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Asterisk - The Open Source Telephony Project GIT-master-f36a736
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=== DEVELOPMENT SUPPORT === We'd like to thank the following companies for helping fund development of Asterisk. * Pilosoft, Inc. - for supporting ADSI development in Asterisk * Asterlink, Inc. - for supporting broad Asterisk development * GFS - for supporting ALSA development * Telesthetic - for supporting SIP development * Christos Ricudis - for substantial code contributions * nic.at - ENUM support in Asterisk * Paul Bagyenda, Digital Solutions - for initial Voicetronix driver development. * 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, 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 === WISHLIST CONTRIBUTORS === 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 === HARDWARE DONORS === We'd like to thank the following for granting access to hardware for testing. * Thanks to QuickNet Technologies for their donation of an Internet PhoneJack and Linejack card to the project. (http://www.quicknet.net) * 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) === MISCELLANEOUS PATCHES === We'd like to thank the following for their patches * Jim Dixon - Zapata Telephony and app_rpt http://www.zapatatelephony.org/app_rpt.html * Russell Bryant - Asterisk release manager and countless enhancements and bug fixes. russell(AT)digium.com * 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 * James Golovich - Innumerable contributions, including SIP TCP and TLS support. You can find him and asterisk-perl at http://asterisk.gnuinter.net * Andre Bierwirth - Extension hints and status * Jean-Denis Girard - Various contributions from the South Pacific Islands jd-girard(AT)sysnux.pf http://www.sysnux.pf * William Jordan / Vonage - MySQL enhancements to Voicemail wjordan(AT)vonage.com * Jac Kersing - Various fixes * Steven Critchfield - Seek and Trunc functions for playback and recording critch(AT)basesys.com * Jefferson Noxon - app_lookupcidname, app_db, and various other contributions * Klaus-Peter Junghanns - in-band DTMF on SIP and MGCP * Ross Finlayson - Dynamic RTP payload support * Mahmut Fettahlioglu - Audio recording, music-on-hold changes, alaw file format, and various fixes. Can be contacted at mahmut(AT)oa.com.au * James Dennis - Cisco SIP compatibility patches to work with SIP service providers. Can be contacted at asterisk(AT)jdennis.net * 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 * Jayson Vantuyl - Manager protocol changes, various other bugs. jvantuyl(AT)computingedge.net * 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 * Josh Roberson - chan_zap reload support, Advanced Voicemail Features, & other misc. patches. josh(AT)asteriasgi.com http://www.asteriasgi.com * William Waites - syslog support, SIP NAT traversal for SIP-UA. ww(AT)styx.org * Rich Murphey - Porting to FreeBSD, NetBSD, OpenBSD, and Darwin. rich(AT)whiteoaklabs.com http://whiteoaklabs.com * Simon Lockhart - Porting to Solaris (based on work of Logan ???) simon(AT)slimey.org * 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, many documentation updates/corrections, and many bug fixes. oej(AT)edvina.net, http://edvina.net * Steve Kann - new jitter buffer for IAX2 stevek(AT)stevek.com * Constantine Filin - major contributions to the Asterisk Realtime Architecture * Steve Murphy - privacy support, $[ ] parser upgrade, AEL2 parser upgrade. murf(AT)digium.com * Claude Patry - bug fixes, feature enhancements, and bug marshalling cpatry(AT)gmail.com * Miroslav Nachev, miro(AT)space-comm.com COSMOS Software Enterprises, Ltd. Variable for No Answer Timeout for Attended Transfer * Slav Klenov & Vanheuverzwijn Joachim - development of the generic jitterbuffer Securax Ltd. info(AT)securax.be * Roy Sigurd Karlsbakk - providing funding for generic jitterbuffer development roy(AT)karlsbakk.net, Briiz Telecom AS * Voop AS, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - rewrite of SIP transfers * Philippe Sultan - RADIUS CDR module, many fixes to res_jabber and gtalk/jingle channel drivers. INRIA, http://www.inria.fr/ * John Martin, Aupix - Improved video support in the SIP channel T.140 text support in RTP/SIP * Steve Underwood - Provided T.38 pass through support. * George Konstantoulakis - Support for Greek in voicemail added by InAccess Networks (work funded by HOL, www.hol.gr) gkon(AT)inaccessnetworks.com * Daniel Nylander - Support for Swedish and Norwegian languages in voicemail. http://www.danielnylander.se/ * 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 * Bartosz Supczinski - Support for Polish added by DIR (www.dir.pl) Bartosz.Supczinski(AT)dir.pl * James Rothenberger - Support for IMAP storage integration added by OneBizTone LLC Work funded by University of Pennsylvania jar(AT)onebiztone.com * Paul Cadach - Bringing chan_h323 up to date, bug fixes, and more! * Voop AS - Financial support for a lot of work with the SIP driver and the IAX trunk MTU patch * Cedric Hans - Development of chan_unistim cedric.hans(AT)mlkj.net * Takao Takahashi & Mina Naguib - chan_unistim improvements for smaller devices * Sergio Fadda - console_video: video support for chan_oss and chan_alsa * Marta Carbone - console_video and the astobj2 framework * Luigi Rizzo - astobj2, console_video, windows build, chan_oss cleanup, and a bunch of infrastructure work (loader, new_cli, ...) * 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 * 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. * Moises Silva (moy) - for writing LibOpenR2, and providing support for it in chan_dahdi moises.silva(AT)gmail.com * Eliel C. Sardanons - XML documentation implementation, and various other contributions eliels(AT)gmail.com * 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 Exchange Web Services. * University of Oslo (uio.no), Norway - SIP Max-Forwards setting support (developed by oej) * FCCN, Lissabon, Portugal - SIP show channels CLI command (developed by oej) * Viagenie, Canada - IPv6 support in socket layers and SIP implementation Developers: Marc Blanchet, Simon Perreault and Jean-Philippe Dionne * ClearIT AB, Sweden - res_mutestream, queue_exists and various other patches (developed by oej) * 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 * 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 === 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> === HOLD MUSIC === We'd like to thank the following for hold music * Music provided by www.opsound.org === OTHER SOURCE CODE IN ASTERISK === We'd like to thank the following for their code use * 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. * 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 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 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.
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