Fri Jan 29 14:25:13 2010

Asterisk developer's documentation


chan_oss.c

Go to the documentation of this file.
00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (C) 1999 - 2005, Digium, Inc.
00005  *
00006  * Mark Spencer <markster@digium.com>
00007  *
00008  * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25
00009  * note-this code best seen with ts=8 (8-spaces tabs) in the editor
00010  *
00011  * See http://www.asterisk.org for more information about
00012  * the Asterisk project. Please do not directly contact
00013  * any of the maintainers of this project for assistance;
00014  * the project provides a web site, mailing lists and IRC
00015  * channels for your use.
00016  *
00017  * This program is free software, distributed under the terms of
00018  * the GNU General Public License Version 2. See the LICENSE file
00019  * at the top of the source tree.
00020  */
00021 
00022 /*! \file
00023  *
00024  * \brief Channel driver for OSS sound cards
00025  *
00026  * \author Mark Spencer <markster@digium.com>
00027  * \author Luigi Rizzo
00028  *
00029  * \par See also
00030  * \arg \ref Config_oss
00031  *
00032  * \ingroup channel_drivers
00033  */
00034 
00035 /*** MODULEINFO
00036    <depend>ossaudio</depend>
00037  ***/
00038 
00039 #include "asterisk.h"
00040 
00041 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 211528 $")
00042 
00043 #include <stdio.h>
00044 #include <ctype.h>
00045 #include <math.h>
00046 #include <string.h>
00047 #include <unistd.h>
00048 #include <sys/ioctl.h>
00049 #include <fcntl.h>
00050 #include <sys/time.h>
00051 #include <stdlib.h>
00052 #include <errno.h>
00053 
00054 #ifdef __linux
00055 #include <linux/soundcard.h>
00056 #elif defined(__FreeBSD__)
00057 #include <sys/soundcard.h>
00058 #else
00059 #include <soundcard.h>
00060 #endif
00061 
00062 #include "asterisk/lock.h"
00063 #include "asterisk/frame.h"
00064 #include "asterisk/logger.h"
00065 #include "asterisk/callerid.h"
00066 #include "asterisk/channel.h"
00067 #include "asterisk/module.h"
00068 #include "asterisk/options.h"
00069 #include "asterisk/pbx.h"
00070 #include "asterisk/config.h"
00071 #include "asterisk/cli.h"
00072 #include "asterisk/utils.h"
00073 #include "asterisk/causes.h"
00074 #include "asterisk/endian.h"
00075 #include "asterisk/stringfields.h"
00076 #include "asterisk/abstract_jb.h"
00077 #include "asterisk/musiconhold.h"
00078 
00079 /* ringtones we use */
00080 #include "busy_tone.h"
00081 #include "ring_tone.h"
00082 #include "ring10.h"
00083 #include "answer.h"
00084 
00085 /*! Global jitterbuffer configuration - by default, jb is disabled */
00086 static struct ast_jb_conf default_jbconf =
00087 {
00088    .flags = 0,
00089    .max_size = -1,
00090    .resync_threshold = -1,
00091    .impl = "",
00092 };
00093 static struct ast_jb_conf global_jbconf;
00094 
00095 /*
00096  * Basic mode of operation:
00097  *
00098  * we have one keyboard (which receives commands from the keyboard)
00099  * and multiple headset's connected to audio cards.
00100  * Cards/Headsets are named as the sections of oss.conf.
00101  * The section called [general] contains the default parameters.
00102  *
00103  * At any time, the keyboard is attached to one card, and you
00104  * can switch among them using the command 'console foo'
00105  * where 'foo' is the name of the card you want.
00106  *
00107  * oss.conf parameters are
00108 START_CONFIG
00109 
00110 [general]
00111     ; General config options, with default values shown.
00112     ; You should use one section per device, with [general] being used
00113     ; for the first device and also as a template for other devices.
00114     ;
00115     ; All but 'debug' can go also in the device-specific sections.
00116     ;
00117     ; debug = 0x0    ; misc debug flags, default is 0
00118 
00119     ; Set the device to use for I/O
00120     ; device = /dev/dsp
00121 
00122     ; Optional mixer command to run upon startup (e.g. to set
00123     ; volume levels, mutes, etc.
00124     ; mixer =
00125 
00126     ; Software mic volume booster (or attenuator), useful for sound
00127     ; cards or microphones with poor sensitivity. The volume level
00128     ; is in dB, ranging from -20.0 to +20.0
00129     ; boost = n         ; mic volume boost in dB
00130 
00131     ; Set the callerid for outgoing calls
00132     ; callerid = John Doe <555-1234>
00133 
00134     ; autoanswer = no      ; no autoanswer on call
00135     ; autohangup = yes     ; hangup when other party closes
00136     ; extension = s     ; default extension to call
00137     ; context = default    ; default context for outgoing calls
00138     ; language = ""     ; default language
00139 
00140     ; Default Music on Hold class to use when this channel is placed on hold in
00141     ; the case that the music class is not set on the channel with
00142     ; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
00143     ; putting this one on hold did not suggest a class to use.
00144     ;
00145     ; mohinterpret=default
00146 
00147     ; If you set overridecontext to 'yes', then the whole dial string
00148     ; will be interpreted as an extension, which is extremely useful
00149     ; to dial SIP, IAX and other extensions which use the '@' character.
00150     ; The default is 'no' just for backward compatibility, but the
00151     ; suggestion is to change it.
00152     ; overridecontext = no ; if 'no', the last @ will start the context
00153             ; if 'yes' the whole string is an extension.
00154 
00155     ; low level device parameters in case you have problems with the
00156     ; device driver on your operating system. You should not touch these
00157     ; unless you know what you are doing.
00158     ; queuesize = 10    ; frames in device driver
00159     ; frags = 8         ; argument to SETFRAGMENT
00160 
00161     ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
00162     ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of an
00163                                   ; OSS channel. Defaults to "no". An enabled jitterbuffer will
00164                                   ; be used only if the sending side can create and the receiving
00165                                   ; side can not accept jitter. The OSS channel can't accept jitter,
00166                                   ; thus an enabled jitterbuffer on the receive OSS side will always
00167                                   ; be used if the sending side can create jitter.
00168 
00169     ; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
00170 
00171     ; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
00172                                   ; resynchronized. Useful to improve the quality of the voice, with
00173                                   ; big jumps in/broken timestamps, usualy sent from exotic devices
00174                                   ; and programs. Defaults to 1000.
00175 
00176     ; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of an OSS
00177                                   ; channel. Two implementations are currenlty available - "fixed"
00178                                   ; (with size always equals to jbmax-size) and "adaptive" (with
00179                                   ; variable size, actually the new jb of IAX2). Defaults to fixed.
00180 
00181     ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
00182     ;-----------------------------------------------------------------------------------
00183 
00184 [card1]
00185     ; device = /dev/dsp1   ; alternate device
00186 
00187 END_CONFIG
00188 
00189 .. and so on for the other cards.
00190 
00191  */
00192 
00193 /*
00194  * Helper macros to parse config arguments. They will go in a common
00195  * header file if their usage is globally accepted. In the meantime,
00196  * we define them here. Typical usage is as below.
00197  * Remember to open a block right before M_START (as it declares
00198  * some variables) and use the M_* macros WITHOUT A SEMICOLON:
00199  *
00200  * {
00201  *    M_START(v->name, v->value) 
00202  *
00203  *    M_BOOL("dothis", x->flag1)
00204  *    M_STR("name", x->somestring)
00205  *    M_F("bar", some_c_code)
00206  *    M_END(some_final_statement)
00207  *    ... other code in the block
00208  * }
00209  *
00210  * XXX NOTE these macros should NOT be replicated in other parts of asterisk. 
00211  * Likely we will come up with a better way of doing config file parsing.
00212  */
00213 #define M_START(var, val) \
00214         char *__s = var; char *__val = val;
00215 #define M_END(x)   x;
00216 #define M_F(tag, f)        if (!strcasecmp((__s), tag)) { f; } else
00217 #define M_BOOL(tag, dst)   M_F(tag, (dst) = ast_true(__val) )
00218 #define M_UINT(tag, dst)   M_F(tag, (dst) = strtoul(__val, NULL, 0) )
00219 #define M_STR(tag, dst)    M_F(tag, ast_copy_string(dst, __val, sizeof(dst)))
00220 
00221 /*
00222  * The following parameters are used in the driver:
00223  *
00224  *  FRAME_SIZE the size of an audio frame, in samples.
00225  *    160 is used almost universally, so you should not change it.
00226  *
00227  *  FRAGS   the argument for the SETFRAGMENT ioctl.
00228  *    Overridden by the 'frags' parameter in oss.conf
00229  *
00230  *    Bits 0-7 are the base-2 log of the device's block size,
00231  *    bits 16-31 are the number of blocks in the driver's queue.
00232  *    There are a lot of differences in the way this parameter
00233  *    is supported by different drivers, so you may need to
00234  *    experiment a bit with the value.
00235  *    A good default for linux is 30 blocks of 64 bytes, which
00236  *    results in 6 frames of 320 bytes (160 samples).
00237  *    FreeBSD works decently with blocks of 256 or 512 bytes,
00238  *    leaving the number unspecified.
00239  *    Note that this only refers to the device buffer size,
00240  *    this module will then try to keep the lenght of audio
00241  *    buffered within small constraints.
00242  *
00243  *  QUEUE_SIZE The max number of blocks actually allowed in the device
00244  *    driver's buffer, irrespective of the available number.
00245  *    Overridden by the 'queuesize' parameter in oss.conf
00246  *
00247  *    Should be >=2, and at most as large as the hw queue above
00248  *    (otherwise it will never be full).
00249  */
00250 
00251 #define FRAME_SIZE   160
00252 #define  QUEUE_SIZE  10
00253 
00254 #if defined(__FreeBSD__)
00255 #define  FRAGS 0x8
00256 #else
00257 #define  FRAGS ( ( (6 * 5) << 16 ) | 0x6 )
00258 #endif
00259 
00260 /*
00261  * XXX text message sizes are probably 256 chars, but i am
00262  * not sure if there is a suitable definition anywhere.
00263  */
00264 #define TEXT_SIZE 256
00265 
00266 #if 0
00267 #define  TRYOPEN  1           /* try to open on startup */
00268 #endif
00269 #define  O_CLOSE  0x444       /* special 'close' mode for device */
00270 /* Which device to use */
00271 #if defined( __OpenBSD__ ) || defined( __NetBSD__ )
00272 #define DEV_DSP "/dev/audio"
00273 #else
00274 #define DEV_DSP "/dev/dsp"
00275 #endif
00276 
00277 #ifndef MIN
00278 #define MIN(a,b) ((a) < (b) ? (a) : (b))
00279 #endif
00280 #ifndef MAX
00281 #define MAX(a,b) ((a) > (b) ? (a) : (b))
00282 #endif
00283 
00284 static char *config = "oss.conf";   /* default config file */
00285 
00286 static int oss_debug;
00287 
00288 /*
00289  * Each sound is made of 'datalen' samples of sound, repeated as needed to
00290  * generate 'samplen' samples of data, then followed by 'silencelen' samples
00291  * of silence. The loop is repeated if 'repeat' is set.
00292  */
00293 struct sound {
00294    int ind;
00295    char *desc;
00296    short *data;
00297    int datalen;
00298    int samplen;
00299    int silencelen;
00300    int repeat;
00301 };
00302 
00303 static struct sound sounds[] = {
00304    { AST_CONTROL_RINGING, "RINGING", ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
00305    { AST_CONTROL_BUSY, "BUSY", busy, sizeof(busy)/2, 4000, 4000, 1 },
00306    { AST_CONTROL_CONGESTION, "CONGESTION", busy, sizeof(busy)/2, 2000, 2000, 1 },
00307    { AST_CONTROL_RING, "RING10", ring10, sizeof(ring10)/2, 16000, 32000, 1 },
00308    { AST_CONTROL_ANSWER, "ANSWER", answer, sizeof(answer)/2, 2200, 0, 0 },
00309    { -1, NULL, 0, 0, 0, 0 },  /* end marker */
00310 };
00311 
00312 
00313 /*
00314  * descriptor for one of our channels.
00315  * There is one used for 'default' values (from the [general] entry in
00316  * the configuration file), and then one instance for each device
00317  * (the default is cloned from [general], others are only created
00318  * if the relevant section exists).
00319  */
00320 struct chan_oss_pvt {
00321    struct chan_oss_pvt *next;
00322 
00323    char *name;
00324    /*
00325     * cursound indicates which in struct sound we play. -1 means nothing,
00326     * any other value is a valid sound, in which case sampsent indicates
00327     * the next sample to send in [0..samplen + silencelen]
00328     * nosound is set to disable the audio data from the channel
00329     * (so we can play the tones etc.).
00330     */
00331    int sndcmd[2];          /* Sound command pipe */
00332    int cursound;           /* index of sound to send */
00333    int sampsent;           /* # of sound samples sent  */
00334    int nosound;            /* set to block audio from the PBX */
00335 
00336    int total_blocks;       /* total blocks in the output device */
00337    int sounddev;
00338    enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
00339    int autoanswer;             /*!< Boolean: whether to answer the immediately upon calling */
00340    int autohangup;             /*!< Boolean: whether to hangup the call when the remote end hangs up */
00341    int hookstate;              /*!< Boolean: 1 if offhook; 0 if onhook */
00342    char *mixer_cmd;        /* initial command to issue to the mixer */
00343    unsigned int queuesize;    /* max fragments in queue */
00344    unsigned int frags;        /* parameter for SETFRAGMENT */
00345 
00346    int warned;             /* various flags used for warnings */
00347 #define WARN_used_blocks   1
00348 #define WARN_speed      2
00349 #define WARN_frag    4
00350    int w_errors;           /* overfull in the write path */
00351    struct timeval lastopen;
00352 
00353    int overridecontext;
00354    int mute;
00355 
00356    /* boost support. BOOST_SCALE * 10 ^(BOOST_MAX/20) must
00357     * be representable in 16 bits to avoid overflows.
00358     */
00359 #define  BOOST_SCALE (1<<9)
00360 #define  BOOST_MAX   40       /* slightly less than 7 bits */
00361    int boost;              /* input boost, scaled by BOOST_SCALE */
00362    char device[64];        /* device to open */
00363 
00364    pthread_t sthread;
00365 
00366    struct ast_channel *owner;
00367    char ext[AST_MAX_EXTENSION];
00368    char ctx[AST_MAX_CONTEXT];
00369    char language[MAX_LANGUAGE];
00370    char cid_name[256];         /*!< Initial CallerID name */
00371    char cid_num[256];          /*!< Initial CallerID number  */
00372    char mohinterpret[MAX_MUSICCLASS];
00373 
00374    /* buffers used in oss_write */
00375    char oss_write_buf[FRAME_SIZE * 2];
00376    int oss_write_dst;
00377    /* buffers used in oss_read - AST_FRIENDLY_OFFSET space for headers
00378     * plus enough room for a full frame
00379     */
00380    char oss_read_buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
00381    int readpos;            /* read position above */
00382    struct ast_frame read_f;   /* returned by oss_read */
00383 };
00384 
00385 static struct chan_oss_pvt oss_default = {
00386    .cursound = -1,
00387    .sounddev = -1,
00388    .duplex = M_UNSET,         /* XXX check this */
00389    .autoanswer = 1,
00390    .autohangup = 1,
00391    .queuesize = QUEUE_SIZE,
00392    .frags = FRAGS,
00393    .ext = "s",
00394    .ctx = "default",
00395    .readpos = AST_FRIENDLY_OFFSET,  /* start here on reads */
00396    .lastopen = { 0, 0 },
00397    .boost = BOOST_SCALE,
00398 };
00399 
00400 static char *oss_active;    /* the active device */
00401 
00402 static int setformat(struct chan_oss_pvt *o, int mode);
00403 
00404 static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause);
00405 static int oss_digit_begin(struct ast_channel *c, char digit);
00406 static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration);
00407 static int oss_text(struct ast_channel *c, const char *text);
00408 static int oss_hangup(struct ast_channel *c);
00409 static int oss_answer(struct ast_channel *c);
00410 static struct ast_frame *oss_read(struct ast_channel *chan);
00411 static int oss_call(struct ast_channel *c, char *dest, int timeout);
00412 static int oss_write(struct ast_channel *chan, struct ast_frame *f);
00413 static int oss_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
00414 static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
00415 static char tdesc[] = "OSS Console Channel Driver";
00416 
00417 static const struct ast_channel_tech oss_tech = {
00418    .type = "Console",
00419    .description = tdesc,
00420    .capabilities = AST_FORMAT_SLINEAR,
00421    .requester = oss_request,
00422    .send_digit_begin = oss_digit_begin,
00423    .send_digit_end = oss_digit_end,
00424    .send_text = oss_text,
00425    .hangup = oss_hangup,
00426    .answer = oss_answer,
00427    .read = oss_read,
00428    .call = oss_call,
00429    .write = oss_write,
00430    .indicate = oss_indicate,
00431    .fixup = oss_fixup,
00432 };
00433 
00434 /*
00435  * returns a pointer to the descriptor with the given name
00436  */
00437 static struct chan_oss_pvt *find_desc(char *dev)
00438 {
00439    struct chan_oss_pvt *o = NULL;
00440 
00441    if (!dev)
00442       ast_log(LOG_WARNING, "null dev\n");
00443 
00444    for (o = oss_default.next; o && o->name && dev && strcmp(o->name, dev) != 0; o = o->next);
00445 
00446    if (!o)
00447       ast_log(LOG_WARNING, "could not find <%s>\n", dev ? dev : "--no-device--");
00448 
00449    return o;
00450 }
00451 
00452 /*
00453  * split a string in extension-context, returns pointers to malloc'ed
00454  * strings.
00455  * If we do not have 'overridecontext' then the last @ is considered as
00456  * a context separator, and the context is overridden.
00457  * This is usually not very necessary as you can play with the dialplan,
00458  * and it is nice not to need it because you have '@' in SIP addresses.
00459  * Return value is the buffer address.
00460  */
00461 static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
00462 {
00463    struct chan_oss_pvt *o = find_desc(oss_active);
00464 
00465    if (ext == NULL || ctx == NULL)
00466       return NULL;         /* error */
00467 
00468    *ext = *ctx = NULL;
00469 
00470    if (src && *src != '\0')
00471       *ext = ast_strdup(src);
00472 
00473    if (*ext == NULL)
00474       return NULL;
00475 
00476    if (!o->overridecontext) {
00477       /* parse from the right */
00478       *ctx = strrchr(*ext, '@');
00479       if (*ctx)
00480          *(*ctx)++ = '\0';
00481    }
00482 
00483    return *ext;
00484 }
00485 
00486 /*
00487  * Returns the number of blocks used in the audio output channel
00488  */
00489 static int used_blocks(struct chan_oss_pvt *o)
00490 {
00491    struct audio_buf_info info;
00492 
00493    if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) {
00494       if (!(o->warned & WARN_used_blocks)) {
00495          ast_log(LOG_WARNING, "Error reading output space\n");
00496          o->warned |= WARN_used_blocks;
00497       }
00498       return 1;
00499    }
00500 
00501    if (o->total_blocks == 0) {
00502       if (0)               /* debugging */
00503          ast_log(LOG_WARNING, "fragtotal %d size %d avail %d\n", info.fragstotal, info.fragsize, info.fragments);
00504       o->total_blocks = info.fragments;
00505    }
00506 
00507    return o->total_blocks - info.fragments;
00508 }
00509 
00510 /* Write an exactly FRAME_SIZE sized frame */
00511 static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
00512 {
00513    int res;
00514 
00515    if (o->sounddev < 0)
00516       setformat(o, O_RDWR);
00517    if (o->sounddev < 0)
00518       return 0;            /* not fatal */
00519    /*
00520     * Nothing complex to manage the audio device queue.
00521     * If the buffer is full just drop the extra, otherwise write.
00522     * XXX in some cases it might be useful to write anyways after
00523     * a number of failures, to restart the output chain.
00524     */
00525    res = used_blocks(o);
00526    if (res > o->queuesize) {  /* no room to write a block */
00527       if (o->w_errors++ == 0 && (oss_debug & 0x4))
00528          ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, o->w_errors);
00529       return 0;
00530    }
00531    o->w_errors = 0;
00532    return write(o->sounddev, ((void *) data), FRAME_SIZE * 2);
00533 }
00534 
00535 /*
00536  * Handler for 'sound writable' events from the sound thread.
00537  * Builds a frame from the high level description of the sounds,
00538  * and passes it to the audio device.
00539  * The actual sound is made of 1 or more sequences of sound samples
00540  * (s->datalen, repeated to make s->samplen samples) followed by
00541  * s->silencelen samples of silence. The position in the sequence is stored
00542  * in o->sampsent, which goes between 0 .. s->samplen+s->silencelen.
00543  * In case we fail to write a frame, don't update o->sampsent.
00544  */
00545 static void send_sound(struct chan_oss_pvt *o)
00546 {
00547    short myframe[FRAME_SIZE];
00548    int ofs, l, start;
00549    int l_sampsent = o->sampsent;
00550    struct sound *s;
00551 
00552    if (o->cursound < 0)    /* no sound to send */
00553       return;
00554 
00555    s = &sounds[o->cursound];
00556 
00557    for (ofs = 0; ofs < FRAME_SIZE; ofs += l) {
00558       l = s->samplen - l_sampsent;  /* # of available samples */
00559       if (l > 0) {
00560          start = l_sampsent % s->datalen; /* source offset */
00561          if (l > FRAME_SIZE - ofs)  /* don't overflow the frame */
00562             l = FRAME_SIZE - ofs;
00563          if (l > s->datalen - start)   /* don't overflow the source */
00564             l = s->datalen - start;
00565          bcopy(s->data + start, myframe + ofs, l * 2);
00566          if (0)
00567             ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n", l_sampsent, l, s->samplen, ofs);
00568          l_sampsent += l;
00569       } else {          /* end of samples, maybe some silence */
00570          static const short silence[FRAME_SIZE] = { 0, };
00571 
00572          l += s->silencelen;
00573          if (l > 0) {
00574             if (l > FRAME_SIZE - ofs)
00575                l = FRAME_SIZE - ofs;
00576             bcopy(silence, myframe + ofs, l * 2);
00577             l_sampsent += l;
00578          } else {       /* silence is over, restart sound if loop */
00579             if (s->repeat == 0) {   /* last block */
00580                o->cursound = -1;
00581                o->nosound = 0;   /* allow audio data */
00582                if (ofs < FRAME_SIZE)   /* pad with silence */
00583                   bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs) * 2);
00584             }
00585             l_sampsent = 0;
00586          }
00587       }
00588    }
00589    l = soundcard_writeframe(o, myframe);
00590    if (l > 0)
00591       o->sampsent = l_sampsent;  /* update status */
00592 }
00593 
00594 static void *sound_thread(void *arg)
00595 {
00596    char ign[4096];
00597    struct chan_oss_pvt *o = (struct chan_oss_pvt *) arg;
00598 
00599    /*
00600     * Just in case, kick the driver by trying to read from it.
00601     * Ignore errors - this read is almost guaranteed to fail.
00602     */
00603    if (read(o->sounddev, ign, sizeof(ign)) < 0) {
00604    }
00605    for (;;) {
00606       fd_set rfds, wfds;
00607       int maxfd, res;
00608 
00609       FD_ZERO(&rfds);
00610       FD_ZERO(&wfds);
00611       FD_SET(o->sndcmd[0], &rfds);
00612       maxfd = o->sndcmd[0];   /* pipe from the main process */
00613       if (o->cursound > -1 && o->sounddev < 0)
00614          setformat(o, O_RDWR);   /* need the channel, try to reopen */
00615       else if (o->cursound == -1 && o->owner == NULL)
00616          setformat(o, O_CLOSE);  /* can close */
00617       if (o->sounddev > -1) {
00618          if (!o->owner) {  /* no one owns the audio, so we must drain it */
00619             FD_SET(o->sounddev, &rfds);
00620             maxfd = MAX(o->sounddev, maxfd);
00621          }
00622          if (o->cursound > -1) {
00623             FD_SET(o->sounddev, &wfds);
00624             maxfd = MAX(o->sounddev, maxfd);
00625          }
00626       }
00627       /* ast_select emulates linux behaviour in terms of timeout handling */
00628       res = ast_select(maxfd + 1, &rfds, &wfds, NULL, NULL);
00629       if (res < 1) {
00630          ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
00631          sleep(1);
00632          continue;
00633       }
00634       if (FD_ISSET(o->sndcmd[0], &rfds)) {
00635          /* read which sound to play from the pipe */
00636          int i, what = -1;
00637 
00638          if (read(o->sndcmd[0], &what, sizeof(what)) != sizeof(what)) {
00639             ast_log(LOG_WARNING, "read() failed: %s\n", strerror(errno));
00640             continue;
00641          }
00642          for (i = 0; sounds[i].ind != -1; i++) {
00643             if (sounds[i].ind == what) {
00644                o->cursound = i;
00645                o->sampsent = 0;
00646                o->nosound = 1;   /* block audio from pbx */
00647                break;
00648             }
00649          }
00650          if (sounds[i].ind == -1)
00651             ast_log(LOG_WARNING, "invalid sound index: %d\n", what);
00652       }
00653       if (o->sounddev > -1) {
00654          if (FD_ISSET(o->sounddev, &rfds))   /* read and ignore errors */
00655             if (read(o->sounddev, ign, sizeof(ign)) < 0) {
00656             }
00657          if (FD_ISSET(o->sounddev, &wfds))
00658             send_sound(o);
00659       }
00660    }
00661    return NULL;            /* Never reached */
00662 }
00663 
00664 /*
00665  * reset and close the device if opened,
00666  * then open and initialize it in the desired mode,
00667  * trigger reads and writes so we can start using it.
00668  */
00669 static int setformat(struct chan_oss_pvt *o, int mode)
00670 {
00671    int fmt, desired, res, fd;
00672 
00673    if (o->sounddev >= 0) {
00674       ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
00675       close(o->sounddev);
00676       o->duplex = M_UNSET;
00677       o->sounddev = -1;
00678    }
00679    if (mode == O_CLOSE)    /* we are done */
00680       return 0;
00681    if (ast_tvdiff_ms(ast_tvnow(), o->lastopen) < 1000)
00682       return -1;           /* don't open too often */
00683    o->lastopen = ast_tvnow();
00684    fd = o->sounddev = open(o->device, mode | O_NONBLOCK);
00685    if (fd < 0) {
00686       ast_log(LOG_WARNING, "Unable to re-open DSP device %s: %s\n", o->device, strerror(errno));
00687       return -1;
00688    }
00689    if (o->owner)
00690       o->owner->fds[0] = fd;
00691 
00692 #if __BYTE_ORDER == __LITTLE_ENDIAN
00693    fmt = AFMT_S16_LE;
00694 #else
00695    fmt = AFMT_S16_BE;
00696 #endif
00697    res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
00698    if (res < 0) {
00699       ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
00700       return -1;
00701    }
00702    switch (mode) {
00703       case O_RDWR:
00704          res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
00705          /* Check to see if duplex set (FreeBSD Bug) */
00706          res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
00707          if (res == 0 && (fmt & DSP_CAP_DUPLEX)) {
00708             if (option_verbose > 1)
00709                ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
00710             o->duplex = M_FULL;
00711          };
00712          break;
00713       case O_WRONLY:
00714          o->duplex = M_WRITE;
00715          break;
00716       case O_RDONLY:
00717          o->duplex = M_READ;
00718          break;
00719    }
00720 
00721    fmt = 0;
00722    res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
00723    if (res < 0) {
00724       ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
00725       return -1;
00726    }
00727    fmt = desired = DEFAULT_SAMPLE_RATE;   /* 8000 Hz desired */
00728    res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
00729 
00730    if (res < 0) {
00731       ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
00732       return -1;
00733    }
00734    if (fmt != desired) {
00735       if (!(o->warned & WARN_speed)) {
00736          ast_log(LOG_WARNING,
00737              "Requested %d Hz, got %d Hz -- sound may be choppy\n",
00738              desired, fmt);
00739          o->warned |= WARN_speed;
00740       }
00741    }
00742    /*
00743     * on Freebsd, SETFRAGMENT does not work very well on some cards.
00744     * Default to use 256 bytes, let the user override
00745     */
00746    if (o->frags) {
00747       fmt = o->frags;
00748       res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
00749       if (res < 0) {
00750          if (!(o->warned & WARN_frag)) {
00751             ast_log(LOG_WARNING,
00752                "Unable to set fragment size -- sound may be choppy\n");
00753             o->warned |= WARN_frag;
00754          }
00755       }
00756    }
00757    /* on some cards, we need SNDCTL_DSP_SETTRIGGER to start outputting */
00758    res = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT;
00759    res = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &res);
00760    /* it may fail if we are in half duplex, never mind */
00761    return 0;
00762 }
00763 
00764 /*
00765  * some of the standard methods supported by channels.
00766  */
00767 static int oss_digit_begin(struct ast_channel *c, char digit)
00768 {
00769    return 0;
00770 }
00771 
00772 static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration)
00773 {
00774    /* no better use for received digits than print them */
00775    ast_verbose(" << Console Received digit %c of duration %u ms >> \n", 
00776       digit, duration);
00777    return 0;
00778 }
00779 
00780 static int oss_text(struct ast_channel *c, const char *text)
00781 {
00782    /* print received messages */
00783    ast_verbose(" << Console Received text %s >> \n", text);
00784    return 0;
00785 }
00786 
00787 /* Play ringtone 'x' on device 'o' */
00788 static void ring(struct chan_oss_pvt *o, int x)
00789 {
00790    if (write(o->sndcmd[1], &x, sizeof(x)) < 0) {
00791       ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
00792    }
00793 }
00794 
00795 
00796 /*
00797  * handler for incoming calls. Either autoanswer, or start ringing
00798  */
00799 static int oss_call(struct ast_channel *c, char *dest, int timeout)
00800 {
00801    struct chan_oss_pvt *o = c->tech_pvt;
00802    struct ast_frame f = { 0, };
00803 
00804    ast_verbose(" << Call to device '%s' dnid '%s' rdnis '%s' on console from '%s' <%s> >>\n", dest, c->cid.cid_dnid, c->cid.cid_rdnis, c->cid.cid_name, c->cid.cid_num);
00805    if (o->autoanswer) {
00806       ast_verbose(" << Auto-answered >> \n");
00807       f.frametype = AST_FRAME_CONTROL;
00808       f.subclass = AST_CONTROL_ANSWER;
00809       ast_queue_frame(c, &f);
00810       o->hookstate = 1;
00811    } else {
00812       ast_verbose("<< Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
00813       f.frametype = AST_FRAME_CONTROL;
00814       f.subclass = AST_CONTROL_RINGING;
00815       ast_queue_frame(c, &f);
00816       ring(o, AST_CONTROL_RING);
00817    }
00818    return 0;
00819 }
00820 
00821 /*
00822  * remote side answered the phone
00823  */
00824 static int oss_answer(struct ast_channel *c)
00825 {
00826    struct chan_oss_pvt *o = c->tech_pvt;
00827 
00828    ast_verbose(" << Console call has been answered >> \n");
00829 #if 0
00830    /* play an answer tone (XXX do we really need it ?) */
00831    ring(o, AST_CONTROL_ANSWER);
00832 #endif
00833    ast_setstate(c, AST_STATE_UP);
00834    o->cursound = -1;
00835    o->nosound = 0;
00836    o->hookstate = 1;
00837    return 0;
00838 }
00839 
00840 static int oss_hangup(struct ast_channel *c)
00841 {
00842    struct chan_oss_pvt *o = c->tech_pvt;
00843 
00844    o->cursound = -1;
00845    o->nosound = 0;
00846    c->tech_pvt = NULL;
00847    o->owner = NULL;
00848    ast_verbose(" << Hangup on console >> \n");
00849    ast_module_unref(ast_module_info->self);
00850    if (o->hookstate) {
00851       if (o->autoanswer || o->autohangup) {
00852          /* Assume auto-hangup too */
00853          o->hookstate = 0;
00854          setformat(o, O_CLOSE);
00855       } else {
00856          /* Make congestion noise */
00857          ring(o, AST_CONTROL_CONGESTION);
00858       }
00859    }
00860    return 0;
00861 }
00862 
00863 /* used for data coming from the network */
00864 static int oss_write(struct ast_channel *c, struct ast_frame *f)
00865 {
00866    int src;
00867    struct chan_oss_pvt *o = c->tech_pvt;
00868 
00869    /* Immediately return if no sound is enabled */
00870    if (o->nosound)
00871       return 0;
00872    /* Stop any currently playing sound */
00873    o->cursound = -1;
00874    /*
00875     * we could receive a block which is not a multiple of our
00876     * FRAME_SIZE, so buffer it locally and write to the device
00877     * in FRAME_SIZE chunks.
00878     * Keep the residue stored for future use.
00879     */
00880    src = 0;             /* read position into f->data */
00881    while (src < f->datalen) {
00882       /* Compute spare room in the buffer */
00883       int l = sizeof(o->oss_write_buf) - o->oss_write_dst;
00884 
00885       if (f->datalen - src >= l) {  /* enough to fill a frame */
00886          memcpy(o->oss_write_buf + o->oss_write_dst, f->data + src, l);
00887          soundcard_writeframe(o, (short *) o->oss_write_buf);
00888          src += l;
00889          o->oss_write_dst = 0;
00890       } else {          /* copy residue */
00891          l = f->datalen - src;
00892          memcpy(o->oss_write_buf + o->oss_write_dst, f->data + src, l);
00893          src += l;         /* but really, we are done */
00894          o->oss_write_dst += l;
00895       }
00896    }
00897    return 0;
00898 }
00899 
00900 static struct ast_frame *oss_read(struct ast_channel *c)
00901 {
00902    int res;
00903    struct chan_oss_pvt *o = c->tech_pvt;
00904    struct ast_frame *f = &o->read_f;
00905 
00906    /* XXX can be simplified returning &ast_null_frame */
00907    /* prepare a NULL frame in case we don't have enough data to return */
00908    bzero(f, sizeof(struct ast_frame));
00909    f->frametype = AST_FRAME_NULL;
00910    f->src = oss_tech.type;
00911 
00912    res = read(o->sounddev, o->oss_read_buf + o->readpos, sizeof(o->oss_read_buf) - o->readpos);
00913    if (res < 0)            /* audio data not ready, return a NULL frame */
00914       return f;
00915 
00916    o->readpos += res;
00917    if (o->readpos < sizeof(o->oss_read_buf)) /* not enough samples */
00918       return f;
00919 
00920    if (o->mute)
00921       return f;
00922 
00923    o->readpos = AST_FRIENDLY_OFFSET;   /* reset read pointer for next frame */
00924    if (c->_state != AST_STATE_UP)   /* drop data if frame is not up */
00925       return f;
00926    /* ok we can build and deliver the frame to the caller */
00927    f->frametype = AST_FRAME_VOICE;
00928    f->subclass = AST_FORMAT_SLINEAR;
00929    f->samples = FRAME_SIZE;
00930    f->datalen = FRAME_SIZE * 2;
00931    f->data = o->oss_read_buf + AST_FRIENDLY_OFFSET;
00932    if (o->boost != BOOST_SCALE) {   /* scale and clip values */
00933       int i, x;
00934       int16_t *p = (int16_t *) f->data;
00935       for (i = 0; i < f->samples; i++) {
00936          x = (p[i] * o->boost) / BOOST_SCALE;
00937          if (x > 32767)
00938             x = 32767;
00939          else if (x < -32768)
00940             x = -32768;
00941          p[i] = x;
00942       }
00943    }
00944 
00945    f->offset = AST_FRIENDLY_OFFSET;
00946    return f;
00947 }
00948 
00949 static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
00950 {
00951    struct chan_oss_pvt *o = newchan->tech_pvt;
00952    o->owner = newchan;
00953    return 0;
00954 }
00955 
00956 static int oss_indicate(struct ast_channel *c, int cond, const void *data, size_t datalen)
00957 {
00958    struct chan_oss_pvt *o = c->tech_pvt;
00959    int res = -1;
00960 
00961    switch (cond) {
00962    case AST_CONTROL_BUSY:
00963    case AST_CONTROL_CONGESTION:
00964    case AST_CONTROL_RINGING:
00965          res = cond;
00966          break;
00967          
00968    case -1:
00969       o->cursound = -1;
00970       o->nosound = 0;      /* when cursound is -1 nosound must be 0 */
00971       return 0;
00972       
00973    case AST_CONTROL_VIDUPDATE:
00974       res = -1;
00975       break;
00976    case AST_CONTROL_HOLD:
00977       ast_verbose(" << Console Has Been Placed on Hold >> \n");
00978       ast_moh_start(c, data, o->mohinterpret);
00979          break;
00980    case AST_CONTROL_UNHOLD:
00981       ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
00982       ast_moh_stop(c);
00983       break;
00984    case AST_CONTROL_SRCUPDATE:
00985       break;
00986    default:
00987       ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, c->name);
00988       return -1;
00989    }
00990 
00991    if (res > -1)
00992       ring(o, res);
00993 
00994    return 0;
00995 }
00996 
00997 /*
00998  * allocate a new channel.
00999  */
01000 static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx, int state)
01001 {
01002    struct ast_channel *c;
01003 
01004    c = ast_channel_alloc(1, state, o->cid_num, o->cid_name, "", ext, ctx, 0, "Console/%s", o->device + 5);
01005    if (c == NULL)
01006       return NULL;
01007    c->tech = &oss_tech;
01008    if (o->sounddev < 0)
01009       setformat(o, O_RDWR);
01010    c->fds[0] = o->sounddev;   /* -1 if device closed, override later */
01011    c->nativeformats = AST_FORMAT_SLINEAR;
01012    c->readformat = AST_FORMAT_SLINEAR;
01013    c->writeformat = AST_FORMAT_SLINEAR;
01014    c->tech_pvt = o;
01015 
01016    if (!ast_strlen_zero(o->language))
01017       ast_string_field_set(c, language, o->language);
01018    /* Don't use ast_set_callerid() here because it will
01019     * generate a needless NewCallerID event */
01020    c->cid.cid_ani = ast_strdup(o->cid_num);
01021    if (!ast_strlen_zero(ext))
01022       c->cid.cid_dnid = ast_strdup(ext);
01023 
01024    o->owner = c;
01025    ast_module_ref(ast_module_info->self);
01026    ast_jb_configure(c, &global_jbconf);
01027    if (state != AST_STATE_DOWN) {
01028       if (ast_pbx_start(c)) {
01029          ast_log(LOG_WARNING, "Unable to start PBX on %s\n", c->name);
01030          ast_hangup(c);
01031          o->owner = c = NULL;
01032       }
01033    }
01034 
01035    return c;
01036 }
01037 
01038 static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause)
01039 {
01040    struct ast_channel *c;
01041    struct chan_oss_pvt *o = find_desc(data);
01042 
01043    ast_log(LOG_WARNING, "oss_request ty <%s> data 0x%p <%s>\n", type, data, (char *) data);
01044    if (o == NULL) {
01045       ast_log(LOG_NOTICE, "Device %s not found\n", (char *) data);
01046       /* XXX we could default to 'dsp' perhaps ? */
01047       return NULL;
01048    }
01049    if ((format & AST_FORMAT_SLINEAR) == 0) {
01050       ast_log(LOG_NOTICE, "Format 0x%x unsupported\n", format);
01051       return NULL;
01052    }
01053    if (o->owner) {
01054       ast_log(LOG_NOTICE, "Already have a call (chan %p) on the OSS channel\n", o->owner);
01055       *cause = AST_CAUSE_BUSY;
01056       return NULL;
01057    }
01058    c = oss_new(o, NULL, NULL, AST_STATE_DOWN);
01059    if (c == NULL) {
01060       ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
01061       return NULL;
01062    }
01063    return c;
01064 }
01065 
01066 static int console_autoanswer_deprecated(int fd, int argc, char *argv[])
01067 {
01068    struct chan_oss_pvt *o = find_desc(oss_active);
01069 
01070    if (argc == 1) {
01071       ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off");
01072       return RESULT_SUCCESS;
01073    }
01074    if (argc != 2)
01075       return RESULT_SHOWUSAGE;
01076    if (o == NULL) {
01077       ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n", oss_active);
01078       return RESULT_FAILURE;
01079    }
01080    if (!strcasecmp(argv[1], "on"))
01081       o->autoanswer = -1;
01082    else if (!strcasecmp(argv[1], "off"))
01083       o->autoanswer = 0;
01084    else
01085       return RESULT_SHOWUSAGE;
01086    return RESULT_SUCCESS;
01087 }
01088 
01089 static int console_autoanswer(int fd, int argc, char *argv[])
01090 {
01091    struct chan_oss_pvt *o = find_desc(oss_active);
01092 
01093    if (argc == 2) {
01094       ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off");
01095       return RESULT_SUCCESS;
01096    }
01097    if (argc != 3)
01098       return RESULT_SHOWUSAGE;
01099    if (o == NULL) {
01100       ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
01101           oss_active);
01102       return RESULT_FAILURE;
01103    }
01104    if (!strcasecmp(argv[2], "on"))
01105       o->autoanswer = -1;
01106    else if (!strcasecmp(argv[2], "off"))
01107       o->autoanswer = 0;
01108    else
01109       return RESULT_SHOWUSAGE;
01110    return RESULT_SUCCESS;
01111 }
01112 
01113 static char *autoanswer_complete_deprecated(const char *line, const char *word, int pos, int state)
01114 {
01115    static char *choices[] = { "on", "off", NULL };
01116 
01117    return (pos != 2) ? NULL : ast_cli_complete(word, choices, state);
01118 }
01119 
01120 static char *autoanswer_complete(const char *line, const char *word, int pos, int state)
01121 {
01122    static char *choices[] = { "on", "off", NULL };
01123 
01124    return (pos != 3) ? NULL : ast_cli_complete(word, choices, state);
01125 }
01126 
01127 static char autoanswer_usage[] =
01128    "Usage: console autoanswer [on|off]\n"
01129    "       Enables or disables autoanswer feature.  If used without\n"
01130    "       argument, displays the current on/off status of autoanswer.\n"
01131    "       The default value of autoanswer is in 'oss.conf'.\n";
01132 
01133 /*
01134  * answer command from the console
01135  */
01136 static int console_answer_deprecated(int fd, int argc, char *argv[])
01137 {
01138    struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
01139    struct chan_oss_pvt *o = find_desc(oss_active);
01140 
01141    if (argc != 1)
01142       return RESULT_SHOWUSAGE;
01143    if (!o->owner) {
01144       ast_cli(fd, "No one is calling us\n");
01145       return RESULT_FAILURE;
01146    }
01147    o->hookstate = 1;
01148    o->cursound = -1;
01149    o->nosound = 0;
01150    ast_queue_frame(o->owner, &f);
01151 #if 0
01152    /* XXX do we really need it ? considering we shut down immediately... */
01153    ring(o, AST_CONTROL_ANSWER);
01154 #endif
01155    return RESULT_SUCCESS;
01156 }
01157 
01158 static int console_answer(int fd, int argc, char *argv[])
01159 {
01160    struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
01161    struct chan_oss_pvt *o = find_desc(oss_active);
01162 
01163    if (argc != 2)
01164       return RESULT_SHOWUSAGE;
01165    if (!o->owner) {
01166       ast_cli(fd, "No one is calling us\n");
01167       return RESULT_FAILURE;
01168    }
01169    o->hookstate = 1;
01170    o->cursound = -1;
01171    o->nosound = 0;
01172    ast_queue_frame(o->owner, &f);
01173 #if 0
01174    /* XXX do we really need it ? considering we shut down immediately... */
01175    ring(o, AST_CONTROL_ANSWER);
01176 #endif
01177    return RESULT_SUCCESS;
01178 }
01179 
01180 static char answer_usage[] =
01181    "Usage: console answer\n"
01182    "       Answers an incoming call on the console (OSS) channel.\n";
01183 
01184 /*
01185  * concatenate all arguments into a single string. argv is NULL-terminated
01186  * so we can use it right away
01187  */
01188 static int console_sendtext_deprecated(int fd, int argc, char *argv[])
01189 {
01190    struct chan_oss_pvt *o = find_desc(oss_active);
01191    char buf[TEXT_SIZE];
01192 
01193    if (argc < 2)
01194       return RESULT_SHOWUSAGE;
01195    if (!o->owner) {
01196       ast_cli(fd, "Not in a call\n");
01197       return RESULT_FAILURE;
01198    }
01199    ast_join(buf, sizeof(buf) - 1, argv + 2);
01200    if (!ast_strlen_zero(buf)) {
01201       struct ast_frame f = { 0, };
01202       int i = strlen(buf);
01203       buf[i] = '\n';
01204       f.frametype = AST_FRAME_TEXT;
01205       f.subclass = 0;
01206       f.data = buf;
01207       f.datalen = i + 1;
01208       ast_queue_frame(o->owner, &f);
01209    }
01210    return RESULT_SUCCESS;
01211 }
01212 
01213 static int console_sendtext(int fd, int argc, char *argv[])
01214 {
01215    struct chan_oss_pvt *o = find_desc(oss_active);
01216    char buf[TEXT_SIZE];
01217 
01218    if (argc < 3)
01219       return RESULT_SHOWUSAGE;
01220    if (!o->owner) {
01221       ast_cli(fd, "Not in a call\n");
01222       return RESULT_FAILURE;
01223    }
01224    ast_join(buf, sizeof(buf) - 1, argv + 3);
01225    if (!ast_strlen_zero(buf)) {
01226       struct ast_frame f = { 0, };
01227       int i = strlen(buf);
01228       buf[i] = '\n';
01229       f.frametype = AST_FRAME_TEXT;
01230       f.subclass = 0;
01231       f.data = buf;
01232       f.datalen = i + 1;
01233       ast_queue_frame(o->owner, &f);
01234    }
01235    return RESULT_SUCCESS;
01236 }
01237 
01238 static char sendtext_usage[] =
01239    "Usage: console send text <message>\n"
01240    "       Sends a text message for display on the remote terminal.\n";
01241 
01242 static int console_hangup_deprecated(int fd, int argc, char *argv[])
01243 {
01244    struct chan_oss_pvt *o = find_desc(oss_active);
01245 
01246    if (argc != 1)
01247       return RESULT_SHOWUSAGE;
01248    o->cursound = -1;
01249    o->nosound = 0;
01250    if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
01251       ast_cli(fd, "No call to hang up\n");
01252       return RESULT_FAILURE;
01253    }
01254    o->hookstate = 0;
01255    if (o->owner)
01256       ast_queue_hangup(o->owner);
01257    setformat(o, O_CLOSE);
01258    return RESULT_SUCCESS;
01259 }
01260 
01261 static int console_hangup(int fd, int argc, char *argv[])
01262 {
01263    struct chan_oss_pvt *o = find_desc(oss_active);
01264 
01265    if (argc != 2)
01266       return RESULT_SHOWUSAGE;
01267    o->cursound = -1;
01268    o->nosound = 0;
01269    if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
01270       ast_cli(fd, "No call to hang up\n");
01271       return RESULT_FAILURE;
01272    }
01273    o->hookstate = 0;
01274    if (o->owner)
01275       ast_queue_hangup(o->owner);
01276    setformat(o, O_CLOSE);
01277    return RESULT_SUCCESS;
01278 }
01279 
01280 static char hangup_usage[] =
01281    "Usage: console hangup\n"
01282    "       Hangs up any call currently placed on the console.\n";
01283 
01284 static int console_flash_deprecated(int fd, int argc, char *argv[])
01285 {
01286    struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
01287    struct chan_oss_pvt *o = find_desc(oss_active);
01288 
01289    if (argc != 1)
01290       return RESULT_SHOWUSAGE;
01291    o->cursound = -1;
01292    o->nosound = 0; /* when cursound is -1 nosound must be 0 */
01293    if (!o->owner) { /* XXX maybe !o->hookstate too ? */
01294       ast_cli(fd, "No call to flash\n");
01295       return RESULT_FAILURE;
01296    }
01297    o->hookstate = 0;
01298    if (o->owner)
01299       ast_queue_frame(o->owner, &f);
01300    return RESULT_SUCCESS;
01301 }
01302 
01303 static int console_flash(int fd, int argc, char *argv[])
01304 {
01305    struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
01306    struct chan_oss_pvt *o = find_desc(oss_active);
01307 
01308    if (argc != 2)
01309       return RESULT_SHOWUSAGE;
01310    o->cursound = -1;
01311    o->nosound = 0;            /* when cursound is -1 nosound must be 0 */
01312    if (!o->owner) {        /* XXX maybe !o->hookstate too ? */
01313       ast_cli(fd, "No call to flash\n");
01314       return RESULT_FAILURE;
01315    }
01316    o->hookstate = 0;
01317    if (o->owner)
01318       ast_queue_frame(o->owner, &f);
01319    return RESULT_SUCCESS;
01320 }
01321 
01322 static char flash_usage[] =
01323    "Usage: console flash\n"
01324    "       Flashes the call currently placed on the console.\n";
01325 
01326 static int console_dial_deprecated(int fd, int argc, char *argv[])
01327 {
01328    char *s = NULL, *mye = NULL, *myc = NULL;
01329    struct chan_oss_pvt *o = find_desc(oss_active);
01330 
01331    if (argc != 1 && argc != 2)
01332       return RESULT_SHOWUSAGE;
01333    if (o->owner) { /* already in a call */
01334       int i;
01335       struct ast_frame f = { AST_FRAME_DTMF, 0 };
01336 
01337       if (argc == 1) { /* argument is mandatory here */
01338          ast_cli(fd, "Already in a call. You can only dial digits until you hangup.\n");
01339          return RESULT_FAILURE;
01340       }
01341       s = argv[1];
01342       /* send the string one char at a time */
01343       for (i = 0; i < strlen(s); i++) {
01344          f.subclass = s[i];
01345          ast_queue_frame(o->owner, &f);
01346       }
01347       return RESULT_SUCCESS;
01348    }
01349    /* if we have an argument split it into extension and context */
01350    if (argc == 2)
01351       s = ast_ext_ctx(argv[1], &mye, &myc);
01352    /* supply default values if needed */
01353    if (mye == NULL)
01354       mye = o->ext;
01355    if (myc == NULL)
01356       myc = o->ctx;
01357    if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
01358       o->hookstate = 1;
01359       oss_new(o, mye, myc, AST_STATE_RINGING);
01360    } else
01361       ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
01362    if (s)
01363       free(s);
01364    return RESULT_SUCCESS;
01365 }
01366 
01367 static int console_dial(int fd, int argc, char *argv[])
01368 {
01369    char *s = NULL, *mye = NULL, *myc = NULL;
01370    struct chan_oss_pvt *o = find_desc(oss_active);
01371 
01372    if (argc != 2 && argc != 3)
01373       return RESULT_SHOWUSAGE;
01374    if (o->owner) {   /* already in a call */
01375       int i;
01376       struct ast_frame f = { AST_FRAME_DTMF, 0 };
01377 
01378       if (argc == 2) {  /* argument is mandatory here */
01379          ast_cli(fd, "Already in a call. You can only dial digits until you hangup.\n");
01380          return RESULT_FAILURE;
01381       }
01382       s = argv[2];
01383       /* send the string one char at a time */
01384       for (i = 0; i < strlen(s); i++) {
01385          f.subclass = s[i];
01386          ast_queue_frame(o->owner, &f);
01387       }
01388       return RESULT_SUCCESS;
01389    }
01390    /* if we have an argument split it into extension and context */
01391    if (argc == 3)
01392       s = ast_ext_ctx(argv[2], &mye, &myc);
01393    /* supply default values if needed */
01394    if (mye == NULL)
01395       mye = o->ext;
01396    if (myc == NULL)
01397       myc = o->ctx;
01398    if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
01399       o->hookstate = 1;
01400       oss_new(o, mye, myc, AST_STATE_RINGING);
01401    } else
01402       ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
01403    if (s)
01404       free(s);
01405    return RESULT_SUCCESS;
01406 }
01407 
01408 static char dial_usage[] =
01409    "Usage: console dial [extension[@context]]\n"
01410    "       Dials a given extension (and context if specified)\n";
01411 
01412 static int __console_mute_unmute(int mute)
01413 {
01414    struct chan_oss_pvt *o = find_desc(oss_active);
01415    
01416    o->mute = mute;
01417    return RESULT_SUCCESS;
01418 }
01419 
01420 static int console_mute_deprecated(int fd, int argc, char *argv[])
01421 {
01422    if (argc != 1)
01423       return RESULT_SHOWUSAGE;
01424 
01425    return __console_mute_unmute(1);
01426 }
01427 
01428 static int console_mute(int fd, int argc, char *argv[])
01429 {
01430    if (argc != 2)
01431       return RESULT_SHOWUSAGE;
01432 
01433    return __console_mute_unmute(1);
01434 }
01435 
01436 static char mute_usage[] =
01437    "Usage: console mute\nMutes the microphone\n";
01438 
01439 static int console_unmute_deprecated(int fd, int argc, char *argv[])
01440 {
01441    if (argc != 1)
01442       return RESULT_SHOWUSAGE;
01443 
01444    return __console_mute_unmute(0);
01445 }
01446 
01447 static int console_unmute(int fd, int argc, char *argv[])
01448 {
01449    if (argc != 2)
01450       return RESULT_SHOWUSAGE;
01451 
01452    return __console_mute_unmute(0);
01453 }
01454 
01455 static char unmute_usage[] =
01456    "Usage: console unmute\nUnmutes the microphone\n";
01457 
01458 static int console_transfer_deprecated(int fd, int argc, char *argv[])
01459 {
01460    struct chan_oss_pvt *o = find_desc(oss_active);
01461    struct ast_channel *b = NULL;
01462    char *tmp, *ext, *ctx;
01463 
01464    if (argc != 2)
01465       return RESULT_SHOWUSAGE;
01466    if (o == NULL)
01467       return RESULT_FAILURE;
01468    if (o->owner ==NULL || (b = ast_bridged_channel(o->owner)) == NULL) {
01469       ast_cli(fd, "There is no call to transfer\n");
01470       return RESULT_SUCCESS;
01471    }
01472 
01473    tmp = ast_ext_ctx(argv[1], &ext, &ctx);
01474    if (ctx == NULL)     /* supply default context if needed */
01475       ctx = o->owner->context;
01476    if (!ast_exists_extension(b, ctx, ext, 1, b->cid.cid_num))
01477       ast_cli(fd, "No such extension exists\n");
01478    else {
01479       ast_cli(fd, "Whee, transferring %s to %s@%s.\n",
01480          b->name, ext, ctx);
01481       if (ast_async_goto(b, ctx, ext, 1))
01482          ast_cli(fd, "Failed to transfer :(\n");
01483    }
01484    if (tmp)
01485       free(tmp);
01486    return RESULT_SUCCESS;
01487 }
01488 
01489 static int console_transfer(int fd, int argc, char *argv[])
01490 {
01491    struct chan_oss_pvt *o = find_desc(oss_active);
01492    struct ast_channel *b = NULL;
01493    char *tmp, *ext, *ctx;
01494 
01495    if (argc != 3)
01496       return RESULT_SHOWUSAGE;
01497    if (o == NULL)
01498       return RESULT_FAILURE;
01499    if (o->owner == NULL || (b = ast_bridged_channel(o->owner)) == NULL) {
01500       ast_cli(fd, "There is no call to transfer\n");
01501       return RESULT_SUCCESS;
01502    }
01503 
01504    tmp = ast_ext_ctx(argv[2], &ext, &ctx);
01505    if (ctx == NULL)        /* supply default context if needed */
01506       ctx = o->owner->context;
01507    if (!ast_exists_extension(b, ctx, ext, 1, b->cid.cid_num))
01508       ast_cli(fd, "No such extension exists\n");
01509    else {
01510       ast_cli(fd, "Whee, transferring %s to %s@%s.\n", b->name, ext, ctx);
01511       if (ast_async_goto(b, ctx, ext, 1))
01512          ast_cli(fd, "Failed to transfer :(\n");
01513    }
01514    if (tmp)
01515       free(tmp);
01516    return RESULT_SUCCESS;
01517 }
01518 
01519 static char transfer_usage[] =
01520    "Usage: console transfer <extension>[@context]\n"
01521    "       Transfers the currently connected call to the given extension (and\n"
01522    "context if specified)\n";
01523 
01524 static int console_active_deprecated(int fd, int argc, char *argv[])
01525 {
01526    if (argc == 1)
01527       ast_cli(fd, "active console is [%s]\n", oss_active);
01528    else if (argc != 2)
01529       return RESULT_SHOWUSAGE;
01530    else {
01531       struct chan_oss_pvt *o;
01532       if (strcmp(argv[1], "show") == 0) {
01533          for (o = oss_default.next; o; o = o->next)
01534             ast_cli(fd, "device [%s] exists\n", o->name);
01535          return RESULT_SUCCESS;
01536       }
01537       o = find_desc(argv[1]);
01538       if (o == NULL)
01539          ast_cli(fd, "No device [%s] exists\n", argv[1]);
01540       else
01541          oss_active = o->name;
01542    }
01543    return RESULT_SUCCESS;
01544 }
01545 
01546 static int console_active(int fd, int argc, char *argv[])
01547 {
01548    if (argc == 2)
01549       ast_cli(fd, "active console is [%s]\n", oss_active);
01550    else if (argc != 3)
01551       return RESULT_SHOWUSAGE;
01552    else {
01553       struct chan_oss_pvt *o;
01554       if (strcmp(argv[2], "show") == 0) {
01555          for (o = oss_default.next; o; o = o->next)
01556             ast_cli(fd, "device [%s] exists\n", o->name);
01557          return RESULT_SUCCESS;
01558       }
01559       o = find_desc(argv[2]);
01560       if (o == NULL)
01561          ast_cli(fd, "No device [%s] exists\n", argv[2]);
01562       else
01563          oss_active = o->name;
01564    }
01565    return RESULT_SUCCESS;
01566 }
01567 
01568 static char active_usage[] =
01569    "Usage: console active [device]\n"
01570    "       If used without a parameter, displays which device is the current\n"
01571    "console.  If a device is specified, the console sound device is changed to\n"
01572    "the device specified.\n";
01573 
01574 /*
01575  * store the boost factor
01576  */
01577 static void store_boost(struct chan_oss_pvt *o, char *s)
01578 {
01579    double boost = 0;
01580    if (sscanf(s, "%30lf", &boost) != 1) {
01581       ast_log(LOG_WARNING, "invalid boost <%s>\n", s);
01582       return;
01583    }
01584    if (boost < -BOOST_MAX) {
01585       ast_log(LOG_WARNING, "boost %s too small, using %d\n", s, -BOOST_MAX);
01586       boost = -BOOST_MAX;
01587    } else if (boost > BOOST_MAX) {
01588       ast_log(LOG_WARNING, "boost %s too large, using %d\n", s, BOOST_MAX);
01589       boost = BOOST_MAX;
01590    }
01591    boost = exp(log(10) * boost / 20) * BOOST_SCALE;
01592    o->boost = boost;
01593    ast_log(LOG_WARNING, "setting boost %s to %d\n", s, o->boost);
01594 }
01595 
01596 static int do_boost(int fd, int argc, char *argv[])
01597 {
01598    struct chan_oss_pvt *o = find_desc(oss_active);
01599 
01600    if (argc == 2)
01601       ast_cli(fd, "boost currently %5.1f\n", 20 * log10(((double) o->boost / (double) BOOST_SCALE)));
01602    else if (argc == 3)
01603       store_boost(o, argv[2]);
01604    return RESULT_SUCCESS;
01605 }
01606 
01607 static struct ast_cli_entry cli_oss_answer_deprecated = {
01608    { "answer", NULL },
01609    console_answer_deprecated, NULL,
01610    NULL };
01611 
01612 static struct ast_cli_entry cli_oss_hangup_deprecated = {
01613    { "hangup", NULL },
01614    console_hangup_deprecated, NULL,
01615    NULL };
01616 
01617 static struct ast_cli_entry cli_oss_flash_deprecated = {
01618    { "flash", NULL },
01619    console_flash_deprecated, NULL,
01620    NULL };
01621 
01622 static struct ast_cli_entry cli_oss_dial_deprecated = {
01623    { "dial", NULL },
01624    console_dial_deprecated, NULL,
01625         NULL };
01626 
01627 static struct ast_cli_entry cli_oss_mute_deprecated = {
01628    { "mute", NULL },
01629    console_mute_deprecated, NULL,
01630         NULL };
01631 
01632 static struct ast_cli_entry cli_oss_unmute_deprecated = {
01633    { "unmute", NULL },
01634    console_unmute_deprecated, NULL,
01635         NULL };
01636 
01637 static struct ast_cli_entry cli_oss_transfer_deprecated = {
01638    { "transfer", NULL },
01639    console_transfer_deprecated, NULL,
01640         NULL };
01641 
01642 static struct ast_cli_entry cli_oss_send_text_deprecated = {
01643    { "send", "text", NULL },
01644    console_sendtext_deprecated, NULL,
01645         NULL };
01646 
01647 static struct ast_cli_entry cli_oss_autoanswer_deprecated = {
01648    { "autoanswer", NULL },
01649    console_autoanswer_deprecated, NULL,
01650         NULL, autoanswer_complete_deprecated };
01651 
01652 static struct ast_cli_entry cli_oss_boost_deprecated = {
01653    { "oss", "boost", NULL },
01654    do_boost, NULL,
01655    NULL };
01656 
01657 static struct ast_cli_entry cli_oss_active_deprecated = {
01658    { "console", NULL },
01659    console_active_deprecated, NULL,
01660         NULL };
01661 
01662 static struct ast_cli_entry cli_oss[] = {
01663    { { "console", "answer", NULL },
01664    console_answer, "Answer an incoming console call",
01665    answer_usage, NULL, &cli_oss_answer_deprecated },
01666 
01667    { { "console", "hangup", NULL },
01668    console_hangup, "Hangup a call on the console",
01669    hangup_usage, NULL, &cli_oss_hangup_deprecated },
01670 
01671    { { "console", "flash", NULL },
01672    console_flash, "Flash a call on the console",
01673    flash_usage, NULL, &cli_oss_flash_deprecated },
01674 
01675    { { "console", "dial", NULL },
01676    console_dial, "Dial an extension on the console",
01677    dial_usage, NULL, &cli_oss_dial_deprecated },
01678 
01679    { { "console", "mute", NULL },
01680    console_mute, "Disable mic input",
01681    mute_usage, NULL, &cli_oss_mute_deprecated },
01682 
01683    { { "console", "unmute", NULL },
01684    console_unmute, "Enable mic input",
01685    unmute_usage, NULL, &cli_oss_unmute_deprecated },
01686 
01687    { { "console", "transfer", NULL },
01688    console_transfer, "Transfer a call to a different extension",
01689    transfer_usage, NULL, &cli_oss_transfer_deprecated },
01690 
01691    { { "console", "send", "text", NULL },
01692    console_sendtext, "Send text to the remote device",
01693    sendtext_usage, NULL, &cli_oss_send_text_deprecated },
01694 
01695    { { "console", "autoanswer", NULL },
01696    console_autoanswer, "Sets/displays autoanswer",
01697    autoanswer_usage, autoanswer_complete, &cli_oss_autoanswer_deprecated },
01698 
01699    { { "console", "boost", NULL },
01700    do_boost, "Sets/displays mic boost in dB",
01701    NULL, NULL, &cli_oss_boost_deprecated },
01702 
01703    { { "console", "active", NULL },
01704    console_active, "Sets/displays active console",
01705    active_usage, NULL, &cli_oss_active_deprecated },
01706 };
01707 
01708 /*
01709  * store the mixer argument from the config file, filtering possibly
01710  * invalid or dangerous values (the string is used as argument for
01711  * system("mixer %s")
01712  */
01713 static void store_mixer(struct chan_oss_pvt *o, char *s)
01714 {
01715    int i;
01716 
01717    for (i = 0; i < strlen(s); i++) {
01718       if (!isalnum(s[i]) && strchr(" \t-/", s[i]) == NULL) {
01719          ast_log(LOG_WARNING, "Suspect char %c in mixer cmd, ignoring:\n\t%s\n", s[i], s);
01720          return;
01721       }
01722    }
01723    if (o->mixer_cmd)
01724       free(o->mixer_cmd);
01725    o->mixer_cmd = ast_strdup(s);
01726    ast_log(LOG_WARNING, "setting mixer %s\n", s);
01727 }
01728 
01729 /*
01730  * store the callerid components
01731  */
01732 static void store_callerid(struct chan_oss_pvt *o, char *s)
01733 {
01734    ast_callerid_split(s, o->cid_name, sizeof(o->cid_name), o->cid_num, sizeof(o->cid_num));
01735 }
01736 
01737 /*
01738  * grab fields from the config file, init the descriptor and open the device.
01739  */
01740 static struct chan_oss_pvt *store_config(struct ast_config *cfg, char *ctg)
01741 {
01742    struct ast_variable *v;
01743    struct chan_oss_pvt *o;
01744 
01745    if (ctg == NULL) {
01746       o = &oss_default;
01747       ctg = "general";
01748    } else {
01749       if (!(o = ast_calloc(1, sizeof(*o))))
01750          return NULL;
01751       *o = oss_default;
01752       /* "general" is also the default thing */
01753       if (strcmp(ctg, "general") == 0) {
01754          o->name = ast_strdup("dsp");
01755          oss_active = o->name;
01756          goto openit;
01757       }
01758       o->name = ast_strdup(ctg);
01759    }
01760 
01761    strcpy(o->mohinterpret, "default");
01762 
01763    o->lastopen = ast_tvnow(); /* don't leave it 0 or tvdiff may wrap */
01764    /* fill other fields from configuration */
01765    for (v = ast_variable_browse(cfg, ctg); v; v = v->next) {
01766       M_START(v->name, v->value);
01767 
01768       /* handle jb conf */
01769       if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
01770          continue;
01771 
01772       M_BOOL("autoanswer", o->autoanswer)
01773          M_BOOL("autohangup", o->autohangup)
01774          M_BOOL("overridecontext", o->overridecontext)
01775          M_STR("device", o->device)
01776          M_UINT("frags", o->frags)
01777          M_UINT("debug", oss_debug)
01778          M_UINT("queuesize", o->queuesize)
01779          M_STR("context", o->ctx)
01780          M_STR("language", o->language)
01781          M_STR("mohinterpret", o->mohinterpret)
01782          M_STR("extension", o->ext)
01783          M_F("mixer", store_mixer(o, v->value))
01784          M_F("callerid", store_callerid(o, v->value))
01785          M_F("boost", store_boost(o, v->value))
01786          M_END(;
01787          );
01788    }
01789    if (ast_strlen_zero(o->device))
01790       ast_copy_string(o->device, DEV_DSP, sizeof(o->device));
01791    if (o->mixer_cmd) {
01792       char *cmd;
01793 
01794       if (asprintf(&cmd, "mixer %s", o->mixer_cmd) < 0) {
01795          ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
01796       } else {
01797          ast_log(LOG_WARNING, "running [%s]\n", cmd);
01798          if (system(cmd) < 0) {
01799             ast_log(LOG_WARNING, "system() failed: %s\n", strerror(errno));
01800          }
01801          free(cmd);
01802       }
01803    }
01804    if (o == &oss_default)     /* we are done with the default */
01805       return NULL;
01806 
01807   openit:
01808 #if TRYOPEN
01809    if (setformat(o, O_RDWR) < 0) {  /* open device */
01810       if (option_verbose > 0) {
01811          ast_verbose(VERBOSE_PREFIX_2 "Device %s not detected\n", ctg);
01812          ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding " "'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
01813       }
01814       goto error;
01815    }
01816    if (o->duplex != M_FULL)
01817       ast_log(LOG_WARNING, "XXX I don't work right with non " "full-duplex sound cards XXX\n");
01818 #endif /* TRYOPEN */
01819    if (pipe(o->sndcmd) != 0) {
01820       ast_log(LOG_ERROR, "Unable to create pipe\n");
01821       goto error;
01822    }
01823    ast_pthread_create_background(&o->sthread, NULL, sound_thread, o);
01824    /* link into list of devices */
01825    if (o != &oss_default) {
01826       o->next = oss_default.next;
01827       oss_default.next = o;
01828    }
01829    return o;
01830 
01831   error:
01832    if (o != &oss_default)
01833       free(o);
01834    return NULL;
01835 }
01836 
01837 static int load_module(void)
01838 {
01839    struct ast_config *cfg = NULL;
01840    char *ctg = NULL;
01841 
01842    /* Copy the default jb config over global_jbconf */
01843    memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
01844 
01845    /* load config file */
01846    if (!(cfg = ast_config_load(config))) {
01847       ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
01848       return AST_MODULE_LOAD_DECLINE;
01849    }
01850 
01851    do {
01852       store_config(cfg, ctg);
01853    } while ( (ctg = ast_category_browse(cfg, ctg)) != NULL);
01854 
01855    ast_config_destroy(cfg);
01856 
01857    if (find_desc(oss_active) == NULL) {
01858       ast_log(LOG_NOTICE, "Device %s not found\n", oss_active);
01859       /* XXX we could default to 'dsp' perhaps ? */
01860       /* XXX should cleanup allocated memory etc. */
01861       return AST_MODULE_LOAD_FAILURE;
01862    }
01863 
01864    if (ast_channel_register(&oss_tech)) {
01865       ast_log(LOG_ERROR, "Unable to register channel type 'OSS'\n");
01866       return AST_MODULE_LOAD_FAILURE;
01867    }
01868 
01869    ast_cli_register_multiple(cli_oss, sizeof(cli_oss) / sizeof(struct ast_cli_entry));
01870 
01871    return AST_MODULE_LOAD_SUCCESS;
01872 }
01873 
01874 
01875 static int unload_module(void)
01876 {
01877    struct chan_oss_pvt *o;
01878 
01879    ast_channel_unregister(&oss_tech);
01880    ast_cli_unregister_multiple(cli_oss, sizeof(cli_oss) / sizeof(struct ast_cli_entry));
01881 
01882    for (o = oss_default.next; o; o = o->next) {
01883       close(o->sounddev);
01884       if (o->sndcmd[0] > 0) {
01885          close(o->sndcmd[0]);
01886          close(o->sndcmd[1]);
01887       }
01888       if (o->owner)
01889          ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD);
01890       if (o->owner)
01891          return -1;
01892       /* XXX what about the thread ? */
01893       /* XXX what about the memory allocated ? */
01894    }
01895    return 0;
01896 }
01897 
01898 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "OSS Console Channel Driver");

Generated on Fri Jan 29 14:25:13 2010 for Asterisk - the Open Source PBX by  doxygen 1.4.7