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    === DEVELOPMENT SUPPORT ===
    
    
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     We'd like to thank the following companies for helping fund development of
     Asterisk.
    
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    	* Pilosoft, Inc. - for supporting ADSI development in Asterisk
    
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    	* Asterlink, Inc. - for supporting broad Asterisk development
    
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    	* GFS - for supporting ALSA development
    
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    	* Telesthetic - for supporting SIP development
    
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    	* Christos Ricudis - for substantial code contributions
    
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    	* nic.at - ENUM support in Asterisk
    
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    	* Paul Bagyenda, Digital Solutions - for initial Voicetronix driver
    		development.
    
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    	* John Todd, TalkPlus, Inc.  and JR Richardson, Ntegrated Solutions. 
    		for funding the development of SIP Session Timers support.
    
    
    	* Omnitor AB, Gunnar Hellström, for funding work with videocaps, 
    
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    		T.140 RED, originate with video/text and many more 
    		contributions.
    
    	* ClearIT AB for work with meetme, res_mutestream, RTCP, manager and 
    		tonezones.
    
    	* NetNation Communications (www.netnation.com)
    		Kevin Lindsay <kevinl@netnation.com>
    		Persistent Dynamic Queue Members
    
    	* inAccess Networks (work funded by Hellas On Line (HOL) www.hol.gr)
    		Priorities in queues
    
    	* Voop AS, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - funding for
    		rewrite of SIP transfers
    
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    === WISHLIST CONTRIBUTORS ===
    
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     We'd like to thank the following for contributing to our wishlist
    
    	* Jeremy McNamara - SpeeX support
    
    	* Nick Seraphin - RDNIS support
    
    	* Gary - Phonejack ADSI (in progress)
    
    	* Wasim - Hangup detect
    
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    === HARDWARE DONORS === 
    
    
     We'd like to thank the following for granting access to hardware for testing.
    
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    	* Thanks to QuickNet Technologies for their donation of an Internet
    		PhoneJack and Linejack card to the project.  
    		(http://www.quicknet.net)
    
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    	* Thanks to VoipSupply for their donation of Sipura ATAs to the project
    		for T.38 testing. (http://www.voipsupply.com)
    
    	* Thanks to Grandstream for their donation of ATAs to the project for
    		T.38 testing. (http://www.grandstream.com)
    
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    === MISCELLANEOUS PATCHES ===
    
     We'd like to thank the following for their patches
    
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    	* Jim Dixon - Zapata Telephony and app_rpt
    		http://www.zapatatelephony.org/app_rpt.html
    
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    	* Russell Bryant - Asterisk release manager and countless enhancements
    		and bug fixes. russell(AT)digium.com
    
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    	* Anthony Minessale II - Countless big and small fixes, and relentless
    		forward	push. ChanSpy, ForkCDR, ControlPlayback, While/EndWhile,
    		DumpChan, Dictate, MacroIf, ExecIf, ExecIfTime, RetryDial, 
    		MixMonitor applications; many realtime concepts and 
    		implementation pieces, including res_config_odbc; format_slin; 
    		cdr_custom; several features in Dial including L(), G() and 
    		enhancements to M() and D(); several CDR enhancements including
    		CDR variables; attended transfer; one touch record; native MOH;
    		manager eventmask; command line '-t' flag to allow 
    		recording/voicemail on nfs shares; #exec command and multiline 
    		comments in config files; setvar in iax and sip configs.
    		anthmct(AT)yahoo.com http://www.asterlink.com
    
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    	* James Golovich - Innumerable contributions, including SIP TCP and TLS
    		support. You can find him and asterisk-perl at 
    		http://asterisk.gnuinter.net
    
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    	* Andre Bierwirth - Extension hints and status
    
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    	* Jean-Denis Girard - Various contributions from the South Pacific
    
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    		Islands jd-girard(AT)sysnux.pf http://www.sysnux.pf
    
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    	* William Jordan / Vonage - MySQL enhancements to Voicemail
    		wjordan(AT)vonage.com
    
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    	* Jac Kersing - Various fixes
    
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    	* Steven Critchfield - Seek and Trunc functions for playback and 
    		recording critch(AT)basesys.com
    
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    	* Jefferson Noxon - app_lookupcidname, app_db, and various other
    		contributions
    
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    	* Klaus-Peter Junghanns - in-band DTMF on SIP and MGCP
    
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    	* Ross Finlayson - Dynamic RTP payload support
    
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    	* Mahmut Fettahlioglu - Audio recording, music-on-hold changes, alaw
    		file format, and various fixes. Can be contacted at 
    		mahmut(AT)oa.com.au
    
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    	* James Dennis - Cisco SIP compatibility patches to work with SIP
    		service providers. Can be contacted at asterisk(AT)jdennis.net
    
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    	* Tilghman Lesher - ast_localtime(); ast_say_date_with_format(); 
    		GotoIfTime, SayUnixTime, HasNewVoicemail applications;
    		CUT, SORT, EVAL, CURL, FIELDQTY, STRFTIME, some QUEUE* 
    		functions; func_odbc, cdr_adaptive_odbc, and other innumerable
    		bug fixes. tilghman(AT)digium.com 
    		http://asterisk.drunkcoder.com
    
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    	* Jayson Vantuyl - Manager protocol changes, various other bugs.
    		jvantuyl(AT)computingedge.net
    
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    	* Thorsten Lockert - OpenBSD, FreeBSD ports, making MacOS X port run on
    		10.3, dialplan include verification, route lookup on OpenBSD,
    		SNMP agent support (res_snmp), various other bugs. 
    		tholo(AT)sigmasoft.com
    
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    	* Josh Roberson - chan_zap reload support, Advanced Voicemail Features,
    		& other misc. patches. josh(AT)asteriasgi.com 
    		http://www.asteriasgi.com
    
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    	* William Waites - syslog support, SIP NAT traversal for SIP-UA. 
    		ww(AT)styx.org
    
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    	* Rich Murphey - Porting to FreeBSD, NetBSD, OpenBSD, and Darwin.
    		rich(AT)whiteoaklabs.com  http://whiteoaklabs.com
    
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    	* Simon Lockhart - Porting to Solaris (based on work of Logan ???)	
    		simon(AT)slimey.org
    
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    	* Olle E. Johansson - SIP RFC compliance, documentation and testing,
    		testing, SIP outbound proxy support, Manager 1.1 update, SIP 
    		transfer support, SIP presence support, SIP call state updates
    		(dialog-info), QUEUE_EXISTS function, device state provider
    		architecture, multiparking (together with mvanbaak), meetme and
    		parking device states, MiniVM - the small voicemail system,
    
    		RTP improvements, RTCP enhancements, DTMF timing fixes,
    
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    		many documentation updates/corrections, and many bug fixes.
    		oej(AT)edvina.net, http://edvina.net
    
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    	* Steve Kann - new jitter buffer for IAX2
    		stevek(AT)stevek.com
    
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    	* Constantine Filin - major contributions to the Asterisk Realtime
    		Architecture
    
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    	* Steve Murphy - privacy support, $[ ] parser upgrade, AEL2 parser
    		upgrade. murf(AT)digium.com
    
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    	* Claude Patry - bug fixes, feature enhancements, and bug marshalling
    		cpatry(AT)gmail.com
    
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    	* Miroslav Nachev, miro(AT)space-comm.com 
    		COSMOS Software Enterprises, Ltd.
    		Variable for No Answer Timeout for Attended Transfer
    
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    	* Slav Klenov & Vanheuverzwijn Joachim - development of the generic
    		jitterbuffer Securax Ltd. info(AT)securax.be
    
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    	* Roy Sigurd Karlsbakk - providing funding for generic jitterbuffer
    		development roy(AT)karlsbakk.net, Briiz Telecom AS
    
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    	* Voop AS, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - rewrite
    		of SIP transfers
    
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    	* Philippe Sultan - RADIUS CDR module, many fixes to res_jabber and 
    		gtalk/jingle channel drivers. INRIA, http://www.inria.fr/
    
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    	* John Martin, Aupix - Improved video support in the SIP channel
    		T.140 text support in RTP/SIP
    
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    	* Steve Underwood - Provided T.38 pass through support.
    
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    	* George Konstantoulakis - Support for Greek in voicemail added by 
    		InAccess Networks (work funded by HOL, www.hol.gr) 
    		gkon(AT)inaccessnetworks.com
    
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    	* Daniel Nylander - Support for Swedish and Norwegian languages in
    		voicemail. http://www.danielnylander.se/
    
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    	* Stojan Sljivic - An option for maximum number of messsages per
    		mailbox in voicemail.  Also an issue with voicemail 
    		synchronization has been fixed. GDS Partners 
    		www.gdspartners.com stojan.sljivic(AT)gdspartners.com
    
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    	* Bartosz Supczinski - Support for Polish added by DIR (www.dir.pl)
    		Bartosz.Supczinski(AT)dir.pl
    
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    	* James Rothenberger - Support for IMAP storage integration added by
    		OneBizTone LLC Work funded by University of Pennsylvania 
    		jar(AT)onebiztone.com
    
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    	* Paul Cadach - Bringing chan_h323 up to date, bug fixes, and more!
    
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    	* Voop AS - Financial support for a lot of work with the SIP driver 
    		and the IAX trunk MTU patch
    
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    	* Cedric Hans - Development of chan_unistim cedric.hans(AT)mlkj.net
    
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    	* Takao Takahashi & Mina Naguib - chan_unistim improvements for 
    		smaller devices
    
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    	* Sergio Fadda - console_video: video support for chan_oss and 
    		chan_alsa
    
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    	* Marta Carbone - console_video and the astobj2 framework
    
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    	* Luigi Rizzo - astobj2, console_video, windows build, chan_oss cleanup,
    		and a bunch of infrastructure work (loader, new_cli, ...)
    
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    	* Brett Bryant - digit option for musiconhold selection, ENUMQUERY and 
    		ENUMRESULT functions, feature group configuration for 
    		features.conf, per-file CLI debug and verbose settings, TCP and
    		TLS support for SIP, and various bug fixes.
    		brettbryant(AT)gmail.com
    
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    	* Sergey Tamkovich - Realtime support for MusicOnHold, store and destroy
    		realtime methods and implementations for odbc, sqlite, and pgsql
    		realtime drivers, attended transfer updates, multiple speeds for
    		ControlPlayback, and multiple bug fixes See 
    		http://voip-info.org/users/view/sergee serg(AT)voipsolutions.ru
    
    	* Klaus Darillon - the SIPremoveHeader function in chan_sip and SIP Path
    		Support.
    
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    	* Moises Silva (moy) - for writing LibOpenR2, and providing support for
    		it in chan_dahdi moises.silva(AT)gmail.com
    
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    	* Eliel C. Sardanons - XML documentation implementation, and various
    		other contributions eliels(AT)gmail.com
    
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    	* Sean Bright - Snom call pickup, newt interface for menuselect,
    		cdr_tds rewrite, countless other improvements, fixes, and good
    		ideas. sean(AT)malleable.com
    
    	* Jan Kaláb - Calendaring support for Exchange Server 2007+ via 
    
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    		Exchange Web Services.
    
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    	* University of Oslo (uio.no), Norway - SIP Max-Forwards setting 
    		support (developed by oej)
    
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    	* FCCN, Lissabon, Portugal - SIP show channels CLI command 
    		(developed by oej)
    
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    	* Viagenie, Canada - IPv6 support in socket layers and SIP 
    		implementation Developers: Marc Blanchet, Simon Perreault and 
    		Jean-Philippe Dionne
    
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    	* ClearIT AB, Sweden - res_mutestream, queue_exists and various other
    		patches (developed by oej)
    
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    	* Despegar.com, Argentina - AstData API implementation, also sponsored
    		by Google as part of the gsoc/2009 program (developed by Eliel)
    
    	* Philippe Lindheimer - DEV_STATE additions to CCSS
    
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    	* Andrew "lathama" Latham <lathama at gmail dot com>
    		Doxygen, HTTP-Static, Phoneprov, make update
    
    
    	* George Joseph - PJSIP CLI commands, PJSIP_HEADER dialplan function
    
    
    === OTHER CONTRIBUTIONS ===
    
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     We'd like to thank the following for their listed contributions.
    
    	* John Todd - Monkey sounds and associated teletorture prompt
    
    	* Michael Jerris - bug marshaling
    	
    	* Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
    		available under a Creative Commons License at
    		http://www.asteriskdocs.org
    
    	* Brian M. Clapper - poll.c emulation 
    		This product includes software developed by 
    		Brian M. Clapper <bmc(AT)clapper.org>
    
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     We'd like to thank the following for hold music
    
    	* Music provided by www.opsound.org
    
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    === OTHER SOURCE CODE IN ASTERISK ===
    
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     We'd like to thank the following for their code use
    
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    	* Asterisk uses libedit, the lightweight readline replacement from
    		NetBSD.
    	* The cdr_radius module uses libradiusclient-ng, which is also from
    		NetBSD.
    	* They are BSD-licensed and require the following statement:
    		This product includes software developed by the NetBSD
    		Foundation, Inc. and its contributors.
    
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    	* Digium did not implement the codecs in Asterisk.
    		Here is the copyright on the GSM source:
    		Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
    		Technische Universitaet Berlin
    
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    Any use of this software is permitted provided that this notice is not
    removed and that neither the authors nor the Technische Universitaet Berlin
    are deemed to have made any representations as to the suitability of this
    software for any purpose nor are held responsible for any defects of
    this software.  THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
    
    As a matter of courtesy, the authors request to be informed about uses
    this software has found, about bugs in this software, and about any
    improvements that may be of general interest.
    
    Berlin, 28.11.1994
    Jutta Degener
    Carsten Bormann
    
    And the copyright on the ADPCM source:
    
    Copyright 1992 by Stichting Mathematisch Centrum, Amsterdam, The
    Netherlands.
    
    
    	                 All Rights Reserved
    
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    Permission to use, copy, modify, and distribute this software and its
    documentation for any purpose and without fee is hereby granted,
    provided that the above copyright notice appear in all copies and that
    both that copyright notice and this permission notice appear in
    supporting documentation, and that the names of Stichting Mathematisch
    Centrum or CWI not be used in advertising or publicity pertaining to
    distribution of the software without specific, written prior permission.
    
    STICHTING MATHEMATISCH CENTRUM DISCLAIMS ALL WARRANTIES WITH REGARD TO
    THIS SOFTWARE, INCLUDING ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND
    FITNESS, IN NO EVENT SHALL STICHTING MATHEMATISCH CENTRUM BE LIABLE
    FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
    WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
    ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT
    OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.