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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: https://origsvn.digium.com/svn/asterisk/trunk@313744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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288 lines
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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 development
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John Todd, TalkPlus, Inc. and JR Richardson, Ntegrated Solutions. - for funding
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the development of SIP Session Timers support.
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Omnitor AB, Gunnar Hellström, for funding work with videocaps, T.140 RED,
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originate with video/text and many more contributions.
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=== WISHLIST CONTRIBUTERS ===
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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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* Thanks to QuickNet Technologies for their donation of an Internet
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PhoneJack and Linejack card to the project. (http://www.quicknet.net)
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* Thanks to VoipSupply for their donation of Sipura ATAs to the project for
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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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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 and bug
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fixes.
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russell(AT)digium.com
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Anthony Minessale II - Countless big and small fixes, and relentless forward
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push. ChanSpy, ForkCDR, ControlPlayback, While/EndWhile, DumpChan, Dictate,
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MacroIf, ExecIf, ExecIfTime, RetryDial, MixMonitor applications; many
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realtime concepts and implementation pieces, including res_config_odbc;
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format_slin; cdr_custom; several features in Dial including L(), G() and
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enhancements to M() and D(); several CDR enhancements including CDR
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variables; attended transfer; one touch record; native MOH; manager
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eventmask; command line '-t' flag to allow recording/voicemail on nfs
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shares; #exec command and multiline comments in config files; setvar in iax
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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 support.
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You can find him and asterisk-perl at 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 Islands
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jd-girard(AT)esoft.pf http://www.esoft.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 recording
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critch(AT)basesys.com
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Jefferson Noxon - app_lookupcidname, app_db, and various other 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 file
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format, and various fixes. Can be contacted at mahmut(AT)oa.com.au
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James Dennis - Cisco SIP compatibility patches to work with SIP service
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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* functions;
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func_odbc, cdr_adaptive_odbc, and other innumerable bug fixes.
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tilghman(AT)digium.com 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 10.3,
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dialplan include verification, route lookup on OpenBSD, SNMP agent
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support (res_snmp), various other bugs. tholo(AT)sigmasoft.com
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Josh Roberson - chan_zap reload support, Advanced Voicemail Features, & other
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misc. patches. - josh(AT)asteriasgi.com, http://www.asteriasgi.com
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William Waites - syslog support, SIP NAT traversal for SIP-UA. 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, testing,
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SIP outbound proxy support, Manager 1.1 update, SIP transfer support,
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SIP presence support, SIP call state updates (dialog-info),
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QUEUE_EXISTS function, device state provider architecture,
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multiparking (together with mvanbaak), meetme and parking device states,
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MiniVM - the small voicemail system, many documentation
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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 Architecture
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Steve Murphy - privacy support, $[ ] parser upgrade, AEL2 parser upgrade.
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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 COSMOS Software Enterprises, Ltd.
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- for Variable for No Answer Timeout for Attended Transfer
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Slav Klenov & Vanheuverzwijn Joachim - development of the generic jitterbuffer
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Securax Ltd. info(AT)securax.be
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Roy Sigurd Karlsbakk - providing funding for generic jitterbuffer development
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roy(AT)karlsbakk.net, Briiz Telecom AS
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Voop AS, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - funding for rewrite
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of SIP transfers
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Philippe Sultan - RADIUS CDR module, many fixes to res_jabber and gtalk/jingle
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channel drivers.
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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 InAccess
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Networks (work funded by HOL, www.hol.gr) gkon(AT)inaccessnetworks.com
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Daniel Nylander - Support for Swedish and Norwegian languages in voicemail.
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http://www.danielnylander.se/
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Stojan Sljivic - An option for maximum number of messsages per mailbox in
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voicemail. Also an issue with voicemail synchronization has been fixed.
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GDS Partners 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 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 and the IAX
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trunk MTU patch
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Cedric Hans - Development of chan_unistim
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cedric.hans(AT)mlkj.net
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Takao Takahashi & Mina Naguib - chan_unistim improvements for smaller devices
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Sergio Fadda - console_video: video support for chan_oss and 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 ENUMRESULT functions,
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feature group configuration for features.conf, per-file CLI debug and verbose settings,
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TCP and 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 realtime methods and
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implementations for odbc, sqlite, and pgsql realtime drivers, attended transfer updates,
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multiple speeds for ControlPlayback, and multiple bug fixes
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- See http://voip-info.org/users/view/sergee
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serg(AT)voipsolutions.ru
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Klaus Darillon - the SIPremoveHeader function in chan_sip
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Moises Silva (moy) - for writing LibOpenR2, and providing support for it in chan_dahdi
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moises.silva(AT)gmail.com
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Eliel C. Sardanons - XML documentation implementation, and various other contributions
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eliels(AT)gmail.com
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Sean Bright - Snom call pickup, newt interface for menuselect, cdr_tds rewrite,
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countless other improvements, fixes, and good ideas.
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sean(AT)malleable.com
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Jan Kaláb - Calendaring support for Exchange Server 2007+ via Exchange Web Services.
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University of Oslo (uio.no), Norway - SIP Max-Forwards setting support (developed by oej)
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FCCN, Lissabon, Portugal - SIP show channels CLI command (developed by oej)
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Viagenie, Canada - IPv6 support in socket layers and SIP implementation
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Developers: Marc Blanchet, Simon Perreault and Jean-Philippe Dionne
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ClearIT AB, Sweden - res_mutestream, queue_exists and various other patches (developed by oej)
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Despegar.com, Argentina - AstData API implementation, also sponsored by Google as part of the
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gsoc/2009 program (developed by Eliel)
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Philippe Lindheimer - DEV_STATE additions to CCSS
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=== OTHER 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 http://www.asteriskdocs.org
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Brian M. Clapper - poll.c emulation
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This product includes software developed by Brian M. Clapper <bmc(AT)clapper.org>
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=== 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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Asterisk uses libedit, the lightweight readline replacement from NetBSD.
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The cdr_radius module uses libradiusclient-ng, which is also from 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. Here is the copyright on the
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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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