357 lines
13 KiB
Plaintext
357 lines
13 KiB
Plaintext
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=== DEVELOPMENT SUPPORT ===
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We'd like to thank the following companies for helping fund development of
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Asterisk.
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* Pilosoft, Inc. - for supporting ADSI development in Asterisk
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* Asterlink, Inc. - for supporting broad Asterisk development
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* GFS - for supporting ALSA development
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* Telesthetic - for supporting SIP development
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* Christos Ricudis - for substantial code contributions
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* nic.at - ENUM support in Asterisk
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* Paul Bagyenda, Digital Solutions - for initial Voicetronix driver
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development.
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* John Todd, TalkPlus, Inc. and JR Richardson, Ntegrated Solutions.
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for funding the development of SIP Session Timers support.
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* Omnitor AB, Gunnar Hellström, for funding work with videocaps,
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T.140 RED, originate with video/text and many more
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contributions.
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* ClearIT AB for work with meetme, res_mutestream, RTCP, manager and
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tonezones.
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* NetNation Communications (www.netnation.com)
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Kevin Lindsay <kevinl@netnation.com>
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Persistent Dynamic Queue Members
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* inAccess Networks (work funded by Hellas On Line (HOL) www.hol.gr)
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Priorities in queues
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* Voop AS, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - funding for
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rewrite of SIP transfers
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=== WISHLIST CONTRIBUTORS ===
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We'd like to thank the following for contributing to our wishlist
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* Jeremy McNamara - SpeeX support
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* Nick Seraphin - RDNIS support
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* Gary - Phonejack ADSI (in progress)
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* Wasim - Hangup detect
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=== HARDWARE DONORS ===
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We'd like to thank the following for granting access to hardware for testing.
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* Thanks to QuickNet Technologies for their donation of an Internet
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PhoneJack and Linejack card to the project.
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(http://www.quicknet.net)
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* Thanks to VoipSupply for their donation of Sipura ATAs to the project
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for T.38 testing. (http://www.voipsupply.com)
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* Thanks to Grandstream for their donation of ATAs to the project for
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T.38 testing. (http://www.grandstream.com)
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=== MISCELLANEOUS PATCHES ===
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We'd like to thank the following for their patches
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* Jim Dixon - Zapata Telephony and app_rpt
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http://www.zapatatelephony.org/app_rpt.html
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* Russell Bryant - Asterisk release manager and countless enhancements
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and bug fixes. russell(AT)digium.com
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* Anthony Minessale II - Countless big and small fixes, and relentless
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forward push. ChanSpy, ForkCDR, ControlPlayback, While/EndWhile,
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DumpChan, Dictate, MacroIf, ExecIf, ExecIfTime, RetryDial,
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MixMonitor applications; many realtime concepts and
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implementation pieces, including res_config_odbc; format_slin;
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cdr_custom; several features in Dial including L(), G() and
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enhancements to M() and D(); several CDR enhancements including
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CDR variables; attended transfer; one touch record; native MOH;
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manager eventmask; command line '-t' flag to allow
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recording/voicemail on nfs shares; #exec command and multiline
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comments in config files; setvar in iax and sip configs.
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anthmct(AT)yahoo.com http://www.asterlink.com
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* James Golovich - Innumerable contributions, including SIP TCP and TLS
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support. You can find him and asterisk-perl at
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http://asterisk.gnuinter.net
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* Andre Bierwirth - Extension hints and status
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* Jean-Denis Girard - Various contributions from the South Pacific
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Islands jd-girard(AT)sysnux.pf http://www.sysnux.pf
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* William Jordan / Vonage - MySQL enhancements to Voicemail
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wjordan(AT)vonage.com
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* Jac Kersing - Various fixes
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* Steven Critchfield - Seek and Trunc functions for playback and
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recording critch(AT)basesys.com
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* Jefferson Noxon - app_lookupcidname, app_db, and various other
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contributions
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* Klaus-Peter Junghanns - in-band DTMF on SIP and MGCP
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* Ross Finlayson - Dynamic RTP payload support
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* Mahmut Fettahlioglu - Audio recording, music-on-hold changes, alaw
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file format, and various fixes. Can be contacted at
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mahmut(AT)oa.com.au
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* James Dennis - Cisco SIP compatibility patches to work with SIP
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service providers. Can be contacted at asterisk(AT)jdennis.net
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* Tilghman Lesher - ast_localtime(); ast_say_date_with_format();
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GotoIfTime, SayUnixTime, HasNewVoicemail applications;
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CUT, SORT, EVAL, CURL, FIELDQTY, STRFTIME, some QUEUE*
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functions; func_odbc, cdr_adaptive_odbc, and other innumerable
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bug fixes. tilghman(AT)digium.com
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http://asterisk.drunkcoder.com
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* Jayson Vantuyl - Manager protocol changes, various other bugs.
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jvantuyl(AT)computingedge.net
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* Thorsten Lockert - OpenBSD, FreeBSD ports, making MacOS X port run on
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10.3, dialplan include verification, route lookup on OpenBSD,
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SNMP agent support (res_snmp), various other bugs.
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tholo(AT)sigmasoft.com
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* Josh Roberson - chan_zap reload support, Advanced Voicemail Features,
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& other misc. patches. josh(AT)asteriasgi.com
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http://www.asteriasgi.com
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* William Waites - syslog support, SIP NAT traversal for SIP-UA.
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ww(AT)styx.org
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* Rich Murphey - Porting to FreeBSD, NetBSD, OpenBSD, and Darwin.
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rich(AT)whiteoaklabs.com http://whiteoaklabs.com
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* Simon Lockhart - Porting to Solaris (based on work of Logan ???)
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simon(AT)slimey.org
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* Olle E. Johansson - SIP RFC compliance, documentation and testing,
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testing, SIP outbound proxy support, Manager 1.1 update, SIP
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transfer support, SIP presence support, SIP call state updates
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(dialog-info), QUEUE_EXISTS function, device state provider
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architecture, multiparking (together with mvanbaak), meetme and
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parking device states, MiniVM - the small voicemail system,
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RTP improvements, RTCP enhancements, DTMF timing fixes,
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many documentation updates/corrections, and many bug fixes.
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oej(AT)edvina.net, http://edvina.net
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* Steve Kann - new jitter buffer for IAX2
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stevek(AT)stevek.com
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* Constantine Filin - major contributions to the Asterisk Realtime
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Architecture
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* Steve Murphy - privacy support, $[ ] parser upgrade, AEL2 parser
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upgrade. murf(AT)digium.com
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* Claude Patry - bug fixes, feature enhancements, and bug marshalling
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cpatry(AT)gmail.com
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* Miroslav Nachev, miro(AT)space-comm.com
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COSMOS Software Enterprises, Ltd.
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Variable for No Answer Timeout for Attended Transfer
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* Slav Klenov & Vanheuverzwijn Joachim - development of the generic
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jitterbuffer Securax Ltd. info(AT)securax.be
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* Roy Sigurd Karlsbakk - providing funding for generic jitterbuffer
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development roy(AT)karlsbakk.net, Briiz Telecom AS
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* Voop AS, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - rewrite
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of SIP transfers
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* Philippe Sultan - RADIUS CDR module, many fixes to res_jabber and
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gtalk/jingle channel drivers. INRIA, http://www.inria.fr/
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* John Martin, Aupix - Improved video support in the SIP channel
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T.140 text support in RTP/SIP
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* Steve Underwood - Provided T.38 pass through support.
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* George Konstantoulakis - Support for Greek in voicemail added by
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InAccess Networks (work funded by HOL, www.hol.gr)
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gkon(AT)inaccessnetworks.com
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* Daniel Nylander - Support for Swedish and Norwegian languages in
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voicemail. http://www.danielnylander.se/
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* Stojan Sljivic - An option for maximum number of messsages per
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mailbox in voicemail. Also an issue with voicemail
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synchronization has been fixed. GDS Partners
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www.gdspartners.com stojan.sljivic(AT)gdspartners.com
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* Bartosz Supczinski - Support for Polish added by DIR (www.dir.pl)
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Bartosz.Supczinski(AT)dir.pl
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* James Rothenberger - Support for IMAP storage integration added by
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OneBizTone LLC Work funded by University of Pennsylvania
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jar(AT)onebiztone.com
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* Paul Cadach - Bringing chan_h323 up to date, bug fixes, and more!
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* Voop AS - Financial support for a lot of work with the SIP driver
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and the IAX trunk MTU patch
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* Cedric Hans - Development of chan_unistim cedric.hans(AT)mlkj.net
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* Takao Takahashi & Mina Naguib - chan_unistim improvements for
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smaller devices
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* Sergio Fadda - console_video: video support for chan_oss and
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chan_alsa
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* Marta Carbone - console_video and the astobj2 framework
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* Luigi Rizzo - astobj2, console_video, windows build, chan_oss cleanup,
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and a bunch of infrastructure work (loader, new_cli, ...)
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* Brett Bryant - digit option for musiconhold selection, ENUMQUERY and
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ENUMRESULT functions, feature group configuration for
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features.conf, per-file CLI debug and verbose settings, TCP and
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TLS support for SIP, and various bug fixes.
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brettbryant(AT)gmail.com
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* Sergey Tamkovich - Realtime support for MusicOnHold, store and destroy
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realtime methods and implementations for odbc, sqlite, and pgsql
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realtime drivers, attended transfer updates, multiple speeds for
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ControlPlayback, and multiple bug fixes See
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http://voip-info.org/users/view/sergee serg(AT)voipsolutions.ru
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* Klaus Darillon - the SIPremoveHeader function in chan_sip and SIP Path
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Support.
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* Moises Silva (moy) - for writing LibOpenR2, and providing support for
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it in chan_dahdi moises.silva(AT)gmail.com
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* Eliel C. Sardanons - XML documentation implementation, and various
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other contributions eliels(AT)gmail.com
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* Sean Bright - Snom call pickup, newt interface for menuselect,
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cdr_tds rewrite, countless other improvements, fixes, and good
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ideas. sean(AT)malleable.com
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* Jan Kaláb - Calendaring support for Exchange Server 2007+ via
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Exchange Web Services.
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* University of Oslo (uio.no), Norway - SIP Max-Forwards setting
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support (developed by oej)
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* FCCN, Lissabon, Portugal - SIP show channels CLI command
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(developed by oej)
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* Viagenie, Canada - IPv6 support in socket layers and SIP
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implementation Developers: Marc Blanchet, Simon Perreault and
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Jean-Philippe Dionne
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* ClearIT AB, Sweden - res_mutestream, queue_exists and various other
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patches (developed by oej)
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* Despegar.com, Argentina - AstData API implementation, also sponsored
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by Google as part of the gsoc/2009 program (developed by Eliel)
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* Philippe Lindheimer - DEV_STATE additions to CCSS
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* Andrew "lathama" Latham <lathama at gmail dot com>
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Doxygen, HTTP-Static, Phoneprov, make update
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* George Joseph - PJSIP CLI commands, PJSIP_HEADER dialplan function
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=== OTHER CONTRIBUTIONS ===
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We'd like to thank the following for their listed contributions.
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* John Todd - Monkey sounds and associated teletorture prompt
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* Michael Jerris - bug marshaling
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* Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
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available under a Creative Commons License at
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http://www.asteriskdocs.org
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* Brian M. Clapper - poll.c emulation
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This product includes software developed by
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Brian M. Clapper <bmc(AT)clapper.org>
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=== HOLD MUSIC ===
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We'd like to thank the following for hold music
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* Music provided by www.opsound.org
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=== OTHER SOURCE CODE IN ASTERISK ===
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We'd like to thank the following for their code use
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* Asterisk uses libedit, the lightweight readline replacement from
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NetBSD.
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* The cdr_radius module uses libradiusclient-ng, which is also from
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NetBSD.
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* They are BSD-licensed and require the following statement:
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This product includes software developed by the NetBSD
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Foundation, Inc. and its contributors.
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* Digium did not implement the codecs in Asterisk.
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Here is the copyright on the GSM source:
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Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
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Technische Universitaet Berlin
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Any use of this software is permitted provided that this notice is not
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removed and that neither the authors nor the Technische Universitaet Berlin
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are deemed to have made any representations as to the suitability of this
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software for any purpose nor are held responsible for any defects of
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this software. THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
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As a matter of courtesy, the authors request to be informed about uses
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this software has found, about bugs in this software, and about any
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improvements that may be of general interest.
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Berlin, 28.11.1994
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Jutta Degener
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Carsten Bormann
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And the copyright on the ADPCM source:
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Copyright 1992 by Stichting Mathematisch Centrum, Amsterdam, The
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Netherlands.
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All Rights Reserved
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Permission to use, copy, modify, and distribute this software and its
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documentation for any purpose and without fee is hereby granted,
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provided that the above copyright notice appear in all copies and that
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both that copyright notice and this permission notice appear in
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supporting documentation, and that the names of Stichting Mathematisch
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Centrum or CWI not be used in advertising or publicity pertaining to
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distribution of the software without specific, written prior permission.
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STICHTING MATHEMATISCH CENTRUM DISCLAIMS ALL WARRANTIES WITH REGARD TO
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THIS SOFTWARE, INCLUDING ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND
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FITNESS, IN NO EVENT SHALL STICHTING MATHEMATISCH CENTRUM BE LIABLE
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FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
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WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
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ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT
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OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
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