(CAPI*) chan_capi a Common ISDN API 2.0 implementation for Asterisk

 Copyright (C) 2005 Cytronics & Melware
 Armin Schindler <armin@melware.de>
 
 Reworked, but based on the work of
 Copyright (C) 2002-2005 Junghanns.NET GmbH
 Klaus-Peter Junghanns <kpj@junghanns.net>

This program is free software and may be modified and distributed under
the terms of the GNU Public License. There is _NO_ warranty for this!

Thanks go to the debuggers and bugfixers (listed in chronological order) :)
===========================================================================
Lele Forzani <lele@windmill.it>
Florian Overkamp <florian@obsimref.com>
Gareth Watts <gareth@omnipotent.net>
Jeff Noxon <jeff@planetfall.com>
Petr Michalek <petr.michalek@aca.cz>
Jan Stocker
(...and all the others that i forgot..) :-)

This chan_capi version includes:
================================

- multiple controller support
- CID,DNID (callling party, called party)
- CLIR/CLIP
- supplementary services, CD,HOLD,RETRIEVE,ECT
- DTMF (dependend on card) + software DTMF support
- early B3 connects (always,success,never)
- digital audio (what did you think?)
- incoming/outgoing calls
- overlap sending (dialtone)
- E(xplicit) C(all) T(ransfer) (...although it's done implicit .. but dont tell!)
- tuneable latency ;) you can configure the size of B3 blocks at compile time
  (in chan_capi_pvt.h, AST_CAPI_MAX_B3_BLOCK_SIZE)
  the default is 160 samples, for non-VoIP use you can tune it down to 130
- use asterisk's internal dsp functions for dtmf
- alaw support 
- ulaw support! 
- Eicon CAPI echo cancelation (echocancel=1)
- reject call waiting (ACO)
- DID for Point to Point mode (a.k.a overlap receiving)
- experimental echo squelching (echosquelch=1)
- call progress, no need to add ||r to your dialstring anymore
- rx/tx gains (rxgain=1.0)
- call deflection on circuitbusy (makefile option) (deflect=12345678)
- (inter)national dialing prefix (for callerid) configurable in capi.conf
- CLI command "capi info" shows B channel status
- capiECT will announce the callerID since it gets lost on most isdn pbxes
  the called party can press # to drop the call
- audio syncing (timing outgoing dataB3 on incoming dataB3), supposed to fix
  the DATA_B3_REQ (error = 0x1103) problem
- catch all MSN (incomingmsn=*)
- some configuration enhancements (msn=123,124,125 and controller=1,2,3,4)
- accountcode= added.
- finally the echo squelching works!
- callgroup support
- fixed pipe leak
- updated to support the new frame->delivery field
- compiles with latest cvs with a makefile option (LOOK AT THE MAKEFILE)
- fixed channel name bug in p2p mode
- added app_capiNoES for disabling the primitive echo suppressor, use this before
  you start recording voicemail or your files may get choppy
- fixed for latest cvs (AST_MUTEX_DEFINE_STATIC)
- fixed for latest cvs (asterisk/parking.h -> asterisk/features.h)
- fixed for latest cvs ast_pthread_create

- ATTENTION! the dialstring syntax now uses the zaptel dialstring syntax 
  it used to be:  Dial(CAPI/[@]<outgoingMSN>:[b|B]<destination>)
 
  now it is:      Dial(CAPI/g<group>/[b|B]<destination>)
  or:             Dial(CAPI/contr<controller>/[b|B]<destination>)
 
  CLIP/CLIR is now uses the calling presentation of the calling channel, this can
  be modified using the CallingPres() application. Use CallinPres(32) for CLIR.
  That is why the msn= param in capi.conf is now obsolete. The callerID is also
  taken from the calling channel.

- fixes for BSD (Jan Stocker)

Helper applications
===================
kapejod says: "No No No, dont use those yet....!" (except maybe HOLD,ECT...)

app_capiCD.c		forwards an unanswered call to another phone (does not rely on sservice CD)
			example:
			exten => s,1,Wait,1
			exten => s,2,capiCD,12345678
			
app_capiHOLD.c		puts an answered call on hold, this has nothing to do with asterisk's onhold thingie (music et al)
			after putting a call onhold, never use the Wait application!

app_capiRETRIEVE.c	gets the holded call back

app_capiECT.c		explicit call transfer of the holded call (must put call on hold first!)
			example:
			exten => s,1,Answer
			exten => s,2,capiHOLD
			exten => s,3,capiECT,55:50
			will ECT the call to 50 using 55 as the callerid/outgoing msn


Using CLIR
==========
Use the CallingPres() application before you dial:
exten => _X.,1,CallingPres(32)
exten => _X.,2,Dial(CAPI/contr1/${EXTEN})    

Enjoying early B3 connects (inband call progress, tones and announcements)
==========================================================================
early B3 is now configurable in the dialstring :)
if you prefix the destination number with a 'b' early B3 will always be used, also if the call fails
because the number is unprovisioned, etc ...
if you prefix it with a 'B' early B3 will only be used on successful calls, giving you ring indication,etc...

dont use indications in the Dial command, your local exchange will do that for you:
exten => _X.,1,Dial(CAPI/contr1/B${EXTEN},30)		(early B3 on success)
exten => _X.,1,Dial(CAPI/contr1/b${EXTEN},30)		(always early B3)
exten => _X.,1,Dial(CAPI/contr1/${EXTEN},30,r)		(no early B3, fake ring indication)

exten => _X.,1,Dial(CAPI/contr1/b${EXTEN},30,r)		(always early B3, fake indicatons if the exchange
							does not give us indications)
exten => _X.,1,Dial(CAPI/contr1/B${EXTEN},30,r)		(early B3 on success, fake indicatons if the exchange
							does not give us indications)
    
For normal PBX usage you would use the "b" option, always early B3.

Overlap sending (a.k.a. real dialtone)
======================================
when you dial an empty number, and have early B3 enabled, with:
    Dial(CAPI/g1/b)
the channel will come up at once and give you the dialtone it gets from the local exchange.
at this point the channel is like a legacy phone, now you can send dtmf digits to dial.    

Example context for incoming calls on MSN 12345678:
===================================================

[capi-in]
exten => 12345678,1,Dial(SIP/phone1)
exten => 12345678,2,Hangup


More information/documentation and commercial support can be found at:
	http://www.junghanns.net/asterisk/