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asterisk/apps/app_fax.c

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/*
* Asterisk -- A telephony toolkit for Linux.
*
* Simple fax applications
*
* 2007-2008, Dmitry Andrianov <asterisk@dima.spb.ru>
*
* Code based on original implementation by Steve Underwood <steveu@coppice.org>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*
*/
/*** MODULEINFO
<defaultenabled>no</defaultenabled>
<depend>spandsp</depend>
<conflict>res_fax</conflict>
<support_level>extended</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <inttypes.h>
#include <pthread.h>
#include <errno.h>
#include <tiffio.h>
#define SPANDSP_EXPOSE_INTERNAL_STRUCTURES
#include <spandsp.h>
#include <spandsp/version.h>
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/app.h"
#include "asterisk/dsp.h"
#include "asterisk/module.h"
#include "asterisk/manager.h"
/*** DOCUMENTATION
<application name="SendFAX" language="en_US">
<synopsis>
Send a Fax
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>Filename of TIFF file to fax</para>
</parameter>
<parameter name="a" required="false">
<para>Makes the application behave as the answering machine</para>
<para>(Default behavior is as calling machine)</para>
</parameter>
</syntax>
<description>
<para>Send a given TIFF file to the channel as a FAX.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="LOCALSTATIONID">
<para>To identify itself to the remote end</para>
</variable>
<variable name="LOCALHEADERINFO">
<para>To generate a header line on each page</para>
</variable>
<variable name="FAXSTATUS">
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
<variable name="FAXERROR">
<para>Cause of failure</para>
</variable>
<variable name="REMOTESTATIONID">
<para>The CSID of the remote side</para>
</variable>
<variable name="FAXPAGES">
<para>Number of pages sent</para>
</variable>
<variable name="FAXBITRATE">
<para>Transmission rate</para>
</variable>
<variable name="FAXRESOLUTION">
<para>Resolution of sent fax</para>
</variable>
</variablelist>
</description>
</application>
<application name="ReceiveFAX" language="en_US">
<synopsis>
Receive a Fax
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>Filename of TIFF file save incoming fax</para>
</parameter>
<parameter name="c" required="false">
<para>Makes the application behave as the calling machine</para>
<para>(Default behavior is as answering machine)</para>
</parameter>
</syntax>
<description>
<para>Receives a FAX from the channel into the given filename
overwriting the file if it already exists.</para>
<para>File created will be in TIFF format.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="LOCALSTATIONID">
<para>To identify itself to the remote end</para>
</variable>
<variable name="LOCALHEADERINFO">
<para>To generate a header line on each page</para>
</variable>
<variable name="FAXSTATUS">
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
<variable name="FAXERROR">
<para>Cause of failure</para>
</variable>
<variable name="REMOTESTATIONID">
<para>The CSID of the remote side</para>
</variable>
<variable name="FAXPAGES">
<para>Number of pages sent</para>
</variable>
<variable name="FAXBITRATE">
<para>Transmission rate</para>
</variable>
<variable name="FAXRESOLUTION">
<para>Resolution of sent fax</para>
</variable>
</variablelist>
</description>
</application>
***/
static const char app_sndfax_name[] = "SendFAX";
static const char app_rcvfax_name[] = "ReceiveFAX";
#define MAX_SAMPLES 240
/* Watchdog. I have seen situations when remote fax disconnects (because of poor line
quality) while SpanDSP continues staying in T30_STATE_IV_CTC state forever.
To avoid this, we terminate when we see that T30 state does not change for 5 minutes.
We also terminate application when more than 30 minutes passed regardless of
state changes. This is just a precaution measure - no fax should take that long */
#define WATCHDOG_TOTAL_TIMEOUT 30 * 60
#define WATCHDOG_STATE_TIMEOUT 5 * 60
typedef struct {
struct ast_channel *chan;
enum ast_t38_state t38state; /* T38 state of the channel */
int direction; /* Fax direction: 0 - receiving, 1 - sending */
int caller_mode;
char *file_name;
struct ast_control_t38_parameters t38parameters;
volatile int finished;
} fax_session;
static void span_message(int level, const char *msg)
{
if (level == SPAN_LOG_ERROR) {
ast_log(LOG_ERROR, "%s", msg);
} else if (level == SPAN_LOG_WARNING) {
ast_log(LOG_WARNING, "%s", msg);
} else {
ast_debug(1, "%s", msg);
}
}
static int t38_tx_packet_handler(t38_core_state_t *s, void *user_data, const uint8_t *buf, int len, int count)
{
struct ast_channel *chan = (struct ast_channel *) user_data;
struct ast_frame outf = {
.frametype = AST_FRAME_MODEM,
.subclass.integer = AST_MODEM_T38,
.src = __FUNCTION__,
};
/* TODO: Asterisk does not provide means of resending the same packet multiple
times so count is ignored at the moment */
AST_FRAME_SET_BUFFER(&outf, buf, 0, len);
if (ast_write(chan, &outf) < 0) {
ast_log(LOG_WARNING, "Unable to write frame to channel; %s\n", strerror(errno));
return -1;
}
return 0;
}
static void phase_e_handler(t30_state_t *f, void *user_data, int result)
{
const char *local_ident;
const char *far_ident;
char buf[20];
fax_session *s = (fax_session *) user_data;
t30_stats_t stat;
int pages_transferred;
ast_debug(1, "Fax phase E handler. result=%d\n", result);
t30_get_transfer_statistics(f, &stat);
s = (fax_session *) user_data;
if (result != T30_ERR_OK) {
s->finished = -1;
/* FAXSTATUS is already set to FAILED */
pbx_builtin_setvar_helper(s->chan, "FAXERROR", t30_completion_code_to_str(result));
ast_log(LOG_WARNING, "Error transmitting fax. result=%d: %s.\n", result, t30_completion_code_to_str(result));
return;
}
s->finished = 1;
local_ident = S_OR(t30_get_tx_ident(f), "");
far_ident = S_OR(t30_get_rx_ident(f), "");
pbx_builtin_setvar_helper(s->chan, "FAXSTATUS", "SUCCESS");
pbx_builtin_setvar_helper(s->chan, "FAXERROR", NULL);
pbx_builtin_setvar_helper(s->chan, "REMOTESTATIONID", far_ident);
#if SPANDSP_RELEASE_DATE >= 20090220
pages_transferred = (s->direction) ? stat.pages_tx : stat.pages_rx;
#else
pages_transferred = stat.pages_transferred;
#endif
snprintf(buf, sizeof(buf), "%d", pages_transferred);
pbx_builtin_setvar_helper(s->chan, "FAXPAGES", buf);
snprintf(buf, sizeof(buf), "%d", stat.y_resolution);
pbx_builtin_setvar_helper(s->chan, "FAXRESOLUTION", buf);
snprintf(buf, sizeof(buf), "%d", stat.bit_rate);
pbx_builtin_setvar_helper(s->chan, "FAXBITRATE", buf);
ast_debug(1, "Fax transmitted successfully.\n");
ast_debug(1, " Remote station ID: %s\n", far_ident);
ast_debug(1, " Pages transferred: %d\n", pages_transferred);
ast_debug(1, " Image resolution: %d x %d\n", stat.x_resolution, stat.y_resolution);
ast_debug(1, " Transfer Rate: %d\n", stat.bit_rate);
ast_manager_event(s->chan, EVENT_FLAG_CALL,
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
s->direction ? "FaxSent" : "FaxReceived",
"Channel: %s\r\n"
"Exten: %s\r\n"
"CallerID: %s\r\n"
"CallerIDName: %s\r\n"
"ConnectedLineNum: %s\r\n"
"ConnectedLineName: %s\r\n"
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
"RemoteStationID: %s\r\n"
"LocalStationID: %s\r\n"
"PagesTransferred: %d\r\n"
"Resolution: %d\r\n"
"TransferRate: %d\r\n"
"FileName: %s\r\n",
s->chan->name,
s->chan->exten,
S_COR(s->chan->caller.id.number.valid, s->chan->caller.id.number.str, ""),
S_COR(s->chan->caller.id.name.valid, s->chan->caller.id.name.str, ""),
S_COR(s->chan->connected.id.number.valid, s->chan->connected.id.number.str, ""),
S_COR(s->chan->connected.id.name.valid, s->chan->connected.id.name.str, ""),
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
far_ident,
local_ident,
pages_transferred,
stat.y_resolution,
stat.bit_rate,
s->file_name);
}
/* === Helper functions to configure fax === */
/* Setup SPAN logging according to Asterisk debug level */
static int set_logging(logging_state_t *state)
{
int level = SPAN_LOG_WARNING + option_debug;
span_log_set_message_handler(state, span_message);
span_log_set_level(state, SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | level);
return 0;
}
static void set_local_info(t30_state_t *state, fax_session *s)
{
const char *x;
x = pbx_builtin_getvar_helper(s->chan, "LOCALSTATIONID");
if (!ast_strlen_zero(x))
t30_set_tx_ident(state, x);
x = pbx_builtin_getvar_helper(s->chan, "LOCALHEADERINFO");
if (!ast_strlen_zero(x))
t30_set_tx_page_header_info(state, x);
}
static void set_file(t30_state_t *state, fax_session *s)
{
if (s->direction)
t30_set_tx_file(state, s->file_name, -1, -1);
else
t30_set_rx_file(state, s->file_name, -1);
}
static void set_ecm(t30_state_t *state, int ecm)
{
t30_set_ecm_capability(state, ecm);
t30_set_supported_compressions(state, T30_SUPPORT_T4_1D_COMPRESSION | T30_SUPPORT_T4_2D_COMPRESSION | T30_SUPPORT_T6_COMPRESSION);
}
/* === Generator === */
/* This function is only needed to return passed params so
generator_activate will save it to channel's generatordata */
static void *fax_generator_alloc(struct ast_channel *chan, void *params)
{
return params;
}
static int fax_generator_generate(struct ast_channel *chan, void *data, int len, int samples)
{
fax_state_t *fax = (fax_state_t*) data;
uint8_t buffer[AST_FRIENDLY_OFFSET + MAX_SAMPLES * sizeof(uint16_t)];
int16_t *buf = (int16_t *) (buffer + AST_FRIENDLY_OFFSET);
struct ast_frame outf = {
.frametype = AST_FRAME_VOICE,
.src = __FUNCTION__,
};
ast_format_set(&outf.subclass.format, AST_FORMAT_SLINEAR, 0);
if (samples > MAX_SAMPLES) {
ast_log(LOG_WARNING, "Only generating %d samples, where %d requested\n", MAX_SAMPLES, samples);
samples = MAX_SAMPLES;
}
if ((len = fax_tx(fax, buf, samples)) > 0) {
outf.samples = len;
AST_FRAME_SET_BUFFER(&outf, buffer, AST_FRIENDLY_OFFSET, len * sizeof(int16_t));
if (ast_write(chan, &outf) < 0) {
ast_log(LOG_WARNING, "Failed to write frame to '%s': %s\n", chan->name, strerror(errno));
return -1;
}
}
return 0;
}
static struct ast_generator generator = {
alloc: fax_generator_alloc,
generate: fax_generator_generate,
};
/* === Transmission === */
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", s->chan->name);
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", s->chan->name);
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", s->chan->name);
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", s->chan->name);
res = 1;
break;
case AST_T38_REFUSED:
ast_log(LOG_WARNING, "channel '%s' refused to negotiate T.38\n", s->chan->name);
break;
default:
ast_log(LOG_ERROR, "channel '%s' failed to negotiate T.38\n", s->chan->name);
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, &s->chan->readformat);
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, &s->chan->writeformat);
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", s->chan->name);
}
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", s->chan->name);
}
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;
}
Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208464 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-23 21:57:24 +00:00
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", s->chan->name);
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", s->chan->name);
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", s->chan->name);
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", s->chan->name);
break;
case AST_T38_REFUSED:
ast_log(LOG_WARNING, "channel '%s' refused to disable T.38\n", s->chan->name);
break;
default:
ast_log(LOG_ERROR, "channel '%s' failed to disable T.38\n", s->chan->name);
break;
}
ast_frfree(inf);
break;
}
ast_frfree(inf);
}
}
}
return res;
}
static int transmit(fax_session *s)
{
int res = 0;
/* Clear all channel variables which to be set by the application.
Pre-set status to error so in case of any problems we can just leave */
pbx_builtin_setvar_helper(s->chan, "FAXSTATUS", "FAILED");
pbx_builtin_setvar_helper(s->chan, "FAXERROR", "Channel problems");
pbx_builtin_setvar_helper(s->chan, "FAXMODE", NULL);
pbx_builtin_setvar_helper(s->chan, "REMOTESTATIONID", NULL);
pbx_builtin_setvar_helper(s->chan, "FAXPAGES", "0");
pbx_builtin_setvar_helper(s->chan, "FAXRESOLUTION", NULL);
pbx_builtin_setvar_helper(s->chan, "FAXBITRATE", NULL);
if (s->chan->_state != AST_STATE_UP) {
/* Shouldn't need this, but checking to see if channel is already answered
* Theoretically asterisk should already have answered before running the app */
res = ast_answer(s->chan);
if (res) {
ast_log(LOG_WARNING, "Could not answer channel '%s'\n", s->chan->name);
return res;
}
}
s->t38state = ast_channel_get_t38_state(s->chan);
if (s->t38state != T38_STATE_NEGOTIATED) {
/* T38 is not negotiated on the channel yet. First start regular transmission. If it switches to T38, follow */
pbx_builtin_setvar_helper(s->chan, "FAXMODE", "audio");
res = transmit_audio(s);
if (res > 0) {
/* transmit_audio reports switchover to T38. Update t38state */
s->t38state = ast_channel_get_t38_state(s->chan);
if (s->t38state != T38_STATE_NEGOTIATED) {
ast_log(LOG_ERROR, "Audio loop reports T38 switchover but t38state != T38_STATE_NEGOTIATED\n");
}
}
}
if (s->t38state == T38_STATE_NEGOTIATED) {
pbx_builtin_setvar_helper(s->chan, "FAXMODE", "T38");
res = transmit_t38(s);
}
if (res) {
ast_log(LOG_WARNING, "Transmission error\n");
res = -1;
} else if (s->finished < 0) {
ast_log(LOG_WARNING, "Transmission failed\n");
} else if (s->finished > 0) {
ast_debug(1, "Transmission finished Ok\n");
}
return res;
}
/* === Application functions === */
static int sndfax_exec(struct ast_channel *chan, const char *data)
{
int res = 0;
char *parse;
fax_session session = { 0, };
char restore_digit_detect = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(file_name);
AST_APP_ARG(options);
);
if (chan == NULL) {
ast_log(LOG_ERROR, "Fax channel is NULL. Giving up.\n");
return -1;
}
/* The next few lines of code parse out the filename and header from the input string */
if (ast_strlen_zero(data)) {
/* No data implies no filename or anything is present */
ast_log(LOG_ERROR, "SendFAX requires an argument (filename)\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
session.caller_mode = TRUE;
if (args.options) {
if (strchr(args.options, 'a'))
session.caller_mode = FALSE;
}
/* Done parsing */
session.direction = 1;
session.file_name = args.file_name;
session.chan = chan;
session.finished = 0;
/* get current digit detection mode, then disable digit detection if enabled */
{
int dummy = sizeof(restore_digit_detect);
ast_channel_queryoption(chan, AST_OPTION_DIGIT_DETECT, &restore_digit_detect, &dummy, 0);
}
if (restore_digit_detect) {
char new_digit_detect = 0;
ast_channel_setoption(chan, AST_OPTION_DIGIT_DETECT, &new_digit_detect, sizeof(new_digit_detect), 0);
}
/* disable FAX tone detection if enabled */
{
char new_fax_detect = 0;
ast_channel_setoption(chan, AST_OPTION_FAX_DETECT, &new_fax_detect, sizeof(new_fax_detect), 0);
}
res = transmit(&session);
if (restore_digit_detect) {
ast_channel_setoption(chan, AST_OPTION_DIGIT_DETECT, &restore_digit_detect, sizeof(restore_digit_detect), 0);
}
return res;
}
static int rcvfax_exec(struct ast_channel *chan, const char *data)
{
int res = 0;
char *parse;
fax_session session;
char restore_digit_detect = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(file_name);
AST_APP_ARG(options);
);
if (chan == NULL) {
ast_log(LOG_ERROR, "Fax channel is NULL. Giving up.\n");
return -1;
}
/* The next few lines of code parse out the filename and header from the input string */
if (ast_strlen_zero(data)) {
/* No data implies no filename or anything is present */
ast_log(LOG_ERROR, "ReceiveFAX requires an argument (filename)\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
session.caller_mode = FALSE;
if (args.options) {
if (strchr(args.options, 'c'))
session.caller_mode = TRUE;
}
/* Done parsing */
session.direction = 0;
session.file_name = args.file_name;
session.chan = chan;
session.finished = 0;
/* get current digit detection mode, then disable digit detection if enabled */
{
int dummy = sizeof(restore_digit_detect);
ast_channel_queryoption(chan, AST_OPTION_DIGIT_DETECT, &restore_digit_detect, &dummy, 0);
}
if (restore_digit_detect) {
char new_digit_detect = 0;
ast_channel_setoption(chan, AST_OPTION_DIGIT_DETECT, &new_digit_detect, sizeof(new_digit_detect), 0);
}
/* disable FAX tone detection if enabled */
{
char new_fax_detect = 0;
ast_channel_setoption(chan, AST_OPTION_FAX_DETECT, &new_fax_detect, sizeof(new_fax_detect), 0);
}
res = transmit(&session);
if (restore_digit_detect) {
ast_channel_setoption(chan, AST_OPTION_DIGIT_DETECT, &restore_digit_detect, sizeof(restore_digit_detect), 0);
}
return res;
}
static int unload_module(void)
{
int res;
res = ast_unregister_application(app_sndfax_name);
res |= ast_unregister_application(app_rcvfax_name);
return res;
}
static int load_module(void)
{
int res ;
res = ast_register_application_xml(app_sndfax_name, sndfax_exec);
res |= ast_register_application_xml(app_rcvfax_name, rcvfax_exec);
/* The default SPAN message handler prints to stderr. It is something we do not want */
span_set_message_handler(NULL);
return res;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Simple FAX Application",
.load = load_module,
.unload = unload_module,
);