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	https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
		
			
				
	
	
		
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			278 lines
		
	
	
		
			9.3 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * Asterisk -- An open source telephony toolkit.
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|  *
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|  * Copyright (C) 2010 Digium, Inc.
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|  *
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|  * Richard Mudgett <rmudgett@digium.com>
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|  *
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|  * See http://www.asterisk.org for more information about
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|  * the Asterisk project. Please do not directly contact
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|  * any of the maintainers of this project for assistance;
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|  * the project provides a web site, mailing lists and IRC
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|  * channels for your use.
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|  *
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|  * This program is free software, distributed under the terms of
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|  * the GNU General Public License Version 2. See the LICENSE file
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|  * at the top of the source tree.
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|  */
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| 
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| /*!
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|  * \file
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|  * \brief Interface header for SS7 signaling module.
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|  *
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|  * \author Matthew Fredrickson <creslin@digium.com>
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|  * \author Richard Mudgett <rmudgett@digium.com>
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|  *
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|  * See Also:
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|  * \arg \ref AstCREDITS
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|  */
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| 
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| #ifndef _ASTERISK_SIG_SS7_H
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| #define _ASTERISK_SIG_SS7_H
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| 
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| #include "asterisk/channel.h"
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| #include <libss7.h>
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| 
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| #ifdef __cplusplus
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| extern "C" {
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| #endif
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| 
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| /* ------------------------------------------------------------------- */
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| 
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| /*! SS7 debug message flags when SS7 debugging is turned on at the command line. */
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| #define SIG_SS7_DEBUG	\
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| 	(SS7_DEBUG_MTP2 | SS7_DEBUG_MTP3 | SS7_DEBUG_ISUP)
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| 
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| #if 0
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| /*! SS7 debug message flags set on initial startup. */
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| #define SIG_SS7_DEBUG_DEFAULT	SIG_SS7_DEBUG
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| #else
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| /*! SS7 debug message flags set on initial startup. */
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| #define SIG_SS7_DEBUG_DEFAULT	0
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| #endif
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| 
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| /* ------------------------------------------------------------------- */
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| 
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| #define SIG_SS7_NUM_DCHANS		4		/*!< No more than 4 d-channels */
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| #define SIG_SS7_MAX_CHANNELS	672		/*!< No more than a DS3 per trunk group */
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| 
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| #define SIG_SS7		(0x1000000 | DAHDI_SIG_CLEAR)
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| 
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| #define LINKSTATE_INALARM	(1 << 0)
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| #define LINKSTATE_STARTING	(1 << 1)
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| #define LINKSTATE_UP		(1 << 2)
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| #define LINKSTATE_DOWN		(1 << 3)
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| 
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| #define SS7_NAI_DYNAMIC		-1
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| 
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| #define LINKSET_FLAG_EXPLICITACM (1 << 0)
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| 
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| 
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| enum sig_ss7_tone {
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| 	SIG_SS7_TONE_RINGTONE = 0,
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| 	SIG_SS7_TONE_STUTTER,
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| 	SIG_SS7_TONE_CONGESTION,
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| 	SIG_SS7_TONE_DIALTONE,
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| 	SIG_SS7_TONE_DIALRECALL,
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| 	SIG_SS7_TONE_INFO,
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| 	SIG_SS7_TONE_BUSY,
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| };
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| 
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| enum sig_ss7_law {
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| 	SIG_SS7_DEFLAW = 0,
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| 	SIG_SS7_ULAW,
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| 	SIG_SS7_ALAW
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| };
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| 
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| /*! Call establishment life cycle level for simple comparisons. */
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| enum sig_ss7_call_level {
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| 	/*! Call does not exist. */
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| 	SIG_SS7_CALL_LEVEL_IDLE,
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| 	/*! Call is present but has no response yet. (SETUP) */
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| 	SIG_SS7_CALL_LEVEL_SETUP,
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| 	/*! Call routing is happening. (PROCEEDING) */
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| 	SIG_SS7_CALL_LEVEL_PROCEEDING,
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| 	/*! Called party is being alerted of the call. (ALERTING) */
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| 	SIG_SS7_CALL_LEVEL_ALERTING,
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| 	/*! Call is connected/answered. (CONNECT) */
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| 	SIG_SS7_CALL_LEVEL_CONNECT,
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| };
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| 
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| struct sig_ss7_linkset;
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| 
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| struct sig_ss7_callback {
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| 	/* Unlock the private in the signaling private structure. */
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| 	void (* const unlock_private)(void *pvt);
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| 	/* Lock the private in the signaling private structure. */
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| 	void (* const lock_private)(void *pvt);
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| 
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| 	int (* const set_echocanceller)(void *pvt, int enable);
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| 	void (* const set_loopback)(void *pvt, int enable);
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| 
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| 	struct ast_channel * (* const new_ast_channel)(void *pvt, int state, enum sig_ss7_law law, char *exten, const struct ast_channel *requestor);
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| 	int (* const play_tone)(void *pvt, enum sig_ss7_tone tone);
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| 
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| 	void (* const handle_link_exception)(struct sig_ss7_linkset *linkset, int which);
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| 	void (* const set_alarm)(void *pvt, int in_alarm);
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| 	void (* const set_dialing)(void *pvt, int is_dialing);
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| 	void (* const set_digital)(void *pvt, int is_digital);
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| 	void (* const set_inservice)(void *pvt, int is_inservice);
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| 	void (* const set_locallyblocked)(void *pvt, int is_blocked);
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| 	void (* const set_remotelyblocked)(void *pvt, int is_blocked);
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| 	void (* const set_callerid)(void *pvt, const struct ast_party_caller *caller);
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| 	void (* const set_dnid)(void *pvt, const char *dnid);
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| 
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| 	void (* const queue_control)(void *pvt, int subclass);
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| };
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| 
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| struct sig_ss7_chan {
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| 	struct sig_ss7_callback *calls;
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| 	void *chan_pvt;					/*!< Private structure of the user of this module. */
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| 	struct sig_ss7_linkset *ss7;
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| 	struct ast_channel *owner;
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| 
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| 	/*! \brief Opaque libss7 call control structure */
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| 	struct isup_call *ss7call;
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| 
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| 	/*! Call establishment life cycle level for simple comparisons. */
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| 	enum sig_ss7_call_level call_level;
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| 
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| 	int channel;					/*!< Channel Number */
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| 	int cic;						/*!< CIC associated with channel */
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| 	unsigned int dpc;				/*!< CIC's DPC */
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| 
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| 	/* Options to be set by user */
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| 	/*!
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| 	 * \brief Number of most significant digits/characters to strip from the dialed number.
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| 	 * \note Feature is deprecated.  Use dialplan logic.
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| 	 */
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| 	int stripmsd;
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| 	/*!
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| 	 * \brief TRUE if the outgoing caller ID is blocked/hidden.
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| 	 */
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| 	unsigned int hidecallerid:1;
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| 	/*! \brief TRUE if caller ID is used on this channel. */
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| 	unsigned int use_callerid:1;
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| 	/*!
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| 	 * \brief TRUE if we will use the calling presentation setting
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| 	 * from the Asterisk channel for outgoing calls.
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| 	 */
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| 	unsigned int use_callingpres:1;
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| 	unsigned int immediate:1;		/*!< Answer before getting digits? */
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| 
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| 	/*! \brief TRUE if the channel is locally blocked.  Set by user and link. */
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| 	unsigned int locallyblocked:1;
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| 	/*! \brief TRUE if the channel is remotely blocked.  Set by user and link. */
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| 	unsigned int remotelyblocked:1;
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| 
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| 	char context[AST_MAX_CONTEXT];
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| 	char mohinterpret[MAX_MUSICCLASS];
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| 
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| 	/* Options to be checked by user */
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| 	int cid_ani2;					/*!< Automatic Number Identification number (Alternate PRI caller ID number) */
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| 	int cid_ton;					/*!< Type Of Number (TON) */
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| 	int callingpres;				/*!< The value of calling presentation that we're going to use when placing a PRI call */
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| 	char cid_num[AST_MAX_EXTENSION];
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| 	char cid_subaddr[AST_MAX_EXTENSION];/*!< XXX SS7 may not support. */
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| 	char cid_name[AST_MAX_EXTENSION];
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| 	char cid_ani[AST_MAX_EXTENSION];
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| 	char exten[AST_MAX_EXTENSION];
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| 
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| 	/* Options to be checked by user that are stuffed into channel variables. */
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| 	char charge_number[50];
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| 	char gen_add_number[50];
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| 	char gen_dig_number[50];
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| 	char orig_called_num[50];
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| 	char redirecting_num[50];
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| 	char generic_name[50];
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| 	unsigned char gen_add_num_plan;
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| 	unsigned char gen_add_nai;
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| 	unsigned char gen_add_pres_ind;
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| 	unsigned char gen_add_type;
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| 	unsigned char gen_dig_type;
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| 	unsigned char gen_dig_scheme;
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| 	char jip_number[50];
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| #if 0
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| 	unsigned char lspi_type;
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| 	unsigned char lspi_scheme;
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| 	unsigned char lspi_context;
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| #endif
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| 	char lspi_ident[50];
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| 	unsigned int call_ref_ident;
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| 	unsigned int call_ref_pc;
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| 	unsigned char calling_party_cat;
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| 
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| 	/*
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| 	 * Channel status bits.
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| 	 */
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| 	/*! TRUE if channel is associated with a link that is down. */
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| 	unsigned int inalarm:1;
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| 	/*! TRUE if this channel is being used for an outgoing call. */
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| 	unsigned int outgoing:1;
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| 	/*! \brief TRUE if the call has seen inband-information progress through the network. */
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| 	unsigned int progress:1;
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| 	/*! \brief TRUE if the call has already gone/hungup */
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| 	unsigned int alreadyhungup:1;
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| 	/*! \brief XXX BOOLEAN Purpose??? */
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| 	unsigned int rlt:1;
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| 	/*! TRUE if this channel is in loopback. */
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| 	unsigned int loopedback:1;
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| };
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| 
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| struct sig_ss7_linkset {
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| 	pthread_t master;					/*!< Thread of master */
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| 	ast_mutex_t lock;					/*!< libss7 access lock */
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| 	struct sig_ss7_callback *calls;
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| 	struct ss7 *ss7;
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| 	struct sig_ss7_chan *pvts[SIG_SS7_MAX_CHANNELS];/*!< Member channel pvt structs */
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| 	int fds[SIG_SS7_NUM_DCHANS];
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| 	int numsigchans;
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| 	int linkstate[SIG_SS7_NUM_DCHANS];
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| 	int numchans;
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| 	int span;							/*!< span number put into user output messages */
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| 	enum {
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| 		LINKSET_STATE_DOWN = 0,
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| 		LINKSET_STATE_UP
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| 	} state;
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| 
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| 	/* Options to be set by user */
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| 	int flags;							/*!< Linkset flags (LINKSET_FLAG_EXPLICITACM) */
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| 	int type;							/*!< SS7 type ITU/ANSI. Used for companding selection. */
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| 	char called_nai;					/*!< Called Nature of Address Indicator */
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| 	char calling_nai;					/*!< Calling Nature of Address Indicator */
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| 	char internationalprefix[10];		/*!< country access code ('00' for european dialplans) */
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| 	char nationalprefix[10];			/*!< area access code ('0' for european dialplans) */
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| 	char subscriberprefix[20];			/*!< area access code + area code ('0'+area code for european dialplans) */
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| 	char unknownprefix[20];				/*!< for unknown dialplans */
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| };
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| 
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| void sig_ss7_set_alarm(struct sig_ss7_chan *p, int in_alarm);
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| 
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| void *ss7_linkset(void *data);
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| 
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| void sig_ss7_link_alarm(struct sig_ss7_linkset *linkset, int which);
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| void sig_ss7_link_noalarm(struct sig_ss7_linkset *linkset, int which);
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| int sig_ss7_add_sigchan(struct sig_ss7_linkset *linkset, int which, int ss7type, int transport, int inalarm, int networkindicator, int pointcode, int adjpointcode);
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| 
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| int sig_ss7_available(struct sig_ss7_chan *p);
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| int sig_ss7_call(struct sig_ss7_chan *p, struct ast_channel *ast, char *rdest);
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| int sig_ss7_hangup(struct sig_ss7_chan *p, struct ast_channel *ast);
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| int sig_ss7_answer(struct sig_ss7_chan *p, struct ast_channel *ast);
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| void sig_ss7_fixup(struct ast_channel *oldchan, struct ast_channel *newchan, struct sig_ss7_chan *pchan);
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| int sig_ss7_indicate(struct sig_ss7_chan *p, struct ast_channel *chan, int condition, const void *data, size_t datalen);
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| struct ast_channel *sig_ss7_request(struct sig_ss7_chan *p, enum sig_ss7_law law, const struct ast_channel *requestor, int transfercapability);
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| void sig_ss7_chan_delete(struct sig_ss7_chan *doomed);
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| struct sig_ss7_chan *sig_ss7_chan_new(void *pvt_data, struct sig_ss7_callback *callback, struct sig_ss7_linkset *ss7);
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| void sig_ss7_init_linkset(struct sig_ss7_linkset *ss7);
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| 
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| 
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| /* ------------------------------------------------------------------- */
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| 
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| #ifdef __cplusplus
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| }
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| #endif
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| 
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| #endif	/* _ASTERISK_SIG_SS7_H */
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| /* ------------------------------------------------------------------- */
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| /* end sig_ss7.h */
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