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MelwareDE 83ba42f167 Asterisk 1.8 calling party name 2010-09-14 22:05:40 +00:00
divastreaming Add Diva streaming for resource PLCI 2010-08-23 15:21:52 +00:00
libcapi20 Use Diva QSIG CAPI extensions for processing of Calling Party Name 2010-09-14 19:54:25 +00:00
CHANGES Use Diva QSIG CAPI extensions for processing of Calling Party Name 2010-09-14 19:54:25 +00:00
INSTALL Reviewed by doc control 2009-03-25 19:38:16 +00:00
LICENSE Initial import of new chan_capi. 2005-06-02 18:47:35 +00:00
Makefile Clan up implementation of Diva streaming 2010-06-30 08:33:41 +00:00
README Added capicommand(keypad) 2010-07-23 08:08:40 +00:00
README.Diva.fax Restore behavior of 'receivefax' to one provided by version 628 of chan_capi.c: Use fax T.30 extended for Diva hardware or if required by 'receivefax' option only. By default allow fine resolution for fax T.30 extended. 2009-08-06 08:49:25 +00:00
README.Diva.qsig Processed by doc control 2009-04-03 09:11:18 +00:00
README.Diva.streaming Description of Diva streaming 2010-06-30 11:40:00 +00:00
README.media Added chat option 'h' to auto hangup a caller if alone too long in conference. 2009-08-03 10:40:03 +00:00
README.qsig Reviewed by doc control 2009-08-03 16:08:23 +00:00
capi.conf Add echocancelpath configuration option 2009-04-15 13:39:57 +00:00
chan_capi.c Use 'connected' instead of 'caller' name for outgoing calls 2010-09-14 20:50:43 +00:00
chan_capi.h Use Diva QSIG CAPI extensions for processing of Calling Party Name 2010-09-14 19:54:25 +00:00
chan_capi20.h Added header file capiutils.h 2005-09-20 18:33:40 +00:00
chan_capi_chat.c Customer reported unexpected behavior after incoming call with redirection number information element. The problem is caused by destruction of internal structures of libc and arrives not on every system but 2010-08-19 08:44:21 +00:00
chan_capi_chat.h Add chat_command. Add r/l/o/a chat commands to remove users/listeners/operatirs/all users from conference 2009-07-16 14:16:13 +00:00
chan_capi_command.c Customer reported unexpected behavior after incoming call with redirection number information element. The problem is caused by destruction of internal structures of libc and arrives not on every system but 2010-08-19 08:44:21 +00:00
chan_capi_command.h Use received from IP tone events for VC command 2009-05-18 22:37:21 +00:00
chan_capi_platform.h Typo 2009-03-12 17:02:22 +00:00
chan_capi_qsig.h Customer reported unexpected behavior after incoming call with redirection number information element. The problem is caused by destruction of internal structures of libc and arrives not on every system but 2010-08-19 08:44:21 +00:00
chan_capi_qsig_asn197ade.c Customer reported unexpected behavior after incoming call with redirection number information element. The problem is caused by destruction of internal structures of libc and arrives not on every system but 2010-08-19 08:44:21 +00:00
chan_capi_qsig_asn197ade.h updated copyright 2008-02-05 11:16:25 +00:00
chan_capi_qsig_asn197no.c Runtime optimization 2009-03-12 15:56:20 +00:00
chan_capi_qsig_asn197no.h updated copyright 2008-02-05 11:16:25 +00:00
chan_capi_qsig_core.c Asterisk 1.8 calling party name 2010-09-14 22:05:40 +00:00
chan_capi_qsig_ecma.c Customer reported unexpected behavior after incoming call with redirection number information element. The problem is caused by destruction of internal structures of libc and arrives not on every system but 2010-08-19 08:44:21 +00:00
chan_capi_qsig_ecma.h Asterissk 1.8 2010-08-13 21:36:12 +00:00
chan_capi_rtp.c asterisk trunk API adaptions 2010-03-01 17:59:31 +00:00
chan_capi_rtp.h Update copyright years. 2009-01-17 17:35:55 +00:00
chan_capi_supplementary.c From CAPI thread locks are taken in the wrong order. The right ordr is ast_channel_lock followed by i->lock. But from CAPI thread 2010-09-12 08:52:18 +00:00
chan_capi_supplementary.h From CAPI thread locks are taken in the wrong order. The right ordr is ast_channel_lock followed by i->lock. But from CAPI thread 2010-09-12 08:52:18 +00:00
chan_capi_utils.c Asterisk 1.8 calling party name 2010-09-14 22:05:40 +00:00
chan_capi_utils.h Asterisk 1.8 calling party name 2010-09-14 22:05:40 +00:00
create_config.sh Compatibility with Asterisk 1.8 2010-08-13 20:31:35 +00:00
dlist.c source code is not executable 2009-02-14 20:18:08 +00:00
dlist.h source code is not executable 2009-02-14 20:18:08 +00:00
xlaw.c - more restructuring 2007-04-29 22:28:30 +00:00
xlaw.h - more restructuring 2007-04-29 22:28:30 +00:00


chan_capi a Common ISDN API 2.0 implementation for Asterisk

 Copyright (C) 2005-2010 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, bugfixers and contributors :)
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
Frank Sautter, levigo group
Hans Petter Selasky
Simon Peter <dn.tlp@gmx.net>
Mario Goegel <m.goegel@gmx.de>

(...and all the others that have been forgotten...) :-)

No support for Asterisk 1.0.x any more, you need at least
Asterisk 1.2.x , 1.4.x or 1.6.x.

Eicon DIVA Server cards are now named Dialogic(R) Diva(R) Media Boards.

This chan_capi version includes:
- Multiple controller support
- CID,DNID (callling party, called party)
- Supplementary services: CD (deflect/reroute),HOLD,RETRIEVE,ECT,3PTY
- 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 and additional digits)
- E(xplicit) C(all) T(ransfer) (...although it's done implicit-but don't tell!)
- Use asterisks internal DSP functions for DTMF
- a-Law support 
- u-Law support! 
- Dialogic(R) Diva(R) software DSP echo cancellation (echocancel=1)
- Reject call waiting (ACO)
- DID for Point to Point mode (a.k.a overlap receiving)
- Rx/Tx gains using positive linar value (rxgain=1.0, txgain=1.0 means no change) 
- (Inter)national/subscriber dialing prefix (for callerid) configurable in capi.conf
- CLI command "capi info" shows B channel status of chan_capi
- Catch all MSN (incomingmsn=*)
- Some configuration enhancements (msn=123,124,125)
- Added accountcode= 
- Echo squelching (echosquelch=1)
- Callgroup support
- report correct DIALSTATUS and HANGUPCAUSE.
- Updated to support the new frame->delivery field
- Compiles with different Asterisk versions (automatic build configuration)
- receive/send faxes for CAPI (see below)
- Fixes and compatibility for BSD (Jan Stocker and Hans Petter Selasky)
- Support 'type of number'.
- ISDN hold.
- CAPI Line Interconnect.
- CLI command "capi show channels" shows details on channel status.
- Asterisk 1.4 jitterbuffer configuration.
- some QSIG extensions (see README.qsig)
- CCBS (call completion on busy subscriber)
- CAPI CHAT (CAPI MeetMe using onboard DSPs)
- KEYPAD digits detection
- provide DATE/TIME in NT-mode.

The Dial string

  Example:        Dial(CAPI/g<group>/[<callerid>:]<destination>[/<params>])
  Or:             Dial(CAPI/contr<controller>/[<callerid>:]<destination>[/<params>])
  Or:             Dial(CAPI/<interface-name>/[<callerid>:]<destination>[/<params>])

  'group' can be a value, comma separated list of values or a range
  using '-'. The according interface is found by searching a match with
  the 'group' specified in the capi.conf for each interface.

  The optional <callerid> followed by an ':' can be used to set a caller ID
  for this dial() command, without changing the original channel's caller ID.

  'params' is an optional part to set special settings for this call.
  The string consists of a list of characters with the following meaning:
   'b' : early B3 always.
   'B' : early B3 on successful calls only.
   'd' : use the default caller ID that is set by defaultcid= in capi.conf
   'o' : use overlap sending of number.
         (Useful if additional digits should be send afterwards or together
          with 'b' to get the dialtone and then send the number, e.g., if 
          no progress tones are available)
   's' : activate 'stay-online': do not disconnect CAPI connection on hangup.
         This is needed to give additional commands like CCBS after hangup.
         To really hang up the CAPI connection, use either capicommand(hangup)
         or wait for chan_capi/network timeout (about 20 seconds).
   'G' : early Line-Interconnect / bridge: Use Line-Interconnect as soon as
         both B-channels are up. Both channels must be of type the CAPI and
         the incoming call may need 'capicommand(progress)' to enable Early B3
         on it as well as the dial option 'b' for the outgoing channel.
         Before Dial(), the capicommand(peerlink) must be used to signal the
         dialed channel its peer.

  If the <interface-name> is used in dialstring, be sure the name (specified
  in capi.conf) does not start with 'contr' or 'g'.

  CLIP/CLIR uses the calling presentation of the calling channel, which can
  be modified using the SetCallerPres() application. Use
  SetCallerPres(prohib_not_screened) for CLIR. That is why the msn= param in
  capi.conf has been removed. The callerID is also taken from the calling channel.

CLI commands

capi info:
    Show chan_capi version info.
    Show status of available B-channels.

capi debug:
    Enable CAPI message verbosity.

capi no debug:
    Disable CAPI message verbosity.

capi show channels:
    Display detailed information on CAPI B-channels.
    (Description see below)

capi chatinfo:
    Show status of CAPI CHAT.

CAPI command application
chan_capi provides an additional Asterisk application
With this application, special CAPI commands and features can be used.

Call Deflection:
    Forwards an unanswered call to another number.
        exten => s,1,capicommand(deflect|12345678)
Fax receive:
    Receives a fax using CAPI.
        exten => s,1,capicommand(receivefax|/tmp/${UNIQUEID}|+49 6137 555123|Asterisk|kX)
	(more see below)

Fax send:
    Sends a fax using CAPI.
        exten => s,1,capicommand(sendfax|/path/to/faxfile.sff|+49 6137 555123|Asterisk)
	(more see below)

Enable/Disable echosquelch:
    Enables or disable a very primitive echo suppressor.
    Disable this option before you start recording voicemail or your files may get choppy.
        exten => s,1,capicommand(echosquelch|yes)
        exten => s,1,capicommand(echosquelch|no)

Enable/Disable echocancel:
    Enables or disables echo-cancel provided by CAPI driver/hardware.
    You might need to disable echo-cancel when the data/fax transmission is handled
    by a non-CAPI application. After hangup, this setting is restored to the value
    set in capi.conf.
        exten => s,1,capicommand(echocancel|yes)
        exten => s,1,capicommand(echocancel|no)

    By default chan_capi deactivates echo-cancel for transit connections
    (Dial command called from existing call). You can use 'econtransitconn'
    configurations parameter (capi.conf) to change this behavior:
    econtransitconn=0 ; Deactivate EC on transit connections
    econtransitconn=1 ; Deactivate EC on secondary call of transit connection only
    econtransitconn=2 ; Deactivate EC on primary call of transit connection only
    econtransitconn=3 ; Do not deactivate EC on transit connections

Send Keypad Facility:
    Send Keypad Faciliaty information.
        exten => s,1,capicommand(keypad|<digits>)

Malicious Call Identification:
    Reports a call of malicious nature.
        exten => s,1,capicommand(malicious)

    Puts an answered call on hold, this has nothing to do with Asterisk's onhold
    (music et al).
    An optional parameter is the name of the variable, which should be set with
    the reference ID of the call on hold.
        exten => s,1,capicommand(hold)
        exten => s,1,capicommand(hold|MYHOLDVAR)

    Sets the type of 'hold'. When Asterisk wants to put the call on hold, the specified method
    will be used.
        exten => s,1,capicommand(holdtype|local)  ;no hold, Asterisk can play MOH
        exten => s,1,capicommand(holdtype|hold)   ;ISDN-HOLD 
        ; not yet implemented
        exten => s,1,capicommand(holdtype|notify) ;notify the peer only, Asterisk can play MOH

    Gets back the holded call. An optional parameter is the reference ID of the call
    on hold.
        exten => s,1,capicommand(retrieve)
        exten => s,1,capicommand(retrieve|${MYHOLDVAR})

    Explicit Call Transfer of the call on hold (must put call on hold first!)
        exten => s,1,capicommand(ect|${MYHOLDVAR})
        exten => s,1,capicommand(ect)
        exten => s,1,capicommand(hold)
        exten => s,2,Wait(1)
        exten => s,3,Dial(CAPI/contr1/1234,60,M(capiect))
    Note: Normaly a PBX needs 'implicit call transfer', which is done by default
    with this command. But if the line needs real 'explicit call transfer', use
        exten => s,1,capicommand(ect|x)

    Initiates a Three-Party Conference (must have one call on hold and one active call!).
        exten => s,1,capicommand(3pty_begin|${MYHOLDVAR})
        exten => s,1,capicommand(3pty_begin)
        exten => s,1,capicommand(hold)
        exten => s,2,Dial(CAPI/contr1/1234,,M(capi3pty))

Peer link creation:
    Creates a reference for chan_capi to know who is the calling channel on Dial().
    This is needed if you want to use CCBS/CCNR afterwards.
        exten => s,1,capicommand(peerlink)

Hangup in mode 'stay-online':
    After hangup in 'stay-online' mode, the line is not really disconnected
    until timeout or command:
        exten => s,1,capicommand(hangup)
    This works after capicommand(peerlink) only.

Set local party to 'busy' or 'free':
    Sets the local phone to the status 'busy' or 'free' when
    awaiting a callback for CCBS/CCNR. If the network wants to
    call you back for CCBS/CCNR, chan_capi normaly does not know
    about the status of the extension who started the callback.
    By default, chan_capi assumes 'free', but you can change that
        exten => s,1,capicommand(ccpartybusy|${CCLINKAGEID}|yes)
        exten => s,1,capicommand(ccpartybusy|${CCLINKAGEID}|no)

Call completion on subscriber busy (CCBS):
    To receive a callback when the dialed and busy party becomes free, aka
    call completion on subscriber busy, you can do the following:
        exten => s,1,capicommand(peerlink)  ;to let chan_capi know who is the calling channel.
        exten => s,2,Dial(CAPI/contr1/123456,60,g)  ;'g' to go-on with the dialplan on busy.
        exten => s,3,NoOp(${CCLINKAGEID})  ;if this variable now contains a number, CCBS is possible.
            ;here you can ask the caller if CCBS shall be activated...
        exten => s,4,capicommand(ccbs|${CCLINKAGEID}|<context>|<exten>|<priority>)
        exten => s,5,NoOp(${CCBSSTATUS})  ;if CCBS was successfully enabled, it is set to "ACTIVATED".
    If the remote party becomes 'non-busy', the network initiates the callback that will be
    sent to the provided context/exten/priority. Of course, this only happens if your local
    phone is set to 'free' with capicommand(ccpartybusy), which is the default.
    In this context/exten/priority you should just setup a callfile to initiate an outgoing
    call from your extension to
        exten => s,1,Dial(CAPI/ccbs/${CCLINKAGEID}/)

Deactivate CCBS:
    To deactivate a previously activated CCBS, use the following command:
        exten => s,1,capicommand(ccbsstop|${CCLINKAGEID})

Chat (MeetMe/Conference):
    See also README.media for details!
    If the CAPI card/driver supports it, the caller can be put into a chat-room:
    (This uses the DSPs onboard a Dialogic(R) Diva(R) Rev.2 Media Board.)
        exten => s,1,capicommand(chat|<roomname>|<options>|controller)
            exten => s,1,capicommand(chat|salesmeeting|m|1,3-6)
    Possible options:
    'm' = The first caller will get music-on-hold until second caller arrives.
    'h<sec>' = Hangup after <sec> seconds if caller is alone in conference.
    'o' = The caller is operator
    'l' = The caller is listener

Progress / Early-B3 on incoming calls:
    Activate Early-B3 on incoming channels to signal progress tones
    when in NT-mode or if the Telco-line supports this.
        exten => s,1,capicommand(progress)

Get CAPI application ID:
    To store the CAPI application ID in an Asterisk dialplan variable, use:
            exten => s,1,capicommand(getid,CAPI_ID)
            exten => s,2,NoOp(CAPI appl-id is ${CAPI_ID})

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

Enjoying Early B3 connects (inband call progress, tones and announcements)
Early B3 is configurable in the dialstring parameters.
If you set a 'b', early B3 will always be used, also if the call fails,
because the number is unprovisioned, etc ...
If you set a 'B', early B3 will only be used on successful calls, 
giving you ring indication,etc...

Do not use indications in the Dial command, your local exchange will do that for 
	exten => _X.,1,Dial(CAPI/contr1/${EXTEN}/B,30)		
		(early B3 on success)
	exten => _X.,1,Dial(CAPI/contr1/${EXTEN}/b,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/${EXTEN}/b,30,r)		
		(always early B3, fake indicatons if the exchange does not give us 
	exten => _X.,1,Dial(CAPI/contr1/${EXTEN}/B,30,r)		
		(early B3 on success, fake indicatons if the exchange does not give us 
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:
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:

exten => 12345678,1,Dial(SIP/phone1)
exten => 12345678,2,Hangup

Short HOWTO of capicommand(receivefax...) and capicommand(sendfax...):
For those of you who have a CAPI card with an on-board DSP (like Dialogic(R) Diva(R) Media Boards),
this allows you to receive/send faxes.

If you want to answer a channel in fax mode, use capicommand(receivefax|...)
instead of Answer(). If you use Answer(), you will be in voice mode. 
If the hardware DSP detects fax tone, you can switch from voice to fax mode 
by calling capicommand(receivefax|...).
The parameter <filename> is mandatory and the parameters <stationid>,
<headline> and <options> are optional.
By default, if fax reception was not successful, the file is deleted. If you want even
partly received or broken fax files, use 'k' for "keep bad fax" in the <options>:
You can activate extended fax support by 'X' in the <options>:
Please read details on extended fax options in README.Diva.fax.

To enable fax tone detection and redirect to extension 'fax', use config variable
'faxdetect' in capi.conf. This can be limited to the first 'n' seconds of a connection
using the setting 'faxdetecttime' in capi.conf.

Example of use :
line number 123, play something, if a fax tone is detected, handle it
line number 124, answer directly in fax mode

exten => 123,1,Answer()
exten => 123,2,BackGround(jpop)
exten => 124,1,Goto(handle_fax,s,1)
exten => fax,1,Goto(handle_fax,s,1)

exten => s,1,capicommand(receivefax|/tmp/${UNIQUEID}[|<stationid>|<headline>|<options>])
exten => s,2,Hangup()
exten => h,1,deadagi,fax.php // Run sfftobmp and mail it.

The output of capicommand(receivefax|...) is a SFF file.
Use sfftobmp to convert it.
With a Dialogic(R) Diva(R) Media Board, the following features are provided:
 - fax up to 33600
 - high resolution
 - Color Fax 
 - JPEG Compression is disabled (not tested yet)
See README.Diva.fax for more details and description of all options.

To send a fax, you can use the same mechanism like with receivefax.
Just replace the <filename> with the path to the .SFF file to send.
You can activate extended fax support by 'X' in the <options>:
Please read details on extended fax options in README.Diva.fax.

After disconnect of a fax connection, the following variables
will be set for that channel:
FAXSTATUS     : 0 = OK, 1 = Error.
FAXREASON     : Value of B3 disconnect reason.
FAXREASONTEXT : Decoded text of FAXREASON value.
FAXRATE       : The baud rate of the fax connection.
FAXRESOLUTION : 0 = standard, 1 = high.
FAXFORMAT     : 0 = SFF.
FAXPAGES      : Number of pages received.
FAXID         : The ID of the remote fax maschine.

KEYPAD digits in NT-mode
If the device connected to a NT-mode port sends KEYPAD digits
instead of normal digits, this call is then send to extension
'Kxxx'. Where 'xxx' stands for the KEYPAD digits sent.

CLI command "capi show channels"
This CLI command shows detailed info on all CAPI channels.
Column description:
  Line-Name : the name of the interface as defined in capi.conf
  NTmode    : is the line in NT-mode instead fo TE-mode
  state     : the state of the channel, like 'Conn', 'Disc', 'Dial', ...
  i/o       : incoming or outgoing line
  bproto    : protocol on CAPI ('fax', 'trans' or 'rtp')
  isdnstate : a string which may consists of the following characters
              * = PBX is active
              G = Line-Interconnect (CAPI bridge) active
              B = B-channel is up
              b = B-channel is requested
              P = Progress was signaled
              H = this line is on hold
              T = this line is in transfer (ECT) mode
              S = SETUP[_ACK] was signaled
  ton       : type of number value
  number    : the caller number and destination number

Asterisk variables used/set by chan_capi

    On incomming calls, this variable is set with the B-channel information value:
     '0' : B-channel is used (default)
     '1' : D-channel is used (not implemented yet)
     '2' : neither B nor D-channel is used (e.g., call waiting)
    Call-Waiting: an incoming call with BCHANNELINFO not '0' cannot be accepted.
    Another connection must be dropped before accepting or use
    capicommand(deflect|<number>) to initiate call deflection to another destination.

    Incoming calls: 'type of number' value of the called number is saved in this variable on
    incomming call.
    Outgoing calls: Allows to specity the 'ETS 300 102-1' called party number octet 3
                    (0x80 is used by default)

    exten => _X.,1,Answer
    exten => _X.,n,Set(_CALLEDTON=${CALLEDTON}) ; Use value of incoming call for outgoing call
    exten => _X.,n,Dial(CAPI/ISDN3/100,10)

    exten => _X.,1,Answer
    exten => _X.,n,Set(_CALLEDTON=1) ; Use new value
    exten => _X.,n,Dial(CAPI/ISDN3/100,10)

    The 'type of number' value to overwrite for the caller number on outgoing call.

    If a call is put on hold (ISDN-HOLD), the reference ID is saved in this variable.
    This variable is inherited as CALLERHOLDID to the dialed channel and will be used
    if e.g., capicommand(ect) is used to transfer the held call.

    If set on dial(), the calling subaddress will be set to the content.

    If set on dial(), the called subaddress will be set to the content.

    The real CIP value, not the transformed 'transfercapability'.
    Set on incoming call automatically. If set on outgoing call, it is used
    instead of transfercapability.

    When using capicommand(ccbs|....), this variable is set to either "ERROR" or

    If a Call-Linkage-Id is received for CCBS/CCNR, this variable contains this ID.
    But you need to use capicommand(peerlink) before dialing a CAPI channel, because
    of a design problem in Asterisk, chan_capi is not able to set channel variables
    of the calling channel.

    Can be set before answering and if set, the content is used for
    IE 'Connected Number' on answering.

    When CAPI reports DISCONNECT_IND, some CAPI reason codes are not set
    in HANGUPCAUSE, because they are not valid cause codes. To also
    get those CAPI reason codes, you can check this variable on hangup.
    This is e.g. useful for reason 0x3301, which tells that the ISDN
	connection is physically broken (cable).

    If chan_capi sends the call to extensions 'fax', the original extension number
    is saved in this variable.

    If set, this value will be used as hangup cause on hangup.

    On incoming calls, if the call was redirected to you by someone, the
    number of the redirecting party is saved in this variable.
    RDNIS is set as well.

    If the incoming call was redirected to you, this variable is set
    with the reason value.

    If an outgoing call is redirected, this variable may be set to the
    new destination.

    The two values of the information element PI (Progress Indicator).