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asterisk/channels/misdn/isdn_lib.h

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/*
* Chan_Misdn -- Channel Driver for Asterisk
*
* Interface to mISDN
*
* Copyright (C) 2004, Christian Richter
*
* Christian Richter <crich@beronet.com>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*/
/*! \file
* \brief Interface to mISDN
*
* \author Christian Richter <crich@beronet.com>
*/
#ifndef TE_LIB
#define TE_LIB
#include <mISDNuser/suppserv.h>
/** For initialization usage **/
/* typedef int ie_nothing_t ;*/
/** end of init usage **/
/*
* uncomment the following to make chan_misdn create
* record files in /tmp/misdn-{rx|tx}-PortChannel format
* */
/*#define MISDN_SAVE_DATA*/
#ifdef WITH_BEROEC
typedef int beroec_t;
enum beroec_type {
BEROEC_FULLBAND=0,
BEROEC_SUBBAND,
BEROEC_FASTSUBBAND
};
void beroec_init(void);
void beroec_exit(void);
beroec_t *beroec_new(int tail, enum beroec_type type, int anti_howl,
int tonedisable, int zerocoeff, int adapt, int nlp);
void beroec_destroy(beroec_t *ec);
int beroec_cancel_alaw_chunk(beroec_t *ec,
char *send,
char *receive,
int len);
int beroec_version(void);
#endif
enum tone_e {
TONE_NONE=0,
TONE_DIAL,
TONE_ALERTING,
TONE_FAR_ALERTING,
TONE_BUSY,
TONE_HANGUP,
TONE_CUSTOM,
TONE_FILE
};
#define MAX_BCHANS 31
enum bchannel_state {
BCHAN_CLEANED=0,
BCHAN_EMPTY,
BCHAN_SETUP,
BCHAN_SETUPED,
BCHAN_ACTIVE,
BCHAN_ACTIVATED,
BCHAN_BRIDGE,
BCHAN_BRIDGED,
BCHAN_RELEASE,
BCHAN_RELEASED,
BCHAN_CLEAN,
BCHAN_CLEAN_REQUEST,
BCHAN_ERROR
};
enum misdn_err_e {
ENOCHAN=1
};
enum mISDN_NUMBER_PLAN {
NUMPLAN_UNKNOWN = 0x0,
NUMPLAN_ISDN = 0x1, /* ISDN/Telephony numbering plan E.164 */
NUMPLAN_DATA = 0x3, /* Data numbering plan X.121 */
NUMPLAN_TELEX = 0x4, /* Telex numbering plan F.69 */
NUMPLAN_NATIONAL = 0x8,
NUMPLAN_PRIVATE = 0x9
};
enum mISDN_NUMBER_TYPE {
NUMTYPE_UNKNOWN = 0x0,
NUMTYPE_INTERNATIONAL = 0x1,
NUMTYPE_NATIONAL = 0x2,
NUMTYPE_NETWORK_SPECIFIC = 0x3,
NUMTYPE_SUBSCRIBER = 0x4,
NUMTYPE_ABBREVIATED = 0x5
};
enum event_response_e {
RESPONSE_IGNORE_SETUP_WITHOUT_CLOSE,
RESPONSE_IGNORE_SETUP,
RESPONSE_RELEASE_SETUP,
RESPONSE_ERR,
RESPONSE_OK
};
enum event_e {
EVENT_NOTHING,
EVENT_TONE_GENERATE,
EVENT_BCHAN_DATA,
EVENT_BCHAN_ACTIVATED,
EVENT_BCHAN_ERROR,
EVENT_CLEANUP,
EVENT_PROCEEDING,
EVENT_PROGRESS,
EVENT_SETUP,
EVENT_REGISTER,
EVENT_ALERTING,
EVENT_CONNECT,
EVENT_SETUP_ACKNOWLEDGE,
EVENT_CONNECT_ACKNOWLEDGE ,
EVENT_USER_INFORMATION,
EVENT_SUSPEND_REJECT,
EVENT_RESUME_REJECT,
EVENT_HOLD,
EVENT_SUSPEND,
EVENT_RESUME,
EVENT_HOLD_ACKNOWLEDGE,
EVENT_SUSPEND_ACKNOWLEDGE,
EVENT_RESUME_ACKNOWLEDGE,
EVENT_HOLD_REJECT,
EVENT_RETRIEVE,
EVENT_RETRIEVE_ACKNOWLEDGE,
EVENT_RETRIEVE_REJECT,
EVENT_DISCONNECT,
EVENT_RESTART,
EVENT_RELEASE,
EVENT_RELEASE_COMPLETE,
EVENT_FACILITY,
EVENT_NOTIFY,
EVENT_STATUS_ENQUIRY,
EVENT_INFORMATION,
EVENT_STATUS,
EVENT_TIMEOUT,
EVENT_DTMF_TONE,
EVENT_NEW_L3ID,
EVENT_NEW_BC,
EVENT_PORT_ALARM,
EVENT_NEW_CHANNEL,
EVENT_UNKNOWN
};
enum ie_name_e {
IE_DUMMY,
IE_LAST
};
enum { /* bearer capability */
INFO_CAPABILITY_SPEECH=0,
INFO_CAPABILITY_AUDIO_3_1K=0x10 ,
INFO_CAPABILITY_AUDIO_7K=0x11 ,
INFO_CAPABILITY_VIDEO =0x18,
INFO_CAPABILITY_DIGITAL_UNRESTRICTED =0x8,
INFO_CAPABILITY_DIGITAL_RESTRICTED =0x09,
INFO_CAPABILITY_DIGITAL_UNRESTRICTED_TONES
};
enum { /* progress indicators */
INFO_PI_CALL_NOT_E2E_ISDN =0x01,
INFO_PI_CALLED_NOT_ISDN =0x02,
INFO_PI_CALLER_NOT_ISDN =0x03,
INFO_PI_CALLER_RETURNED_TO_ISDN =0x04,
INFO_PI_INBAND_AVAILABLE =0x08,
INFO_PI_DELAY_AT_INTERF =0x0a,
INFO_PI_INTERWORKING_WITH_PUBLIC =0x10,
INFO_PI_INTERWORKING_NO_RELEASE =0x11,
INFO_PI_INTERWORKING_NO_RELEASE_PRE_ANSWER =0x12,
INFO_PI_INTERWORKING_NO_RELEASE_POST_ANSWER =0x13
};
/*!
* \brief Q.931 encoded redirecting reason
*/
enum mISDN_REDIRECTING_REASON {
mISDN_REDIRECTING_REASON_UNKNOWN = 0x0,
/*! Call forwarding busy or called DTE busy */
mISDN_REDIRECTING_REASON_CALL_FWD_BUSY = 0x1,
/*! Call forwarding no reply */
mISDN_REDIRECTING_REASON_NO_REPLY = 0x2,
/*! Call deflection */
mISDN_REDIRECTING_REASON_DEFLECTION = 0x4,
/*! Called DTE out of order */
mISDN_REDIRECTING_REASON_OUT_OF_ORDER = 0x9,
/*! Call forwarding by the called DTE */
mISDN_REDIRECTING_REASON_CALL_FWD_DTE = 0xA,
/*! Call forwarding unconditional or systematic call redirection */
mISDN_REDIRECTING_REASON_CALL_FWD = 0xF
};
/*!
* \brief Notification description code enumeration
*/
enum mISDN_NOTIFY_CODE {
mISDN_NOTIFY_CODE_INVALID = -1,
/*! Call is placed on hold (Q.931) */
mISDN_NOTIFY_CODE_USER_SUSPEND = 0x00,
/*! Call is taken off of hold (Q.931) */
mISDN_NOTIFY_CODE_USER_RESUME = 0x01,
/*! Call is diverting (EN 300 207-1 Section 7.2.1) */
mISDN_NOTIFY_CODE_CALL_IS_DIVERTING = 0x7B,
/*! Call diversion is enabled (cfu, cfb, cfnr) (EN 300 207-1 Section 7.2.1) */
mISDN_NOTIFY_CODE_DIVERSION_ACTIVATED = 0x68,
/*! Call transfer, alerting (EN 300 369-1 Section 7.2) */
mISDN_NOTIFY_CODE_CALL_TRANSFER_ALERTING = 0x69,
/*! Call transfer, active(answered) (EN 300 369-1 Section 7.2) */
mISDN_NOTIFY_CODE_CALL_TRANSFER_ACTIVE = 0x6A,
};
enum { /*CODECS*/
INFO_CODEC_ULAW=2,
INFO_CODEC_ALAW=3
};
enum layer_e {
L3,
L2,
L1,
UNKNOWN
};
/*! Maximum phone number (address) length plus null terminator */
#define MISDN_MAX_NUMBER_LEN (31 + 1)
/*! Maximum name length plus null terminator (From ECMA-164) */
#define MISDN_MAX_NAME_LEN (50 + 1)
/*! Maximum subaddress length plus null terminator */
#define MISDN_MAX_SUBADDRESS_LEN (23 + 1)
/*! Maximum keypad facility content length plus null terminator */
#define MISDN_MAX_KEYPAD_LEN (31 + 1)
/*! \brief Dialed/Called information struct */
struct misdn_party_dialing {
/*! \brief Type-of-number in ISDN terms for the dialed/called number */
enum mISDN_NUMBER_TYPE number_type;
/*! \brief Type-of-number numbering plan. */
enum mISDN_NUMBER_PLAN number_plan;
/*! \brief Dialed/Called Phone Number (Address) */
char number[MISDN_MAX_NUMBER_LEN];
/*! \brief Dialed/Called Subaddress number */
char subaddress[MISDN_MAX_SUBADDRESS_LEN];
};
/*! \brief Connected-Line/Calling/Redirecting ID info struct */
struct misdn_party_id {
/*! \brief Number presentation restriction code
* 0=Allowed, 1=Restricted, 2=Unavailable
*/
int presentation;
/*! \brief Number screening code
* 0=Unscreened, 1=Passed Screen, 2=Failed Screen, 3=Network Number
*/
int screening;
/*! \brief Type-of-number in ISDN terms for the number */
enum mISDN_NUMBER_TYPE number_type;
/*! \brief Type-of-number numbering plan. */
enum mISDN_NUMBER_PLAN number_plan;
/*! \brief Subscriber Name
* \note The name is currently obtained from Asterisk for
* potential use in display ie's since basic ISDN does
* not support names directly.
*/
char name[MISDN_MAX_NAME_LEN];
/*! \brief Phone number (Address) */
char number[MISDN_MAX_NUMBER_LEN];
/*! \brief Subaddress number */
char subaddress[MISDN_MAX_SUBADDRESS_LEN];
};
/*! \brief Redirecting information struct */
struct misdn_party_redirecting {
/*! \brief Who is redirecting the call (Sent to the party the call is redirected toward) */
struct misdn_party_id from;
/*! \brief Where the call is being redirected toward (Sent to the calling party) */
struct misdn_party_id to;
/*! \brief Reason a call is being redirected (Q.931 field value) */
enum mISDN_REDIRECTING_REASON reason;
/*! \brief Number of times the call has been redirected */
int count;
/*! \brief TRUE if the redirecting.to information has changed */
int to_changed;
};
/*! \brief B channel control structure */
struct misdn_bchannel {
/*! \brief B channel send locking structure */
struct send_lock *send_lock;
#if defined(AST_MISDN_ENHANCEMENTS)
/*! \brief The BC, HLC (optional) and LLC (optional) contents from the SETUP message. */
struct Q931_Bc_Hlc_Llc setup_bc_hlc_llc;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
/*!
* \brief Dialed/Called information struct
* \note The number_type element is set to "dialplan" in /etc/asterisk/misdn.conf for outgoing calls
*/
struct misdn_party_dialing dialed;
/*! \brief Originating/Caller ID information struct
* \note The number_type element can be set to "localdialplan" in /etc/asterisk/misdn.conf for outgoing calls
* \note The number element can be set to "callerid" in /etc/asterisk/misdn.conf for outgoing calls
*/
struct misdn_party_id caller;
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17 15:36:31 +00:00
/*! \brief Incoming Caller ID string tag for special purpose
* \note The element can be set to "incoming_cid_tag" in /etc/asterisk/misdn.conf for incoming calls
*/
char incoming_cid_tag[MISDN_MAX_NAME_LEN];
/*! \brief Connected-Party/Connected-Line ID information struct
* \note The number_type element can be set to "cpndialplan" in /etc/asterisk/misdn.conf for outgoing calls
*/
struct misdn_party_id connected;
/*! \brief Redirecting information struct (Where a call diversion or transfer was invoked)
* \note The redirecting subaddress is not defined in Q.931 so it is not used.
*/
struct misdn_party_redirecting redirecting;
/*! \brief TRUE if this is a dummy BC record */
int dummy;
/*! \brief TRUE if NT side of protocol (TE otherwise) */
int nt;
/*! \brief TRUE if ISDN-PRI (ISDN-BRI otherwise) */
int pri;
/*! \brief Logical Layer 1 port associated with this B channel */
int port;
/** init stuff **/
/*! \brief B Channel mISDN driver stack ID */
int b_stid;
/* int b_addr; */
/*! \brief B Channel mISDN driver layer ID from mISDN_new_layer() */
int layer_id;
/*! \brief B channel layer; set to 3 or 4 */
int layer;
/* state stuff */
/*! \brief TRUE if DISCONNECT needs to be sent to clear a call */
int need_disconnect;
/*! \brief TRUE if RELEASE needs to be sent to clear a call */
int need_release;
/*! \brief TRUE if RELEASE_COMPLETE needs to be sent to clear a call */
int need_release_complete;
/*! \brief TRUE if allocate higher B channels first */
int dec;
/* var stuff */
/*! \brief Layer 3 process ID */
int l3_id;
/*! \brief B channel process ID (1-5000) */
int pid;
/*! \brief Not used. Saved mISDN stack CONNECT_t ces value */
int ces;
/*! \brief B channel to restart if received a RESTART message */
int restart_channel;
/*! \brief Assigned B channel number B1, B2... 0 if not assigned */
int channel;
/*! \brief TRUE if the B channel number is preselected */
int channel_preselected;
/*! \brief TRUE if the B channel is allocated from the REGISTER pool */
int is_register_pool;
/*! \brief TRUE if B channel record is in use */
int in_use;
/*! \brief Time when empty_bc() last called on this record */
struct timeval last_used;
/*! \brief TRUE if call waiting */
int cw;
/*! \brief B Channel mISDN driver layer ID from mISDN_get_layerid() */
int addr;
/*! \brief B channel speech sample data buffer */
char *bframe;
/*! \brief B channel speech sample data buffer size */
int bframe_len;
int time_usec; /* Not used */
/*! \brief Not used. Contents are setup but not used. */
void *astbuf;
/*! \brief TRUE if the TE side should choose the B channel to use
* \note This value is user configurable in /etc/asterisk/misdn.conf
*/
int te_choose_channel;
/*! \brief TRUE if the call progress indicators can indicate an inband audio message for the user to listen to
* \note This value is user configurable in /etc/asterisk/misdn.conf
*/
int early_bconnect;
/*! \brief Last decoded DTMF digit from mISDN driver */
int dtmf;
/*! \brief TRUE if we should produce DTMF tones ourselves
* \note This value is user configurable in /etc/asterisk/misdn.conf
*/
int send_dtmf;
/*! \brief TRUE if we send SETUP_ACKNOWLEDGE on incoming calls anyway (instead of PROCEEDING).
*
* This requests additional INFORMATION messages, so we can
* wait for digits without issues.
* \note This value is user configurable in /etc/asterisk/misdn.conf
*/
int need_more_infos;
/*! \brief TRUE if all digits necessary to complete the call are available.
* No more INFORMATION messages are needed.
*/
int sending_complete;
/*! \brief TRUE if we will not use jollys dsp */
int nodsp;
/*! \brief TRUE if we will not use the jitter buffer system */
int nojitter;
/*! \brief Progress Indicator IE coding standard field.
* \note Collected from the incoming messages but not used.
*/
int progress_coding;
/*! \brief Progress Indicator IE location field.
* \note Collected from the incoming messages but not used.
*/
int progress_location;
/*! \brief Progress Indicator IE progress description field.
* Used to determine if there is an inband audio message present.
*/
int progress_indicator;
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \brief TRUE if waiting for DivertingLegInformation3 to queue redirecting update.
*/
int div_leg_3_rx_wanted;
/*!
* \brief TRUE if a DivertingLegInformation3 needs to be sent with CONNECT.
*/
int div_leg_3_tx_pending;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
/*! \brief Inbound FACILITY message function type and contents */
struct FacParm fac_in;
/*! \brief Outbound FACILITY message function type and contents.
* \note Filled in by misdn facility commands before FACILITY message sent.
*/
struct FacParm fac_out;
/* storing the current AOCD info here */
enum FacFunction AOCDtype;
union {
struct FacAOCDCurrency currency;
struct FacAOCDChargingUnit chargingUnit;
} AOCD;
/*! \brief TRUE if AOCDtype and AOCD data are ready to export to Asterisk */
int AOCD_need_export;
/*! \brief Event waiting for Layer 1 to come up */
enum event_e evq;
/*** CRYPTING STUFF ***/
int crypt; /* Initialized, Not used */
int curprx; /* Initialized, Not used */
int curptx; /* Initialized, Not used */
/*! \brief Blowfish encryption key string (secret) */
char crypt_key[255];
int crypt_state; /* Not used */
/*** CRYPTING STUFF END***/
/*! \brief Seems to have been intended for something to do with the jitter buffer.
* \note Used as a boolean. Only initialized to 0 and referenced in a couple places
*/
int active;
int upset; /* Not used */
/*! \brief TRUE if tone generator allowed to start */
int generate_tone;
/*! \brief Number of tone samples to generate */
int tone_cnt;
/*! \brief Current B Channel state */
enum bchannel_state bc_state;
/*! \brief This is used as a pending bridge join request for when bc_state becomes BCHAN_ACTIVATED */
enum bchannel_state next_bc_state;
/*! \brief Bridging conference ID */
int conf_id;
/*! \brief TRUE if this channel is on hold */
int holded;
/*! \brief TRUE if this channel is on the misdn_stack->holding list
* \note If TRUE this implies that the structure is also malloced.
*/
int stack_holder;
/*!
* \brief Put a display ie in the CONNECT message
* \details
* Put a display ie in the CONNECT message containing the following
* information if it is available (nt port only):
* 0 - Do not put the connected line information in the display ie.
* 1 - Put the available connected line name in the display ie.
* 2 - Put the available connected line number in the display ie.
* 3 - Put the available connected line name and number in the display ie.
*/
int display_connected;
/*!
* \brief Put a display ie in the SETUP message
* \details
* Put a display ie in the SETUP message containing the following
* information if it is available (nt port only):
* 0 - Do not put the caller information in the display ie.
* 1 - Put the available caller name in the display ie.
* 2 - Put the available caller number in the display ie.
* 3 - Put the available caller name and number in the display ie.
*/
int display_setup;
/*!
* \brief Select what to do with outgoing COLP information.
* \details
* 0 - pass (Send out COLP information unaltered.)
* 1 - restricted (Force COLP to restricted on all outgoing COLP information.)
* 2 - block (Do not send COLP information.)
*/
int outgoing_colp;
/*! \brief User set presentation restriction code
* 0=Allowed, 1=Restricted, 2=Unavailable
* \note It is settable by the misdn_set_opt() application.
*/
int presentation;
/*! \brief TRUE if the user set the presentation restriction code */
int set_presentation;
/*! \brief Notification indicator ie description code */
enum mISDN_NOTIFY_CODE notify_description_code;
/*! \brief SETUP message bearer capability field code value */
int capability;
/*! \brief Companding ALaw/uLaw encoding (INFO_CODEC_ALAW / INFO_CODEC_ULAW) */
int law;
/* V110 Stuff */
/*! \brief Q.931 Bearer Capability IE Information Transfer Rate field. Initialized to 0x10 (64kbit). Altered by incoming SETUP messages. */
int rate;
/*! \brief Q.931 Bearer Capability IE Transfer Mode field. Initialized to 0 (Circuit). Altered by incoming SETUP messages. */
int mode;
/*! \brief Q.931 Bearer Capability IE User Information Layer 1 Protocol field code.
* \note Collected from the incoming SETUP message but not used.
*/
int user1;
/*! \brief Q.931 Bearer Capability IE Layer 1 User Rate field.
* \note Collected from the incoming SETUP message and exported to Asterisk variable MISDN_URATE.
*/
int urate;
/*! \brief TRUE if call made in digital HDLC mode
* \note This value is user configurable in /etc/asterisk/misdn.conf.
* It is also settable by the misdn_set_opt() application.
*/
int hdlc;
/* V110 */
/*! \brief Display message that can be displayed by the user phone.
* \note Maximum displayable length is 34 or 82 octets.
* It is also settable by the misdn_set_opt() application.
*/
char display[84];
/*! \brief Q.931 Keypad Facility IE contents
* \note Contents exported and imported to Asterisk variable MISDN_KEYPAD
*/
char keypad[MISDN_MAX_KEYPAD_LEN];
/*! \brief Current overlap dialing digits to/from INFORMATION messages */
char info_dad[MISDN_MAX_NUMBER_LEN];
/*! \brief Collected digits to go into info_dad[] while waiting for a SETUP_ACKNOWLEDGE to come in. */
char infos_pending[MISDN_MAX_NUMBER_LEN];
/* unsigned char info_keypad[MISDN_MAX_KEYPAD_LEN]; */
/* unsigned char clisub[24]; */
/* unsigned char cldsub[24]; */
/*! \brief User-User information string.
* \note Contents exported and imported to Asterisk variable MISDN_USERUSER
* \note We only support ASCII strings (IA5 characters).
*/
char uu[256];
/*! \brief User-User information string length in uu[] */
int uulen;
/*! \brief Q.931 Cause for disconnection code (received)
* \note Need to use the AST_CAUSE_xxx code definitions in causes.h
*/
int cause;
/*! \brief Q.931 Cause for disconnection code (sent)
* \note Need to use the AST_CAUSE_xxx code definitions in causes.h
* \note -1 is used to suppress including the cause code in the RELEASE message.
*/
int out_cause;
/* struct misdn_bchannel hold_bc; */
/** list stuf **/
#ifdef MISDN_1_2
/*! \brief The configuration string for the mISDN dsp pipeline in /etc/asterisk/misdn.conf. */
char pipeline[128];
#else
/*! \brief TRUE if the echo cancellor is enabled */
int ec_enable;
/*! \brief Number of taps in the echo cancellor when enabled.
* \note This value is user configurable in /etc/asterisk/misdn.conf (echocancel)
*/
int ec_deftaps;
#endif
/*! \brief TRUE if the channel was allocated from the available B channels */
int channel_found;
/*! \brief Who originated the call (ORG_AST, ORG_MISDN)
* \note Set but not used when the misdn_set_opt() application enables echo cancellation.
*/
int orig;
/*! \brief Tx gain setting (range -8 to 8)
* \note This value is user configurable in /etc/asterisk/misdn.conf.
* It is also settable by the misdn_set_opt() application.
*/
int txgain;
/*! \brief Rx gain setting (range -8 to 8)
* \note This value is user configurable in /etc/asterisk/misdn.conf.
* It is also settable by the misdn_set_opt() application.
*/
int rxgain;
/*! \brief Next node in the misdn_stack.holding list */
struct misdn_bchannel *next;
};
extern enum event_response_e (*cb_event) (enum event_e event, struct misdn_bchannel *bc, void *user_data);
extern void (*cb_log) (int level, int port, char *tmpl, ...)
__attribute__ ((format (printf, 3, 4)));
extern int (*cb_jb_empty)(struct misdn_bchannel *bc, char *buffer, int len);
struct misdn_lib_iface {
enum event_response_e (*cb_event)(enum event_e event, struct misdn_bchannel *bc, void *user_data);
void (*cb_log)(int level, int port, char *tmpl, ...)
__attribute__ ((format (printf, 3, 4)));
int (*cb_jb_empty)(struct misdn_bchannel *bc, char *buffer, int len);
};
/***** USER IFACE **********/
void misdn_lib_nt_keepcalls(int kc);
void misdn_lib_nt_debug_init( int flags, char *file );
int misdn_lib_init(char *portlist, struct misdn_lib_iface* iface, void *user_data);
int misdn_lib_send_event(struct misdn_bchannel *bc, enum event_e event );
void misdn_lib_destroy(void);
void misdn_lib_isdn_l1watcher(int port);
void misdn_lib_log_ies(struct misdn_bchannel *bc);
char *manager_isdn_get_info(enum event_e event);
struct misdn_bchannel* misdn_lib_get_free_bc(int port, int channel, int inout, int dec);
#if defined(AST_MISDN_ENHANCEMENTS)
struct misdn_bchannel *misdn_lib_get_register_bc(int port);
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
void manager_bchannel_activate(struct misdn_bchannel *bc);
void manager_bchannel_deactivate(struct misdn_bchannel * bc);
int misdn_lib_tx2misdn_frm(struct misdn_bchannel *bc, void *data, int len);
void manager_ph_control(struct misdn_bchannel *bc, int c1, int c2);
void isdn_lib_update_rxgain (struct misdn_bchannel *bc);
void isdn_lib_update_txgain (struct misdn_bchannel *bc);
void isdn_lib_update_ec (struct misdn_bchannel *bc);
void isdn_lib_stop_dtmf (struct misdn_bchannel *bc);
int misdn_lib_port_restart(int port);
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@49321 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-03 11:15:02 +00:00
int misdn_lib_pid_restart(int pid);
int misdn_lib_send_restart(int port, int channel);
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@49321 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-03 11:15:02 +00:00
int misdn_lib_get_port_info(int port);
int misdn_lib_is_port_blocked(int port);
int misdn_lib_port_block(int port);
int misdn_lib_port_unblock(int port);
int misdn_lib_port_is_pri(int port);
int misdn_lib_port_is_nt(int port);
int misdn_lib_port_up(int port, int notcheck);
int misdn_lib_get_port_down(int port);
int misdn_lib_get_port_up (int port) ;
int misdn_lib_maxports_get(void) ;
struct misdn_bchannel *misdn_lib_find_held_bc(int port, int l3_id);
void misdn_lib_release(struct misdn_bchannel *bc);
int misdn_cap_is_speech(int cap);
int misdn_inband_avail(struct misdn_bchannel *bc);
void manager_ec_enable(struct misdn_bchannel *bc);
void manager_ec_disable(struct misdn_bchannel *bc);
void misdn_lib_send_tone(struct misdn_bchannel *bc, enum tone_e tone);
void get_show_stack_details(int port, char *buf);
void misdn_lib_tone_generator_start(struct misdn_bchannel *bc);
void misdn_lib_tone_generator_stop(struct misdn_bchannel *bc);
void misdn_lib_setup_bc(struct misdn_bchannel *bc);
void misdn_lib_bridge( struct misdn_bchannel * bc1, struct misdn_bchannel *bc2);
void misdn_lib_split_bridge( struct misdn_bchannel * bc1, struct misdn_bchannel *bc2);
void misdn_lib_echo(struct misdn_bchannel *bc, int onoff);
int misdn_lib_is_ptp(int port);
int misdn_lib_get_maxchans(int port);
void misdn_lib_reinit_nt_stack(int port);
#define PRI_TRANS_CAP_SPEECH 0x0
#define PRI_TRANS_CAP_DIGITAL 0x08
#define PRI_TRANS_CAP_RESTRICTED_DIGITAL 0x09
#define PRI_TRANS_CAP_3_1K_AUDIO 0x10
#define PRI_TRANS_CAP_7K_AUDIO 0x11
char *bc_state2str(enum bchannel_state state);
void bc_state_change(struct misdn_bchannel *bc, enum bchannel_state state);
void misdn_dump_chanlist(void);
void misdn_make_dummy(struct misdn_bchannel *dummybc, int port, int l3id, int nt, int channel);
#endif /* TE_LIB */