=== 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 <[email protected]>

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.