2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 1999 - 2007, Digium, Inc.
6 * Mark Spencer <markster@digium.com>
8 * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25
9 * note-this code best seen with ts=8 (8-spaces tabs) in the editor
11 * See http://www.asterisk.org for more information about
12 * the Asterisk project. Please do not directly contact
13 * any of the maintainers of this project for assistance;
14 * the project provides a web site, mailing lists and IRC
15 * channels for your use.
17 * This program is free software, distributed under the terms of
18 * the GNU General Public License Version 2. See the LICENSE file
19 * at the top of the source tree.
22 // #define HAVE_VIDEO_CONSOLE // uncomment to enable video
25 * \brief Channel driver for OSS sound cards
27 * \author Mark Spencer <markster@digium.com>
30 * \ingroup channel_drivers
34 * \li The channel chan_oss uses the configuration file \ref oss.conf
35 * \addtogroup configuration_file
38 /*! \page oss.conf oss.conf
39 * \verbinclude oss.conf.sample
44 <support_level>extended</support_level>
49 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
51 #include <ctype.h> /* isalnum() used here */
53 #include <sys/ioctl.h>
56 #include <linux/soundcard.h>
57 #elif defined(__FreeBSD__) || defined(__CYGWIN__) || defined(__GLIBC__)
58 #include <sys/soundcard.h>
60 #include <soundcard.h>
63 #include "asterisk/channel.h"
64 #include "asterisk/file.h"
65 #include "asterisk/callerid.h"
66 #include "asterisk/module.h"
67 #include "asterisk/pbx.h"
68 #include "asterisk/cli.h"
69 #include "asterisk/causes.h"
70 #include "asterisk/musiconhold.h"
71 #include "asterisk/app.h"
73 #include "console_video.h"
75 /*! Global jitterbuffer configuration - by default, jb is disabled
76 * \note Values shown here match the defaults shown in oss.conf.sample */
77 static struct ast_jb_conf default_jbconf =
81 .resync_threshold = 1000,
85 static struct ast_jb_conf global_jbconf;
88 * Basic mode of operation:
90 * we have one keyboard (which receives commands from the keyboard)
91 * and multiple headset's connected to audio cards.
92 * Cards/Headsets are named as the sections of oss.conf.
93 * The section called [general] contains the default parameters.
95 * At any time, the keyboard is attached to one card, and you
96 * can switch among them using the command 'console foo'
97 * where 'foo' is the name of the card you want.
99 * oss.conf parameters are
103 ; General config options, with default values shown.
104 ; You should use one section per device, with [general] being used
105 ; for the first device and also as a template for other devices.
107 ; All but 'debug' can go also in the device-specific sections.
109 ; debug = 0x0 ; misc debug flags, default is 0
111 ; Set the device to use for I/O
114 ; Optional mixer command to run upon startup (e.g. to set
115 ; volume levels, mutes, etc.
118 ; Software mic volume booster (or attenuator), useful for sound
119 ; cards or microphones with poor sensitivity. The volume level
120 ; is in dB, ranging from -20.0 to +20.0
121 ; boost = n ; mic volume boost in dB
123 ; Set the callerid for outgoing calls
124 ; callerid = John Doe <555-1234>
126 ; autoanswer = no ; no autoanswer on call
127 ; autohangup = yes ; hangup when other party closes
128 ; extension = s ; default extension to call
129 ; context = default ; default context for outgoing calls
130 ; language = "" ; default language
132 ; Default Music on Hold class to use when this channel is placed on hold in
133 ; the case that the music class is not set on the channel with
134 ; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
135 ; putting this one on hold did not suggest a class to use.
137 ; mohinterpret=default
139 ; If you set overridecontext to 'yes', then the whole dial string
140 ; will be interpreted as an extension, which is extremely useful
141 ; to dial SIP, IAX and other extensions which use the '@' character.
142 ; The default is 'no' just for backward compatibility, but the
143 ; suggestion is to change it.
144 ; overridecontext = no ; if 'no', the last @ will start the context
145 ; if 'yes' the whole string is an extension.
147 ; low level device parameters in case you have problems with the
148 ; device driver on your operating system. You should not touch these
149 ; unless you know what you are doing.
150 ; queuesize = 10 ; frames in device driver
151 ; frags = 8 ; argument to SETFRAGMENT
153 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
154 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
155 ; OSS channel. Defaults to "no". An enabled jitterbuffer will
156 ; be used only if the sending side can create and the receiving
157 ; side can not accept jitter. The OSS channel can't accept jitter,
158 ; thus an enabled jitterbuffer on the receive OSS side will always
159 ; be used if the sending side can create jitter.
161 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
163 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
164 ; resynchronized. Useful to improve the quality of the voice, with
165 ; big jumps in/broken timestamps, usualy sent from exotic devices
166 ; and programs. Defaults to 1000.
168 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
169 ; channel. Two implementations are currenlty available - "fixed"
170 ; (with size always equals to jbmax-size) and "adaptive" (with
171 ; variable size, actually the new jb of IAX2). Defaults to fixed.
173 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
174 ;-----------------------------------------------------------------------------------
177 ; device = /dev/dsp1 ; alternate device
181 .. and so on for the other cards.
186 * The following parameters are used in the driver:
188 * FRAME_SIZE the size of an audio frame, in samples.
189 * 160 is used almost universally, so you should not change it.
191 * FRAGS the argument for the SETFRAGMENT ioctl.
192 * Overridden by the 'frags' parameter in oss.conf
194 * Bits 0-7 are the base-2 log of the device's block size,
195 * bits 16-31 are the number of blocks in the driver's queue.
196 * There are a lot of differences in the way this parameter
197 * is supported by different drivers, so you may need to
198 * experiment a bit with the value.
199 * A good default for linux is 30 blocks of 64 bytes, which
200 * results in 6 frames of 320 bytes (160 samples).
201 * FreeBSD works decently with blocks of 256 or 512 bytes,
202 * leaving the number unspecified.
203 * Note that this only refers to the device buffer size,
204 * this module will then try to keep the lenght of audio
205 * buffered within small constraints.
207 * QUEUE_SIZE The max number of blocks actually allowed in the device
208 * driver's buffer, irrespective of the available number.
209 * Overridden by the 'queuesize' parameter in oss.conf
211 * Should be >=2, and at most as large as the hw queue above
212 * (otherwise it will never be full).
215 #define FRAME_SIZE 160
216 #define QUEUE_SIZE 10
218 #if defined(__FreeBSD__)
221 #define FRAGS ( ( (6 * 5) << 16 ) | 0x6 )
225 * XXX text message sizes are probably 256 chars, but i am
226 * not sure if there is a suitable definition anywhere.
228 #define TEXT_SIZE 256
231 #define TRYOPEN 1 /* try to open on startup */
233 #define O_CLOSE 0x444 /* special 'close' mode for device */
234 /* Which device to use */
235 #if defined( __OpenBSD__ ) || defined( __NetBSD__ )
236 #define DEV_DSP "/dev/audio"
238 #define DEV_DSP "/dev/dsp"
241 static char *config = "oss.conf"; /* default config file */
243 static int oss_debug;
246 * \brief descriptor for one of our channels.
248 * There is one used for 'default' values (from the [general] entry in
249 * the configuration file), and then one instance for each device
250 * (the default is cloned from [general], others are only created
251 * if the relevant section exists).
253 struct chan_oss_pvt {
254 struct chan_oss_pvt *next;
257 int total_blocks; /*!< total blocks in the output device */
259 enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
260 int autoanswer; /*!< Boolean: whether to answer the immediately upon calling */
261 int autohangup; /*!< Boolean: whether to hangup the call when the remote end hangs up */
262 int hookstate; /*!< Boolean: 1 if offhook; 0 if onhook */
263 char *mixer_cmd; /*!< initial command to issue to the mixer */
264 unsigned int queuesize; /*!< max fragments in queue */
265 unsigned int frags; /*!< parameter for SETFRAGMENT */
267 int warned; /*!< various flags used for warnings */
268 #define WARN_used_blocks 1
271 int w_errors; /*!< overfull in the write path */
272 struct timeval lastopen;
277 /*! boost support. BOOST_SCALE * 10 ^(BOOST_MAX/20) must
278 * be representable in 16 bits to avoid overflows.
280 #define BOOST_SCALE (1<<9)
281 #define BOOST_MAX 40 /*!< slightly less than 7 bits */
282 int boost; /*!< input boost, scaled by BOOST_SCALE */
283 char device[64]; /*!< device to open */
287 struct ast_channel *owner;
289 struct video_desc *env; /*!< parameters for video support */
291 char ext[AST_MAX_EXTENSION];
292 char ctx[AST_MAX_CONTEXT];
293 char language[MAX_LANGUAGE];
294 char cid_name[256]; /*!< Initial CallerID name */
295 char cid_num[256]; /*!< Initial CallerID number */
296 char mohinterpret[MAX_MUSICCLASS];
298 /*! buffers used in oss_write */
299 char oss_write_buf[FRAME_SIZE * 2];
301 /*! buffers used in oss_read - AST_FRIENDLY_OFFSET space for headers
302 * plus enough room for a full frame
304 char oss_read_buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
305 int readpos; /*!< read position above */
306 struct ast_frame read_f; /*!< returned by oss_read */
309 /*! forward declaration */
310 static struct chan_oss_pvt *find_desc(const char *dev);
312 static char *oss_active; /*!< the active device */
314 /*! \brief return the pointer to the video descriptor */
315 struct video_desc *get_video_desc(struct ast_channel *c)
317 struct chan_oss_pvt *o = c ? ast_channel_tech_pvt(c) : find_desc(oss_active);
318 return o ? o->env : NULL;
320 static struct chan_oss_pvt oss_default = {
322 .duplex = M_UNSET, /* XXX check this */
325 .queuesize = QUEUE_SIZE,
329 .readpos = AST_FRIENDLY_OFFSET, /* start here on reads */
330 .lastopen = { 0, 0 },
331 .boost = BOOST_SCALE,
335 static int setformat(struct chan_oss_pvt *o, int mode);
337 static struct ast_channel *oss_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor,
338 const char *data, int *cause);
339 static int oss_digit_begin(struct ast_channel *c, char digit);
340 static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration);
341 static int oss_text(struct ast_channel *c, const char *text);
342 static int oss_hangup(struct ast_channel *c);
343 static int oss_answer(struct ast_channel *c);
344 static struct ast_frame *oss_read(struct ast_channel *chan);
345 static int oss_call(struct ast_channel *c, const char *dest, int timeout);
346 static int oss_write(struct ast_channel *chan, struct ast_frame *f);
347 static int oss_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
348 static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
349 static char tdesc[] = "OSS Console Channel Driver";
351 /* cannot do const because need to update some fields at runtime */
352 static struct ast_channel_tech oss_tech = {
354 .description = tdesc,
355 .requester = oss_request,
356 .send_digit_begin = oss_digit_begin,
357 .send_digit_end = oss_digit_end,
358 .send_text = oss_text,
359 .hangup = oss_hangup,
360 .answer = oss_answer,
364 .write_video = console_write_video,
365 .indicate = oss_indicate,
370 * \brief returns a pointer to the descriptor with the given name
372 static struct chan_oss_pvt *find_desc(const char *dev)
374 struct chan_oss_pvt *o = NULL;
377 ast_log(LOG_WARNING, "null dev\n");
379 for (o = oss_default.next; o && o->name && dev && strcmp(o->name, dev) != 0; o = o->next);
382 ast_log(LOG_WARNING, "could not find <%s>\n", dev ? dev : "--no-device--");
388 * \brief split a string in extension-context, returns pointers to malloc'ed
391 * If we do not have 'overridecontext' then the last @ is considered as
392 * a context separator, and the context is overridden.
393 * This is usually not very necessary as you can play with the dialplan,
394 * and it is nice not to need it because you have '@' in SIP addresses.
396 * \return the buffer address.
398 static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
400 struct chan_oss_pvt *o = find_desc(oss_active);
402 if (ext == NULL || ctx == NULL)
403 return NULL; /* error */
407 if (src && *src != '\0')
408 *ext = ast_strdup(src);
413 if (!o->overridecontext) {
414 /* parse from the right */
415 *ctx = strrchr(*ext, '@');
424 * \brief Returns the number of blocks used in the audio output channel
426 static int used_blocks(struct chan_oss_pvt *o)
428 struct audio_buf_info info;
430 if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) {
431 if (!(o->warned & WARN_used_blocks)) {
432 ast_log(LOG_WARNING, "Error reading output space\n");
433 o->warned |= WARN_used_blocks;
438 if (o->total_blocks == 0) {
439 if (0) /* debugging */
440 ast_log(LOG_WARNING, "fragtotal %d size %d avail %d\n", info.fragstotal, info.fragsize, info.fragments);
441 o->total_blocks = info.fragments;
444 return o->total_blocks - info.fragments;
447 /*! Write an exactly FRAME_SIZE sized frame */
448 static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
453 setformat(o, O_RDWR);
455 return 0; /* not fatal */
457 * Nothing complex to manage the audio device queue.
458 * If the buffer is full just drop the extra, otherwise write.
459 * XXX in some cases it might be useful to write anyways after
460 * a number of failures, to restart the output chain.
462 res = used_blocks(o);
463 if (res > o->queuesize) { /* no room to write a block */
464 if (o->w_errors++ == 0 && (oss_debug & 0x4))
465 ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, o->w_errors);
469 return write(o->sounddev, (void *)data, FRAME_SIZE * 2);
473 * reset and close the device if opened,
474 * then open and initialize it in the desired mode,
475 * trigger reads and writes so we can start using it.
477 static int setformat(struct chan_oss_pvt *o, int mode)
479 int fmt, desired, res, fd;
481 if (o->sounddev >= 0) {
482 ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
487 if (mode == O_CLOSE) /* we are done */
489 if (ast_tvdiff_ms(ast_tvnow(), o->lastopen) < 1000)
490 return -1; /* don't open too often */
491 o->lastopen = ast_tvnow();
492 fd = o->sounddev = open(o->device, mode | O_NONBLOCK);
494 ast_log(LOG_WARNING, "Unable to re-open DSP device %s: %s\n", o->device, strerror(errno));
498 ast_channel_set_fd(o->owner, 0, fd);
500 #if __BYTE_ORDER == __LITTLE_ENDIAN
505 res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
507 ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
512 res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
513 /* Check to see if duplex set (FreeBSD Bug) */
514 res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
515 if (res == 0 && (fmt & DSP_CAP_DUPLEX)) {
516 ast_verb(2, "Console is full duplex\n");
531 res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
533 ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
536 fmt = desired = DEFAULT_SAMPLE_RATE; /* 8000 Hz desired */
537 res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
540 ast_log(LOG_WARNING, "Failed to set sample rate to %d\n", desired);
543 if (fmt != desired) {
544 if (!(o->warned & WARN_speed)) {
546 "Requested %d Hz, got %d Hz -- sound may be choppy\n",
548 o->warned |= WARN_speed;
552 * on Freebsd, SETFRAGMENT does not work very well on some cards.
553 * Default to use 256 bytes, let the user override
557 res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
559 if (!(o->warned & WARN_frag)) {
561 "Unable to set fragment size -- sound may be choppy\n");
562 o->warned |= WARN_frag;
566 /* on some cards, we need SNDCTL_DSP_SETTRIGGER to start outputting */
567 res = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT;
568 res = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &res);
569 /* it may fail if we are in half duplex, never mind */
574 * some of the standard methods supported by channels.
576 static int oss_digit_begin(struct ast_channel *c, char digit)
581 static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration)
583 /* no better use for received digits than print them */
584 ast_verbose(" << Console Received digit %c of duration %u ms >> \n",
589 static int oss_text(struct ast_channel *c, const char *text)
591 /* print received messages */
592 ast_verbose(" << Console Received text %s >> \n", text);
597 * \brief handler for incoming calls. Either autoanswer, or start ringing
599 static int oss_call(struct ast_channel *c, const char *dest, int timeout)
601 struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
602 struct ast_frame f = { AST_FRAME_CONTROL, };
603 AST_DECLARE_APP_ARGS(args,
607 char *parse = ast_strdupa(dest);
609 AST_NONSTANDARD_APP_ARGS(args, parse, '/');
611 ast_verbose(" << Call to device '%s' dnid '%s' rdnis '%s' on console from '%s' <%s> >>\n",
613 S_OR(ast_channel_dialed(c)->number.str, ""),
614 S_COR(ast_channel_redirecting(c)->from.number.valid, ast_channel_redirecting(c)->from.number.str, ""),
615 S_COR(ast_channel_caller(c)->id.name.valid, ast_channel_caller(c)->id.name.str, ""),
616 S_COR(ast_channel_caller(c)->id.number.valid, ast_channel_caller(c)->id.number.str, ""));
617 if (!ast_strlen_zero(args.flags) && strcasecmp(args.flags, "answer") == 0) {
618 f.subclass.integer = AST_CONTROL_ANSWER;
619 ast_queue_frame(c, &f);
620 } else if (!ast_strlen_zero(args.flags) && strcasecmp(args.flags, "noanswer") == 0) {
621 f.subclass.integer = AST_CONTROL_RINGING;
622 ast_queue_frame(c, &f);
623 ast_indicate(c, AST_CONTROL_RINGING);
624 } else if (o->autoanswer) {
625 ast_verbose(" << Auto-answered >> \n");
626 f.subclass.integer = AST_CONTROL_ANSWER;
627 ast_queue_frame(c, &f);
630 ast_verbose("<< Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
631 f.subclass.integer = AST_CONTROL_RINGING;
632 ast_queue_frame(c, &f);
633 ast_indicate(c, AST_CONTROL_RINGING);
639 * \brief remote side answered the phone
641 static int oss_answer(struct ast_channel *c)
643 struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
644 ast_verbose(" << Console call has been answered >> \n");
645 ast_setstate(c, AST_STATE_UP);
650 static int oss_hangup(struct ast_channel *c)
652 struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
654 ast_channel_tech_pvt_set(c, NULL);
656 ast_verbose(" << Hangup on console >> \n");
657 console_video_uninit(o->env);
658 ast_module_unref(ast_module_info->self);
660 if (o->autoanswer || o->autohangup) {
661 /* Assume auto-hangup too */
663 setformat(o, O_CLOSE);
669 /*! \brief used for data coming from the network */
670 static int oss_write(struct ast_channel *c, struct ast_frame *f)
673 struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
676 * we could receive a block which is not a multiple of our
677 * FRAME_SIZE, so buffer it locally and write to the device
678 * in FRAME_SIZE chunks.
679 * Keep the residue stored for future use.
681 src = 0; /* read position into f->data */
682 while (src < f->datalen) {
683 /* Compute spare room in the buffer */
684 int l = sizeof(o->oss_write_buf) - o->oss_write_dst;
686 if (f->datalen - src >= l) { /* enough to fill a frame */
687 memcpy(o->oss_write_buf + o->oss_write_dst, f->data.ptr + src, l);
688 soundcard_writeframe(o, (short *) o->oss_write_buf);
690 o->oss_write_dst = 0;
691 } else { /* copy residue */
692 l = f->datalen - src;
693 memcpy(o->oss_write_buf + o->oss_write_dst, f->data.ptr + src, l);
694 src += l; /* but really, we are done */
695 o->oss_write_dst += l;
701 static struct ast_frame *oss_read(struct ast_channel *c)
704 struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
705 struct ast_frame *f = &o->read_f;
707 /* XXX can be simplified returning &ast_null_frame */
708 /* prepare a NULL frame in case we don't have enough data to return */
709 memset(f, '\0', sizeof(struct ast_frame));
710 f->frametype = AST_FRAME_NULL;
711 f->src = oss_tech.type;
713 res = read(o->sounddev, o->oss_read_buf + o->readpos, sizeof(o->oss_read_buf) - o->readpos);
714 if (res < 0) /* audio data not ready, return a NULL frame */
718 if (o->readpos < sizeof(o->oss_read_buf)) /* not enough samples */
724 o->readpos = AST_FRIENDLY_OFFSET; /* reset read pointer for next frame */
725 if (ast_channel_state(c) != AST_STATE_UP) /* drop data if frame is not up */
727 /* ok we can build and deliver the frame to the caller */
728 f->frametype = AST_FRAME_VOICE;
729 ast_format_set(&f->subclass.format, AST_FORMAT_SLINEAR, 0);
730 f->samples = FRAME_SIZE;
731 f->datalen = FRAME_SIZE * 2;
732 f->data.ptr = o->oss_read_buf + AST_FRIENDLY_OFFSET;
733 if (o->boost != BOOST_SCALE) { /* scale and clip values */
735 int16_t *p = (int16_t *) f->data.ptr;
736 for (i = 0; i < f->samples; i++) {
737 x = (p[i] * o->boost) / BOOST_SCALE;
746 f->offset = AST_FRIENDLY_OFFSET;
750 static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
752 struct chan_oss_pvt *o = ast_channel_tech_pvt(newchan);
757 static int oss_indicate(struct ast_channel *c, int cond, const void *data, size_t datalen)
759 struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
763 case AST_CONTROL_INCOMPLETE:
764 case AST_CONTROL_BUSY:
765 case AST_CONTROL_CONGESTION:
766 case AST_CONTROL_RINGING:
767 case AST_CONTROL_PVT_CAUSE_CODE:
771 case AST_CONTROL_PROGRESS:
772 case AST_CONTROL_PROCEEDING:
773 case AST_CONTROL_VIDUPDATE:
774 case AST_CONTROL_SRCUPDATE:
776 case AST_CONTROL_HOLD:
777 ast_verbose(" << Console Has Been Placed on Hold >> \n");
778 ast_moh_start(c, data, o->mohinterpret);
780 case AST_CONTROL_UNHOLD:
781 ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
785 ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, ast_channel_name(c));
793 * \brief allocate a new channel.
795 static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx, int state, const char *linkedid)
797 struct ast_channel *c;
799 c = ast_channel_alloc(1, state, o->cid_num, o->cid_name, "", ext, ctx, linkedid, 0, "Console/%s", o->device + 5);
802 ast_channel_tech_set(c, &oss_tech);
804 setformat(o, O_RDWR);
805 ast_channel_set_fd(c, 0, o->sounddev); /* -1 if device closed, override later */
807 ast_format_set(ast_channel_readformat(c), AST_FORMAT_SLINEAR, 0);
808 ast_format_set(ast_channel_writeformat(c), AST_FORMAT_SLINEAR, 0);
809 ast_format_cap_add(ast_channel_nativeformats(c), ast_channel_readformat(c));
811 /* if the console makes the call, add video to the offer */
812 /* if (state == AST_STATE_RINGING) TODO XXX CONSOLE VIDEO IS DISABLED UNTIL IT GETS A MAINTAINER
813 c->nativeformats |= console_video_formats; */
815 ast_channel_tech_pvt_set(c, o);
817 if (!ast_strlen_zero(o->language))
818 ast_channel_language_set(c, o->language);
819 /* Don't use ast_set_callerid() here because it will
820 * generate a needless NewCallerID event */
821 if (!ast_strlen_zero(o->cid_num)) {
822 ast_channel_caller(c)->ani.number.valid = 1;
823 ast_channel_caller(c)->ani.number.str = ast_strdup(o->cid_num);
825 if (!ast_strlen_zero(ext)) {
826 ast_channel_dialed(c)->number.str = ast_strdup(ext);
830 ast_module_ref(ast_module_info->self);
831 ast_jb_configure(c, &global_jbconf);
832 if (state != AST_STATE_DOWN) {
833 if (ast_pbx_start(c)) {
834 ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(c));
839 console_video_start(get_video_desc(c), c); /* XXX cleanup */
844 static struct ast_channel *oss_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause)
846 struct ast_channel *c;
847 struct chan_oss_pvt *o;
848 AST_DECLARE_APP_ARGS(args,
852 char *parse = ast_strdupa(data);
854 struct ast_format tmpfmt;
856 AST_NONSTANDARD_APP_ARGS(args, parse, '/');
857 o = find_desc(args.name);
859 ast_log(LOG_WARNING, "oss_request ty <%s> data 0x%p <%s>\n", type, data, data);
861 ast_log(LOG_NOTICE, "Device %s not found\n", args.name);
862 /* XXX we could default to 'dsp' perhaps ? */
865 if (!(ast_format_cap_iscompatible(cap, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0)))) {
866 ast_log(LOG_NOTICE, "Format %s unsupported\n", ast_getformatname_multiple(buf, sizeof(buf), cap));
870 ast_log(LOG_NOTICE, "Already have a call (chan %p) on the OSS channel\n", o->owner);
871 *cause = AST_CAUSE_BUSY;
874 c = oss_new(o, NULL, NULL, AST_STATE_DOWN, requestor ? ast_channel_linkedid(requestor) : NULL);
876 ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
882 static void store_config_core(struct chan_oss_pvt *o, const char *var, const char *value);
884 /*! Generic console command handler. Basically a wrapper for a subset
885 * of config file options which are also available from the CLI
887 static char *console_cmd(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
889 struct chan_oss_pvt *o = find_desc(oss_active);
890 const char *var, *value;
893 e->command = CONSOLE_VIDEO_CMDS;
895 "Usage: " CONSOLE_VIDEO_CMDS "...\n"
896 " Generic handler for console commands.\n";
903 if (a->argc < e->args)
904 return CLI_SHOWUSAGE;
906 ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
910 var = a->argv[e->args-1];
911 value = a->argc > e->args ? a->argv[e->args] : NULL;
912 if (value) /* handle setting */
913 store_config_core(o, var, value);
914 if (!console_video_cli(o->env, var, a->fd)) /* print video-related values */
916 /* handle other values */
917 if (!strcasecmp(var, "device")) {
918 ast_cli(a->fd, "device is [%s]\n", o->device);
923 static char *console_autoanswer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
925 struct chan_oss_pvt *o = find_desc(oss_active);
929 e->command = "console {set|show} autoanswer [on|off]";
931 "Usage: console {set|show} autoanswer [on|off]\n"
932 " Enables or disables autoanswer feature. If used without\n"
933 " argument, displays the current on/off status of autoanswer.\n"
934 " The default value of autoanswer is in 'oss.conf'.\n";
941 if (a->argc == e->args - 1) {
942 ast_cli(a->fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off");
945 if (a->argc != e->args)
946 return CLI_SHOWUSAGE;
948 ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
952 if (!strcasecmp(a->argv[e->args-1], "on"))
954 else if (!strcasecmp(a->argv[e->args - 1], "off"))
957 return CLI_SHOWUSAGE;
961 /*! \brief helper function for the answer key/cli command */
962 static char *console_do_answer(int fd)
964 struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_ANSWER } };
965 struct chan_oss_pvt *o = find_desc(oss_active);
968 ast_cli(fd, "No one is calling us\n");
972 ast_queue_frame(o->owner, &f);
977 * \brief answer command from the console
979 static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
983 e->command = "console answer";
985 "Usage: console answer\n"
986 " Answers an incoming call on the console (OSS) channel.\n";
990 return NULL; /* no completion */
992 if (a->argc != e->args)
993 return CLI_SHOWUSAGE;
994 return console_do_answer(a->fd);
998 * \brief Console send text CLI command
1000 * \note concatenate all arguments into a single string. argv is NULL-terminated
1001 * so we can use it right away
1003 static char *console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
1005 struct chan_oss_pvt *o = find_desc(oss_active);
1006 char buf[TEXT_SIZE];
1008 if (cmd == CLI_INIT) {
1009 e->command = "console send text";
1011 "Usage: console send text <message>\n"
1012 " Sends a text message for display on the remote terminal.\n";
1014 } else if (cmd == CLI_GENERATE)
1017 if (a->argc < e->args + 1)
1018 return CLI_SHOWUSAGE;
1020 ast_cli(a->fd, "Not in a call\n");
1023 ast_join(buf, sizeof(buf) - 1, a->argv + e->args);
1024 if (!ast_strlen_zero(buf)) {
1025 struct ast_frame f = { 0, };
1026 int i = strlen(buf);
1028 f.frametype = AST_FRAME_TEXT;
1029 f.subclass.integer = 0;
1032 ast_queue_frame(o->owner, &f);
1037 static char *console_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
1039 struct chan_oss_pvt *o = find_desc(oss_active);
1041 if (cmd == CLI_INIT) {
1042 e->command = "console hangup";
1044 "Usage: console hangup\n"
1045 " Hangs up any call currently placed on the console.\n";
1047 } else if (cmd == CLI_GENERATE)
1050 if (a->argc != e->args)
1051 return CLI_SHOWUSAGE;
1052 if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
1053 ast_cli(a->fd, "No call to hang up\n");
1058 ast_queue_hangup_with_cause(o->owner, AST_CAUSE_NORMAL_CLEARING);
1059 setformat(o, O_CLOSE);
1063 static char *console_flash(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
1065 struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH } };
1066 struct chan_oss_pvt *o = find_desc(oss_active);
1068 if (cmd == CLI_INIT) {
1069 e->command = "console flash";
1071 "Usage: console flash\n"
1072 " Flashes the call currently placed on the console.\n";
1074 } else if (cmd == CLI_GENERATE)
1077 if (a->argc != e->args)
1078 return CLI_SHOWUSAGE;
1079 if (!o->owner) { /* XXX maybe !o->hookstate too ? */
1080 ast_cli(a->fd, "No call to flash\n");
1085 ast_queue_frame(o->owner, &f);
1089 static char *console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
1092 char *mye = NULL, *myc = NULL;
1093 struct chan_oss_pvt *o = find_desc(oss_active);
1095 if (cmd == CLI_INIT) {
1096 e->command = "console dial";
1098 "Usage: console dial [extension[@context]]\n"
1099 " Dials a given extension (and context if specified)\n";
1101 } else if (cmd == CLI_GENERATE)
1104 if (a->argc > e->args + 1)
1105 return CLI_SHOWUSAGE;
1106 if (o->owner) { /* already in a call */
1108 struct ast_frame f = { AST_FRAME_DTMF, { 0 } };
1111 if (a->argc == e->args) { /* argument is mandatory here */
1112 ast_cli(a->fd, "Already in a call. You can only dial digits until you hangup.\n");
1115 digits = a->argv[e->args];
1116 /* send the string one char at a time */
1117 for (i = 0; i < strlen(digits); i++) {
1118 f.subclass.integer = digits[i];
1119 ast_queue_frame(o->owner, &f);
1123 /* if we have an argument split it into extension and context */
1124 if (a->argc == e->args + 1)
1125 s = ast_ext_ctx(a->argv[e->args], &mye, &myc);
1126 /* supply default values if needed */
1131 if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
1133 oss_new(o, mye, myc, AST_STATE_RINGING, NULL);
1135 ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
1141 static char *console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
1143 struct chan_oss_pvt *o = find_desc(oss_active);
1147 if (cmd == CLI_INIT) {
1148 e->command = "console {mute|unmute} [toggle]";
1150 "Usage: console {mute|unmute} [toggle]\n"
1151 " Mute/unmute the microphone.\n";
1153 } else if (cmd == CLI_GENERATE)
1156 if (a->argc > e->args)
1157 return CLI_SHOWUSAGE;
1158 if (a->argc == e->args) {
1159 if (strcasecmp(a->argv[e->args-1], "toggle"))
1160 return CLI_SHOWUSAGE;
1163 s = a->argv[e->args-2];
1164 if (!strcasecmp(s, "mute"))
1165 o->mute = toggle ? !o->mute : 1;
1166 else if (!strcasecmp(s, "unmute"))
1167 o->mute = toggle ? !o->mute : 0;
1169 return CLI_SHOWUSAGE;
1170 ast_cli(a->fd, "Console mic is %s\n", o->mute ? "off" : "on");
1174 static char *console_transfer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
1176 struct chan_oss_pvt *o = find_desc(oss_active);
1177 struct ast_channel *b = NULL;
1178 char *tmp, *ext, *ctx;
1182 e->command = "console transfer";
1184 "Usage: console transfer <extension>[@context]\n"
1185 " Transfers the currently connected call to the given extension (and\n"
1186 " context if specified)\n";
1193 return CLI_SHOWUSAGE;
1196 if (o->owner == NULL || (b = ast_bridged_channel(o->owner)) == NULL) {
1197 ast_cli(a->fd, "There is no call to transfer\n");
1201 tmp = ast_ext_ctx(a->argv[2], &ext, &ctx);
1202 if (ctx == NULL) /* supply default context if needed */
1203 ctx = ast_strdupa(ast_channel_context(o->owner));
1204 if (!ast_exists_extension(b, ctx, ext, 1,
1205 S_COR(ast_channel_caller(b)->id.number.valid, ast_channel_caller(b)->id.number.str, NULL))) {
1206 ast_cli(a->fd, "No such extension exists\n");
1208 ast_cli(a->fd, "Whee, transferring %s to %s@%s.\n", ast_channel_name(b), ext, ctx);
1209 if (ast_async_goto(b, ctx, ext, 1))
1210 ast_cli(a->fd, "Failed to transfer :(\n");
1217 static char *console_active(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
1221 e->command = "console {set|show} active [<device>]";
1223 "Usage: console active [device]\n"
1224 " If used without a parameter, displays which device is the current\n"
1225 " console. If a device is specified, the console sound device is changed to\n"
1226 " the device specified.\n";
1233 ast_cli(a->fd, "active console is [%s]\n", oss_active);
1234 else if (a->argc != 4)
1235 return CLI_SHOWUSAGE;
1237 struct chan_oss_pvt *o;
1238 if (strcmp(a->argv[3], "show") == 0) {
1239 for (o = oss_default.next; o; o = o->next)
1240 ast_cli(a->fd, "device [%s] exists\n", o->name);
1243 o = find_desc(a->argv[3]);
1245 ast_cli(a->fd, "No device [%s] exists\n", a->argv[3]);
1247 oss_active = o->name;
1253 * \brief store the boost factor
1255 static void store_boost(struct chan_oss_pvt *o, const char *s)
1258 if (sscanf(s, "%30lf", &boost) != 1) {
1259 ast_log(LOG_WARNING, "invalid boost <%s>\n", s);
1262 if (boost < -BOOST_MAX) {
1263 ast_log(LOG_WARNING, "boost %s too small, using %d\n", s, -BOOST_MAX);
1265 } else if (boost > BOOST_MAX) {
1266 ast_log(LOG_WARNING, "boost %s too large, using %d\n", s, BOOST_MAX);
1269 boost = exp(log(10) * boost / 20) * BOOST_SCALE;
1271 ast_log(LOG_WARNING, "setting boost %s to %d\n", s, o->boost);
1274 static char *console_boost(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
1276 struct chan_oss_pvt *o = find_desc(oss_active);
1280 e->command = "console boost";
1282 "Usage: console boost [boost in dB]\n"
1283 " Sets or display mic boost in dB\n";
1290 ast_cli(a->fd, "boost currently %5.1f\n", 20 * log10(((double) o->boost / (double) BOOST_SCALE)));
1291 else if (a->argc == 3)
1292 store_boost(o, a->argv[2]);
1296 static struct ast_cli_entry cli_oss[] = {
1297 AST_CLI_DEFINE(console_answer, "Answer an incoming console call"),
1298 AST_CLI_DEFINE(console_hangup, "Hangup a call on the console"),
1299 AST_CLI_DEFINE(console_flash, "Flash a call on the console"),
1300 AST_CLI_DEFINE(console_dial, "Dial an extension on the console"),
1301 AST_CLI_DEFINE(console_mute, "Disable/Enable mic input"),
1302 AST_CLI_DEFINE(console_transfer, "Transfer a call to a different extension"),
1303 AST_CLI_DEFINE(console_cmd, "Generic console command"),
1304 AST_CLI_DEFINE(console_sendtext, "Send text to the remote device"),
1305 AST_CLI_DEFINE(console_autoanswer, "Sets/displays autoanswer"),
1306 AST_CLI_DEFINE(console_boost, "Sets/displays mic boost in dB"),
1307 AST_CLI_DEFINE(console_active, "Sets/displays active console"),
1311 * store the mixer argument from the config file, filtering possibly
1312 * invalid or dangerous values (the string is used as argument for
1313 * system("mixer %s")
1315 static void store_mixer(struct chan_oss_pvt *o, const char *s)
1319 for (i = 0; i < strlen(s); i++) {
1320 if (!isalnum(s[i]) && strchr(" \t-/", s[i]) == NULL) {
1321 ast_log(LOG_WARNING, "Suspect char %c in mixer cmd, ignoring:\n\t%s\n", s[i], s);
1326 ast_free(o->mixer_cmd);
1327 o->mixer_cmd = ast_strdup(s);
1328 ast_log(LOG_WARNING, "setting mixer %s\n", s);
1332 * store the callerid components
1334 static void store_callerid(struct chan_oss_pvt *o, const char *s)
1336 ast_callerid_split(s, o->cid_name, sizeof(o->cid_name), o->cid_num, sizeof(o->cid_num));
1339 static void store_config_core(struct chan_oss_pvt *o, const char *var, const char *value)
1341 CV_START(var, value);
1343 /* handle jb conf */
1344 if (!ast_jb_read_conf(&global_jbconf, var, value))
1347 if (!console_video_config(&o->env, var, value))
1348 return; /* matched there */
1349 CV_BOOL("autoanswer", o->autoanswer);
1350 CV_BOOL("autohangup", o->autohangup);
1351 CV_BOOL("overridecontext", o->overridecontext);
1352 CV_STR("device", o->device);
1353 CV_UINT("frags", o->frags);
1354 CV_UINT("debug", oss_debug);
1355 CV_UINT("queuesize", o->queuesize);
1356 CV_STR("context", o->ctx);
1357 CV_STR("language", o->language);
1358 CV_STR("mohinterpret", o->mohinterpret);
1359 CV_STR("extension", o->ext);
1360 CV_F("mixer", store_mixer(o, value));
1361 CV_F("callerid", store_callerid(o, value)) ;
1362 CV_F("boost", store_boost(o, value));
1368 * grab fields from the config file, init the descriptor and open the device.
1370 static struct chan_oss_pvt *store_config(struct ast_config *cfg, char *ctg)
1372 struct ast_variable *v;
1373 struct chan_oss_pvt *o;
1379 if (!(o = ast_calloc(1, sizeof(*o))))
1382 /* "general" is also the default thing */
1383 if (strcmp(ctg, "general") == 0) {
1384 o->name = ast_strdup("dsp");
1385 oss_active = o->name;
1388 o->name = ast_strdup(ctg);
1391 strcpy(o->mohinterpret, "default");
1393 o->lastopen = ast_tvnow(); /* don't leave it 0 or tvdiff may wrap */
1394 /* fill other fields from configuration */
1395 for (v = ast_variable_browse(cfg, ctg); v; v = v->next) {
1396 store_config_core(o, v->name, v->value);
1398 if (ast_strlen_zero(o->device))
1399 ast_copy_string(o->device, DEV_DSP, sizeof(o->device));
1403 if (ast_asprintf(&cmd, "mixer %s", o->mixer_cmd) >= 0) {
1404 ast_log(LOG_WARNING, "running [%s]\n", cmd);
1405 if (system(cmd) < 0) {
1406 ast_log(LOG_WARNING, "system() failed: %s\n", strerror(errno));
1412 /* if the config file requested to start the GUI, do it */
1413 if (get_gui_startup(o->env))
1414 console_video_start(o->env, NULL);
1416 if (o == &oss_default) /* we are done with the default */
1421 if (setformat(o, O_RDWR) < 0) { /* open device */
1422 ast_verb(1, "Device %s not detected\n", ctg);
1423 ast_verb(1, "Turn off OSS support by adding " "'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
1426 if (o->duplex != M_FULL)
1427 ast_log(LOG_WARNING, "XXX I don't work right with non " "full-duplex sound cards XXX\n");
1428 #endif /* TRYOPEN */
1430 /* link into list of devices */
1431 if (o != &oss_default) {
1432 o->next = oss_default.next;
1433 oss_default.next = o;
1439 if (o != &oss_default)
1446 * \brief Load the module
1448 * Module loading including tests for configuration or dependencies.
1449 * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
1450 * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
1451 * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
1452 * configuration file or other non-critical problem return
1453 * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
1455 static int load_module(void)
1457 struct ast_config *cfg = NULL;
1459 struct ast_flags config_flags = { 0 };
1460 struct ast_format tmpfmt;
1462 /* Copy the default jb config over global_jbconf */
1463 memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
1465 /* load config file */
1466 if (!(cfg = ast_config_load(config, config_flags))) {
1467 ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
1468 return AST_MODULE_LOAD_DECLINE;
1469 } else if (cfg == CONFIG_STATUS_FILEINVALID) {
1470 ast_log(LOG_ERROR, "Config file %s is in an invalid format. Aborting.\n", config);
1471 return AST_MODULE_LOAD_DECLINE;
1475 store_config(cfg, ctg);
1476 } while ( (ctg = ast_category_browse(cfg, ctg)) != NULL);
1478 ast_config_destroy(cfg);
1480 if (find_desc(oss_active) == NULL) {
1481 ast_log(LOG_NOTICE, "Device %s not found\n", oss_active);
1482 /* XXX we could default to 'dsp' perhaps ? */
1483 /* XXX should cleanup allocated memory etc. */
1484 return AST_MODULE_LOAD_FAILURE;
1487 if (!(oss_tech.capabilities = ast_format_cap_alloc())) {
1488 return AST_MODULE_LOAD_FAILURE;
1490 ast_format_cap_add(oss_tech.capabilities, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0));
1492 /* TODO XXX CONSOLE VIDEO IS DISABLE UNTIL IT HAS A MAINTAINER
1493 * add console_video_formats to oss_tech.capabilities once this occurs. */
1495 if (ast_channel_register(&oss_tech)) {
1496 ast_log(LOG_ERROR, "Unable to register channel type 'OSS'\n");
1497 return AST_MODULE_LOAD_DECLINE;
1500 ast_cli_register_multiple(cli_oss, ARRAY_LEN(cli_oss));
1502 return AST_MODULE_LOAD_SUCCESS;
1506 static int unload_module(void)
1508 struct chan_oss_pvt *o, *next;
1510 ast_channel_unregister(&oss_tech);
1511 ast_cli_unregister_multiple(cli_oss, ARRAY_LEN(cli_oss));
1513 o = oss_default.next;
1517 ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD);
1525 oss_tech.capabilities = ast_format_cap_destroy(oss_tech.capabilities);
1529 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "OSS Console Channel Driver");