diff options
Diffstat (limited to '1.2-netsec/channels/chan_oss_old.c')
| -rw-r--r-- | 1.2-netsec/channels/chan_oss_old.c | 1132 |
1 files changed, 0 insertions, 1132 deletions
diff --git a/1.2-netsec/channels/chan_oss_old.c b/1.2-netsec/channels/chan_oss_old.c deleted file mode 100644 index c1613d3b6..000000000 --- a/1.2-netsec/channels/chan_oss_old.c +++ /dev/null @@ -1,1132 +0,0 @@ -/* - * 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. - */ - -/* - * Use /dev/dsp as a channel, and the console to command it :). - * - * The full-duplex "simulation" is pretty weak. This is generally a - * VERY BADLY WRITTEN DRIVER so please don't use it as a model for - * writing a driver. - * - * \ingroup channel_drivers - */ - -#include <unistd.h> -#include <fcntl.h> -#include <errno.h> -#include <sys/ioctl.h> -#include <sys/time.h> -#include <string.h> -#include <stdlib.h> -#include <stdio.h> - -#ifdef __linux -#include <linux/soundcard.h> -#elif defined(__FreeBSD__) -#include <sys/soundcard.h> -#else -#include <soundcard.h> -#endif - -#include "asterisk.h" - -ASTERISK_FILE_VERSION(__FILE__, "$Revision$") - -#include "asterisk/lock.h" -#include "asterisk/frame.h" -#include "asterisk/logger.h" -#include "asterisk/channel.h" -#include "asterisk/module.h" -#include "asterisk/options.h" -#include "asterisk/pbx.h" -#include "asterisk/config.h" -#include "asterisk/cli.h" -#include "asterisk/utils.h" -#include "asterisk/causes.h" -#include "asterisk/endian.h" - -#include "busy.h" -#include "ringtone.h" -#include "ring10.h" -#include "answer.h" - -/* Which device to use */ -#if defined( __OpenBSD__ ) || defined( __NetBSD__ ) -#define DEV_DSP "/dev/audio" -#else -#define DEV_DSP "/dev/dsp" -#endif - -/* Lets use 160 sample frames, just like GSM. */ -#define FRAME_SIZE 160 - -/* When you set the frame size, you have to come up with - the right buffer format as well. */ -/* 5 64-byte frames = one frame */ -#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006); - -/* Don't switch between read/write modes faster than every 300 ms */ -#define MIN_SWITCH_TIME 600 - -static struct timeval lasttime; - -static int usecnt; -static int silencesuppression = 0; -static int silencethreshold = 1000; -static int playbackonly = 0; - - -AST_MUTEX_DEFINE_STATIC(usecnt_lock); - -static const char type[] = "Console"; -static const char desc[] = "OSS Console Channel Driver"; -static const char tdesc[] = "OSS Console Channel Driver"; -static const char config[] = "oss.conf"; - -static char context[AST_MAX_CONTEXT] = "default"; -static char language[MAX_LANGUAGE] = ""; -static char exten[AST_MAX_EXTENSION] = "s"; - -static int hookstate=0; - -static short silence[FRAME_SIZE] = {0, }; - -struct sound { - int ind; - short *data; - int datalen; - int samplen; - int silencelen; - int repeat; -}; - -static struct sound sounds[] = { - { AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 }, - { AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 }, - { AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 }, - { AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 }, - { AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 }, -}; - -/* Sound command pipe */ -static int sndcmd[2]; - -static struct chan_oss_pvt { - /* We only have one OSS structure -- near sighted perhaps, but it - keeps this driver as simple as possible -- as it should be. */ - struct ast_channel *owner; - char exten[AST_MAX_EXTENSION]; - char context[AST_MAX_CONTEXT]; -} oss; - -static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause); -static int oss_digit(struct ast_channel *c, char digit); -static int oss_text(struct ast_channel *c, const char *text); -static int oss_hangup(struct ast_channel *c); -static int oss_answer(struct ast_channel *c); -static struct ast_frame *oss_read(struct ast_channel *chan); -static int oss_call(struct ast_channel *c, char *dest, int timeout); -static int oss_write(struct ast_channel *chan, struct ast_frame *f); -static int oss_indicate(struct ast_channel *chan, int cond); -static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan); - -static const struct ast_channel_tech oss_tech = { - .type = type, - .description = tdesc, - .capabilities = AST_FORMAT_SLINEAR, - .requester = oss_request, - .send_digit = oss_digit, - .send_text = oss_text, - .hangup = oss_hangup, - .answer = oss_answer, - .read = oss_read, - .call = oss_call, - .write = oss_write, - .indicate = oss_indicate, - .fixup = oss_fixup, -}; - -static int time_has_passed(void) -{ - struct timeval tv; - int ms; - gettimeofday(&tv, NULL); - ms = (tv.tv_sec - lasttime.tv_sec) * 1000 + - (tv.tv_usec - lasttime.tv_usec) / 1000; - if (ms > MIN_SWITCH_TIME) - return -1; - return 0; -} - -/* Number of buffers... Each is FRAMESIZE/8 ms long. For example - with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, - usually plenty. */ - -static pthread_t sthread; - -#define MAX_BUFFER_SIZE 100 -static int buffersize = 3; - -static int full_duplex = 0; - -/* Are we reading or writing (simulated full duplex) */ -static int readmode = 1; - -/* File descriptor for sound device */ -static int sounddev = -1; - -static int autoanswer = 1; - -#if 0 -static int calc_loudness(short *frame) -{ - int sum = 0; - int x; - for (x=0;x<FRAME_SIZE;x++) { - if (frame[x] < 0) - sum -= frame[x]; - else - sum += frame[x]; - } - sum = sum/FRAME_SIZE; - return sum; -} -#endif - -static int cursound = -1; -static int sampsent = 0; -static int silencelen=0; -static int offset=0; -static int nosound=0; - -static int send_sound(void) -{ - short myframe[FRAME_SIZE]; - int total = FRAME_SIZE; - short *frame = NULL; - int amt=0; - int res; - int myoff; - audio_buf_info abi; - if (cursound > -1) { - res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi); - if (res) { - ast_log(LOG_WARNING, "Unable to read output space\n"); - return -1; - } - /* Calculate how many samples we can send, max */ - if (total > (abi.fragments * abi.fragsize / 2)) - total = abi.fragments * abi.fragsize / 2; - res = total; - if (sampsent < sounds[cursound].samplen) { - myoff=0; - while(total) { - amt = total; - if (amt > (sounds[cursound].datalen - offset)) - amt = sounds[cursound].datalen - offset; - memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2); - total -= amt; - offset += amt; - sampsent += amt; - myoff += amt; - if (offset >= sounds[cursound].datalen) - offset = 0; - } - /* Set it up for silence */ - if (sampsent >= sounds[cursound].samplen) - silencelen = sounds[cursound].silencelen; - frame = myframe; - } else { - if (silencelen > 0) { - frame = silence; - silencelen -= res; - } else { - if (sounds[cursound].repeat) { - /* Start over */ - sampsent = 0; - offset = 0; - } else { - cursound = -1; - nosound = 0; - } - } - } - if (frame) - res = write(sounddev, frame, res * 2); - if (res > 0) - return 0; - return res; - } - return 0; -} - -static void *sound_thread(void *unused) -{ - fd_set rfds; - fd_set wfds; - int max; - int res; - char ign[4096]; - if (read(sounddev, ign, sizeof(sounddev)) < 0) - ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno)); - for(;;) { - FD_ZERO(&rfds); - FD_ZERO(&wfds); - max = sndcmd[0]; - FD_SET(sndcmd[0], &rfds); - if (!oss.owner) { - FD_SET(sounddev, &rfds); - if (sounddev > max) - max = sounddev; - } - if (cursound > -1) { - FD_SET(sounddev, &wfds); - if (sounddev > max) - max = sounddev; - } - res = ast_select(max + 1, &rfds, &wfds, NULL, NULL); - if (res < 1) { - ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno)); - continue; - } - if (FD_ISSET(sndcmd[0], &rfds)) { - read(sndcmd[0], &cursound, sizeof(cursound)); - silencelen = 0; - offset = 0; - sampsent = 0; - } - if (FD_ISSET(sounddev, &rfds)) { - /* Ignore read */ - if (read(sounddev, ign, sizeof(ign)) < 0) - ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno)); - } - if (FD_ISSET(sounddev, &wfds)) - if (send_sound()) - ast_log(LOG_WARNING, "Failed to write sound\n"); - } - /* Never reached */ - return NULL; -} - -#if 0 -static int silence_suppress(short *buf) -{ -#define SILBUF 3 - int loudness; - static int silentframes = 0; - static char silbuf[FRAME_SIZE * 2 * SILBUF]; - static int silbufcnt=0; - if (!silencesuppression) - return 0; - loudness = calc_loudness((short *)(buf)); - if (option_debug) - ast_log(LOG_DEBUG, "loudness is %d\n", loudness); - if (loudness < silencethreshold) { - silentframes++; - silbufcnt++; - /* Keep track of the last few bits of silence so we can play - them as lead-in when the time is right */ - if (silbufcnt >= SILBUF) { - /* Make way for more buffer */ - memmove(silbuf, silbuf + FRAME_SIZE * 2, FRAME_SIZE * 2 * (SILBUF - 1)); - silbufcnt--; - } - memcpy(silbuf + FRAME_SIZE * 2 * silbufcnt, buf, FRAME_SIZE * 2); - if (silentframes > 10) { - /* We've had plenty of silence, so compress it now */ - return 1; - } - } else { - silentframes=0; - /* Write any buffered silence we have, it may have something - important */ - if (silbufcnt) { - write(sounddev, silbuf, silbufcnt * FRAME_SIZE); - silbufcnt = 0; - } - } - return 0; -} -#endif - -static int setformat(void) -{ - int fmt, desired, res, fd = sounddev; - static int warnedalready = 0; - static int warnedalready2 = 0; - -#if __BYTE_ORDER == __LITTLE_ENDIAN - fmt = AFMT_S16_LE; -#else - fmt = AFMT_S16_BE; -#endif - - res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt); - if (res < 0) { - ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n"); - return -1; - } - res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0); - - /* Check to see if duplex set (FreeBSD Bug)*/ - res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt); - - if ((fmt & DSP_CAP_DUPLEX) && !res) { - if (option_verbose > 1) - ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n"); - full_duplex = -1; - } - fmt = 0; - res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt); - if (res < 0) { - ast_log(LOG_WARNING, "Failed to set audio device to mono\n"); - return -1; - } - /* 8000 Hz desired */ - desired = 8000; - fmt = desired; - res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt); - if (res < 0) { - ast_log(LOG_WARNING, "Failed to set audio device to mono\n"); - return -1; - } - if (fmt != desired) { - if (!warnedalready++) - ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt); - } -#if 1 - fmt = BUFFER_FMT; - res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt); - if (res < 0) { - if (!warnedalready2++) - ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n"); - } -#endif - return 0; -} - -static int soundcard_setoutput(int force) -{ - /* Make sure the soundcard is in output mode. */ - int fd = sounddev; - if (full_duplex || (!readmode && !force)) - return 0; - readmode = 0; - if (force || time_has_passed()) { - ioctl(sounddev, SNDCTL_DSP_RESET, 0); - /* Keep the same fd reserved by closing the sound device and copying stdin at the same - time. */ - /* dup2(0, sound); */ - close(sounddev); - fd = open(DEV_DSP, O_WRONLY |O_NONBLOCK); - if (fd < 0) { - ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno)); - return -1; - } - /* dup2 will close the original and make fd be sound */ - if (dup2(fd, sounddev) < 0) { - ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno)); - return -1; - } - if (setformat()) { - return -1; - } - return 0; - } - return 1; -} - -static int soundcard_setinput(int force) -{ - int fd = sounddev; - if (full_duplex || (readmode && !force)) - return 0; - readmode = -1; - if (force || time_has_passed()) { - ioctl(sounddev, SNDCTL_DSP_RESET, 0); - close(sounddev); - /* dup2(0, sound); */ - fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK); - if (fd < 0) { - ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno)); - return -1; - } - /* dup2 will close the original and make fd be sound */ - if (dup2(fd, sounddev) < 0) { - ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno)); - return -1; - } - if (setformat()) { - return -1; - } - return 0; - } - return 1; -} - -static int soundcard_init(void) -{ - /* Assume it's full duplex for starters */ - int fd = open(DEV_DSP, O_RDWR | O_NONBLOCK); - if (fd < 0) { - ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno)); - return fd; - } - gettimeofday(&lasttime, NULL); - sounddev = fd; - setformat(); - if (!full_duplex) - soundcard_setinput(1); - return sounddev; -} - -static int oss_digit(struct ast_channel *c, char digit) -{ - ast_verbose( " << Console Received digit %c >> \n", digit); - return 0; -} - -static int oss_text(struct ast_channel *c, const char *text) -{ - ast_verbose( " << Console Received text %s >> \n", text); - return 0; -} - -static int oss_call(struct ast_channel *c, char *dest, int timeout) -{ - int res = 3; - struct ast_frame f = { 0, }; - ast_verbose( " << Call placed to '%s' on console >> \n", dest); - if (autoanswer) { - ast_verbose( " << Auto-answered >> \n" ); - f.frametype = AST_FRAME_CONTROL; - f.subclass = AST_CONTROL_ANSWER; - ast_queue_frame(c, &f); - } else { - nosound = 1; - ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n"); - f.frametype = AST_FRAME_CONTROL; - f.subclass = AST_CONTROL_RINGING; - ast_queue_frame(c, &f); - write(sndcmd[1], &res, sizeof(res)); - } - return 0; -} - -static void answer_sound(void) -{ - int res; - nosound = 1; - res = 4; - write(sndcmd[1], &res, sizeof(res)); - -} - -static int oss_answer(struct ast_channel *c) -{ - ast_verbose( " << Console call has been answered >> \n"); - answer_sound(); - ast_setstate(c, AST_STATE_UP); - cursound = -1; - nosound=0; - return 0; -} - -static int oss_hangup(struct ast_channel *c) -{ - int res = 0; - cursound = -1; - c->tech_pvt = NULL; - oss.owner = NULL; - ast_verbose( " << Hangup on console >> \n"); - ast_mutex_lock(&usecnt_lock); - usecnt--; - ast_mutex_unlock(&usecnt_lock); - if (hookstate) { - if (autoanswer) { - /* Assume auto-hangup too */ - hookstate = 0; - } else { - /* Make congestion noise */ - res = 2; - write(sndcmd[1], &res, sizeof(res)); - hookstate = 0; - } - } - return 0; -} - -static int soundcard_writeframe(short *data) -{ - /* Write an exactly FRAME_SIZE sized of frame */ - static int bufcnt = 0; - static short buffer[FRAME_SIZE * MAX_BUFFER_SIZE * 5]; - struct audio_buf_info info; - int res; - int fd = sounddev; - static int warned=0; - if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) { - if (!warned) - ast_log(LOG_WARNING, "Error reading output space\n"); - bufcnt = buffersize; - warned++; - } - if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) { - /* We've run out of stuff, buffer again */ - bufcnt = 0; - } - if (bufcnt == buffersize) { - /* Write sample immediately */ - res = write(fd, ((void *)data), FRAME_SIZE * 2); - } else { - /* Copy the data into our buffer */ - res = FRAME_SIZE * 2; - memcpy(buffer + (bufcnt * FRAME_SIZE), data, FRAME_SIZE * 2); - bufcnt++; - if (bufcnt == buffersize) { - res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize); - } - } - return res; -} - - -static int oss_write(struct ast_channel *chan, struct ast_frame *f) -{ - int res; - static char sizbuf[8000]; - static int sizpos = 0; - int len = sizpos; - int pos; - /* Immediately return if no sound is enabled */ - if (nosound) - return 0; - /* Stop any currently playing sound */ - cursound = -1; - if (!full_duplex && !playbackonly) { - /* If we're half duplex, we have to switch to read mode - to honor immediate needs if necessary. But if we are in play - back only mode, then we don't switch because the console - is only being used one way -- just to playback something. */ - res = soundcard_setinput(1); - if (res < 0) { - ast_log(LOG_WARNING, "Unable to set device to input mode\n"); - return -1; - } - return 0; - } - res = soundcard_setoutput(0); - if (res < 0) { - ast_log(LOG_WARNING, "Unable to set output device\n"); - return -1; - } else if (res > 0) { - /* The device is still in read mode, and it's too soon to change it, - so just pretend we wrote it */ - return 0; - } - /* We have to digest the frame in 160-byte portions */ - if (f->datalen > sizeof(sizbuf) - sizpos) { - ast_log(LOG_WARNING, "Frame too large\n"); - return -1; - } - memcpy(sizbuf + sizpos, f->data, f->datalen); - len += f->datalen; - pos = 0; - while(len - pos > FRAME_SIZE * 2) { - soundcard_writeframe((short *)(sizbuf + pos)); - pos += FRAME_SIZE * 2; - } - if (len - pos) - memmove(sizbuf, sizbuf + pos, len - pos); - sizpos = len - pos; - return 0; -} - -static struct ast_frame *oss_read(struct ast_channel *chan) -{ - static struct ast_frame f; - static char buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET]; - static int readpos = 0; - int res; - -#if 0 - ast_log(LOG_DEBUG, "oss_read()\n"); -#endif - - f.frametype = AST_FRAME_NULL; - f.subclass = 0; - f.samples = 0; - f.datalen = 0; - f.data = NULL; - f.offset = 0; - f.src = type; - f.mallocd = 0; - f.delivery.tv_sec = 0; - f.delivery.tv_usec = 0; - - res = soundcard_setinput(0); - if (res < 0) { - ast_log(LOG_WARNING, "Unable to set input mode\n"); - return NULL; - } - if (res > 0) { - /* Theoretically shouldn't happen, but anyway, return a NULL frame */ - return &f; - } - res = read(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos); - if (res < 0) { - ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno)); -#if 0 - CRASH; -#endif - return NULL; - } - readpos += res; - - if (readpos >= FRAME_SIZE * 2) { - /* A real frame */ - readpos = 0; - if (chan->_state != AST_STATE_UP) { - /* Don't transmit unless it's up */ - return &f; - } - f.frametype = AST_FRAME_VOICE; - f.subclass = AST_FORMAT_SLINEAR; - f.samples = FRAME_SIZE; - f.datalen = FRAME_SIZE * 2; - f.data = buf + AST_FRIENDLY_OFFSET; - f.offset = AST_FRIENDLY_OFFSET; - f.src = type; - f.mallocd = 0; - f.delivery.tv_sec = 0; - f.delivery.tv_usec = 0; -#if 0 - { static int fd = -1; - if (fd < 0) - fd = open("output.raw", O_RDWR | O_TRUNC | O_CREAT); - write(fd, f.data, f.datalen); - } -#endif - } - return &f; -} - -static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan) -{ - struct chan_oss_pvt *p = newchan->tech_pvt; - p->owner = newchan; - return 0; -} - -static int oss_indicate(struct ast_channel *chan, int cond) -{ - int res; - switch(cond) { - case AST_CONTROL_BUSY: - res = 1; - break; - case AST_CONTROL_CONGESTION: - res = 2; - break; - case AST_CONTROL_RINGING: - res = 0; - break; - case -1: - cursound = -1; - return 0; - default: - ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name); - return -1; - } - if (res > -1) { - write(sndcmd[1], &res, sizeof(res)); - } - return 0; -} - -static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state) -{ - struct ast_channel *tmp; - tmp = ast_channel_alloc(1); - if (tmp) { - tmp->tech = &oss_tech; - snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5); - tmp->type = type; - tmp->fds[0] = sounddev; - tmp->nativeformats = AST_FORMAT_SLINEAR; - tmp->readformat = AST_FORMAT_SLINEAR; - tmp->writeformat = AST_FORMAT_SLINEAR; - tmp->tech_pvt = p; - if (!ast_strlen_zero(p->context)) - strncpy(tmp->context, p->context, sizeof(tmp->context)-1); - if (!ast_strlen_zero(p->exten)) - strncpy(tmp->exten, p->exten, sizeof(tmp->exten)-1); - if (!ast_strlen_zero(language)) - strncpy(tmp->language, language, sizeof(tmp->language)-1); - p->owner = tmp; - ast_setstate(tmp, state); - ast_mutex_lock(&usecnt_lock); - usecnt++; - ast_mutex_unlock(&usecnt_lock); - ast_update_use_count(); - if (state != AST_STATE_DOWN) { - if (ast_pbx_start(tmp)) { - ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name); - ast_hangup(tmp); - tmp = NULL; - } - } - } - return tmp; -} - -static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause) -{ - int oldformat = format; - struct ast_channel *tmp; - format &= AST_FORMAT_SLINEAR; - if (!format) { - ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat); - return NULL; - } - if (oss.owner) { - ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n"); - *cause = AST_CAUSE_BUSY; - return NULL; - } - tmp= oss_new(&oss, AST_STATE_DOWN); - if (!tmp) { - ast_log(LOG_WARNING, "Unable to create new OSS channel\n"); - } - return tmp; -} - -static int console_autoanswer(int fd, int argc, char *argv[]) -{ - if ((argc != 1) && (argc != 2)) - return RESULT_SHOWUSAGE; - if (argc == 1) { - ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off"); - return RESULT_SUCCESS; - } else { - if (!strcasecmp(argv[1], "on")) - autoanswer = -1; - else if (!strcasecmp(argv[1], "off")) - autoanswer = 0; - else - return RESULT_SHOWUSAGE; - } - return RESULT_SUCCESS; -} - -static char *autoanswer_complete(char *line, char *word, int pos, int state) -{ -#ifndef MIN -#define MIN(a,b) ((a) < (b) ? (a) : (b)) -#endif - switch(state) { - case 0: - if (!ast_strlen_zero(word) && !strncasecmp(word, "on", MIN(strlen(word), 2))) - return strdup("on"); - case 1: - if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3))) - return strdup("off"); - default: - return NULL; - } - return NULL; -} - -static char autoanswer_usage[] = -"Usage: autoanswer [on|off]\n" -" Enables or disables autoanswer feature. If used without\n" -" argument, displays the current on/off status of autoanswer.\n" -" The default value of autoanswer is in 'oss.conf'.\n"; - -static int console_answer(int fd, int argc, char *argv[]) -{ - struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER }; - if (argc != 1) - return RESULT_SHOWUSAGE; - if (!oss.owner) { - ast_cli(fd, "No one is calling us\n"); - return RESULT_FAILURE; - } - hookstate = 1; - cursound = -1; - ast_queue_frame(oss.owner, &f); - answer_sound(); - return RESULT_SUCCESS; -} - -static char sendtext_usage[] = -"Usage: send text <message>\n" -" Sends a text message for display on the remote terminal.\n"; - -static int console_sendtext(int fd, int argc, char *argv[]) -{ - int tmparg = 2; - char text2send[256] = ""; - struct ast_frame f = { 0, }; - if (argc < 2) - return RESULT_SHOWUSAGE; - if (!oss.owner) { - ast_cli(fd, "No one is calling us\n"); - return RESULT_FAILURE; - } - if (!ast_strlen_zero(text2send)) - ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n"); - text2send[0] = '\0'; - while(tmparg < argc) { - strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1); - strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1); - } - if (!ast_strlen_zero(text2send)) { - f.frametype = AST_FRAME_TEXT; - f.subclass = 0; - f.data = text2send; - f.datalen = strlen(text2send); - ast_queue_frame(oss.owner, &f); - } - return RESULT_SUCCESS; -} - -static char answer_usage[] = -"Usage: answer\n" -" Answers an incoming call on the console (OSS) channel.\n"; - -static int console_hangup(int fd, int argc, char *argv[]) -{ - if (argc != 1) - return RESULT_SHOWUSAGE; - cursound = -1; - if (!oss.owner && !hookstate) { - ast_cli(fd, "No call to hangup up\n"); - return RESULT_FAILURE; - } - hookstate = 0; - if (oss.owner) { - ast_queue_hangup(oss.owner); - } - return RESULT_SUCCESS; -} - -static int console_flash(int fd, int argc, char *argv[]) -{ - struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH }; - if (argc != 1) - return RESULT_SHOWUSAGE; - cursound = -1; - if (!oss.owner) { - ast_cli(fd, "No call to flash\n"); - return RESULT_FAILURE; - } - hookstate = 0; - if (oss.owner) { - ast_queue_frame(oss.owner, &f); - } - return RESULT_SUCCESS; -} - -static char hangup_usage[] = -"Usage: hangup\n" -" Hangs up any call currently placed on the console.\n"; - - -static char flash_usage[] = -"Usage: flash\n" -" Flashes the call currently placed on the console.\n"; - -static int console_dial(int fd, int argc, char *argv[]) -{ - char tmp[256], *tmp2; - char *mye, *myc; - int x; - struct ast_frame f = { AST_FRAME_DTMF, 0 }; - if ((argc != 1) && (argc != 2)) - return RESULT_SHOWUSAGE; - if (oss.owner) { - if (argc == 2) { - for (x=0;x<strlen(argv[1]);x++) { - f.subclass = argv[1][x]; - ast_queue_frame(oss.owner, &f); - } - } else { - ast_cli(fd, "You're already in a call. You can use this only to dial digits until you hangup\n"); - return RESULT_FAILURE; - } - return RESULT_SUCCESS; - } - mye = exten; - myc = context; - if (argc == 2) { - char *stringp=NULL; - strncpy(tmp, argv[1], sizeof(tmp)-1); - stringp=tmp; - strsep(&stringp, "@"); - tmp2 = strsep(&stringp, "@"); - if (!ast_strlen_zero(tmp)) - mye = tmp; - if (!ast_strlen_zero(tmp2)) - myc = tmp2; - } - if (ast_exists_extension(NULL, myc, mye, 1, NULL)) { - strncpy(oss.exten, mye, sizeof(oss.exten)-1); - strncpy(oss.context, myc, sizeof(oss.context)-1); - hookstate = 1; - oss_new(&oss, AST_STATE_RINGING); - } else - ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc); - return RESULT_SUCCESS; -} - -static char dial_usage[] = -"Usage: dial [extension[@context]]\n" -" Dials a given extensison (and context if specified)\n"; - -static int console_transfer(int fd, int argc, char *argv[]) -{ - char tmp[256]; - char *context; - if (argc != 2) - return RESULT_SHOWUSAGE; - if (oss.owner && ast_bridged_channel(oss.owner)) { - strncpy(tmp, argv[1], sizeof(tmp) - 1); - context = strchr(tmp, '@'); - if (context) { - *context = '\0'; - context++; - } else - context = oss.owner->context; - if (ast_exists_extension(ast_bridged_channel(oss.owner), context, tmp, 1, ast_bridged_channel(oss.owner)->cid.cid_num)) { - ast_cli(fd, "Whee, transferring %s to %s@%s.\n", - ast_bridged_channel(oss.owner)->name, tmp, context); - if (ast_async_goto(ast_bridged_channel(oss.owner), context, tmp, 1)) - ast_cli(fd, "Failed to transfer :(\n"); - } else { - ast_cli(fd, "No such extension exists\n"); - } - } else { - ast_cli(fd, "There is no call to transfer\n"); - } - return RESULT_SUCCESS; -} - -static char transfer_usage[] = -"Usage: transfer <extension>[@context]\n" -" Transfers the currently connected call to the given extension (and\n" -"context if specified)\n"; - -static struct ast_cli_entry myclis[] = { - { { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage }, - { { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage }, - { { "flash", NULL }, console_flash, "Flash a call on the console", flash_usage }, - { { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage }, - { { "transfer", NULL }, console_transfer, "Transfer a call to a different extension", transfer_usage }, - { { "send", "text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage }, - { { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete } -}; - -int load_module() -{ - int res; - int x; - struct ast_config *cfg; - struct ast_variable *v; - res = pipe(sndcmd); - if (res) { - ast_log(LOG_ERROR, "Unable to create pipe\n"); - return -1; - } - res = soundcard_init(); - if (res < 0) { - if (option_verbose > 1) { - ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n"); - ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n"); - } - return 0; - } - if (!full_duplex) - ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n"); - res = ast_channel_register(&oss_tech); - if (res < 0) { - ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type); - return -1; - } - for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++) - ast_cli_register(myclis + x); - if ((cfg = ast_config_load(config))) { - v = ast_variable_browse(cfg, "general"); - while(v) { - if (!strcasecmp(v->name, "autoanswer")) - autoanswer = ast_true(v->value); - else if (!strcasecmp(v->name, "silencesuppression")) - silencesuppression = ast_true(v->value); - else if (!strcasecmp(v->name, "silencethreshold")) - silencethreshold = atoi(v->value); - else if (!strcasecmp(v->name, "context")) - strncpy(context, v->value, sizeof(context)-1); - else if (!strcasecmp(v->name, "language")) - strncpy(language, v->value, sizeof(language)-1); - else if (!strcasecmp(v->name, "extension")) - strncpy(exten, v->value, sizeof(exten)-1); - else if (!strcasecmp(v->name, "playbackonly")) - playbackonly = ast_true(v->value); - v=v->next; - } - ast_config_destroy(cfg); - } - ast_pthread_create(&sthread, NULL, sound_thread, NULL); - return 0; -} - - - -int unload_module() -{ - int x; - - ast_channel_unregister(&oss_tech); - for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++) - ast_cli_unregister(myclis + x); - close(sounddev); - if (sndcmd[0] > 0) { - close(sndcmd[0]); - close(sndcmd[1]); - } - if (oss.owner) - ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD); - if (oss.owner) - return -1; - return 0; -} - -char *description() -{ - return (char *) desc; -} - -int usecount() -{ - return usecnt; -} - -char *key() -{ - return ASTERISK_GPL_KEY; -} |