git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1457 lines
41 KiB
C
1457 lines
41 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2005, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Generic Linux Telephony Interface driver
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*
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* \author Mark Spencer <markster@digium.com>
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*
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* \ingroup channel_drivers
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*/
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/*** MODULEINFO
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<depend>ixjuser</depend>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <ctype.h>
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#include <sys/socket.h>
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#include <sys/time.h>
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#include <arpa/inet.h>
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#include <fcntl.h>
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#include <sys/ioctl.h>
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#include <signal.h>
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#ifdef HAVE_LINUX_COMPILER_H
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#include <linux/compiler.h>
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#endif
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#include <linux/telephony.h>
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/* Still use some IXJ specific stuff */
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#include <linux/version.h>
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#include <linux/ixjuser.h>
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#include "asterisk/lock.h"
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#include "asterisk/channel.h"
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#include "asterisk/config.h"
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#include "asterisk/module.h"
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#include "asterisk/pbx.h"
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#include "asterisk/utils.h"
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#include "asterisk/callerid.h"
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#include "asterisk/causes.h"
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#include "asterisk/stringfields.h"
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#include "asterisk/musiconhold.h"
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#include "chan_phone.h"
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#ifdef QTI_PHONEJACK_TJ_PCI /* check for the newer quicknet driver v.3.1.0 which has this symbol */
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#define QNDRV_VER 310
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#else
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#define QNDRV_VER 100
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#endif
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#if QNDRV_VER > 100
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#ifdef __linux__
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#define IXJ_PHONE_RING_START(x) ioctl(p->fd, PHONE_RING_START, &x);
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#else /* FreeBSD and others */
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#define IXJ_PHONE_RING_START(x) ioctl(p->fd, PHONE_RING_START, x);
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#endif /* __linux__ */
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#else /* older driver */
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#define IXJ_PHONE_RING_START(x) ioctl(p->fd, PHONE_RING_START, &x);
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#endif
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#define DEFAULT_CALLER_ID "Unknown"
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#define PHONE_MAX_BUF 480
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#define DEFAULT_GAIN 0x100
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static const char tdesc[] = "Standard Linux Telephony API Driver";
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static const char config[] = "phone.conf";
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/* Default context for dialtone mode */
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static char context[AST_MAX_EXTENSION] = "default";
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/* Default language */
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static char language[MAX_LANGUAGE] = "";
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static int echocancel = AEC_OFF;
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static int silencesupression = 0;
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static int prefformat = AST_FORMAT_G729A | AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW;
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/* Protect the interface list (of phone_pvt's) */
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AST_MUTEX_DEFINE_STATIC(iflock);
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/* Protect the monitoring thread, so only one process can kill or start it, and not
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when it's doing something critical. */
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AST_MUTEX_DEFINE_STATIC(monlock);
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/* Boolean value whether the monitoring thread shall continue. */
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static unsigned int monitor;
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/* This is the thread for the monitor which checks for input on the channels
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which are not currently in use. */
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static pthread_t monitor_thread = AST_PTHREADT_NULL;
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static int restart_monitor(void);
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/* The private structures of the Phone Jack channels are linked for
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selecting outgoing channels */
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#define MODE_DIALTONE 1
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#define MODE_IMMEDIATE 2
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#define MODE_FXO 3
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#define MODE_FXS 4
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#define MODE_SIGMA 5
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static struct phone_pvt {
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int fd; /* Raw file descriptor for this device */
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struct ast_channel *owner; /* Channel we belong to, possibly NULL */
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int mode; /* Is this in the */
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int lastformat; /* Last output format */
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int lastinput; /* Last input format */
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int ministate; /* Miniature state, for dialtone mode */
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char dev[256]; /* Device name */
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struct phone_pvt *next; /* Next channel in list */
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struct ast_frame fr; /* Frame */
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char offset[AST_FRIENDLY_OFFSET];
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char buf[PHONE_MAX_BUF]; /* Static buffer for reading frames */
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int obuflen;
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int dialtone;
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int txgain, rxgain; /* gain control for playing, recording */
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/* 0x100 - 1.0, 0x200 - 2.0, 0x80 - 0.5 */
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int cpt; /* Call Progress Tone playing? */
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int silencesupression;
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char context[AST_MAX_EXTENSION];
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char obuf[PHONE_MAX_BUF * 2];
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char ext[AST_MAX_EXTENSION];
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char language[MAX_LANGUAGE];
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char cid_num[AST_MAX_EXTENSION];
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char cid_name[AST_MAX_EXTENSION];
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} *iflist = NULL;
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static char cid_num[AST_MAX_EXTENSION];
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static char cid_name[AST_MAX_EXTENSION];
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static struct ast_channel *phone_request(const char *type, int format, void *data, int *cause);
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static int phone_digit_begin(struct ast_channel *ast, char digit);
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static int phone_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
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static int phone_call(struct ast_channel *ast, char *dest, int timeout);
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static int phone_hangup(struct ast_channel *ast);
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static int phone_answer(struct ast_channel *ast);
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static struct ast_frame *phone_read(struct ast_channel *ast);
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static int phone_write(struct ast_channel *ast, struct ast_frame *frame);
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static struct ast_frame *phone_exception(struct ast_channel *ast);
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static int phone_send_text(struct ast_channel *ast, const char *text);
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static int phone_fixup(struct ast_channel *old, struct ast_channel *new);
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static int phone_indicate(struct ast_channel *chan, int condition, const void *data, size_t datalen);
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static const struct ast_channel_tech phone_tech = {
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.type = "Phone",
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.description = tdesc,
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.capabilities = AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW | AST_FORMAT_G729A,
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.requester = phone_request,
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.send_digit_begin = phone_digit_begin,
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.send_digit_end = phone_digit_end,
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.call = phone_call,
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.hangup = phone_hangup,
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.answer = phone_answer,
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.read = phone_read,
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.write = phone_write,
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.exception = phone_exception,
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.indicate = phone_indicate,
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.fixup = phone_fixup
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};
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static struct ast_channel_tech phone_tech_fxs = {
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.type = "Phone",
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.description = tdesc,
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.requester = phone_request,
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.send_digit_begin = phone_digit_begin,
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.send_digit_end = phone_digit_end,
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.call = phone_call,
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.hangup = phone_hangup,
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.answer = phone_answer,
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.read = phone_read,
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.write = phone_write,
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.exception = phone_exception,
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.write_video = phone_write,
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.send_text = phone_send_text,
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.indicate = phone_indicate,
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.fixup = phone_fixup
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};
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static struct ast_channel_tech *cur_tech;
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static int phone_indicate(struct ast_channel *chan, int condition, const void *data, size_t datalen)
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{
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struct phone_pvt *p = chan->tech_pvt;
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int res=-1;
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ast_debug(1, "Requested indication %d on channel %s\n", condition, chan->name);
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switch(condition) {
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case AST_CONTROL_FLASH:
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ioctl(p->fd, IXJCTL_PSTN_SET_STATE, PSTN_ON_HOOK);
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usleep(320000);
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ioctl(p->fd, IXJCTL_PSTN_SET_STATE, PSTN_OFF_HOOK);
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p->lastformat = -1;
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res = 0;
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break;
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case AST_CONTROL_HOLD:
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ast_moh_start(chan, data, NULL);
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break;
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case AST_CONTROL_UNHOLD:
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ast_moh_stop(chan);
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break;
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case AST_CONTROL_SRCUPDATE:
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res = 0;
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break;
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default:
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ast_log(LOG_WARNING, "Condition %d is not supported on channel %s\n", condition, chan->name);
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}
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return res;
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}
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static int phone_fixup(struct ast_channel *old, struct ast_channel *new)
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{
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struct phone_pvt *pvt = old->tech_pvt;
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if (pvt && pvt->owner == old)
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pvt->owner = new;
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return 0;
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}
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static int phone_digit_begin(struct ast_channel *chan, char digit)
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{
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/* XXX Modify this callback to let Asterisk support controlling the length of DTMF */
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return 0;
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}
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static int phone_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
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{
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struct phone_pvt *p;
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int outdigit;
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p = ast->tech_pvt;
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ast_debug(1, "Dialed %c\n", digit);
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switch(digit) {
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case '0':
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case '1':
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case '2':
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case '3':
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case '4':
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case '5':
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case '6':
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case '7':
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case '8':
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case '9':
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outdigit = digit - '0';
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break;
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case '*':
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outdigit = 11;
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break;
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case '#':
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outdigit = 12;
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break;
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case 'f': /*flash*/
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case 'F':
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ioctl(p->fd, IXJCTL_PSTN_SET_STATE, PSTN_ON_HOOK);
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usleep(320000);
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ioctl(p->fd, IXJCTL_PSTN_SET_STATE, PSTN_OFF_HOOK);
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p->lastformat = -1;
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return 0;
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default:
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ast_log(LOG_WARNING, "Unknown digit '%c'\n", digit);
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return -1;
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}
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ast_debug(1, "Dialed %d\n", outdigit);
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ioctl(p->fd, PHONE_PLAY_TONE, outdigit);
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p->lastformat = -1;
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return 0;
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}
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static int phone_call(struct ast_channel *ast, char *dest, int timeout)
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{
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struct phone_pvt *p;
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PHONE_CID cid;
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struct timeval UtcTime = ast_tvnow();
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struct ast_tm tm;
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int start;
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ast_localtime(&UtcTime, &tm, NULL);
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memset(&cid, 0, sizeof(PHONE_CID));
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if(&tm != NULL) {
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snprintf(cid.month, sizeof(cid.month), "%02d",(tm.tm_mon + 1));
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snprintf(cid.day, sizeof(cid.day), "%02d", tm.tm_mday);
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snprintf(cid.hour, sizeof(cid.hour), "%02d", tm.tm_hour);
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snprintf(cid.min, sizeof(cid.min), "%02d", tm.tm_min);
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}
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/* the standard format of ast->callerid is: "name" <number>, but not always complete */
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if (ast_strlen_zero(ast->cid.cid_name))
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strcpy(cid.name, DEFAULT_CALLER_ID);
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else
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ast_copy_string(cid.name, ast->cid.cid_name, sizeof(cid.name));
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if (ast->cid.cid_num)
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ast_copy_string(cid.number, ast->cid.cid_num, sizeof(cid.number));
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p = ast->tech_pvt;
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if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
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ast_log(LOG_WARNING, "phone_call called on %s, neither down nor reserved\n", ast->name);
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return -1;
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}
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ast_debug(1, "Ringing %s on %s (%d)\n", dest, ast->name, ast->fds[0]);
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start = IXJ_PHONE_RING_START(cid);
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if (start == -1)
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return -1;
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if (p->mode == MODE_FXS) {
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char *digit = strchr(dest, '/');
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if (digit)
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{
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digit++;
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while (*digit)
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phone_digit_end(ast, *digit++, 0);
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}
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}
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ast_setstate(ast, AST_STATE_RINGING);
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ast_queue_control(ast, AST_CONTROL_RINGING);
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return 0;
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}
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static int phone_hangup(struct ast_channel *ast)
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{
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struct phone_pvt *p;
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p = ast->tech_pvt;
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ast_debug(1, "phone_hangup(%s)\n", ast->name);
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if (!ast->tech_pvt) {
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ast_log(LOG_WARNING, "Asked to hangup channel not connected\n");
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return 0;
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}
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/* XXX Is there anything we can do to really hang up except stop recording? */
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ast_setstate(ast, AST_STATE_DOWN);
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if (ioctl(p->fd, PHONE_REC_STOP))
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ast_log(LOG_WARNING, "Failed to stop recording\n");
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if (ioctl(p->fd, PHONE_PLAY_STOP))
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ast_log(LOG_WARNING, "Failed to stop playing\n");
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if (ioctl(p->fd, PHONE_RING_STOP))
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ast_log(LOG_WARNING, "Failed to stop ringing\n");
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if (ioctl(p->fd, PHONE_CPT_STOP))
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ast_log(LOG_WARNING, "Failed to stop sounds\n");
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/* If it's an FXO, hang them up */
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if (p->mode == MODE_FXO) {
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if (ioctl(p->fd, PHONE_PSTN_SET_STATE, PSTN_ON_HOOK))
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ast_debug(1, "ioctl(PHONE_PSTN_SET_STATE) failed on %s (%s)\n",ast->name, strerror(errno));
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}
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/* If they're off hook, give a busy signal */
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if (ioctl(p->fd, PHONE_HOOKSTATE)) {
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ast_debug(1, "Got hunghup, giving busy signal\n");
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ioctl(p->fd, PHONE_BUSY);
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p->cpt = 1;
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}
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p->lastformat = -1;
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p->lastinput = -1;
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p->ministate = 0;
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p->obuflen = 0;
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p->dialtone = 0;
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memset(p->ext, 0, sizeof(p->ext));
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((struct phone_pvt *)(ast->tech_pvt))->owner = NULL;
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ast_module_unref(ast_module_info->self);
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ast_verb(3, "Hungup '%s'\n", ast->name);
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ast->tech_pvt = NULL;
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ast_setstate(ast, AST_STATE_DOWN);
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restart_monitor();
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return 0;
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}
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static int phone_setup(struct ast_channel *ast)
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{
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struct phone_pvt *p;
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p = ast->tech_pvt;
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ioctl(p->fd, PHONE_CPT_STOP);
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/* Nothing to answering really, just start recording */
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if (ast->rawreadformat == AST_FORMAT_G729A) {
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/* Prefer g729 */
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ioctl(p->fd, PHONE_REC_STOP);
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if (p->lastinput != AST_FORMAT_G729A) {
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p->lastinput = AST_FORMAT_G729A;
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if (ioctl(p->fd, PHONE_REC_CODEC, G729)) {
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ast_log(LOG_WARNING, "Failed to set codec to g729\n");
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return -1;
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}
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}
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} else if (ast->rawreadformat == AST_FORMAT_G723_1) {
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ioctl(p->fd, PHONE_REC_STOP);
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if (p->lastinput != AST_FORMAT_G723_1) {
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p->lastinput = AST_FORMAT_G723_1;
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if (ioctl(p->fd, PHONE_REC_CODEC, G723_63)) {
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ast_log(LOG_WARNING, "Failed to set codec to g723.1\n");
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return -1;
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}
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}
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} else if (ast->rawreadformat == AST_FORMAT_SLINEAR) {
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ioctl(p->fd, PHONE_REC_STOP);
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if (p->lastinput != AST_FORMAT_SLINEAR) {
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p->lastinput = AST_FORMAT_SLINEAR;
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if (ioctl(p->fd, PHONE_REC_CODEC, LINEAR16)) {
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ast_log(LOG_WARNING, "Failed to set codec to signed linear 16\n");
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return -1;
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}
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}
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} else if (ast->rawreadformat == AST_FORMAT_ULAW) {
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ioctl(p->fd, PHONE_REC_STOP);
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if (p->lastinput != AST_FORMAT_ULAW) {
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p->lastinput = AST_FORMAT_ULAW;
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if (ioctl(p->fd, PHONE_REC_CODEC, ULAW)) {
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ast_log(LOG_WARNING, "Failed to set codec to uLaw\n");
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return -1;
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}
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}
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} else if (p->mode == MODE_FXS) {
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ioctl(p->fd, PHONE_REC_STOP);
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if (p->lastinput != ast->rawreadformat) {
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p->lastinput = ast->rawreadformat;
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if (ioctl(p->fd, PHONE_REC_CODEC, ast->rawreadformat)) {
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ast_log(LOG_WARNING, "Failed to set codec to %d\n",
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ast->rawreadformat);
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return -1;
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}
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}
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} else {
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ast_log(LOG_WARNING, "Can't do format %s\n", ast_getformatname(ast->rawreadformat));
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return -1;
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}
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if (ioctl(p->fd, PHONE_REC_START)) {
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ast_log(LOG_WARNING, "Failed to start recording\n");
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return -1;
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}
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/* set the DTMF times (the default is too short) */
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ioctl(p->fd, PHONE_SET_TONE_ON_TIME, 300);
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ioctl(p->fd, PHONE_SET_TONE_OFF_TIME, 200);
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return 0;
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}
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static int phone_answer(struct ast_channel *ast)
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{
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struct phone_pvt *p;
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p = ast->tech_pvt;
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/* In case it's a LineJack, take it off hook */
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if (p->mode == MODE_FXO) {
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if (ioctl(p->fd, PHONE_PSTN_SET_STATE, PSTN_OFF_HOOK))
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ast_debug(1, "ioctl(PHONE_PSTN_SET_STATE) failed on %s (%s)\n", ast->name, strerror(errno));
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else
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ast_debug(1, "Took linejack off hook\n");
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}
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phone_setup(ast);
|
|
ast_debug(1, "phone_answer(%s)\n", ast->name);
|
|
ast->rings = 0;
|
|
ast_setstate(ast, AST_STATE_UP);
|
|
return 0;
|
|
}
|
|
|
|
#if 0
|
|
static char phone_2digit(char c)
|
|
{
|
|
if (c == 12)
|
|
return '#';
|
|
else if (c == 11)
|
|
return '*';
|
|
else if ((c < 10) && (c >= 0))
|
|
return '0' + c - 1;
|
|
else
|
|
return '?';
|
|
}
|
|
#endif
|
|
|
|
static struct ast_frame *phone_exception(struct ast_channel *ast)
|
|
{
|
|
int res;
|
|
union telephony_exception phonee;
|
|
struct phone_pvt *p = ast->tech_pvt;
|
|
char digit;
|
|
|
|
/* Some nice norms */
|
|
p->fr.datalen = 0;
|
|
p->fr.samples = 0;
|
|
p->fr.data.ptr = NULL;
|
|
p->fr.src = "Phone";
|
|
p->fr.offset = 0;
|
|
p->fr.mallocd=0;
|
|
p->fr.delivery = ast_tv(0,0);
|
|
|
|
phonee.bytes = ioctl(p->fd, PHONE_EXCEPTION);
|
|
if (phonee.bits.dtmf_ready) {
|
|
ast_debug(1, "phone_exception(): DTMF\n");
|
|
|
|
/* We've got a digit -- Just handle this nicely and easily */
|
|
digit = ioctl(p->fd, PHONE_GET_DTMF_ASCII);
|
|
p->fr.subclass = digit;
|
|
p->fr.frametype = AST_FRAME_DTMF;
|
|
return &p->fr;
|
|
}
|
|
if (phonee.bits.hookstate) {
|
|
ast_debug(1, "Hookstate changed\n");
|
|
res = ioctl(p->fd, PHONE_HOOKSTATE);
|
|
/* See if we've gone on hook, if so, notify by returning NULL */
|
|
ast_debug(1, "New hookstate: %d\n", res);
|
|
if (!res && (p->mode != MODE_FXO))
|
|
return NULL;
|
|
else {
|
|
if (ast->_state == AST_STATE_RINGING) {
|
|
/* They've picked up the phone */
|
|
p->fr.frametype = AST_FRAME_CONTROL;
|
|
p->fr.subclass = AST_CONTROL_ANSWER;
|
|
phone_setup(ast);
|
|
ast_setstate(ast, AST_STATE_UP);
|
|
return &p->fr;
|
|
} else
|
|
ast_log(LOG_WARNING, "Got off hook in weird state %d\n", ast->_state);
|
|
}
|
|
}
|
|
#if 1
|
|
if (phonee.bits.pstn_ring)
|
|
ast_verbose("Unit is ringing\n");
|
|
if (phonee.bits.caller_id) {
|
|
ast_verbose("We have caller ID\n");
|
|
}
|
|
if (phonee.bits.pstn_wink)
|
|
ast_verbose("Detected Wink\n");
|
|
#endif
|
|
/* Strange -- nothing there.. */
|
|
p->fr.frametype = AST_FRAME_NULL;
|
|
p->fr.subclass = 0;
|
|
return &p->fr;
|
|
}
|
|
|
|
static struct ast_frame *phone_read(struct ast_channel *ast)
|
|
{
|
|
int res;
|
|
struct phone_pvt *p = ast->tech_pvt;
|
|
|
|
|
|
/* Some nice norms */
|
|
p->fr.datalen = 0;
|
|
p->fr.samples = 0;
|
|
p->fr.data.ptr = NULL;
|
|
p->fr.src = "Phone";
|
|
p->fr.offset = 0;
|
|
p->fr.mallocd=0;
|
|
p->fr.delivery = ast_tv(0,0);
|
|
|
|
/* Try to read some data... */
|
|
CHECK_BLOCKING(ast);
|
|
res = read(p->fd, p->buf, PHONE_MAX_BUF);
|
|
ast_clear_flag(ast, AST_FLAG_BLOCKING);
|
|
if (res < 0) {
|
|
#if 0
|
|
if (errno == EAGAIN) {
|
|
ast_log(LOG_WARNING, "Null frame received\n");
|
|
p->fr.frametype = AST_FRAME_NULL;
|
|
p->fr.subclass = 0;
|
|
return &p->fr;
|
|
}
|
|
#endif
|
|
ast_log(LOG_WARNING, "Error reading: %s\n", strerror(errno));
|
|
return NULL;
|
|
}
|
|
p->fr.data.ptr = p->buf;
|
|
if (p->mode != MODE_FXS)
|
|
switch(p->buf[0] & 0x3) {
|
|
case '0':
|
|
case '1':
|
|
/* Normal */
|
|
break;
|
|
case '2':
|
|
case '3':
|
|
/* VAD/CNG, only send two words */
|
|
res = 4;
|
|
break;
|
|
}
|
|
p->fr.samples = 240;
|
|
p->fr.datalen = res;
|
|
p->fr.frametype = p->lastinput <= AST_FORMAT_AUDIO_MASK ?
|
|
AST_FRAME_VOICE :
|
|
p->lastinput <= AST_FORMAT_PNG ? AST_FRAME_IMAGE
|
|
: AST_FRAME_VIDEO;
|
|
p->fr.subclass = p->lastinput;
|
|
p->fr.offset = AST_FRIENDLY_OFFSET;
|
|
/* Byteswap from little-endian to native-endian */
|
|
if (p->fr.subclass == AST_FORMAT_SLINEAR)
|
|
ast_frame_byteswap_le(&p->fr);
|
|
return &p->fr;
|
|
}
|
|
|
|
static int phone_write_buf(struct phone_pvt *p, const char *buf, int len, int frlen, int swap)
|
|
{
|
|
int res;
|
|
/* Store as much of the buffer as we can, then write fixed frames */
|
|
int space = sizeof(p->obuf) - p->obuflen;
|
|
/* Make sure we have enough buffer space to store the frame */
|
|
if (space < len)
|
|
len = space;
|
|
if (swap)
|
|
ast_swapcopy_samples(p->obuf+p->obuflen, buf, len/2);
|
|
else
|
|
memcpy(p->obuf + p->obuflen, buf, len);
|
|
p->obuflen += len;
|
|
while(p->obuflen > frlen) {
|
|
res = write(p->fd, p->obuf, frlen);
|
|
if (res != frlen) {
|
|
if (res < 1) {
|
|
/*
|
|
* Card is in non-blocking mode now and it works well now, but there are
|
|
* lot of messages like this. So, this message is temporarily disabled.
|
|
*/
|
|
return 0;
|
|
} else {
|
|
ast_log(LOG_WARNING, "Only wrote %d of %d bytes\n", res, frlen);
|
|
}
|
|
}
|
|
p->obuflen -= frlen;
|
|
/* Move memory if necessary */
|
|
if (p->obuflen)
|
|
memmove(p->obuf, p->obuf + frlen, p->obuflen);
|
|
}
|
|
return len;
|
|
}
|
|
|
|
static int phone_send_text(struct ast_channel *ast, const char *text)
|
|
{
|
|
int length = strlen(text);
|
|
return phone_write_buf(ast->tech_pvt, text, length, length, 0) ==
|
|
length ? 0 : -1;
|
|
}
|
|
|
|
static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
|
|
{
|
|
struct phone_pvt *p = ast->tech_pvt;
|
|
int res;
|
|
int maxfr=0;
|
|
char *pos;
|
|
int sofar;
|
|
int expected;
|
|
int codecset = 0;
|
|
char tmpbuf[4];
|
|
/* Write a frame of (presumably voice) data */
|
|
if (frame->frametype != AST_FRAME_VOICE && p->mode != MODE_FXS) {
|
|
if (frame->frametype != AST_FRAME_IMAGE)
|
|
ast_log(LOG_WARNING, "Don't know what to do with frame type '%d'\n", frame->frametype);
|
|
return 0;
|
|
}
|
|
if (!(frame->subclass &
|
|
(AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW | AST_FORMAT_G729A)) &&
|
|
p->mode != MODE_FXS) {
|
|
ast_log(LOG_WARNING, "Cannot handle frames in %d format\n", frame->subclass);
|
|
return -1;
|
|
}
|
|
#if 0
|
|
/* If we're not in up mode, go into up mode now */
|
|
if (ast->_state != AST_STATE_UP) {
|
|
ast_setstate(ast, AST_STATE_UP);
|
|
phone_setup(ast);
|
|
}
|
|
#else
|
|
if (ast->_state != AST_STATE_UP) {
|
|
/* Don't try tos end audio on-hook */
|
|
return 0;
|
|
}
|
|
#endif
|
|
if (frame->subclass == AST_FORMAT_G729A) {
|
|
if (p->lastformat != AST_FORMAT_G729A) {
|
|
ioctl(p->fd, PHONE_PLAY_STOP);
|
|
ioctl(p->fd, PHONE_REC_STOP);
|
|
if (ioctl(p->fd, PHONE_PLAY_CODEC, G729)) {
|
|
ast_log(LOG_WARNING, "Unable to set G729 mode\n");
|
|
return -1;
|
|
}
|
|
if (ioctl(p->fd, PHONE_REC_CODEC, G729)) {
|
|
ast_log(LOG_WARNING, "Unable to set G729 mode\n");
|
|
return -1;
|
|
}
|
|
p->lastformat = AST_FORMAT_G729A;
|
|
p->lastinput = AST_FORMAT_G729A;
|
|
/* Reset output buffer */
|
|
p->obuflen = 0;
|
|
codecset = 1;
|
|
}
|
|
if (frame->datalen > 80) {
|
|
ast_log(LOG_WARNING, "Frame size too large for G.729 (%d bytes)\n", frame->datalen);
|
|
return -1;
|
|
}
|
|
maxfr = 80;
|
|
} else if (frame->subclass == AST_FORMAT_G723_1) {
|
|
if (p->lastformat != AST_FORMAT_G723_1) {
|
|
ioctl(p->fd, PHONE_PLAY_STOP);
|
|
ioctl(p->fd, PHONE_REC_STOP);
|
|
if (ioctl(p->fd, PHONE_PLAY_CODEC, G723_63)) {
|
|
ast_log(LOG_WARNING, "Unable to set G723.1 mode\n");
|
|
return -1;
|
|
}
|
|
if (ioctl(p->fd, PHONE_REC_CODEC, G723_63)) {
|
|
ast_log(LOG_WARNING, "Unable to set G723.1 mode\n");
|
|
return -1;
|
|
}
|
|
p->lastformat = AST_FORMAT_G723_1;
|
|
p->lastinput = AST_FORMAT_G723_1;
|
|
/* Reset output buffer */
|
|
p->obuflen = 0;
|
|
codecset = 1;
|
|
}
|
|
if (frame->datalen > 24) {
|
|
ast_log(LOG_WARNING, "Frame size too large for G.723.1 (%d bytes)\n", frame->datalen);
|
|
return -1;
|
|
}
|
|
maxfr = 24;
|
|
} else if (frame->subclass == AST_FORMAT_SLINEAR) {
|
|
if (p->lastformat != AST_FORMAT_SLINEAR) {
|
|
ioctl(p->fd, PHONE_PLAY_STOP);
|
|
ioctl(p->fd, PHONE_REC_STOP);
|
|
if (ioctl(p->fd, PHONE_PLAY_CODEC, LINEAR16)) {
|
|
ast_log(LOG_WARNING, "Unable to set 16-bit linear mode\n");
|
|
return -1;
|
|
}
|
|
if (ioctl(p->fd, PHONE_REC_CODEC, LINEAR16)) {
|
|
ast_log(LOG_WARNING, "Unable to set 16-bit linear mode\n");
|
|
return -1;
|
|
}
|
|
p->lastformat = AST_FORMAT_SLINEAR;
|
|
p->lastinput = AST_FORMAT_SLINEAR;
|
|
codecset = 1;
|
|
/* Reset output buffer */
|
|
p->obuflen = 0;
|
|
}
|
|
maxfr = 480;
|
|
} else if (frame->subclass == AST_FORMAT_ULAW) {
|
|
if (p->lastformat != AST_FORMAT_ULAW) {
|
|
ioctl(p->fd, PHONE_PLAY_STOP);
|
|
ioctl(p->fd, PHONE_REC_STOP);
|
|
if (ioctl(p->fd, PHONE_PLAY_CODEC, ULAW)) {
|
|
ast_log(LOG_WARNING, "Unable to set uLaw mode\n");
|
|
return -1;
|
|
}
|
|
if (ioctl(p->fd, PHONE_REC_CODEC, ULAW)) {
|
|
ast_log(LOG_WARNING, "Unable to set uLaw mode\n");
|
|
return -1;
|
|
}
|
|
p->lastformat = AST_FORMAT_ULAW;
|
|
p->lastinput = AST_FORMAT_ULAW;
|
|
codecset = 1;
|
|
/* Reset output buffer */
|
|
p->obuflen = 0;
|
|
}
|
|
maxfr = 240;
|
|
} else {
|
|
if (p->lastformat != frame->subclass) {
|
|
ioctl(p->fd, PHONE_PLAY_STOP);
|
|
ioctl(p->fd, PHONE_REC_STOP);
|
|
if (ioctl(p->fd, PHONE_PLAY_CODEC, frame->subclass)) {
|
|
ast_log(LOG_WARNING, "Unable to set %d mode\n",
|
|
frame->subclass);
|
|
return -1;
|
|
}
|
|
if (ioctl(p->fd, PHONE_REC_CODEC, frame->subclass)) {
|
|
ast_log(LOG_WARNING, "Unable to set %d mode\n",
|
|
frame->subclass);
|
|
return -1;
|
|
}
|
|
p->lastformat = frame->subclass;
|
|
p->lastinput = frame->subclass;
|
|
codecset = 1;
|
|
/* Reset output buffer */
|
|
p->obuflen = 0;
|
|
}
|
|
maxfr = 480;
|
|
}
|
|
if (codecset) {
|
|
ioctl(p->fd, PHONE_REC_DEPTH, 3);
|
|
ioctl(p->fd, PHONE_PLAY_DEPTH, 3);
|
|
if (ioctl(p->fd, PHONE_PLAY_START)) {
|
|
ast_log(LOG_WARNING, "Failed to start playback\n");
|
|
return -1;
|
|
}
|
|
if (ioctl(p->fd, PHONE_REC_START)) {
|
|
ast_log(LOG_WARNING, "Failed to start recording\n");
|
|
return -1;
|
|
}
|
|
}
|
|
/* If we get here, we have a frame of Appropriate data */
|
|
sofar = 0;
|
|
pos = frame->data.ptr;
|
|
while(sofar < frame->datalen) {
|
|
/* Write in no more than maxfr sized frames */
|
|
expected = frame->datalen - sofar;
|
|
if (maxfr < expected)
|
|
expected = maxfr;
|
|
/* XXX Internet Phone Jack does not handle the 4-byte VAD frame properly! XXX
|
|
we have to pad it to 24 bytes still. */
|
|
if (frame->datalen == 4) {
|
|
if (p->silencesupression) {
|
|
(void) memset(tmpbuf + 4, 0, sizeof(tmpbuf) - 4);
|
|
memcpy(tmpbuf, frame->data.ptr, 4);
|
|
expected = 24;
|
|
res = phone_write_buf(p, tmpbuf, expected, maxfr, 0);
|
|
}
|
|
res = 4;
|
|
expected=4;
|
|
} else {
|
|
int swap = 0;
|
|
#if __BYTE_ORDER == __BIG_ENDIAN
|
|
if (frame->subclass == AST_FORMAT_SLINEAR)
|
|
swap = 1; /* Swap big-endian samples to little-endian as we copy */
|
|
#endif
|
|
res = phone_write_buf(p, pos, expected, maxfr, swap);
|
|
}
|
|
if (res != expected) {
|
|
if ((errno != EAGAIN) && (errno != EINTR)) {
|
|
if (res < 0)
|
|
ast_log(LOG_WARNING, "Write returned error (%s)\n", strerror(errno));
|
|
/*
|
|
* Card is in non-blocking mode now and it works well now, but there are
|
|
* lot of messages like this. So, this message is temporarily disabled.
|
|
*/
|
|
#if 0
|
|
else
|
|
ast_log(LOG_WARNING, "Only wrote %d of %d bytes\n", res, frame->datalen);
|
|
#endif
|
|
return -1;
|
|
} else /* Pretend it worked */
|
|
res = expected;
|
|
}
|
|
sofar += res;
|
|
pos += res;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static struct ast_channel *phone_new(struct phone_pvt *i, int state, char *cntx)
|
|
{
|
|
struct ast_channel *tmp;
|
|
struct phone_codec_data queried_codec;
|
|
tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, "", i->ext, i->context, 0, "Phone/%s", i->dev + 5);
|
|
if (tmp) {
|
|
tmp->tech = cur_tech;
|
|
ast_channel_set_fd(tmp, 0, i->fd);
|
|
/* XXX Switching formats silently causes kernel panics XXX */
|
|
if (i->mode == MODE_FXS &&
|
|
ioctl(i->fd, PHONE_QUERY_CODEC, &queried_codec) == 0) {
|
|
if (queried_codec.type == LINEAR16)
|
|
tmp->nativeformats =
|
|
tmp->rawreadformat =
|
|
tmp->rawwriteformat =
|
|
AST_FORMAT_SLINEAR;
|
|
else {
|
|
tmp->nativeformats =
|
|
tmp->rawreadformat =
|
|
tmp->rawwriteformat =
|
|
prefformat & ~AST_FORMAT_SLINEAR;
|
|
}
|
|
}
|
|
else {
|
|
tmp->nativeformats = prefformat;
|
|
tmp->rawreadformat = prefformat;
|
|
tmp->rawwriteformat = prefformat;
|
|
}
|
|
/* no need to call ast_setstate: the channel_alloc already did its job */
|
|
if (state == AST_STATE_RING)
|
|
tmp->rings = 1;
|
|
tmp->tech_pvt = i;
|
|
ast_copy_string(tmp->context, cntx, sizeof(tmp->context));
|
|
if (!ast_strlen_zero(i->ext))
|
|
ast_copy_string(tmp->exten, i->ext, sizeof(tmp->exten));
|
|
else
|
|
strcpy(tmp->exten, "s");
|
|
if (!ast_strlen_zero(i->language))
|
|
ast_string_field_set(tmp, language, i->language);
|
|
|
|
/* Don't use ast_set_callerid() here because it will
|
|
* generate a NewCallerID event before the NewChannel event */
|
|
tmp->cid.cid_ani = ast_strdup(i->cid_num);
|
|
|
|
i->owner = tmp;
|
|
ast_module_ref(ast_module_info->self);
|
|
if (state != AST_STATE_DOWN) {
|
|
if (state == AST_STATE_RING) {
|
|
ioctl(tmp->fds[0], PHONE_RINGBACK);
|
|
i->cpt = 1;
|
|
}
|
|
if (ast_pbx_start(tmp)) {
|
|
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
|
|
ast_hangup(tmp);
|
|
}
|
|
}
|
|
} else
|
|
ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
|
|
return tmp;
|
|
}
|
|
|
|
static void phone_mini_packet(struct phone_pvt *i)
|
|
{
|
|
int res;
|
|
char buf[1024];
|
|
/* Ignore stuff we read... */
|
|
res = read(i->fd, buf, sizeof(buf));
|
|
if (res < 1) {
|
|
ast_log(LOG_WARNING, "Read returned %d: %s\n", res, strerror(errno));
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void phone_check_exception(struct phone_pvt *i)
|
|
{
|
|
int offhook=0;
|
|
char digit[2] = {0 , 0};
|
|
union telephony_exception phonee;
|
|
/* XXX Do something XXX */
|
|
#if 0
|
|
ast_debug(1, "Exception!\n");
|
|
#endif
|
|
phonee.bytes = ioctl(i->fd, PHONE_EXCEPTION);
|
|
if (phonee.bits.dtmf_ready) {
|
|
digit[0] = ioctl(i->fd, PHONE_GET_DTMF_ASCII);
|
|
if (i->mode == MODE_DIALTONE || i->mode == MODE_FXS || i->mode == MODE_SIGMA) {
|
|
ioctl(i->fd, PHONE_PLAY_STOP);
|
|
ioctl(i->fd, PHONE_REC_STOP);
|
|
ioctl(i->fd, PHONE_CPT_STOP);
|
|
i->dialtone = 0;
|
|
if (strlen(i->ext) < AST_MAX_EXTENSION - 1)
|
|
strncat(i->ext, digit, sizeof(i->ext) - strlen(i->ext) - 1);
|
|
if ((i->mode != MODE_FXS ||
|
|
!(phonee.bytes = ioctl(i->fd, PHONE_EXCEPTION)) ||
|
|
!phonee.bits.dtmf_ready) &&
|
|
ast_exists_extension(NULL, i->context, i->ext, 1, i->cid_num)) {
|
|
/* It's a valid extension in its context, get moving! */
|
|
phone_new(i, AST_STATE_RING, i->context);
|
|
/* No need to restart monitor, we are the monitor */
|
|
} else if (!ast_canmatch_extension(NULL, i->context, i->ext, 1, i->cid_num)) {
|
|
/* There is nothing in the specified extension that can match anymore.
|
|
Try the default */
|
|
if (ast_exists_extension(NULL, "default", i->ext, 1, i->cid_num)) {
|
|
/* Check the default, too... */
|
|
phone_new(i, AST_STATE_RING, "default");
|
|
/* XXX This should probably be justified better XXX */
|
|
} else if (!ast_canmatch_extension(NULL, "default", i->ext, 1, i->cid_num)) {
|
|
/* It's not a valid extension, give a busy signal */
|
|
ast_debug(1, "%s can't match anything in %s or default\n", i->ext, i->context);
|
|
ioctl(i->fd, PHONE_BUSY);
|
|
i->cpt = 1;
|
|
}
|
|
}
|
|
#if 0
|
|
ast_verbose("Extension is %s\n", i->ext);
|
|
#endif
|
|
}
|
|
}
|
|
if (phonee.bits.hookstate) {
|
|
offhook = ioctl(i->fd, PHONE_HOOKSTATE);
|
|
if (offhook) {
|
|
if (i->mode == MODE_IMMEDIATE) {
|
|
phone_new(i, AST_STATE_RING, i->context);
|
|
} else if (i->mode == MODE_DIALTONE) {
|
|
ast_module_ref(ast_module_info->self);
|
|
/* Reset the extension */
|
|
i->ext[0] = '\0';
|
|
/* Play the dialtone */
|
|
i->dialtone++;
|
|
ioctl(i->fd, PHONE_PLAY_STOP);
|
|
ioctl(i->fd, PHONE_PLAY_CODEC, ULAW);
|
|
ioctl(i->fd, PHONE_PLAY_START);
|
|
i->lastformat = -1;
|
|
} else if (i->mode == MODE_SIGMA) {
|
|
ast_module_ref(ast_module_info->self);
|
|
/* Reset the extension */
|
|
i->ext[0] = '\0';
|
|
/* Play the dialtone */
|
|
i->dialtone++;
|
|
ioctl(i->fd, PHONE_DIALTONE);
|
|
}
|
|
} else {
|
|
if (i->dialtone)
|
|
ast_module_unref(ast_module_info->self);
|
|
memset(i->ext, 0, sizeof(i->ext));
|
|
if (i->cpt)
|
|
{
|
|
ioctl(i->fd, PHONE_CPT_STOP);
|
|
i->cpt = 0;
|
|
}
|
|
ioctl(i->fd, PHONE_PLAY_STOP);
|
|
ioctl(i->fd, PHONE_REC_STOP);
|
|
i->dialtone = 0;
|
|
i->lastformat = -1;
|
|
}
|
|
}
|
|
if (phonee.bits.pstn_ring) {
|
|
ast_verbose("Unit is ringing\n");
|
|
phone_new(i, AST_STATE_RING, i->context);
|
|
}
|
|
if (phonee.bits.caller_id)
|
|
ast_verbose("We have caller ID\n");
|
|
|
|
|
|
}
|
|
|
|
static void *do_monitor(void *data)
|
|
{
|
|
fd_set rfds, efds;
|
|
int n, res;
|
|
struct phone_pvt *i;
|
|
int tonepos = 0;
|
|
/* The tone we're playing this round */
|
|
struct timeval wait = {0,0};
|
|
int dotone;
|
|
/* This thread monitors all the frame relay interfaces which are not yet in use
|
|
(and thus do not have a separate thread) indefinitely */
|
|
while (monitor) {
|
|
/* Don't let anybody kill us right away. Nobody should lock the interface list
|
|
and wait for the monitor list, but the other way around is okay. */
|
|
/* Lock the interface list */
|
|
if (ast_mutex_lock(&iflock)) {
|
|
ast_log(LOG_ERROR, "Unable to grab interface lock\n");
|
|
return NULL;
|
|
}
|
|
/* Build the stuff we're going to select on, that is the socket of every
|
|
phone_pvt that does not have an associated owner channel */
|
|
n = -1;
|
|
FD_ZERO(&rfds);
|
|
FD_ZERO(&efds);
|
|
i = iflist;
|
|
dotone = 0;
|
|
while (i) {
|
|
if (FD_ISSET(i->fd, &rfds))
|
|
ast_log(LOG_WARNING, "Descriptor %d appears twice (%s)?\n", i->fd, i->dev);
|
|
if (!i->owner) {
|
|
/* This needs to be watched, as it lacks an owner */
|
|
FD_SET(i->fd, &rfds);
|
|
FD_SET(i->fd, &efds);
|
|
if (i->fd > n)
|
|
n = i->fd;
|
|
if (i->dialtone && i->mode != MODE_SIGMA) {
|
|
/* Remember we're going to have to come back and play
|
|
more dialtones */
|
|
if (ast_tvzero(wait)) {
|
|
/* If we're due for a dialtone, play one */
|
|
if (write(i->fd, DialTone + tonepos, 240) != 240)
|
|
ast_log(LOG_WARNING, "Dial tone write error\n");
|
|
}
|
|
dotone++;
|
|
}
|
|
}
|
|
|
|
i = i->next;
|
|
}
|
|
/* Okay, now that we know what to do, release the interface lock */
|
|
ast_mutex_unlock(&iflock);
|
|
|
|
/* Wait indefinitely for something to happen */
|
|
if (dotone && i && i->mode != MODE_SIGMA) {
|
|
/* If we're ready to recycle the time, set it to 30 ms */
|
|
tonepos += 240;
|
|
if (tonepos >= sizeof(DialTone))
|
|
tonepos = 0;
|
|
if (ast_tvzero(wait)) {
|
|
wait = ast_tv(30000, 0);
|
|
}
|
|
res = ast_select(n + 1, &rfds, NULL, &efds, &wait);
|
|
} else {
|
|
res = ast_select(n + 1, &rfds, NULL, &efds, NULL);
|
|
wait = ast_tv(0,0);
|
|
tonepos = 0;
|
|
}
|
|
/* Okay, select has finished. Let's see what happened. */
|
|
if (res < 0) {
|
|
ast_debug(1, "select return %d: %s\n", res, strerror(errno));
|
|
continue;
|
|
}
|
|
/* If there are no fd's changed, just continue, it's probably time
|
|
to play some more dialtones */
|
|
if (!res)
|
|
continue;
|
|
/* Alright, lock the interface list again, and let's look and see what has
|
|
happened */
|
|
if (ast_mutex_lock(&iflock)) {
|
|
ast_log(LOG_WARNING, "Unable to lock the interface list\n");
|
|
continue;
|
|
}
|
|
|
|
i = iflist;
|
|
for(; i; i=i->next) {
|
|
if (FD_ISSET(i->fd, &rfds)) {
|
|
if (i->owner) {
|
|
continue;
|
|
}
|
|
phone_mini_packet(i);
|
|
}
|
|
if (FD_ISSET(i->fd, &efds)) {
|
|
if (i->owner) {
|
|
continue;
|
|
}
|
|
phone_check_exception(i);
|
|
}
|
|
}
|
|
ast_mutex_unlock(&iflock);
|
|
}
|
|
return NULL;
|
|
|
|
}
|
|
|
|
static int restart_monitor()
|
|
{
|
|
/* If we're supposed to be stopped -- stay stopped */
|
|
if (monitor_thread == AST_PTHREADT_STOP)
|
|
return 0;
|
|
if (ast_mutex_lock(&monlock)) {
|
|
ast_log(LOG_WARNING, "Unable to lock monitor\n");
|
|
return -1;
|
|
}
|
|
if (monitor_thread == pthread_self()) {
|
|
ast_mutex_unlock(&monlock);
|
|
ast_log(LOG_WARNING, "Cannot kill myself\n");
|
|
return -1;
|
|
}
|
|
if (monitor_thread != AST_PTHREADT_NULL) {
|
|
if (ast_mutex_lock(&iflock)) {
|
|
ast_mutex_unlock(&monlock);
|
|
ast_log(LOG_WARNING, "Unable to lock the interface list\n");
|
|
return -1;
|
|
}
|
|
monitor = 0;
|
|
while (pthread_kill(monitor_thread, SIGURG) == 0)
|
|
sched_yield();
|
|
pthread_join(monitor_thread, NULL);
|
|
ast_mutex_unlock(&iflock);
|
|
}
|
|
monitor = 1;
|
|
/* Start a new monitor */
|
|
if (ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL) < 0) {
|
|
ast_mutex_unlock(&monlock);
|
|
ast_log(LOG_ERROR, "Unable to start monitor thread.\n");
|
|
return -1;
|
|
}
|
|
ast_mutex_unlock(&monlock);
|
|
return 0;
|
|
}
|
|
|
|
static struct phone_pvt *mkif(const char *iface, int mode, int txgain, int rxgain)
|
|
{
|
|
/* Make a phone_pvt structure for this interface */
|
|
struct phone_pvt *tmp;
|
|
int flags;
|
|
|
|
tmp = ast_calloc(1, sizeof(*tmp));
|
|
if (tmp) {
|
|
tmp->fd = open(iface, O_RDWR);
|
|
if (tmp->fd < 0) {
|
|
ast_log(LOG_WARNING, "Unable to open '%s'\n", iface);
|
|
ast_free(tmp);
|
|
return NULL;
|
|
}
|
|
if (mode == MODE_FXO) {
|
|
if (ioctl(tmp->fd, IXJCTL_PORT, PORT_PSTN)) {
|
|
ast_debug(1, "Unable to set port to PSTN\n");
|
|
}
|
|
} else {
|
|
if (ioctl(tmp->fd, IXJCTL_PORT, PORT_POTS))
|
|
if (mode != MODE_FXS)
|
|
ast_debug(1, "Unable to set port to POTS\n");
|
|
}
|
|
ioctl(tmp->fd, PHONE_PLAY_STOP);
|
|
ioctl(tmp->fd, PHONE_REC_STOP);
|
|
ioctl(tmp->fd, PHONE_RING_STOP);
|
|
ioctl(tmp->fd, PHONE_CPT_STOP);
|
|
if (ioctl(tmp->fd, PHONE_PSTN_SET_STATE, PSTN_ON_HOOK))
|
|
ast_debug(1, "ioctl(PHONE_PSTN_SET_STATE) failed on %s (%s)\n",iface, strerror(errno));
|
|
if (echocancel != AEC_OFF)
|
|
ioctl(tmp->fd, IXJCTL_AEC_START, echocancel);
|
|
if (silencesupression)
|
|
tmp->silencesupression = 1;
|
|
#ifdef PHONE_VAD
|
|
ioctl(tmp->fd, PHONE_VAD, tmp->silencesupression);
|
|
#endif
|
|
tmp->mode = mode;
|
|
flags = fcntl(tmp->fd, F_GETFL);
|
|
fcntl(tmp->fd, F_SETFL, flags | O_NONBLOCK);
|
|
tmp->owner = NULL;
|
|
tmp->lastformat = -1;
|
|
tmp->lastinput = -1;
|
|
tmp->ministate = 0;
|
|
memset(tmp->ext, 0, sizeof(tmp->ext));
|
|
ast_copy_string(tmp->language, language, sizeof(tmp->language));
|
|
ast_copy_string(tmp->dev, iface, sizeof(tmp->dev));
|
|
ast_copy_string(tmp->context, context, sizeof(tmp->context));
|
|
tmp->next = NULL;
|
|
tmp->obuflen = 0;
|
|
tmp->dialtone = 0;
|
|
tmp->cpt = 0;
|
|
ast_copy_string(tmp->cid_num, cid_num, sizeof(tmp->cid_num));
|
|
ast_copy_string(tmp->cid_name, cid_name, sizeof(tmp->cid_name));
|
|
tmp->txgain = txgain;
|
|
ioctl(tmp->fd, PHONE_PLAY_VOLUME, tmp->txgain);
|
|
tmp->rxgain = rxgain;
|
|
ioctl(tmp->fd, PHONE_REC_VOLUME, tmp->rxgain);
|
|
}
|
|
return tmp;
|
|
}
|
|
|
|
static struct ast_channel *phone_request(const char *type, int format, void *data, int *cause)
|
|
{
|
|
int oldformat;
|
|
struct phone_pvt *p;
|
|
struct ast_channel *tmp = NULL;
|
|
char *name = data;
|
|
|
|
/* Search for an unowned channel */
|
|
if (ast_mutex_lock(&iflock)) {
|
|
ast_log(LOG_ERROR, "Unable to lock interface list???\n");
|
|
return NULL;
|
|
}
|
|
p = iflist;
|
|
while(p) {
|
|
if (p->mode == MODE_FXS ||
|
|
format & (AST_FORMAT_G729A | AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW)) {
|
|
size_t length = strlen(p->dev + 5);
|
|
if (strncmp(name, p->dev + 5, length) == 0 &&
|
|
!isalnum(name[length])) {
|
|
if (!p->owner) {
|
|
tmp = phone_new(p, AST_STATE_DOWN, p->context);
|
|
break;
|
|
} else
|
|
*cause = AST_CAUSE_BUSY;
|
|
}
|
|
}
|
|
p = p->next;
|
|
}
|
|
ast_mutex_unlock(&iflock);
|
|
restart_monitor();
|
|
if (tmp == NULL) {
|
|
oldformat = format;
|
|
format &= (AST_FORMAT_G729A | AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW);
|
|
if (!format) {
|
|
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%d'\n", oldformat);
|
|
return NULL;
|
|
}
|
|
}
|
|
return tmp;
|
|
}
|
|
|
|
/* parse gain value from config file */
|
|
static int parse_gain_value(const char *gain_type, const char *value)
|
|
{
|
|
float gain;
|
|
|
|
/* try to scan number */
|
|
if (sscanf(value, "%f", &gain) != 1)
|
|
{
|
|
ast_log(LOG_ERROR, "Invalid %s value '%s' in '%s' config\n",
|
|
value, gain_type, config);
|
|
return DEFAULT_GAIN;
|
|
}
|
|
|
|
/* multiplicate gain by 1.0 gain value */
|
|
gain = gain * (float)DEFAULT_GAIN;
|
|
|
|
/* percentage? */
|
|
if (value[strlen(value) - 1] == '%')
|
|
return (int)(gain / (float)100);
|
|
|
|
return (int)gain;
|
|
}
|
|
|
|
static int __unload_module(void)
|
|
{
|
|
struct phone_pvt *p, *pl;
|
|
/* First, take us out of the channel loop */
|
|
if (cur_tech)
|
|
ast_channel_unregister(cur_tech);
|
|
if (!ast_mutex_lock(&iflock)) {
|
|
/* Hangup all interfaces if they have an owner */
|
|
p = iflist;
|
|
while(p) {
|
|
if (p->owner)
|
|
ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD);
|
|
p = p->next;
|
|
}
|
|
iflist = NULL;
|
|
ast_mutex_unlock(&iflock);
|
|
} else {
|
|
ast_log(LOG_WARNING, "Unable to lock the monitor\n");
|
|
return -1;
|
|
}
|
|
if (!ast_mutex_lock(&monlock)) {
|
|
if (monitor_thread > AST_PTHREADT_NULL) {
|
|
monitor = 0;
|
|
while (pthread_kill(monitor_thread, SIGURG) == 0)
|
|
sched_yield();
|
|
pthread_join(monitor_thread, NULL);
|
|
}
|
|
monitor_thread = AST_PTHREADT_STOP;
|
|
ast_mutex_unlock(&monlock);
|
|
} else {
|
|
ast_log(LOG_WARNING, "Unable to lock the monitor\n");
|
|
return -1;
|
|
}
|
|
|
|
if (!ast_mutex_lock(&iflock)) {
|
|
/* Destroy all the interfaces and free their memory */
|
|
p = iflist;
|
|
while(p) {
|
|
/* Close the socket, assuming it's real */
|
|
if (p->fd > -1)
|
|
close(p->fd);
|
|
pl = p;
|
|
p = p->next;
|
|
/* Free associated memory */
|
|
ast_free(pl);
|
|
}
|
|
iflist = NULL;
|
|
ast_mutex_unlock(&iflock);
|
|
} else {
|
|
ast_log(LOG_WARNING, "Unable to lock the monitor\n");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int unload_module(void)
|
|
{
|
|
return __unload_module();
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
struct ast_config *cfg;
|
|
struct ast_variable *v;
|
|
struct phone_pvt *tmp;
|
|
int mode = MODE_IMMEDIATE;
|
|
int txgain = DEFAULT_GAIN, rxgain = DEFAULT_GAIN; /* default gain 1.0 */
|
|
struct ast_flags config_flags = { 0 };
|
|
|
|
if ((cfg = ast_config_load(config, config_flags)) == CONFIG_STATUS_FILEINVALID) {
|
|
ast_log(LOG_ERROR, "Config file %s is in an invalid format. Aborting.\n", config);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
/* We *must* have a config file otherwise stop immediately */
|
|
if (!cfg) {
|
|
ast_log(LOG_ERROR, "Unable to load config %s\n", config);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
if (ast_mutex_lock(&iflock)) {
|
|
/* It's a little silly to lock it, but we mind as well just to be sure */
|
|
ast_log(LOG_ERROR, "Unable to lock interface list???\n");
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
v = ast_variable_browse(cfg, "interfaces");
|
|
while(v) {
|
|
/* Create the interface list */
|
|
if (!strcasecmp(v->name, "device")) {
|
|
tmp = mkif(v->value, mode, txgain, rxgain);
|
|
if (tmp) {
|
|
tmp->next = iflist;
|
|
iflist = tmp;
|
|
|
|
} else {
|
|
ast_log(LOG_ERROR, "Unable to register channel '%s'\n", v->value);
|
|
ast_config_destroy(cfg);
|
|
ast_mutex_unlock(&iflock);
|
|
__unload_module();
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
} else if (!strcasecmp(v->name, "silencesupression")) {
|
|
silencesupression = ast_true(v->value);
|
|
} else if (!strcasecmp(v->name, "language")) {
|
|
ast_copy_string(language, v->value, sizeof(language));
|
|
} else if (!strcasecmp(v->name, "callerid")) {
|
|
ast_callerid_split(v->value, cid_name, sizeof(cid_name), cid_num, sizeof(cid_num));
|
|
} else if (!strcasecmp(v->name, "mode")) {
|
|
if (!strncasecmp(v->value, "di", 2))
|
|
mode = MODE_DIALTONE;
|
|
else if (!strncasecmp(v->value, "sig", 3))
|
|
mode = MODE_SIGMA;
|
|
else if (!strncasecmp(v->value, "im", 2))
|
|
mode = MODE_IMMEDIATE;
|
|
else if (!strncasecmp(v->value, "fxs", 3)) {
|
|
mode = MODE_FXS;
|
|
prefformat = 0x01ff0000; /* All non-voice */
|
|
}
|
|
else if (!strncasecmp(v->value, "fx", 2))
|
|
mode = MODE_FXO;
|
|
else
|
|
ast_log(LOG_WARNING, "Unknown mode: %s\n", v->value);
|
|
} else if (!strcasecmp(v->name, "context")) {
|
|
ast_copy_string(context, v->value, sizeof(context));
|
|
} else if (!strcasecmp(v->name, "format")) {
|
|
if (!strcasecmp(v->value, "g729")) {
|
|
prefformat = AST_FORMAT_G729A;
|
|
} else if (!strcasecmp(v->value, "g723.1")) {
|
|
prefformat = AST_FORMAT_G723_1;
|
|
} else if (!strcasecmp(v->value, "slinear")) {
|
|
if (mode == MODE_FXS)
|
|
prefformat |= AST_FORMAT_SLINEAR;
|
|
else prefformat = AST_FORMAT_SLINEAR;
|
|
} else if (!strcasecmp(v->value, "ulaw")) {
|
|
prefformat = AST_FORMAT_ULAW;
|
|
} else
|
|
ast_log(LOG_WARNING, "Unknown format '%s'\n", v->value);
|
|
} else if (!strcasecmp(v->name, "echocancel")) {
|
|
if (!strcasecmp(v->value, "off")) {
|
|
echocancel = AEC_OFF;
|
|
} else if (!strcasecmp(v->value, "low")) {
|
|
echocancel = AEC_LOW;
|
|
} else if (!strcasecmp(v->value, "medium")) {
|
|
echocancel = AEC_MED;
|
|
} else if (!strcasecmp(v->value, "high")) {
|
|
echocancel = AEC_HIGH;
|
|
} else
|
|
ast_log(LOG_WARNING, "Unknown echo cancellation '%s'\n", v->value);
|
|
} else if (!strcasecmp(v->name, "txgain")) {
|
|
txgain = parse_gain_value(v->name, v->value);
|
|
} else if (!strcasecmp(v->name, "rxgain")) {
|
|
rxgain = parse_gain_value(v->name, v->value);
|
|
}
|
|
v = v->next;
|
|
}
|
|
ast_mutex_unlock(&iflock);
|
|
|
|
if (mode == MODE_FXS) {
|
|
phone_tech_fxs.capabilities = prefformat;
|
|
cur_tech = &phone_tech_fxs;
|
|
} else
|
|
cur_tech = (struct ast_channel_tech *) &phone_tech;
|
|
|
|
/* Make sure we can register our Adtranphone channel type */
|
|
|
|
if (ast_channel_register(cur_tech)) {
|
|
ast_log(LOG_ERROR, "Unable to register channel class 'Phone'\n");
|
|
ast_config_destroy(cfg);
|
|
__unload_module();
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
ast_config_destroy(cfg);
|
|
/* And start the monitor for the first time */
|
|
restart_monitor();
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Linux Telephony API Support");
|