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asterisk/channels/chan_phone.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Generic Linux Telephony Interface driver
*
* \author Mark Spencer <markster@digium.com>
*
* \ingroup channel_drivers
*/
/*** MODULEINFO
<depend>ixjuser</depend>
<support_level>extended</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <ctype.h>
#include <sys/socket.h>
#include <sys/time.h>
#include <arpa/inet.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <signal.h>
#ifdef HAVE_LINUX_COMPILER_H
#include <linux/compiler.h>
#endif
#include <linux/telephony.h>
/* Still use some IXJ specific stuff */
#include <linux/version.h>
#include <linux/ixjuser.h>
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/utils.h"
#include "asterisk/callerid.h"
#include "asterisk/causes.h"
#include "asterisk/stringfields.h"
#include "asterisk/musiconhold.h"
#include "chan_phone.h"
#ifdef QTI_PHONEJACK_TJ_PCI /* check for the newer quicknet driver v.3.1.0 which has this symbol */
#define QNDRV_VER 310
#else
#define QNDRV_VER 100
#endif
#if QNDRV_VER > 100
#ifdef __linux__
#define IXJ_PHONE_RING_START(x) ioctl(p->fd, PHONE_RING_START, &x);
#else /* FreeBSD and others */
#define IXJ_PHONE_RING_START(x) ioctl(p->fd, PHONE_RING_START, x);
#endif /* __linux__ */
#else /* older driver */
#define IXJ_PHONE_RING_START(x) ioctl(p->fd, PHONE_RING_START, &x);
#endif
#define DEFAULT_CALLER_ID "Unknown"
#define PHONE_MAX_BUF 480
#define DEFAULT_GAIN 0x100
static const char tdesc[] = "Standard Linux Telephony API Driver";
static const char config[] = "phone.conf";
/* Default context for dialtone mode */
static char context[AST_MAX_EXTENSION] = "default";
/* Default language */
static char language[MAX_LANGUAGE] = "";
static int echocancel = AEC_OFF;
static int silencesupression = 0;
static struct ast_format_cap *prefcap;
/* Protect the interface list (of phone_pvt's) */
AST_MUTEX_DEFINE_STATIC(iflock);
/* Protect the monitoring thread, so only one process can kill or start it, and not
when it's doing something critical. */
AST_MUTEX_DEFINE_STATIC(monlock);
/* Boolean value whether the monitoring thread shall continue. */
static unsigned int monitor;
/* This is the thread for the monitor which checks for input on the channels
which are not currently in use. */
static pthread_t monitor_thread = AST_PTHREADT_NULL;
static int restart_monitor(void);
/* The private structures of the Phone Jack channels are linked for
selecting outgoing channels */
#define MODE_DIALTONE 1
#define MODE_IMMEDIATE 2
#define MODE_FXO 3
#define MODE_FXS 4
#define MODE_SIGMA 5
static struct phone_pvt {
int fd; /* Raw file descriptor for this device */
struct ast_channel *owner; /* Channel we belong to, possibly NULL */
int mode; /* Is this in the */
struct ast_format lastformat; /* Last output format */
struct ast_format lastinput; /* Last input format */
int ministate; /* Miniature state, for dialtone mode */
char dev[256]; /* Device name */
struct phone_pvt *next; /* Next channel in list */
struct ast_frame fr; /* Frame */
char offset[AST_FRIENDLY_OFFSET];
char buf[PHONE_MAX_BUF]; /* Static buffer for reading frames */
int obuflen;
int dialtone;
int txgain, rxgain; /* gain control for playing, recording */
/* 0x100 - 1.0, 0x200 - 2.0, 0x80 - 0.5 */
int cpt; /* Call Progress Tone playing? */
int silencesupression;
char context[AST_MAX_EXTENSION];
char obuf[PHONE_MAX_BUF * 2];
char ext[AST_MAX_EXTENSION];
char language[MAX_LANGUAGE];
char cid_num[AST_MAX_EXTENSION];
char cid_name[AST_MAX_EXTENSION];
} *iflist = NULL;
static char cid_num[AST_MAX_EXTENSION];
static char cid_name[AST_MAX_EXTENSION];
static struct ast_channel *phone_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause);
static int phone_digit_begin(struct ast_channel *ast, char digit);
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static int phone_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
static int phone_call(struct ast_channel *ast, char *dest, int timeout);
static int phone_hangup(struct ast_channel *ast);
static int phone_answer(struct ast_channel *ast);
static struct ast_frame *phone_read(struct ast_channel *ast);
static int phone_write(struct ast_channel *ast, struct ast_frame *frame);
static struct ast_frame *phone_exception(struct ast_channel *ast);
static int phone_send_text(struct ast_channel *ast, const char *text);
static int phone_fixup(struct ast_channel *old, struct ast_channel *new);
static int phone_indicate(struct ast_channel *chan, int condition, const void *data, size_t datalen);
static struct ast_channel_tech phone_tech = {
.type = "Phone",
.description = tdesc,
.requester = phone_request,
.send_digit_begin = phone_digit_begin,
.send_digit_end = phone_digit_end,
.call = phone_call,
.hangup = phone_hangup,
.answer = phone_answer,
.read = phone_read,
.write = phone_write,
.exception = phone_exception,
.indicate = phone_indicate,
.fixup = phone_fixup
};
static struct ast_channel_tech phone_tech_fxs = {
.type = "Phone",
.description = tdesc,
.requester = phone_request,
.send_digit_begin = phone_digit_begin,
.send_digit_end = phone_digit_end,
.call = phone_call,
.hangup = phone_hangup,
.answer = phone_answer,
.read = phone_read,
.write = phone_write,
.exception = phone_exception,
.write_video = phone_write,
.send_text = phone_send_text,
.indicate = phone_indicate,
.fixup = phone_fixup
};
static struct ast_channel_tech *cur_tech;
static int phone_indicate(struct ast_channel *chan, int condition, const void *data, size_t datalen)
{
struct phone_pvt *p = chan->tech_pvt;
int res=-1;
ast_debug(1, "Requested indication %d on channel %s\n", condition, chan->name);
switch(condition) {
case AST_CONTROL_FLASH:
ioctl(p->fd, IXJCTL_PSTN_SET_STATE, PSTN_ON_HOOK);
usleep(320000);
ioctl(p->fd, IXJCTL_PSTN_SET_STATE, PSTN_OFF_HOOK);
ast_format_clear(&p->lastformat);
res = 0;
break;
case AST_CONTROL_HOLD:
ast_moh_start(chan, data, NULL);
break;
case AST_CONTROL_UNHOLD:
ast_moh_stop(chan);
break;
case AST_CONTROL_SRCUPDATE:
res = 0;
break;
default:
ast_log(LOG_WARNING, "Condition %d is not supported on channel %s\n", condition, chan->name);
}
return res;
}
static int phone_fixup(struct ast_channel *old, struct ast_channel *new)
{
struct phone_pvt *pvt = old->tech_pvt;
if (pvt && pvt->owner == old)
pvt->owner = new;
return 0;
}
static int phone_digit_begin(struct ast_channel *chan, char digit)
{
/* XXX Modify this callback to let Asterisk support controlling the length of DTMF */
return 0;
}
2007-01-19 18:06:03 +00:00
static int phone_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
{
struct phone_pvt *p;
int outdigit;
p = ast->tech_pvt;
ast_debug(1, "Dialed %c\n", digit);
switch(digit) {
case '0':
case '1':
case '2':
case '3':
case '4':
case '5':
case '6':
case '7':
case '8':
case '9':
outdigit = digit - '0';
break;
case '*':
outdigit = 11;
break;
case '#':
outdigit = 12;
break;
case 'f': /*flash*/
case 'F':
ioctl(p->fd, IXJCTL_PSTN_SET_STATE, PSTN_ON_HOOK);
usleep(320000);
ioctl(p->fd, IXJCTL_PSTN_SET_STATE, PSTN_OFF_HOOK);
ast_format_clear(&p->lastformat);
return 0;
default:
ast_log(LOG_WARNING, "Unknown digit '%c'\n", digit);
return -1;
}
ast_debug(1, "Dialed %d\n", outdigit);
ioctl(p->fd, PHONE_PLAY_TONE, outdigit);
ast_format_clear(&p->lastformat);
return 0;
}
static int phone_call(struct ast_channel *ast, char *dest, int timeout)
{
struct phone_pvt *p;
PHONE_CID cid;
struct timeval UtcTime = ast_tvnow();
struct ast_tm tm;
int start;
ast_localtime(&UtcTime, &tm, NULL);
memset(&cid, 0, sizeof(PHONE_CID));
snprintf(cid.month, sizeof(cid.month), "%02d",(tm.tm_mon + 1));
snprintf(cid.day, sizeof(cid.day), "%02d", tm.tm_mday);
snprintf(cid.hour, sizeof(cid.hour), "%02d", tm.tm_hour);
snprintf(cid.min, sizeof(cid.min), "%02d", tm.tm_min);
/* the standard format of ast->callerid is: "name" <number>, but not always complete */
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
if (!ast->connected.id.name.valid
|| ast_strlen_zero(ast->connected.id.name.str)) {
strcpy(cid.name, DEFAULT_CALLER_ID);
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
} else {
ast_copy_string(cid.name, ast->connected.id.name.str, sizeof(cid.name));
}
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
if (ast->connected.id.number.valid && ast->connected.id.number.str) {
ast_copy_string(cid.number, ast->connected.id.number.str, sizeof(cid.number));
}
p = ast->tech_pvt;
if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
ast_log(LOG_WARNING, "phone_call called on %s, neither down nor reserved\n", ast->name);
return -1;
}
ast_debug(1, "Ringing %s on %s (%d)\n", dest, ast->name, ast->fds[0]);
start = IXJ_PHONE_RING_START(cid);
if (start == -1)
return -1;
if (p->mode == MODE_FXS) {
char *digit = strchr(dest, '/');
if (digit)
{
digit++;
while (*digit)
2007-01-19 18:06:03 +00:00
phone_digit_end(ast, *digit++, 0);
}
}
ast_setstate(ast, AST_STATE_RINGING);
ast_queue_control(ast, AST_CONTROL_RINGING);
return 0;
}
static int phone_hangup(struct ast_channel *ast)
{
struct phone_pvt *p;
p = ast->tech_pvt;
ast_debug(1, "phone_hangup(%s)\n", ast->name);
if (!ast->tech_pvt) {
ast_log(LOG_WARNING, "Asked to hangup channel not connected\n");
return 0;
}
/* XXX Is there anything we can do to really hang up except stop recording? */
ast_setstate(ast, AST_STATE_DOWN);
if (ioctl(p->fd, PHONE_REC_STOP))
ast_log(LOG_WARNING, "Failed to stop recording\n");
if (ioctl(p->fd, PHONE_PLAY_STOP))
ast_log(LOG_WARNING, "Failed to stop playing\n");
if (ioctl(p->fd, PHONE_RING_STOP))
ast_log(LOG_WARNING, "Failed to stop ringing\n");
if (ioctl(p->fd, PHONE_CPT_STOP))
ast_log(LOG_WARNING, "Failed to stop sounds\n");
/* If it's an FXO, hang them up */
if (p->mode == MODE_FXO) {
if (ioctl(p->fd, PHONE_PSTN_SET_STATE, PSTN_ON_HOOK))
ast_debug(1, "ioctl(PHONE_PSTN_SET_STATE) failed on %s (%s)\n",ast->name, strerror(errno));
}
/* If they're off hook, give a busy signal */
if (ioctl(p->fd, PHONE_HOOKSTATE)) {
ast_debug(1, "Got hunghup, giving busy signal\n");
ioctl(p->fd, PHONE_BUSY);
p->cpt = 1;
}
ast_format_clear(&p->lastformat);
ast_format_clear(&p->lastinput);
p->ministate = 0;
p->obuflen = 0;
p->dialtone = 0;
memset(p->ext, 0, sizeof(p->ext));
((struct phone_pvt *)(ast->tech_pvt))->owner = NULL;
ast_module_unref(ast_module_info->self);
ast_verb(3, "Hungup '%s'\n", ast->name);
ast->tech_pvt = NULL;
ast_setstate(ast, AST_STATE_DOWN);
restart_monitor();
return 0;
}
static int phone_setup(struct ast_channel *ast)
{
struct phone_pvt *p;
p = ast->tech_pvt;
ioctl(p->fd, PHONE_CPT_STOP);
/* Nothing to answering really, just start recording */
if (ast->rawreadformat.id == AST_FORMAT_G729A) {
/* Prefer g729 */
ioctl(p->fd, PHONE_REC_STOP);
if (p->lastinput.id != AST_FORMAT_G729A) {
ast_format_set(&p->lastinput, AST_FORMAT_G729A, 0);
if (ioctl(p->fd, PHONE_REC_CODEC, G729)) {
ast_log(LOG_WARNING, "Failed to set codec to g729\n");
return -1;
}
}
} else if (ast->rawreadformat.id == AST_FORMAT_G723_1) {
ioctl(p->fd, PHONE_REC_STOP);
if (p->lastinput.id != AST_FORMAT_G723_1) {
ast_format_set(&p->lastinput, AST_FORMAT_G723_1, 0);
if (ioctl(p->fd, PHONE_REC_CODEC, G723_63)) {
ast_log(LOG_WARNING, "Failed to set codec to g723.1\n");
return -1;
}
}
} else if (ast->rawreadformat.id == AST_FORMAT_SLINEAR) {
ioctl(p->fd, PHONE_REC_STOP);
if (p->lastinput.id != AST_FORMAT_SLINEAR) {
ast_format_set(&p->lastinput, AST_FORMAT_SLINEAR, 0);
if (ioctl(p->fd, PHONE_REC_CODEC, LINEAR16)) {
ast_log(LOG_WARNING, "Failed to set codec to signed linear 16\n");
return -1;
}
}
} else if (ast->rawreadformat.id == AST_FORMAT_ULAW) {
ioctl(p->fd, PHONE_REC_STOP);
if (p->lastinput.id != AST_FORMAT_ULAW) {
ast_format_set(&p->lastinput, AST_FORMAT_ULAW, 0);
if (ioctl(p->fd, PHONE_REC_CODEC, ULAW)) {
ast_log(LOG_WARNING, "Failed to set codec to uLaw\n");
return -1;
}
}
} else if (p->mode == MODE_FXS) {
ioctl(p->fd, PHONE_REC_STOP);
if (ast_format_cmp(&p->lastinput, &ast->rawreadformat) == AST_FORMAT_CMP_NOT_EQUAL) {
ast_format_copy(&p->lastinput, &ast->rawreadformat);
if (ioctl(p->fd, PHONE_REC_CODEC, &ast->rawreadformat)) {
ast_log(LOG_WARNING, "Failed to set codec to %s\n",
ast_getformatname(&ast->rawreadformat));
return -1;
}
}
} else {
ast_log(LOG_WARNING, "Can't do format %s\n", ast_getformatname(&ast->rawreadformat));
return -1;
}
if (ioctl(p->fd, PHONE_REC_START)) {
ast_log(LOG_WARNING, "Failed to start recording\n");
return -1;
}
/* set the DTMF times (the default is too short) */
ioctl(p->fd, PHONE_SET_TONE_ON_TIME, 300);
ioctl(p->fd, PHONE_SET_TONE_OFF_TIME, 200);
return 0;
}
static int phone_answer(struct ast_channel *ast)
{
struct phone_pvt *p;
p = ast->tech_pvt;
/* In case it's a LineJack, take it off hook */
if (p->mode == MODE_FXO) {
if (ioctl(p->fd, PHONE_PSTN_SET_STATE, PSTN_OFF_HOOK))
ast_debug(1, "ioctl(PHONE_PSTN_SET_STATE) failed on %s (%s)\n", ast->name, strerror(errno));
else
ast_debug(1, "Took linejack off hook\n");
}
phone_setup(ast);
ast_debug(1, "phone_answer(%s)\n", ast->name);
ast->rings = 0;
ast_setstate(ast, AST_STATE_UP);
return 0;
}
#if 0
static char phone_2digit(char c)
{
if (c == 12)
return '#';
else if (c == 11)
return '*';
else if ((c < 10) && (c >= 0))
return '0' + c - 1;
else
return '?';
}
#endif
static struct ast_frame *phone_exception(struct ast_channel *ast)
{
int res;
union telephony_exception phonee;
struct phone_pvt *p = ast->tech_pvt;
char digit;
/* Some nice norms */
p->fr.datalen = 0;
p->fr.samples = 0;
p->fr.data.ptr = NULL;
p->fr.src = "Phone";
p->fr.offset = 0;
p->fr.mallocd=0;
p->fr.delivery = ast_tv(0,0);
phonee.bytes = ioctl(p->fd, PHONE_EXCEPTION);
if (phonee.bits.dtmf_ready) {
ast_debug(1, "phone_exception(): DTMF\n");
/* We've got a digit -- Just handle this nicely and easily */
digit = ioctl(p->fd, PHONE_GET_DTMF_ASCII);
p->fr.subclass.integer = digit;
p->fr.frametype = AST_FRAME_DTMF;
return &p->fr;
}
if (phonee.bits.hookstate) {
ast_debug(1, "Hookstate changed\n");
res = ioctl(p->fd, PHONE_HOOKSTATE);
/* See if we've gone on hook, if so, notify by returning NULL */
ast_debug(1, "New hookstate: %d\n", res);
if (!res && (p->mode != MODE_FXO))
return NULL;
else {
if (ast->_state == AST_STATE_RINGING) {
/* They've picked up the phone */
p->fr.frametype = AST_FRAME_CONTROL;
p->fr.subclass.integer = AST_CONTROL_ANSWER;
phone_setup(ast);
ast_setstate(ast, AST_STATE_UP);
return &p->fr;
} else
ast_log(LOG_WARNING, "Got off hook in weird state %d\n", ast->_state);
}
}
#if 1
if (phonee.bits.pstn_ring)
ast_verbose("Unit is ringing\n");
if (phonee.bits.caller_id) {
ast_verbose("We have caller ID\n");
}
if (phonee.bits.pstn_wink)
ast_verbose("Detected Wink\n");
#endif
/* Strange -- nothing there.. */
p->fr.frametype = AST_FRAME_NULL;
p->fr.subclass.integer = 0;
return &p->fr;
}
static struct ast_frame *phone_read(struct ast_channel *ast)
{
int res;
struct phone_pvt *p = ast->tech_pvt;
/* Some nice norms */
p->fr.datalen = 0;
p->fr.samples = 0;
p->fr.data.ptr = NULL;
p->fr.src = "Phone";
p->fr.offset = 0;
p->fr.mallocd=0;
p->fr.delivery = ast_tv(0,0);
/* Try to read some data... */
CHECK_BLOCKING(ast);
res = read(p->fd, p->buf, PHONE_MAX_BUF);
ast_clear_flag(ast, AST_FLAG_BLOCKING);
if (res < 0) {
#if 0
if (errno == EAGAIN) {
ast_log(LOG_WARNING, "Null frame received\n");
p->fr.frametype = AST_FRAME_NULL;
p->fr.subclass = 0;
return &p->fr;
}
#endif
ast_log(LOG_WARNING, "Error reading: %s\n", strerror(errno));
return NULL;
}
p->fr.data.ptr = p->buf;
if (p->mode != MODE_FXS)
switch(p->buf[0] & 0x3) {
case '0':
case '1':
/* Normal */
break;
case '2':
case '3':
/* VAD/CNG, only send two words */
res = 4;
break;
}
p->fr.samples = 240;
p->fr.datalen = res;
p->fr.frametype = AST_FORMAT_GET_TYPE(p->lastinput.id) == AST_FORMAT_TYPE_AUDIO ?
AST_FRAME_VOICE : AST_FORMAT_GET_TYPE(p->lastinput.id) == AST_FORMAT_TYPE_IMAGE ?
AST_FRAME_IMAGE : AST_FRAME_VIDEO;
ast_format_copy(&p->fr.subclass.format, &p->lastinput);
p->fr.offset = AST_FRIENDLY_OFFSET;
/* Byteswap from little-endian to native-endian */
if (p->fr.subclass.format.id == AST_FORMAT_SLINEAR)
ast_frame_byteswap_le(&p->fr);
return &p->fr;
}
static int phone_write_buf(struct phone_pvt *p, const char *buf, int len, int frlen, int swap)
{
int res;
/* Store as much of the buffer as we can, then write fixed frames */
int space = sizeof(p->obuf) - p->obuflen;
/* Make sure we have enough buffer space to store the frame */
if (space < len)
len = space;
if (swap)
ast_swapcopy_samples(p->obuf+p->obuflen, buf, len/2);
else
memcpy(p->obuf + p->obuflen, buf, len);
p->obuflen += len;
while(p->obuflen > frlen) {
res = write(p->fd, p->obuf, frlen);
if (res != frlen) {
if (res < 1) {
/*
* Card is in non-blocking mode now and it works well now, but there are
* lot of messages like this. So, this message is temporarily disabled.
*/
return 0;
} else {
ast_log(LOG_WARNING, "Only wrote %d of %d bytes\n", res, frlen);
}
}
p->obuflen -= frlen;
/* Move memory if necessary */
if (p->obuflen)
memmove(p->obuf, p->obuf + frlen, p->obuflen);
}
return len;
}
static int phone_send_text(struct ast_channel *ast, const char *text)
{
int length = strlen(text);
return phone_write_buf(ast->tech_pvt, text, length, length, 0) ==
length ? 0 : -1;
}
static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct phone_pvt *p = ast->tech_pvt;
int res;
int maxfr=0;
char *pos;
int sofar;
int expected;
int codecset = 0;
char tmpbuf[4];
/* Write a frame of (presumably voice) data */
if (frame->frametype != AST_FRAME_VOICE && p->mode != MODE_FXS) {
if (frame->frametype != AST_FRAME_IMAGE)
ast_log(LOG_WARNING, "Don't know what to do with frame type '%d'\n", frame->frametype);
return 0;
}
if (!(frame->subclass.format.id == AST_FORMAT_G723_1 ||
frame->subclass.format.id == AST_FORMAT_SLINEAR ||
frame->subclass.format.id == AST_FORMAT_ULAW ||
frame->subclass.format.id == AST_FORMAT_G729A) &&
p->mode != MODE_FXS) {
ast_log(LOG_WARNING, "Cannot handle frames in %s format\n", ast_getformatname(&frame->subclass.format));
return -1;
}
#if 0
/* If we're not in up mode, go into up mode now */
if (ast->_state != AST_STATE_UP) {
ast_setstate(ast, AST_STATE_UP);
phone_setup(ast);
}
#else
if (ast->_state != AST_STATE_UP) {
/* Don't try tos end audio on-hook */
return 0;
}
#endif
if (frame->subclass.format.id == AST_FORMAT_G729A) {
if (p->lastformat.id != AST_FORMAT_G729A) {
ioctl(p->fd, PHONE_PLAY_STOP);
ioctl(p->fd, PHONE_REC_STOP);
if (ioctl(p->fd, PHONE_PLAY_CODEC, G729)) {
ast_log(LOG_WARNING, "Unable to set G729 mode\n");
return -1;
}
if (ioctl(p->fd, PHONE_REC_CODEC, G729)) {
ast_log(LOG_WARNING, "Unable to set G729 mode\n");
return -1;
}
ast_format_set(&p->lastformat, AST_FORMAT_G729A, 0);
ast_format_set(&p->lastinput, AST_FORMAT_G729A, 0);
/* Reset output buffer */
p->obuflen = 0;
codecset = 1;
}
if (frame->datalen > 80) {
ast_log(LOG_WARNING, "Frame size too large for G.729 (%d bytes)\n", frame->datalen);
return -1;
}
maxfr = 80;
} else if (frame->subclass.format.id == AST_FORMAT_G723_1) {
if (p->lastformat.id != AST_FORMAT_G723_1) {
ioctl(p->fd, PHONE_PLAY_STOP);
ioctl(p->fd, PHONE_REC_STOP);
if (ioctl(p->fd, PHONE_PLAY_CODEC, G723_63)) {
ast_log(LOG_WARNING, "Unable to set G723.1 mode\n");
return -1;
}
if (ioctl(p->fd, PHONE_REC_CODEC, G723_63)) {
ast_log(LOG_WARNING, "Unable to set G723.1 mode\n");
return -1;
}
ast_format_set(&p->lastformat, AST_FORMAT_G723_1, 0);
ast_format_set(&p->lastinput, AST_FORMAT_G723_1, 0);
/* Reset output buffer */
p->obuflen = 0;
codecset = 1;
}
if (frame->datalen > 24) {
ast_log(LOG_WARNING, "Frame size too large for G.723.1 (%d bytes)\n", frame->datalen);
return -1;
}
maxfr = 24;
} else if (frame->subclass.format.id == AST_FORMAT_SLINEAR) {
if (p->lastformat.id != AST_FORMAT_SLINEAR) {
ioctl(p->fd, PHONE_PLAY_STOP);
ioctl(p->fd, PHONE_REC_STOP);
if (ioctl(p->fd, PHONE_PLAY_CODEC, LINEAR16)) {
ast_log(LOG_WARNING, "Unable to set 16-bit linear mode\n");
return -1;
}
if (ioctl(p->fd, PHONE_REC_CODEC, LINEAR16)) {
ast_log(LOG_WARNING, "Unable to set 16-bit linear mode\n");
return -1;
}
ast_format_set(&p->lastformat, AST_FORMAT_SLINEAR, 0);
ast_format_set(&p->lastinput, AST_FORMAT_SLINEAR, 0);
codecset = 1;
/* Reset output buffer */
p->obuflen = 0;
}
maxfr = 480;
} else if (frame->subclass.format.id == AST_FORMAT_ULAW) {
if (p->lastformat.id != AST_FORMAT_ULAW) {
ioctl(p->fd, PHONE_PLAY_STOP);
ioctl(p->fd, PHONE_REC_STOP);
if (ioctl(p->fd, PHONE_PLAY_CODEC, ULAW)) {
ast_log(LOG_WARNING, "Unable to set uLaw mode\n");
return -1;
}
if (ioctl(p->fd, PHONE_REC_CODEC, ULAW)) {
ast_log(LOG_WARNING, "Unable to set uLaw mode\n");
return -1;
}
ast_format_set(&p->lastformat, AST_FORMAT_ULAW, 0);
ast_format_set(&p->lastinput, AST_FORMAT_ULAW, 0);
codecset = 1;
/* Reset output buffer */
p->obuflen = 0;
}
maxfr = 240;
} else {
if (ast_format_cmp(&p->lastformat, &frame->subclass.format) != AST_FORMAT_CMP_EQUAL) {
ioctl(p->fd, PHONE_PLAY_STOP);
ioctl(p->fd, PHONE_REC_STOP);
if (ioctl(p->fd, PHONE_PLAY_CODEC, (int) frame->subclass.format.id)) {
ast_log(LOG_WARNING, "Unable to set %s mode\n",
ast_getformatname(&frame->subclass.format));
return -1;
}
if (ioctl(p->fd, PHONE_REC_CODEC, (int) frame->subclass.format.id)) {
ast_log(LOG_WARNING, "Unable to set %s mode\n",
ast_getformatname(&frame->subclass.format));
return -1;
}
ast_format_copy(&p->lastformat, &frame->subclass.format);
ast_format_copy(&p->lastinput, &frame->subclass.format);
codecset = 1;
/* Reset output buffer */
p->obuflen = 0;
}
maxfr = 480;
}
if (codecset) {
ioctl(p->fd, PHONE_REC_DEPTH, 3);
ioctl(p->fd, PHONE_PLAY_DEPTH, 3);
if (ioctl(p->fd, PHONE_PLAY_START)) {
ast_log(LOG_WARNING, "Failed to start playback\n");
return -1;
}
if (ioctl(p->fd, PHONE_REC_START)) {
ast_log(LOG_WARNING, "Failed to start recording\n");
return -1;
}
}
/* If we get here, we have a frame of Appropriate data */
sofar = 0;
pos = frame->data.ptr;
while(sofar < frame->datalen) {
/* Write in no more than maxfr sized frames */
expected = frame->datalen - sofar;
if (maxfr < expected)
expected = maxfr;
/* XXX Internet Phone Jack does not handle the 4-byte VAD frame properly! XXX
we have to pad it to 24 bytes still. */
if (frame->datalen == 4) {
if (p->silencesupression) {
memcpy(tmpbuf, frame->data.ptr, 4);
expected = 24;
res = phone_write_buf(p, tmpbuf, expected, maxfr, 0);
}
res = 4;
expected=4;
} else {
int swap = 0;
#if __BYTE_ORDER == __BIG_ENDIAN
if (frame->subclass.format.id == AST_FORMAT_SLINEAR)
swap = 1; /* Swap big-endian samples to little-endian as we copy */
#endif
res = phone_write_buf(p, pos, expected, maxfr, swap);
}
if (res != expected) {
if ((errno != EAGAIN) && (errno != EINTR)) {
if (res < 0)
ast_log(LOG_WARNING, "Write returned error (%s)\n", strerror(errno));
/*
* Card is in non-blocking mode now and it works well now, but there are
* lot of messages like this. So, this message is temporarily disabled.
*/
#if 0
else
ast_log(LOG_WARNING, "Only wrote %d of %d bytes\n", res, frame->datalen);
#endif
return -1;
} else /* Pretend it worked */
res = expected;
}
sofar += res;
pos += res;
}
return 0;
}
static struct ast_channel *phone_new(struct phone_pvt *i, int state, char *cntx, const char *linkedid)
{
struct ast_channel *tmp;
struct phone_codec_data queried_codec;
struct ast_format tmpfmt;
tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, "", i->ext, i->context, linkedid, 0, "Phone/%s", i->dev + 5);
if (tmp) {
tmp->tech = cur_tech;
ast_channel_set_fd(tmp, 0, i->fd);
/* XXX Switching formats silently causes kernel panics XXX */
if (i->mode == MODE_FXS &&
ioctl(i->fd, PHONE_QUERY_CODEC, &queried_codec) == 0) {
if (queried_codec.type == LINEAR16) {
ast_format_cap_add(tmp->nativeformats, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0));
ast_format_copy(&tmp->rawreadformat, &tmpfmt);
ast_format_copy(&tmp->rawwriteformat, &tmpfmt);
} else {
ast_format_cap_remove(prefcap, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0));
}
} else {
ast_format_cap_copy(tmp->nativeformats, prefcap);
ast_best_codec(tmp->nativeformats, &tmpfmt);
ast_format_copy(&tmp->rawreadformat, &tmpfmt);
ast_format_copy(&tmp->rawwriteformat, &tmpfmt);
}
/* no need to call ast_setstate: the channel_alloc already did its job */
if (state == AST_STATE_RING)
tmp->rings = 1;
tmp->tech_pvt = i;
ast_copy_string(tmp->context, cntx, sizeof(tmp->context));
if (!ast_strlen_zero(i->ext))
ast_copy_string(tmp->exten, i->ext, sizeof(tmp->exten));
else
strcpy(tmp->exten, "s");
if (!ast_strlen_zero(i->language))
ast_string_field_set(tmp, language, i->language);
/* Don't use ast_set_callerid() here because it will
* generate a NewCallerID event before the NewChannel event */
if (!ast_strlen_zero(i->cid_num)) {
tmp->caller.ani.number.valid = 1;
tmp->caller.ani.number.str = ast_strdup(i->cid_num);
}
i->owner = tmp;
ast_module_ref(ast_module_info->self);
if (state != AST_STATE_DOWN) {
if (state == AST_STATE_RING) {
ioctl(tmp->fds[0], PHONE_RINGBACK);
i->cpt = 1;
}
if (ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
ast_hangup(tmp);
}
}
} else
ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
return tmp;
}
static void phone_mini_packet(struct phone_pvt *i)
{
int res;
char buf[1024];
/* Ignore stuff we read... */
res = read(i->fd, buf, sizeof(buf));
if (res < 1) {
ast_log(LOG_WARNING, "Read returned %d: %s\n", res, strerror(errno));
return;
}
}
static void phone_check_exception(struct phone_pvt *i)
{
int offhook=0;
char digit[2] = {0 , 0};
union telephony_exception phonee;
/* XXX Do something XXX */
#if 0
ast_debug(1, "Exception!\n");
#endif
phonee.bytes = ioctl(i->fd, PHONE_EXCEPTION);
if (phonee.bits.dtmf_ready) {
digit[0] = ioctl(i->fd, PHONE_GET_DTMF_ASCII);
if (i->mode == MODE_DIALTONE || i->mode == MODE_FXS || i->mode == MODE_SIGMA) {
ioctl(i->fd, PHONE_PLAY_STOP);
ioctl(i->fd, PHONE_REC_STOP);
ioctl(i->fd, PHONE_CPT_STOP);
i->dialtone = 0;
if (strlen(i->ext) < AST_MAX_EXTENSION - 1)
strncat(i->ext, digit, sizeof(i->ext) - strlen(i->ext) - 1);
if ((i->mode != MODE_FXS ||
!(phonee.bytes = ioctl(i->fd, PHONE_EXCEPTION)) ||
!phonee.bits.dtmf_ready) &&
ast_exists_extension(NULL, i->context, i->ext, 1, i->cid_num)) {
/* It's a valid extension in its context, get moving! */
phone_new(i, AST_STATE_RING, i->context, NULL);
/* No need to restart monitor, we are the monitor */
} else if (!ast_canmatch_extension(NULL, i->context, i->ext, 1, i->cid_num)) {
/* There is nothing in the specified extension that can match anymore.
Try the default */
if (ast_exists_extension(NULL, "default", i->ext, 1, i->cid_num)) {
/* Check the default, too... */
phone_new(i, AST_STATE_RING, "default", NULL);
/* XXX This should probably be justified better XXX */
} else if (!ast_canmatch_extension(NULL, "default", i->ext, 1, i->cid_num)) {
/* It's not a valid extension, give a busy signal */
ast_debug(1, "%s can't match anything in %s or default\n", i->ext, i->context);
ioctl(i->fd, PHONE_BUSY);
i->cpt = 1;
}
}
#if 0
ast_verbose("Extension is %s\n", i->ext);
#endif
}
}
if (phonee.bits.hookstate) {
offhook = ioctl(i->fd, PHONE_HOOKSTATE);
if (offhook) {
if (i->mode == MODE_IMMEDIATE) {
phone_new(i, AST_STATE_RING, i->context, NULL);
} else if (i->mode == MODE_DIALTONE) {
ast_module_ref(ast_module_info->self);
/* Reset the extension */
i->ext[0] = '\0';
/* Play the dialtone */
i->dialtone++;
ioctl(i->fd, PHONE_PLAY_STOP);
ioctl(i->fd, PHONE_PLAY_CODEC, ULAW);
ioctl(i->fd, PHONE_PLAY_START);
ast_format_clear(&i->lastformat);
} else if (i->mode == MODE_SIGMA) {
ast_module_ref(ast_module_info->self);
/* Reset the extension */
i->ext[0] = '\0';
/* Play the dialtone */
i->dialtone++;
ioctl(i->fd, PHONE_DIALTONE);
}
} else {
if (i->dialtone)
ast_module_unref(ast_module_info->self);
memset(i->ext, 0, sizeof(i->ext));
if (i->cpt)
{
ioctl(i->fd, PHONE_CPT_STOP);
i->cpt = 0;
}
ioctl(i->fd, PHONE_PLAY_STOP);
ioctl(i->fd, PHONE_REC_STOP);
i->dialtone = 0;
ast_format_clear(&i->lastformat);
}
}
if (phonee.bits.pstn_ring) {
ast_verbose("Unit is ringing\n");
phone_new(i, AST_STATE_RING, i->context, NULL);
}
if (phonee.bits.caller_id)
ast_verbose("We have caller ID\n");
}
static void *do_monitor(void *data)
{
struct pollfd *fds = NULL;
int nfds = 0, inuse_fds = 0, res;
struct phone_pvt *i;
int tonepos = 0;
/* The tone we're playing this round */
struct timeval tv = { 0, 0 };
int dotone;
/* This thread monitors all the frame relay interfaces which are not yet in use
(and thus do not have a separate thread) indefinitely */
while (monitor) {
/* Don't let anybody kill us right away. Nobody should lock the interface list
and wait for the monitor list, but the other way around is okay. */
/* Lock the interface list */
if (ast_mutex_lock(&iflock)) {
ast_log(LOG_ERROR, "Unable to grab interface lock\n");
return NULL;
}
/* Build the stuff we're going to select on, that is the socket of every
phone_pvt that does not have an associated owner channel */
i = iflist;
dotone = 0;
inuse_fds = 0;
for (i = iflist; i; i = i->next) {
if (!i->owner) {
/* This needs to be watched, as it lacks an owner */
if (inuse_fds == nfds) {
void *tmp = ast_realloc(fds, (nfds + 1) * sizeof(*fds));
if (!tmp) {
/* Avoid leaking */
continue;
}
fds = tmp;
nfds++;
}
fds[inuse_fds].fd = i->fd;
fds[inuse_fds].events = POLLIN | POLLERR;
fds[inuse_fds].revents = 0;
inuse_fds++;
if (i->dialtone && i->mode != MODE_SIGMA) {
/* Remember we're going to have to come back and play
more dialtones */
if (ast_tvzero(tv)) {
/* If we're due for a dialtone, play one */
if (write(i->fd, DialTone + tonepos, 240) != 240) {
ast_log(LOG_WARNING, "Dial tone write error\n");
}
}
dotone++;
}
}
}
/* Okay, now that we know what to do, release the interface lock */
ast_mutex_unlock(&iflock);
/* Wait indefinitely for something to happen */
if (dotone && i && i->mode != MODE_SIGMA) {
/* If we're ready to recycle the time, set it to 30 ms */
tonepos += 240;
if (tonepos >= sizeof(DialTone)) {
tonepos = 0;
}
if (ast_tvzero(tv)) {
tv = ast_tv(0, 30000);
}
res = ast_poll2(fds, inuse_fds, &tv);
} else {
res = ast_poll(fds, inuse_fds, -1);
tv = ast_tv(0, 0);
tonepos = 0;
}
/* Okay, select has finished. Let's see what happened. */
if (res < 0) {
ast_debug(1, "poll returned %d: %s\n", res, strerror(errno));
continue;
}
/* If there are no fd's changed, just continue, it's probably time
to play some more dialtones */
if (!res) {
continue;
}
/* Alright, lock the interface list again, and let's look and see what has
happened */
if (ast_mutex_lock(&iflock)) {
ast_log(LOG_WARNING, "Unable to lock the interface list\n");
continue;
}
for (i = iflist; i; i = i->next) {
int j;
/* Find the record */
for (j = 0; j < inuse_fds; j++) {
if (fds[j].fd == i->fd) {
break;
}
}
/* Not found? */
if (j == inuse_fds) {
continue;
}
if (fds[j].revents & POLLIN) {
if (i->owner) {
continue;
}
phone_mini_packet(i);
}
if (fds[j].revents & POLLERR) {
if (i->owner) {
continue;
}
phone_check_exception(i);
}
}
ast_mutex_unlock(&iflock);
}
return NULL;
}
static int restart_monitor()
{
/* If we're supposed to be stopped -- stay stopped */
if (monitor_thread == AST_PTHREADT_STOP)
return 0;
if (ast_mutex_lock(&monlock)) {
ast_log(LOG_WARNING, "Unable to lock monitor\n");
return -1;
}
if (monitor_thread == pthread_self()) {
ast_mutex_unlock(&monlock);
ast_log(LOG_WARNING, "Cannot kill myself\n");
return -1;
}
if (monitor_thread != AST_PTHREADT_NULL) {
if (ast_mutex_lock(&iflock)) {
ast_mutex_unlock(&monlock);
ast_log(LOG_WARNING, "Unable to lock the interface list\n");
return -1;
}
monitor = 0;
while (pthread_kill(monitor_thread, SIGURG) == 0)
sched_yield();
pthread_join(monitor_thread, NULL);
ast_mutex_unlock(&iflock);
}
monitor = 1;
/* Start a new monitor */
if (ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL) < 0) {
ast_mutex_unlock(&monlock);
ast_log(LOG_ERROR, "Unable to start monitor thread.\n");
return -1;
}
ast_mutex_unlock(&monlock);
return 0;
}
static struct phone_pvt *mkif(const char *iface, int mode, int txgain, int rxgain)
{
/* Make a phone_pvt structure for this interface */
struct phone_pvt *tmp;
int flags;
tmp = ast_calloc(1, sizeof(*tmp));
if (tmp) {
tmp->fd = open(iface, O_RDWR);
if (tmp->fd < 0) {
ast_log(LOG_WARNING, "Unable to open '%s'\n", iface);
ast_free(tmp);
return NULL;
}
if (mode == MODE_FXO) {
if (ioctl(tmp->fd, IXJCTL_PORT, PORT_PSTN)) {
ast_debug(1, "Unable to set port to PSTN\n");
}
} else {
if (ioctl(tmp->fd, IXJCTL_PORT, PORT_POTS))
if (mode != MODE_FXS)
ast_debug(1, "Unable to set port to POTS\n");
}
ioctl(tmp->fd, PHONE_PLAY_STOP);
ioctl(tmp->fd, PHONE_REC_STOP);
ioctl(tmp->fd, PHONE_RING_STOP);
ioctl(tmp->fd, PHONE_CPT_STOP);
if (ioctl(tmp->fd, PHONE_PSTN_SET_STATE, PSTN_ON_HOOK))
ast_debug(1, "ioctl(PHONE_PSTN_SET_STATE) failed on %s (%s)\n",iface, strerror(errno));
if (echocancel != AEC_OFF)
ioctl(tmp->fd, IXJCTL_AEC_START, echocancel);
if (silencesupression)
tmp->silencesupression = 1;
#ifdef PHONE_VAD
ioctl(tmp->fd, PHONE_VAD, tmp->silencesupression);
#endif
tmp->mode = mode;
flags = fcntl(tmp->fd, F_GETFL);
fcntl(tmp->fd, F_SETFL, flags | O_NONBLOCK);
tmp->owner = NULL;
ast_format_clear(&tmp->lastformat);
ast_format_clear(&tmp->lastinput);
tmp->ministate = 0;
memset(tmp->ext, 0, sizeof(tmp->ext));
ast_copy_string(tmp->language, language, sizeof(tmp->language));
ast_copy_string(tmp->dev, iface, sizeof(tmp->dev));
ast_copy_string(tmp->context, context, sizeof(tmp->context));
tmp->next = NULL;
tmp->obuflen = 0;
tmp->dialtone = 0;
tmp->cpt = 0;
ast_copy_string(tmp->cid_num, cid_num, sizeof(tmp->cid_num));
ast_copy_string(tmp->cid_name, cid_name, sizeof(tmp->cid_name));
tmp->txgain = txgain;
ioctl(tmp->fd, PHONE_PLAY_VOLUME, tmp->txgain);
tmp->rxgain = rxgain;
ioctl(tmp->fd, PHONE_REC_VOLUME, tmp->rxgain);
}
return tmp;
}
static struct ast_channel *phone_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause)
{
struct phone_pvt *p;
struct ast_channel *tmp = NULL;
char *name = data;
/* Search for an unowned channel */
if (ast_mutex_lock(&iflock)) {
ast_log(LOG_ERROR, "Unable to lock interface list???\n");
return NULL;
}
p = iflist;
while(p) {
if (p->mode == MODE_FXS || (ast_format_cap_has_joint(cap, phone_tech.capabilities))) {
size_t length = strlen(p->dev + 5);
if (strncmp(name, p->dev + 5, length) == 0 &&
!isalnum(name[length])) {
if (!p->owner) {
tmp = phone_new(p, AST_STATE_DOWN, p->context, requestor ? requestor->linkedid : NULL);
break;
} else
*cause = AST_CAUSE_BUSY;
}
}
p = p->next;
}
ast_mutex_unlock(&iflock);
restart_monitor();
if (tmp == NULL) {
if (!(ast_format_cap_has_joint(cap, phone_tech.capabilities))) {
char buf[256];
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%s'\n", ast_getformatname_multiple(buf, sizeof(buf), cap));
return NULL;
}
}
return tmp;
}
/* parse gain value from config file */
static int parse_gain_value(const char *gain_type, const char *value)
{
float gain;
/* try to scan number */
if (sscanf(value, "%30f", &gain) != 1)
{
ast_log(LOG_ERROR, "Invalid %s value '%s' in '%s' config\n",
value, gain_type, config);
return DEFAULT_GAIN;
}
/* multiplicate gain by 1.0 gain value */
gain = gain * (float)DEFAULT_GAIN;
/* percentage? */
if (value[strlen(value) - 1] == '%')
return (int)(gain / (float)100);
return (int)gain;
}
static int __unload_module(void)
{
struct phone_pvt *p, *pl;
/* First, take us out of the channel loop */
if (cur_tech)
ast_channel_unregister(cur_tech);
if (!ast_mutex_lock(&iflock)) {
/* Hangup all interfaces if they have an owner */
p = iflist;
while(p) {
if (p->owner)
ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD);
p = p->next;
}
iflist = NULL;
ast_mutex_unlock(&iflock);
} else {
ast_log(LOG_WARNING, "Unable to lock the monitor\n");
return -1;
}
if (!ast_mutex_lock(&monlock)) {
if (monitor_thread > AST_PTHREADT_NULL) {
monitor = 0;
while (pthread_kill(monitor_thread, SIGURG) == 0)
sched_yield();
pthread_join(monitor_thread, NULL);
}
monitor_thread = AST_PTHREADT_STOP;
ast_mutex_unlock(&monlock);
} else {
ast_log(LOG_WARNING, "Unable to lock the monitor\n");
return -1;
}
if (!ast_mutex_lock(&iflock)) {
/* Destroy all the interfaces and free their memory */
p = iflist;
while(p) {
/* Close the socket, assuming it's real */
if (p->fd > -1)
close(p->fd);
pl = p;
p = p->next;
/* Free associated memory */
ast_free(pl);
}
iflist = NULL;
ast_mutex_unlock(&iflock);
} else {
ast_log(LOG_WARNING, "Unable to lock the monitor\n");
return -1;
}
phone_tech.capabilities = ast_format_cap_destroy(phone_tech.capabilities);
phone_tech_fxs.capabilities = ast_format_cap_destroy(phone_tech_fxs.capabilities);
prefcap = ast_format_cap_destroy(prefcap);
return 0;
}
static int unload_module(void)
{
return __unload_module();
}
static int load_module(void)
{
struct ast_config *cfg;
struct ast_variable *v;
struct phone_pvt *tmp;
int mode = MODE_IMMEDIATE;
int txgain = DEFAULT_GAIN, rxgain = DEFAULT_GAIN; /* default gain 1.0 */
struct ast_flags config_flags = { 0 };
struct ast_format tmpfmt;
if (!(phone_tech.capabilities = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_DECLINE;
}
ast_format_cap_add(phone_tech.capabilities, ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0));
ast_format_cap_add(phone_tech.capabilities, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0));
ast_format_cap_add(phone_tech.capabilities, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0));
ast_format_cap_add(phone_tech.capabilities, ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0));
if (!(prefcap = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_DECLINE;
}
ast_format_cap_copy(prefcap, phone_tech.capabilities);
if (!(phone_tech_fxs.capabilities = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_DECLINE;
}
if ((cfg = ast_config_load(config, config_flags)) == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file %s is in an invalid format. Aborting.\n", config);
return AST_MODULE_LOAD_DECLINE;
}
/* We *must* have a config file otherwise stop immediately */
if (!cfg) {
ast_log(LOG_ERROR, "Unable to load config %s\n", config);
return AST_MODULE_LOAD_DECLINE;
}
if (ast_mutex_lock(&iflock)) {
/* It's a little silly to lock it, but we mind as well just to be sure */
ast_log(LOG_ERROR, "Unable to lock interface list???\n");
return AST_MODULE_LOAD_FAILURE;
}
v = ast_variable_browse(cfg, "interfaces");
while(v) {
/* Create the interface list */
if (!strcasecmp(v->name, "device")) {
tmp = mkif(v->value, mode, txgain, rxgain);
if (tmp) {
tmp->next = iflist;
iflist = tmp;
} else {
ast_log(LOG_ERROR, "Unable to register channel '%s'\n", v->value);
ast_config_destroy(cfg);
ast_mutex_unlock(&iflock);
__unload_module();
return AST_MODULE_LOAD_FAILURE;
}
} else if (!strcasecmp(v->name, "silencesupression")) {
silencesupression = ast_true(v->value);
} else if (!strcasecmp(v->name, "language")) {
ast_copy_string(language, v->value, sizeof(language));
} else if (!strcasecmp(v->name, "callerid")) {
ast_callerid_split(v->value, cid_name, sizeof(cid_name), cid_num, sizeof(cid_num));
} else if (!strcasecmp(v->name, "mode")) {
if (!strncasecmp(v->value, "di", 2))
mode = MODE_DIALTONE;
else if (!strncasecmp(v->value, "sig", 3))
mode = MODE_SIGMA;
else if (!strncasecmp(v->value, "im", 2))
mode = MODE_IMMEDIATE;
else if (!strncasecmp(v->value, "fxs", 3)) {
mode = MODE_FXS;
ast_format_cap_remove_bytype(prefcap, AST_FORMAT_TYPE_AUDIO); /* All non-voice */
}
else if (!strncasecmp(v->value, "fx", 2))
mode = MODE_FXO;
else
ast_log(LOG_WARNING, "Unknown mode: %s\n", v->value);
} else if (!strcasecmp(v->name, "context")) {
ast_copy_string(context, v->value, sizeof(context));
} else if (!strcasecmp(v->name, "format")) {
struct ast_format tmpfmt;
if (!strcasecmp(v->value, "g729")) {
ast_format_cap_set(prefcap, ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0));
} else if (!strcasecmp(v->value, "g723.1")) {
ast_format_cap_set(prefcap, ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0));
} else if (!strcasecmp(v->value, "slinear")) {
ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0);
if (mode == MODE_FXS) {
ast_format_cap_add(prefcap, &tmpfmt);
} else {
ast_format_cap_set(prefcap, &tmpfmt);
}
} else if (!strcasecmp(v->value, "ulaw")) {
ast_format_cap_set(prefcap, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0));
} else
ast_log(LOG_WARNING, "Unknown format '%s'\n", v->value);
} else if (!strcasecmp(v->name, "echocancel")) {
if (!strcasecmp(v->value, "off")) {
echocancel = AEC_OFF;
} else if (!strcasecmp(v->value, "low")) {
echocancel = AEC_LOW;
} else if (!strcasecmp(v->value, "medium")) {
echocancel = AEC_MED;
} else if (!strcasecmp(v->value, "high")) {
echocancel = AEC_HIGH;
} else
ast_log(LOG_WARNING, "Unknown echo cancellation '%s'\n", v->value);
} else if (!strcasecmp(v->name, "txgain")) {
txgain = parse_gain_value(v->name, v->value);
} else if (!strcasecmp(v->name, "rxgain")) {
rxgain = parse_gain_value(v->name, v->value);
}
v = v->next;
}
ast_mutex_unlock(&iflock);
if (mode == MODE_FXS) {
ast_format_cap_copy(phone_tech_fxs.capabilities, prefcap);
cur_tech = &phone_tech_fxs;
} else
cur_tech = (struct ast_channel_tech *) &phone_tech;
/* Make sure we can register our Adtranphone channel type */
if (ast_channel_register(cur_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class 'Phone'\n");
ast_config_destroy(cfg);
__unload_module();
return AST_MODULE_LOAD_FAILURE;
}
ast_config_destroy(cfg);
/* And start the monitor for the first time */
restart_monitor();
return AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Linux Telephony API Support");