->frametype, (unsigned int) inf->subclass.format.id, inf->datalen);

		/* Check the frame type. Format also must be checked because there is a chance
		   that a frame in old format was already queued before we set channel format
		   to slinear so it will still be received by ast_read */
		if (inf->frametype == AST_FRAME_VOICE && inf->subclass.format.id == AST_FORMAT_SLINEAR) {
			if (fax_rx(&fax, inf->data.ptr, inf->samples) < 0) {
				/* I know fax_rx never returns errors. The check here is for good style only */
				ast_log(LOG_WARNING, "fax_rx returned error\n");
				res = -1;
				break;
			}
			if (last_state != t30state->state) {
				state_change = ast_tvnow();
				last_state = t30state->state;
			}
		} else if ((inf->frametype == AST_FRAME_CONTROL) &&
			   (inf->subclass.integer == AST_CONTROL_T38_PARAMETERS)) {
			struct ast_control_t38_parameters *parameters = inf->data.ptr;

			if (parameters->request_response == AST_T38_NEGOTIATED) {
				/* T38 switchover completed */
				s->t38parameters = *parameters;
				ast_debug(1, "T38 negotiated, finishing audio loop\n");
				res = 1;
				break;
			} else if (parameters->request_response == AST_T38_REQUEST_NEGOTIATE) {
				t38_parameters.request_response = AST_T38_NEGOTIATED;
				ast_debug(1, "T38 request received, accepting\n");
				/* Complete T38 switchover */
				ast_indicate_data(s->chan, AST_CONTROL_T38_PARAMETERS, &t38_parameters, sizeof(t38_parameters));
				/* Do not break audio loop, wait until channel driver finally acks switchover
				 * with AST_T38_NEGOTIATED
				 */
			}
		}

		ast_frfree(inf);
		inf = NULL;
	}

	ast_debug(1, "Loop finished, res=%d\n", res);

	if (inf)
		ast_frfree(inf);

	ast_deactivate_generator(s->chan);

	/* If we are switching to T38, remove phase E handler. Otherwise it will be executed
	   by t30_terminate, display diagnostics and set status variables although no transmittion
	   has taken place yet. */
	if (res > 0) {
		t30_set_phase_e_handler(t30state, NULL, NULL);
	}

	t30_terminate(t30state);
	fax_release(&fax);

done:
	if (original_write_fmt.id != AST_FORMAT_SLINEAR) {
		if (ast_set_write_format(s->chan, &original_write_fmt) < 0)
			ast_log(LOG_WARNING, "Unable to restore write format on '%s'\n", ast_channel_name(s->chan));
	}

	if (original_read_fmt.id != AST_FORMAT_SLINEAR) {
		if (ast_set_read_format(s->chan, &original_read_fmt) < 0)
			ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", ast_channel_name(s->chan));
	}

	return res;

}

static int transmit_t38(fax_session *s)
{
	int res = 0;
	t38_terminal_state_t t38;
	struct ast_frame *inf = NULL;
	int last_state = 0;
	struct timeval now, start, state_change, last_frame;
	t30_state_t *t30state;
	t38_core_state_t *t38state;

#if SPANDSP_RELEASE_DATE >= 20080725
	/* for spandsp shaphots 0.0.6 and higher */
	t30state = &t38.t30;
	t38state = &t38.t38_fe.t38;
#else
	/* for spandsp releases 0.0.5 */
	t30state = &t38.t30_state;
	t38state = &t38.t38;
#endif

	/* Initialize terminal */
	memset(&t38, 0, sizeof(t38));
	if (t38_terminal_init(&t38, s->caller_mode, t38_tx_packet_handler, s->chan) == NULL) {
		ast_log(LOG_WARNING, "Unable to start T.38 termination.\n");
		res = -1;
		goto disable_t38;
	}

	t38_set_max_datagram_size(t38state, s->t38parameters.max_ifp);

	if (s->t38parameters.fill_bit_removal) {
		t38_set_fill_bit_removal(t38state, TRUE);
	}
	if (s->t38parameters.transcoding_mmr) {
		t38_set_mmr_transcoding(t38state, TRUE);
	}
	if (s->t38parameters.transcoding_jbig) {
		t38_set_jbig_transcoding(t38state, TRUE);
	}

	/* Setup logging */
	set_logging(&t38.logging);
	set_logging(&t30state->logging);
	set_logging(&t38state->logging);

	/* Configure terminal */
	set_local_info(t30state, s);
	set_file(t30state, s);
	set_ecm(t30state, TRUE);

	t30_set_phase_e_handler(t30state, phase_e_handler, s);

	now = start = state_change = ast_tvnow();

	while (!s->finished) {
		inf = NULL;

		if ((res = ast_waitfor(s->chan, 25)) < 0) {
			ast_debug(1, "Error waiting for a frame\n");
			break;
		}

		last_frame = now;

		/* Watchdog */
		now = ast_tvnow();
		if (ast_tvdiff_sec(now, start) > WATCHDOG_TOTAL_TIMEOUT || ast_tvdiff_sec(now, state_change) > WATCHDOG_STATE_TIMEOUT) {
			ast_log(LOG_WARNING, "It looks like we hung. Aborting.\n");
			res = -1;
			break;
		}

		t38_terminal_send_timeout(&t38, ast_tvdiff_us(now, last_frame) / (1000000 / 8000));

		if (!res) {
			/* There was timeout waiting for a frame. Loop around and wait again */
			continue;
		}

		/* There is a frame available. Get it */
		res = 0;

		if (!(inf = ast_read(s->chan))) {
			ast_debug(1, "Channel hangup\n");
			res = -1;
			break;
		}

		ast_debug(10, "frame %d/%d, len=%d\n", inf->frametype, inf->subclass.integer, inf->datalen);

		if (inf->frametype == AST_FRAME_MODEM && inf->subclass.integer == AST_MODEM_T38) {
			t38_core_rx_ifp_packet(t38state, inf->data.ptr, inf->datalen, inf->seqno);
			if (last_state != t30state->state) {
				state_change = ast_tvnow();
				last_state = t30state->state;
			}
		} else if (inf->frametype == AST_FRAME_CONTROL && inf->subclass.integer == AST_CONTROL_T38_PARAMETERS) {
			struct ast_control_t38_parameters *parameters = inf->data.ptr;
			if (parameters->request_response == AST_T38_TERMINATED) {
				ast_debug(1, "T38 down, finishing\n");
				break;
			}
		}

		ast_frfree(inf);
		inf = NULL;
	}

	ast_debug(1, "Loop finished, res=%d\n", res);

	if (inf)
		ast_frfree(inf);

	t30_terminate(t30state);
	t38_terminal_release(&t38);

disable_t38:
	/* if we are not the caller, it's our job to shut down the T.38
	 * session when the FAX transmisson is complete.
	 */
	if ((s->caller_mode == FALSE) &&
	    (ast_channel_get_t38_state(s->chan) == T38_STATE_NEGOTIATED)) {
		struct ast_control_t38_parameters t38_parameters = { .request_response = AST_T38_REQUEST_TERMINATE, };

		if (ast_indicate_data(s->chan, AST_CONTROL_T38_PARAMETERS, &t38_parameters, sizeof(t38_parameters)) == 0) {
			/* wait up to five seconds for negotiation to complete */
			unsigned int timeout = 5000;
			int ms;

			ast_debug(1, "Shutting down T.38 on %s\n", ast_channel_name(s->chan));
			while (timeout > 0) {
				ms = ast_waitfor(s->chan, 1000);
				if (ms < 0) {
					ast_log(LOG_WARNING, "something bad happened while channel '%s' was polling.\n", ast_channel_name(s->chan));
					return -1;
				}
				if (!ms) {
					/* nothing happened */
					if (timeout > 0) {
						timeout -= 1000;
						continue;
					} else {
						ast_log(LOG_WARNING, "channel '%s' timed-out during the T.38 shutdown.\n", ast_channel_name(s->chan));
						break;
					}
				}
				if (!(inf = ast_read(s->chan))) {
					return -1;
				}
				if ((inf->frametype == AST_FRAME_CONTROL) &&
				    (inf->subclass.integer == AST_CONTROL_T38_PARAMETERS) &&
				    (inf->datalen == sizeof(t38_parameters))) {
					struct ast_control_t38_parameters *parameters = inf->data.ptr;

					switch (parameters->request_response) {
					case AST_T38_TERMINATED:
						ast_debug(1, "Shut down T.38 on %s\n", ast_channel_name(s->chan));
						break;
					case AST_T38_REFUSED:
						ast_log(LOG_WARNING, "channel '%s' refused to disable T.38\n", ast_channel_name(s->chan));
						break;
					default:
						ast_log(LOG_ERROR, "channel '%s' failed to disable T.38\n", ast_channel_name(s->chan));
						break;
					}
					ast_frfree(inf);
					break;
				}
				ast_frfree(inf);
			}
		}
	}

	return res;
}
コード例 #11
0
struct ast_format *ast_format_from_old_bitfield(struct ast_format *dst, uint64_t src)
{
	switch (src) {
	/*! G.723.1 compression */
	case (1ULL << 0):
		return ast_format_set(dst, AST_FORMAT_G723_1, 0);
	/*! GSM compression */
	case (1ULL << 1):
		return ast_format_set(dst, AST_FORMAT_GSM, 0);
	/*! Raw mu-law data (G.711) */
	case (1ULL << 2):
		return ast_format_set(dst, AST_FORMAT_ULAW, 0);
	/*! Raw A-law data (G.711) */
	case (1ULL << 3):
		return ast_format_set(dst, AST_FORMAT_ALAW, 0);
	/*! ADPCM (G.726, 32kbps, AAL2 codeword packing) */
	case (1ULL << 4):
		return ast_format_set(dst, AST_FORMAT_G726_AAL2, 0);
	/*! ADPCM (IMA) */
	case (1ULL << 5):
		return ast_format_set(dst, AST_FORMAT_ADPCM, 0);
	/*! Raw 16-bit Signed Linear (8000 Hz) PCM */
	case (1ULL << 6):
		return ast_format_set(dst, AST_FORMAT_SLINEAR, 0);
	/*! LPC10, 180 samples/frame */
	case (1ULL << 7):
		return ast_format_set(dst, AST_FORMAT_LPC10, 0);
	/*! G.729A audio */
	case (1ULL << 8):
		return ast_format_set(dst, AST_FORMAT_G729A, 0);
	/*! SpeeX Free Compression */
	case (1ULL << 9):
		return ast_format_set(dst, AST_FORMAT_SPEEX, 0);
	/*! iLBC Free Compression */
	case (1ULL << 10):
		return ast_format_set(dst, AST_FORMAT_ILBC, 0);
	/*! ADPCM (G.726, 32kbps, RFC3551 codeword packing) */
	case (1ULL << 11):
		return ast_format_set(dst, AST_FORMAT_G726, 0);
	/*! G.722 */
	case (1ULL << 12):
		return ast_format_set(dst, AST_FORMAT_G722, 0);
	/*! G.722.1 (also known as Siren7, 32kbps assumed) */
	case (1ULL << 13):
		return ast_format_set(dst, AST_FORMAT_SIREN7, 0);
	/*! G.722.1 Annex C (also known as Siren14, 48kbps assumed) */
	case (1ULL << 14):
		return ast_format_set(dst, AST_FORMAT_SIREN14, 0);
	/*! Raw 16-bit Signed Linear (16000 Hz) PCM */
	case (1ULL << 15):
		return ast_format_set(dst, AST_FORMAT_SLINEAR16, 0);
	/*! G.719 (64 kbps assumed) */
	case (1ULL << 32):
		return ast_format_set(dst, AST_FORMAT_G719, 0);
	/*! SpeeX Wideband (16kHz) Free Compression */
	case (1ULL << 33):
		return ast_format_set(dst, AST_FORMAT_SPEEX16, 0);
	/*! Raw mu-law data (G.711) */
	case (1ULL << 47):
		return ast_format_set(dst, AST_FORMAT_TESTLAW, 0);

	/*! H.261 Video */
	case (1ULL << 18):
		return ast_format_set(dst, AST_FORMAT_H261, 0);
	/*! H.263 Video */
	case (1ULL << 19):
		return ast_format_set(dst, AST_FORMAT_H263, 0);
	/*! H.263+ Video */
	case (1ULL << 20):
		return ast_format_set(dst, AST_FORMAT_H263_PLUS, 0);
	/*! H.264 Video */
	case (1ULL << 21):
		return ast_format_set(dst, AST_FORMAT_H264, 0);
	/*! MPEG4 Video */
	case (1ULL << 22):
		return ast_format_set(dst, AST_FORMAT_MP4_VIDEO, 0);

	/*! JPEG Images */
	case (1ULL << 16):
		return ast_format_set(dst, AST_FORMAT_JPEG, 0);
	/*! PNG Images */
	case (1ULL << 17):
		return ast_format_set(dst, AST_FORMAT_PNG, 0);

	/*! T.140 RED Text format RFC 4103 */
	case (1ULL << 26):
		return ast_format_set(dst, AST_FORMAT_T140RED, 0);
	/*! T.140 Text format - ITU T.140, RFC 4103 */
	case (1ULL << 27):
		return ast_format_set(dst, AST_FORMAT_T140, 0);
	}
	ast_format_clear(dst);
	return NULL;
}
コード例 #12
0
ファイル: app_nbscat.c プロジェクト: mtulio/mtulio
static int NBScat_exec(struct ast_channel *chan, const char *data)
{
	int res=0;
	int fds[2];
	int ms = -1;
	int pid = -1;
	struct ast_format owriteformat;
	struct timeval next;
	struct ast_frame *f;
	struct myframe {
		struct ast_frame f;
		char offset[AST_FRIENDLY_OFFSET];
		short frdata[160];
	} myf;

	ast_format_clear(&owriteformat);
	if (socketpair(AF_LOCAL, SOCK_STREAM, 0, fds)) {
		ast_log(LOG_WARNING, "Unable to create socketpair\n");
		return -1;
	}
	
	ast_stopstream(chan);

	ast_format_copy(&owriteformat, &chan->writeformat);
	res = ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR);
	if (res < 0) {
		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
		return -1;
	}
	
	res = NBScatplay(fds[1]);
	/* Wait 1000 ms first */
	next = ast_tvnow();
	next.tv_sec += 1;
	if (res >= 0) {
		pid = res;
		/* Order is important -- there's almost always going to be mp3...  we want to prioritize the
		   user */
		for (;;) {
			ms = ast_tvdiff_ms(next, ast_tvnow());
			if (ms <= 0) {
				res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata));
				if (res > 0) {
					myf.f.frametype = AST_FRAME_VOICE;
					ast_format_set(&myf.f.subclass.format, AST_FORMAT_SLINEAR, 0);
					myf.f.datalen = res;
					myf.f.samples = res / 2;
					myf.f.mallocd = 0;
					myf.f.offset = AST_FRIENDLY_OFFSET;
					myf.f.src = __PRETTY_FUNCTION__;
					myf.f.delivery.tv_sec = 0;
					myf.f.delivery.tv_usec = 0;
					myf.f.data.ptr = myf.frdata;
					if (ast_write(chan, &myf.f) < 0) {
						res = -1;
						break;
					}
				} else {
					ast_debug(1, "No more mp3\n");
					res = 0;
					break;
				}
				next = ast_tvadd(next, ast_samp2tv(myf.f.samples, 8000));
			} else {
				ms = ast_waitfor(chan, ms);
				if (ms < 0) {
					ast_debug(1, "Hangup detected\n");
					res = -1;
					break;
				}
				if (ms) {
					f = ast_read(chan);
					if (!f) {
						ast_debug(1, "Null frame == hangup() detected\n");
						res = -1;
						break;
					}
					if (f->frametype == AST_FRAME_DTMF) {
						ast_debug(1, "User pressed a key\n");
						ast_frfree(f);
						res = 0;
						break;
					}
					ast_frfree(f);
				} 
			}
		}
	}
	close(fds[0]);
	close(fds[1]);
	
	if (pid > -1)
		kill(pid, SIGKILL);
	if (!res && owriteformat.id)
		ast_set_write_format(chan, &owriteformat);

	return res;
}
コード例 #13
0
ファイル: app_ices.c プロジェクト: bugrahantopall/asterisk
static int ices_exec(struct ast_channel *chan, const char *data)
{
	int res = 0;
	int fds[2];
	int ms = -1;
	int pid = -1;
	int flags;
	struct ast_format oreadformat;
	struct ast_frame *f;
	char filename[256]="";
	char *c;

	ast_format_clear(&oreadformat);
	if (ast_strlen_zero(data)) {
		ast_log(LOG_WARNING, "ICES requires an argument (configfile.xml)\n");
		return -1;
	}
	
	if (pipe(fds)) {
		ast_log(LOG_WARNING, "Unable to create pipe\n");
		return -1;
	}
	flags = fcntl(fds[1], F_GETFL);
	fcntl(fds[1], F_SETFL, flags | O_NONBLOCK);
	
	ast_stopstream(chan);

	if (ast_channel_state(chan) != AST_STATE_UP)
		res = ast_answer(chan);
		
	if (res) {
		close(fds[0]);
		close(fds[1]);
		ast_log(LOG_WARNING, "Answer failed!\n");
		return -1;
	}

	ast_format_copy(&oreadformat, ast_channel_readformat(chan));
	res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
	if (res < 0) {
		close(fds[0]);
		close(fds[1]);
		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
		return -1;
	}
	if (((char *)data)[0] == '/')
		ast_copy_string(filename, (char *) data, sizeof(filename));
	else
		snprintf(filename, sizeof(filename), "%s/%s", ast_config_AST_CONFIG_DIR, (char *)data);
	/* Placeholder for options */		
	c = strchr(filename, '|');
	if (c)
		*c = '\0';	
	res = icesencode(filename, fds[0]);
	if (res >= 0) {
		pid = res;
		for (;;) {
			/* Wait for audio, and stream */
			ms = ast_waitfor(chan, -1);
			if (ms < 0) {
				ast_debug(1, "Hangup detected\n");
				res = -1;
				break;
			}
			f = ast_read(chan);
			if (!f) {
				ast_debug(1, "Null frame == hangup() detected\n");
				res = -1;
				break;
			}
			if (f->frametype == AST_FRAME_VOICE) {
				res = write(fds[1], f->data.ptr, f->datalen);
				if (res < 0) {
					if (errno != EAGAIN) {
						ast_log(LOG_WARNING, "Write failed to pipe: %s\n", strerror(errno));
						res = -1;
						ast_frfree(f);
						break;
					}
				}
			}
			ast_frfree(f);
		}
	}
	close(fds[0]);
	close(fds[1]);

	if (pid > -1)
		kill(pid, SIGKILL);
	if (!res && oreadformat.id)
		ast_set_read_format(chan, &oreadformat);

	return res;
}
コード例 #14
0
ファイル: app_chanspy.c プロジェクト: xhook/asterisk-v11
static int extenspy_exec(struct ast_channel *chan, const char *data)
{
	char *ptr, *exten = NULL;
	char *mygroup = NULL;
	char *recbase = NULL;
	int fd = 0;
	struct ast_flags flags;
	struct spy_dtmf_options user_options = {
		.cycle = '*',
		.volume = '#',
		.exit = '\0',
	};
	struct ast_format oldwf;
	int volfactor = 0;
	int res;
	char *mailbox = NULL;
	char *name_context = NULL;
	AST_DECLARE_APP_ARGS(args,
		AST_APP_ARG(context);
		AST_APP_ARG(options);
	);
	char *parse = ast_strdupa(data);

	AST_STANDARD_APP_ARGS(args, parse);
	ast_format_clear(&oldwf);

	if (!ast_strlen_zero(args.context) && (ptr = strchr(args.context, '@'))) {
		exten = args.context;
		*ptr++ = '\0';
		args.context = ptr;
	}
	if (ast_strlen_zero(args.context))
		args.context = ast_strdupa(ast_channel_context(chan));

	if (args.options) {
		char *opts[OPT_ARG_ARRAY_SIZE];
		char tmp;

		ast_app_parse_options(spy_opts, &flags, opts, args.options);
		if (ast_test_flag(&flags, OPTION_GROUP))
			mygroup = opts[OPT_ARG_GROUP];

		if (ast_test_flag(&flags, OPTION_RECORD) &&
			!(recbase = opts[OPT_ARG_RECORD]))
			recbase = "chanspy";

		if (ast_test_flag(&flags, OPTION_DTMF_EXIT) && opts[OPT_ARG_EXIT]) {
			tmp = opts[OPT_ARG_EXIT][0];
			if (strchr("0123456789*#", tmp) && tmp != '\0') {
				user_options.exit = tmp;
			} else {
				ast_log(LOG_NOTICE, "Argument for option 'x' must be a valid DTMF digit.\n");
			}
		}

		if (ast_test_flag(&flags, OPTION_DTMF_CYCLE) && opts[OPT_ARG_CYCLE]) {
			tmp = opts[OPT_ARG_CYCLE][0];
			if (strchr("0123456789*#", tmp) && tmp != '\0') {
				user_options.cycle = tmp;
			} else {
				ast_log(LOG_NOTICE, "Argument for option 'c' must be a valid DTMF digit.\n");
			}
		}

		if (ast_test_flag(&flags, OPTION_VOLUME) && opts[OPT_ARG_VOLUME]) {
			int vol;

			if ((sscanf(opts[OPT_ARG_VOLUME], "%30d", &vol) != 1) || (vol > 4) || (vol < -4))
				ast_log(LOG_NOTICE, "Volume factor must be a number between -4 and 4\n");
			else
				volfactor = vol;
		}

		if (ast_test_flag(&flags, OPTION_PRIVATE))
			ast_set_flag(&flags, OPTION_WHISPER);

		if (ast_test_flag(&flags, OPTION_NAME)) {
			if (!ast_strlen_zero(opts[OPT_ARG_NAME])) {
				char *delimiter;
				if ((delimiter = strchr(opts[OPT_ARG_NAME], '@'))) {
					mailbox = opts[OPT_ARG_NAME];
					*delimiter++ = '\0';
					name_context = delimiter;
				} else {
					mailbox = opts[OPT_ARG_NAME];
				}
			}
		}

	} else {
		/* Coverity - This uninit_use should be ignored since this macro initializes the flags */
		ast_clear_flag(&flags, AST_FLAGS_ALL);
	}

	ast_format_copy(&oldwf, ast_channel_writeformat(chan));
	if (ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR) < 0) {
		ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
		return -1;
	}

	if (recbase) {
		char filename[PATH_MAX];

		snprintf(filename, sizeof(filename), "%s/%s.%d.raw", ast_config_AST_MONITOR_DIR, recbase, (int) time(NULL));
		if ((fd = open(filename, O_CREAT | O_WRONLY | O_TRUNC, AST_FILE_MODE)) <= 0) {
			ast_log(LOG_WARNING, "Cannot open '%s' for recording\n", filename);
			fd = 0;
		}
	}


	res = common_exec(chan, &flags, volfactor, fd, &user_options, mygroup, NULL, NULL, exten, args.context, mailbox, name_context);

	if (fd)
		close(fd);

	if (oldwf.id && ast_set_write_format(chan, &oldwf) < 0)
		ast_log(LOG_ERROR, "Could Not Set Write Format.\n");

	return res;
}
コード例 #15
0
static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, int length, char *intkeys)
{
	int res = 0;
	int fds[2];
	int needed = 0;
	struct ast_format owriteformat;
	struct ast_frame *f;
	struct myframe {
		struct ast_frame f;
		char offset[AST_FRIENDLY_OFFSET];
		char frdata[2048];
	} myf = {
		.f = { 0, },
	};

	ast_format_clear(&owriteformat);
	if (pipe(fds)) {
		ast_log(LOG_WARNING, "Unable to create pipe\n");
		return -1;
	}

	/* Answer if it's not already going */
	if (ast_channel_state(chan) != AST_STATE_UP)
		ast_answer(chan);
	ast_stopstream(chan);
	ast_indicate(chan, -1);
	
	ast_format_copy(&owriteformat, ast_channel_writeformat(chan));
	res = ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR);
	if (res < 0) {
		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
		return -1;
	}
	
	res = send_waveform_to_fd(waveform, length, fds[1]);
	if (res >= 0) {
		/* Order is important -- there's almost always going to be mp3...  we want to prioritize the
		   user */
		for (;;) {
			res = ast_waitfor(chan, 1000);
			if (res < 1) {
				res = -1;
				break;
			}
			f = ast_read(chan);
			if (!f) {
				ast_log(LOG_WARNING, "Null frame == hangup() detected\n");
				res = -1;
				break;
			}
			if (f->frametype == AST_FRAME_DTMF) {
				ast_debug(1, "User pressed a key\n");
				if (intkeys && strchr(intkeys, f->subclass.integer)) {
					res = f->subclass.integer;
					ast_frfree(f);
					break;
				}
			}
			if (f->frametype == AST_FRAME_VOICE) {
				/* Treat as a generator */
				needed = f->samples * 2;
				if (needed > sizeof(myf.frdata)) {
					ast_log(LOG_WARNING, "Only able to deliver %d of %d requested samples\n",
						(int)sizeof(myf.frdata) / 2, needed/2);
					needed = sizeof(myf.frdata);
				}
				res = read(fds[0], myf.frdata, needed);
				if (res > 0) {
					myf.f.frametype = AST_FRAME_VOICE;
					ast_format_set(&myf.f.subclass.format, AST_FORMAT_SLINEAR, 0);
					myf.f.datalen = res;
					myf.f.samples = res / 2;
					myf.f.offset = AST_FRIENDLY_OFFSET;
					myf.f.src = __PRETTY_FUNCTION__;
					myf.f.data.ptr = myf.frdata;
					if (ast_write(chan, &myf.f) < 0) {
						res = -1;
						ast_frfree(f);
						break;
					}
					if (res < needed) { /* last frame */
						ast_debug(1, "Last frame\n");
						res = 0;
						ast_frfree(f);
						break;
					}
				} else {
					ast_debug(1, "No more waveform\n");
					res = 0;
				}
			}
			ast_frfree(f);
		}
	}
	close(fds[0]);
	close(fds[1]);

	if (!res && owriteformat.id)
		ast_set_write_format(chan, &owriteformat);
	return res;
}
コード例 #16
0
ファイル: app_swift.c プロジェクト: mojolingo/app_swift
static int app_exec(struct ast_channel *chan, const char *data)
{
	int res = 0, argc = 0, max_digits = 0, timeout = 0, alreadyran = 0;
	int ms, len, availatend;
	char *argv[3], *parse = NULL, *text = NULL, *rc = NULL;
	char tmp_exten[2], results[20];
	struct ast_module_user *u;
	struct ast_frame *f;
	struct timeval next;
	struct stuff *ps;
	struct ast_format oldwf;

	struct myframe {
		struct ast_frame f;
		char offset[AST_FRIENDLY_OFFSET];
		char frdata[framesize];
	} myf;

	ast_format_clear(&oldwf);

	swift_engine *engine;
	swift_port *port = NULL;
	swift_voice *voice;
	swift_params *params;
	swift_result_t sresult;
	swift_background_t tts_stream;
	unsigned int event_mask;

	memset(results, 0 ,20);
	memset(tmp_exten, 0, 2);
	memset(argv, 0, 3);

	parse = ast_strdupa(data);
	u = ast_module_user_add(chan);
	argc = ast_app_separate_args(parse, '|', argv, 3);
	text = argv[0];

	if (!ast_strlen_zero(argv[1])) {
		timeout = strtol(argv[1], NULL, 0);
	}
	if (!ast_strlen_zero(argv[2])) {
		max_digits = strtol(argv[2], NULL, 0);
	}
	if (ast_strlen_zero(text)) {
		ast_log(LOG_WARNING, "%s requires text to speak!\n", app);
		return -1;
	}
	if (!ast_strlen_zero(text)) {
		ast_log(LOG_DEBUG, "Text to Speak : %s\n", text);
	}
	if (timeout > 0) {
		ast_log(LOG_DEBUG, "Timeout : %d\n", timeout);
	}
	if (max_digits > 0) {
		ast_log(LOG_DEBUG, "Max Digits : %d\n", max_digits);
	}

	ps = malloc(sizeof(struct stuff));
	swift_init_stuff(ps);

	/* Setup synthesis */

	if ((engine = swift_engine_open(NULL)) == NULL) {
		ast_log(LOG_ERROR, "Failed to open Swift Engine.\n");
		goto exception;
	}

	params = swift_params_new(NULL);
	swift_params_set_string(params, "audio/encoding", "ulaw");
	swift_params_set_string(params, "audio/sampling-rate", "8000");
	swift_params_set_string(params, "audio/output-format", "raw");
	swift_params_set_string(params, "tts/text-encoding", "utf-8");

	/* Additional swift parameters
	 *
	 * swift_params_set_float(params, "speech/pitch/shift", 1.0);
	 * swift_params_set_int(params, "speech/rate", 150);
	 * swift_params_set_int(params, "audio/volume", 110);
	 * swift_params_set_int(params, "audio/deadair", 0);
	 */

	if ((port = swift_port_open(engine, params)) == NULL) {
		ast_log(LOG_ERROR, "Failed to open Swift Port.\n");
		goto exception;
	}
	if ((voice = swift_port_set_voice_by_name(port, cfg_voice)) == NULL) {
		ast_log(LOG_ERROR, "Failed to set voice.\n");
		goto exception;
	}

	event_mask = SWIFT_EVENT_AUDIO | SWIFT_EVENT_END;
	swift_port_set_callback(port, &swift_cb, event_mask, ps);

	if (SWIFT_FAILED(swift_port_speak_text(port, text, 0, NULL, &tts_stream, NULL))) {
		ast_log(LOG_ERROR, "Failed to speak.\n");
		goto exception;
	}
	if (chan->_state != AST_STATE_UP) {
		ast_answer(chan);
	}

	ast_stopstream(chan);

	ast_format_copy(&oldwf, &chan->writeformat);

	if (ast_set_write_format_by_id(chan, AST_FORMAT_ULAW) < 0) {
		ast_log(LOG_WARNING, "Unable to set write format.\n");
		goto exception;
	}

	res = 0;

	/* Wait 100ms first for synthesis to start crankin'; if that's not
	 * enough the
	 */

	next = ast_tvadd(ast_tvnow(), ast_tv(0, 100000));

	while (swift_generator_running(ps)) {
		ms = ast_tvdiff_ms(next, ast_tvnow());

		if (ms <= 0) {
			if (swift_bytes_available(ps) > 0) {
				ASTOBJ_WRLOCK(ps);
				len = fmin(framesize, ps->qc);
				availatend = cfg_buffer_size - (ps->pq_r - ps->q);

				if (len > availatend) {
					/* read #1: to end of q buf */
					memcpy(myf.frdata, ps->pq_r, availatend);
					ps->qc -= availatend;

					/* read #2: reset to start of q buf and get rest */
					ps->pq_r = ps->q;
					memcpy(myf.frdata + availatend, ps->pq_r, len - availatend);
					ps->qc -= len - availatend;
					ps->pq_r += len - availatend;
				} else {
					ast_log(LOG_DEBUG, "Easy read; %d bytes and %d at end, %d free\n", len, availatend, cfg_buffer_size - ps->qc);
					memcpy(myf.frdata, ps->pq_r, len);
					ps->qc -= len;
					ps->pq_r += len;
				}

				myf.f.frametype = AST_FRAME_VOICE;
				ast_format_set(&myf.f.subclass.format, AST_FORMAT_ULAW, 0);
				myf.f.datalen = len;
				myf.f.samples = len;
				myf.f.data.ptr = myf.frdata;
				myf.f.mallocd = 0;
				myf.f.offset = AST_FRIENDLY_OFFSET;
				myf.f.src = __PRETTY_FUNCTION__;
				myf.f.delivery.tv_sec = 0;
				myf.f.delivery.tv_usec = 0;

				if (ast_write(chan, &myf.f) < 0) {
					ast_log(LOG_DEBUG, "ast_write failed\n");
				}

				ast_log(LOG_DEBUG, "wrote a frame of %d\n", len);

				if (ps->qc < 0) {
					ast_log(LOG_DEBUG, "queue claims to contain negative bytes. Huh? qc < 0\n");
				}

				ASTOBJ_UNLOCK(ps);
				next = ast_tvadd(next, ast_samp2tv(myf.f.samples, samplerate));
			} else {
				next = ast_tvadd(next, ast_samp2tv(framesize / 2, samplerate));
				ast_log(LOG_DEBUG, "Whoops, writer starved for audio\n");
			}
		} else {
			ms = ast_waitfor(chan, ms);

			if (ms < 0) {
				ast_log(LOG_DEBUG, "Hangup detected\n");
				res = -1;
				ASTOBJ_WRLOCK(ps);
				ps->immediate_exit = 1;
				ASTOBJ_UNLOCK(ps);
			} else if (ms) {
				f = ast_read(chan);

				if (!f) {
					ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
					res = -1;
					ASTOBJ_WRLOCK(ps);
					ps->immediate_exit = 1;
					ASTOBJ_UNLOCK(ps);
				} else {
					if (f->frametype == AST_FRAME_DTMF && timeout > 0 && max_digits > 0) {
						char originalDTMF = f->subclass.integer;
						alreadyran = 1;
						res = 0;
						ASTOBJ_WRLOCK(ps);
						ps->immediate_exit = 1;
						ASTOBJ_UNLOCK(ps);

						if (max_digits > 1) {
							rc = listen_for_dtmf(chan, timeout, max_digits - 1);
						}
						if (rc) {
							sprintf(results, "%c%s", originalDTMF, rc);
						} else {
							sprintf(results, "%c", originalDTMF);
						}

						ast_log(LOG_NOTICE, "DTMF = %s\n", results);
						pbx_builtin_setvar_helper(chan, "SWIFT_DTMF", results);
					}
				}
				ast_frfree(f);
			}
		}

		ASTOBJ_RDLOCK(ps);

		if (ps->immediate_exit && !ps->generating_done) {
			if (SWIFT_FAILED(sresult = swift_port_stop(port, tts_stream, SWIFT_EVENT_NOW))) {
				ast_log(LOG_NOTICE, "Early top of swift port failed\n");
			}
		}

		ASTOBJ_UNLOCK(ps);
	}
	if (alreadyran == 0 && timeout > 0 && max_digits > 0) {
		rc = listen_for_dtmf(chan, timeout, max_digits);

		if (rc != NULL) {
			sprintf(results, "%s", rc);
			ast_log(LOG_NOTICE, "DTMF = %s\n", results);
			pbx_builtin_setvar_helper(chan, "SWIFT_DTMF", results);
		}
	}
	if (max_digits >= 1 && results != NULL) {
		if (cfg_goto_exten) {
			ast_log(LOG_NOTICE, "GoTo(%s|%s|%d) : ", chan->context, results, 1);

			if (ast_exists_extension (chan, chan->context, results, 1, chan->caller.id.number.str)) {
				ast_log(LOG_NOTICE, "OK\n");
				ast_copy_string(chan->exten, results, sizeof(chan->exten) - 1);
				chan->priority = 0;
			} else {
				ast_log(LOG_NOTICE, "FAILED\n");
			}
		}
	}

	exception:

	if (port != NULL) {
		swift_port_close(port);
	}
	if (engine != NULL) {
		swift_engine_close(engine);
	}
	if (ps && ps->q) {
		ast_free(ps->q);
		ps->q = NULL;
	}
	if (ps) {
		ast_free(ps);
		ps = NULL;
	}
	if (!res) {
		ast_set_write_format(chan, &oldwf);
	}

	ast_module_user_remove(u);
	return res;
}