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asterisk/channels/chan_unistim.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* UNISTIM channel driver for asterisk
*
* Copyright (C) 2005 - 2007, Cedric Hans
*
* Cedric Hans <cedric.hans@mlkj.net>
*
* Asterisk 1.4 patch by Peter Be
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*!
* \file
*
* \brief chan_unistim channel driver for Asterisk
* \author Cedric Hans <cedric.hans@mlkj.net>
*
* Unistim (Unified Networks IP Stimulus) channel driver
* for Nortel i2002, i2004 and i2050
*
* \ingroup channel_drivers
*/
/*** MODULEINFO
<support_level>extended</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <sys/stat.h>
#include <signal.h>
#if defined(__CYGWIN__)
/*
* cygwin headers are partly inconsistent. struct iovec is defined in sys/uio.h
* which is not included by default by sys/socket.h - in_pktinfo is defined in
* w32api/ws2tcpip.h but this probably has compatibility problems with sys/socket.h
* So for the time being we simply disable HAVE_PKTINFO when building under cygwin.
* This should be done in some common header, but for now this is the only file
* using iovec and in_pktinfo so it suffices to apply the fix here.
*/
#ifdef HAVE_PKTINFO
#undef HAVE_PKTINFO
#endif
#endif /* __CYGWIN__ */
#include "asterisk/paths.h" /* ast_config_AST_LOG_DIR used in (too ?) many places */
#include "asterisk/network.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/event.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/netsock.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/causes.h"
#include "asterisk/indications.h"
/*! Beware, G729 and G723 are not supported by asterisk, except with the proper licence */
#define DEFAULTCONTEXT "default"
#define DEFAULTCALLERID "Unknown"
#define DEFAULTCALLERNAME " "
#define DEFAULTHEIGHT 3
#define USTM_LOG_DIR "unistimHistory"
/*! Size of the transmit buffer */
#define MAX_BUF_SIZE 64
/*! Number of slots for the transmit queue */
#define MAX_BUF_NUMBER 50
/*! Try x times before removing the phone */
#define NB_MAX_RETRANSMIT 8
/*! Nb of milliseconds waited when no events are scheduled */
#define IDLE_WAIT 1000
/*! Wait x milliseconds before resending a packet */
#define RETRANSMIT_TIMER 2000
/*! How often the mailbox is checked for new messages */
#define TIMER_MWI 10000
/*! Not used */
#define DEFAULT_CODEC 0x00
#define SIZE_PAGE 4096
#define DEVICE_NAME_LEN 16
#define AST_CONFIG_MAX_PATH 255
#define MAX_ENTRY_LOG 30
#define SUB_REAL 0
#define SUB_THREEWAY 1
#define MAX_SUBS 2
struct ast_format_cap *global_cap;
enum autoprovision {
AUTOPROVISIONING_NO = 0,
AUTOPROVISIONING_YES,
AUTOPROVISIONING_DB,
AUTOPROVISIONING_TN
};
enum autoprov_extn {
/*! Do not create an extension into the default dialplan */
EXTENSION_NONE = 0,
/*! Prompt user for an extension number and register it */
EXTENSION_ASK,
/*! Register an extension with the line=> value */
EXTENSION_LINE,
/*! Used with AUTOPROVISIONING_TN */
EXTENSION_TN
};
#define OUTPUT_HANDSET 0xC0
#define OUTPUT_HEADPHONE 0xC1
#define OUTPUT_SPEAKER 0xC2
#define VOLUME_LOW 0x01
#define VOLUME_LOW_SPEAKER 0x03
#define VOLUME_NORMAL 0x02
#define VOLUME_INSANELY_LOUD 0x07
#define MUTE_OFF 0x00
#define MUTE_ON 0xFF
#define MUTE_ON_DISCRET 0xCE
#define SIZE_HEADER 6
#define SIZE_MAC_ADDR 17
#define TEXT_LENGTH_MAX 24
#define TEXT_LINE0 0x00
#define TEXT_LINE1 0x20
#define TEXT_LINE2 0x40
#define TEXT_NORMAL 0x05
#define TEXT_INVERSE 0x25
#define STATUS_LENGTH_MAX 28
#define FAV_ICON_NONE 0x00
#define FAV_ICON_ONHOOK_BLACK 0x20
#define FAV_ICON_ONHOOK_WHITE 0x21
#define FAV_ICON_SPEAKER_ONHOOK_BLACK 0x22
#define FAV_ICON_SPEAKER_ONHOOK_WHITE 0x23
#define FAV_ICON_OFFHOOK_BLACK 0x24
#define FAV_ICON_OFFHOOK_WHITE 0x25
#define FAV_ICON_ONHOLD_BLACK 0x26
#define FAV_ICON_ONHOLD_WHITE 0x27
#define FAV_ICON_SPEAKER_OFFHOOK_BLACK 0x28
#define FAV_ICON_SPEAKER_OFFHOOK_WHITE 0x29
#define FAV_ICON_PHONE_BLACK 0x2A
#define FAV_ICON_PHONE_WHITE 0x2B
#define FAV_ICON_SPEAKER_ONHOLD_BLACK 0x2C
#define FAV_ICON_SPEAKER_ONHOLD_WHITE 0x2D
#define FAV_ICON_HEADPHONES 0x2E
#define FAV_ICON_HEADPHONES_ONHOLD 0x2F
#define FAV_ICON_HOME 0x30
#define FAV_ICON_CITY 0x31
#define FAV_ICON_SHARP 0x32
#define FAV_ICON_PAGER 0x33
#define FAV_ICON_CALL_CENTER 0x34
#define FAV_ICON_FAX 0x35
#define FAV_ICON_MAILBOX 0x36
#define FAV_ICON_REFLECT 0x37
#define FAV_ICON_COMPUTER 0x38
#define FAV_ICON_FORWARD 0x39
#define FAV_ICON_LOCKED 0x3A
#define FAV_ICON_TRASH 0x3B
#define FAV_ICON_INBOX 0x3C
#define FAV_ICON_OUTBOX 0x3D
#define FAV_ICON_MEETING 0x3E
#define FAV_ICON_BOX 0x3F
#define FAV_BLINK_FAST 0x20
#define FAV_BLINK_SLOW 0x40
#define FAV_MAX_LENGTH 0x0A
static void dummy(char *unused, ...)
{
return;
}
/*! \brief Global jitterbuffer configuration - by default, jb is disabled
* \note Values shown here match the defaults shown in unistim.conf.sample */
static struct ast_jb_conf default_jbconf =
{
.flags = 0,
.max_size = 200,
.resync_threshold = 1000,
.impl = "fixed",
.target_extra = 40,
};
static struct ast_jb_conf global_jbconf;
/* #define DUMP_PACKET 1 */
/* #define DEBUG_TIMER ast_verbose */
#define DEBUG_TIMER dummy
/*! Enable verbose output. can also be set with the CLI */
static int unistimdebug = 0;
static int unistim_port;
static enum autoprovision autoprovisioning = AUTOPROVISIONING_NO;
static int unistim_keepalive;
static int unistimsock = -1;
static struct {
unsigned int tos;
unsigned int tos_audio;
unsigned int cos;
unsigned int cos_audio;
} qos = { 0, 0, 0, 0 };
static struct io_context *io;
static struct ast_sched_context *sched;
static struct sockaddr_in public_ip = { 0, };
/*! give the IP address for the last packet received */
static struct sockaddr_in address_from;
/*! size of the sockaddr_in (in WSARecvFrom) */
static unsigned int size_addr_from = sizeof(address_from);
/*! Receive buffer address */
static unsigned char *buff;
static int unistim_reloading = 0;
AST_MUTEX_DEFINE_STATIC(unistim_reload_lock);
AST_MUTEX_DEFINE_STATIC(usecnt_lock);
static int usecnt = 0;
/* extern char ast_config_AST_LOG_DIR[AST_CONFIG_MAX_PATH]; */
/*! This is the thread for the monitor which checks for input on the channels
* which are not currently in use. */
static pthread_t monitor_thread = AST_PTHREADT_NULL;
/*! Protect the monitoring thread, so only one process can kill or start it, and not
* when it's doing something critical. */
AST_MUTEX_DEFINE_STATIC(monlock);
/*! Protect the session list */
AST_MUTEX_DEFINE_STATIC(sessionlock);
/*! Protect the device list */
AST_MUTEX_DEFINE_STATIC(devicelock);
enum phone_state {
STATE_INIT,
STATE_AUTHDENY,
STATE_MAINPAGE,
STATE_EXTENSION,
STATE_DIALPAGE,
STATE_RINGING,
STATE_CALL,
STATE_SELECTCODEC,
STATE_CLEANING,
STATE_HISTORY
};
enum handset_state {
STATE_ONHOOK,
STATE_OFFHOOK,
};
enum phone_key {
KEY_0 = 0x40,
KEY_1 = 0x41,
KEY_2 = 0x42,
KEY_3 = 0x43,
KEY_4 = 0x44,
KEY_5 = 0x45,
KEY_6 = 0x46,
KEY_7 = 0x47,
KEY_8 = 0x48,
KEY_9 = 0x49,
KEY_STAR = 0x4a,
KEY_SHARP = 0x4b,
KEY_UP = 0x4c,
KEY_DOWN = 0x4d,
KEY_RIGHT = 0x4e,
KEY_LEFT = 0x4f,
KEY_QUIT = 0x50,
KEY_COPY = 0x51,
KEY_FUNC1 = 0x54,
KEY_FUNC2 = 0x55,
KEY_FUNC3 = 0x56,
KEY_FUNC4 = 0x57,
KEY_ONHOLD = 0x5b,
KEY_HANGUP = 0x5c,
KEY_MUTE = 0x5d,
KEY_HEADPHN = 0x5e,
KEY_LOUDSPK = 0x5f,
KEY_FAV0 = 0x60,
KEY_FAV1 = 0x61,
KEY_FAV2 = 0x62,
KEY_FAV3 = 0x63,
KEY_FAV4 = 0x64,
KEY_FAV5 = 0x65,
KEY_COMPUTR = 0x7b,
KEY_CONF = 0x7c,
KEY_SNDHIST = 0x7d,
KEY_RCVHIST = 0x7e,
KEY_INDEX = 0x7f
};
struct tone_zone_unistim {
char country[3];
int freq1;
int freq2;
};
static const struct tone_zone_unistim frequency[] = {
{"us", 350, 440},
{"fr", 440, 0},
{"au", 413, 438},
{"nl", 425, 0},
{"uk", 350, 440},
{"fi", 425, 0},
{"es", 425, 0},
{"jp", 400, 0},
{"no", 425, 0},
{"at", 420, 0},
{"nz", 400, 0},
{"tw", 350, 440},
{"cl", 400, 0},
{"se", 425, 0},
{"be", 425, 0},
{"sg", 425, 0},
{"il", 414, 0},
{"br", 425, 0},
{"hu", 425, 0},
{"lt", 425, 0},
{"pl", 425, 0},
{"za", 400, 0},
{"pt", 425, 0},
{"ee", 425, 0},
{"mx", 425, 0},
{"in", 400, 0},
{"de", 425, 0},
{"ch", 425, 0},
{"dk", 425, 0},
{"cn", 450, 0},
{"--", 0, 0}
};
struct wsabuf {
u_long len;
unsigned char *buf;
};
struct systemtime {
unsigned short w_year;
unsigned short w_month;
unsigned short w_day_of_week;
unsigned short w_day;
unsigned short w_hour;
unsigned short w_minute;
unsigned short w_second;
unsigned short w_milliseconds;
};
struct unistim_subchannel {
ast_mutex_t lock;
/*! SUBS_REAL or SUBS_THREEWAY */
unsigned int subtype;
/*! Asterisk channel used by the subchannel */
struct ast_channel *owner;
/*! Unistim line */
struct unistim_line *parent;
/*! RTP handle */
struct ast_rtp_instance *rtp;
int alreadygone;
char ringvolume;
char ringstyle;
};
/*!
* \todo Convert to stringfields
*/
struct unistim_line {
ast_mutex_t lock;
/*! Like 200 */
char name[80];
/*! Like USTM/200\@black */
char fullname[80];
/*! pointer to our current connection, channel... */
struct unistim_subchannel *subs[MAX_SUBS];
/*! Extension where to start */
char exten[AST_MAX_EXTENSION];
/*! Context to start in */
char context[AST_MAX_EXTENSION];
/*! Language for asterisk sounds */
char language[MAX_LANGUAGE];
/*! CallerID Number */
char cid_num[AST_MAX_EXTENSION];
/*! Mailbox for MWI */
char mailbox[AST_MAX_EXTENSION];
/*! Used by MWI */
int lastmsgssent;
/*! Used by MWI */
time_t nextmsgcheck;
/*! MusicOnHold class */
char musicclass[MAX_MUSICCLASS];
/*! Call group */
unsigned int callgroup;
/*! Pickup group */
unsigned int pickupgroup;
/*! Account code (for billing) */
char accountcode[80];
/*! AMA flags (for billing) */
int amaflags;
/*! Codec supported */
struct ast_format_cap *cap;
/*! Parkinglot */
char parkinglot[AST_MAX_CONTEXT];
struct unistim_line *next;
struct unistim_device *parent;
};
/*!
* \brief A device containing one or more lines
*/
static struct unistim_device {
int receiver_state; /*!< state of the receiver (see ReceiverState) */
int size_phone_number; /*!< size of the phone number */
char phone_number[16]; /*!< the phone number entered by the user */
char redial_number[16]; /*!< the last phone number entered by the user */
int phone_current; /*!< Number of the current phone */
int pos_fav; /*!< Position of the displayed favorites (used for scrolling) */
char id[18]; /*!< mac address of the current phone in ascii */
char name[DEVICE_NAME_LEN]; /*!< name of the device */
int softkeylinepos; /*!< position of the line softkey (default 0) */
char softkeylabel[6][11]; /*!< soft key label */
char softkeynumber[6][16]; /*!< number dialed when the soft key is pressed */
char softkeyicon[6]; /*!< icon number */
char softkeydevice[6][16]; /*!< name of the device monitored */
struct unistim_device *sp[6]; /*!< pointer to the device monitored by this soft key */
int height; /*!< The number of lines the phone can display */
char maintext0[25]; /*!< when the phone is idle, display this string on line 0 */
char maintext1[25]; /*!< when the phone is idle, display this string on line 1 */
char maintext2[25]; /*!< when the phone is idle, display this string on line 2 */
char titledefault[13]; /*!< title (text before date/time) */
char datetimeformat; /*!< format used for displaying time/date */
char contrast; /*!< contrast */
char country[3]; /*!< country used for dial tone frequency */
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
struct ast_tone_zone *tz; /*!< Tone zone for res_indications (ring, busy, congestion) */
char ringvolume; /*!< Ring volume */
char ringstyle; /*!< Ring melody */
int rtp_port; /*!< RTP port used by the phone */
int rtp_method; /*!< Select the unistim data used to establish a RTP session */
int status_method; /*!< Select the unistim packet used for sending status text */
char codec_number; /*!< The current codec used to make calls */
int missed_call; /*!< Number of call unanswered */
int callhistory; /*!< Allowed to record call history */
char lst_cid[TEXT_LENGTH_MAX]; /*!< Last callerID received */
char lst_cnm[TEXT_LENGTH_MAX]; /*!< Last callername recevied */
char call_forward[AST_MAX_EXTENSION]; /*!< Forward number */
int output; /*!< Handset, headphone or speaker */
int previous_output; /*!< Previous output */
int volume; /*!< Default volume */
int mute; /*!< Mute mode */
int moh; /*!< Music on hold in progress */
int nat; /*!< Used by the obscure ast_rtp_setnat */
enum autoprov_extn extension; /*!< See ifdef EXTENSION for valid values */
char extension_number[11]; /*!< Extension number entered by the user */
char to_delete; /*!< Used in reload */
time_t start_call_timestamp; /*!< timestamp for the length calculation of the call */
struct ast_silence_generator *silence_generator;
struct unistim_line *lines;
struct ast_ha *ha;
struct unistimsession *session;
struct unistim_device *next;
} *devices = NULL;
static struct unistimsession {
ast_mutex_t lock;
struct sockaddr_in sin; /*!< IP address of the phone */
struct sockaddr_in sout; /*!< IP address of server */
int timeout; /*!< time-out in ticks : resend packet if no ack was received before the timeout occured */
unsigned short seq_phone; /*!< sequence number for the next packet (when we receive a request) */
unsigned short seq_server; /*!< sequence number for the next packet (when we send a request) */
unsigned short last_seq_ack; /*!< sequence number of the last ACK received */
unsigned long tick_next_ping; /*!< time for the next ping */
int last_buf_available; /*!< number of a free slot */
int nb_retransmit; /*!< number of retransmition */
int state; /*!< state of the phone (see phone_state) */
int size_buff_entry; /*!< size of the buffer used to enter datas */
char buff_entry[16]; /*!< Buffer for temporary datas */
char macaddr[18]; /*!< mac adress of the phone (not always available) */
struct wsabuf wsabufsend[MAX_BUF_NUMBER]; /*!< Size of each paquet stored in the buffer array & pointer to this buffer */
unsigned char buf[MAX_BUF_NUMBER][MAX_BUF_SIZE]; /*!< Buffer array used to keep the lastest non-acked paquets */
struct unistim_device *device;
struct unistimsession *next;
} *sessions = NULL;
/*!
* \page Unistim datagram formats
*
* Format of datagrams :
* bytes 0 & 1 : ffff for discovery packet, 0000 for everything else
* byte 2 : sequence number (high part)
* byte 3 : sequence number (low part)
* byte 4 : 2 = ask question or send info, 1 = answer or ACK, 0 = retransmit request
* byte 5 : direction, 1 = server to phone, 2 = phone to server arguments
*/
static const unsigned char packet_rcv_discovery[] =
{ 0xff, 0xff, 0xff, 0xff, 0x02, 0x02, 0xff, 0xff, 0xff, 0xff, 0x9e, 0x03, 0x08 };
static const unsigned char packet_send_discovery_ack[] =
{ 0x00, 0x00, /*Initial Seq (2 bytes) */ 0x00, 0x00, 0x00, 0x01 };
static const unsigned char packet_recv_firm_version[] =
{ 0x00, 0x00, 0x00, 0x13, 0x9a, 0x0a, 0x02 };
static const unsigned char packet_recv_pressed_key[] =
{ 0x00, 0x00, 0x00, 0x13, 0x99, 0x04, 0x00 };
static const unsigned char packet_recv_pick_up[] =
{ 0x00, 0x00, 0x00, 0x13, 0x99, 0x03, 0x04 };
static const unsigned char packet_recv_hangup[] =
{ 0x00, 0x00, 0x00, 0x13, 0x99, 0x03, 0x03 };
static const unsigned char packet_recv_r2[] = { 0x00, 0x00, 0x00, 0x13, 0x96, 0x03, 0x03 };
/*! TransportAdapter */
static const unsigned char packet_recv_resume_connection_with_server[] =
{ 0xff, 0xff, 0xff, 0xff, 0x9e, 0x03, 0x08 };
static const unsigned char packet_recv_mac_addr[] =
{ 0xff, 0xff, 0xff, 0xff, 0x9a, 0x0d, 0x07 /*MacAddr */ };
static const unsigned char packet_send_date_time3[] =
{ 0x11, 0x09, 0x02, 0x02, /*Month */ 0x05, /*Day */ 0x06, /*Hour */ 0x07,
/*Minutes */ 0x08, 0x32
};
static const unsigned char packet_send_date_time[] =
{ 0x11, 0x09, 0x02, 0x0a, /*Month */ 0x05, /*Day */ 0x06, /*Hour */ 0x07, /*Minutes */
0x08, 0x32, 0x17, 0x04, 0x24, 0x07, 0x19,
0x04, 0x07, 0x00, 0x19, 0x05, 0x09, 0x3e, 0x0f, 0x16, 0x05, 0x00, 0x80, 0x00, 0x1e,
0x05, 0x12, 0x00, 0x78
};
static const unsigned char packet_send_no_ring[] =
{ 0x16, 0x04, 0x1a, 0x00, 0x16, 0x04, 0x11, 0x00 };
static const unsigned char packet_send_s4[] =
{ 0x16, 0x04, 0x1a, 0x00, 0x16, 0x04, 0x11, 0x00, 0x16, 0x06, 0x32, 0xdf, 0x00, 0xff,
0x16, 0x05, 0x1c, 0x00, 0x00, 0x17, 0x05,
0x0b, 0x00, 0x00, 0x19, 0x04, 0x00, 0x00, 0x19, 0x04, 0x00, 0x08, 0x19, 0x04, 0x00,
0x10, 0x19, 0x04, 0x00, 0x18, 0x16, 0x05,
0x31, 0x00, 0x00, 0x16, 0x05, 0x04, 0x00, 0x00
};
static const unsigned char packet_send_call[] =
{ 0x16, 0x04, 0x1a, 0x00, 0x16, 0x04, 0x11, 0x00, 0x16, 0x06, 0x32, 0xdf,
0x00, 0xff, 0x16, 0x05, 0x1c, 0x00, 0x00, 0x16, 0x0a, 0x38, 0x00, 0x12, 0xca, 0x03,
0xc0, 0xc3, 0xc5, 0x16, 0x16, 0x30, 0x00,
0x00, /*codec */ 0x12, 0x12, /* frames per packet */ 0x01, 0x5c, 0x00, /*port RTP */
0x0f, 0xa0, /* port RTCP */ 0x9c, 0x41,
/*port RTP */ 0x0f, 0xa0, /* port RTCP */ 0x9c, 0x41, /* IP Address */ 0x0a, 0x01,
0x16, 0x66
};
static const unsigned char packet_send_stream_based_tone_off[] =
{ 0x16, 0x05, 0x1c, 0x00, 0x00 };
/* static const unsigned char packet_send_Mute[] = { 0x16, 0x05, 0x04, 0x00, 0x00 };
static const unsigned char packet_send_CloseAudioStreamRX[] = { 0x16, 0x05, 0x31, 0x00, 0xff };
static const unsigned char packet_send_CloseAudioStreamTX[] = { 0x16, 0x05, 0x31, 0xff, 0x00 };*/
static const unsigned char packet_send_stream_based_tone_on[] =
{ 0x16, 0x06, 0x1b, 0x00, 0x00, 0x05 };
static const unsigned char packet_send_stream_based_tone_single_freq[] =
{ 0x16, 0x06, 0x1d, 0x00, 0x01, 0xb8 };
static const unsigned char packet_send_stream_based_tone_dial_freq[] =
{ 0x16, 0x08, 0x1d, 0x00, 0x01, 0xb8, 0x01, 0x5e };
static const unsigned char packet_send_select_output[] =
{ 0x16, 0x06, 0x32, 0xc0, 0x01, 0x00 };
static const unsigned char packet_send_ring[] =
{ 0x16, 0x06, 0x32, 0xdf, 0x00, 0xff, 0x16, 0x05, 0x1c, 0x00, 0x00, 0x16,
0x04, 0x1a, 0x01, 0x16, 0x05, 0x12, 0x13 /* Ring type 10 to 17 */ , 0x18, 0x16, 0x04, 0x18, /* volume 00, 10, 20... */
0x20, 0x16, 0x04, 0x10, 0x00
};
static const unsigned char packet_send_end_call[] =
{ 0x16, 0x06, 0x32, 0xdf, 0x00, 0xff, 0x16, 0x05, 0x31, 0x00, 0x00, 0x19, 0x04, 0x00,
0x10, 0x19, 0x04, 0x00, 0x18, 0x16, 0x05,
0x04, 0x00, 0x00, 0x16, 0x04, 0x37, 0x10
};
static const unsigned char packet_send_s9[] =
{ 0x16, 0x06, 0x32, 0xdf, 0x00, 0xff, 0x19, 0x04, 0x00, 0x10, 0x16, 0x05, 0x1c, 0x00,
0x00 };
static const unsigned char packet_send_rtp_packet_size[] =
{ 0x16, 0x08, 0x38, 0x00, 0x00, 0xe0, 0x00, 0xa0 };
static const unsigned char packet_send_jitter_buffer_conf[] =
{ 0x16, 0x0e, 0x3a, 0x00, /* jitter */ 0x02, /* high water mark */ 0x04, 0x00, 0x00,
/* early packet resync 2 bytes */ 0x3e, 0x80,
0x00, 0x00, /* late packet resync 2 bytes */ 0x3e, 0x80
};
/* Duration in ms div 2 (0x20 = 64ms, 0x08 = 16ms)
static unsigned char packet_send_StreamBasedToneCad[] =
{ 0x16, 0x0a, 0x1e, 0x00, duration on 0x0a, duration off 0x0d, duration on 0x0a, duration off 0x0d, duration on 0x0a, duration off 0x2b }; */
static const unsigned char packet_send_open_audio_stream_rx[] =
{ 0x16, 0x1a, 0x30, 0x00, 0xff, /* Codec */ 0x00, 0x00, 0x01, 0x00, 0xb8, 0xb8, 0x0e,
0x0e, 0x01, /* Port */ 0x14, 0x50, 0x00,
0x00, /* Port */ 0x14, 0x50, 0x00, 0x00, /* Dest IP */ 0x0a, 0x93, 0x69, 0x05
};
static const unsigned char packet_send_open_audio_stream_tx[] =
{ 0x16, 0x1a, 0x30, 0xff, 0x00, 0x00, /* Codec */ 0x00, 0x01, 0x00, 0xb8, 0xb8, 0x0e,
0x0e, 0x01, /* Local port */ 0x14, 0x50,
0x00, 0x00, /* Rmt Port */ 0x14, 0x50, 0x00, 0x00, /* Dest IP */ 0x0a, 0x93, 0x69, 0x05
};
static const unsigned char packet_send_open_audio_stream_rx3[] =
{ 0x16, 0x1a, 0x30, 0x00, 0xff, /* Codec */ 0x00, 0x00, 0x02, 0x01, 0xb8, 0xb8, 0x06,
0x06, 0x81, /* RTP Port */ 0x14, 0x50,
/* RTCP Port */ 0x14,
0x51, /* RTP Port */ 0x14, 0x50, /* RTCP Port */ 0x00, 0x00, /* Dest IP */ 0x0a, 0x93,
0x69, 0x05
};
static const unsigned char packet_send_open_audio_stream_tx3[] =
{ 0x16, 0x1a, 0x30, 0xff, 0x00, 0x00, /* Codec */ 0x00, 0x02, 0x01, 0xb8, 0xb8, 0x06,
0x06, 0x81, /* RTP Local port */ 0x14, 0x50,
/* RTCP Port */ 0x00, 0x00, /* RTP Rmt Port */ 0x14, 0x50, /* RTCP Port */ 0x00, 0x00,
/* Dest IP */ 0x0a, 0x93, 0x69, 0x05
};
static const unsigned char packet_send_arrow[] = { 0x17, 0x04, 0x04, 0x00 };
static const unsigned char packet_send_blink_cursor[] = { 0x17, 0x04, 0x10, 0x86 };
static const unsigned char packet_send_date_time2[] = { 0x17, 0x04, 0x17, 0x3d, 0x11, 0x09, 0x02, 0x0a, /*Month */ 0x05, /*Day */
0x06, /*Hour */ 0x07, /*Minutes */ 0x08, 0x32
};
static const unsigned char packet_send_Contrast[] =
{ 0x17, 0x04, 0x24, /*Contrast */ 0x08 };
static const unsigned char packet_send_StartTimer[] =
{ 0x17, 0x05, 0x0b, 0x05, 0x00, 0x17, 0x08, 0x16, /* Text */ 0x44, 0x75, 0x72, 0xe9,
0x65 };
static const unsigned char packet_send_stop_timer[] = { 0x17, 0x05, 0x0b, 0x02, 0x00 };
static const unsigned char packet_send_icon[] = { 0x17, 0x05, 0x14, /*pos */ 0x00, /*icon */ 0x25 }; /* display an icon in front of the text zone */
static const unsigned char packet_send_S7[] = { 0x17, 0x06, 0x0f, 0x30, 0x07, 0x07 };
static const unsigned char packet_send_set_pos_cursor[] =
{ 0x17, 0x06, 0x10, 0x81, 0x04, /*pos */ 0x20 };
/*static unsigned char packet_send_MonthLabelsDownload[] =
{ 0x17, 0x0a, 0x15, Month (3 char) 0x46, 0x65, 0x62, 0x4d, 0xe4, 0x72, 0x20 }; */
static const unsigned char packet_send_favorite[] =
{ 0x17, 0x0f, 0x19, 0x10, /*pos */ 0x01, /*name */ 0x20, 0x20, 0x20, 0x20, 0x20, 0x20,
0x20, 0x20, 0x20, 0x20, /*end_name */ 0x19,
0x05, 0x0f, /*pos */ 0x01, /*icone */ 0x00
};
static const unsigned char packet_send_title[] =
{ 0x17, 0x10, 0x19, 0x02, /*text */ 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20,
0x20, 0x20, 0x20, 0x20 /*end_text */ };
static const unsigned char packet_send_text[] =
{ 0x17, 0x1e, 0x1b, 0x04, /*pos */ 0x00, /*inverse */ 0x25, /*text */ 0x20, 0x20,
0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20,
0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20,
/*end_text */ 0x17, 0x04, 0x10, 0x87
};
static const unsigned char packet_send_status[] =
{ 0x17, 0x20, 0x19, 0x08, /*text */ 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20,
0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20,
0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20, 0x20 /*end_text */
};
static const unsigned char packet_send_status2[] =
{ 0x17, 0x0b, 0x19, /* pos [08|28|48|68] */ 0x00, /* text */ 0x20, 0x20, 0x20, 0x20,
0x20, 0x20, 0x20 /* end_text */ };
static const unsigned char packet_send_led_update[] = { 0x19, 0x04, 0x00, 0x00 };
static const unsigned char packet_send_query_basic_manager_04[] = { 0x1a, 0x04, 0x01, 0x04 };
static const unsigned char packet_send_query_mac_address[] = { 0x1a, 0x04, 0x01, 0x08 };
static const unsigned char packet_send_query_basic_manager_10[] = { 0x1a, 0x04, 0x01, 0x10 };
static const unsigned char packet_send_S1[] = { 0x1a, 0x07, 0x07, 0x00, 0x00, 0x00, 0x13 };
static unsigned char packet_send_ping[] =
{ 0x1e, 0x05, 0x12, 0x00, /*Watchdog timer */ 0x78 };
#define BUFFSEND unsigned char buffsend[64] = { 0x00, 0x00, 0xaa, 0xbb, 0x02, 0x01 }
static const char tdesc[] = "UNISTIM Channel Driver";
static const char channel_type[] = "USTM";
/*! Protos */
static struct ast_channel *unistim_new(struct unistim_subchannel *sub, int state, const char *linkedid);
static int load_module(void);
static int reload(void);
static int unload_module(void);
static int reload_config(void);
static void show_main_page(struct unistimsession *pte);
static struct ast_channel *unistim_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor,
void *data, int *cause);
static int unistim_call(struct ast_channel *ast, char *dest, int timeout);
static int unistim_hangup(struct ast_channel *ast);
static int unistim_answer(struct ast_channel *ast);
static struct ast_frame *unistim_read(struct ast_channel *ast);
static int unistim_write(struct ast_channel *ast, struct ast_frame *frame);
static int unistim_indicate(struct ast_channel *ast, int ind, const void *data,
size_t datalen);
static int unistim_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static int unistim_senddigit_begin(struct ast_channel *ast, char digit);
static int unistim_senddigit_end(struct ast_channel *ast, char digit,
unsigned int duration);
static int unistim_sendtext(struct ast_channel *ast, const char *text);
static int write_entry_history(struct unistimsession *pte, FILE * f, char c,
char *line1);
static void change_callerid(struct unistimsession *pte, int type, char *callerid);
static struct ast_channel_tech unistim_tech = {
.type = channel_type,
.description = tdesc,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
.requester = unistim_request,
.call = unistim_call,
.hangup = unistim_hangup,
.answer = unistim_answer,
.read = unistim_read,
.write = unistim_write,
.indicate = unistim_indicate,
.fixup = unistim_fixup,
.send_digit_begin = unistim_senddigit_begin,
.send_digit_end = unistim_senddigit_end,
.send_text = unistim_sendtext,
.bridge = ast_rtp_instance_bridge,
};
static void display_last_error(const char *sz_msg)
{
time_t cur_time;
time(&cur_time);
/* Display the error message */
ast_log(LOG_WARNING, "%s %s : (%u) %s\n", ctime(&cur_time), sz_msg, errno,
strerror(errno));
}
static unsigned int get_tick_count(void)
{
struct timeval now = ast_tvnow();
return (now.tv_sec * 1000) + (now.tv_usec / 1000);
}
/* Send data to a phone without retransmit nor buffering */
static void send_raw_client(int size, const unsigned char *data, struct sockaddr_in *addr_to,
const struct sockaddr_in *addr_ourip)
{
#ifdef HAVE_PKTINFO
struct iovec msg_iov;
struct msghdr msg;
char buffer[CMSG_SPACE(sizeof(struct in_pktinfo))];
struct cmsghdr *ip_msg = (struct cmsghdr *) buffer;
struct in_pktinfo *pki = (struct in_pktinfo *) CMSG_DATA(ip_msg);
/* cast this to a non-const pointer, since the sendmsg() API
* does not provide read-only and write-only flavors of the
* structures used for its arguments, but in this case we know
* the data will not be modified
*/
msg_iov.iov_base = (char *) data;
msg_iov.iov_len = size;
msg.msg_name = addr_to; /* optional address */
msg.msg_namelen = sizeof(struct sockaddr_in); /* size of address */
msg.msg_iov = &msg_iov; /* scatter/gather array */
msg.msg_iovlen = 1; /* # elements in msg_iov */
msg.msg_control = ip_msg; /* ancillary data */
msg.msg_controllen = sizeof(buffer); /* ancillary data buffer len */
msg.msg_flags = 0; /* flags on received message */
ip_msg->cmsg_len = CMSG_LEN(sizeof(*pki));
ip_msg->cmsg_level = IPPROTO_IP;
ip_msg->cmsg_type = IP_PKTINFO;
pki->ipi_ifindex = 0; /* Interface index, 0 = use interface specified in routing table */
pki->ipi_spec_dst.s_addr = addr_ourip->sin_addr.s_addr; /* Local address */
/* pki->ipi_addr = ; Header Destination address - ignored by kernel */
#ifdef DUMP_PACKET
if (unistimdebug) {
int tmp;
char iabuf[INET_ADDRSTRLEN];
char iabuf2[INET_ADDRSTRLEN];
ast_verb(0, "\n**> From %s sending %d bytes to %s ***\n",
ast_inet_ntoa(addr_ourip->sin_addr), (int) size,
ast_inet_ntoa(addr_to->sin_addr));
for (tmp = 0; tmp < size; tmp++)
ast_verb(0, "%.2x ", (unsigned char) data[tmp]);
ast_verb(0, "\n******************************************\n");
}
#endif
if (sendmsg(unistimsock, &msg, 0) == -1)
display_last_error("Error sending datas");
#else
if (sendto(unistimsock, data, size, 0, (struct sockaddr *) addr_to, sizeof(*addr_to))
== -1)
display_last_error("Error sending datas");
#endif
}
static void send_client(int size, const unsigned char *data, struct unistimsession *pte)
{
unsigned int tick;
int buf_pos;
unsigned short *sdata = (unsigned short *) data;
ast_mutex_lock(&pte->lock);
buf_pos = pte->last_buf_available;
if (buf_pos >= MAX_BUF_NUMBER) {
ast_log(LOG_WARNING, "Error : send queue overflow\n");
ast_mutex_unlock(&pte->lock);
return;
}
sdata[1] = ntohs(++(pte->seq_server));
pte->wsabufsend[buf_pos].len = size;
memcpy(pte->wsabufsend[buf_pos].buf, data, size);
tick = get_tick_count();
pte->timeout = tick + RETRANSMIT_TIMER;
/*#ifdef DUMP_PACKET */
if (unistimdebug)
ast_verb(6, "Sending datas with seq #0x%.4x Using slot #%d :\n", pte->seq_server, buf_pos);
/*#endif */
send_raw_client(pte->wsabufsend[buf_pos].len, pte->wsabufsend[buf_pos].buf, &(pte->sin),
&(pte->sout));
pte->last_buf_available++;
ast_mutex_unlock(&pte->lock);
}
static void send_ping(struct unistimsession *pte)
{
BUFFSEND;
if (unistimdebug)
ast_verb(6, "Sending ping\n");
pte->tick_next_ping = get_tick_count() + unistim_keepalive;
memcpy(buffsend + SIZE_HEADER, packet_send_ping, sizeof(packet_send_ping));
send_client(SIZE_HEADER + sizeof(packet_send_ping), buffsend, pte);
}
static int get_to_address(int fd, struct sockaddr_in *toAddr)
{
#ifdef HAVE_PKTINFO
int err;
struct msghdr msg;
struct {
struct cmsghdr cm;
int len;
struct in_addr address;
} ip_msg;
/* Zero out the structures before we use them */
/* This sets several key values to NULL */
memset(&msg, 0, sizeof(msg));
memset(&ip_msg, 0, sizeof(ip_msg));
/* Initialize the message structure */
msg.msg_control = &ip_msg;
msg.msg_controllen = sizeof(ip_msg);
/* Get info about the incoming packet */
err = recvmsg(fd, &msg, MSG_PEEK);
if (err == -1)
ast_log(LOG_WARNING, "recvmsg returned an error: %s\n", strerror(errno));
memcpy(&toAddr->sin_addr, &ip_msg.address, sizeof(struct in_addr));
return err;
#else
memcpy(&toAddr, &public_ip, sizeof(&toAddr));
return 0;
#endif
}
/* Allocate memory & initialize structures for a new phone */
/* addr_from : ip address of the phone */
static struct unistimsession *create_client(const struct sockaddr_in *addr_from)
{
int tmp;
struct unistimsession *s;
if (!(s = ast_calloc(1, sizeof(*s))))
return NULL;
memcpy(&s->sin, addr_from, sizeof(struct sockaddr_in));
get_to_address(unistimsock, &s->sout);
if (unistimdebug) {
ast_verb(0, "Creating a new entry for the phone from %s received via server ip %s\n",
ast_inet_ntoa(addr_from->sin_addr), ast_inet_ntoa(s->sout.sin_addr));
}
ast_mutex_init(&s->lock);
ast_mutex_lock(&sessionlock);
s->next = sessions;
sessions = s;
s->timeout = get_tick_count() + RETRANSMIT_TIMER;
s->seq_phone = (short) 0x0000;
s->seq_server = (short) 0x0000;
s->last_seq_ack = (short) 0x000;
s->last_buf_available = 0;
s->nb_retransmit = 0;
s->state = STATE_INIT;
s->tick_next_ping = get_tick_count() + unistim_keepalive;
/* Initialize struct wsabuf */
for (tmp = 0; tmp < MAX_BUF_NUMBER; tmp++) {
s->wsabufsend[tmp].buf = s->buf[tmp];
}
ast_mutex_unlock(&sessionlock);
return s;
}
static void send_end_call(struct unistimsession *pte)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending end call\n");
memcpy(buffsend + SIZE_HEADER, packet_send_end_call, sizeof(packet_send_end_call));
send_client(SIZE_HEADER + sizeof(packet_send_end_call), buffsend, pte);
}
static void set_ping_timer(struct unistimsession *pte)
{
unsigned int tick = 0; /* XXX what is this for, anyways */
pte->timeout = pte->tick_next_ping;
DEBUG_TIMER("tick = %u next ping at %u tick\n", tick, pte->timeout);
return;
}
/* Checking if our send queue is empty,
* if true, setting up a timer for keepalive */
static void check_send_queue(struct unistimsession *pte)
{
/* Check if our send queue contained only one element */
if (pte->last_buf_available == 1) {
if (unistimdebug)
ast_verb(6, "Our single packet was ACKed.\n");
pte->last_buf_available--;
set_ping_timer(pte);
return;
}
/* Check if this ACK catch up our latest packet */
else if (pte->last_seq_ack + 1 == pte->seq_server + 1) {
if (unistimdebug)
ast_verb(6, "Our send queue is completely ACKed.\n");
pte->last_buf_available = 0; /* Purge the send queue */
set_ping_timer(pte);
return;
}
if (unistimdebug)
ast_verb(6, "We still have packets in our send queue\n");
return;
}
static void send_start_timer(struct unistimsession *pte)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending start timer\n");
memcpy(buffsend + SIZE_HEADER, packet_send_StartTimer, sizeof(packet_send_StartTimer));
send_client(SIZE_HEADER + sizeof(packet_send_StartTimer), buffsend, pte);
}
static void send_stop_timer(struct unistimsession *pte)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending stop timer\n");
memcpy(buffsend + SIZE_HEADER, packet_send_stop_timer, sizeof(packet_send_stop_timer));
send_client(SIZE_HEADER + sizeof(packet_send_stop_timer), buffsend, pte);
}
static void Sendicon(unsigned char pos, unsigned char status, struct unistimsession *pte)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending icon pos %d with status 0x%.2x\n", pos, status);
memcpy(buffsend + SIZE_HEADER, packet_send_icon, sizeof(packet_send_icon));
buffsend[9] = pos;
buffsend[10] = status;
send_client(SIZE_HEADER + sizeof(packet_send_icon), buffsend, pte);
}
static void send_tone(struct unistimsession *pte, uint16_t tone1, uint16_t tone2)
{
BUFFSEND;
if (!tone1) {
if (unistimdebug)
ast_verb(0, "Sending Stream Based Tone Off\n");
memcpy(buffsend + SIZE_HEADER, packet_send_stream_based_tone_off,
sizeof(packet_send_stream_based_tone_off));
send_client(SIZE_HEADER + sizeof(packet_send_stream_based_tone_off), buffsend, pte);
return;
}
/* Since most of the world use a continuous tone, it's useless
if (unistimdebug)
ast_verb(0, "Sending Stream Based Tone Cadence Download\n");
memcpy (buffsend + SIZE_HEADER, packet_send_StreamBasedToneCad, sizeof (packet_send_StreamBasedToneCad));
send_client (SIZE_HEADER + sizeof (packet_send_StreamBasedToneCad), buffsend, pte); */
if (unistimdebug)
ast_verb(0, "Sending Stream Based Tone Frequency Component List Download %d %d\n", tone1, tone2);
tone1 *= 8;
if (!tone2) {
memcpy(buffsend + SIZE_HEADER, packet_send_stream_based_tone_single_freq,
sizeof(packet_send_stream_based_tone_single_freq));
buffsend[10] = (tone1 & 0xff00) >> 8;
buffsend[11] = (tone1 & 0x00ff);
send_client(SIZE_HEADER + sizeof(packet_send_stream_based_tone_single_freq), buffsend,
pte);
} else {
tone2 *= 8;
memcpy(buffsend + SIZE_HEADER, packet_send_stream_based_tone_dial_freq,
sizeof(packet_send_stream_based_tone_dial_freq));
buffsend[10] = (tone1 & 0xff00) >> 8;
buffsend[11] = (tone1 & 0x00ff);
buffsend[12] = (tone2 & 0xff00) >> 8;
buffsend[13] = (tone2 & 0x00ff);
send_client(SIZE_HEADER + sizeof(packet_send_stream_based_tone_dial_freq), buffsend,
pte);
}
if (unistimdebug)
ast_verb(0, "Sending Stream Based Tone On\n");
memcpy(buffsend + SIZE_HEADER, packet_send_stream_based_tone_on,
sizeof(packet_send_stream_based_tone_on));
send_client(SIZE_HEADER + sizeof(packet_send_stream_based_tone_on), buffsend, pte);
}
/* Positions for favorites
|--------------------|
| 5 2 |
| 4 1 |
| 3 0 |
*/
/* status (icons) : 00 = nothing, 2x/3x = see parser.h, 4x/5x = blink fast, 6x/7x = blink slow */
static void
send_favorite(unsigned char pos, unsigned char status, struct unistimsession *pte,
const char *text)
{
BUFFSEND;
int i;
if (unistimdebug)
ast_verb(0, "Sending favorite pos %d with status 0x%.2x\n", pos, status);
memcpy(buffsend + SIZE_HEADER, packet_send_favorite, sizeof(packet_send_favorite));
buffsend[10] = pos;
buffsend[24] = pos;
buffsend[25] = status;
i = strlen(text);
if (i > FAV_MAX_LENGTH)
i = FAV_MAX_LENGTH;
memcpy(buffsend + FAV_MAX_LENGTH + 1, text, i);
send_client(SIZE_HEADER + sizeof(packet_send_favorite), buffsend, pte);
}
static void refresh_all_favorite(struct unistimsession *pte)
{
int i = 0;
if (unistimdebug)
ast_verb(0, "Refreshing all favorite\n");
for (i = 0; i < 6; i++) {
if ((pte->device->softkeyicon[i] <= FAV_ICON_HEADPHONES_ONHOLD) &&
(pte->device->softkeylinepos != i))
send_favorite((unsigned char) i, pte->device->softkeyicon[i] + 1, pte,
pte->device->softkeylabel[i]);
else
send_favorite((unsigned char) i, pte->device->softkeyicon[i], pte,
pte->device->softkeylabel[i]);
}
}
/* Change the status for this phone (pte) and update for each phones where pte is bookmarked
* use FAV_ICON_*_BLACK constant in status parameters */
static void change_favorite_icon(struct unistimsession *pte, unsigned char status)
{
struct unistim_device *d = devices;
int i;
/* Update the current phone */
if (pte->state != STATE_CLEANING)
send_favorite(pte->device->softkeylinepos, status, pte,
pte->device->softkeylabel[pte->device->softkeylinepos]);
/* Notify other phones if we're in their bookmark */
while (d) {
for (i = 0; i < 6; i++) {
if (d->sp[i] == pte->device) { /* It's us ? */
if (d->softkeyicon[i] != status) { /* Avoid resending the same icon */
d->softkeyicon[i] = status;
if (d->session)
send_favorite(i, status + 1, d->session, d->softkeylabel[i]);
}
}
}
d = d->next;
}
}
static int RegisterExtension(const struct unistimsession *pte)
{
if (unistimdebug)
ast_verb(0, "Trying to register extension '%s' into context '%s' to %s\n",
pte->device->extension_number, pte->device->lines->context,
pte->device->lines->fullname);
return ast_add_extension(pte->device->lines->context, 0,
pte->device->extension_number, 1, NULL, NULL, "Dial",
pte->device->lines->fullname, 0, "Unistim");
}
static int UnregisterExtension(const struct unistimsession *pte)
{
if (unistimdebug)
ast_verb(0, "Trying to unregister extension '%s' context '%s'\n",
pte->device->extension_number, pte->device->lines->context);
return ast_context_remove_extension(pte->device->lines->context,
pte->device->extension_number, 1, "Unistim");
}
/* Free memory allocated for a phone */
static void close_client(struct unistimsession *s)
{
struct unistim_subchannel *sub;
struct unistimsession *cur, *prev = NULL;
ast_mutex_lock(&sessionlock);
cur = sessions;
/* Looking for the session in the linked chain */
while (cur) {
if (cur == s)
break;
prev = cur;
cur = cur->next;
}
if (cur) { /* Session found ? */
if (cur->device) { /* This session was registered ? */
s->state = STATE_CLEANING;
if (unistimdebug)
ast_verb(0, "close_client session %p device %p lines %p sub %p\n",
s, s->device, s->device->lines,
s->device->lines->subs[SUB_REAL]);
change_favorite_icon(s, FAV_ICON_NONE);
sub = s->device->lines->subs[SUB_REAL];
if (sub) {
if (sub->owner) { /* Call in progress ? */
if (unistimdebug)
ast_verb(0, "Aborting call\n");
ast_queue_hangup_with_cause(sub->owner, AST_CAUSE_NETWORK_OUT_OF_ORDER);
}
} else
ast_log(LOG_WARNING, "Freeing a client with no subchannel !\n");
if (!ast_strlen_zero(s->device->extension_number))
UnregisterExtension(s);
cur->device->session = NULL;
} else {
if (unistimdebug)
ast_verb(0, "Freeing an unregistered client\n");
}
if (prev)
prev->next = cur->next;
else
sessions = cur->next;
ast_mutex_destroy(&s->lock);
ast_free(s);
} else
ast_log(LOG_WARNING, "Trying to delete non-existent session %p?\n", s);
ast_mutex_unlock(&sessionlock);
return;
}
/* Return 1 if the session chained link was modified */
static int send_retransmit(struct unistimsession *pte)
{
int i;
ast_mutex_lock(&pte->lock);
if (++pte->nb_retransmit >= NB_MAX_RETRANSMIT) {
if (unistimdebug)
ast_verb(0, "Too many retransmit - freeing client\n");
ast_mutex_unlock(&pte->lock);
close_client(pte);
return 1;
}
pte->timeout = get_tick_count() + RETRANSMIT_TIMER;
for (i = pte->last_buf_available - (pte->seq_server - pte->last_seq_ack);
i < pte->last_buf_available; i++) {
if (i < 0) {
ast_log(LOG_WARNING,
"Asked to retransmit an ACKed slot ! last_buf_available=%d, seq_server = #0x%.4x last_seq_ack = #0x%.4x\n",
pte->last_buf_available, pte->seq_server, pte->last_seq_ack);
continue;
}
if (unistimdebug) {
unsigned short *sbuf = (unsigned short *) pte->wsabufsend[i].buf;
unsigned short seq;
seq = ntohs(sbuf[1]);
ast_verb(0, "Retransmit slot #%d (seq=#0x%.4x), last ack was #0x%.4x\n", i,
seq, pte->last_seq_ack);
}
send_raw_client(pte->wsabufsend[i].len, pte->wsabufsend[i].buf, &pte->sin,
&pte->sout);
}
ast_mutex_unlock(&pte->lock);
return 0;
}
/* inverse : TEXT_INVERSE : yes, TEXT_NORMAL : no */
static void
send_text(unsigned char pos, unsigned char inverse, struct unistimsession *pte,
const char *text)
{
int i;
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending text at pos %d, inverse flag %d\n", pos, inverse);
memcpy(buffsend + SIZE_HEADER, packet_send_text, sizeof(packet_send_text));
buffsend[10] = pos;
buffsend[11] = inverse;
i = strlen(text);
if (i > TEXT_LENGTH_MAX)
i = TEXT_LENGTH_MAX;
memcpy(buffsend + 12, text, i);
send_client(SIZE_HEADER + sizeof(packet_send_text), buffsend, pte);
}
static void send_text_status(struct unistimsession *pte, const char *text)
{
BUFFSEND;
int i;
if (unistimdebug)
ast_verb(0, "Sending status text\n");
if (pte->device) {
if (pte->device->status_method == 1) { /* For new firmware and i2050 soft phone */
int n = strlen(text);
/* Must send individual button separately */
int j;
for (i = 0, j = 0; i < 4; i++, j += 7) {
int pos = 0x08 + (i * 0x20);
memcpy(buffsend + SIZE_HEADER, packet_send_status2,
sizeof(packet_send_status2));
buffsend[9] = pos;
memcpy(buffsend + 10, (j < n) ? (text + j) : " ", 7);
send_client(SIZE_HEADER + sizeof(packet_send_status2), buffsend, pte);
}
return;
}
}
memcpy(buffsend + SIZE_HEADER, packet_send_status, sizeof(packet_send_status));
i = strlen(text);
if (i > STATUS_LENGTH_MAX)
i = STATUS_LENGTH_MAX;
memcpy(buffsend + 10, text, i);
send_client(SIZE_HEADER + sizeof(packet_send_status), buffsend, pte);
}
/* led values in hexa : 0 = bar off, 1 = bar on, 2 = bar 1s on/1s off, 3 = bar 2.5s on/0.5s off
* 4 = bar 0.6s on/0.3s off, 5 = bar 0.5s on/0.5s off, 6 = bar 2s on/0.5s off
* 7 = bar off, 8 = speaker off, 9 = speaker on, 10 = headphone off, 11 = headphone on
* 18 = mute off, 19 mute on */
static void send_led_update(struct unistimsession *pte, unsigned char led)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending led_update (%x)\n", led);
memcpy(buffsend + SIZE_HEADER, packet_send_led_update, sizeof(packet_send_led_update));
buffsend[9] = led;
send_client(SIZE_HEADER + sizeof(packet_send_led_update), buffsend, pte);
}
/* output = OUTPUT_HANDSET, OUTPUT_HEADPHONE or OUTPUT_SPEAKER
* volume = VOLUME_LOW, VOLUME_NORMAL, VOLUME_INSANELY_LOUD
* mute = MUTE_OFF, MUTE_ON */
static void
send_select_output(struct unistimsession *pte, unsigned char output, unsigned char volume,
unsigned char mute)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending select output packet output=%x volume=%x mute=%x\n", output,
volume, mute);
memcpy(buffsend + SIZE_HEADER, packet_send_select_output,
sizeof(packet_send_select_output));
buffsend[9] = output;
if (output == OUTPUT_SPEAKER)
volume = VOLUME_LOW_SPEAKER;
else
volume = VOLUME_LOW;
buffsend[10] = volume;
if (mute == MUTE_ON_DISCRET)
buffsend[11] = MUTE_ON;
else
buffsend[11] = mute;
send_client(SIZE_HEADER + sizeof(packet_send_select_output), buffsend, pte);
if (mute == MUTE_OFF)
send_led_update(pte, 0x18);
else if (mute == MUTE_ON)
send_led_update(pte, 0x19);
pte->device->mute = mute;
if (output == OUTPUT_HANDSET) {
if (mute == MUTE_ON)
change_favorite_icon(pte, FAV_ICON_ONHOLD_BLACK);
else
change_favorite_icon(pte, FAV_ICON_OFFHOOK_BLACK);
send_led_update(pte, 0x08);
send_led_update(pte, 0x10);
} else if (output == OUTPUT_HEADPHONE) {
if (mute == MUTE_ON)
change_favorite_icon(pte, FAV_ICON_HEADPHONES_ONHOLD);
else
change_favorite_icon(pte, FAV_ICON_HEADPHONES);
send_led_update(pte, 0x08);
send_led_update(pte, 0x11);
} else if (output == OUTPUT_SPEAKER) {
send_led_update(pte, 0x10);
send_led_update(pte, 0x09);
if (pte->device->receiver_state == STATE_OFFHOOK) {
if (mute == MUTE_ON)
change_favorite_icon(pte, FAV_ICON_SPEAKER_ONHOLD_BLACK);
else
change_favorite_icon(pte, FAV_ICON_SPEAKER_ONHOOK_BLACK);
} else {
if (mute == MUTE_ON)
change_favorite_icon(pte, FAV_ICON_SPEAKER_ONHOLD_BLACK);
else
change_favorite_icon(pte, FAV_ICON_SPEAKER_OFFHOOK_BLACK);
}
} else
ast_log(LOG_WARNING, "Invalid output (%d)\n", output);
if (output != pte->device->output)
pte->device->previous_output = pte->device->output;
pte->device->output = output;
}
static void send_ring(struct unistimsession *pte, char volume, char style)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending ring packet\n");
memcpy(buffsend + SIZE_HEADER, packet_send_ring, sizeof(packet_send_ring));
buffsend[24] = style + 0x10;
buffsend[29] = volume * 0x10;
send_client(SIZE_HEADER + sizeof(packet_send_ring), buffsend, pte);
}
static void send_no_ring(struct unistimsession *pte)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending no ring packet\n");
memcpy(buffsend + SIZE_HEADER, packet_send_no_ring, sizeof(packet_send_no_ring));
send_client(SIZE_HEADER + sizeof(packet_send_no_ring), buffsend, pte);
}
static void send_texttitle(struct unistimsession *pte, const char *text)
{
BUFFSEND;
int i;
if (unistimdebug)
ast_verb(0, "Sending title text\n");
memcpy(buffsend + SIZE_HEADER, packet_send_title, sizeof(packet_send_title));
i = strlen(text);
if (i > 12)
i = 12;
memcpy(buffsend + 10, text, i);
send_client(SIZE_HEADER + sizeof(packet_send_title), buffsend, pte);
}
static void send_date_time(struct unistimsession *pte)
{
BUFFSEND;
struct timeval now = ast_tvnow();
struct ast_tm atm = { 0, };
if (unistimdebug)
ast_verb(0, "Sending Time & Date\n");
memcpy(buffsend + SIZE_HEADER, packet_send_date_time, sizeof(packet_send_date_time));
ast_localtime(&now, &atm, NULL);
buffsend[10] = (unsigned char) atm.tm_mon + 1;
buffsend[11] = (unsigned char) atm.tm_mday;
buffsend[12] = (unsigned char) atm.tm_hour;
buffsend[13] = (unsigned char) atm.tm_min;
send_client(SIZE_HEADER + sizeof(packet_send_date_time), buffsend, pte);
}
static void send_date_time2(struct unistimsession *pte)
{
BUFFSEND;
struct timeval now = ast_tvnow();
struct ast_tm atm = { 0, };
if (unistimdebug)
ast_verb(0, "Sending Time & Date #2\n");
memcpy(buffsend + SIZE_HEADER, packet_send_date_time2, sizeof(packet_send_date_time2));
ast_localtime(&now, &atm, NULL);
if (pte->device)
buffsend[9] = pte->device->datetimeformat;
else
buffsend[9] = 61;
buffsend[14] = (unsigned char) atm.tm_mon + 1;
buffsend[15] = (unsigned char) atm.tm_mday;
buffsend[16] = (unsigned char) atm.tm_hour;
buffsend[17] = (unsigned char) atm.tm_min;
send_client(SIZE_HEADER + sizeof(packet_send_date_time2), buffsend, pte);
}
static void send_date_time3(struct unistimsession *pte)
{
BUFFSEND;
struct timeval now = ast_tvnow();
struct ast_tm atm = { 0, };
if (unistimdebug)
ast_verb(0, "Sending Time & Date #3\n");
memcpy(buffsend + SIZE_HEADER, packet_send_date_time3, sizeof(packet_send_date_time3));
ast_localtime(&now, &atm, NULL);
buffsend[10] = (unsigned char) atm.tm_mon + 1;
buffsend[11] = (unsigned char) atm.tm_mday;
buffsend[12] = (unsigned char) atm.tm_hour;
buffsend[13] = (unsigned char) atm.tm_min;
send_client(SIZE_HEADER + sizeof(packet_send_date_time3), buffsend, pte);
}
static void send_blink_cursor(struct unistimsession *pte)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending set blink\n");
memcpy(buffsend + SIZE_HEADER, packet_send_blink_cursor, sizeof(packet_send_blink_cursor));
send_client(SIZE_HEADER + sizeof(packet_send_blink_cursor), buffsend, pte);
return;
}
/* pos : 0xab (a=0/2/4 = line ; b = row) */
static void send_cursor_pos(struct unistimsession *pte, unsigned char pos)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending set cursor position\n");
memcpy(buffsend + SIZE_HEADER, packet_send_set_pos_cursor,
sizeof(packet_send_set_pos_cursor));
buffsend[11] = pos;
send_client(SIZE_HEADER + sizeof(packet_send_set_pos_cursor), buffsend, pte);
return;
}
static void rcv_resume_connection_with_server(struct unistimsession *pte)
{
BUFFSEND;
if (unistimdebug) {
ast_verb(0, "ResumeConnectionWithServer received\n");
ast_verb(0, "Sending packet_send_query_mac_address\n");
}
memcpy(buffsend + SIZE_HEADER, packet_send_query_mac_address,
sizeof(packet_send_query_mac_address));
send_client(SIZE_HEADER + sizeof(packet_send_query_mac_address), buffsend, pte);
return;
}
static int unistim_register(struct unistimsession *s)
{
struct unistim_device *d;
ast_mutex_lock(&devicelock);
d = devices;
while (d) {
if (!strcasecmp(s->macaddr, d->id)) {
/* XXX Deal with IP authentication */
s->device = d;
d->session = s;
d->codec_number = DEFAULT_CODEC;
d->pos_fav = 0;
d->missed_call = 0;
d->receiver_state = STATE_ONHOOK;
break;
}
d = d->next;
}
ast_mutex_unlock(&devicelock);
if (!d)
return 0;
return 1;
}
static void unistim_line_copy(struct unistim_line *dst, struct unistim_line *src)
{
struct ast_format_cap *tmp = src->cap;
memcpy(dst, src, sizeof(*dst)); /* this over writes the cap ptr, so we have to reset it */
src->cap = tmp;
ast_format_cap_copy(src->cap, dst->cap);
}
static struct unistim_line *unistim_line_destroy(struct unistim_line *l)
{
if (!l) {
return NULL;
}
l->cap = ast_format_cap_destroy(l->cap);
ast_free(l);
return NULL;
}
static struct unistim_line *unistim_line_alloc(void)
{
struct unistim_line *l;
if (!(l = ast_calloc(1, sizeof(*l)))) {
return NULL;
}
if (!(l->cap = ast_format_cap_alloc_nolock())) {
ast_free(l);
return NULL;
}
return l;
}
static int alloc_sub(struct unistim_line *l, int x)
{
struct unistim_subchannel *sub;
if (!(sub = ast_calloc(1, sizeof(*sub))))
return 0;
if (unistimdebug)
ast_verb(3, "Allocating UNISTIM subchannel #%d on %s@%s ptr=%p\n", x, l->name, l->parent->name, sub);
sub->parent = l;
sub->subtype = x;
l->subs[x] = sub;
ast_mutex_init(&sub->lock);
return 1;
}
static int unalloc_sub(struct unistim_line *p, int x)
{
if (!x) {
ast_log(LOG_WARNING, "Trying to unalloc the real channel %s@%s?!?\n", p->name,
p->parent->name);
return -1;
}
if (unistimdebug)
ast_debug(1, "Released sub %d of channel %s@%s\n", x, p->name,
p->parent->name);
ast_mutex_destroy(&p->lock);
ast_free(p->subs[x]);
p->subs[x] = 0;
return 0;
}
static void rcv_mac_addr(struct unistimsession *pte, const unsigned char *buf)
{
BUFFSEND;
int tmp, i = 0;
char addrmac[19];
int res = 0;
if (unistimdebug)
ast_verb(0, "Mac Address received : ");
for (tmp = 15; tmp < 15 + SIZE_HEADER; tmp++) {
sprintf(&addrmac[i], "%.2x", (unsigned char) buf[tmp]);
i += 2;
}
if (unistimdebug)
ast_verb(0, "%s\n", addrmac);
strcpy(pte->macaddr, addrmac);
res = unistim_register(pte);
if (!res) {
switch (autoprovisioning) {
case AUTOPROVISIONING_NO:
ast_log(LOG_WARNING, "No entry found for this phone : %s\n", addrmac);
pte->state = STATE_AUTHDENY;
break;
case AUTOPROVISIONING_YES:
{
struct unistim_device *d, *newd;
struct unistim_line *newl;
if (unistimdebug)
ast_verb(0, "New phone, autoprovisioning on\n");
/* First : locate the [template] section */
ast_mutex_lock(&devicelock);
d = devices;
while (d) {
if (!strcasecmp(d->name, "template")) {
/* Found, cloning this entry */
if (!(newd = ast_malloc(sizeof(*newd)))) {
ast_mutex_unlock(&devicelock);
return;
}
memcpy(newd, d, sizeof(*newd));
if (!(newl = unistim_line_alloc())) {
ast_free(newd);
ast_mutex_unlock(&devicelock);
return;
}
unistim_line_copy(d->lines, newl);
if (!alloc_sub(newl, SUB_REAL)) {
ast_free(newd);
unistim_line_destroy(newl);
ast_mutex_unlock(&devicelock);
return;
}
/* Ok, now updating some fields */
ast_copy_string(newd->id, addrmac, sizeof(newd->id));
ast_copy_string(newd->name, addrmac, sizeof(newd->name));
if (newd->extension == EXTENSION_NONE)
newd->extension = EXTENSION_ASK;
newd->lines = newl;
newd->receiver_state = STATE_ONHOOK;
newd->session = pte;
newd->to_delete = -1;
pte->device = newd;
newd->next = NULL;
newl->parent = newd;
strcpy(newl->name, d->lines->name);
snprintf(d->lines->name, sizeof(d->lines->name), "%d",
atoi(d->lines->name) + 1);
snprintf(newl->fullname, sizeof(newl->fullname), "USTM/%s@%s",
newl->name, newd->name);
/* Go to the end of the linked chain */
while (d->next) {
d = d->next;
}
d->next = newd;
d = newd;
break;
}
d = d->next;
}
ast_mutex_unlock(&devicelock);
if (!d) {
ast_log(LOG_WARNING, "No entry [template] found in unistim.conf\n");
pte->state = STATE_AUTHDENY;
}
}
break;
case AUTOPROVISIONING_TN:
pte->state = STATE_AUTHDENY;
break;
case AUTOPROVISIONING_DB:
ast_log(LOG_WARNING,
"Autoprovisioning with database is not yet functional\n");
break;
default:
ast_log(LOG_WARNING, "Internal error : unknown autoprovisioning value = %d\n",
autoprovisioning);
}
}
if (pte->state != STATE_AUTHDENY) {
ast_verb(3, "Device '%s' successfuly registered\n", pte->device->name);
switch (pte->device->extension) {
case EXTENSION_NONE:
pte->state = STATE_MAINPAGE;
break;
case EXTENSION_ASK:
/* Checking if we already have an extension number */
if (ast_strlen_zero(pte->device->extension_number))
pte->state = STATE_EXTENSION;
else {
/* Yes, because of a phone reboot. We don't ask again for the TN */
if (RegisterExtension(pte))
pte->state = STATE_EXTENSION;
else
pte->state = STATE_MAINPAGE;
}
break;
case EXTENSION_LINE:
ast_copy_string(pte->device->extension_number, pte->device->lines->name,
sizeof(pte->device->extension_number));
if (RegisterExtension(pte))
pte->state = STATE_EXTENSION;
else
pte->state = STATE_MAINPAGE;
break;
case EXTENSION_TN:
/* If we are here, it's because of a phone reboot */
pte->state = STATE_MAINPAGE;
break;
default:
ast_log(LOG_WARNING, "Internal error, extension value unknown : %d\n",
pte->device->extension);
pte->state = STATE_AUTHDENY;
break;
}
}
if (pte->state == STATE_EXTENSION) {
if (pte->device->extension != EXTENSION_TN)
pte->device->extension = EXTENSION_ASK;
pte->device->extension_number[0] = '\0';
}
if (unistimdebug)
ast_verb(0, "\nSending S1\n");
memcpy(buffsend + SIZE_HEADER, packet_send_S1, sizeof(packet_send_S1));
send_client(SIZE_HEADER + sizeof(packet_send_S1), buffsend, pte);
if (unistimdebug)
ast_verb(0, "Sending query_basic_manager_04\n");
memcpy(buffsend + SIZE_HEADER, packet_send_query_basic_manager_04,
sizeof(packet_send_query_basic_manager_04));
send_client(SIZE_HEADER + sizeof(packet_send_query_basic_manager_04), buffsend, pte);
if (unistimdebug)
ast_verb(0, "Sending query_basic_manager_10\n");
memcpy(buffsend + SIZE_HEADER, packet_send_query_basic_manager_10,
sizeof(packet_send_query_basic_manager_10));
send_client(SIZE_HEADER + sizeof(packet_send_query_basic_manager_10), buffsend, pte);
send_date_time(pte);
return;
}
static int write_entry_history(struct unistimsession *pte, FILE * f, char c, char *line1)
{
if (fwrite(&c, 1, 1, f) != 1) {
display_last_error("Unable to write history log header.");
return -1;
}
if (fwrite(line1, TEXT_LENGTH_MAX, 1, f) != 1) {
display_last_error("Unable to write history entry - date.");
return -1;
}
if (fwrite(pte->device->lst_cid, TEXT_LENGTH_MAX, 1, f) != 1) {
display_last_error("Unable to write history entry - callerid.");
return -1;
}
if (fwrite(pte->device->lst_cnm, TEXT_LENGTH_MAX, 1, f) != 1) {
display_last_error("Unable to write history entry - callername.");
return -1;
}
return 0;
}
static int write_history(struct unistimsession *pte, char way, char ismissed)
{
char tmp[AST_CONFIG_MAX_PATH], tmp2[AST_CONFIG_MAX_PATH];
char line1[TEXT_LENGTH_MAX + 1];
char count = 0, *histbuf;
int size;
FILE *f, *f2;
struct timeval now = ast_tvnow();
struct ast_tm atm = { 0, };
if (!pte->device)
return -1;
if (!pte->device->callhistory)
return 0;
if (strchr(pte->device->name, '/') || (pte->device->name[0] == '.')) {
ast_log(LOG_WARNING, "Account code '%s' insecure for writing file\n",
pte->device->name);
return -1;
}
snprintf(tmp, sizeof(tmp), "%s/%s", ast_config_AST_LOG_DIR, USTM_LOG_DIR);
if (ast_mkdir(tmp, 0770)) {
if (errno != EEXIST) {
display_last_error("Unable to create directory for history");
return -1;
}
}
ast_localtime(&now, &atm, NULL);
if (ismissed) {
if (way == 'i')
strcpy(tmp2, "Miss");
else
strcpy(tmp2, "Fail");
} else
strcpy(tmp2, "Answ");
snprintf(line1, sizeof(line1), "%04d/%02d/%02d %02d:%02d:%02d %s",
atm.tm_year + 1900, atm.tm_mon + 1, atm.tm_mday, atm.tm_hour,
atm.tm_min, atm.tm_sec, tmp2);
snprintf(tmp, sizeof(tmp), "%s/%s/%s-%c.csv", ast_config_AST_LOG_DIR,
USTM_LOG_DIR, pte->device->name, way);
if ((f = fopen(tmp, "r"))) {
struct stat bufstat;
if (stat(tmp, &bufstat)) {
display_last_error("Unable to stat history log.");
fclose(f);
return -1;
}
size = 1 + (MAX_ENTRY_LOG * TEXT_LENGTH_MAX * 3);
if (bufstat.st_size != size) {
ast_log(LOG_WARNING,
"History file %s has an incorrect size (%d instead of %d). It will be replaced by a new one.",
tmp, (int) bufstat.st_size, size);
fclose(f);
f = NULL;
count = 1;
}
}
/* If we can't open the log file, we create a brand new one */
if (!f) {
char c = 1;
int i;
if ((errno != ENOENT) && (count == 0)) {
display_last_error("Unable to open history log.");
return -1;
}
f = fopen(tmp, "w");
if (!f) {
display_last_error("Unable to create history log.");
return -1;
}
if (write_entry_history(pte, f, c, line1)) {
fclose(f);
return -1;
}
memset(line1, ' ', TEXT_LENGTH_MAX);
for (i = 3; i < MAX_ENTRY_LOG * 3; i++) {
if (fwrite(line1, TEXT_LENGTH_MAX, 1, f) != 1) {
display_last_error("Unable to write history entry - stuffing.");
fclose(f);
return -1;
}
}
if (fclose(f))
display_last_error("Unable to close history - creation.");
return 0;
}
/* We can open the log file, we create a temporary one, we add our entry and copy the rest */
if (fread(&count, 1, 1, f) != 1) {
display_last_error("Unable to read history header.");
fclose(f);
return -1;
}
if (count > MAX_ENTRY_LOG) {
ast_log(LOG_WARNING, "Invalid count in history header of %s (%d max %d)\n", tmp,
count, MAX_ENTRY_LOG);
fclose(f);
return -1;
}
snprintf(tmp2, sizeof(tmp2), "%s/%s/%s-%c.csv.tmp", ast_config_AST_LOG_DIR,
USTM_LOG_DIR, pte->device->name, way);
if (!(f2 = fopen(tmp2, "w"))) {
display_last_error("Unable to create temporary history log.");
fclose(f);
return -1;
}
if (++count > MAX_ENTRY_LOG)
count = MAX_ENTRY_LOG;
if (write_entry_history(pte, f2, count, line1)) {
fclose(f);
fclose(f2);
return -1;
}
size = (MAX_ENTRY_LOG - 1) * TEXT_LENGTH_MAX * 3;
if (!(histbuf = ast_malloc(size))) {
fclose(f);
fclose(f2);
return -1;
}
if (fread(histbuf, size, 1, f) != 1) {
ast_free(histbuf);
fclose(f);
fclose(f2);
display_last_error("Unable to read previous history entries.");
return -1;
}
if (fwrite(histbuf, size, 1, f2) != 1) {
ast_free(histbuf);
fclose(f);
fclose(f2);
display_last_error("Unable to write previous history entries.");
return -1;
}
ast_free(histbuf);
if (fclose(f))
display_last_error("Unable to close history log.");
if (fclose(f2))
display_last_error("Unable to close temporary history log.");
if (unlink(tmp))
display_last_error("Unable to remove old history log.");
if (rename(tmp2, tmp))
display_last_error("Unable to rename new history log.");
return 0;
}
static void cancel_dial(struct unistimsession *pte)
{
send_no_ring(pte);
pte->device->missed_call++;
write_history(pte, 'i', 1);
show_main_page(pte);
return;
}
static void swap_subs(struct unistim_line *p, int a, int b)
{
/* struct ast_channel *towner; */
struct ast_rtp_instance *rtp;
int fds;
if (unistimdebug)
ast_verb(0, "Swapping %d and %d\n", a, b);
if ((!p->subs[a]->owner) || (!p->subs[b]->owner)) {
ast_log(LOG_WARNING,
"Attempted to swap subchannels with a null owner : sub #%d=%p sub #%d=%p\n",
a, p->subs[a]->owner, b, p->subs[b]->owner);
return;
}
rtp = p->subs[a]->rtp;
p->subs[a]->rtp = p->subs[b]->rtp;
p->subs[b]->rtp = rtp;
fds = p->subs[a]->owner->fds[0];
p->subs[a]->owner->fds[0] = p->subs[b]->owner->fds[0];
p->subs[b]->owner->fds[0] = fds;
fds = p->subs[a]->owner->fds[1];
p->subs[a]->owner->fds[1] = p->subs[b]->owner->fds[1];
p->subs[b]->owner->fds[1] = fds;
}
static int attempt_transfer(struct unistim_subchannel *p1, struct unistim_subchannel *p2)
{
int res = 0;
struct ast_channel
*chana = NULL, *chanb = NULL, *bridgea = NULL, *bridgeb = NULL, *peera =
NULL, *peerb = NULL, *peerc = NULL;
if (!p1->owner || !p2->owner) {
ast_log(LOG_WARNING, "Transfer attempted without dual ownership?\n");
return -1;
}
chana = p1->owner;
chanb = p2->owner;
bridgea = ast_bridged_channel(chana);
bridgeb = ast_bridged_channel(chanb);
if (bridgea) {
peera = chana;
peerb = chanb;
peerc = bridgea;
} else if (bridgeb) {
peera = chanb;
peerb = chana;
peerc = bridgeb;
}
if (peera && peerb && peerc && (peerb != peerc)) {
/*ast_quiet_chan(peera);
ast_quiet_chan(peerb);
ast_quiet_chan(peerc);
ast_quiet_chan(peerd); */
if (peera->cdr && peerb->cdr) {
peerb->cdr = ast_cdr_append(peerb->cdr, peera->cdr);
} else if (peera->cdr) {
peerb->cdr = peera->cdr;
}
peera->cdr = NULL;
if (peerb->cdr && peerc->cdr) {
peerb->cdr = ast_cdr_append(peerb->cdr, peerc->cdr);
} else if (peerc->cdr) {
peerb->cdr = peerc->cdr;
}
peerc->cdr = NULL;
if (ast_channel_masquerade(peerb, peerc)) {
ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", peerb->name,
peerc->name);
res = -1;
}
return res;
} else {
ast_log(LOG_NOTICE,
"Transfer attempted with no appropriate bridged calls to transfer\n");
if (chana)
ast_softhangup_nolock(chana, AST_SOFTHANGUP_DEV);
if (chanb)
ast_softhangup_nolock(chanb, AST_SOFTHANGUP_DEV);
return -1;
}
return 0;
}
void change_callerid(struct unistimsession *pte, int type, char *callerid)
{
char *data;
int size;
if (type)
data = pte->device->lst_cnm;
else
data = pte->device->lst_cid;
/* This is very nearly strncpy(), except that the remaining buffer
* is padded with ' ', instead of '\0' */
memset(data, ' ', TEXT_LENGTH_MAX);
size = strlen(callerid);
if (size > TEXT_LENGTH_MAX)
size = TEXT_LENGTH_MAX;
memcpy(data, callerid, size);
}
static void close_call(struct unistimsession *pte)
{
struct unistim_subchannel *sub;
struct unistim_line *l = pte->device->lines;
sub = pte->device->lines->subs[SUB_REAL];
send_stop_timer(pte);
if (sub->owner) {
sub->alreadygone = 1;
if (l->subs[SUB_THREEWAY]) {
l->subs[SUB_THREEWAY]->alreadygone = 1;
if (attempt_transfer(sub, l->subs[SUB_THREEWAY]) < 0)
ast_verb(0, "attempt_transfer failed.\n");
} else
ast_queue_hangup(sub->owner);
} else {
if (l->subs[SUB_THREEWAY]) {
if (l->subs[SUB_THREEWAY]->owner)
ast_queue_hangup_with_cause(l->subs[SUB_THREEWAY]->owner, AST_CAUSE_NORMAL_CLEARING);
else
ast_log(LOG_WARNING, "threeway sub without owner\n");
} else
ast_verb(0, "USTM(%s@%s-%d) channel already destroyed\n", sub->parent->name,
sub->parent->parent->name, sub->subtype);
}
change_callerid(pte, 0, pte->device->redial_number);
change_callerid(pte, 1, "");
write_history(pte, 'o', pte->device->missed_call);
pte->device->missed_call = 0;
show_main_page(pte);
return;
}
static void IgnoreCall(struct unistimsession *pte)
{
send_no_ring(pte);
return;
}
static void *unistim_ss(void *data)
{
struct ast_channel *chan = data;
struct unistim_subchannel *sub = chan->tech_pvt;
struct unistim_line *l = sub->parent;
struct unistimsession *s = l->parent->session;
int res;
ast_verb(3, "Starting switch on '%s@%s-%d' to %s\n", l->name, l->parent->name, sub->subtype, s->device->phone_number);
ast_copy_string(chan->exten, s->device->phone_number, sizeof(chan->exten));
ast_copy_string(s->device->redial_number, s->device->phone_number,
sizeof(s->device->redial_number));
ast_setstate(chan, AST_STATE_RING);
res = ast_pbx_run(chan);
if (res) {
ast_log(LOG_WARNING, "PBX exited non-zero\n");
send_tone(s, 1000, 0);;
}
return NULL;
}
static void start_rtp(struct unistim_subchannel *sub)
{
BUFFSEND;
struct sockaddr_in us = { 0, };
struct sockaddr_in public = { 0, };
struct sockaddr_in sin = { 0, };
int codec;
struct sockaddr_in sout = { 0, };
struct ast_sockaddr us_tmp;
struct ast_sockaddr sin_tmp;
struct ast_sockaddr sout_tmp;
/* Sanity checks */
if (!sub) {
ast_log(LOG_WARNING, "start_rtp with a null subchannel !\n");
return;
}
if (!sub->parent) {
ast_log(LOG_WARNING, "start_rtp with a null line !\n");
return;
}
if (!sub->parent->parent) {
ast_log(LOG_WARNING, "start_rtp with a null device !\n");
return;
}
if (!sub->parent->parent->session) {
ast_log(LOG_WARNING, "start_rtp with a null session !\n");
return;
}
sout = sub->parent->parent->session->sout;
ast_mutex_lock(&sub->lock);
/* Allocate the RTP */
if (unistimdebug)
ast_verb(0, "Starting RTP. Bind on %s\n", ast_inet_ntoa(sout.sin_addr));
ast_sockaddr_from_sin(&sout_tmp, &sout);
sub->rtp = ast_rtp_instance_new("asterisk", sched, &sout_tmp, NULL);
if (!sub->rtp) {
ast_log(LOG_WARNING, "Unable to create RTP session: %s binaddr=%s\n",
strerror(errno), ast_inet_ntoa(sout.sin_addr));
ast_mutex_unlock(&sub->lock);
return;
}
ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_RTCP, 1);
if (sub->owner) {
sub->owner->fds[0] = ast_rtp_instance_fd(sub->rtp, 0);
sub->owner->fds[1] = ast_rtp_instance_fd(sub->rtp, 1);
}
ast_rtp_instance_set_qos(sub->rtp, qos.tos_audio, qos.cos_audio, "UNISTIM RTP");
ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_NAT, sub->parent->parent->nat);
/* Create the RTP connection */
ast_rtp_instance_get_local_address(sub->rtp, &us_tmp);
ast_sockaddr_to_sin(&us_tmp, &us);
sin.sin_family = AF_INET;
/* Setting up RTP for our side */
memcpy(&sin.sin_addr, &sub->parent->parent->session->sin.sin_addr,
sizeof(sin.sin_addr));
sin.sin_port = htons(sub->parent->parent->rtp_port);
ast_sockaddr_from_sin(&sin_tmp, &sin);
ast_rtp_instance_set_remote_address(sub->rtp, &sin_tmp);
if (!(ast_format_cap_iscompatible(sub->owner->nativeformats, &sub->owner->readformat))) {
struct ast_format tmpfmt;
char tmp[256];
ast_best_codec(sub->owner->nativeformats, &tmpfmt);
ast_log(LOG_WARNING,
"Our read/writeformat has been changed to something incompatible: %s, using %s best codec from %s\n",
ast_getformatname(&sub->owner->readformat),
ast_getformatname(&tmpfmt),
ast_getformatname_multiple(tmp, sizeof(tmp), sub->owner->nativeformats));
ast_format_copy(&sub->owner->readformat, &tmpfmt);
ast_format_copy(&sub->owner->writeformat, &tmpfmt);
}
codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(sub->rtp), 1, &sub->owner->readformat, 0);
/* Setting up RTP of the phone */
if (public_ip.sin_family == 0) /* NAT IP override ? */
memcpy(&public, &us, sizeof(public)); /* No defined, using IP from recvmsg */
else
memcpy(&public, &public_ip, sizeof(public)); /* override */
if (unistimdebug) {
ast_verb(0, "RTP started : Our IP/port is : %s:%hd with codec %s\n",
ast_inet_ntoa(us.sin_addr),
htons(us.sin_port), ast_getformatname(&sub->owner->readformat));
ast_verb(0, "Starting phone RTP stack. Our public IP is %s\n",
ast_inet_ntoa(public.sin_addr));
}
if ((sub->owner->readformat.id == AST_FORMAT_ULAW) ||
(sub->owner->readformat.id == AST_FORMAT_ALAW)) {
if (unistimdebug)
ast_verb(0, "Sending packet_send_rtp_packet_size for codec %d\n", codec);
memcpy(buffsend + SIZE_HEADER, packet_send_rtp_packet_size,
sizeof(packet_send_rtp_packet_size));
buffsend[10] = (int) codec & 0xffffffffLL;
send_client(SIZE_HEADER + sizeof(packet_send_rtp_packet_size), buffsend,
sub->parent->parent->session);
}
if (unistimdebug)
ast_verb(0, "Sending Jitter Buffer Parameters Configuration\n");
memcpy(buffsend + SIZE_HEADER, packet_send_jitter_buffer_conf,
sizeof(packet_send_jitter_buffer_conf));
send_client(SIZE_HEADER + sizeof(packet_send_jitter_buffer_conf), buffsend,
sub->parent->parent->session);
if (sub->parent->parent->rtp_method != 0) {
uint16_t rtcpsin_port = htons(us.sin_port) + 1; /* RTCP port is RTP + 1 */
if (unistimdebug)
ast_verb(0, "Sending OpenAudioStreamTX using method #%d\n",
sub->parent->parent->rtp_method);
if (sub->parent->parent->rtp_method == 3)
memcpy(buffsend + SIZE_HEADER, packet_send_open_audio_stream_tx3,
sizeof(packet_send_open_audio_stream_tx3));
else
memcpy(buffsend + SIZE_HEADER, packet_send_open_audio_stream_tx,
sizeof(packet_send_open_audio_stream_tx));
if (sub->parent->parent->rtp_method != 2) {
memcpy(buffsend + 28, &public.sin_addr, sizeof(public.sin_addr));
buffsend[20] = (htons(sin.sin_port) & 0xff00) >> 8;
buffsend[21] = (htons(sin.sin_port) & 0x00ff);
buffsend[23] = (rtcpsin_port & 0x00ff);
buffsend[22] = (rtcpsin_port & 0xff00) >> 8;
buffsend[25] = (us.sin_port & 0xff00) >> 8;
buffsend[24] = (us.sin_port & 0x00ff);
buffsend[27] = (rtcpsin_port & 0x00ff);
buffsend[26] = (rtcpsin_port & 0xff00) >> 8;
} else {
memcpy(buffsend + 23, &public.sin_addr, sizeof(public.sin_addr));
buffsend[15] = (htons(sin.sin_port) & 0xff00) >> 8;
buffsend[16] = (htons(sin.sin_port) & 0x00ff);
buffsend[20] = (us.sin_port & 0xff00) >> 8;
buffsend[19] = (us.sin_port & 0x00ff);
buffsend[11] = codec;
}
buffsend[12] = codec;
send_client(SIZE_HEADER + sizeof(packet_send_open_audio_stream_tx), buffsend,
sub->parent->parent->session);
if (unistimdebug)
ast_verb(0, "Sending OpenAudioStreamRX\n");
if (sub->parent->parent->rtp_method == 3)
memcpy(buffsend + SIZE_HEADER, packet_send_open_audio_stream_rx3,
sizeof(packet_send_open_audio_stream_rx3));
else
memcpy(buffsend + SIZE_HEADER, packet_send_open_audio_stream_rx,
sizeof(packet_send_open_audio_stream_rx));
if (sub->parent->parent->rtp_method != 2) {
memcpy(buffsend + 28, &public.sin_addr, sizeof(public.sin_addr));
buffsend[20] = (htons(sin.sin_port) & 0xff00) >> 8;
buffsend[21] = (htons(sin.sin_port) & 0x00ff);
buffsend[23] = (rtcpsin_port & 0x00ff);
buffsend[22] = (rtcpsin_port & 0xff00) >> 8;
buffsend[25] = (us.sin_port & 0xff00) >> 8;
buffsend[24] = (us.sin_port & 0x00ff);
buffsend[27] = (rtcpsin_port & 0x00ff);
buffsend[26] = (rtcpsin_port & 0xff00) >> 8;
} else {
memcpy(buffsend + 23, &public.sin_addr, sizeof(public.sin_addr));
buffsend[15] = (htons(sin.sin_port) & 0xff00) >> 8;
buffsend[16] = (htons(sin.sin_port) & 0x00ff);
buffsend[20] = (us.sin_port & 0xff00) >> 8;
buffsend[19] = (us.sin_port & 0x00ff);
buffsend[12] = codec;
}
buffsend[11] = codec;
send_client(SIZE_HEADER + sizeof(packet_send_open_audio_stream_rx), buffsend,
sub->parent->parent->session);
} else {
uint16_t rtcpsin_port = htons(us.sin_port) + 1; /* RTCP port is RTP + 1 */
if (unistimdebug)
ast_verb(0, "Sending packet_send_call default method\n");
memcpy(buffsend + SIZE_HEADER, packet_send_call, sizeof(packet_send_call));
memcpy(buffsend + 53, &public.sin_addr, sizeof(public.sin_addr));
/* Destination port when sending RTP */
buffsend[49] = (us.sin_port & 0x00ff);
buffsend[50] = (us.sin_port & 0xff00) >> 8;
/* Destination port when sending RTCP */
buffsend[52] = (rtcpsin_port & 0x00ff);
buffsend[51] = (rtcpsin_port & 0xff00) >> 8;
/* Codec */
buffsend[40] = codec;
buffsend[41] = codec;
if (sub->owner->readformat.id == AST_FORMAT_ULAW)
buffsend[42] = 1; /* 1 = 20ms (160 bytes), 2 = 40ms (320 bytes) */
else if (sub->owner->readformat.id == AST_FORMAT_ALAW)
buffsend[42] = 1; /* 1 = 20ms (160 bytes), 2 = 40ms (320 bytes) */
else if (sub->owner->readformat.id == AST_FORMAT_G723_1)
buffsend[42] = 2; /* 1 = 30ms (24 bytes), 2 = 60 ms (48 bytes) */
else if (sub->owner->readformat.id == AST_FORMAT_G729A)
buffsend[42] = 2; /* 1 = 10ms (10 bytes), 2 = 20ms (20 bytes) */
else
ast_log(LOG_WARNING, "Unsupported codec %s!\n",
ast_getformatname(&sub->owner->readformat));
/* Source port for transmit RTP and Destination port for receiving RTP */
buffsend[45] = (htons(sin.sin_port) & 0xff00) >> 8;
buffsend[46] = (htons(sin.sin_port) & 0x00ff);
buffsend[47] = (rtcpsin_port & 0xff00) >> 8;
buffsend[48] = (rtcpsin_port & 0x00ff);
send_client(SIZE_HEADER + sizeof(packet_send_call), buffsend,
sub->parent->parent->session);
}
ast_mutex_unlock(&sub->lock);
}
static void SendDialTone(struct unistimsession *pte)
{
int i;
/* No country defined ? Using US tone */
if (ast_strlen_zero(pte->device->country)) {
if (unistimdebug)
ast_verb(0, "No country defined, using US tone\n");
send_tone(pte, 350, 440);
return;
}
if (strlen(pte->device->country) != 2) {
if (unistimdebug)
ast_verb(0, "Country code != 2 char, using US tone\n");
send_tone(pte, 350, 440);
return;
}
i = 0;
while (frequency[i].freq1) {
if ((frequency[i].country[0] == pte->device->country[0]) &&
(frequency[i].country[1] == pte->device->country[1])) {
if (unistimdebug)
ast_verb(0, "Country code found (%s), freq1=%d freq2=%d\n",
frequency[i].country, frequency[i].freq1, frequency[i].freq2);
send_tone(pte, frequency[i].freq1, frequency[i].freq2);
}
i++;
}
}
static void handle_dial_page(struct unistimsession *pte)
{
pte->state = STATE_DIALPAGE;
if (pte->device->call_forward[0] == -1) {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "");
send_text(TEXT_LINE1, TEXT_NORMAL, pte, "Enter forward");
send_text_status(pte, "ForwardCancel BackSpcErase");
if (pte->device->call_forward[1] != 0) {
char tmp[TEXT_LENGTH_MAX + 1];
ast_copy_string(pte->device->phone_number, pte->device->call_forward + 1,
sizeof(pte->device->phone_number));
pte->device->size_phone_number = strlen(pte->device->phone_number);
if (pte->device->size_phone_number > 15)
pte->device->size_phone_number = 15;
strcpy(tmp, "Number : ...............");
memcpy(tmp + 9, pte->device->phone_number, pte->device->size_phone_number);
if (pte->device->height == 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, tmp);
send_blink_cursor(pte);
send_cursor_pos(pte,
(unsigned char) (TEXT_LINE0 + 0x09 +
pte->device->size_phone_number));
} else {
send_text(TEXT_LINE2, TEXT_NORMAL, pte, tmp);
send_blink_cursor(pte);
send_cursor_pos(pte,
(unsigned char) (TEXT_LINE2 + 0x09 +
pte->device->size_phone_number));
}
send_led_update(pte, 0);
return;
}
} else {
if ((pte->device->output == OUTPUT_HANDSET) &&
(pte->device->receiver_state == STATE_ONHOOK))
send_select_output(pte, OUTPUT_SPEAKER, pte->device->volume, MUTE_OFF);
else
send_select_output(pte, pte->device->output, pte->device->volume, MUTE_OFF);
SendDialTone(pte);
if (pte->device->height > 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Enter the number to dial");
send_text(TEXT_LINE1, TEXT_NORMAL, pte, "and press Call");
}
send_text_status(pte, "Call Redial BackSpcErase");
}
if (pte->device->height == 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Number : ...............");
send_blink_cursor(pte);
send_cursor_pos(pte, TEXT_LINE0 + 0x09);
} else {
send_text(TEXT_LINE2, TEXT_NORMAL, pte, "Number : ...............");
send_blink_cursor(pte);
send_cursor_pos(pte, TEXT_LINE2 + 0x09);
}
pte->device->size_phone_number = 0;
pte->device->phone_number[0] = 0;
change_favorite_icon(pte, FAV_ICON_PHONE_BLACK);
Sendicon(TEXT_LINE0, FAV_ICON_NONE, pte);
pte->device->missed_call = 0;
send_led_update(pte, 0);
return;
}
/* Step 1 : Music On Hold for peer, Dialing screen for us */
static void TransferCallStep1(struct unistimsession *pte)
{
struct unistim_subchannel *sub;
struct unistim_line *p = pte->device->lines;
sub = p->subs[SUB_REAL];
if (!sub->owner) {
ast_log(LOG_WARNING, "Unable to find subchannel for music on hold\n");
return;
}
if (p->subs[SUB_THREEWAY]) {
if (unistimdebug)
ast_verb(0, "Transfer canceled, hangup our threeway channel\n");
if (p->subs[SUB_THREEWAY]->owner)
ast_queue_hangup_with_cause(p->subs[SUB_THREEWAY]->owner, AST_CAUSE_NORMAL_CLEARING);
else
ast_log(LOG_WARNING, "Canceling a threeway channel without owner\n");
return;
}
/* Start music on hold if appropriate */
if (pte->device->moh)
ast_log(LOG_WARNING, "Transfer with peer already listening music on hold\n");
else {
if (ast_bridged_channel(p->subs[SUB_REAL]->owner)) {
ast_moh_start(ast_bridged_channel(p->subs[SUB_REAL]->owner),
pte->device->lines->musicclass, NULL);
pte->device->moh = 1;
} else {
ast_log(LOG_WARNING, "Unable to find peer subchannel for music on hold\n");
return;
}
}
/* Silence our channel */
if (!pte->device->silence_generator) {
pte->device->silence_generator =
ast_channel_start_silence_generator(p->subs[SUB_REAL]->owner);
if (pte->device->silence_generator == NULL)
ast_log(LOG_WARNING, "Unable to start a silence generator.\n");
else if (unistimdebug)
ast_verb(0, "Starting silence generator\n");
}
handle_dial_page(pte);
}
/* From phone to PBX */
static void HandleCallOutgoing(struct unistimsession *s)
{
struct ast_channel *c;
struct unistim_subchannel *sub;
pthread_t t;
s->state = STATE_CALL;
sub = s->device->lines->subs[SUB_REAL];
if (!sub) {
ast_log(LOG_NOTICE, "No available lines on: %s\n", s->device->name);
return;
}
if (!sub->owner) { /* A call is already in progress ? */
c = unistim_new(sub, AST_STATE_DOWN, NULL); /* No, starting a new one */
if (c) {
/* Need to start RTP before calling ast_pbx_run */
if (!sub->rtp)
start_rtp(sub);
send_select_output(s, s->device->output, s->device->volume, MUTE_OFF);
if (s->device->height == 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, s, s->device->phone_number);
} else {
send_text(TEXT_LINE0, TEXT_NORMAL, s, "Calling :");
send_text(TEXT_LINE1, TEXT_NORMAL, s, s->device->phone_number);
send_text(TEXT_LINE2, TEXT_NORMAL, s, "Dialing...");
}
send_text_status(s, "Hangup");
/* start switch */
if (ast_pthread_create(&t, NULL, unistim_ss, c)) {
display_last_error("Unable to create switch thread");
ast_queue_hangup_with_cause(c, AST_CAUSE_SWITCH_CONGESTION);
}
} else
ast_log(LOG_WARNING, "Unable to create channel for %s@%s\n",
sub->parent->name, s->device->name);
} else { /* We already have a call, so we switch in a threeway call */
if (s->device->moh) {
struct unistim_subchannel *subchannel;
struct unistim_line *p = s->device->lines;
subchannel = p->subs[SUB_REAL];
if (!subchannel->owner) {
ast_log(LOG_WARNING, "Unable to find subchannel for music on hold\n");
return;
}
if (p->subs[SUB_THREEWAY]) {
ast_log(LOG_WARNING,
"Can't transfer while an another transfer is taking place\n");
return;
}
if (!alloc_sub(p, SUB_THREEWAY)) {
ast_log(LOG_WARNING, "Unable to allocate three-way subchannel\n");
return;
}
/* Stop the silence generator */
if (s->device->silence_generator) {
if (unistimdebug)
ast_verb(0, "Stopping silence generator\n");
ast_channel_stop_silence_generator(subchannel->owner,
s->device->silence_generator);
s->device->silence_generator = NULL;
}
send_tone(s, 0, 0);
/* Make new channel */
c = unistim_new(p->subs[SUB_THREEWAY], AST_STATE_DOWN, NULL);
if (!c) {
ast_log(LOG_WARNING, "Cannot allocate new structure on channel %p\n", p);
return;
}
/* Swap things around between the three-way and real call */
swap_subs(p, SUB_THREEWAY, SUB_REAL);
send_select_output(s, s->device->output, s->device->volume, MUTE_OFF);
if (s->device->height == 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, s, s->device->phone_number);
} else {
send_text(TEXT_LINE0, TEXT_NORMAL, s, "Calling (pre-transfer)");
send_text(TEXT_LINE1, TEXT_NORMAL, s, s->device->phone_number);
send_text(TEXT_LINE2, TEXT_NORMAL, s, "Dialing...");
}
send_text_status(s, "TransfrCancel");
if (ast_pthread_create(&t, NULL, unistim_ss, p->subs[SUB_THREEWAY]->owner)) {
ast_log(LOG_WARNING, "Unable to start simple switch on channel %p\n", p);
ast_hangup(c);
return;
}
if (unistimdebug)
ast_verb(0, "Started three way call on channel %p (%s) subchan %d\n",
p->subs[SUB_THREEWAY]->owner, p->subs[SUB_THREEWAY]->owner->name,
p->subs[SUB_THREEWAY]->subtype);
} else
ast_debug(1, "Current sub [%s] already has owner\n", sub->owner->name);
}
return;
}
/* From PBX to phone */
static void HandleCallIncoming(struct unistimsession *s)
{
struct unistim_subchannel *sub;
s->state = STATE_CALL;
s->device->missed_call = 0;
send_no_ring(s);
sub = s->device->lines->subs[SUB_REAL];
if (!sub) {
ast_log(LOG_NOTICE, "No available lines on: %s\n", s->device->name);
return;
} else if (unistimdebug)
ast_verb(0, "Handle Call Incoming for %s@%s\n", sub->parent->name,
s->device->name);
start_rtp(sub);
if (!sub->rtp)
ast_log(LOG_WARNING, "Unable to create channel for %s@%s\n", sub->parent->name,
s->device->name);
ast_queue_control(sub->owner, AST_CONTROL_ANSWER);
send_text(TEXT_LINE2, TEXT_NORMAL, s, "is on-line");
send_text_status(s, "Hangup Transf");
send_start_timer(s);
if ((s->device->output == OUTPUT_HANDSET) &&
(s->device->receiver_state == STATE_ONHOOK))
send_select_output(s, OUTPUT_SPEAKER, s->device->volume, MUTE_OFF);
else
send_select_output(s, s->device->output, s->device->volume, MUTE_OFF);
s->device->start_call_timestamp = time(0);
write_history(s, 'i', 0);
return;
}
static int unistim_do_senddigit(struct unistimsession *pte, char digit)
{
struct ast_frame f = { .frametype = AST_FRAME_DTMF, .subclass.integer = digit, .src = "unistim" };
struct unistim_subchannel *sub;
sub = pte->device->lines->subs[SUB_REAL];
if (!sub->owner || sub->alreadygone) {
ast_log(LOG_WARNING, "Unable to find subchannel in dtmf senddigit\n");
return -1;
}
/* Send DTMF indication _before_ playing sounds */
ast_queue_frame(sub->owner, &f);
if (unistimdebug)
ast_verb(0, "Send Digit %c\n", digit);
switch (digit) {
case '0':
send_tone(pte, 941, 1336);
break;
case '1':
send_tone(pte, 697, 1209);
break;
case '2':
send_tone(pte, 697, 1336);
break;
case '3':
send_tone(pte, 697, 1477);
break;
case '4':
send_tone(pte, 770, 1209);
break;
case '5':
send_tone(pte, 770, 1336);
break;
case '6':
send_tone(pte, 770, 1477);
break;
case '7':
send_tone(pte, 852, 1209);
break;
case '8':
send_tone(pte, 852, 1336);
break;
case '9':
send_tone(pte, 852, 1477);
break;
case 'A':
send_tone(pte, 697, 1633);
break;
case 'B':
send_tone(pte, 770, 1633);
break;
case 'C':
send_tone(pte, 852, 1633);
break;
case 'D':
send_tone(pte, 941, 1633);
break;
case '*':
send_tone(pte, 941, 1209);
break;
case '#':
send_tone(pte, 941, 1477);
break;
default:
send_tone(pte, 500, 2000);
}
usleep(150000); /* XXX Less than perfect, blocking an important thread is not a good idea */
send_tone(pte, 0, 0);
return 0;
}
static void key_call(struct unistimsession *pte, char keycode)
{
if ((keycode >= KEY_0) && (keycode <= KEY_SHARP)) {
if (keycode == KEY_SHARP)
keycode = '#';
else if (keycode == KEY_STAR)
keycode = '*';
else
keycode -= 0x10;
unistim_do_senddigit(pte, keycode);
return;
}
switch (keycode) {
case KEY_HANGUP:
case KEY_FUNC1:
close_call(pte);
break;
case KEY_FUNC2:
TransferCallStep1(pte);
break;
case KEY_HEADPHN:
if (pte->device->output == OUTPUT_HEADPHONE)
send_select_output(pte, OUTPUT_HANDSET, pte->device->volume, MUTE_OFF);
else
send_select_output(pte, OUTPUT_HEADPHONE, pte->device->volume, MUTE_OFF);
break;
case KEY_LOUDSPK:
if (pte->device->output != OUTPUT_SPEAKER)
send_select_output(pte, OUTPUT_SPEAKER, pte->device->volume, MUTE_OFF);
else
send_select_output(pte, pte->device->previous_output, pte->device->volume,
MUTE_OFF);
break;
case KEY_MUTE:
if (!pte->device->moh) {
if (pte->device->mute == MUTE_ON)
send_select_output(pte, pte->device->output, pte->device->volume, MUTE_OFF);
else
send_select_output(pte, pte->device->output, pte->device->volume, MUTE_ON);
break;
}
case KEY_ONHOLD:
{
struct unistim_subchannel *sub;
struct ast_channel *bridgepeer = NULL;
sub = pte->device->lines->subs[SUB_REAL];
if (!sub->owner) {
ast_log(LOG_WARNING, "Unable to find subchannel for music on hold\n");
return;
}
if ((bridgepeer = ast_bridged_channel(sub->owner))) {
if (pte->device->moh) {
ast_moh_stop(bridgepeer);
pte->device->moh = 0;
send_select_output(pte, pte->device->output, pte->device->volume,
MUTE_OFF);
} else {
ast_moh_start(bridgepeer, pte->device->lines->musicclass, NULL);
pte->device->moh = 1;
send_select_output(pte, pte->device->output, pte->device->volume,
MUTE_ON);
}
} else
ast_log(LOG_WARNING,
"Unable to find peer subchannel for music on hold\n");
break;
}
}
return;
}
static void key_ringing(struct unistimsession *pte, char keycode)
{
if (keycode == KEY_FAV0 + pte->device->softkeylinepos) {
HandleCallIncoming(pte);
return;
}
switch (keycode) {
case KEY_HANGUP:
case KEY_FUNC4:
IgnoreCall(pte);
break;
case KEY_FUNC1:
HandleCallIncoming(pte);
break;
}
return;
}
static void Keyfavorite(struct unistimsession *pte, char keycode)
{
int fav;
if ((keycode < KEY_FAV1) && (keycode > KEY_FAV5)) {
ast_log(LOG_WARNING, "It's not a favorite key\n");
return;
}
if (keycode == KEY_FAV0)
return;
fav = keycode - KEY_FAV0;
if (pte->device->softkeyicon[fav] == 0)
return;
ast_copy_string(pte->device->phone_number, pte->device->softkeynumber[fav],
sizeof(pte->device->phone_number));
HandleCallOutgoing(pte);
return;
}
static void key_dial_page(struct unistimsession *pte, char keycode)
{
if (keycode == KEY_FUNC3) {
if (pte->device->size_phone_number <= 1)
keycode = KEY_FUNC4;
else {
pte->device->size_phone_number -= 2;
keycode = pte->device->phone_number[pte->device->size_phone_number] + 0x10;
}
}
if ((keycode >= KEY_0) && (keycode <= KEY_SHARP)) {
char tmpbuf[] = "Number : ...............";
int i = 0;
if (pte->device->size_phone_number >= 15)
return;
if (pte->device->size_phone_number == 0)
send_tone(pte, 0, 0);
while (i < pte->device->size_phone_number) {
tmpbuf[i + 9] = pte->device->phone_number[i];
i++;
}
if (keycode == KEY_SHARP)
keycode = '#';
else if (keycode == KEY_STAR)
keycode = '*';
else
keycode -= 0x10;
tmpbuf[i + 9] = keycode;
pte->device->phone_number[i] = keycode;
pte->device->size_phone_number++;
pte->device->phone_number[i + 1] = 0;
if (pte->device->height == 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, tmpbuf);
} else {
send_text(TEXT_LINE2, TEXT_NORMAL, pte, tmpbuf);
}
send_blink_cursor(pte);
send_cursor_pos(pte, (unsigned char) (TEXT_LINE2 + 0x0a + i));
return;
}
if (keycode == KEY_FUNC4) {
pte->device->size_phone_number = 0;
if (pte->device->height == 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Number : ...............");
send_blink_cursor(pte);
send_cursor_pos(pte, TEXT_LINE0 + 0x09);
} else {
send_text(TEXT_LINE2, TEXT_NORMAL, pte, "Number : ...............");
send_blink_cursor(pte);
send_cursor_pos(pte, TEXT_LINE2 + 0x09);
}
return;
}
if (pte->device->call_forward[0] == -1) {
if (keycode == KEY_FUNC1) {
ast_copy_string(pte->device->call_forward, pte->device->phone_number,
sizeof(pte->device->call_forward));
show_main_page(pte);
} else if ((keycode == KEY_FUNC2) || (keycode == KEY_HANGUP)) {
pte->device->call_forward[0] = '\0';
show_main_page(pte);
}
return;
}
switch (keycode) {
case KEY_FUNC2:
if (ast_strlen_zero(pte->device->redial_number))
break;
ast_copy_string(pte->device->phone_number, pte->device->redial_number,
sizeof(pte->device->phone_number));
case KEY_FUNC1:
HandleCallOutgoing(pte);
break;
case KEY_HANGUP:
if (pte->device->lines->subs[SUB_REAL]->owner) {
/* Stop the silence generator */
if (pte->device->silence_generator) {
if (unistimdebug)
ast_verb(0, "Stopping silence generator\n");
ast_channel_stop_silence_generator(pte->device->lines->subs[SUB_REAL]->
owner, pte->device->silence_generator);
pte->device->silence_generator = NULL;
}
send_tone(pte, 0, 0);
ast_moh_stop(ast_bridged_channel(pte->device->lines->subs[SUB_REAL]->owner));
pte->device->moh = 0;
pte->state = STATE_CALL;
if (pte->device->height == 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Dial Cancel,back to priv. call.");
} else {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Dialing canceled,");
send_text(TEXT_LINE1, TEXT_NORMAL, pte, "switching back to");
send_text(TEXT_LINE2, TEXT_NORMAL, pte, "previous call.");
}
send_text_status(pte, "Hangup Transf");
} else
show_main_page(pte);
break;
case KEY_FAV1:
case KEY_FAV2:
case KEY_FAV3:
case KEY_FAV4:
case KEY_FAV5:
Keyfavorite(pte, keycode);
break;
case KEY_LOUDSPK:
if (pte->device->output == OUTPUT_SPEAKER) {
if (pte->device->receiver_state == STATE_OFFHOOK)
send_select_output(pte, pte->device->previous_output, pte->device->volume,
MUTE_OFF);
else
show_main_page(pte);
} else
send_select_output(pte, OUTPUT_SPEAKER, pte->device->volume, MUTE_OFF);
break;
case KEY_HEADPHN:
if (pte->device->output == OUTPUT_HEADPHONE) {
if (pte->device->receiver_state == STATE_OFFHOOK)
send_select_output(pte, OUTPUT_HANDSET, pte->device->volume, MUTE_OFF);
else
show_main_page(pte);
} else
send_select_output(pte, OUTPUT_HEADPHONE, pte->device->volume, MUTE_OFF);
break;
}
return;
}
#define SELECTCODEC_START_ENTRY_POS 15
#define SELECTCODEC_MAX_LENGTH 2
#define SELECTCODEC_MSG "Codec number : .."
static void HandleSelectCodec(struct unistimsession *pte)
{
char buf[30], buf2[5];
pte->state = STATE_SELECTCODEC;
strcpy(buf, "Using codec ");
sprintf(buf2, "%d", pte->device->codec_number);
strcat(buf, buf2);
strcat(buf, " (G711u=0,");
send_text(TEXT_LINE0, TEXT_NORMAL, pte, buf);
send_text(TEXT_LINE1, TEXT_NORMAL, pte, "G723=4,G711a=8,G729A=18)");
send_text(TEXT_LINE2, TEXT_INVERSE, pte, SELECTCODEC_MSG);
send_blink_cursor(pte);
send_cursor_pos(pte, TEXT_LINE2 + SELECTCODEC_START_ENTRY_POS);
pte->size_buff_entry = 0;
send_text_status(pte, "Select BackSpcErase Cancel");
return;
}
static void key_select_codec(struct unistimsession *pte, char keycode)
{
if (keycode == KEY_FUNC2) {
if (pte->size_buff_entry <= 1)
keycode = KEY_FUNC3;
else {
pte->size_buff_entry -= 2;
keycode = pte->buff_entry[pte->size_buff_entry] + 0x10;
}
}
if ((keycode >= KEY_0) && (keycode <= KEY_9)) {
char tmpbuf[] = SELECTCODEC_MSG;
int i = 0;
if (pte->size_buff_entry >= SELECTCODEC_MAX_LENGTH)
return;
while (i < pte->size_buff_entry) {
tmpbuf[i + SELECTCODEC_START_ENTRY_POS] = pte->buff_entry[i];
i++;
}
tmpbuf[i + SELECTCODEC_START_ENTRY_POS] = keycode - 0x10;
pte->buff_entry[i] = keycode - 0x10;
pte->size_buff_entry++;
send_text(TEXT_LINE2, TEXT_INVERSE, pte, tmpbuf);
send_blink_cursor(pte);
send_cursor_pos(pte,
(unsigned char) (TEXT_LINE2 + SELECTCODEC_START_ENTRY_POS + 1 + i));
return;
}
switch (keycode) {
case KEY_FUNC1:
if (pte->size_buff_entry == 1)
pte->device->codec_number = pte->buff_entry[0] - 48;
else if (pte->size_buff_entry == 2)
pte->device->codec_number =
((pte->buff_entry[0] - 48) * 10) + (pte->buff_entry[1] - 48);
show_main_page(pte);
break;
case KEY_FUNC3:
pte->size_buff_entry = 0;
send_text(TEXT_LINE2, TEXT_INVERSE, pte, SELECTCODEC_MSG);
send_blink_cursor(pte);
send_cursor_pos(pte, TEXT_LINE2 + SELECTCODEC_START_ENTRY_POS);
break;
case KEY_HANGUP:
case KEY_FUNC4:
show_main_page(pte);
break;
}
return;
}
#define SELECTEXTENSION_START_ENTRY_POS 0
#define SELECTEXTENSION_MAX_LENGTH 10
#define SELECTEXTENSION_MSG ".........."
static void ShowExtensionPage(struct unistimsession *pte)
{
pte->state = STATE_EXTENSION;
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Please enter a Terminal");
send_text(TEXT_LINE1, TEXT_NORMAL, pte, "Number (TN) :");
send_text(TEXT_LINE2, TEXT_NORMAL, pte, SELECTEXTENSION_MSG);
send_blink_cursor(pte);
send_cursor_pos(pte, TEXT_LINE2 + SELECTEXTENSION_START_ENTRY_POS);
send_text_status(pte, "Enter BackSpcErase");
pte->size_buff_entry = 0;
return;
}
static void key_select_extension(struct unistimsession *pte, char keycode)
{
if (keycode == KEY_FUNC2) {
if (pte->size_buff_entry <= 1)
keycode = KEY_FUNC3;
else {
pte->size_buff_entry -= 2;
keycode = pte->buff_entry[pte->size_buff_entry] + 0x10;
}
}
if ((keycode >= KEY_0) && (keycode <= KEY_9)) {
char tmpbuf[] = SELECTEXTENSION_MSG;
int i = 0;
if (pte->size_buff_entry >= SELECTEXTENSION_MAX_LENGTH)
return;
while (i < pte->size_buff_entry) {
tmpbuf[i + SELECTEXTENSION_START_ENTRY_POS] = pte->buff_entry[i];
i++;
}
tmpbuf[i + SELECTEXTENSION_START_ENTRY_POS] = keycode - 0x10;
pte->buff_entry[i] = keycode - 0x10;
pte->size_buff_entry++;
send_text(TEXT_LINE2, TEXT_NORMAL, pte, tmpbuf);
send_blink_cursor(pte);
send_cursor_pos(pte,
(unsigned char) (TEXT_LINE2 + SELECTEXTENSION_START_ENTRY_POS + 1 +
i));
return;
}
switch (keycode) {
case KEY_FUNC1:
if (pte->size_buff_entry < 1)
return;
if (autoprovisioning == AUTOPROVISIONING_TN) {
struct unistim_device *d;
/* First step : looking for this TN in our device list */
ast_mutex_lock(&devicelock);
d = devices;
pte->buff_entry[pte->size_buff_entry] = '\0';
while (d) {
if (d->id[0] == 'T') { /* It's a TN device ? */
/* It's the TN we're looking for ? */
if (!strcmp((d->id) + 1, pte->buff_entry)) {
pte->device = d;
d->session = pte;
d->codec_number = DEFAULT_CODEC;
d->pos_fav = 0;
d->missed_call = 0;
d->receiver_state = STATE_ONHOOK;
strcpy(d->id, pte->macaddr);
pte->device->extension_number[0] = 'T';
pte->device->extension = EXTENSION_TN;
ast_copy_string((pte->device->extension_number) + 1,
pte->buff_entry, pte->size_buff_entry + 1);
ast_mutex_unlock(&devicelock);
show_main_page(pte);
refresh_all_favorite(pte);
return;
}
}
d = d->next;
}
ast_mutex_unlock(&devicelock);
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Invalid Terminal Number.");
send_text(TEXT_LINE1, TEXT_NORMAL, pte, "Please try again :");
send_cursor_pos(pte,
(unsigned char) (TEXT_LINE2 + SELECTEXTENSION_START_ENTRY_POS +
pte->size_buff_entry));
send_blink_cursor(pte);
} else {
ast_copy_string(pte->device->extension_number, pte->buff_entry,
pte->size_buff_entry + 1);
if (RegisterExtension(pte)) {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Invalid extension.");
send_text(TEXT_LINE1, TEXT_NORMAL, pte, "Please try again :");
send_cursor_pos(pte,
(unsigned char) (TEXT_LINE2 +
SELECTEXTENSION_START_ENTRY_POS +
pte->size_buff_entry));
send_blink_cursor(pte);
} else
show_main_page(pte);
}
break;
case KEY_FUNC3:
pte->size_buff_entry = 0;
send_text(TEXT_LINE2, TEXT_NORMAL, pte, SELECTEXTENSION_MSG);
send_blink_cursor(pte);
send_cursor_pos(pte, TEXT_LINE2 + SELECTEXTENSION_START_ENTRY_POS);
break;
}
return;
}
static int ReformatNumber(char *number)
{
int pos = 0, i = 0, size = strlen(number);
for (; i < size; i++) {
if ((number[i] >= '0') && (number[i] <= '9')) {
if (i == pos) {
pos++;
continue;
}
number[pos] = number[i];
pos++;
}
}
number[pos] = 0;
return pos;
}
static void show_entry_history(struct unistimsession *pte, FILE ** f)
{
char line[TEXT_LENGTH_MAX + 1], status[STATUS_LENGTH_MAX + 1], func1[10], func2[10],
func3[10];
if (fread(line, TEXT_LENGTH_MAX, 1, *f) != 1) {
display_last_error("Can't read history date entry");
fclose(*f);
return;
}
line[sizeof(line) - 1] = '\0';
send_text(TEXT_LINE0, TEXT_NORMAL, pte, line);
if (fread(line, TEXT_LENGTH_MAX, 1, *f) != 1) {
display_last_error("Can't read callerid entry");
fclose(*f);
return;
}
line[sizeof(line) - 1] = '\0';
ast_copy_string(pte->device->lst_cid, line, sizeof(pte->device->lst_cid));
send_text(TEXT_LINE1, TEXT_NORMAL, pte, line);
if (fread(line, TEXT_LENGTH_MAX, 1, *f) != 1) {
display_last_error("Can't read callername entry");
fclose(*f);
return;
}
line[sizeof(line) - 1] = '\0';
send_text(TEXT_LINE2, TEXT_NORMAL, pte, line);
fclose(*f);
snprintf(line, sizeof(line), "Call %03d/%03d", pte->buff_entry[2],
pte->buff_entry[1]);
send_texttitle(pte, line);
if (pte->buff_entry[2] == 1)
strcpy(func1, " ");
else
strcpy(func1, "Prvious");
if (pte->buff_entry[2] >= pte->buff_entry[1])
strcpy(func2, " ");
else
strcpy(func2, "Next ");
if (ReformatNumber(pte->device->lst_cid))
strcpy(func3, "Redial ");
else
strcpy(func3, " ");
snprintf(status, sizeof(status), "%s%s%sCancel", func1, func2, func3);
send_text_status(pte, status);
}
static char OpenHistory(struct unistimsession *pte, char way, FILE ** f)
{
char tmp[AST_CONFIG_MAX_PATH];
char count;
snprintf(tmp, sizeof(tmp), "%s/%s/%s-%c.csv", ast_config_AST_LOG_DIR,
USTM_LOG_DIR, pte->device->name, way);
*f = fopen(tmp, "r");
if (!*f) {
display_last_error("Unable to open history file");
return 0;
}
if (fread(&count, 1, 1, *f) != 1) {
display_last_error("Unable to read history header - display.");
fclose(*f);
return 0;
}
if (count > MAX_ENTRY_LOG) {
ast_log(LOG_WARNING, "Invalid count in history header of %s (%d max %d)\n", tmp,
count, MAX_ENTRY_LOG);
fclose(*f);
return 0;
}
return count;
}
static void show_history(struct unistimsession *pte, char way)
{
FILE *f;
char count;
if (!pte->device)
return;
if (!pte->device->callhistory)
return;
count = OpenHistory(pte, way, &f);
if (!count)
return;
pte->buff_entry[0] = way;
pte->buff_entry[1] = count;
pte->buff_entry[2] = 1;
show_entry_history(pte, &f);
pte->state = STATE_HISTORY;
}
static void show_main_page(struct unistimsession *pte)
{
char tmpbuf[TEXT_LENGTH_MAX + 1];
if ((pte->device->extension == EXTENSION_ASK) &&
(ast_strlen_zero(pte->device->extension_number))) {
ShowExtensionPage(pte);
return;
}
pte->state = STATE_MAINPAGE;
send_tone(pte, 0, 0);
send_select_output(pte, pte->device->output, pte->device->volume, MUTE_ON_DISCRET);
pte->device->lines->lastmsgssent = 0;
send_favorite(pte->device->softkeylinepos, FAV_ICON_ONHOOK_BLACK, pte,
pte->device->softkeylabel[pte->device->softkeylinepos]);
if (!ast_strlen_zero(pte->device->call_forward)) {
if (pte->device->height == 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Forwarding ON");
} else {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Call forwarded to :");
send_text(TEXT_LINE1, TEXT_NORMAL, pte, pte->device->call_forward);
}
Sendicon(TEXT_LINE0, FAV_ICON_REFLECT + FAV_BLINK_SLOW, pte);
send_text_status(pte, "Dial Redial NoForwd");
} else {
if ((pte->device->extension == EXTENSION_ASK) ||
(pte->device->extension == EXTENSION_TN))
send_text_status(pte, "Dial Redial ForwardUnregis");
else
send_text_status(pte, "Dial Redial Forward");
send_text(TEXT_LINE1, TEXT_NORMAL, pte, pte->device->maintext1);
if (pte->device->missed_call == 0)
send_text(TEXT_LINE0, TEXT_NORMAL, pte, pte->device->maintext0);
else {
sprintf(tmpbuf, "%d unanswered call(s)", pte->device->missed_call);
send_text(TEXT_LINE0, TEXT_NORMAL, pte, tmpbuf);
Sendicon(TEXT_LINE0, FAV_ICON_CALL_CENTER + FAV_BLINK_SLOW, pte);
}
}
if (ast_strlen_zero(pte->device->maintext2)) {
strcpy(tmpbuf, "IP : ");
strcat(tmpbuf, ast_inet_ntoa(pte->sin.sin_addr));
send_text(TEXT_LINE2, TEXT_NORMAL, pte, tmpbuf);
} else
send_text(TEXT_LINE2, TEXT_NORMAL, pte, pte->device->maintext2);
send_texttitle(pte, pte->device->titledefault);
change_favorite_icon(pte, FAV_ICON_ONHOOK_BLACK);
}
static void key_main_page(struct unistimsession *pte, char keycode)
{
if (pte->device->missed_call) {
Sendicon(TEXT_LINE0, FAV_ICON_NONE, pte);
pte->device->missed_call = 0;
}
if ((keycode >= KEY_0) && (keycode <= KEY_SHARP)) {
handle_dial_page(pte);
key_dial_page(pte, keycode);
return;
}
switch (keycode) {
case KEY_FUNC1:
handle_dial_page(pte);
break;
case KEY_FUNC2:
if (ast_strlen_zero(pte->device->redial_number))
break;
if ((pte->device->output == OUTPUT_HANDSET) &&
(pte->device->receiver_state == STATE_ONHOOK))
send_select_output(pte, OUTPUT_SPEAKER, pte->device->volume, MUTE_OFF);
else
send_select_output(pte, pte->device->output, pte->device->volume, MUTE_OFF);
ast_copy_string(pte->device->phone_number, pte->device->redial_number,
sizeof(pte->device->phone_number));
HandleCallOutgoing(pte);
break;
case KEY_FUNC3:
if (!ast_strlen_zero(pte->device->call_forward)) {
/* Cancel call forwarding */
memmove(pte->device->call_forward + 1, pte->device->call_forward,
sizeof(pte->device->call_forward));
pte->device->call_forward[0] = '\0';
Sendicon(TEXT_LINE0, FAV_ICON_NONE, pte);
pte->device->output = OUTPUT_HANDSET; /* Seems to be reseted somewhere */
show_main_page(pte);
break;
}
pte->device->call_forward[0] = -1;
handle_dial_page(pte);
break;
case KEY_FUNC4:
if (pte->device->extension == EXTENSION_ASK) {
UnregisterExtension(pte);
pte->device->extension_number[0] = '\0';
ShowExtensionPage(pte);
} else if (pte->device->extension == EXTENSION_TN) {
ast_mutex_lock(&devicelock);
strcpy(pte->device->id, pte->device->extension_number);
pte->buff_entry[0] = '\0';
pte->size_buff_entry = 0;
pte->device->session = NULL;
pte->device = NULL;
ast_mutex_unlock(&devicelock);
ShowExtensionPage(pte);
}
break;
case KEY_FAV0:
handle_dial_page(pte);
break;
case KEY_FAV1:
case KEY_FAV2:
case KEY_FAV3:
case KEY_FAV4:
case KEY_FAV5:
if ((pte->device->output == OUTPUT_HANDSET) &&
(pte->device->receiver_state == STATE_ONHOOK))
send_select_output(pte, OUTPUT_SPEAKER, pte->device->volume, MUTE_OFF);
else
send_select_output(pte, pte->device->output, pte->device->volume, MUTE_OFF);
Keyfavorite(pte, keycode);
break;
case KEY_CONF:
HandleSelectCodec(pte);
break;
case KEY_LOUDSPK:
send_select_output(pte, OUTPUT_SPEAKER, pte->device->volume, MUTE_OFF);
handle_dial_page(pte);
break;
case KEY_HEADPHN:
send_select_output(pte, OUTPUT_HEADPHONE, pte->device->volume, MUTE_OFF);
handle_dial_page(pte);
break;
case KEY_SNDHIST:
show_history(pte, 'o');
break;
case KEY_RCVHIST:
show_history(pte, 'i');
break;
}
return;
}
static void key_history(struct unistimsession *pte, char keycode)
{
FILE *f;
char count;
long offset;
switch (keycode) {
case KEY_UP:
case KEY_LEFT:
case KEY_FUNC1:
if (pte->buff_entry[2] <= 1)
return;
pte->buff_entry[2]--;
count = OpenHistory(pte, pte->buff_entry[0], &f);
if (!count)
return;
offset = ((pte->buff_entry[2] - 1) * TEXT_LENGTH_MAX * 3);
if (fseek(f, offset, SEEK_CUR)) {
display_last_error("Unable to seek history entry.");
fclose(f);
return;
}
show_entry_history(pte, &f);
break;
case KEY_DOWN:
case KEY_RIGHT:
case KEY_FUNC2:
if (pte->buff_entry[2] >= pte->buff_entry[1])
return;
pte->buff_entry[2]++;
count = OpenHistory(pte, pte->buff_entry[0], &f);
if (!count)
return;
offset = ((pte->buff_entry[2] - 1) * TEXT_LENGTH_MAX * 3);
if (fseek(f, offset, SEEK_CUR)) {
display_last_error("Unable to seek history entry.");
fclose(f);
return;
}
show_entry_history(pte, &f);
break;
case KEY_FUNC3:
if (!ReformatNumber(pte->device->lst_cid))
break;
ast_copy_string(pte->device->redial_number, pte->device->lst_cid,
sizeof(pte->device->redial_number));
key_main_page(pte, KEY_FUNC2);
break;
case KEY_FUNC4:
case KEY_HANGUP:
show_main_page(pte);
break;
case KEY_SNDHIST:
if (pte->buff_entry[0] == 'i')
show_history(pte, 'o');
else
show_main_page(pte);
break;
case KEY_RCVHIST:
if (pte->buff_entry[0] == 'i')
show_main_page(pte);
else
show_history(pte, 'i');
break;
}
return;
}
static void init_phone_step2(struct unistimsession *pte)
{
BUFFSEND;
if (unistimdebug)
ast_verb(0, "Sending S4\n");
memcpy(buffsend + SIZE_HEADER, packet_send_s4, sizeof(packet_send_s4));
send_client(SIZE_HEADER + sizeof(packet_send_s4), buffsend, pte);
send_date_time2(pte);
send_date_time3(pte);
if (unistimdebug)
ast_verb(0, "Sending S7\n");
memcpy(buffsend + SIZE_HEADER, packet_send_S7, sizeof(packet_send_S7));
send_client(SIZE_HEADER + sizeof(packet_send_S7), buffsend, pte);
if (unistimdebug)
ast_verb(0, "Sending Contrast\n");
memcpy(buffsend + SIZE_HEADER, packet_send_Contrast, sizeof(packet_send_Contrast));
if (pte->device != NULL)
buffsend[9] = pte->device->contrast;
send_client(SIZE_HEADER + sizeof(packet_send_Contrast), buffsend, pte);
if (unistimdebug)
ast_verb(0, "Sending S9\n");
memcpy(buffsend + SIZE_HEADER, packet_send_s9, sizeof(packet_send_s9));
send_client(SIZE_HEADER + sizeof(packet_send_s9), buffsend, pte);
send_no_ring(pte);
if (unistimdebug)
ast_verb(0, "Sending S7\n");
memcpy(buffsend + SIZE_HEADER, packet_send_S7, sizeof(packet_send_S7));
send_client(SIZE_HEADER + sizeof(packet_send_S7), buffsend, pte);
send_led_update(pte, 0);
send_ping(pte);
if (pte->state < STATE_MAINPAGE) {
if (autoprovisioning == AUTOPROVISIONING_TN) {
ShowExtensionPage(pte);
return;
} else {
int i;
char tmp[30];
for (i = 1; i < 6; i++)
send_favorite(i, 0, pte, "");
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Sorry, this phone is not");
send_text(TEXT_LINE1, TEXT_NORMAL, pte, "registered in unistim.cfg");
strcpy(tmp, "MAC = ");
strcat(tmp, pte->macaddr);
send_text(TEXT_LINE2, TEXT_NORMAL, pte, tmp);
send_text_status(pte, "");
send_texttitle(pte, "UNISTIM for*");
return;
}
}
show_main_page(pte);
refresh_all_favorite(pte);
if (unistimdebug)
ast_verb(0, "Sending arrow\n");
memcpy(buffsend + SIZE_HEADER, packet_send_arrow, sizeof(packet_send_arrow));
send_client(SIZE_HEADER + sizeof(packet_send_arrow), buffsend, pte);
return;
}
static void process_request(int size, unsigned char *buf, struct unistimsession *pte)
{
char tmpbuf[255];
if (memcmp
(buf + SIZE_HEADER, packet_recv_resume_connection_with_server,
sizeof(packet_recv_resume_connection_with_server)) == 0) {
rcv_resume_connection_with_server(pte);
return;
}
if (memcmp(buf + SIZE_HEADER, packet_recv_firm_version, sizeof(packet_recv_firm_version)) ==
0) {
buf[size] = 0;
if (unistimdebug)
ast_verb(0, "Got the firmware version : '%s'\n", buf + 13);
init_phone_step2(pte);
return;
}
if (memcmp(buf + SIZE_HEADER, packet_recv_mac_addr, sizeof(packet_recv_mac_addr)) == 0) {
rcv_mac_addr(pte, buf);
return;
}
if (memcmp(buf + SIZE_HEADER, packet_recv_r2, sizeof(packet_recv_r2)) == 0) {
if (unistimdebug)
ast_verb(0, "R2 received\n");
return;
}
if (pte->state < STATE_MAINPAGE) {
if (unistimdebug)
ast_verb(0, "Request not authorized in this state\n");
return;
}
if (!memcmp(buf + SIZE_HEADER, packet_recv_pressed_key, sizeof(packet_recv_pressed_key))) {
char keycode = buf[13];
if (unistimdebug)
ast_verb(0, "Key pressed : keycode = 0x%.2x - current state : %d\n", keycode,
pte->state);
switch (pte->state) {
case STATE_INIT:
if (unistimdebug)
ast_verb(0, "No keys allowed in the init state\n");
break;
case STATE_AUTHDENY:
if (unistimdebug)
ast_verb(0, "No keys allowed in authdeny state\n");
break;
case STATE_MAINPAGE:
key_main_page(pte, keycode);
break;
case STATE_DIALPAGE:
key_dial_page(pte, keycode);
break;
case STATE_RINGING:
key_ringing(pte, keycode);
break;
case STATE_CALL:
key_call(pte, keycode);
break;
case STATE_EXTENSION:
key_select_extension(pte, keycode);
break;
case STATE_SELECTCODEC:
key_select_codec(pte, keycode);
break;
case STATE_HISTORY:
key_history(pte, keycode);
break;
default:
ast_log(LOG_WARNING, "Key : Unknown state\n");
}
return;
}
if (memcmp(buf + SIZE_HEADER, packet_recv_pick_up, sizeof(packet_recv_pick_up)) == 0) {
if (unistimdebug)
ast_verb(0, "Handset off hook\n");
if (!pte->device) /* We are not yet registered (asking for a TN in AUTOPROVISIONING_TN) */
return;
pte->device->receiver_state = STATE_OFFHOOK;
if (pte->device->output == OUTPUT_HEADPHONE)
send_select_output(pte, OUTPUT_HEADPHONE, pte->device->volume, MUTE_OFF);
else
send_select_output(pte, OUTPUT_HANDSET, pte->device->volume, MUTE_OFF);
if (pte->state == STATE_RINGING)
HandleCallIncoming(pte);
else if ((pte->state == STATE_DIALPAGE) || (pte->state == STATE_CALL))
send_select_output(pte, OUTPUT_HANDSET, pte->device->volume, MUTE_OFF);
else if (pte->state == STATE_EXTENSION) /* We must have a TN before calling */
return;
else {
send_select_output(pte, OUTPUT_HANDSET, pte->device->volume, MUTE_OFF);
handle_dial_page(pte);
}
return;
}
if (memcmp(buf + SIZE_HEADER, packet_recv_hangup, sizeof(packet_recv_hangup)) == 0) {
if (unistimdebug)
ast_verb(0, "Handset on hook\n");
if (!pte->device)
return;
pte->device->receiver_state = STATE_ONHOOK;
if (pte->state == STATE_CALL)
close_call(pte);
else if (pte->device->lines->subs[SUB_REAL]->owner)
close_call(pte);
else if (pte->state == STATE_EXTENSION)
return;
else
show_main_page(pte);
return;
}
strcpy(tmpbuf, ast_inet_ntoa(pte->sin.sin_addr));
strcat(tmpbuf, " Unknown request packet\n");
if (unistimdebug)
ast_debug(1, "%s", tmpbuf);
return;
}
static void parsing(int size, unsigned char *buf, struct unistimsession *pte,
struct sockaddr_in *addr_from)
{
unsigned short *sbuf = (unsigned short *) buf;
unsigned short seq;
char tmpbuf[255];
strcpy(tmpbuf, ast_inet_ntoa(addr_from->sin_addr));
if (size < 10) {
if (size == 0) {
ast_log(LOG_WARNING, "%s Read error\n", tmpbuf);
} else {
ast_log(LOG_NOTICE, "%s Packet too short - ignoring\n", tmpbuf);
}
return;
}
if (sbuf[0] == 0xffff) { /* Starting with 0xffff ? *//* Yes, discovery packet ? */
if (size != sizeof(packet_rcv_discovery)) {
ast_log(LOG_NOTICE, "%s Invalid size of a discovery packet\n", tmpbuf);
} else {
if (memcmp(buf, packet_rcv_discovery, sizeof(packet_rcv_discovery)) == 0) {
if (unistimdebug)
ast_verb(0, "Discovery packet received - Sending Discovery ACK\n");
if (pte) { /* A session was already active for this IP ? */
if (pte->state == STATE_INIT) { /* Yes, but it's a dupe */
if (unistimdebug)
ast_verb(1, "Duplicated Discovery packet\n");
send_raw_client(sizeof(packet_send_discovery_ack),
packet_send_discovery_ack, addr_from, &pte->sout);
pte->seq_phone = (short) 0x0000; /* reset sequence number */
} else { /* No, probably a reboot, phone side */
close_client(pte); /* Cleanup the previous session */
if (create_client(addr_from))
send_raw_client(sizeof(packet_send_discovery_ack),
packet_send_discovery_ack, addr_from, &pte->sout);
}
} else {
/* Creating new entry in our phone list */
if ((pte = create_client(addr_from)))
send_raw_client(sizeof(packet_send_discovery_ack),
packet_send_discovery_ack, addr_from, &pte->sout);
}
return;
}
ast_log(LOG_NOTICE, "%s Invalid discovery packet\n", tmpbuf);
}
return;
}
if (!pte) {
if (unistimdebug)
ast_verb(0, "%s Not a discovery packet from an unknown source : ignoring\n",
tmpbuf);
return;
}
if (sbuf[0] != 0) { /* Starting with something else than 0x0000 ? */
ast_log(LOG_NOTICE, "Unknown packet received - ignoring\n");
return;
}
if (buf[5] != 2) {
ast_log(LOG_NOTICE, "%s Wrong direction : got 0x%.2x expected 0x02\n", tmpbuf,
buf[5]);
return;
}
seq = ntohs(sbuf[1]);
if (buf[4] == 1) {
ast_mutex_lock(&pte->lock);
if (unistimdebug)
ast_verb(6, "ACK received for packet #0x%.4x\n", seq);
pte->nb_retransmit = 0;
if ((pte->last_seq_ack) + 1 == seq) {
pte->last_seq_ack++;
check_send_queue(pte);
ast_mutex_unlock(&pte->lock);
return;
}
if (pte->last_seq_ack > seq) {
if (pte->last_seq_ack == 0xffff) {
ast_verb(0, "ACK at 0xffff, restarting counter.\n");
pte->last_seq_ack = 0;
} else
ast_log(LOG_NOTICE,
"%s Warning : ACK received for an already ACKed packet : #0x%.4x we are at #0x%.4x\n",
tmpbuf, seq, pte->last_seq_ack);
ast_mutex_unlock(&pte->lock);
return;
}
if (pte->seq_server < seq) {
ast_log(LOG_NOTICE,
"%s Error : ACK received for a non-existent packet : #0x%.4x\n",
tmpbuf, pte->seq_server);
ast_mutex_unlock(&pte->lock);
return;
}
if (unistimdebug)
ast_verb(0, "%s ACK gap : Received ACK #0x%.4x, previous was #0x%.4x\n",
tmpbuf, seq, pte->last_seq_ack);
pte->last_seq_ack = seq;
check_send_queue(pte);
ast_mutex_unlock(&pte->lock);
return;
}
if (buf[4] == 2) {
if (unistimdebug)
ast_verb(0, "Request received\n");
if (pte->seq_phone == seq) {
/* Send ACK */
buf[4] = 1;
buf[5] = 1;
send_raw_client(SIZE_HEADER, buf, addr_from, &pte->sout);
pte->seq_phone++;
process_request(size, buf, pte);
return;
}
if (pte->seq_phone > seq) {
ast_log(LOG_NOTICE,
"%s Warning : received a retransmitted packet : #0x%.4x (we are at #0x%.4x)\n",
tmpbuf, seq, pte->seq_phone);
/* BUG ? pte->device->seq_phone = seq; */
/* Send ACK */
buf[4] = 1;
buf[5] = 1;
send_raw_client(SIZE_HEADER, buf, addr_from, &pte->sout);
return;
}
ast_log(LOG_NOTICE,
"%s Warning : we lost a packet : received #0x%.4x (we are at #0x%.4x)\n",
tmpbuf, seq, pte->seq_phone);
return;
}
if (buf[4] == 0) {
ast_log(LOG_NOTICE, "%s Retransmit request for packet #0x%.4x\n", tmpbuf, seq);
if (pte->last_seq_ack > seq) {
ast_log(LOG_NOTICE,
"%s Error : received a request for an already ACKed packet : #0x%.4x\n",
tmpbuf, pte->last_seq_ack);
return;
}
if (pte->seq_server < seq) {
ast_log(LOG_NOTICE,
"%s Error : received a request for a non-existent packet : #0x%.4x\n",
tmpbuf, pte->seq_server);
return;
}
send_retransmit(pte);
return;
}
ast_log(LOG_NOTICE, "%s Unknown request : got 0x%.2x expected 0x00,0x01 or 0x02\n",
tmpbuf, buf[4]);
return;
}
static struct unistimsession *channel_to_session(struct ast_channel *ast)
{
struct unistim_subchannel *sub;
if (!ast) {
ast_log(LOG_WARNING, "Unistim callback function called with a null channel\n");
return NULL;
}
if (!ast->tech_pvt) {
ast_log(LOG_WARNING, "Unistim callback function called without a tech_pvt\n");
return NULL;
}
sub = ast->tech_pvt;
if (!sub->parent) {
ast_log(LOG_WARNING, "Unistim callback function called without a line\n");
return NULL;
}
if (!sub->parent->parent) {
ast_log(LOG_WARNING, "Unistim callback function called without a device\n");
return NULL;
}
if (!sub->parent->parent->session) {
ast_log(LOG_WARNING, "Unistim callback function called without a session\n");
return NULL;
}
return sub->parent->parent->session;
}
/*--- unistim_call: Initiate UNISTIM call from PBX ---*/
/* used from the dial() application */
static int unistim_call(struct ast_channel *ast, char *dest, int timeout)
{
int res = 0;
struct unistim_subchannel *sub;
struct unistimsession *session;
session = channel_to_session(ast);
if (!session) {
ast_log(LOG_ERROR, "Device not registered, cannot call %s\n", dest);
return -1;
}
sub = ast->tech_pvt;
if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
ast_log(LOG_WARNING, "unistim_call called on %s, neither down nor reserved\n",
ast->name);
return -1;
}
if (unistimdebug)
ast_verb(3, "unistim_call(%s)\n", ast->name);
session->state = STATE_RINGING;
Sendicon(TEXT_LINE0, FAV_ICON_NONE, session);
if (sub->owner) {
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
if (sub->owner->connected.id.number.valid
&& sub->owner->connected.id.number.str) {
if (session->device->height == 1) {
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
send_text(TEXT_LINE0, TEXT_NORMAL, session, sub->owner->connected.id.number.str);
} else {
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
send_text(TEXT_LINE1, TEXT_NORMAL, session, sub->owner->connected.id.number.str);
}
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
change_callerid(session, 0, sub->owner->connected.id.number.str);
} else {
if (session->device->height == 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, session, DEFAULTCALLERID);
} else {
send_text(TEXT_LINE1, TEXT_NORMAL, session, DEFAULTCALLERID);
}
change_callerid(session, 0, DEFAULTCALLERID);
}
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
if (sub->owner->connected.id.name.valid
&& sub->owner->connected.id.name.str) {
send_text(TEXT_LINE0, TEXT_NORMAL, session, sub->owner->connected.id.name.str);
change_callerid(session, 1, sub->owner->connected.id.name.str);
} else {
send_text(TEXT_LINE0, TEXT_NORMAL, session, DEFAULTCALLERNAME);
change_callerid(session, 1, DEFAULTCALLERNAME);
}
}
send_text(TEXT_LINE2, TEXT_NORMAL, session, "is calling you.");
send_text_status(session, "Accept Ignore");
if (sub->ringstyle == -1)
send_ring(session, session->device->ringvolume, session->device->ringstyle);
else {
if (sub->ringvolume == -1)
send_ring(session, session->device->ringvolume, sub->ringstyle);
else
send_ring(session, sub->ringvolume, sub->ringstyle);
}
change_favorite_icon(session, FAV_ICON_SPEAKER_ONHOOK_BLACK + FAV_BLINK_FAST);
ast_setstate(ast, AST_STATE_RINGING);
ast_queue_control(ast, AST_CONTROL_RINGING);
return res;
}
/*--- unistim_hangup: Hangup UNISTIM call */
static int unistim_hangup(struct ast_channel *ast)
{
struct unistim_subchannel *sub;
struct unistim_line *l;
struct unistimsession *s;
s = channel_to_session(ast);
sub = ast->tech_pvt;
if (!s) {
ast_debug(1, "Asked to hangup channel not connected\n");
ast_mutex_lock(&sub->lock);
sub->owner = NULL;
ast->tech_pvt = NULL;
sub->alreadygone = 0;
ast_mutex_unlock(&sub->lock);
if (sub->rtp) {
if (unistimdebug)
ast_verb(0, "Destroying RTP session\n");
ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
return 0;
}
l = sub->parent;
if (unistimdebug)
ast_verb(0, "unistim_hangup(%s) on %s@%s\n", ast->name, l->name, l->parent->name);
if ((l->subs[SUB_THREEWAY]) && (sub->subtype == SUB_REAL)) {
if (unistimdebug)
ast_verb(0, "Real call disconnected while talking to threeway\n");
sub->owner = NULL;
ast->tech_pvt = NULL;
return 0;
}
if ((l->subs[SUB_REAL]->owner) && (sub->subtype == SUB_THREEWAY) &&
(sub->alreadygone == 0)) {
if (unistimdebug)
ast_verb(0, "threeway call disconnected, switching to real call\n");
send_text(TEXT_LINE0, TEXT_NORMAL, s, "Three way call canceled,");
send_text(TEXT_LINE1, TEXT_NORMAL, s, "switching back to");
send_text(TEXT_LINE2, TEXT_NORMAL, s, "previous call.");
send_text_status(s, "Hangup Transf");
ast_moh_stop(ast_bridged_channel(l->subs[SUB_REAL]->owner));
swap_subs(l, SUB_THREEWAY, SUB_REAL);
l->parent->moh = 0;
ast_mutex_lock(&sub->lock);
sub->owner = NULL;
ast->tech_pvt = NULL;
ast_mutex_unlock(&sub->lock);
unalloc_sub(l, SUB_THREEWAY);
return 0;
}
ast_mutex_lock(&sub->lock);
sub->owner = NULL;
ast->tech_pvt = NULL;
sub->alreadygone = 0;
ast_mutex_unlock(&sub->lock);
if (!s) {
if (unistimdebug)
ast_verb(0, "Asked to hangup channel not connected (no session)\n");
if (sub->rtp) {
if (unistimdebug)
ast_verb(0, "Destroying RTP session\n");
ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
return 0;
}
if (sub->subtype == SUB_REAL) {
/* Stop the silence generator */
if (s->device->silence_generator) {
if (unistimdebug)
ast_verb(0, "Stopping silence generator\n");
if (sub->owner)
ast_channel_stop_silence_generator(sub->owner,
s->device->silence_generator);
else
ast_log(LOG_WARNING,
"Trying to stop silence generator on a null channel !\n");
s->device->silence_generator = NULL;
}
}
l->parent->moh = 0;
send_no_ring(s);
send_end_call(s);
if (sub->rtp) {
if (unistimdebug)
ast_verb(0, "Destroying RTP session\n");
ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
} else if (unistimdebug)
ast_verb(0, "No RTP session to destroy\n");
if (l->subs[SUB_THREEWAY]) {
if (unistimdebug)
ast_verb(0, "Cleaning other subchannels\n");
unalloc_sub(l, SUB_THREEWAY);
}
if (s->state == STATE_RINGING)
cancel_dial(s);
else if (s->state == STATE_CALL)
close_call(s);
return 0;
}
/*--- unistim_answer: Answer UNISTIM call */
static int unistim_answer(struct ast_channel *ast)
{
int res = 0;
struct unistim_subchannel *sub;
struct unistim_line *l;
struct unistimsession *s;
s = channel_to_session(ast);
if (!s) {
ast_log(LOG_WARNING, "unistim_answer on a disconnected device ?\n");
return -1;
}
sub = ast->tech_pvt;
l = sub->parent;
if ((!sub->rtp) && (!l->subs[SUB_THREEWAY]))
start_rtp(sub);
if (unistimdebug)
ast_verb(0, "unistim_answer(%s) on %s@%s-%d\n", ast->name, l->name,
l->parent->name, sub->subtype);
send_text(TEXT_LINE2, TEXT_NORMAL, l->parent->session, "is now on-line");
if (l->subs[SUB_THREEWAY])
send_text_status(l->parent->session, "Transf Cancel");
else
send_text_status(l->parent->session, "Hangup Transf");
send_start_timer(l->parent->session);
if (ast->_state != AST_STATE_UP)
ast_setstate(ast, AST_STATE_UP);
return res;
}
/*--- unistimsock_read: Read data from UNISTIM socket ---*/
/* Successful messages is connected to UNISTIM call and forwarded to parsing() */
static int unistimsock_read(int *id, int fd, short events, void *ignore)
{
struct sockaddr_in addr_from = { 0, };
struct unistimsession *cur = NULL;
int found = 0;
int tmp = 0;
int dw_num_bytes_rcvd;
#ifdef DUMP_PACKET
int dw_num_bytes_rcvdd;
char iabuf[INET_ADDRSTRLEN];
#endif
dw_num_bytes_rcvd =
recvfrom(unistimsock, buff, SIZE_PAGE, 0, (struct sockaddr *) &addr_from,
&size_addr_from);
if (dw_num_bytes_rcvd == -1) {
if (errno == EAGAIN)
ast_log(LOG_NOTICE, "UNISTIM: Received packet with bad UDP checksum\n");
else if (errno != ECONNREFUSED)
ast_log(LOG_WARNING, "Recv error %d (%s)\n", errno, strerror(errno));
return 1;
}
/* Looking in the phone list if we already have a registration for him */
ast_mutex_lock(&sessionlock);
cur = sessions;
while (cur) {
if (cur->sin.sin_addr.s_addr == addr_from.sin_addr.s_addr) {
found = 1;
break;
}
tmp++;
cur = cur->next;
}
ast_mutex_unlock(&sessionlock);
#ifdef DUMP_PACKET
if (unistimdebug)
ast_verb(0, "\n*** Dump %d bytes from %s - phone_table[%d] ***\n",
dw_num_bytes_rcvd, ast_inet_ntoa(addr_from.sin_addr), tmp);
for (dw_num_bytes_rcvdd = 0; dw_num_bytes_rcvdd < dw_num_bytes_rcvd;
dw_num_bytes_rcvdd++)
ast_verb(0, "%.2x ", (unsigned char) buff[dw_num_bytes_rcvdd]);
ast_verb(0, "\n******************************************\n");
#endif
if (!found) {
if (unistimdebug)
ast_verb(0, "Received a packet from an unknown source\n");
parsing(dw_num_bytes_rcvd, buff, NULL, (struct sockaddr_in *) &addr_from);
} else
parsing(dw_num_bytes_rcvd, buff, cur, (struct sockaddr_in *) &addr_from);
return 1;
}
static struct ast_frame *unistim_rtp_read(const struct ast_channel *ast,
const struct unistim_subchannel *sub)
{
/* Retrieve audio/etc from channel. Assumes sub->lock is already held. */
struct ast_frame *f;
if (!ast) {
ast_log(LOG_WARNING, "Channel NULL while reading\n");
return &ast_null_frame;
}
if (!sub->rtp) {
ast_log(LOG_WARNING, "RTP handle NULL while reading on subchannel %d\n",
sub->subtype);
return &ast_null_frame;
}
switch (ast->fdno) {
case 0:
f = ast_rtp_instance_read(sub->rtp, 0); /* RTP Audio */
break;
case 1:
f = ast_rtp_instance_read(sub->rtp, 1); /* RTCP Control Channel */
break;
default:
f = &ast_null_frame;
}
if (sub->owner) {
/* We already hold the channel lock */
if (f->frametype == AST_FRAME_VOICE) {
if (!(ast_format_cap_iscompatible(sub->owner->nativeformats, &f->subclass.format))) {
char tmp[256];
ast_debug(1,
"Oooh, format changed from %s to %s\n",
ast_getformatname_multiple(tmp, sizeof(tmp), sub->owner->nativeformats),
ast_getformatname(&f->subclass.format));
ast_format_cap_set(sub->owner->nativeformats, &f->subclass.format);
ast_set_read_format(sub->owner, &sub->owner->readformat);
ast_set_write_format(sub->owner, &sub->owner->writeformat);
}
}
}
return f;
}
static struct ast_frame *unistim_read(struct ast_channel *ast)
{
struct ast_frame *fr;
struct unistim_subchannel *sub = ast->tech_pvt;
ast_mutex_lock(&sub->lock);
fr = unistim_rtp_read(ast, sub);
ast_mutex_unlock(&sub->lock);
return fr;
}
static int unistim_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct unistim_subchannel *sub = ast->tech_pvt;
int res = 0;
if (frame->frametype != AST_FRAME_VOICE) {
if (frame->frametype == AST_FRAME_IMAGE)
return 0;
else {
ast_log(LOG_WARNING, "Can't send %d type frames with unistim_write\n",
frame->frametype);
return 0;
}
} else {
if (!(ast_format_cap_iscompatible(ast->nativeformats, &frame->subclass.format))) {
char tmp[256];
ast_log(LOG_WARNING,
"Asked to transmit frame type %s, while native formats is %s (read/write = (%s/%s)\n",
ast_getformatname(&frame->subclass.format),
ast_getformatname_multiple(tmp, sizeof(tmp), ast->nativeformats),
ast_getformatname(&ast->readformat),
ast_getformatname(&ast->writeformat));
return -1;
}
}
if (sub) {
ast_mutex_lock(&sub->lock);
if (sub->rtp) {
res = ast_rtp_instance_write(sub->rtp, frame);
}
ast_mutex_unlock(&sub->lock);
}
return res;
}
static int unistim_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
struct unistim_subchannel *p = newchan->tech_pvt;
struct unistim_line *l = p->parent;
ast_mutex_lock(&p->lock);
ast_debug(1, "New owner for channel USTM/%s@%s-%d is %s\n", l->name,
l->parent->name, p->subtype, newchan->name);
if (p->owner != oldchan) {
ast_log(LOG_WARNING, "old channel wasn't %s (%p) but was %s (%p)\n",
oldchan->name, oldchan, p->owner->name, p->owner);
return -1;
}
p->owner = newchan;
ast_mutex_unlock(&p->lock);
return 0;
}
static char *control2str(int ind)
{
switch (ind) {
case AST_CONTROL_HANGUP:
return "Other end has hungup";
case AST_CONTROL_RING:
return "Local ring";
case AST_CONTROL_RINGING:
return "Remote end is ringing";
case AST_CONTROL_ANSWER:
return "Remote end has answered";
case AST_CONTROL_BUSY:
return "Remote end is busy";
case AST_CONTROL_TAKEOFFHOOK:
return "Make it go off hook";
case AST_CONTROL_OFFHOOK:
return "Line is off hook";
case AST_CONTROL_CONGESTION:
return "Congestion (circuits busy)";
case AST_CONTROL_FLASH:
return "Flash hook";
case AST_CONTROL_WINK:
return "Wink";
case AST_CONTROL_OPTION:
return "Set a low-level option";
case AST_CONTROL_RADIO_KEY:
return "Key Radio";
case AST_CONTROL_RADIO_UNKEY:
return "Un-Key Radio";
case -1:
return "Stop tone";
}
return "UNKNOWN";
}
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
static void in_band_indication(struct ast_channel *ast, const struct ast_tone_zone *tz,
const char *indication)
{
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
struct ast_tone_zone_sound *ts = NULL;
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
if ((ts = ast_get_indication_tone(tz, indication))) {
ast_playtones_start(ast, 0, ts->data, 1);
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
ts = ast_tone_zone_sound_unref(ts);
} else {
ast_log(LOG_WARNING, "Unable to get indication tone for %s\n", indication);
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
}
}
static int unistim_indicate(struct ast_channel *ast, int ind, const void *data,
size_t datalen)
{
struct unistim_subchannel *sub;
struct unistim_line *l;
struct unistimsession *s;
if (unistimdebug) {
ast_verb(3, "Asked to indicate '%s' condition on channel %s\n",
control2str(ind), ast->name);
}
s = channel_to_session(ast);
if (!s)
return -1;
sub = ast->tech_pvt;
l = sub->parent;
switch (ind) {
case AST_CONTROL_RINGING:
if (ast->_state != AST_STATE_UP) {
send_text(TEXT_LINE2, TEXT_NORMAL, s, "Ringing...");
in_band_indication(ast, l->parent->tz, "ring");
s->device->missed_call = -1;
break;
}
return -1;
case AST_CONTROL_BUSY:
if (ast->_state != AST_STATE_UP) {
sub->alreadygone = 1;
send_text(TEXT_LINE2, TEXT_NORMAL, s, "Busy");
in_band_indication(ast, l->parent->tz, "busy");
s->device->missed_call = -1;
break;
}
return -1;
case AST_CONTROL_CONGESTION:
if (ast->_state != AST_STATE_UP) {
sub->alreadygone = 1;
send_text(TEXT_LINE2, TEXT_NORMAL, s, "Congestion");
in_band_indication(ast, l->parent->tz, "congestion");
s->device->missed_call = -1;
break;
}
return -1;
case AST_CONTROL_HOLD:
ast_moh_start(ast, data, NULL);
break;
case AST_CONTROL_UNHOLD:
ast_moh_stop(ast);
break;
case AST_CONTROL_PROGRESS:
case AST_CONTROL_SRCUPDATE:
break;
case -1:
ast_playtones_stop(ast);
s->device->missed_call = 0;
break;
case AST_CONTROL_PROCEEDING:
break;
default:
ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);
return -1;
}
return 0;
}
static struct unistim_subchannel *find_subchannel_by_name(const char *dest)
{
struct unistim_line *l;
struct unistim_device *d;
char line[256];
char *at;
char *device;
ast_copy_string(line, dest, sizeof(line));
at = strchr(line, '@');
if (!at) {
ast_log(LOG_NOTICE, "Device '%s' has no @ (at) sign!\n", dest);
return NULL;
}
*at = '\0';
at++;
device = at;
ast_mutex_lock(&devicelock);
d = devices;
at = strchr(device, '/'); /* Extra options ? */
if (at)
*at = '\0';
while (d) {
if (!strcasecmp(d->name, device)) {
if (unistimdebug)
ast_verb(0, "Found device: %s\n", d->name);
/* Found the device */
l = d->lines;
while (l) {
/* Search for the right line */
if (!strcasecmp(l->name, line)) {
l->subs[SUB_REAL]->ringvolume = -1;
l->subs[SUB_REAL]->ringstyle = -1;
if (at) { /* Other options ? */
at++; /* Skip slash */
if (*at == 'r') { /* distinctive ring */
at++;
if ((*at < '0') || (*at > '7')) /* ring style */
ast_log(LOG_WARNING, "Invalid ring selection (%s)", at);
else {
char ring_volume = -1;
char ring_style = *at - '0';
at++;
if ((*at >= '0') && (*at <= '3')) /* ring volume */
ring_volume = *at - '0';
if (unistimdebug)
ast_verb(0, "Distinctive ring : style #%d volume %d\n",
ring_style, ring_volume);
l->subs[SUB_REAL]->ringvolume = ring_volume;
l->subs[SUB_REAL]->ringstyle = ring_style;
}
}
}
ast_mutex_unlock(&devicelock);
return l->subs[SUB_REAL];
}
l = l->next;
}
}
d = d->next;
}
/* Device not found */
ast_mutex_unlock(&devicelock);
return NULL;
}
static int unistim_senddigit_begin(struct ast_channel *ast, char digit)
{
struct unistimsession *pte = channel_to_session(ast);
if (!pte)
return -1;
return unistim_do_senddigit(pte, digit);
}
static int unistim_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration)
{
struct unistimsession *pte = channel_to_session(ast);
struct ast_frame f = { 0, };
struct unistim_subchannel *sub;
sub = pte->device->lines->subs[SUB_REAL];
if (!sub->owner || sub->alreadygone) {
ast_log(LOG_WARNING, "Unable to find subchannel in dtmf senddigit_end\n");
return -1;
}
if (unistimdebug)
ast_verb(0, "Send Digit off %c\n", digit);
if (!pte)
return -1;
send_tone(pte, 0, 0);
f.frametype = AST_FRAME_DTMF;
f.subclass.integer = digit;
f.src = "unistim";
ast_queue_frame(sub->owner, &f);
return 0;
}
/*--- unistim_sendtext: Display a text on the phone screen ---*/
/* Called from PBX core text message functions */
static int unistim_sendtext(struct ast_channel *ast, const char *text)
{
struct unistimsession *pte = channel_to_session(ast);
int size;
char tmp[TEXT_LENGTH_MAX + 1];
if (unistimdebug)
ast_verb(0, "unistim_sendtext called\n");
if (!text) {
ast_log(LOG_WARNING, "unistim_sendtext called with a null text\n");
return 1;
}
size = strlen(text);
if (text[0] == '@') {
int pos = 0, i = 1, tok = 0, sz = 0;
char label[11];
char number[16];
char icon = '\0';
char cur = '\0';
memset(label, 0, 11);
memset(number, 0, 16);
while (text[i]) {
cur = text[i++];
switch (tok) {
case 0:
if ((cur < '0') && (cur > '5')) {
ast_log(LOG_WARNING,
"sendtext failed : position must be a number beetween 0 and 5\n");
return 1;
}
pos = cur - '0';
tok = 1;
continue;
case 1:
if (cur != '@') {
ast_log(LOG_WARNING, "sendtext failed : invalid position\n");
return 1;
}
tok = 2;
continue;
case 2:
if ((cur < '3') && (cur > '6')) {
ast_log(LOG_WARNING,
"sendtext failed : icon must be a number beetween 32 and 63 (first digit invalid)\n");
return 1;
}
icon = (cur - '0') * 10;
tok = 3;
continue;
case 3:
if ((cur < '0') && (cur > '9')) {
ast_log(LOG_WARNING,
"sendtext failed : icon must be a number beetween 32 and 63 (second digit invalid)\n");
return 1;
}
icon += (cur - '0');
tok = 4;
continue;
case 4:
if (cur != '@') {
ast_log(LOG_WARNING,
"sendtext failed : icon must be a number beetween 32 and 63 (too many digits)\n");
return 1;
}
tok = 5;
continue;
case 5:
if (cur == '@') {
tok = 6;
sz = 0;
continue;
}
if (sz > 10)
continue;
label[sz] = cur;
sz++;
continue;
case 6:
if (sz > 15) {
ast_log(LOG_WARNING,
"sendtext failed : extension too long = %d (15 car max)\n",
sz);
return 1;
}
number[sz] = cur;
sz++;
continue;
}
}
if (tok != 6) {
ast_log(LOG_WARNING, "sendtext failed : incomplet command\n");
return 1;
}
if (!pte->device) {
ast_log(LOG_WARNING, "sendtext failed : no device ?\n");
return 1;
}
strcpy(pte->device->softkeylabel[pos], label);
strcpy(pte->device->softkeynumber[pos], number);
pte->device->softkeyicon[pos] = icon;
send_favorite(pos, icon, pte, label);
return 0;
}
if (size <= TEXT_LENGTH_MAX * 2) {
if (pte->device->height == 1) {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, text);
} else {
send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Message :");
send_text(TEXT_LINE1, TEXT_NORMAL, pte, text);
}
if (size <= TEXT_LENGTH_MAX) {
send_text(TEXT_LINE2, TEXT_NORMAL, pte, "");
return 0;
}
memcpy(tmp, text + TEXT_LENGTH_MAX, TEXT_LENGTH_MAX);
tmp[sizeof(tmp) - 1] = '\0';
send_text(TEXT_LINE2, TEXT_NORMAL, pte, tmp);
return 0;
}
send_text(TEXT_LINE0, TEXT_NORMAL, pte, text);
memcpy(tmp, text + TEXT_LENGTH_MAX, TEXT_LENGTH_MAX);
tmp[sizeof(tmp) - 1] = '\0';
send_text(TEXT_LINE1, TEXT_NORMAL, pte, tmp);
memcpy(tmp, text + TEXT_LENGTH_MAX * 2, TEXT_LENGTH_MAX);
tmp[sizeof(tmp) - 1] = '\0';
send_text(TEXT_LINE2, TEXT_NORMAL, pte, tmp);
return 0;
}
/*--- unistim_send_mwi_to_peer: Send message waiting indication ---*/
static int unistim_send_mwi_to_peer(struct unistimsession *s, unsigned int tick)
{
struct ast_event *event;
int new;
char *mailbox, *context;
struct unistim_line *peer = s->device->lines;
context = mailbox = ast_strdupa(peer->mailbox);
strsep(&context, "@");
if (ast_strlen_zero(context))
context = "default";
event = ast_event_get_cached(AST_EVENT_MWI,
AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox,
AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, context,
AST_EVENT_IE_END);
if (event) {
new = ast_event_get_ie_uint(event, AST_EVENT_IE_NEWMSGS);
ast_event_destroy(event);
} else { /* Fall back on checking the mailbox directly */
new = ast_app_has_voicemail(peer->mailbox, "INBOX");
}
peer->nextmsgcheck = tick + TIMER_MWI;
/* Return now if it's the same thing we told them last time */
if (new == peer->lastmsgssent) {
return 0;
}
peer->lastmsgssent = new;
if (new == 0) {
send_led_update(s, 0);
} else {
send_led_update(s, 1);
}
return 0;
}
/*--- unistim_new: Initiate a call in the UNISTIM channel */
/* called from unistim_request (calls from the pbx ) */
static struct ast_channel *unistim_new(struct unistim_subchannel *sub, int state, const char *linkedid)
{
struct ast_channel *tmp;
struct unistim_line *l;
struct ast_format tmpfmt;
if (!sub) {
ast_log(LOG_WARNING, "subchannel null in unistim_new\n");
return NULL;
}
if (!sub->parent) {
ast_log(LOG_WARNING, "no line for subchannel %p\n", sub);
return NULL;
}
l = sub->parent;
Convert the ast_channel data structure over to the astobj2 framework. There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24 14:04:26 +00:00
tmp = ast_channel_alloc(1, state, l->cid_num, NULL, l->accountcode, l->exten,
l->context, linkedid, l->amaflags, "%s@%s-%d", l->name, l->parent->name, sub->subtype);
if (unistimdebug)
ast_verb(0, "unistim_new sub=%d (%p) chan=%p\n", sub->subtype, sub, tmp);
if (!tmp) {
ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
return NULL;
}
ast_format_cap_copy(tmp->nativeformats, l->cap);
if (ast_format_cap_is_empty(tmp->nativeformats))
ast_format_cap_copy(tmp->nativeformats, global_cap);
ast_best_codec(tmp->nativeformats, &tmpfmt);
if (unistimdebug) {
char tmp1[256], tmp2[256], tmp3[256];
ast_verb(0, "Best codec = %s from nativeformats %s (line cap=%s global=%s)\n",
ast_getformatname(&tmpfmt),
ast_getformatname_multiple(tmp1, sizeof(tmp1), tmp->nativeformats),
ast_getformatname_multiple(tmp2, sizeof(tmp2), l->cap),
ast_getformatname_multiple(tmp3, sizeof(tmp3), global_cap));
}
if ((sub->rtp) && (sub->subtype == 0)) {
if (unistimdebug)
ast_verb(0, "New unistim channel with a previous rtp handle ?\n");
tmp->fds[0] = ast_rtp_instance_fd(sub->rtp, 0);
tmp->fds[1] = ast_rtp_instance_fd(sub->rtp, 1);
}
if (sub->rtp)
ast_jb_configure(tmp, &global_jbconf);
/* tmp->type = type; */
ast_setstate(tmp, state);
if (state == AST_STATE_RING)
tmp->rings = 1;
tmp->adsicpe = AST_ADSI_UNAVAILABLE;
ast_format_copy(&tmp->writeformat, &tmpfmt);
ast_format_copy(&tmp->rawwriteformat, &tmpfmt);
ast_format_copy(&tmp->readformat, &tmpfmt);
ast_format_copy(&tmp->rawreadformat, &tmpfmt);
tmp->tech_pvt = sub;
tmp->tech = &unistim_tech;
if (!ast_strlen_zero(l->language))
ast_string_field_set(tmp, language, l->language);
sub->owner = tmp;
ast_mutex_lock(&usecnt_lock);
usecnt++;
ast_mutex_unlock(&usecnt_lock);
ast_update_use_count();
tmp->callgroup = l->callgroup;
tmp->pickupgroup = l->pickupgroup;
ast_string_field_set(tmp, call_forward, l->parent->call_forward);
if (!ast_strlen_zero(l->cid_num)) {
char *name, *loc, *instr;
instr = ast_strdup(l->cid_num);
if (instr) {
ast_callerid_parse(instr, &name, &loc);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
tmp->caller.id.number.valid = 1;
ast_free(tmp->caller.id.number.str);
tmp->caller.id.number.str = ast_strdup(loc);
tmp->caller.id.name.valid = 1;
ast_free(tmp->caller.id.name.str);
tmp->caller.id.name.str = ast_strdup(name);
ast_free(instr);
}
}
tmp->priority = 1;
if (state != AST_STATE_DOWN) {
if (unistimdebug)
ast_verb(0, "Starting pbx in unistim_new\n");
if (ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
ast_hangup(tmp);
tmp = NULL;
}
}
return tmp;
}
static void *do_monitor(void *data)
{
struct unistimsession *cur = NULL;
unsigned int dw_timeout = 0;
unsigned int tick;
int res;
int reloading;
/* Add an I/O event to our UDP socket */
if (unistimsock > -1)
ast_io_add(io, unistimsock, unistimsock_read, AST_IO_IN, NULL);
/* This thread monitors our UDP socket and timers */
for (;;) {
/* This loop is executed at least every IDLE_WAITus (1s) or every time a packet is received */
/* Looking for the smallest time-out value */
tick = get_tick_count();
dw_timeout = UINT_MAX;
ast_mutex_lock(&sessionlock);
cur = sessions;
DEBUG_TIMER("checking timeout for session %p with tick = %u\n", cur, tick);
while (cur) {
DEBUG_TIMER("checking timeout for session %p timeout = %u\n", cur,
cur->timeout);
/* Check if we have miss something */
if (cur->timeout <= tick) {
DEBUG_TIMER("Event for session %p\n", cur);
/* If the queue is empty, send a ping */
if (cur->last_buf_available == 0)
send_ping(cur);
else {
if (send_retransmit(cur)) {
DEBUG_TIMER("The chained link was modified, restarting...\n");
cur = sessions;
dw_timeout = UINT_MAX;
continue;
}
}
}
if (dw_timeout > cur->timeout - tick)
dw_timeout = cur->timeout - tick;
/* Checking if the phone is logged on for a new MWI */
if (cur->device) {
if ((!ast_strlen_zero(cur->device->lines->mailbox)) &&
((tick >= cur->device->lines->nextmsgcheck))) {
DEBUG_TIMER("Checking mailbox for MWI\n");
unistim_send_mwi_to_peer(cur, tick);
break;
}
}
cur = cur->next;
}
ast_mutex_unlock(&sessionlock);
DEBUG_TIMER("Waiting for %dus\n", dw_timeout);
res = dw_timeout;
/* We should not wait more than IDLE_WAIT */
if ((res < 0) || (res > IDLE_WAIT))
res = IDLE_WAIT;
/* Wait for UDP messages for a maximum of res us */
res = ast_io_wait(io, res); /* This function will call unistimsock_read if a packet is received */
/* Check for a reload request */
ast_mutex_lock(&unistim_reload_lock);
reloading = unistim_reloading;
unistim_reloading = 0;
ast_mutex_unlock(&unistim_reload_lock);
if (reloading) {
ast_verb(1, "Reloading unistim.conf...\n");
reload_config();
}
pthread_testcancel();
}
/* Never reached */
return NULL;
}
/*--- restart_monitor: Start the channel monitor thread ---*/
static int restart_monitor(void)
{
pthread_attr_t attr;
/* 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) {
/* Wake up the thread */
pthread_kill(monitor_thread, SIGURG);
} else {
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
/* Start a new monitor */
if (ast_pthread_create(&monitor_thread, &attr, 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;
}
/*--- unistim_request: PBX interface function ---*/
/* UNISTIM calls initiated by the PBX arrive here */
static struct ast_channel *unistim_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data,
int *cause)
{
struct unistim_subchannel *sub;
struct ast_channel *tmpc = NULL;
char tmp[256];
char tmp2[256];
char *dest = data;
if (!(ast_format_cap_has_joint(cap, global_cap))) {
ast_log(LOG_NOTICE,
"Asked to get a channel of unsupported format %s while capability is %s\n",
ast_getformatname_multiple(tmp2, sizeof(tmp2), cap), ast_getformatname_multiple(tmp, sizeof(tmp), global_cap));
return NULL;
}
ast_copy_string(tmp, dest, sizeof(tmp));
if (ast_strlen_zero(tmp)) {
ast_log(LOG_NOTICE, "Unistim channels require a device\n");
return NULL;
}
sub = find_subchannel_by_name(tmp);
if (!sub) {
ast_log(LOG_NOTICE, "No available lines on: %s\n", dest);
*cause = AST_CAUSE_CONGESTION;
return NULL;
}
ast_verb(3, "unistim_request(%s)\n", tmp);
/* Busy ? */
if (sub->owner) {
if (unistimdebug)
ast_verb(0, "Can't create channel : Busy !\n");
*cause = AST_CAUSE_BUSY;
return NULL;
}
ast_format_cap_copy(sub->parent->cap, cap);
tmpc = unistim_new(sub, AST_STATE_DOWN, requestor ? requestor->linkedid : NULL);
if (!tmpc)
ast_log(LOG_WARNING, "Unable to make channel for '%s'\n", tmp);
if (unistimdebug)
ast_verb(0, "unistim_request owner = %p\n", sub->owner);
restart_monitor();
/* and finish */
return tmpc;
}
static char *unistim_info(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct unistim_device *device = devices;
struct unistim_line *line;
struct unistim_subchannel *sub;
struct unistimsession *s;
int i;
struct ast_channel *tmp;
switch (cmd) {
case CLI_INIT:
e->command = "unistim show info";
e->usage =
"Usage: unistim show info\n"
" Dump internal structures.\n";
return NULL;
case CLI_GENERATE:
return NULL; /* no completion */
}
if (a->argc != e->args)
return CLI_SHOWUSAGE;
ast_cli(a->fd, "Dumping internal structures :\ndevice\n->line\n-->sub\n");
while (device) {
ast_cli(a->fd, "\nname=%s id=%s line=%p ha=%p sess=%p device=%p\n",
device->name, device->id, device->lines, device->ha, device->session,
device);
line = device->lines;
while (line) {
char tmp2[256];
ast_cli(a->fd,
"->name=%s fullname=%s exten=%s callid=%s cap=%s device=%p line=%p\n",
line->name, line->fullname, line->exten, line->cid_num,
ast_getformatname_multiple(tmp2, sizeof(tmp2), line->cap), line->parent, line);
for (i = 0; i < MAX_SUBS; i++) {
sub = line->subs[i];
if (!sub)
continue;
if (!sub->owner)
tmp = (void *) -42;
else
tmp = sub->owner->_bridge;
if (sub->subtype != i)
ast_cli(a->fd, "Warning ! subchannel->subs[%d] have a subtype=%d\n", i,
sub->subtype);
ast_cli(a->fd,
"-->subtype=%d chan=%p rtp=%p bridge=%p line=%p alreadygone=%d\n",
sub->subtype, sub->owner, sub->rtp, tmp, sub->parent,
sub->alreadygone);
}
line = line->next;
}
device = device->next;
}
ast_cli(a->fd, "\nSessions:\n");
ast_mutex_lock(&sessionlock);
s = sessions;
while (s) {
ast_cli(a->fd,
"sin=%s timeout=%u state=%d macaddr=%s device=%p session=%p\n",
ast_inet_ntoa(s->sin.sin_addr), s->timeout, s->state, s->macaddr,
s->device, s);
s = s->next;
}
ast_mutex_unlock(&sessionlock);
return CLI_SUCCESS;
}
static char *unistim_sp(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
BUFFSEND;
struct unistim_subchannel *sub;
int i, j = 0, len;
unsigned char c, cc;
char tmp[256];
switch (cmd) {
case CLI_INIT:
e->command = "unistim send packet";
e->usage =
"Usage: unistim send packet USTM/line@name hexa\n"
" unistim send packet USTM/1000@hans 19040004\n";
return NULL;
case CLI_GENERATE:
return NULL; /* no completion */
}
if (a->argc < 5)
return CLI_SHOWUSAGE;
if (strlen(a->argv[3]) < 9)
return CLI_SHOWUSAGE;
len = strlen(a->argv[4]);
if (len % 2)
return CLI_SHOWUSAGE;
ast_copy_string(tmp, a->argv[3] + 5, sizeof(tmp));
sub = find_subchannel_by_name(tmp);
if (!sub) {
ast_cli(a->fd, "Can't find '%s'\n", tmp);
return CLI_SUCCESS;
}
if (!sub->parent->parent->session) {
ast_cli(a->fd, "'%s' is not connected\n", tmp);
return CLI_SUCCESS;
}
ast_cli(a->fd, "Sending '%s' to %s (%p)\n", a->argv[4], tmp, sub->parent->parent->session);
for (i = 0; i < len; i++) {
c = a->argv[4][i];
if (c >= 'a')
c -= 'a' - 10;
else
c -= '0';
i++;
cc = a->argv[4][i];
if (cc >= 'a')
cc -= 'a' - 10;
else
cc -= '0';
tmp[j++] = (c << 4) | cc;
}
memcpy(buffsend + SIZE_HEADER, tmp, j);
send_client(SIZE_HEADER + j, buffsend, sub->parent->parent->session);
return CLI_SUCCESS;
}
static char *unistim_do_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "unistim set debug {on|off}";
e->usage =
"Usage: unistim set debug\n"
" Display debug messages.\n";
return NULL;
case CLI_GENERATE:
return NULL; /* no completion */
}
if (a->argc != e->args)
return CLI_SHOWUSAGE;
if (!strcasecmp(a->argv[3], "on")) {
unistimdebug = 1;
ast_cli(a->fd, "UNISTIM Debugging Enabled\n");
} else if (!strcasecmp(a->argv[3], "off")) {
unistimdebug = 0;
ast_cli(a->fd, "UNISTIM Debugging Disabled\n");
} else
return CLI_SHOWUSAGE;
return CLI_SUCCESS;
}
/*! \brief --- unistim_reload: Force reload of module from cli ---
* Runs in the asterisk main thread, so don't do anything useful
* but setting a flag and waiting for do_monitor to do the job
* in our thread */
static char *unistim_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "unistim reload";
e->usage =
"Usage: unistim reload\n"
" Reloads UNISTIM configuration from unistim.conf\n";
return NULL;
case CLI_GENERATE:
return NULL; /* no completion */
}
if (e && a && a->argc != e->args)
return CLI_SHOWUSAGE;
if (unistimdebug)
ast_verb(0, "reload unistim\n");
ast_mutex_lock(&unistim_reload_lock);
if (!unistim_reloading)
unistim_reloading = 1;
ast_mutex_unlock(&unistim_reload_lock);
restart_monitor();
return CLI_SUCCESS;
}
static struct ast_cli_entry unistim_cli[] = {
AST_CLI_DEFINE(unistim_reload, "Reload UNISTIM configuration"),
AST_CLI_DEFINE(unistim_info, "Show UNISTIM info"),
AST_CLI_DEFINE(unistim_sp, "Send packet (for reverse engineering)"),
AST_CLI_DEFINE(unistim_do_debug, "Toggle UNITSTIM debugging"),
};
static void unquote(char *out, const char *src, int maxlen)
{
int len = strlen(src);
if (!len)
return;
if ((len > 1) && src[0] == '\"') {
/* This is a quoted string */
src++;
/* Don't take more than what's there */
len--;
if (maxlen > len - 1)
maxlen = len - 1;
memcpy(out, src, maxlen);
((char *) out)[maxlen] = '\0';
} else
memcpy(out, src, maxlen);
return;
}
static int ParseBookmark(const char *text, struct unistim_device *d)
{
char line[256];
char *at;
char *number;
char *icon;
int p;
int len = strlen(text);
ast_copy_string(line, text, sizeof(line));
/* Position specified ? */
if ((len > 2) && (line[1] == '@')) {
p = line[0];
if ((p >= '0') && (p <= '5'))
p -= '0';
else {
ast_log(LOG_WARNING,
"Invalid position for bookmark : must be between 0 and 5\n");
return 0;
}
if (d->softkeyicon[p] != 0) {
ast_log(LOG_WARNING, "Invalid position %d for bookmark : already used\n:", p);
return 0;
}
memmove(line, line + 2, sizeof(line));
} else {
/* No position specified, looking for a free slot */
for (p = 0; p <= 5; p++) {
if (!d->softkeyicon[p])
break;
}
if (p > 5) {
ast_log(LOG_WARNING, "No more free bookmark position\n");
return 0;
}
}
at = strchr(line, '@');
if (!at) {
ast_log(LOG_NOTICE, "Bookmark entry '%s' has no @ (at) sign!\n", text);
return 0;
}
*at = '\0';
at++;
number = at;
at = strchr(at, '@');
if (ast_strlen_zero(number)) {
ast_log(LOG_NOTICE, "Bookmark entry '%s' has no number\n", text);
return 0;
}
if (ast_strlen_zero(line)) {
ast_log(LOG_NOTICE, "Bookmark entry '%s' has no description\n", text);
return 0;
}
at = strchr(number, '@');
if (!at)
d->softkeyicon[p] = FAV_ICON_SHARP; /* default icon */
else {
*at = '\0';
at++;
icon = at;
if (ast_strlen_zero(icon)) {
ast_log(LOG_NOTICE, "Bookmark entry '%s' has no icon value\n", text);
return 0;
}
if (strncmp(icon, "USTM/", 5))
d->softkeyicon[p] = atoi(icon);
else {
d->softkeyicon[p] = 1;
ast_copy_string(d->softkeydevice[p], icon + 5, sizeof(d->softkeydevice[p]));
}
}
ast_copy_string(d->softkeylabel[p], line, sizeof(d->softkeylabel[p]));
ast_copy_string(d->softkeynumber[p], number, sizeof(d->softkeynumber[p]));
if (unistimdebug)
ast_verb(0, "New bookmark at pos %d label='%s' number='%s' icon=%x\n",
p, d->softkeylabel[p], d->softkeynumber[p], d->softkeyicon[p]);
return 1;
}
/* Looking for dynamic icons entries in bookmarks */
static void finish_bookmark(void)
{
struct unistim_device *d = devices;
int i;
while (d) {
for (i = 0; i < 6; i++) {
if (d->softkeyicon[i] == 1) { /* Something for us */
struct unistim_device *d2 = devices;
while (d2) {
if (!strcmp(d->softkeydevice[i], d2->name)) {
d->sp[i] = d2;
d->softkeyicon[i] = 0;
break;
}
d2 = d2->next;
}
if (d->sp[i] == NULL)
ast_log(LOG_NOTICE, "Bookmark entry with device %s not found\n",
d->softkeydevice[i]);
}
}
d = d->next;
}
}
static struct unistim_device *build_device(const char *cat, const struct ast_variable *v)
{
struct unistim_device *d;
struct unistim_line *l = NULL;
int create = 1;
int nbsoftkey, dateformat, timeformat, callhistory;
char linelabel[AST_MAX_EXTENSION];
char context[AST_MAX_EXTENSION];
char ringvolume, ringstyle;
/* First, we need to know if we already have this name in our list */
/* Get a lock for the device chained list */
ast_mutex_lock(&devicelock);
d = devices;
while (d) {
if (!strcmp(d->name, cat)) {
/* Yep, we alreay have this one */
if (unistimsock < 0) {
/* It's a dupe */
ast_log(LOG_WARNING, "Duplicate entry found (%s), ignoring.\n", cat);
ast_mutex_unlock(&devicelock);
return NULL;
}
/* we're reloading right now */
create = 0;
l = d->lines;
break;
}
d = d->next;
}
ast_mutex_unlock(&devicelock);
if (create) {
if (!(d = ast_calloc(1, sizeof(*d))))
return NULL;
if (!(l = unistim_line_alloc())) {
ast_free(d);
return NULL;
}
ast_copy_string(d->name, cat, sizeof(d->name));
}
ast_copy_string(context, DEFAULTCONTEXT, sizeof(context));
d->contrast = -1;
d->output = OUTPUT_HANDSET;
d->previous_output = OUTPUT_HANDSET;
d->volume = VOLUME_LOW;
d->mute = MUTE_OFF;
d->height = DEFAULTHEIGHT;
linelabel[0] = '\0';
dateformat = 1;
timeformat = 1;
ringvolume = 2;
callhistory = 1;
ringstyle = 3;
nbsoftkey = 0;
while (v) {
if (!strcasecmp(v->name, "rtp_port"))
d->rtp_port = atoi(v->value);
else if (!strcasecmp(v->name, "rtp_method"))
d->rtp_method = atoi(v->value);
else if (!strcasecmp(v->name, "status_method"))
d->status_method = atoi(v->value);
else if (!strcasecmp(v->name, "device"))
ast_copy_string(d->id, v->value, sizeof(d->id));
else if (!strcasecmp(v->name, "tn"))
ast_copy_string(d->extension_number, v->value, sizeof(d->extension_number));
else if (!strcasecmp(v->name, "permit") || !strcasecmp(v->name, "deny"))
d->ha = ast_append_ha(v->name, v->value, d->ha, NULL);
else if (!strcasecmp(v->name, "context"))
ast_copy_string(context, v->value, sizeof(context));
else if (!strcasecmp(v->name, "maintext0"))
unquote(d->maintext0, v->value, sizeof(d->maintext0) - 1);
else if (!strcasecmp(v->name, "maintext1"))
unquote(d->maintext1, v->value, sizeof(d->maintext1) - 1);
else if (!strcasecmp(v->name, "maintext2"))
unquote(d->maintext2, v->value, sizeof(d->maintext2) - 1);
else if (!strcasecmp(v->name, "titledefault"))
unquote(d->titledefault, v->value, sizeof(d->titledefault) - 1);
else if (!strcasecmp(v->name, "dateformat"))
dateformat = atoi(v->value);
else if (!strcasecmp(v->name, "timeformat"))
timeformat = atoi(v->value);
else if (!strcasecmp(v->name, "contrast")) {
d->contrast = atoi(v->value);
if ((d->contrast < 0) || (d->contrast > 15)) {
ast_log(LOG_WARNING, "constrast must be beetween 0 and 15");
d->contrast = 8;
}
} else if (!strcasecmp(v->name, "nat"))
d->nat = ast_true(v->value);
else if (!strcasecmp(v->name, "ringvolume"))
ringvolume = atoi(v->value);
else if (!strcasecmp(v->name, "ringstyle"))
ringstyle = atoi(v->value);
else if (!strcasecmp(v->name, "callhistory"))
callhistory = atoi(v->value);
else if (!strcasecmp(v->name, "callerid")) {
if (!strcasecmp(v->value, "asreceived"))
l->cid_num[0] = '\0';
else
ast_copy_string(l->cid_num, v->value, sizeof(l->cid_num));
} else if (!strcasecmp(v->name, "language"))
ast_copy_string(l->language, v->value, sizeof(l->language));
else if (!strcasecmp(v->name, "country"))
ast_copy_string(d->country, v->value, sizeof(d->country));
else if (!strcasecmp(v->name, "accountcode"))
ast_copy_string(l->accountcode, v->value, sizeof(l->accountcode));
else if (!strcasecmp(v->name, "amaflags")) {
int y;
y = ast_cdr_amaflags2int(v->value);
if (y < 0)
ast_log(LOG_WARNING, "Invalid AMA flags: %s at line %d\n", v->value,
v->lineno);
else
l->amaflags = y;
} else if (!strcasecmp(v->name, "musiconhold"))
ast_copy_string(l->musicclass, v->value, sizeof(l->musicclass));
else if (!strcasecmp(v->name, "callgroup"))
l->callgroup = ast_get_group(v->value);
else if (!strcasecmp(v->name, "pickupgroup"))
l->pickupgroup = ast_get_group(v->value);
else if (!strcasecmp(v->name, "mailbox"))
ast_copy_string(l->mailbox, v->value, sizeof(l->mailbox));
else if (!strcasecmp(v->name, "parkinglot"))
ast_copy_string(l->parkinglot, v->value, sizeof(l->parkinglot));
else if (!strcasecmp(v->name, "linelabel"))
unquote(linelabel, v->value, sizeof(linelabel) - 1);
else if (!strcasecmp(v->name, "extension")) {
if (!strcasecmp(v->value, "none"))
d->extension = EXTENSION_NONE;
else if (!strcasecmp(v->value, "ask"))
d->extension = EXTENSION_ASK;
else if (!strcasecmp(v->value, "line"))
d->extension = EXTENSION_LINE;
else
ast_log(LOG_WARNING, "Unknown extension option.\n");
} else if (!strcasecmp(v->name, "bookmark")) {
if (nbsoftkey > 5)
ast_log(LOG_WARNING,
"More than 6 softkeys defined. Ignoring new entries.\n");
else {
if (ParseBookmark(v->value, d))
nbsoftkey++;
}
} else if (!strcasecmp(v->name, "line")) {
int len = strlen(linelabel);
if (nbsoftkey) {
ast_log(LOG_WARNING,
"You must use bookmark AFTER line=>. Only one line is supported in this version\n");
if (create) {
ast_free(d);
unistim_line_destroy(l);
}
return NULL;
}
if (create) {
ast_mutex_init(&l->lock);
} else {
d->to_delete = 0;
/* reset bookmarks */
memset(d->softkeylabel, 0, sizeof(d->softkeylabel));
memset(d->softkeynumber, 0, sizeof(d->softkeynumber));
memset(d->softkeyicon, 0, sizeof(d->softkeyicon));
memset(d->softkeydevice, 0, sizeof(d->softkeydevice));
memset(d->sp, 0, sizeof(d->sp));
}
ast_copy_string(l->name, v->value, sizeof(l->name));
snprintf(l->fullname, sizeof(l->fullname), "USTM/%s@%s", l->name, d->name);
d->softkeyicon[0] = FAV_ICON_ONHOOK_BLACK;
if (!len) /* label is undefined ? */
ast_copy_string(d->softkeylabel[0], v->value, sizeof(d->softkeylabel[0]));
else {
if ((len > 2) && (linelabel[1] == '@')) {
d->softkeylinepos = linelabel[0];
if ((d->softkeylinepos >= '0') && (d->softkeylinepos <= '5')) {
d->softkeylinepos -= '0';
d->softkeyicon[0] = 0;
} else {
ast_log(LOG_WARNING,
"Invalid position for linelabel : must be between 0 and 5\n");
d->softkeylinepos = 0;
}
ast_copy_string(d->softkeylabel[d->softkeylinepos], linelabel + 2,
sizeof(d->softkeylabel[d->softkeylinepos]));
d->softkeyicon[d->softkeylinepos] = FAV_ICON_ONHOOK_BLACK;
} else
ast_copy_string(d->softkeylabel[0], linelabel,
sizeof(d->softkeylabel[0]));
}
nbsoftkey++;
ast_copy_string(l->context, context, sizeof(l->context));
if (!ast_strlen_zero(l->mailbox)) {
if (unistimdebug)
ast_verb(3, "Setting mailbox '%s' on %s@%s\n", l->mailbox, d->name, l->name);
}
ast_format_cap_copy(l->cap, global_cap);
l->parent = d;
if (create) {
if (!alloc_sub(l, SUB_REAL)) {
ast_mutex_destroy(&l->lock);
unistim_line_destroy(l);
ast_free(d);
return NULL;
}
l->next = d->lines;
d->lines = l;
}
} else if (!strcasecmp(v->name, "height")) {
/* Allow the user to lower the expected display lines on the phone
* For example the Nortal I2001 and I2002 only have one ! */
d->height = atoi(v->value);
} else
ast_log(LOG_WARNING, "Don't know keyword '%s' at line %d\n", v->name,
v->lineno);
v = v->next;
}
d->ringvolume = ringvolume;
d->ringstyle = ringstyle;
d->callhistory = callhistory;
d->tz = ast_get_indication_zone(d->country);
if ((d->tz == NULL) && !ast_strlen_zero(d->country))
ast_log(LOG_WARNING, "Country '%s' was not found in indications.conf\n",
d->country);
d->datetimeformat = 56 + (dateformat * 4);
d->datetimeformat += timeformat;
if (!d->lines) {
ast_log(LOG_ERROR, "An Unistim device must have at least one line!\n");
ast_mutex_destroy(&l->lock);
unistim_line_destroy(l);
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
if (d->tz) {
d->tz = ast_tone_zone_unref(d->tz);
}
ast_free(d);
return NULL;
}
if ((autoprovisioning == AUTOPROVISIONING_TN) &&
(!ast_strlen_zero(d->extension_number))) {
d->extension = EXTENSION_TN;
if (!ast_strlen_zero(d->id))
ast_log(LOG_WARNING,
"tn= and device= can't be used together. Ignoring device= entry\n");
d->id[0] = 'T'; /* magic : this is a tn entry */
ast_copy_string((d->id) + 1, d->extension_number, sizeof(d->id) - 1);
d->extension_number[0] = '\0';
} else if (ast_strlen_zero(d->id)) {
if (strcmp(d->name, "template")) {
ast_log(LOG_ERROR, "You must specify the mac address with device=\n");
ast_mutex_destroy(&l->lock);
unistim_line_destroy(l);
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
if (d->tz) {
d->tz = ast_tone_zone_unref(d->tz);
}
ast_free(d);
return NULL;
} else
strcpy(d->id, "000000000000");
}
if (!d->rtp_port)
d->rtp_port = 10000;
if (d->contrast == -1)
d->contrast = 8;
if (ast_strlen_zero(d->maintext0))
strcpy(d->maintext0, "Welcome");
if (ast_strlen_zero(d->maintext1))
strcpy(d->maintext1, d->name);
if (ast_strlen_zero(d->titledefault)) {
struct ast_tm tm = { 0, };
struct timeval cur_time = ast_tvnow();
if ((ast_localtime(&cur_time, &tm, 0)) == 0 || ast_strlen_zero(tm.tm_zone)) {
display_last_error("Error in ast_localtime()");
ast_copy_string(d->titledefault, "UNISTIM for*", 12);
} else {
if (strlen(tm.tm_zone) < 4) {
strcpy(d->titledefault, "TimeZone ");
strcat(d->titledefault, tm.tm_zone);
} else if (strlen(tm.tm_zone) < 9) {
strcpy(d->titledefault, "TZ ");
strcat(d->titledefault, tm.tm_zone);
} else
ast_copy_string(d->titledefault, tm.tm_zone, 12);
}
}
/* Update the chained link if it's a new device */
if (create) {
ast_mutex_lock(&devicelock);
d->next = devices;
devices = d;
ast_mutex_unlock(&devicelock);
ast_verb(3, "Added device '%s'\n", d->name);
} else {
ast_verb(3, "Device '%s' reloaded\n", d->name);
}
return d;
}
/*--- reload_config: Re-read unistim.conf config file ---*/
static int reload_config(void)
{
struct ast_config *cfg;
struct ast_variable *v;
struct ast_hostent ahp;
struct hostent *hp;
struct sockaddr_in bindaddr = { 0, };
char *config = "unistim.conf";
char *cat;
struct unistim_device *d;
const int reuseFlag = 1;
struct unistimsession *s;
struct ast_flags config_flags = { 0, };
cfg = ast_config_load(config, config_flags);
/* We *must* have a config file otherwise stop immediately */
if (!cfg) {
ast_log(LOG_ERROR, "Unable to load config %s\n", config);
return -1;
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file %s is in an invalid format. Aborting.\n", config);
return -1;
}
/* Copy the default jb config over global_jbconf */
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
unistim_keepalive = 120;
unistim_port = 0;
v = ast_variable_browse(cfg, "general");
while (v) {
/* handle jb conf */
if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
continue;
if (!strcasecmp(v->name, "keepalive"))
unistim_keepalive = atoi(v->value);
else if (!strcasecmp(v->name, "port"))
unistim_port = atoi(v->value);
else if (!strcasecmp(v->name, "tos")) {
if (ast_str2tos(v->value, &qos.tos))
ast_log(LOG_WARNING, "Invalid tos value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "tos_audio")) {
if (ast_str2tos(v->value, &qos.tos_audio))
ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos")) {
if (ast_str2cos(v->value, &qos.cos))
ast_log(LOG_WARNING, "Invalid cos value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos_audio")) {
if (ast_str2cos(v->value, &qos.cos_audio))
ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "autoprovisioning")) {
if (!strcasecmp(v->value, "no"))
autoprovisioning = AUTOPROVISIONING_NO;
else if (!strcasecmp(v->value, "yes"))
autoprovisioning = AUTOPROVISIONING_YES;
else if (!strcasecmp(v->value, "db"))
autoprovisioning = AUTOPROVISIONING_DB;
else if (!strcasecmp(v->value, "tn"))
autoprovisioning = AUTOPROVISIONING_TN;
else
ast_log(LOG_WARNING, "Unknown autoprovisioning option.\n");
} else if (!strcasecmp(v->name, "public_ip")) {
if (!ast_strlen_zero(v->value)) {
if (!(hp = ast_gethostbyname(v->value, &ahp)))
ast_log(LOG_WARNING, "Invalid address: %s\n", v->value);
else {
memcpy(&public_ip.sin_addr, hp->h_addr, sizeof(public_ip.sin_addr));
public_ip.sin_family = AF_INET;
}
}
}
v = v->next;
}
if ((unistim_keepalive < 10) ||
(unistim_keepalive >
255 - (((NB_MAX_RETRANSMIT + 1) * RETRANSMIT_TIMER) / 1000))) {
ast_log(LOG_ERROR, "keepalive is invalid in %s\n", config);
ast_config_destroy(cfg);
return -1;
}
packet_send_ping[4] =
unistim_keepalive + (((NB_MAX_RETRANSMIT + 1) * RETRANSMIT_TIMER) / 1000);
if ((unistim_port < 1) || (unistim_port > 65535)) {
ast_log(LOG_ERROR, "port is not set or invalid in %s\n", config);
ast_config_destroy(cfg);
return -1;
}
unistim_keepalive *= 1000;
ast_mutex_lock(&devicelock);
d = devices;
while (d) {
if (d->to_delete >= 0)
d->to_delete = 1;
d = d->next;
}
ast_mutex_unlock(&devicelock);
/* load the device sections */
cat = ast_category_browse(cfg, NULL);
while (cat) {
if (strcasecmp(cat, "general")) {
d = build_device(cat, ast_variable_browse(cfg, cat));
}
cat = ast_category_browse(cfg, cat);
}
ast_mutex_lock(&devicelock);
d = devices;
while (d) {
if (d->to_delete) {
int i;
if (unistimdebug)
ast_verb(0, "Removing device '%s'\n", d->name);
if (!d->lines) {
ast_log(LOG_ERROR, "Device '%s' without a line !, aborting\n", d->name);
ast_config_destroy(cfg);
return 0;
}
if (!d->lines->subs[0]) {
ast_log(LOG_ERROR, "Device '%s' without a subchannel !, aborting\n",
d->name);
ast_config_destroy(cfg);
return 0;
}
if (d->lines->subs[0]->owner) {
ast_log(LOG_WARNING,
"Device '%s' was not deleted : a call is in progress. Try again later.\n",
d->name);
d = d->next;
continue;
}
ast_mutex_destroy(&d->lines->subs[0]->lock);
ast_free(d->lines->subs[0]);
for (i = 1; i < MAX_SUBS; i++) {
if (d->lines->subs[i]) {
ast_log(LOG_WARNING,
"Device '%s' with threeway call subchannels allocated, aborting.\n",
d->name);
break;
}
}
if (i < MAX_SUBS) {
d = d->next;
continue;
}
ast_mutex_destroy(&d->lines->lock);
ast_free(d->lines);
if (d->session) {
if (sessions == d->session)
sessions = d->session->next;
else {
s = sessions;
while (s) {
if (s->next == d->session) {
s->next = d->session->next;
break;
}
s = s->next;
}
}
ast_mutex_destroy(&d->session->lock);
ast_free(d->session);
}
if (devices == d)
devices = d->next;
else {
struct unistim_device *d2 = devices;
while (d2) {
if (d2->next == d) {
d2->next = d->next;
break;
}
d2 = d2->next;
}
}
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
if (d->tz) {
d->tz = ast_tone_zone_unref(d->tz);
}
ast_free(d);
d = devices;
continue;
}
d = d->next;
}
finish_bookmark();
ast_mutex_unlock(&devicelock);
ast_config_destroy(cfg);
ast_mutex_lock(&sessionlock);
s = sessions;
while (s) {
if (s->device)
refresh_all_favorite(s);
s = s->next;
}
ast_mutex_unlock(&sessionlock);
/* We don't recreate a socket when reloading (locks would be necessary). */
if (unistimsock > -1)
return 0;
bindaddr.sin_addr.s_addr = INADDR_ANY;
bindaddr.sin_port = htons(unistim_port);
bindaddr.sin_family = AF_INET;
unistimsock = socket(AF_INET, SOCK_DGRAM, 0);
if (unistimsock < 0) {
ast_log(LOG_WARNING, "Unable to create UNISTIM socket: %s\n", strerror(errno));
return -1;
}
#ifdef HAVE_PKTINFO
{
const int pktinfoFlag = 1;
setsockopt(unistimsock, IPPROTO_IP, IP_PKTINFO, &pktinfoFlag,
sizeof(pktinfoFlag));
}
#else
if (public_ip.sin_family == 0) {
ast_log(LOG_WARNING,
"Your OS does not support IP_PKTINFO, you must set public_ip.\n");
unistimsock = -1;
return -1;
}
#endif
setsockopt(unistimsock, SOL_SOCKET, SO_REUSEADDR, (const char *) &reuseFlag,
sizeof(reuseFlag));
if (bind(unistimsock, (struct sockaddr *) &bindaddr, sizeof(bindaddr)) < 0) {
ast_log(LOG_WARNING, "Failed to bind to %s:%d: %s\n",
ast_inet_ntoa(bindaddr.sin_addr), htons(bindaddr.sin_port),
strerror(errno));
close(unistimsock);
unistimsock = -1;
} else {
ast_verb(2, "UNISTIM Listening on %s:%d\n", ast_inet_ntoa(bindaddr.sin_addr), htons(bindaddr.sin_port));
ast_netsock_set_qos(unistimsock, qos.tos, qos.cos, "UNISTIM");
}
return 0;
}
static enum ast_rtp_glue_result unistim_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct unistim_subchannel *sub = chan->tech_pvt;
ao2_ref(sub->rtp, +1);
*instance = sub->rtp;
return AST_RTP_GLUE_RESULT_LOCAL;
}
static struct ast_rtp_glue unistim_rtp_glue = {
.type = channel_type,
.get_rtp_info = unistim_get_rtp_peer,
};
/*--- load_module: PBX load module - initialization ---*/
int load_module(void)
{
int res;
struct ast_format tmpfmt;
if (!(global_cap = ast_format_cap_alloc())) {
goto buff_failed;
}
if (!(unistim_tech.capabilities = ast_format_cap_alloc())) {
goto buff_failed;
}
ast_format_cap_add(global_cap, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0));
ast_format_cap_add(global_cap, ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0));
ast_format_cap_copy(unistim_tech.capabilities, global_cap);
if (!(buff = ast_malloc(SIZE_PAGE)))
goto buff_failed;
io = io_context_create();
if (!io) {
ast_log(LOG_ERROR, "Failed to allocate IO context\n");
goto io_failed;
}
sched = ast_sched_context_create();
if (!sched) {
ast_log(LOG_ERROR, "Failed to allocate scheduler context\n");
goto sched_failed;
}
res = reload_config();
if (res)
return AST_MODULE_LOAD_DECLINE;
/* Make sure we can register our unistim channel type */
if (ast_channel_register(&unistim_tech)) {
ast_log(LOG_ERROR, "Unable to register channel type '%s'\n", channel_type);
goto chanreg_failed;
}
ast_rtp_glue_register(&unistim_rtp_glue);
ast_cli_register_multiple(unistim_cli, ARRAY_LEN(unistim_cli));
restart_monitor();
return AST_MODULE_LOAD_SUCCESS;
chanreg_failed:
/*! XXX \todo Leaking anything allocated by reload_config() ... */
ast_sched_context_destroy(sched);
sched = NULL;
sched_failed:
io_context_destroy(io);
io = NULL;
io_failed:
ast_free(buff);
buff = NULL;
global_cap = ast_format_cap_destroy(global_cap);
unistim_tech.capabilities = ast_format_cap_destroy(unistim_tech.capabilities);
buff_failed:
return AST_MODULE_LOAD_FAILURE;
}
static int unload_module(void)
{
/* First, take us out of the channel loop */
if (sched) {
ast_sched_context_destroy(sched);
}
ast_cli_unregister_multiple(unistim_cli, ARRAY_LEN(unistim_cli));
ast_channel_unregister(&unistim_tech);
ast_rtp_glue_unregister(&unistim_rtp_glue);
ast_mutex_lock(&monlock);
if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) {
pthread_cancel(monitor_thread);
pthread_kill(monitor_thread, SIGURG);
pthread_join(monitor_thread, NULL);
}
monitor_thread = AST_PTHREADT_STOP;
ast_mutex_unlock(&monlock);
if (buff)
ast_free(buff);
if (unistimsock > -1)
close(unistimsock);
global_cap = ast_format_cap_destroy(global_cap);
unistim_tech.capabilities = ast_format_cap_destroy(unistim_tech.capabilities);
return 0;
}
/*! reload: Part of Asterisk module interface ---*/
int reload(void)
{
unistim_reload(NULL, 0, NULL);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "UNISTIM Protocol (USTM)",
.load = load_module,
.unload = unload_module,
.reload = reload,
);