dect
/
asterisk
Archived
13
0
Fork 0
This repository has been archived on 2022-02-17. You can view files and clone it, but cannot push or open issues or pull requests.
asterisk/CREDITS

288 lines
11 KiB
Plaintext
Raw Normal View History

=== 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
Merge the changes from issue #10665 from the team/group/sip_session_timers branch. This set of changes introduces SIP session timers support (RFC 4028). In short, this prevents stuck SIP sessions that were not properly torn down due to network or endpoint failures during an established SIP session. To quote some of the documentation supplied with the patch: "The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE request at a negotiated interval. If a session refresh fails then all the entities that support Session- Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path that do not support Session-Timers)." (closes issue #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by rjain (license 226) chan_sip.c.diff uploaded by rjain (license 226) sip.conf.sample.diff uploaded by rjain (license 226) proc_422_rsp_comment.diff uploaded by rjain (license 226) chan_sip.c.cache.diff uploaded by rjain (license 226) chan_sip.memalloc uploaded by rjain (license 226) chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches tracked in team/group/sip_session_timers, with some additional fixes by russell and oej. Tested by: jtodd, rjain, loloski git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98978 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-16 21:53:10 +00:00
John Todd, TalkPlus, Inc. and JR Richardson, Ntegrated Solutions. - for funding
the development of SIP Session Timers support.
Omnitor AB, Gunnar Hellstr<74>m, for funding work with videocaps, T.140 RED,
originate with video/text and many more contributions.
=== WISHLIST CONTRIBUTERS ===
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)
=== 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 Kal<61>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)
Add Device State Information CCSS for Generic Devices. Add Asterisk Device State information and callbacks to the Call Completion Supplemental Services for generic agents. There are currently not many devices that have native support for CCSS. Even as the devices become available there may be other reasons why one may choose to not take advantage of the native abilities and stick with the generic implementation. The generic implementation is quite capable and could be greatly enhanced by adding device state capabilities. A phone could then subscribe to the device state with a BLF key in conjunction with Asterisk hints. The advantages of the device state information would allow a single button to: request CCSS, cancel a CCSS request, and display the current state of a CCSS request. For example, you may have a single button that when not lit, there is no active CCSS request. When you press that button, the dialplan can query the DEVICE_STATE() associated with that caller to determine whether they should be calling CallCompletionRequest() or CallCompletionCancel(). If there is currently a pending request, then the dialplan would cancel it. This also has the advantage of showing the true state of a request, which is an asynchronous call, even when CallCompletionRequest() thinks it was successful. The actual request could ultimately fail. Once lit, further feedback can be provided to the caller about the current state of their request since it will be updated by the CCSS State Machine as appropriate. The DEVICE_STATE mapping is configurable since the BLF being used on a given phone type may vary. The idea is to allow some level of customization as to the phone's behavior. As an example, you may want the BLF key to go solid once you have requested a callback. You may then want the LED to blink (typically ringing) when either the callback is in process, which is a visual indication that the incoming call is the desired callback. You may want it to blink when the callee is ready but you are busy, giving you a visual indication that the target is available as you may want to get off the line so that the callback can be successful. Device state information is sent back via the ast_devstate_prov_add() callback for any generic CCSS device as it traverses through the state machine. You simply provide a map between CC_STATE values and the corresponding AST_DEVICE state values. You could then generate hints against these states similar to what is possible today with Custom Devstates or MeetMe states. For example, you may have an extension 3000 that is currently associated with device SIP/3000. You could then create a feature code for that extension that may look something like: exten => *823000,hint,ccss:sip/3000 You would then subscribe a BLF button to *823000 which would point to the dialplan that handled CCSS requests/cancels using the available DEVICE_STATE() information about ccss:sip/3000 to make the decision about what to do. (closes issue #18788) Reported by: p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p lindheimer (license 558) Modified with final reviewboard comments. Tested by: p_lindheimer, loloski Review: https://reviewboard.asterisk.org/r/1105/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@313744 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-14 18:22:35 +00:00
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>
=== 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
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.