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  1. === DEVELOPMENT SUPPORT ===
  2. We'd like to thank the following companies for helping fund development of
  3. Asterisk.
  4. * Pilosoft, Inc. - for supporting ADSI development in Asterisk
  5. * Asterlink, Inc. - for supporting broad Asterisk development
  6. * GFS - for supporting ALSA development
  7. * Telesthetic - for supporting SIP development
  8. * Christos Ricudis - for substantial code contributions
  9. * nic.at - ENUM support in Asterisk
  10. * Paul Bagyenda, Digital Solutions - for initial Voicetronix driver
  11. development.
  12. * John Todd, TalkPlus, Inc. and JR Richardson, Ntegrated Solutions.
  13. for funding the development of SIP Session Timers support.
  14. * Omnitor AB, Gunnar Hellström, for funding work with videocaps,
  15. T.140 RED, originate with video/text and many more
  16. contributions.
  17. * ClearIT AB for work with meetme, res_mutestream, RTCP, manager and
  18. tonezones.
  19. * NetNation Communications (www.netnation.com)
  20. Kevin Lindsay <kevinl@netnation.com>
  21. Persistent Dynamic Queue Members
  22. * inAccess Networks (work funded by Hellas On Line (HOL) www.hol.gr)
  23. Priorities in queues
  24. * Voop AS, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - funding for
  25. rewrite of SIP transfers
  26. === WISHLIST CONTRIBUTORS ===
  27. We'd like to thank the following for contributing to our wishlist
  28. * Jeremy McNamara - SpeeX support
  29. * Nick Seraphin - RDNIS support
  30. * Gary - Phonejack ADSI (in progress)
  31. * Wasim - Hangup detect
  32. === HARDWARE DONORS ===
  33. We'd like to thank the following for granting access to hardware for testing.
  34. * Thanks to QuickNet Technologies for their donation of an Internet
  35. PhoneJack and Linejack card to the project.
  36. (http://www.quicknet.net)
  37. * Thanks to VoipSupply for their donation of Sipura ATAs to the project
  38. for T.38 testing. (http://www.voipsupply.com)
  39. * Thanks to Grandstream for their donation of ATAs to the project for
  40. T.38 testing. (http://www.grandstream.com)
  41. === MISCELLANEOUS PATCHES ===
  42. We'd like to thank the following for their patches
  43. * Jim Dixon - Zapata Telephony and app_rpt
  44. http://www.zapatatelephony.org/app_rpt.html
  45. * Russell Bryant - Asterisk release manager and countless enhancements
  46. and bug fixes. russell(AT)digium.com
  47. * Anthony Minessale II - Countless big and small fixes, and relentless
  48. forward push. ChanSpy, ForkCDR, ControlPlayback, While/EndWhile,
  49. DumpChan, Dictate, MacroIf, ExecIf, ExecIfTime, RetryDial,
  50. MixMonitor applications; many realtime concepts and
  51. implementation pieces, including res_config_odbc; format_slin;
  52. cdr_custom; several features in Dial including L(), G() and
  53. enhancements to M() and D(); several CDR enhancements including
  54. CDR variables; attended transfer; one touch record; native MOH;
  55. manager eventmask; command line '-t' flag to allow
  56. recording/voicemail on nfs shares; #exec command and multiline
  57. comments in config files; setvar in iax and sip configs.
  58. anthmct(AT)yahoo.com http://www.asterlink.com
  59. * James Golovich - Innumerable contributions, including SIP TCP and TLS
  60. support. You can find him and asterisk-perl at
  61. http://asterisk.gnuinter.net
  62. * Andre Bierwirth - Extension hints and status
  63. * Jean-Denis Girard - Various contributions from the South Pacific
  64. Islands jd-girard(AT)sysnux.pf http://www.sysnux.pf
  65. * William Jordan / Vonage - MySQL enhancements to Voicemail
  66. wjordan(AT)vonage.com
  67. * Jac Kersing - Various fixes
  68. * Steven Critchfield - Seek and Trunc functions for playback and
  69. recording critch(AT)basesys.com
  70. * Jefferson Noxon - app_lookupcidname, app_db, and various other
  71. contributions
  72. * Klaus-Peter Junghanns - in-band DTMF on SIP and MGCP
  73. * Ross Finlayson - Dynamic RTP payload support
  74. * Mahmut Fettahlioglu - Audio recording, music-on-hold changes, alaw
  75. file format, and various fixes. Can be contacted at
  76. mahmut(AT)oa.com.au
  77. * James Dennis - Cisco SIP compatibility patches to work with SIP
  78. service providers. Can be contacted at asterisk(AT)jdennis.net
  79. * Tilghman Lesher - ast_localtime(); ast_say_date_with_format();
  80. GotoIfTime, SayUnixTime, HasNewVoicemail applications;
  81. CUT, SORT, EVAL, CURL, FIELDQTY, STRFTIME, some QUEUE*
  82. functions; func_odbc, cdr_adaptive_odbc, and other innumerable
  83. bug fixes. tilghman(AT)digium.com
  84. http://asterisk.drunkcoder.com
  85. * Jayson Vantuyl - Manager protocol changes, various other bugs.
  86. jvantuyl(AT)computingedge.net
  87. * Thorsten Lockert - OpenBSD, FreeBSD ports, making MacOS X port run on
  88. 10.3, dialplan include verification, route lookup on OpenBSD,
  89. SNMP agent support (res_snmp), various other bugs.
  90. tholo(AT)sigmasoft.com
  91. * Josh Roberson - chan_zap reload support, Advanced Voicemail Features,
  92. & other misc. patches. josh(AT)asteriasgi.com
  93. http://www.asteriasgi.com
  94. * William Waites - syslog support, SIP NAT traversal for SIP-UA.
  95. ww(AT)styx.org
  96. * Rich Murphey - Porting to FreeBSD, NetBSD, OpenBSD, and Darwin.
  97. rich(AT)whiteoaklabs.com http://whiteoaklabs.com
  98. * Simon Lockhart - Porting to Solaris (based on work of Logan ???)
  99. simon(AT)slimey.org
  100. * Olle E. Johansson - SIP RFC compliance, documentation and testing,
  101. testing, SIP outbound proxy support, Manager 1.1 update, SIP
  102. transfer support, SIP presence support, SIP call state updates
  103. (dialog-info), QUEUE_EXISTS function, device state provider
  104. architecture, multiparking (together with mvanbaak), meetme and
  105. parking device states, MiniVM - the small voicemail system,
  106. RTP improvements, RTCP enhancements, DTMF timing fixes,
  107. many documentation updates/corrections, and many bug fixes.
  108. oej(AT)edvina.net, http://edvina.net
  109. * Steve Kann - new jitter buffer for IAX2
  110. stevek(AT)stevek.com
  111. * Constantine Filin - major contributions to the Asterisk Realtime
  112. Architecture
  113. * Steve Murphy - privacy support, $[ ] parser upgrade, AEL2 parser
  114. upgrade. murf(AT)digium.com
  115. * Claude Patry - bug fixes, feature enhancements, and bug marshalling
  116. cpatry(AT)gmail.com
  117. * Miroslav Nachev, miro(AT)space-comm.com
  118. COSMOS Software Enterprises, Ltd.
  119. Variable for No Answer Timeout for Attended Transfer
  120. * Slav Klenov & Vanheuverzwijn Joachim - development of the generic
  121. jitterbuffer Securax Ltd. info(AT)securax.be
  122. * Roy Sigurd Karlsbakk - providing funding for generic jitterbuffer
  123. development roy(AT)karlsbakk.net, Briiz Telecom AS
  124. * Voop AS, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - rewrite
  125. of SIP transfers
  126. * Philippe Sultan - RADIUS CDR module, many fixes to res_jabber and
  127. gtalk/jingle channel drivers. INRIA, http://www.inria.fr/
  128. * John Martin, Aupix - Improved video support in the SIP channel
  129. T.140 text support in RTP/SIP
  130. * Steve Underwood - Provided T.38 pass through support.
  131. * George Konstantoulakis - Support for Greek in voicemail added by
  132. InAccess Networks (work funded by HOL, www.hol.gr)
  133. gkon(AT)inaccessnetworks.com
  134. * Daniel Nylander - Support for Swedish and Norwegian languages in
  135. voicemail. http://www.danielnylander.se/
  136. * Stojan Sljivic - An option for maximum number of messsages per
  137. mailbox in voicemail. Also an issue with voicemail
  138. synchronization has been fixed. GDS Partners
  139. www.gdspartners.com stojan.sljivic(AT)gdspartners.com
  140. * Bartosz Supczinski - Support for Polish added by DIR (www.dir.pl)
  141. Bartosz.Supczinski(AT)dir.pl
  142. * James Rothenberger - Support for IMAP storage integration added by
  143. OneBizTone LLC Work funded by University of Pennsylvania
  144. jar(AT)onebiztone.com
  145. * Paul Cadach - Bringing chan_h323 up to date, bug fixes, and more!
  146. * Voop AS - Financial support for a lot of work with the SIP driver
  147. and the IAX trunk MTU patch
  148. * Cedric Hans - Development of chan_unistim cedric.hans(AT)mlkj.net
  149. * Takao Takahashi & Mina Naguib - chan_unistim improvements for
  150. smaller devices
  151. * Sergio Fadda - console_video: video support for chan_oss and
  152. chan_alsa
  153. * Marta Carbone - console_video and the astobj2 framework
  154. * Luigi Rizzo - astobj2, console_video, windows build, chan_oss cleanup,
  155. and a bunch of infrastructure work (loader, new_cli, ...)
  156. * Brett Bryant - digit option for musiconhold selection, ENUMQUERY and
  157. ENUMRESULT functions, feature group configuration for
  158. features.conf, per-file CLI debug and verbose settings, TCP and
  159. TLS support for SIP, and various bug fixes.
  160. brettbryant(AT)gmail.com
  161. * Sergey Tamkovich - Realtime support for MusicOnHold, store and destroy
  162. realtime methods and implementations for odbc, sqlite, and pgsql
  163. realtime drivers, attended transfer updates, multiple speeds for
  164. ControlPlayback, and multiple bug fixes See
  165. http://voip-info.org/users/view/sergee serg(AT)voipsolutions.ru
  166. * Klaus Darillon - the SIPremoveHeader function in chan_sip and SIP Path
  167. Support.
  168. * Moises Silva (moy) - for writing LibOpenR2, and providing support for
  169. it in chan_dahdi moises.silva(AT)gmail.com
  170. * Eliel C. Sardanons - XML documentation implementation, and various
  171. other contributions eliels(AT)gmail.com
  172. * Sean Bright - Snom call pickup, newt interface for menuselect,
  173. cdr_tds rewrite, countless other improvements, fixes, and good
  174. ideas. sean(AT)malleable.com
  175. * Jan Kaláb - Calendaring support for Exchange Server 2007+ via
  176. Exchange Web Services.
  177. * University of Oslo (uio.no), Norway - SIP Max-Forwards setting
  178. support (developed by oej)
  179. * FCCN, Lissabon, Portugal - SIP show channels CLI command
  180. (developed by oej)
  181. * Viagenie, Canada - IPv6 support in socket layers and SIP
  182. implementation Developers: Marc Blanchet, Simon Perreault and
  183. Jean-Philippe Dionne
  184. * ClearIT AB, Sweden - res_mutestream, queue_exists and various other
  185. patches (developed by oej)
  186. * Despegar.com, Argentina - AstData API implementation, also sponsored
  187. by Google as part of the gsoc/2009 program (developed by Eliel)
  188. * Philippe Lindheimer - DEV_STATE additions to CCSS
  189. * Andrew "lathama" Latham <lathama at gmail dot com>
  190. Doxygen, HTTP-Static, Phoneprov, make update
  191. * George Joseph - PJSIP CLI commands, PJSIP_HEADER dialplan function
  192. === OTHER CONTRIBUTIONS ===
  193. We'd like to thank the following for their listed contributions.
  194. * John Todd - Monkey sounds and associated teletorture prompt
  195. * Michael Jerris - bug marshaling
  196. * Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
  197. available under a Creative Commons License at
  198. http://www.asteriskdocs.org
  199. * Brian M. Clapper - poll.c emulation
  200. This product includes software developed by
  201. Brian M. Clapper <bmc(AT)clapper.org>
  202. === HOLD MUSIC ===
  203. We'd like to thank the following for hold music
  204. * Music provided by www.opsound.org
  205. === OTHER SOURCE CODE IN ASTERISK ===
  206. We'd like to thank the following for their code use
  207. * Asterisk uses libedit, the lightweight readline replacement from
  208. NetBSD.
  209. * The cdr_radius module uses libradiusclient-ng, which is also from
  210. NetBSD.
  211. * They are BSD-licensed and require the following statement:
  212. This product includes software developed by the NetBSD
  213. Foundation, Inc. and its contributors.
  214. * Digium did not implement the codecs in Asterisk.
  215. Here is the copyright on the GSM source:
  216. Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
  217. Technische Universitaet Berlin
  218. Any use of this software is permitted provided that this notice is not
  219. removed and that neither the authors nor the Technische Universitaet Berlin
  220. are deemed to have made any representations as to the suitability of this
  221. software for any purpose nor are held responsible for any defects of
  222. this software. THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
  223. As a matter of courtesy, the authors request to be informed about uses
  224. this software has found, about bugs in this software, and about any
  225. improvements that may be of general interest.
  226. Berlin, 28.11.1994
  227. Jutta Degener
  228. Carsten Bormann
  229. And the copyright on the ADPCM source:
  230. Copyright 1992 by Stichting Mathematisch Centrum, Amsterdam, The
  231. Netherlands.
  232. All Rights Reserved
  233. Permission to use, copy, modify, and distribute this software and its
  234. documentation for any purpose and without fee is hereby granted,
  235. provided that the above copyright notice appear in all copies and that
  236. both that copyright notice and this permission notice appear in
  237. supporting documentation, and that the names of Stichting Mathematisch
  238. Centrum or CWI not be used in advertising or publicity pertaining to
  239. distribution of the software without specific, written prior permission.
  240. STICHTING MATHEMATISCH CENTRUM DISCLAIMS ALL WARRANTIES WITH REGARD TO
  241. THIS SOFTWARE, INCLUDING ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND
  242. FITNESS, IN NO EVENT SHALL STICHTING MATHEMATISCH CENTRUM BE LIABLE
  243. FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
  244. WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
  245. ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT
  246. OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.