Skip to content

HTTPS clone URL

Subversion checkout URL

You can clone with HTTPS or Subversion.

Download ZIP
branch: for-trunk
Fetching contributors…

Cannot retrieve contributors at this time

356 lines (245 sloc) 12.857 kb
=== 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 Hellstr�m, 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 CONTRIBUTERS ===
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,
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 Kal�b - 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.
Jump to Line
Something went wrong with that request. Please try again.