1029 lines
24 KiB
C
Executable File
1029 lines
24 KiB
C
Executable File
/*
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* Asterisk -- A telephony toolkit for Linux.
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*
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* Use /dev/dsp as a channel, and the console to command it :).
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*
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* The full-duplex "simulation" is pretty weak. This is generally a
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* VERY BADLY WRITTEN DRIVER so please don't use it as a model for
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* writing a driver.
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*
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* Copyright (C) 1999, Mark Spencer
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*
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* Mark Spencer <markster@linux-support.net>
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License
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*/
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#include <asterisk/frame.h>
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#include <asterisk/logger.h>
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#include <asterisk/channel.h>
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#include <asterisk/module.h>
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#include <asterisk/channel_pvt.h>
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#include <asterisk/options.h>
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#include <asterisk/pbx.h>
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#include <asterisk/config.h>
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#include <asterisk/cli.h>
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#include <unistd.h>
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#include <fcntl.h>
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#include <errno.h>
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#include <sys/ioctl.h>
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#include <sys/time.h>
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#include <string.h>
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#include <stdlib.h>
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#include <stdio.h>
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#include <linux/soundcard.h>
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#include "busy.h"
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#include "ringtone.h"
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#include "ring10.h"
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#include "answer.h"
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/* Which device to use */
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#define DEV_DSP "/dev/dsp"
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/* Lets use 160 sample frames, just like GSM. */
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#define FRAME_SIZE 160
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/* When you set the frame size, you have to come up with
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the right buffer format as well. */
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/* 5 64-byte frames = one frame */
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#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
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/* Don't switch between read/write modes faster than every 300 ms */
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#define MIN_SWITCH_TIME 600
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static struct timeval lasttime;
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static int usecnt;
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static int needanswer = 0;
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static int needringing = 0;
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static int needhangup = 0;
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static int silencesuppression = 0;
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static int silencethreshold = 1000;
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static char digits[80] = "";
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static char text2send[80] = "";
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static pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
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static char *type = "Console";
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static char *desc = "OSS Console Channel Driver";
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static char *tdesc = "OSS Console Channel Driver";
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static char *config = "oss.conf";
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static char context[AST_MAX_EXTENSION] = "default";
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static char language[MAX_LANGUAGE] = "";
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static char exten[AST_MAX_EXTENSION] = "s";
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/* Command pipe */
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static int cmd[2];
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int hookstate=0;
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static short silence[FRAME_SIZE] = {0, };
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struct sound {
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int ind;
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short *data;
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int datalen;
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int samplen;
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int silencelen;
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int repeat;
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};
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static struct sound sounds[] = {
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{ AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
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{ AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 },
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{ AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 },
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{ AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 },
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{ AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 },
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};
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/* Sound command pipe */
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static int sndcmd[2];
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static struct chan_oss_pvt {
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/* We only have one OSS structure -- near sighted perhaps, but it
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keeps this driver as simple as possible -- as it should be. */
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struct ast_channel *owner;
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char exten[AST_MAX_EXTENSION];
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char context[AST_MAX_EXTENSION];
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} oss;
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static int time_has_passed()
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{
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struct timeval tv;
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int ms;
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gettimeofday(&tv, NULL);
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ms = (tv.tv_sec - lasttime.tv_sec) * 1000 +
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(tv.tv_usec - lasttime.tv_usec) / 1000;
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if (ms > MIN_SWITCH_TIME)
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return -1;
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return 0;
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}
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/* Number of buffers... Each is FRAMESIZE/8 ms long. For example
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with 160 sample frames, and a buffer size of 3, we have a 60ms buffer,
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usually plenty. */
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pthread_t sthread;
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#define MAX_BUFFER_SIZE 100
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static int buffersize = 3;
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static int full_duplex = 0;
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/* Are we reading or writing (simulated full duplex) */
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static int readmode = 1;
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/* File descriptor for sound device */
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static int sounddev = -1;
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static int autoanswer = 1;
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static int calc_loudness(short *frame)
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{
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int sum = 0;
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int x;
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for (x=0;x<FRAME_SIZE;x++) {
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if (frame[x] < 0)
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sum -= frame[x];
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else
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sum += frame[x];
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}
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sum = sum/FRAME_SIZE;
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return sum;
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}
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static int cursound = -1;
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static int sampsent = 0;
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static int silencelen=0;
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static int offset=0;
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static int nosound=0;
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static int send_sound(void)
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{
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short myframe[FRAME_SIZE];
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int total = FRAME_SIZE;
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short *frame = NULL;
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int amt=0;
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int res;
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int myoff;
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audio_buf_info abi;
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if (cursound > -1) {
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res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi);
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if (res) {
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ast_log(LOG_WARNING, "Unable to read output space\n");
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return -1;
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}
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/* Calculate how many samples we can send, max */
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if (total > (abi.fragments * abi.fragsize / 2))
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total = abi.fragments * abi.fragsize / 2;
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res = total;
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if (sampsent < sounds[cursound].samplen) {
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myoff=0;
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while(total) {
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amt = total;
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if (amt > (sounds[cursound].datalen - offset))
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amt = sounds[cursound].datalen - offset;
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memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2);
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total -= amt;
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offset += amt;
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sampsent += amt;
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myoff += amt;
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if (offset >= sounds[cursound].datalen)
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offset = 0;
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}
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/* Set it up for silence */
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if (sampsent >= sounds[cursound].samplen)
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silencelen = sounds[cursound].silencelen;
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frame = myframe;
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} else {
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if (silencelen > 0) {
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frame = silence;
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silencelen -= res;
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} else {
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if (sounds[cursound].repeat) {
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/* Start over */
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sampsent = 0;
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offset = 0;
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} else {
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cursound = -1;
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nosound = 0;
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}
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}
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}
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res = write(sounddev, frame, res * 2);
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if (res > 0)
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return 0;
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return res;
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}
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return 0;
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}
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static void *sound_thread(void *unused)
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{
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fd_set rfds;
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fd_set wfds;
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int max;
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int res;
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for(;;) {
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FD_ZERO(&rfds);
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FD_ZERO(&wfds);
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max = sndcmd[0];
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FD_SET(sndcmd[0], &rfds);
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if (cursound > -1) {
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FD_SET(sounddev, &wfds);
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if (sounddev > max)
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max = sounddev;
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}
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res = select(max + 1, &rfds, &wfds, NULL, NULL);
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if (res < 1) {
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ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
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continue;
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}
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if (FD_ISSET(sndcmd[0], &rfds)) {
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read(sndcmd[0], &cursound, sizeof(cursound));
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silencelen = 0;
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offset = 0;
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sampsent = 0;
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}
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if (FD_ISSET(sounddev, &wfds))
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if (send_sound())
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ast_log(LOG_WARNING, "Failed to write sound\n");
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}
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/* Never reached */
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return NULL;
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}
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#if 0
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static int silence_suppress(short *buf)
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{
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#define SILBUF 3
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int loudness;
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static int silentframes = 0;
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static char silbuf[FRAME_SIZE * 2 * SILBUF];
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static int silbufcnt=0;
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if (!silencesuppression)
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return 0;
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loudness = calc_loudness((short *)(buf));
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if (option_debug)
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ast_log(LOG_DEBUG, "loudness is %d\n", loudness);
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if (loudness < silencethreshold) {
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silentframes++;
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silbufcnt++;
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/* Keep track of the last few bits of silence so we can play
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them as lead-in when the time is right */
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if (silbufcnt >= SILBUF) {
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/* Make way for more buffer */
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memmove(silbuf, silbuf + FRAME_SIZE * 2, FRAME_SIZE * 2 * (SILBUF - 1));
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silbufcnt--;
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}
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memcpy(silbuf + FRAME_SIZE * 2 * silbufcnt, buf, FRAME_SIZE * 2);
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if (silentframes > 10) {
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/* We've had plenty of silence, so compress it now */
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return 1;
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}
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} else {
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silentframes=0;
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/* Write any buffered silence we have, it may have something
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important */
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if (silbufcnt) {
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write(sounddev, silbuf, silbufcnt * FRAME_SIZE);
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silbufcnt = 0;
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}
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}
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return 0;
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}
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#endif
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static int setformat(void)
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{
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int fmt, desired, res, fd = sounddev;
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static int warnedalready = 0;
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static int warnedalready2 = 0;
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fmt = AFMT_S16_LE;
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res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
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if (res < 0) {
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ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
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return -1;
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}
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res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
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if (res >= 0) {
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if (option_verbose > 1)
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ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
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full_duplex = -1;
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}
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fmt = 0;
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res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
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if (res < 0) {
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ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
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return -1;
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}
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/* 8000 Hz desired */
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desired = 8000;
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fmt = desired;
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res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
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if (res < 0) {
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ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
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return -1;
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}
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if (fmt != desired) {
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if (!warnedalready++)
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ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
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}
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#if 1
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fmt = BUFFER_FMT;
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res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
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if (res < 0) {
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if (!warnedalready2++)
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ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
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}
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#endif
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return 0;
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}
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static int soundcard_setoutput(int force)
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{
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/* Make sure the soundcard is in output mode. */
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int fd = sounddev;
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if (full_duplex || (!readmode && !force))
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return 0;
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readmode = 0;
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if (force || time_has_passed()) {
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ioctl(sounddev, SNDCTL_DSP_RESET);
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/* Keep the same fd reserved by closing the sound device and copying stdin at the same
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time. */
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/* dup2(0, sound); */
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close(sounddev);
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fd = open(DEV_DSP, O_WRONLY |O_NONBLOCK);
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if (fd < 0) {
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ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
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return -1;
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}
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/* dup2 will close the original and make fd be sound */
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if (dup2(fd, sounddev) < 0) {
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ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
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return -1;
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}
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if (setformat()) {
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return -1;
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}
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return 0;
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}
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return 1;
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}
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static int soundcard_setinput(int force)
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{
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int fd = sounddev;
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if (full_duplex || (readmode && !force))
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return 0;
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readmode = -1;
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if (force || time_has_passed()) {
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ioctl(sounddev, SNDCTL_DSP_RESET);
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close(sounddev);
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/* dup2(0, sound); */
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fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK);
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if (fd < 0) {
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ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
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return -1;
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}
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/* dup2 will close the original and make fd be sound */
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if (dup2(fd, sounddev) < 0) {
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ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
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return -1;
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}
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if (setformat()) {
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return -1;
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}
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return 0;
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}
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return 1;
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}
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static int soundcard_init()
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{
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/* Assume it's full duplex for starters */
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int fd = open(DEV_DSP, O_RDWR | O_NONBLOCK);
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if (fd < 0) {
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ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
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return fd;
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}
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gettimeofday(&lasttime, NULL);
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sounddev = fd;
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setformat();
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if (!full_duplex)
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soundcard_setinput(1);
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return sounddev;
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}
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static int oss_digit(struct ast_channel *c, char digit)
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{
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ast_verbose( " << Console Received digit %c >> \n", digit);
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return 0;
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}
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static int oss_text(struct ast_channel *c, char *text)
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{
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ast_verbose( " << Console Received text %s >> \n", text);
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return 0;
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}
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static int oss_call(struct ast_channel *c, char *dest, int timeout)
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{
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int res = 3;
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ast_verbose( " << Call placed to '%s' on console >> \n", dest);
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if (autoanswer) {
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ast_verbose( " << Auto-answered >> \n" );
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needanswer = 1;
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} else {
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ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
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needringing = 1;
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write(sndcmd[1], &res, sizeof(res));
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}
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return 0;
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}
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static void answer_sound(void)
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{
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int res;
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nosound = 1;
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res = 4;
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write(sndcmd[1], &res, sizeof(res));
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}
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static int oss_answer(struct ast_channel *c)
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{
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ast_verbose( " << Console call has been answered >> \n");
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answer_sound();
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c->state = AST_STATE_UP;
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cursound = -1;
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return 0;
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}
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static int oss_hangup(struct ast_channel *c)
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{
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int res;
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cursound = -1;
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c->pvt->pvt = NULL;
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oss.owner = NULL;
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ast_verbose( " << Hangup on console >> \n");
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ast_pthread_mutex_lock(&usecnt_lock);
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usecnt--;
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ast_pthread_mutex_unlock(&usecnt_lock);
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needhangup = 0;
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needanswer = 0;
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if (hookstate) {
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res = 2;
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write(sndcmd[1], &res, sizeof(res));
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}
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return 0;
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}
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|
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static int soundcard_writeframe(short *data)
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{
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/* Write an exactly FRAME_SIZE sized of frame */
|
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static int bufcnt = 0;
|
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static short buffer[FRAME_SIZE * MAX_BUFFER_SIZE * 5];
|
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struct audio_buf_info info;
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int res;
|
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int fd = sounddev;
|
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static int warned=0;
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if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) {
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if (!warned)
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ast_log(LOG_WARNING, "Error reading output space\n");
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bufcnt = buffersize;
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warned++;
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}
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if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) {
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/* We've run out of stuff, buffer again */
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bufcnt = 0;
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}
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if (bufcnt == buffersize) {
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/* Write sample immediately */
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res = write(fd, ((void *)data), FRAME_SIZE * 2);
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} else {
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/* Copy the data into our buffer */
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res = FRAME_SIZE * 2;
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memcpy(buffer + (bufcnt * FRAME_SIZE), data, FRAME_SIZE * 2);
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bufcnt++;
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if (bufcnt == buffersize) {
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res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
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}
|
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}
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return res;
|
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}
|
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|
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|
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static int oss_write(struct ast_channel *chan, struct ast_frame *f)
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{
|
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int res;
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static char sizbuf[8000];
|
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static int sizpos = 0;
|
|
int len = sizpos;
|
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int pos;
|
|
/* Immediately return if no sound is enabled */
|
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if (nosound)
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return 0;
|
|
/* Stop any currently playing sound */
|
|
cursound = -1;
|
|
if (!full_duplex && (strlen(digits) || needhangup || needanswer)) {
|
|
/* If we're half duplex, we have to switch to read mode
|
|
to honor immediate needs if necessary */
|
|
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;
|
|
int b;
|
|
int nonull=0;
|
|
|
|
#if 0
|
|
ast_log(LOG_DEBUG, "oss_read()\n");
|
|
#endif
|
|
|
|
/* Acknowledge any pending cmd */
|
|
res = read(cmd[0], &b, sizeof(b));
|
|
if (res > 0)
|
|
nonull = 1;
|
|
|
|
f.frametype = AST_FRAME_NULL;
|
|
f.subclass = 0;
|
|
f.timelen = 0;
|
|
f.datalen = 0;
|
|
f.data = NULL;
|
|
f.offset = 0;
|
|
f.src = type;
|
|
f.mallocd = 0;
|
|
|
|
if (needringing) {
|
|
f.frametype = AST_FRAME_CONTROL;
|
|
f.subclass = AST_CONTROL_RINGING;
|
|
needringing = 0;
|
|
return &f;
|
|
}
|
|
|
|
if (needhangup) {
|
|
needhangup = 0;
|
|
return NULL;
|
|
}
|
|
if (strlen(text2send)) {
|
|
f.frametype = AST_FRAME_TEXT;
|
|
f.subclass = 0;
|
|
f.data = text2send;
|
|
f.datalen = strlen(text2send);
|
|
strcpy(text2send,"");
|
|
return &f;
|
|
}
|
|
if (strlen(digits)) {
|
|
f.frametype = AST_FRAME_DTMF;
|
|
f.subclass = digits[0];
|
|
for (res=0;res<strlen(digits);res++)
|
|
digits[res] = digits[res + 1];
|
|
return &f;
|
|
}
|
|
|
|
if (needanswer) {
|
|
needanswer = 0;
|
|
f.frametype = AST_FRAME_CONTROL;
|
|
f.subclass = AST_CONTROL_ANSWER;
|
|
chan->state = AST_STATE_UP;
|
|
return &f;
|
|
}
|
|
|
|
if (nonull)
|
|
return &f;
|
|
|
|
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.timelen = FRAME_SIZE / 8;
|
|
f.datalen = FRAME_SIZE * 2;
|
|
f.data = buf + AST_FRIENDLY_OFFSET;
|
|
f.offset = AST_FRIENDLY_OFFSET;
|
|
f.src = type;
|
|
f.mallocd = 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->pvt->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;
|
|
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();
|
|
if (tmp) {
|
|
snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
|
|
tmp->type = type;
|
|
tmp->fds[0] = sounddev;
|
|
tmp->fds[1] = cmd[0];
|
|
tmp->nativeformats = AST_FORMAT_SLINEAR;
|
|
tmp->pvt->pvt = p;
|
|
tmp->pvt->send_digit = oss_digit;
|
|
tmp->pvt->send_text = oss_text;
|
|
tmp->pvt->hangup = oss_hangup;
|
|
tmp->pvt->answer = oss_answer;
|
|
tmp->pvt->read = oss_read;
|
|
tmp->pvt->call = oss_call;
|
|
tmp->pvt->write = oss_write;
|
|
tmp->pvt->indicate = oss_indicate;
|
|
tmp->pvt->fixup = oss_fixup;
|
|
if (strlen(p->context))
|
|
strncpy(tmp->context, p->context, sizeof(tmp->context));
|
|
if (strlen(p->exten))
|
|
strncpy(tmp->exten, p->exten, sizeof(tmp->exten));
|
|
if (strlen(language))
|
|
strncpy(tmp->language, language, sizeof(tmp->language));
|
|
p->owner = tmp;
|
|
tmp->state = state;
|
|
ast_pthread_mutex_lock(&usecnt_lock);
|
|
usecnt++;
|
|
ast_pthread_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(char *type, int format, void *data)
|
|
{
|
|
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");
|
|
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 (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
|
|
return strdup("on");
|
|
case 1:
|
|
if (strlen(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[])
|
|
{
|
|
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;
|
|
needanswer++;
|
|
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 = 1;
|
|
if (argc < 1)
|
|
return RESULT_SHOWUSAGE;
|
|
if (!oss.owner) {
|
|
ast_cli(fd, "No one is calling us\n");
|
|
return RESULT_FAILURE;
|
|
}
|
|
if (strlen(text2send))
|
|
ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n");
|
|
strcpy(text2send, "");
|
|
while(tmparg <= argc) {
|
|
strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send));
|
|
strncat(text2send, " ", sizeof(text2send) - strlen(text2send));
|
|
}
|
|
needanswer++;
|
|
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)
|
|
needhangup++;
|
|
return RESULT_SUCCESS;
|
|
}
|
|
|
|
static char hangup_usage[] =
|
|
"Usage: hangup\n"
|
|
" Hangs up any 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 b = 0;
|
|
if ((argc != 1) && (argc != 2))
|
|
return RESULT_SHOWUSAGE;
|
|
if (oss.owner) {
|
|
if (argc == 2) {
|
|
strncat(digits, argv[1], sizeof(digits) - strlen(digits));
|
|
/* Wake up the polling thread */
|
|
write(cmd[1], &b, sizeof(b));
|
|
} 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) {
|
|
strncpy(tmp, argv[1], sizeof(tmp));
|
|
strtok(tmp, "@");
|
|
tmp2 = strtok(NULL, "@");
|
|
if (strlen(tmp))
|
|
mye = tmp;
|
|
if (tmp2 && strlen(tmp2))
|
|
myc = tmp2;
|
|
}
|
|
if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
|
|
strncpy(oss.exten, mye, sizeof(oss.exten));
|
|
strncpy(oss.context, myc, sizeof(oss.context));
|
|
hookstate = 1;
|
|
oss_new(&oss, AST_STATE_UP);
|
|
} 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 (";
|
|
|
|
|
|
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 },
|
|
{ { "dial", NULL }, console_dial, "Dial an extension on the console", dial_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;
|
|
int flags;
|
|
struct ast_config *cfg = ast_load(config);
|
|
struct ast_variable *v;
|
|
res = pipe(cmd);
|
|
res = pipe(sndcmd);
|
|
if (res) {
|
|
ast_log(LOG_ERROR, "Unable to create pipe\n");
|
|
return -1;
|
|
}
|
|
flags = fcntl(cmd[0], F_GETFL);
|
|
fcntl(cmd[0], F_SETFL, flags | O_NONBLOCK);
|
|
flags = fcntl(cmd[1], F_GETFL);
|
|
fcntl(cmd[1], F_SETFL, flags | O_NONBLOCK);
|
|
res = soundcard_init();
|
|
if (res < 0) {
|
|
close(cmd[1]);
|
|
close(cmd[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(type, tdesc, AST_FORMAT_SLINEAR, oss_request);
|
|
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) {
|
|
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));
|
|
else if (!strcasecmp(v->name, "language"))
|
|
strncpy(language, v->value, sizeof(language));
|
|
else if (!strcasecmp(v->name, "extension"))
|
|
strncpy(exten, v->value, sizeof(exten));
|
|
v=v->next;
|
|
}
|
|
ast_destroy(cfg);
|
|
}
|
|
pthread_create(&sthread, NULL, sound_thread, NULL);
|
|
return 0;
|
|
}
|
|
|
|
|
|
|
|
int unload_module()
|
|
{
|
|
int x;
|
|
for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
|
|
ast_cli_unregister(myclis + x);
|
|
close(sounddev);
|
|
if (cmd[0] > 0) {
|
|
close(cmd[0]);
|
|
close(cmd[1]);
|
|
}
|
|
if (sndcmd[0] > 0) {
|
|
close(sndcmd[0]);
|
|
close(sndcmd[1]);
|
|
}
|
|
if (oss.owner)
|
|
ast_softhangup(oss.owner);
|
|
if (oss.owner)
|
|
return -1;
|
|
return 0;
|
|
}
|
|
|
|
char *description()
|
|
{
|
|
return desc;
|
|
}
|
|
|
|
int usecount()
|
|
{
|
|
int res;
|
|
ast_pthread_mutex_lock(&usecnt_lock);
|
|
res = usecnt;
|
|
ast_pthread_mutex_unlock(&usecnt_lock);
|
|
return res;
|
|
}
|
|
|
|
char *key()
|
|
{
|
|
return ASTERISK_GPL_KEY;
|
|
}
|