Files
asterisk/include/asterisk/callerid.h

627 lines
23 KiB
C
Raw Normal View History

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, 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 CallerID (and other GR30) management and generation
* Includes code and algorithms from the Zapata library.
*
* \ref CID
*
*/
/*!
* \page CID Caller ID names and numbers
*
* Caller ID names are currently 8 bit characters, propably
* ISO8859-1, depending on what your channel drivers handle.
*
* IAX2 and SIP caller ID names are UTF8
* On ISDN Caller ID names are 7 bit, Almost ASCII
* (See http://www.zytrax.com/tech/ia5.html )
*
* \note Asterisk does not currently support SIP utf8 caller ID names or caller ID's.
*
* \par See also
* \arg \ref callerid.c
* \arg \ref callerid.h
* \arg \ref Def_CallerPres
*/
#ifndef _ASTERISK_CALLERID_H
#define _ASTERISK_CALLERID_H
#include "asterisk/format.h"
#define MAX_CALLERID_SIZE 32000
#define CID_PRIVATE_NAME (1 << 0)
#define CID_PRIVATE_NUMBER (1 << 1)
#define CID_UNKNOWN_NAME (1 << 2)
#define CID_UNKNOWN_NUMBER (1 << 3)
#define CID_MSGWAITING (1 << 4)
#define CID_NOMSGWAITING (1 << 5)
chan_dahdi: Fix buggy and missing Caller ID parameters There are several things wrong with analog Caller ID handling that are fixed by this commit: callerid.c's Caller ID generation function contains the logic to use the presentation to properly send the proper Caller ID. However, currently, DAHDI does not pass any presentation information to the Caller ID module, which means that presentation is completely ignored on all calls. This means that lines could be getting Caller ID information they aren't supposed to. Part of the reason this has been obscured is because the simple switch logic for handling the built in *67 and *82 is completely wrong. Rather than modifying the presentation for the call accordingly (which is what it's supposed to do), it simply blanks out the Caller ID or fills it in. This is wrong, so wrong that it makes a mockery of the specification. Additionally, it would leave to the "UNAVAILABLE" disposition being used for Caller ID generation as opposed to the "PRIVATE" disposition that it should have been using. This is now fixed to only update the presentation and not modify the number and name, so that the simple switch *67/*82 work correctly. Next, sig_analog currently only copies over the name and number, nothing else, when it is filling in a duplicated caller id structure. Thus, we also now copy over the presentation information so that is available for the Caller ID spill. Additionally, this meant that "valid" was implicitly 0, and as such presentation would always fail to "Unavailable". The validity is therefore also copied over so it can be used by ast_party_id_presentation. As part of this fix, new API is added so that all the relevant Caller ID information can be passed in to the Caller ID generation functions. Parameters that are also completely missing from the Caller ID spill have also been added, to enhance the compatibility, correctness, and completeness of the Asterisk Caller ID implementation. ASTERISK-29991 #close Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15
2022-03-29 01:35:43 +00:00
#define CID_QUALIFIER (1 << 6)
#define CID_SIG_BELL 1
#define CID_SIG_V23 2
#define CID_SIG_DTMF 3
#define CID_SIG_V23_JP 4
#define CID_SIG_SMDI 5
#define CID_START_RING 1
#define CID_START_POLARITY 2
#define CID_START_POLARITY_IN 3
#define CID_START_DTMF_NOALERT 4
chan_dahdi: Fix buggy and missing Caller ID parameters There are several things wrong with analog Caller ID handling that are fixed by this commit: callerid.c's Caller ID generation function contains the logic to use the presentation to properly send the proper Caller ID. However, currently, DAHDI does not pass any presentation information to the Caller ID module, which means that presentation is completely ignored on all calls. This means that lines could be getting Caller ID information they aren't supposed to. Part of the reason this has been obscured is because the simple switch logic for handling the built in *67 and *82 is completely wrong. Rather than modifying the presentation for the call accordingly (which is what it's supposed to do), it simply blanks out the Caller ID or fills it in. This is wrong, so wrong that it makes a mockery of the specification. Additionally, it would leave to the "UNAVAILABLE" disposition being used for Caller ID generation as opposed to the "PRIVATE" disposition that it should have been using. This is now fixed to only update the presentation and not modify the number and name, so that the simple switch *67/*82 work correctly. Next, sig_analog currently only copies over the name and number, nothing else, when it is filling in a duplicated caller id structure. Thus, we also now copy over the presentation information so that is available for the Caller ID spill. Additionally, this meant that "valid" was implicitly 0, and as such presentation would always fail to "Unavailable". The validity is therefore also copied over so it can be used by ast_party_id_presentation. As part of this fix, new API is added so that all the relevant Caller ID information can be passed in to the Caller ID generation functions. Parameters that are also completely missing from the Caller ID spill have also been added, to enhance the compatibility, correctness, and completeness of the Asterisk Caller ID implementation. ASTERISK-29991 #close Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15
2022-03-29 01:35:43 +00:00
/* Caller ID message formats */
/*! SDMF - number only */
#define CID_TYPE_SDMF 0x00
/*! MDMF - name, number, etc. */
#define CID_TYPE_MDMF 0x01
/* defines dealing with message waiting indication generation */
/*! MWI SDMF format */
#define CID_MWI_TYPE_SDMF 0x00
/*! MWI MDMF format -- generate only MWI field */
#define CID_MWI_TYPE_MDMF 0x01
/*! MWI MDMF format -- generate name, callerid, date and MWI fields */
#define CID_MWI_TYPE_MDMF_FULL 0x02
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
#define AST_LIN2X(a) ((ast_format_cmp(codec, ast_format_alaw) == AST_FORMAT_CMP_EQUAL) ? (AST_LIN2A(a)) : (AST_LIN2MU(a)))
#define AST_XLAW(a) ((ast_format_cmp(codec, ast_format_alaw) == AST_FORMAT_CMP_EQUAL) ? (AST_ALAW(a)) : (AST_MULAW(a)))
struct callerid_state;
typedef struct callerid_state CIDSTATE;
/*! \brief CallerID Initialization
* \par
* Initializes the callerid system. Mostly stuff for inverse FFT
*/
void callerid_init(void);
/*! \brief Generates a CallerID FSK stream in ulaw format suitable for transmission.
* \param buf Buffer to use. If "buf" is supplied, it will use that buffer instead of allocating its own.
* "buf" must be at least 32000 bytes in size of you want to be sure you don't have an overrun.
* \param number Use NULL for no number or "P" for "private"
* \param name name to be used
* \param flags passed flags
* \param callwaiting callwaiting flag
* \param codec -- either AST_FORMAT_ULAW or AST_FORMAT_ALAW
* \details
* This function creates a stream of callerid (a callerid spill) data in ulaw format.
* \return It returns the size
* (in bytes) of the data (if it returns a size of 0, there is probably an error)
*/
chan_dahdi: Fix buggy and missing Caller ID parameters There are several things wrong with analog Caller ID handling that are fixed by this commit: callerid.c's Caller ID generation function contains the logic to use the presentation to properly send the proper Caller ID. However, currently, DAHDI does not pass any presentation information to the Caller ID module, which means that presentation is completely ignored on all calls. This means that lines could be getting Caller ID information they aren't supposed to. Part of the reason this has been obscured is because the simple switch logic for handling the built in *67 and *82 is completely wrong. Rather than modifying the presentation for the call accordingly (which is what it's supposed to do), it simply blanks out the Caller ID or fills it in. This is wrong, so wrong that it makes a mockery of the specification. Additionally, it would leave to the "UNAVAILABLE" disposition being used for Caller ID generation as opposed to the "PRIVATE" disposition that it should have been using. This is now fixed to only update the presentation and not modify the number and name, so that the simple switch *67/*82 work correctly. Next, sig_analog currently only copies over the name and number, nothing else, when it is filling in a duplicated caller id structure. Thus, we also now copy over the presentation information so that is available for the Caller ID spill. Additionally, this meant that "valid" was implicitly 0, and as such presentation would always fail to "Unavailable". The validity is therefore also copied over so it can be used by ast_party_id_presentation. As part of this fix, new API is added so that all the relevant Caller ID information can be passed in to the Caller ID generation functions. Parameters that are also completely missing from the Caller ID spill have also been added, to enhance the compatibility, correctness, and completeness of the Asterisk Caller ID implementation. ASTERISK-29991 #close Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15
2022-03-29 01:35:43 +00:00
int callerid_generate(unsigned char *buf, const char *number, const char *name, int flags, int callwaiting, struct ast_format *codec);
/*! \brief Generates a CallerID FSK stream in ulaw format suitable for transmission.
* \param buf Buffer to use. If "buf" is supplied, it will use that buffer instead of allocating its own.
* "buf" must be at least 32000 bytes in size of you want to be sure you don't have an overrun.
* \param number Use NULL for no number or "P" for "private"
* \param name name to be used
* \param ddn Dialable Directory Number (or NULL)
* \param redirecting Redirecting reason
* \param flags passed flags
* \param format Message format
* \param callwaiting callwaiting flag
* \param codec -- either AST_FORMAT_ULAW or AST_FORMAT_ALAW
* \details
* This function creates a stream of callerid (a callerid spill) data in ulaw format.
* \return It returns the size
* (in bytes) of the data (if it returns a size of 0, there is probably an error)
*/
int callerid_full_generate(unsigned char *buf, const char *number, const char *name, const char *ddn, int redirecting,
int flags, int format, int callwaiting, struct ast_format *codec);
/*! \brief Generates a CallerID FSK stream in ulaw format suitable for transmission.
* \param buf Buffer to use. If "buf" is supplied, it will use that buffer instead of allocating its own.
* "buf" must be at least 32000 bytes in size of you want to be sure you don't have an overrun.
* \param number Use NULL for no number or "P" for "private"
* \param name name to be used
* \param ddn Dialable Directory Number (or NULL)
* \param redirecting Redirecting reason
* \param flags passed flags
* \param format Message format
* \param callwaiting callwaiting flag
* \param codec -- either AST_FORMAT_ULAW or AST_FORMAT_ALAW
* \param tz TZ-format time zone to use for date/time (NULL for system default)
* \details
* This function creates a stream of callerid (a callerid spill) data in ulaw format.
* \return It returns the size
* (in bytes) of the data (if it returns a size of 0, there is probably an error)
*/
int callerid_full_tz_generate(unsigned char *buf, const char *number, const char *name, const char *ddn, int redirecting,
int flags, int format, int callwaiting, struct ast_format *codec, const char *tz);
/*! \brief Create a callerID state machine
* \param cid_signalling Type of signalling in use
*
* \details
* This function returns a malloc'd instance of the callerid_state data structure.
* \return Returns a pointer to a malloc'd callerid_state structure, or NULL on error.
*/
struct callerid_state *callerid_new(int cid_signalling);
/*! \brief Read samples into the state machine.
* \param cid Which state machine to act upon
* \param ubuf containing your samples
* \param samples number of samples contained within the buffer.
* \param codec which codec (AST_FORMAT_ALAW or AST_FORMAT_ULAW)
*
* \details
* Send received audio to the Caller*ID demodulator.
* \retval -1 on error
* \retval 0 for "needs more samples"
* \retval 1 if the CallerID spill reception is complete.
*/
int callerid_feed(struct callerid_state *cid, unsigned char *ubuf, int samples, struct ast_format *codec);
/*! \brief Read samples into the state machine.
* \param cid Which state machine to act upon
* \param ubuf containing your samples
* \param samples number of samples contained within the buffer.
* \param codec which codec (AST_FORMAT_ALAW or AST_FORMAT_ULAW)
*
* \details
* Send received audio to the Caller*ID demodulator (for japanese style lines).
* \retval -1 on error
* \retval 0 for "needs more samples"
* \retval 1 if the CallerID spill reception is complete.
*/
int callerid_feed_jp(struct callerid_state *cid, unsigned char *ubuf, int samples, struct ast_format *codec);
/*! \brief Extract info out of callerID state machine. Flags are listed above
* \param cid Callerid state machine to act upon
* \param number Pass the address of a pointer-to-char (will contain the phone number)
* \param name Pass the address of a pointer-to-char (will contain the name)
* \param flags Pass the address of an int variable (will contain the various callerid flags - presentation flags and call qualifier)
*
* \details
* This function extracts a callerid string out of a callerid_state state machine.
* If no number is found, *number will be set to NULL. Likewise for the name.
* Flags can contain any of the following: CID_PRIVATE_NAME, CID_PRIVATE_NUMBER, CID_UNKNOWN_NAME, CID_UNKNOWN_NUMBER, CID_MSGWAITING, CID_NOMSGWAITING, CID_QUALIFIER
*/
void callerid_get(struct callerid_state *cid, char **number, char **name, int *flags);
/*! \brief Extract info out of callerID state machine. Flags are listed above
* \param cid Callerid state machine to act upon
* \param[out] number Pass the address of a pointer-to-char (will contain the phone number)
* \param[out] name Pass the address of a pointer-to-char (will contain the name)
* \param[out] flags Pass the address of an int variable (will contain the various callerid flags)
* \param[out] redirecting Pass the address of an int variable (will contain the redirecting reason, if received - presentation flags and call qualifier)
*
* \details
* This function extracts a callerid string out of a callerid_state state machine.
* If no number is found, *number will be set to NULL. Likewise for the name.
* Flags can contain any of the following: CID_PRIVATE_NAME, CID_PRIVATE_NUMBER, CID_UNKNOWN_NAME, CID_UNKNOWN_NUMBER, CID_MSGWAITING, CID_NOMSGWAITING, CID_QUALIFIER
*/
void callerid_get_with_redirecting(struct callerid_state *cid, char **name, char **number, int *flags, int *redirecting);
/*!
* \brief Get and parse DTMF-based callerid
* \param cidstring The actual transmitted string.
* \param number The cid number is returned here.
* \param flags The cid flags are returned here.
*/
void callerid_get_dtmf(char *cidstring, char *number, int *flags);
/*! \brief This function frees callerid_state cid.
* \param cid This is the callerid_state state machine to free
*/
void callerid_free(struct callerid_state *cid);
/*! \brief Generate Caller-ID spill from the "callerid" field of asterisk (in e-mail address like format)
* \param buf buffer for output samples. See callerid_generate() for details regarding buffer.
* \param name Caller-ID Name
* \param number Caller-ID Number
* \param codec Asterisk codec (either AST_FORMAT_ALAW or AST_FORMAT_ULAW)
*
* \details
* Acts like callerid_generate except uses an asterisk format callerid string.
*/
int ast_callerid_generate(unsigned char *buf, const char *name, const char *number, struct ast_format *codec);
chan_dahdi: Fix buggy and missing Caller ID parameters There are several things wrong with analog Caller ID handling that are fixed by this commit: callerid.c's Caller ID generation function contains the logic to use the presentation to properly send the proper Caller ID. However, currently, DAHDI does not pass any presentation information to the Caller ID module, which means that presentation is completely ignored on all calls. This means that lines could be getting Caller ID information they aren't supposed to. Part of the reason this has been obscured is because the simple switch logic for handling the built in *67 and *82 is completely wrong. Rather than modifying the presentation for the call accordingly (which is what it's supposed to do), it simply blanks out the Caller ID or fills it in. This is wrong, so wrong that it makes a mockery of the specification. Additionally, it would leave to the "UNAVAILABLE" disposition being used for Caller ID generation as opposed to the "PRIVATE" disposition that it should have been using. This is now fixed to only update the presentation and not modify the number and name, so that the simple switch *67/*82 work correctly. Next, sig_analog currently only copies over the name and number, nothing else, when it is filling in a duplicated caller id structure. Thus, we also now copy over the presentation information so that is available for the Caller ID spill. Additionally, this meant that "valid" was implicitly 0, and as such presentation would always fail to "Unavailable". The validity is therefore also copied over so it can be used by ast_party_id_presentation. As part of this fix, new API is added so that all the relevant Caller ID information can be passed in to the Caller ID generation functions. Parameters that are also completely missing from the Caller ID spill have also been added, to enhance the compatibility, correctness, and completeness of the Asterisk Caller ID implementation. ASTERISK-29991 #close Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15
2022-03-29 01:35:43 +00:00
/*! \brief Generate Caller-ID spill from the "callerid" field of asterisk (in e-mail address like format)
* \param buf buffer for output samples. See callerid_generate() for details regarding buffer.
* \param name Caller-ID Name
* \param number Caller-ID Number
* \param ddn Dialable Directory Number (or NULL)
* \param redirecting Redirecting Reason (-1 if N/A)
* \param pres Presentation (0 for default)
* \param qualifier Call Qualifier (0 for no, 1 for yes)
* \param format Message Format
* \param codec Asterisk codec (either AST_FORMAT_ALAW or AST_FORMAT_ULAW)
*
* \details
* Like ast_callerid_generate but with additional parameters.
*/
int ast_callerid_full_generate(unsigned char *buf, const char *name, const char *number,
const char *ddn, int redirecting, int pres, int qualifier, int format, struct ast_format *codec);
/*! \brief Generate Caller-ID spill from the "callerid" field of asterisk (in e-mail address like format)
* \param buf buffer for output samples. See callerid_generate() for details regarding buffer.
* \param name Caller-ID Name
* \param number Caller-ID Number
* \param ddn Dialable Directory Number (or NULL)
* \param redirecting Redirecting Reason (-1 if N/A)
* \param pres Presentation (0 for default)
* \param qualifier Call Qualifier (0 for no, 1 for yes)
* \param format Message Format
* \param codec Asterisk codec (either AST_FORMAT_ALAW or AST_FORMAT_ULAW)
* \param tz TZ-format time zone name to use for date/time (NULL for system default)
*
* \details
* Like ast_callerid_generate but with additional parameters.
*/
int ast_callerid_full_tz_generate(unsigned char *buf, const char *name, const char *number,
const char *ddn, int redirecting, int pres, int qualifier, int format, struct ast_format *codec, const char *tz);
/*!
* \brief Generate message waiting indicator
* \param buf
* \param active The message indicator state
* -- either 0 no messages in mailbox or 1 messages in mailbox
* \param type Format of message (any of CID_MWI_TYPE_*)
* \param codec
* \param name
* \param number
* \param flags
* \see callerid_generate() for more info as it uses the same encoding
* \version 1.6.1 changed mdmf parameter to type, added name, number and flags for caller id message generation
*/
int ast_callerid_vmwi_generate(unsigned char *buf, int active, int type, struct ast_format *codec, const char *name,
const char *number, int flags);
/*! \brief Generate Caller-ID spill but in a format suitable for Call Waiting(tm)'s Caller*ID(tm)
* \see ast_callerid_generate() for other details
*/
int ast_callerid_callwaiting_generate(unsigned char *buf, const char *name, const char *number, struct ast_format *codec);
chan_dahdi: Fix buggy and missing Caller ID parameters There are several things wrong with analog Caller ID handling that are fixed by this commit: callerid.c's Caller ID generation function contains the logic to use the presentation to properly send the proper Caller ID. However, currently, DAHDI does not pass any presentation information to the Caller ID module, which means that presentation is completely ignored on all calls. This means that lines could be getting Caller ID information they aren't supposed to. Part of the reason this has been obscured is because the simple switch logic for handling the built in *67 and *82 is completely wrong. Rather than modifying the presentation for the call accordingly (which is what it's supposed to do), it simply blanks out the Caller ID or fills it in. This is wrong, so wrong that it makes a mockery of the specification. Additionally, it would leave to the "UNAVAILABLE" disposition being used for Caller ID generation as opposed to the "PRIVATE" disposition that it should have been using. This is now fixed to only update the presentation and not modify the number and name, so that the simple switch *67/*82 work correctly. Next, sig_analog currently only copies over the name and number, nothing else, when it is filling in a duplicated caller id structure. Thus, we also now copy over the presentation information so that is available for the Caller ID spill. Additionally, this meant that "valid" was implicitly 0, and as such presentation would always fail to "Unavailable". The validity is therefore also copied over so it can be used by ast_party_id_presentation. As part of this fix, new API is added so that all the relevant Caller ID information can be passed in to the Caller ID generation functions. Parameters that are also completely missing from the Caller ID spill have also been added, to enhance the compatibility, correctness, and completeness of the Asterisk Caller ID implementation. ASTERISK-29991 #close Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15
2022-03-29 01:35:43 +00:00
/*! \brief Generate Caller-ID spill but in a format suitable for Call Waiting(tm)'s Caller*ID(tm)
* \see ast_callerid_generate() for other details
*/
int ast_callerid_callwaiting_full_generate(unsigned char *buf, const char *name, const char *number,
const char *ddn, int redirecting, int pres, int qualifier, struct ast_format *codec);
/*! \brief Generate Caller-ID spill but in a format suitable for Call Waiting(tm)'s Caller*ID(tm)
* \param tz TZ-format time zone for date/time (NULL for system default)
* \see ast_callerid_generate() for other details
*/
int ast_callerid_callwaiting_full_tz_generate(unsigned char *buf, const char *name, const char *number,
const char *ddn, int redirecting, int pres, int qualifier, struct ast_format *codec, const char *tz);
/*! \brief Destructively parse inbuf into name and location (or number)
* \details
* Parses callerid stream from inbuf and changes into useable form, outputted in name and location.
* \param instr buffer of callerid stream (in audio form) to be parsed. Warning, data in buffer is changed.
* \param name address of a pointer-to-char for the name value of the stream.
* \param location address of a pointer-to-char for the phone number value of the stream.
* \note XXX 'name' is not parsed consistently e.g. we have
* input location name
* " foo bar " <123> 123 ' foo bar ' (with spaces around)
* " foo bar " NULL 'foo bar' (without spaces around)
* The parsing of leading and trailing space/quotes should be more consistent.
* \retval 0 on success
* \retval -1 on failure
*/
int ast_callerid_parse(char *instr, char **name, char **location);
/*!
* \brief Generate a CAS (CPE Alert Signal) tone for 'n' samples
* \param outbuf Allocated buffer for data. Must be at least 2400 bytes unless no SAS is desired
* \param sas Non-zero if CAS should be preceeded by SAS
* \param len How many samples to generate.
* \param codec Which codec (AST_FORMAT_ALAW or AST_FORMAT_ULAW)
* \retval -1 on error (if len is less than 2400)
* \retval 0 on success
*/
int ast_gen_cas(unsigned char *outbuf, int sas, int len, struct ast_format *codec);
/*!
* \brief Shrink a phone number in place to just digits (more accurately it just removes ()'s, .'s, and -'s...
* \param n The number to be stripped/shrunk
*/
void ast_shrink_phone_number(char *n);
/*!
* \brief Check if a string consists only of digits and + \#
* \param n number to be checked.
* \retval 0 if \p n is a number
* \retval 1 if not
*/
int ast_isphonenumber(const char *n);
/*!
* \brief Check if a string consists only of digits and + \# ( ) - .
* (meaning it can be cleaned with ast_shrink_phone_number)
* \param exten The extension (or URI) to be checked.
* \retval 1 if \p exten is valid AST shrinkable phone number
* \retval 0 if not
*/
int ast_is_shrinkable_phonenumber(const char *exten);
int ast_callerid_split(const char *src, char *name, int namelen, char *num, int numlen);
char *ast_callerid_merge(char *buf, int bufsiz, const char *name, const char *num, const char *unknown);
/*
* Caller*ID and other GR-30 compatible generation
* routines (used by ADSI for example)
*/
extern float cid_dr[4];
extern float cid_di[4];
extern float clidsb;
static inline float callerid_getcarrier(float *cr, float *ci, int bit)
{
/* Move along. There's nothing to see here... */
float t;
t = *cr * cid_dr[bit] - *ci * cid_di[bit];
*ci = *cr * cid_di[bit] + *ci * cid_dr[bit];
*cr = t;
t = 2.0 - (*cr * *cr + *ci * *ci);
*cr *= t;
*ci *= t;
return *cr;
}
#define PUT_BYTE(a) do { \
*(buf++) = (a); \
bytes++; \
} while(0)
#define PUT_AUDIO_SAMPLE(y) do { \
int __sample_idx = (short)(rint(8192.0 * (y))); \
*(buf++) = AST_LIN2X(__sample_idx); \
bytes++; \
} while(0)
#define PUT_CLID_MARKMS do { \
int __clid_x; \
for (__clid_x=0;__clid_x<8;__clid_x++) \
PUT_AUDIO_SAMPLE(callerid_getcarrier(&cr, &ci, 1)); \
} while(0)
#define PUT_CLID_BAUD(bit) do { \
while(scont < clidsb) { \
PUT_AUDIO_SAMPLE(callerid_getcarrier(&cr, &ci, bit)); \
scont += 1.0; \
} \
scont -= clidsb; \
} while(0)
#define PUT_CLID(byte) do { \
int z; \
unsigned char b = (byte); \
PUT_CLID_BAUD(0); /* Start bit */ \
for (z=0;z<8;z++) { \
PUT_CLID_BAUD(b & 1); \
b >>= 1; \
} \
PUT_CLID_BAUD(1); /* Stop bit */ \
} while(0)
/* Various defines and bits for handling PRI- and SS7-type restriction */
#define AST_PRES_NUMBER_TYPE 0x03
#define AST_PRES_USER_NUMBER_UNSCREENED 0x00
#define AST_PRES_USER_NUMBER_PASSED_SCREEN 0x01
#define AST_PRES_USER_NUMBER_FAILED_SCREEN 0x02
#define AST_PRES_NETWORK_NUMBER 0x03
#define AST_PRES_RESTRICTION 0x60
#define AST_PRES_ALLOWED 0x00
#define AST_PRES_RESTRICTED 0x20
#define AST_PRES_UNAVAILABLE 0x40
#define AST_PRES_RESERVED 0x60
#define AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED \
(AST_PRES_ALLOWED | AST_PRES_USER_NUMBER_UNSCREENED)
#define AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN \
(AST_PRES_ALLOWED | AST_PRES_USER_NUMBER_PASSED_SCREEN)
#define AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN \
(AST_PRES_ALLOWED | AST_PRES_USER_NUMBER_FAILED_SCREEN)
#define AST_PRES_ALLOWED_NETWORK_NUMBER \
(AST_PRES_ALLOWED | AST_PRES_NETWORK_NUMBER)
#define AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED \
(AST_PRES_RESTRICTED | AST_PRES_USER_NUMBER_UNSCREENED)
#define AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN \
(AST_PRES_RESTRICTED | AST_PRES_USER_NUMBER_PASSED_SCREEN)
#define AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN \
(AST_PRES_RESTRICTED | AST_PRES_USER_NUMBER_FAILED_SCREEN)
#define AST_PRES_PROHIB_NETWORK_NUMBER \
(AST_PRES_RESTRICTED | AST_PRES_NETWORK_NUMBER)
#define AST_PRES_NUMBER_NOT_AVAILABLE \
(AST_PRES_UNAVAILABLE | AST_PRES_NETWORK_NUMBER)
int ast_parse_caller_presentation(const char *data);
const char *ast_describe_caller_presentation(int data);
const char *ast_named_caller_presentation(int data);
/*!
* \page Def_CallerPres Caller ID Presentation
*
* Caller ID presentation values are used to set properties to a
* caller ID in PSTN networks, and as RPID value in SIP transactions.
*
* The following values are available to use:
* \arg \b Defined value, text string in config file, explanation
*
* \arg \b AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED, "allowed_not_screened", Presentation Allowed, Not Screened,
* \arg \b AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN, "allowed_passed_screen", Presentation Allowed, Passed Screen,
* \arg \b AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN, "allowed_failed_screen", Presentation Allowed, Failed Screen,
* \arg \b AST_PRES_ALLOWED_NETWORK_NUMBER, "allowed", Presentation Allowed, Network Number,
* \arg \b AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED, "prohib_not_screened", Presentation Prohibited, Not Screened,
* \arg \b AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN, "prohib_passed_screen", Presentation Prohibited, Passed Screen,
* \arg \b AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN, "prohib_failed_screen", Presentation Prohibited, Failed Screen,
* \arg \b AST_PRES_PROHIB_NETWORK_NUMBER, "prohib", Presentation Prohibited, Network Number,
*
* \par References
* \arg \ref callerid.h Definitions
* \arg \ref callerid.c Functions
* \arg \ref CID Caller ID names and numbers
*/
/*!
* \brief redirecting reason codes.
*
* This list attempts to encompass redirecting reasons
* as defined by several channel technologies.
*/
enum AST_REDIRECTING_REASON {
AST_REDIRECTING_REASON_UNKNOWN,
AST_REDIRECTING_REASON_USER_BUSY,
AST_REDIRECTING_REASON_NO_ANSWER,
AST_REDIRECTING_REASON_UNAVAILABLE,
AST_REDIRECTING_REASON_UNCONDITIONAL,
AST_REDIRECTING_REASON_TIME_OF_DAY,
AST_REDIRECTING_REASON_DO_NOT_DISTURB,
AST_REDIRECTING_REASON_DEFLECTION,
AST_REDIRECTING_REASON_FOLLOW_ME,
AST_REDIRECTING_REASON_OUT_OF_ORDER,
AST_REDIRECTING_REASON_AWAY,
AST_REDIRECTING_REASON_CALL_FWD_DTE, /* This is something defined in Q.931, and no I don't know what it means */
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
AST_REDIRECTING_REASON_SEND_TO_VM,
};
/*!
* \since 1.8
* \brief Convert redirecting reason text code to value (used in config file parsing)
*
* \param data text string from config file
*
* \retval Q931_REDIRECTING_REASON from callerid.h
* \retval -1 if not in table
*/
int ast_redirecting_reason_parse(const char *data);
/*!
* \since 1.8
* \brief Convert redirecting reason value to explanatory string
*
* \param data Q931_REDIRECTING_REASON from callerid.h
*
* \return string for human presentation
*/
const char *ast_redirecting_reason_describe(int data);
struct ast_party_redirecting_reason;
/*!
* \since 1.8
* \brief Convert redirecting reason value to text code
*
* \param data ast_party_redirecting_reason structure from channel.h
*
* \return string for config file
*/
const char *ast_redirecting_reason_name(const struct ast_party_redirecting_reason *data);
/*!
* \brief Connected line update source code
*/
enum AST_CONNECTED_LINE_UPDATE_SOURCE {
/*! Update for unknown reason (May be interpreted to mean from answer) */
AST_CONNECTED_LINE_UPDATE_SOURCE_UNKNOWN,
/*! Update from normal call answering */
AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER,
/*! Update from call diversion (Deprecated, use REDIRECTING updates instead.) */
AST_CONNECTED_LINE_UPDATE_SOURCE_DIVERSION,
/*! Update from call transfer(active) (Party has already answered) */
AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER,
/*! Update from call transfer(alerting) (Party has not answered yet) */
AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER_ALERTING
};
/*!
* \since 1.8
* \brief Convert connected line update source text code to value (used in config file parsing)
*
* \param data text string from config file
*
* \retval AST_CONNECTED_LINE_UPDATE_SOURCE from callerid.h
* \retval -1 if not in table
*/
int ast_connected_line_source_parse(const char *data);
/*!
* \since 1.8
* \brief Convert connected line update source value to explanatory string
*
* \param data AST_CONNECTED_LINE_UPDATE_SOURCE from callerid.h
*
* \return string for human presentation
*/
const char *ast_connected_line_source_describe(int data);
/*!
* \since 1.8
* \brief Convert connected line update source value to text code
*
* \param data AST_CONNECTED_LINE_UPDATE_SOURCE from callerid.h
*
* \return string for config file
*/
const char *ast_connected_line_source_name(int data);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
/*!
* \since 1.8
* \brief Convert ast_party_name.char_set text code to value (used in config file parsing)
*
* \param data text string from config file
*
* \retval AST_PARTY_CHAR_SET from channel.h
* \retval -1 if not in table
*/
int ast_party_name_charset_parse(const char *data);
/*!
* \since 1.8
* \brief Convert ast_party_name.char_set value to explanatory string
*
* \param data AST_PARTY_CHAR_SET from channel.h
*
* \return string for human presentation
*/
const char *ast_party_name_charset_describe(int data);
/*!
* \since 1.8
* \brief Convert ast_party_name.char_set value to text code
*
* \param data AST_PARTY_CHAR_SET from channel.h
*
* \return string for config file
*/
const char *ast_party_name_charset_str(int data);
#endif /* _ASTERISK_CALLERID_H */