asterisk/channels/sig_analog.h

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#ifndef _SIG_ANALOG_H
#define _SIG_ANALOG_H
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2009, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* 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 Interface header for analog signaling module
*
* \author Matthew Fredrickson <creslin@digium.com>
*/
#include "asterisk/channel.h"
#include "asterisk/frame.h"
/* Signalling types supported */
enum analog_sigtype {
ANALOG_SIG_NONE = -1,
ANALOG_SIG_FXOLS = 1,
ANALOG_SIG_FXOKS,
ANALOG_SIG_FXOGS,
ANALOG_SIG_FXSLS,
ANALOG_SIG_FXSKS,
ANALOG_SIG_FXSGS,
ANALOG_SIG_EMWINK,
ANALOG_SIG_EM,
ANALOG_SIG_EM_E1,
ANALOG_SIG_FEATD,
ANALOG_SIG_FEATDMF,
ANALOG_SIG_E911,
ANALOG_SIG_FGC_CAMA,
ANALOG_SIG_FGC_CAMAMF,
ANALOG_SIG_FEATB,
ANALOG_SIG_SFWINK,
ANALOG_SIG_SF,
ANALOG_SIG_SF_FEATD,
ANALOG_SIG_SF_FEATDMF,
ANALOG_SIG_FEATDMF_TA,
ANALOG_SIG_SF_FEATB,
};
enum analog_tone {
ANALOG_TONE_RINGTONE = 0,
ANALOG_TONE_STUTTER,
ANALOG_TONE_CONGESTION,
ANALOG_TONE_DIALTONE,
ANALOG_TONE_DIALRECALL,
ANALOG_TONE_INFO,
};
enum analog_event {
ANALOG_EVENT_NONE = 0,
ANALOG_EVENT_ONHOOK,
ANALOG_EVENT_RINGOFFHOOK,
ANALOG_EVENT_WINKFLASH,
ANALOG_EVENT_ALARM,
ANALOG_EVENT_NOALARM,
ANALOG_EVENT_DIALCOMPLETE,
ANALOG_EVENT_RINGERON,
ANALOG_EVENT_RINGEROFF,
ANALOG_EVENT_HOOKCOMPLETE,
ANALOG_EVENT_PULSE_START,
ANALOG_EVENT_POLARITY,
ANALOG_EVENT_RINGBEGIN,
ANALOG_EVENT_EC_DISABLED,
ANALOG_EVENT_REMOVED,
ANALOG_EVENT_NEONMWI_ACTIVE,
ANALOG_EVENT_NEONMWI_INACTIVE,
ANALOG_EVENT_TX_CED_DETECTED,
ANALOG_EVENT_RX_CED_DETECTED,
ANALOG_EVENT_EC_NLP_DISABLED,
ANALOG_EVENT_EC_NLP_ENABLED,
ANALOG_EVENT_ERROR, /* not a DAHDI event */
ANALOG_EVENT_DTMFCID, /* not a DAHDI event */
ANALOG_EVENT_PULSEDIGIT = (1 << 16),
ANALOG_EVENT_DTMFDOWN = (1 << 17),
ANALOG_EVENT_DTMFUP = (1 << 18),
};
enum analog_sub {
ANALOG_SUB_REAL = 0, /*!< Active call */
ANALOG_SUB_CALLWAIT, /*!< Call-Waiting call on hold */
ANALOG_SUB_THREEWAY, /*!< Three-way call */
};
enum analog_dsp_digitmode {
ANALOG_DIGITMODE_DTMF = 1,
ANALOG_DIGITMODE_MF,
};
enum analog_cid_start {
ANALOG_CID_START_POLARITY = 1,
ANALOG_CID_START_POLARITY_IN,
ANALOG_CID_START_RING,
ANALOG_CID_START_DTMF_NOALERT,
};
#define ANALOG_MAX_CID 300
enum dialop {
ANALOG_DIAL_OP_REPLACE = 2,
};
struct analog_dialoperation {
enum dialop op;
char dialstr[256];
};
struct analog_callback {
/* Unlock the private in the signalling private structure. This is used for three way calling madness. */
void (* const unlock_private)(void *pvt);
/* Lock the private in the signalling private structure. ... */
void (* const lock_private)(void *pvt);
/* Function which is called back to handle any other DTMF up events that are received. Called by analog_handle_event. Why is this
* important to use, instead of just directly using events received before they are passed into the library? Because sometimes,
* (CWCID) the library absorbs DTMF events received. */
void (* const handle_dtmfup)(void *pvt, struct ast_channel *ast, enum analog_sub analog_index, struct ast_frame **dest);
int (* const get_event)(void *pvt);
int (* const wait_event)(void *pvt);
int (* const is_off_hook)(void *pvt);
int (* const is_dialing)(void *pvt, enum analog_sub sub);
/* Start a trunk type signalling protocol (everything except phone ports basically */
int (* const start)(void *pvt);
int (* const ring)(void *pvt);
int (* const flash)(void *pvt);
/*! \brief Set channel on hook */
int (* const on_hook)(void *pvt);
/*! \brief Set channel off hook */
int (* const off_hook)(void *pvt);
void (* const set_needringing)(void *pvt, int value);
/* We're assuming that we're going to only wink on ANALOG_SUB_REAL - even though in the code there's an argument to the index
* function */
int (* const wink)(void *pvt, enum analog_sub sub);
int (* const dial_digits)(void *pvt, enum analog_sub sub, struct analog_dialoperation *dop);
int (* const send_fsk)(void *pvt, struct ast_channel *ast, char *fsk);
int (* const play_tone)(void *pvt, enum analog_sub sub, enum analog_tone tone);
int (* const set_echocanceller)(void *pvt, int enable);
int (* const train_echocanceller)(void *pvt);
int (* const dsp_set_digitmode)(void *pvt, enum analog_dsp_digitmode mode);
int (* const dsp_reset_and_flush_digits)(void *pvt);
int (* const send_callerid)(void *pvt, int cwcid, struct ast_callerid *cid);
/* Returns 0 if CID received. Returns 1 if event received, and -1 if error. name and num are size ANALOG_MAX_CID */
int (* const get_callerid)(void *pvt, char *name, char *num, enum analog_event *ev, size_t timeout);
/* Start CID detection */
int (* const start_cid_detect)(void *pvt, int cid_signalling);
/* Stop CID detection */
int (* const stop_cid_detect)(void *pvt);
/* Play the CAS callwait tone on the REAL sub, then repeat after 10 seconds, and then stop */
int (* const callwait)(void *pvt);
/* Stop playing any CAS call waiting announcement tones that might be running on the REAL sub */
int (* const stop_callwait)(void *pvt);
/* Bearer control related (non signalling) callbacks */
int (* const allocate_sub)(void *pvt, enum analog_sub sub);
int (* const unallocate_sub)(void *pvt, enum analog_sub sub);
/*! This function is for swapping of the owners with the underlying subs. Typically it means you need to change the fds
* of the new owner to be the fds of the sub specified, for each of the two subs given */
void (* const swap_subs)(void *pvt, enum analog_sub a, struct ast_channel *new_a_owner, enum analog_sub b, struct ast_channel *new_b_owner);
struct ast_channel * (* const new_ast_channel)(void *pvt, int state, int startpbx, enum analog_sub sub, const struct ast_channel *requestor);
/* Add the given sub to a conference */
int (* const conf_add)(void *pvt, enum analog_sub sub);
/* Delete the given sub from any conference that might be running on the channels */
int (* const conf_del)(void *pvt, enum analog_sub sub);
/* If you would like to do any optimizations after the conference members have been added and removed,
* you can do so here */
int (* const complete_conference_update)(void *pvt, int needconf);
/* This is called when there are no more subchannels on the given private that are left up,
* for any cleanup or whatever else you would like to do. Called from analog_hangup() */
void (* const all_subchannels_hungup)(void *pvt);
int (* const has_voicemail)(void *pvt);
int (* const check_for_conference)(void *pvt);
void (* const handle_notify_message)(struct ast_channel *chan, void *pvt, int cid_flags, int neon_mwievent);
/* callbacks for increasing and decreasing ss_thread_count, will handle locking and condition signal */
void (* const increase_ss_count)(void);
void (* const decrease_ss_count)(void);
int (* const distinctive_ring)(struct ast_channel *chan, void *pvt, int idx, int *ringdata);
/* Sets the specified sub-channel in and out of signed linear mode, returns the value that was overwritten */
int (* const set_linear_mode)(void *pvt, int idx, int linear_mode);
void (* const get_and_handle_alarms)(void *pvt);
void * (* const get_sigpvt_bridged_channel)(struct ast_channel *chan);
int (* const get_sub_fd)(void *pvt, enum analog_sub sub);
void (* const set_cadence)(void *pvt, int *cidrings, struct ast_channel *chan);
void (* const set_alarm)(void *pvt, int in_alarm);
void (* const set_dialing)(void *pvt, int is_dialing);
void (* const set_ringtimeout)(void *pvt, int ringt);
void (* const set_waitingfordt)(void *pvt, struct ast_channel *ast);
int (* const check_waitingfordt)(void *pvt);
void (* const set_confirmanswer)(void *pvt, int flag);
int (* const check_confirmanswer)(void *pvt);
void (* const cancel_cidspill)(void *pvt);
int (* const confmute)(void *pvt, int mute);
void (* const set_pulsedial)(void *pvt, int flag);
Merge Call completion support into trunk. From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 15:31:32 +00:00
const char *(* const get_orig_dialstring)(void *pvt);
};
#define READ_SIZE 160
struct analog_subchannel {
struct ast_channel *owner;
struct ast_frame f; /*!< One frame for each channel. How did this ever work before? */
unsigned int inthreeway:1;
/* Have we allocated a subchannel yet or not */
unsigned int allocd:1;
};
struct analog_pvt {
/* Analog signalling type used in this private */
enum analog_sigtype sig;
/* To contain the private structure passed into the channel callbacks */
void *chan_pvt;
/* Callbacks for various functions needed by the analog API */
struct analog_callback *calls;
/* All members after this are giong to be transient, and most will probably change */
struct ast_channel *owner; /*!< Our current active owner (if applicable) */
struct analog_subchannel subs[3]; /*!< Sub-channels */
struct analog_dialoperation dop;
int onhooktime; /*< Time the interface went on-hook. */
int fxsoffhookstate; /*< TRUE if the FXS port is off-hook */
/*! \brief -1 = unknown, 0 = no messages, 1 = new messages available */
int msgstate;
/* XXX: Option Variables - Set by allocator of private structure */
unsigned int answeronpolarityswitch:1;
unsigned int callreturn:1;
unsigned int cancallforward:1;
unsigned int canpark:1;
unsigned int dahditrcallerid:1; /*!< should we use the callerid from incoming call on dahdi transfer or not */
unsigned int hanguponpolarityswitch:1;
unsigned int immediate:1;
unsigned int permcallwaiting:1;
unsigned int permhidecallerid:1; /*!< Whether to hide our outgoing caller ID or not */
unsigned int pulse:1;
unsigned int threewaycalling:1;
unsigned int transfer:1;
unsigned int transfertobusy:1; /*!< allow flash-transfers to busy channels */
unsigned int use_callerid:1; /*!< Whether or not to use caller id on this channel */
const struct ast_channel_tech *chan_tech;
/*!
* \brief TRUE if distinctive rings are to be detected.
* \note For FXO lines
* \note Set indirectly from the "usedistinctiveringdetection" value read in from chan_dahdi.conf
*/
unsigned int usedistinctiveringdetection:1;
/* Not used for anything but log messages. Could be just the TCID */
int channel; /*!< Channel Number */
enum analog_sigtype outsigmod;
int echotraining;
int cid_signalling; /*!< Asterisk callerid type we're using */
int polarityonanswerdelay;
int stripmsd;
enum analog_cid_start cid_start;
int callwaitingcallerid;
char mohsuggest[MAX_MUSICCLASS];
char cid_num[AST_MAX_EXTENSION];
char cid_name[AST_MAX_EXTENSION];
/* XXX: All variables after this are internal */
unsigned int callwaiting:1;
unsigned int dialednone:1;
unsigned int dialing:1; /*!< TRUE if in the process of dialing digits or sending something */
unsigned int dnd:1; /*!< TRUE if Do-Not-Disturb is enabled. */
unsigned int echobreak:1;
unsigned int hidecallerid:1;
unsigned int outgoing:1;
char callwait_num[AST_MAX_EXTENSION];
char callwait_name[AST_MAX_EXTENSION];
char lastcid_num[AST_MAX_EXTENSION];
char lastcid_name[AST_MAX_EXTENSION];
struct ast_callerid cid;
int cidrings; /*!< Which ring to deliver CID on */
char echorest[20];
int polarity;
struct timeval polaritydelaytv;
char dialdest[256];
time_t guardtime; /*!< Must wait this much time before using for new call */
struct timeval flashtime; /*!< Last flash-hook time */
int whichwink; /*!< SIG_FEATDMF_TA Which wink are we on? */
char finaldial[64];
char *origcid_num; /*!< malloced original callerid */
char *origcid_name; /*!< malloced original callerid */
char call_forward[AST_MAX_EXTENSION];
/* Ast channel to pass to __ss_analog_thread */
struct ast_channel *ss_astchan;
/* All variables after this are definitely going to be audited */
unsigned int inalarm:1;
unsigned int unknown_alarm:1;
int callwaitcas;
int ringt;
int ringt_base;
};
struct analog_pvt *analog_new(enum analog_sigtype signallingtype, struct analog_callback *c, void *private_data);
void analog_delete(struct analog_pvt *doomed);
void analog_free(struct analog_pvt *p);
int analog_call(struct analog_pvt *p, struct ast_channel *ast, char *rdest, int timeout);
int analog_hangup(struct analog_pvt *p, struct ast_channel *ast);
int analog_answer(struct analog_pvt *p, struct ast_channel *ast);
struct ast_frame *analog_exception(struct analog_pvt *p, struct ast_channel *ast);
struct ast_channel * analog_request(struct analog_pvt *p, int *callwait, const struct ast_channel *requestor);
int analog_available(struct analog_pvt *p);
void *analog_handle_init_event(struct analog_pvt *i, int event);
int analog_config_complete(struct analog_pvt *p);
void analog_handle_dtmfup(struct analog_pvt *p, struct ast_channel *ast, enum analog_sub index, struct ast_frame **dest);
enum analog_cid_start analog_str_to_cidstart(const char *value);
const char *analog_cidstart_to_str(enum analog_cid_start cid_start);
enum analog_sigtype analog_str_to_sigtype(const char *name);
const char *analog_sigtype_to_str(enum analog_sigtype sigtype);
unsigned int analog_str_to_cidtype(const char *name);
const char *analog_cidtype_to_str(unsigned int cid_type);
int analog_ss_thread_start(struct analog_pvt *p, struct ast_channel *ast);
int analog_fixup(struct ast_channel *oldchan, struct ast_channel *newchan, void *newp);
int analog_dnd(struct analog_pvt *p, int flag);
#endif /* _SIG_ANSLOG_H */