asterisk/channels/sig_pri.h

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#ifndef _SIG_PRI_H
#define _SIG_PRI_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 PRI signaling module
*
* \author Matthew Fredrickson <creslin@digium.com>
*/
#include "asterisk/channel.h"
#include "asterisk/frame.h"
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 An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch.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
#include "asterisk/ccss.h"
#include <libpri.h>
#include <dahdi/user.h>
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
#define SIG_PRI_AOC_GRANT_S (1 << 0)
#define SIG_PRI_AOC_GRANT_D (1 << 1)
#define SIG_PRI_AOC_GRANT_E (1 << 2)
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
#if defined(HAVE_PRI_CCSS)
/*! PRI debug message flags when normal PRI debugging is turned on at the command line. */
#define SIG_PRI_DEBUG_NORMAL \
(PRI_DEBUG_APDU | PRI_DEBUG_Q931_DUMP | PRI_DEBUG_Q931_STATE | PRI_DEBUG_Q921_STATE \
| PRI_DEBUG_CC)
/*! PRI debug message flags when intense PRI debugging is turned on at the command line. */
#define SIG_PRI_DEBUG_INTENSE \
(PRI_DEBUG_APDU | PRI_DEBUG_Q931_DUMP | PRI_DEBUG_Q931_STATE | PRI_DEBUG_Q921_STATE \
| PRI_DEBUG_CC | PRI_DEBUG_Q921_RAW | PRI_DEBUG_Q921_DUMP)
#else
/*! PRI debug message flags when normal PRI debugging is turned on at the command line. */
#define SIG_PRI_DEBUG_NORMAL \
(PRI_DEBUG_APDU | PRI_DEBUG_Q931_DUMP | PRI_DEBUG_Q931_STATE | PRI_DEBUG_Q921_STATE)
/*! PRI debug message flags when intense PRI debugging is turned on at the command line. */
#define SIG_PRI_DEBUG_INTENSE \
(PRI_DEBUG_APDU | PRI_DEBUG_Q931_DUMP | PRI_DEBUG_Q931_STATE | PRI_DEBUG_Q921_STATE \
| PRI_DEBUG_Q921_RAW | PRI_DEBUG_Q921_DUMP)
#endif /* !defined(HAVE_PRI_CCSS) */
#if 0
/*! PRI debug message flags set on initial startup. */
#define SIG_PRI_DEBUG_DEFAULT SIG_PRI_DEBUG_NORMAL
#else
/*! PRI debug message flags set on initial startup. */
#define SIG_PRI_DEBUG_DEFAULT 0
#endif
enum sig_pri_tone {
SIG_PRI_TONE_RINGTONE = 0,
SIG_PRI_TONE_STUTTER,
SIG_PRI_TONE_CONGESTION,
SIG_PRI_TONE_DIALTONE,
SIG_PRI_TONE_DIALRECALL,
SIG_PRI_TONE_INFO,
SIG_PRI_TONE_BUSY,
};
enum sig_pri_law {
SIG_PRI_DEFLAW = 0,
SIG_PRI_ULAW,
SIG_PRI_ALAW
};
struct sig_pri_pri;
struct sig_pri_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 dial_digits)(void *pvt, enum analog_sub sub, struct analog_dialoperation *dop);
int (* const play_tone)(void *pvt, enum sig_pri_tone tone);
int (* const set_echocanceller)(void *pvt, int enable);
int (* const train_echocanceller)(void *pvt);
int (* const dsp_reset_and_flush_digits)(void *pvt);
struct ast_channel * (* const new_ast_channel)(void *pvt, int state, int startpbx, enum sig_pri_law law, int transfercapability, char *exten, const struct ast_channel *chan);
void (* const fixup_chans)(void *old_chan, void *new_chan);
/* Note: Called with PRI lock held */
void (* const handle_dchan_exception)(struct sig_pri_pri *pri, int index);
void (* const set_alarm)(void *pvt, int in_alarm);
void (* const set_dialing)(void *pvt, int is_dialing);
void (* const set_digital)(void *pvt, int is_digital);
void (* const set_callerid)(void *pvt, const struct ast_party_caller *caller);
void (* const set_dnid)(void *pvt, const char *dnid);
void (* const set_rdnis)(void *pvt, const char *rdnis);
void (* const queue_control)(void *pvt, int subclass);
int (* const new_nobch_intf)(struct sig_pri_pri *pri);
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);
void (* const make_cc_dialstring)(void *pvt, char *buf, size_t buf_size);
void (* const update_span_devstate)(struct sig_pri_pri *pri);
/*! Reference the parent module. */
void (*module_ref)(void);
/*! Unreference the parent module. */
void (*module_unref)(void);
};
#define SIG_PRI_NUM_DCHANS 4 /*!< No more than 4 d-channels */
#define SIG_PRI_MAX_CHANNELS 672 /*!< No more than a DS3 per trunk group */
#define SIG_PRI DAHDI_SIG_CLEAR
#define SIG_BRI (0x2000000 | DAHDI_SIG_CLEAR)
#define SIG_BRI_PTMP (0X4000000 | DAHDI_SIG_CLEAR)
/* QSIG channel mapping option types */
#define DAHDI_CHAN_MAPPING_PHYSICAL 0
#define DAHDI_CHAN_MAPPING_LOGICAL 1
/* Overlap dialing option types */
#define DAHDI_OVERLAPDIAL_NONE 0
#define DAHDI_OVERLAPDIAL_OUTGOING 1
#define DAHDI_OVERLAPDIAL_INCOMING 2
#define DAHDI_OVERLAPDIAL_BOTH (DAHDI_OVERLAPDIAL_INCOMING|DAHDI_OVERLAPDIAL_OUTGOING)
#if defined(HAVE_PRI_SERVICE_MESSAGES)
/*! \brief Persistent Service State */
#define SRVST_DBKEY "service-state"
/*! \brief The out-of-service SERVICE state */
#define SRVST_TYPE_OOS "O"
/*! \brief SRVST_INITIALIZED is used to indicate a channel being out-of-service
* The SRVST_INITIALIZED is mostly used maintain backwards compatibility but also may
* mean that the channel has not yet received a RESTART message. If a channel is
* out-of-service with this reason a RESTART message will result in the channel
* being put into service. */
#define SRVST_INITIALIZED 0
/*! \brief SRVST_NEAREND is used to indicate that the near end was put out-of-service */
#define SRVST_NEAREND (1 << 0)
/*! \brief SRVST_FAREND is used to indicate that the far end was taken out-of-service */
#define SRVST_FAREND (1 << 1)
/*! \brief SRVST_BOTH is used to indicate that both sides of the channel are out-of-service */
#define SRVST_BOTH (SRVST_NEAREND | SRVST_FAREND)
/*! \brief The AstDB family */
static const char dahdi_db[] = "dahdi/registry";
#endif /* defined(HAVE_PRI_SERVICE_MESSAGES) */
struct sig_pri_chan {
/* Options to be set by user */
unsigned int hidecallerid:1;
unsigned int hidecalleridname:1; /*!< Hide just the name not the number for legacy PBX use */
unsigned int immediate:1; /*!< Answer before getting digits? */
unsigned int priexclusive:1; /*!< Whether or not to override and use exculsive mode for channel selection */
unsigned int priindication_oob:1;
unsigned int use_callerid:1; /*!< Whether or not to use caller id on this channel */
unsigned int use_callingpres:1; /*!< Whether to use the callingpres the calling switch sends */
char context[AST_MAX_CONTEXT];
char mohinterpret[MAX_MUSICCLASS];
int stripmsd;
int channel; /*!< Channel Number or CRV */
/* Options to be checked by user */
int cid_ani2; /*!< Automatic Number Identification number (Alternate PRI caller ID number) */
int cid_ton; /*!< Type Of Number (TON) */
int callingpres; /*!< The value of calling presentation that we're going to use when placing a PRI call */
char cid_num[AST_MAX_EXTENSION];
char cid_subaddr[AST_MAX_EXTENSION];
char cid_name[AST_MAX_EXTENSION];
char cid_ani[AST_MAX_EXTENSION];
char exten[AST_MAX_EXTENSION];
/* Internal variables -- Don't touch */
/* Probably will need DS0 number, DS1 number, and a few other things */
char dialdest[256]; /* Queued up digits for overlap dialing. They will be sent out as information messages when setup ACK is received */
#if defined(HAVE_PRI_SETUP_KEYPAD)
/*! \brief Keypad digits that came in with the SETUP message. */
char keypad_digits[AST_MAX_EXTENSION];
#endif /* defined(HAVE_PRI_SETUP_KEYPAD) */
#if defined(HAVE_PRI_AOC_EVENTS)
struct pri_subcmd_aoc_e aoc_e;
int aoc_s_request_invoke_id; /*!< If an AOC-S request was present for the call, this is the invoke_id to use for the response */
unsigned int aoc_s_request_invoke_id_valid:1; /*!< This is set when the AOC-S invoke id is present */
unsigned int waiting_for_aoce:1; /*!< Delaying hangup for AOC-E msg. If this is set and AOC-E is recieved, continue with hangup before timeout period. */
unsigned int holding_aoce:1; /*!< received AOC-E msg from asterisk. holding for disconnect/release */
#endif /* defined(HAVE_PRI_AOC_EVENTS) */
unsigned int inalarm:1;
unsigned int alerting:1; /*!< TRUE if channel is alerting/ringing */
unsigned int alreadyhungup:1; /*!< TRUE if the call has already gone/hungup */
unsigned int isidlecall:1; /*!< TRUE if this is an idle call */
unsigned int proceeding:1; /*!< TRUE if call is in a proceeding state */
unsigned int progress:1; /*!< TRUE if the call has seen progress through the network */
unsigned int resetting:1; /*!< TRUE if this channel is being reset/restarted */
unsigned int setup_ack:1; /*!< TRUE if this channel has received a SETUP_ACKNOWLEDGE */
unsigned int outgoing:1;
unsigned int digital:1;
/*! \brief TRUE if this interface has no B channel. (call hold and call waiting) */
unsigned int no_b_channel:1;
struct ast_channel *owner;
struct sig_pri_pri *pri;
q931_call *call; /*!< opaque libpri call control structure */
int prioffset; /*!< channel number in span */
int logicalspan; /*!< logical span number within trunk group */
int mastertrunkgroup; /*!< what trunk group is our master */
#if defined(HAVE_PRI_SERVICE_MESSAGES)
/*! \brief Active SRVST_DBKEY out-of-service status value. */
unsigned service_status;
#endif /* defined(HAVE_PRI_SERVICE_MESSAGES) */
struct sig_pri_callback *calls;
void *chan_pvt; /*!< Private structure of the user of this module. */
#if defined(HAVE_PRI_REVERSE_CHARGE)
/*!
* \brief Reverse charging indication
* \details
* -1 - No reverse charging,
* 1 - Reverse charging,
* 0,2-7 - Reserved for future use
*/
int reverse_charging_indication;
#endif
};
struct sig_pri_pri {
/* Should be set by user */
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
struct ast_cc_config_params *cc_params; /*!< CC config parameters for each new call. */
int pritimers[PRI_MAX_TIMERS];
int overlapdial; /*!< In overlap dialing mode */
int qsigchannelmapping; /*!< QSIG channel mapping type */
int discardremoteholdretrieval; /*!< shall remote hold or remote retrieval notifications be discarded? */
int facilityenable; /*!< Enable facility IEs */
int dchan_logical_span[SIG_PRI_NUM_DCHANS]; /*!< Logical offset the DCHAN sits in */
int fds[SIG_PRI_NUM_DCHANS]; /*!< FD's for d-channels */
#if defined(HAVE_PRI_AOC_EVENTS)
int aoc_passthrough_flag; /*!< Represents what AOC messages (S,D,E) are allowed to pass-through */
int aoce_delayhangup:1; /*!< defines whether the aoce_delayhangup option is enabled or not */
#endif /* defined(HAVE_PRI_AOC_EVENTS) */
#if defined(HAVE_PRI_SERVICE_MESSAGES)
unsigned int enable_service_message_support:1; /*!< enable SERVICE message support */
#endif /* defined(HAVE_PRI_SERVICE_MESSAGES) */
#ifdef HAVE_PRI_INBANDDISCONNECT
unsigned int inbanddisconnect:1; /*!< Should we support inband audio after receiving DISCONNECT? */
#endif
#if defined(HAVE_PRI_CALL_HOLD)
/*! \brief TRUE if held calls are transferred on disconnect. */
unsigned int hold_disconnect_transfer:1;
#endif /* defined(HAVE_PRI_CALL_HOLD) */
/*!
* \brief TRUE if call transfer is enabled for the span.
* \note Support switch-side transfer (called 2BCT, RLT or other names)
*/
unsigned int transfer:1;
int dialplan; /*!< Dialing plan */
int localdialplan; /*!< Local dialing plan */
char internationalprefix[10]; /*!< country access code ('00' for european dialplans) */
char nationalprefix[10]; /*!< area access code ('0' for european dialplans) */
char localprefix[20]; /*!< area access code + area code ('0'+area code for european dialplans) */
char privateprefix[20]; /*!< for private dialplans */
char unknownprefix[20]; /*!< for unknown dialplans */
long resetinterval; /*!< Interval (in seconds) for resetting unused channels */
char msn_list[AST_MAX_EXTENSION]; /*!< Comma separated list of MSNs to handle. Empty if disabled. */
char idleext[AST_MAX_EXTENSION]; /*!< Where to idle extra calls */
char idlecontext[AST_MAX_CONTEXT]; /*!< What context to use for idle */
char idledial[AST_MAX_EXTENSION]; /*!< What to dial before dumping */
int minunused; /*!< Min # of channels to keep empty */
int minidle; /*!< Min # of "idling" calls to keep active */
int nodetype; /*!< Node type */
int switchtype; /*!< Type of switch to emulate */
int nsf; /*!< Network-Specific Facilities */
int trunkgroup; /*!< What our trunkgroup is */
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
#if defined(HAVE_PRI_CCSS)
int cc_ptmp_recall_mode; /*!< CC PTMP recall mode. globalRecall(0), specificRecall(1) */
int cc_qsig_signaling_link_req; /*!< CC Q.SIG signaling link retention (Party A) release(0), retain(1), do-not-care(2) */
int cc_qsig_signaling_link_rsp; /*!< CC Q.SIG signaling link retention (Party B) release(0), retain(1) */
#endif /* defined(HAVE_PRI_CCSS) */
int dchanavail[SIG_PRI_NUM_DCHANS]; /*!< Whether each channel is available */
int debug; /*!< set to true if to dump PRI event info (tested but never set) */
int span; /*!< span number put into user output messages */
int resetting; /*!< true if span is being reset/restarted */
int resetpos; /*!< current position during a reset (-1 if not started) */
int sig; /*!< ISDN signalling type (SIG_PRI, SIG_BRI, SIG_BRI_PTMP, etc...) */
int new_chan_seq; /*!< New struct ast_channel sequence number */
/* Everything after here is internally set */
struct pri *dchans[SIG_PRI_NUM_DCHANS]; /*!< Actual d-channels */
struct pri *pri; /*!< Currently active D-channel */
/*!
* List of private structures of the user of this module for no B channel
* interfaces. (hold and call waiting interfaces)
*/
void *no_b_chan_iflist;
/*!
* List of private structures of the user of this module for no B channel
* interfaces. (hold and call waiting interfaces)
*/
void *no_b_chan_end;
int numchans; /*!< Num of channels we represent */
struct sig_pri_chan *pvts[SIG_PRI_MAX_CHANNELS];/*!< Member channel pvt structs */
pthread_t master; /*!< Thread of master */
ast_mutex_t lock; /*!< libpri access Mutex */
time_t lastreset; /*!< time when unused channels were last reset */
struct sig_pri_callback *calls;
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
/*!
* \brief Congestion device state of the span.
* \details
* AST_DEVICE_NOT_INUSE - Span does not have all B channels in use.
* AST_DEVICE_BUSY - All B channels are in use.
* AST_DEVICE_UNAVAILABLE - Span is in alarm.
* \note
* Device name: DAHDI/I<span>/congestion
*/
int congestion_devstate;
#if defined(THRESHOLD_DEVSTATE_PLACEHOLDER)
/*! \todo An ISDN span threshold device state could be useful in determining how often a span utilization goes over a configurable threshold. */
/*!
* \brief User threshold device state of the span.
* \details
* AST_DEVICE_NOT_INUSE - There are no B channels in use.
* AST_DEVICE_INUSE - The number of B channels in use is less than
* the configured threshold but not zero.
* AST_DEVICE_BUSY - The number of B channels in use meets or exceeds
* the configured threshold.
* AST_DEVICE_UNAVAILABLE - Span is in alarm.
* \note
* Device name: DAHDI/I<span>/threshold
*/
int threshold_devstate;
/*!
* \brief Number of B channels in use to consider the span in a busy state.
* \note Setting the threshold to zero is interpreted as all B channels.
*/
int user_busy_threshold;
#endif /* defined(THRESHOLD_DEVSTATE_PLACEHOLDER) */
};
void sig_pri_extract_called_num_subaddr(struct sig_pri_chan *p, const char *rdest, char *called, size_t called_buff_size);
int sig_pri_call(struct sig_pri_chan *p, struct ast_channel *ast, char *rdest, int timeout, int layer1);
int sig_pri_hangup(struct sig_pri_chan *p, struct ast_channel *ast);
int sig_pri_indicate(struct sig_pri_chan *p, struct ast_channel *chan, int condition, const void *data, size_t datalen);
int sig_pri_answer(struct sig_pri_chan *p, struct ast_channel *ast);
int sig_pri_available(struct sig_pri_chan *p);
void sig_pri_init_pri(struct sig_pri_pri *pri);
/* If return 0, it means this function was able to handle it (pre setup digits). If non zero, the user of this
* functions should handle it normally (generate inband DTMF) */
int sig_pri_digit_begin(struct sig_pri_chan *pvt, struct ast_channel *ast, char digit);
int sig_pri_start_pri(struct sig_pri_pri *pri);
void sig_pri_chan_alarm_notify(struct sig_pri_chan *p, int noalarm);
void pri_event_alarm(struct sig_pri_pri *pri, int index, int before_start_pri);
void pri_event_noalarm(struct sig_pri_pri *pri, int index, int before_start_pri);
struct ast_channel *sig_pri_request(struct sig_pri_chan *p, enum sig_pri_law law, const struct ast_channel *requestor, int transfercapability);
struct sig_pri_chan *sig_pri_chan_new(void *pvt_data, struct sig_pri_callback *callback, struct sig_pri_pri *pri, int logicalspan, int channo, int trunkgroup);
void sig_pri_chan_delete(struct sig_pri_chan *doomed);
int pri_is_up(struct sig_pri_pri *pri);
void sig_pri_cli_show_spans(int fd, int span, struct sig_pri_pri *pri);
void sig_pri_cli_show_span(int fd, int *dchannels, struct sig_pri_pri *pri);
int pri_send_keypad_facility_exec(struct sig_pri_chan *p, const char *digits);
int pri_send_callrerouting_facility_exec(struct sig_pri_chan *p, enum ast_channel_state chanstate, const char *destination, const char *original, const char *reason);
#if defined(HAVE_PRI_SERVICE_MESSAGES)
int pri_maintenance_bservice(struct pri *pri, struct sig_pri_chan *p, int changestatus);
#endif /* defined(HAVE_PRI_SERVICE_MESSAGES) */
void sig_pri_fixup(struct ast_channel *oldchan, struct ast_channel *newchan, struct sig_pri_chan *pchan);
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
int sig_pri_cc_agent_init(struct ast_cc_agent *agent, struct sig_pri_chan *pvt_chan);
int sig_pri_cc_agent_start_offer_timer(struct ast_cc_agent *agent);
int sig_pri_cc_agent_stop_offer_timer(struct ast_cc_agent *agent);
void sig_pri_cc_agent_req_ack(struct ast_cc_agent *agent);
int sig_pri_cc_agent_status_req(struct ast_cc_agent *agent);
int sig_pri_cc_agent_stop_ringing(struct ast_cc_agent *agent);
int sig_pri_cc_agent_party_b_free(struct ast_cc_agent *agent);
int sig_pri_cc_agent_start_monitoring(struct ast_cc_agent *agent);
int sig_pri_cc_agent_callee_available(struct ast_cc_agent *agent);
void sig_pri_cc_agent_destructor(struct ast_cc_agent *agent);
int sig_pri_cc_monitor_req_cc(struct ast_cc_monitor *monitor, int *available_timer_id);
int sig_pri_cc_monitor_suspend(struct ast_cc_monitor *monitor);
int sig_pri_cc_monitor_unsuspend(struct ast_cc_monitor *monitor);
int sig_pri_cc_monitor_status_rsp(struct ast_cc_monitor *monitor, enum ast_device_state devstate);
int sig_pri_cc_monitor_cancel_available_timer(struct ast_cc_monitor *monitor, int *sched_id);
void sig_pri_cc_monitor_destructor(void *monitor_pvt);
int sig_pri_load(const char *cc_type_name);
void sig_pri_unload(void);
#endif /* _SIG_PRI_H */