unified socket application interface (for asterisk and maybe other apps)

[lcr.git] / README

Read the documentation at http://www.linux-call-router.de


Changes in Version 20021228 
- first release

Changes in Version 20030111 (buggy and unuseable)
- support dtmf for callback and dtmf dialing mode (dial through via dtmf)
- bug fixes
- dialing improvement: dialing h323 now possible with port and alias
- other stuff
- new Makefile: make install will now install binaries and data on your system

Changes in Version 20030118 (buggy and unuseable)
- information exchange between isdn/h323-ports, endpoints and calls are messages now
  previousely using direct calls with pointers were dangerous
- removed bug since 200301011, which caued h323 calls to deadlock

Changes in Version 20030120
- login function
- callback authentication
- h323 audio bug. no h323 audio transmission sice version 20030111
- some other bug fixes

Changes in Version 20030206 (first beta release 1.0)
- callerid (CLIP/COLP) is now processed correctly with all features
- hold sound and active/inactive notification
- CD notification
- many callerid display function
- bug fixes
- documentation as word document (partly done)
- Note: This week I have my vakation, so there is no response to any question in the mailing list.

Changes in Version 1.0 
- first release
- finished the first version of the documentation
- new style internet page with documentation in html
- all enities (port, endpoint, call) are now c++ objects, rather than structures
- tones may now be played and recorded using wave (8bit mono, 16bit mono, 16bit stereo) or law (alwa/ulaw mono)
- fixed corruption in wave-file creation
- now call forwad (cfb, cfnr) is implemented and working
- an answering machine with playback function is implemented.
- lots of bug fixes

Changes in Version 1.1
- option to fetch tones into memory, causing faster access, then from hard disk
- Memory is now locked using mlockall(), to prevent paging which causes jitter and interruption.
- Answering machine now works with keypad information.
- callback on internal port, if hangup with call on hold.
- A pocket calcularor with simple terms (*, /, +, -) and floating point is added.
- If a calls had more than two endpoints, any message has been ignored.
  This caused not to receive the retreive notification, which caused the hold
  music to continue to play.
- minor buf fixes
- Disconnect/release "causes" are now processed by priority, if a multipoint call is made.

Changes in Version 1.2
- bugfix: dialing of vbox-caller now works.
- bugfix: minor answering machine announcement bug
- fixed compiling error: h323_con.cpp (p_type not declared)  thanx arne!
- added include definition for kernel api in Makefile. Hope it works...
- fix: dummyid is used for external calls when no caller id is available. the
  dummy id is transmitted as restricted id. if the police is called, it will
  see the dummyid rather than the pbx line id. this is used on forwarded calls
  when the caller id is not available.
- doc: added a simple instruction to build a cross over cable.

Changes in Version 2.0pre
- NEW ISDN DRIVER SUPPORT (Forget the HiSax, now use 'mISDN'.)
- NEW KERNEL REALTIME MODULE (mISDN_dsp.o)
- NEW NT-MODE LIBRARY SUPPORT (now MULTIPOINT!!!)
- NEW TE-MODE STACK SUPPORT
- support of call suspension and retrieval (switch between calls)
- call waiting on internal phone (calls when no bchannel is available)
- doc: Now headlines are moved to the next page if they are at the bottom of a
  page.
- vbox: minor speech syntax bugfix
- up to 50 (compiler flag) dialed numbers can be recalled.
- up to 50 (compiler flag) received calls are listed and can be replied.
- Dialing informations are now queued by endpoint until port has received 
  setup acknowledge on the outgoing connection.
- Starting PBX without parameter gives a list of options.
- Query option for listing available ports/cards.
- CNIP (calling name) Some Simens switches and telephones support this.
- Extensions no have names.
- Timeouts can now be specified for different call states on ISDN phones.
- Tones/Announcements can now be overridden at different call states.
- isdn.cpp is completely reworked.
- Tones/Announcements can be played externally, if supported by the external
  line.
- Commandline parameters must be given on startup of pbx.
- query option to check out the current isdn cards and protocolls.
- Debug flags can now be used to speciallize the debug output.
- vbox: Recorded calls can now be sent as attached sound file via email, or
  just a notifaction mail without sound file can be sent.
- PBX now runs with highest prio (99) to get as much cpu as possible.
- CPU scheduler now runs PBX4Linux as real time process.
- Internal calls now use internal extension number as caller id.
- Rework of hold/conference.
- New option to allow an incoming h323-call to be connected at ringing state.
- New codecs supported with h323 (speex and law)
- COLP now works with h323
- Answering machine now delays to avoid the dtmf tone when start recording
- Answering machine now adds the beep behind the announcement file.
- keypad facility dialing option
- Conference now really works using call hold feature in conjunction with
  keypad feature
- Picking of an incoming call on isdn now really works.
- picking of a call forwarded to vbox
- I fixed a bug that did not queue the dialed digits correctly before getting
  an external connecting.
- Logfiles of calls now show the correct year.
- Callback now rejects if no password is given or the given extension doesn't
  exist.
- Incoming H.323 calls may now instantly connect using a special option.
- H323 compiles faster, because the H323 includes are only used for the H323
  code part.
- Using 'curses' for a state debugging. The current state of all instances
  is displayed on the screen including calls, endpoints and ports.
- manny, manny more things that I forgot

Changes in Version 2.0
- fixed memory leak
- fixed COLP bug
- finally SUSPEND/RESUME (call parking), resume is allowed from any ISDN-port
- some CRITICAL bugfixes
- Fixed and reworked sample counting when playing tones. Now fast wind and
  rewind works correctly. Also 8-bit recording can be played back now.
  I hope it works now without problems.
- An internal phone without caller id will now be rejected rather than
  treated as an extenal call.
- Fixed bug in rejecting an external call.
- Corrected handeling of 'inbandpatterns'.
- Data calls will not enable DTMF detection.
- Data calls will not use any audio transmission from user space.
- Forward to VBOX only if call is an audio call.
- Fixed library bug, that caused not to process keypad-information during
  setup message.
- Fixed a 'release_complete' bug.
- Debug information now have the correct month.
- Fixed bug of wave-playback of voicebox recoding, caused by rework of the
  audio routines. It caused a SEGMENTATION FAULT!
- Using threads now to send email and using libcrypto.
- Introducing encryption of external calls using Blowfish.
- Key exchange using RSA.
- Fixed a bug in dialing H323-IP with numerical digits.
- Fixed a bug that causes endpoint, which receives audio data, to crash when
  no port is related to it.
- Fixed a bug that did not release endpoint, when it receives a disconnect
  if it has no port (parked).
- Fixed a channel assignment bug when retrieving call. (second B-channel)
- Now COLP with H.323 works. No more crash!
- Park attribute was not set, which caused a crash.
- Conference now works correctly with dsp-module.
- Fixed a serious NT-mode process handling problem. (crash after some calls)
- Added log file which is also displayed on the 'state' screen.
- * Happy new year 2004 *

Changes in Version 2.1
- Fixed a bug that caused not to reply external calls (also VBOX).
- 'genrc' now supports loading HFC-4S, HFC-8S and HFC-E1 drivers
- Outgoing setup now expires after 8 seconds!
- Fixed a bug that causes mISDNuser to crash during cleanup.
- Fixed memory bug, thanx Paul!
- hfc-4s/8s driver support (mISDN)
- Improvement of isdn audio processing, hardware support.
- Fixed diplay callerid bug "anonymousunknown anon"
- Added more stable malloc (calloc) / free handling

Changes in Version 2.2
- PRI proof (2 MBit interface support when using HFC-E1)
- Fixed data call bug
- Improved display of PRI channels
- Now VBOX playback says "no messages" if the last message has been deleted
  and will not play the last but one, unless the "previous" button has been
  pressed.

Changes in Version 2.3
- Fixed HFC_MULTI driver activation problem (HW_RESET was not implemented)
- Fixed login prefix bug. Thanx Karsten V.
- MISDN: better layer 2 check
- Now facility informations are transfered during call to terminal
  (finally advice of charge is displayed) MUST BE ENABLED BY SETTINGS!
- Fixed 'reply' dialing bug, that caused a crash. Thanx Karsten V.
- Added L1 activation for NT-Mode. Fixed problems with inactive links.
- Fixed a bug that caused subsequent data calls after a data call. Thanx JC.
- mISDN: layer 1 now works correct with E1 cards.

Changes in Version 2.4
- Fixed parallel ringing to multiple external numbers.
- Fixed login again (was still buggy).

Changes in Version 2.5
- Fixed callback bug. (International numbers were not detected.)
- Fixed typos (mostly "incomming") - thanx Lars.
- Fixed vbox-email bug - thanx Martin. (and also the compiler error)
- Fixed compiler bug, that caused compiling without crypto lib to fail.
- Fixed some mISDN crash problems.
- Now it should also compile with the original CVS tree.
- Fixed hfc_multi unloading bug - thanx Karsten!
- Now disabling DSP_MODULE really causes DSP to be disabled.
- Now disabling real time scheduling really works.
- mISDNuser (CVS) now compiles with the mISDN (CVS)
- Adding the outdial prefix to the caller ID is now possible.
- Fixed bug that caused echo test not to work.
- And finally hardware echo now works on HFC 4s/8s/E1 (hfc_multi)
  For echo dial 993 (Test + 3) for standard configuration.
- Added new hfc_multi vendor IDs including "Beronet Cards".

Changes in Version 2.6
- Fixed hookflash bug in conjunction with prefix. Thanx Tobias!
- Fixed cleanup bug when loading of ISDN driver failed.
- Fixed mISDN bug that caused cards not to be found, if loaded in different
  order as found by kernel.
- Fixed a bug that causes a segfault when a phone disconnects while
  parallel ringing multiple phones/ports.
- Added capability for Point-To-Point in NT mode, including PRI.
- Added L1 link control for NT mode.
- Fixed bug in hfc_multi and mISDN driver that caused mISDN not to work
  with kernel > 2.6.7.
- Fixed a but when detecting different cards with hfc_multi.
- Fixed timer bug that caused timers of multiple NT ports not to work
  correctly.

Changes in Version 2.7
- Fixed lots of bugs.
- Now receive stream from mISDN is disabled when not needed.
- Added NT mode support for incomming "SETUP_ACKNOWLEDGE".

Changes in Version 3.0
- Advanced routing capability to replace the numbering_*.conf
  (Don't worry, internal and external numbering is a feature of the routing
  capability and is easy to convert.)
- Now correct cause location is generated and handled.
- New cause display feature. Location is displayed with the cause number.
- Many source cleanups.
- New interface (Unix socket) to administrate. Status informations are now
  viewable without restarting PBX. Even may processes may view status info.
  Starting / stopping state debugging, doesn't require to restart PBX.
- Status information now has selectable details.
- Better structure for debugging functions and better logging. (code)
- Dialing may also be done via command line interface.
- Now internel/external dialtone and ringing depends on internal/external call.
- Now endpoints (partys) can be released via command line (admin tool).
- Watchdog "pbxwatch" to automatically restart and even debug PBX4Linux.
- Removed problem with uninitialized variable in ISDNPort object causing to
  crash. It did not happen very often.(only after some hundred/thousand calls)
- HFC-E1 cards did not correctly synchronize to external lines.
- dsp.o now allocates only one timeslot per call, as expected.
- mISDNuser now correctly connects PRI calls.
- PRI improvements and bugfixes.
- Support for conference rooms.
- Voice box is now able to play announcement before connecting the call.
  A special feature on the external line is required to send audio before
  answering the call.
- It is now possible to include seconds (time) in the connect message. This
  might not be supported by all telephones, so it is an optional feature.
- Moved open and close of recording audio to the "Port" class, where it
  belongs. The mixer will be more performant this way.
- Notify is now supported by mISDN and also correctly handled and queued
  by PBX.
- Fixed bug that caused not to free broadcast process IDs in certain cases.
  This would cause calls to internal phones (from extern or intern) not to
  work after a while.
- Added HFC-S USB to 'genrc' tool.
- printisdn now shows corret month.

Changes in Version 3.0-fix1
- Rule for changing the forwarding now works. Enter "pbx rules forward" for
  description. Also the example in "defaults/route.conf" is corrected.
- Forking now forks twice and suppresses debug output. Closing of shell is
  possible.

Changes in Version 3.0-fix2
- Fixed memory leak bug in pbxadmin that caused to eat all memory and make it
  stop.
- Fixed audio handling that cause forking calls to be mute. (Parallel
  forwarding causes calls to fork to multiple destinations.)

Changes in Version 3.0-fix3
- Added "nopassword" parameter for login action.
- Fixed bolean condition bug.
- pbxadmin will not exit if terminal size changes.

Changes in Version 3.1
- Internal structure changed. "Endpoints" and "applications" are now two
  linked classes. The code is now reusable for other projects than
  "PBX4Linux". (No added features!)
- Some source cleanups.
- Now keypad must be enabled for each extension if required. (settings)
- Removed a new bug that caused remotely parked/holded calls not to be removed
  from conference. The conference got disturbed by park/hold sound from
  remote.
- Removed bug that caused printing of unset pointers.

Changes in Version 3.2
- PBX now works with mqueue branch. This is the latest CVS source:
  * HFCmulti is ported
  * HFC-PCI is ported
  * DSP is ported
  * nt-mode lib (libi4l) is ported
  * source is now SMP (multiple processors or hyperthreading) save.
- Fixed bug that caused not to record if annoucement is missing
- A prefix may be specified with callback for predefined dialing after
  callback.
- Now b-channels are displayed more compressed on admin tool.

Changes in Version 3.3
- * te-mode works
- * te-mode layer 1 and layer 2 control works (SHORTMESSAGE)

Changes in Version 3.3-fix2
- OpenH323 midas release compiles
- Fixed bug in MESSAGE_NOTIFY which cases display information not to dliver.
- OpenH323 midas release works currently only with law-codecs
- Dixed some dial string parsing for Openh323.

Changes in Version 3.3-fix3
- Rework of kernel audio briding. Much faster (less delay), dynamically
  handles jitter. Ready for future RTP / ISDNoIP modules.

Changes in Version 3.4
- Removed DSP_MODULE switch, because it will be essential for PBX operation.
- Fixed pbxadmin offset bug.
- Added special feature "efi" to announce caller's ID. You call, and it tells
  your caller ID. (if available) Sample set is not complete!
- Now caller ID and type can be given for external call rule.
- Now caller ID and type can be given for changing caller ID.
- Removed a display bug in pbxadmin, that caused busy channels to be omitted.
- Fixed layer 2 handling bug.
- Increased performance of pbx-status screen. Many interfaces/calls caused
  lock up of machine.
- Timeout condition seems to work now.
- Timeout action seems to work now.

New Verion for new name: LCR

Changes in Version 1.0
- Statefull b-channel open and closing
- Rebuild audio flow
  Made much simpler
  Preloading and keeping transmit buffer for seamless tones and patterns.
  Recording of what is actually transmitted and received by party.
- Logging is replaced by trace
- New isdn interface and port structure with many features
  Interfaces can be changed at runtime.
  Interfaces can be loaded and unloaded at runtime.
- mISDN stack fixes
- DDI in and out on all stacks
- Layer 1 over IP supports interconnection via IP
- Rebuild line and b-channel hunting with individual lists
- Screen lists for changing caller IDs
- Multiplexing calls to multiple extensions
- Removed all VoIP stuff to make core fast and stable (Use Asterisk for VoIP.)
- Fixed a bug that caused some isdn connections to hang during disconnect
- Many bug fixes
- Many minor improvements
- New bugs of course...
- Rename of 'Call' instances to 'Join', because they join parties together.
- A new remote interface for external applications is integrated
  -> Our first application is (-: *ASTERISK CHANNEL DRIVER* :-)