[svn-commits] lathama: trunk r374887 - /trunk/CREDITS
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Oct 11 17:35:43 CDT 2012
Author: lathama
Date: Thu Oct 11 17:35:41 2012
New Revision: 374887
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=374887
Log:
CREDITS clean up
As discussed online http://lists.digium.com/pipermail/asterisk-dev/2012-October/057245.html the credits file needs some cleaning. This is 95% whitespace with a few additions found in file headers. Further additions should be added here instead of in the file being updated.
(issue ASTERISK-20259)
Modified:
trunk/CREDITS
Modified: trunk/CREDITS
URL: http://svnview.digium.com/svn/asterisk/trunk/CREDITS?view=diff&rev=374887&r1=374886&r2=374887
==============================================================================
--- trunk/CREDITS (original)
+++ trunk/CREDITS Thu Oct 11 17:35:41 2012
@@ -1,255 +1,316 @@
=== 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
+
+ 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 at 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 CONTRIBUTERS ===
-Jeremy McNamara - SpeeX support
-Nick Seraphin - RDNIS support
-Gary - Phonejack ADSI (in progress)
-Wasim - Hangup detect
+
+ 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 ===
-* 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)
+
+ We'd like to thank the followwing 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 ===
-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)esoft.pf http://www.esoft.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, 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.
- - for 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 - funding for 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
-
-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
+
+ We'd like to thank the flollowing 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)esoft.pf http://www.esoft.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,
+ 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
+
+ * 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
=== OTHER 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>
+
+ 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 ===
-Music provided by www.opsound.org
+
+ We'd like to thank the following for hold music
+
+ * Music provided by www.opsound.org
=== OTHER SOURCE CODE IN ASTERISK ===
-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
+
+ 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
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