asterisk/channels/sig_analog.h
Richard Mudgett ccdc417ab5 Merged revisions 296167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r296167 | rmudgett | 2010-11-24 16:49:48 -0600 (Wed, 24 Nov 2010) | 57 lines
  
  Merged revisions 296166 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r296166 | rmudgett | 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) | 50 lines
    
    Merged revisions 296165 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines
      
      Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.
      
      The FXS connected phone has to have CW/CID support to fail, as it will
      send back a DTMF 'A' or 'D' when it's ready to receive CallerID.  A normal
      phone with no CID never fails.  Also the SIP phone does not hear MOH when
      the CW call is answered.
      
      The DTMF end frame is suppressed when the phone acknowledges the CW signal
      for CID.  The problem is the DTMF begin frame needs to be suppressed as
      well.  The DTMF begin frame is causing SIP to start sending the DTMF RTP
      frames.  Since the DTMF end frame is suppressed, SIP will not stop sending
      those DTMF RTP packets.
      
      * Suppress the DTMF begin and end frames when the channel driver is
      looking for DTMF digits.
      
      * Fixed a couple issues caused by not cleaning up the CID spill if you
      answer the CW call while it is sending the CID spill.
      
      * Fixed not sending CW/CID spill to the phone when the call is natively
      bridged.  (Fixed by not using native bridge if CW/CID is possible.)
      
      * Suppress received audio when sending CW/CID spills.  The other parties
      involved do not need to hear the CW/CID spills and may be confused if the
      CW call is for them.
      
      (closes issue #18129)
      Reported by: alecdavis
      Patches:
            issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664)
      Tested by: alecdavis, rmudgett
      
      
      NOTE:
      
      * v1.4 does not have the main problem fixed by suppressing the DTMF start
      frames.  The other three items fixed are relevant.
      
      * If you really must restore native bridging between analog ports, you
      need to disable CW/CID either by configuring chan_dahdi.conf
      callwaitingcallerid=no or dialing *70 before dialing the number to
      temporarily disable CW.
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24 22:52:07 +00:00

390 lines
15 KiB
C

#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"
#include "asterisk/smdi.h"
#define ANALOG_SMDI_MD_WAIT_TIMEOUT 1500 /* 1.5 seconds */
#define ANALOG_MAX_CID 300
#define READ_SIZE 160
#define RING_PATTERNS 3
/* 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,
};
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);
/* Do deadlock avoidance for the private signaling structure lock. */
void (* const deadlock_avoidance_private)(void *pvt);
/* Function which is called back to handle any other DTMF 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_dtmf)(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);
/*! \brief Set FXS line polarity to 0=IDLE NZ=REVERSED */
void (* const set_polarity)(void *pvt, int value);
/*! \brief Reset FXS line polarity to IDLE, based on answeronpolarityswitch and hanguponpolarityswitch */
void (* const start_polarityswitch)(void *pvt);
/*! \brief Switch FXS line polarity, based on answeronpolarityswitch=yes */
void (* const answer_polarityswitch)(void *pvt);
/*! \brief Switch FXS line polarity, based on answeronpolarityswitch and hanguponpolarityswitch */
void (* const hangup_polarityswitch)(void *pvt);
/* 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_party_caller *caller);
/* 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, enum analog_sub sub, int linear_mode);
void (* const set_inthreeway)(void *pvt, enum analog_sub sub, int inthreeway);
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 set_callwaiting)(void *pvt, int callwaiting_enable);
void (* const cancel_cidspill)(void *pvt);
int (* const confmute)(void *pvt, int mute);
void (* const set_pulsedial)(void *pvt, int flag);
void (* const set_new_owner)(void *pvt, struct ast_channel *new_owner);
const char *(* const get_orig_dialstring)(void *pvt);
};
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; /*!< TRUE if call waiting is enabled. (Configured option) */
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 */
unsigned int callwaitingcallerid:1; /*!< TRUE if send caller ID for Call Waiting */
/*!
* \brief TRUE if SMDI (Simplified Message Desk Interface) is enabled
*/
unsigned int use_smdi:1;
/*! \brief The SMDI interface to get SMDI messages from. */
struct ast_smdi_interface *smdi_iface;
const struct ast_channel_tech *chan_tech;
/* 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;
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; /*!< TRUE if call waiting is enabled. (Active option) */
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;
unsigned int inalarm:1;
/*!
* \brief TRUE if Call Waiting (CW) CPE Alert Signal (CAS) is being sent.
* \note
* After CAS is sent, the call waiting caller id will be sent if the phone
* gives a positive reply.
*/
unsigned int callwaitcas: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_party_caller caller;
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 */
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_dtmf(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 */