Reviewed by doc control

master
MelwareDE 14 years ago
parent 4c42d3fe26
commit 35137fc921
  1. 60
      CHANGES
  2. 8
      INSTALL
  3. 32
      capi.conf

@ -5,7 +5,7 @@ HEAD
------------------
- fixed buffer length error with internal libcapi debug code.
- performance optimizations, use debug code when needed only.
- added commands for media features supported by Dialogic Diva cards.
- added commands for media features supported by Dialogic(R) Diva(R) Media Boards
- added variable CAPI_CIP for full access to all bearer capabilities.
@ -14,15 +14,15 @@ chan_capi-1.1.2
- added config setting 'faxdetecttime' to limit the fax detection for a given amount of seconds.
- added config option for subscriber prefix. Some lines may show local calls without area code
signaled as subscriber-number. Here the complete prefix including area code must be added.
- better counting of active b-channels.
- make capicommand(progress) "early-B3" usable for non NT-mode incoming channels too.
- support early Line-Interconnect (bridge) as soon as both b-channels are up. This bridges b-channels
from beginning of call-establishment, even before calls are connected and the bridge command is received.
Dial() option 'G' is used to activate this feature.
- better counting of active B-channels.
- make capicommand(progress) "early-B3" usable for non NT-mode incoming channels as well.
- support early Line-Interconnect (bridge) as soon as both B-channels are up. This bridges B-channels
from beginning of the call-establishment, even before calls are connected and the bridge command is received.
The dial() option 'G' is used to activate this feature.
- fixed big-endian issue for DATA_B3 messages in internal libcapi code.
- fixed NULL-pointer when no digits are signaled in DID mode.
- adapt to new Asterisk 1.6.1 changes.
- added capicommand to set CAPI application id into an Asterisk dialplan variable.
- added capicommand to set CAPI application ID into an Asterisk dialplan variable.
chan_capi-1.1.1
@ -43,7 +43,7 @@ chan_capi-1.1.0
- fixed reading capi profile on big-endian
- increased maximum number of CAPI controllers to 64 (needed for big PBX).
- if immedіate=yes is set in DID mode, accept calls to empty DNID (needed for Austrian lines).
- use own libcapi20 implementation by default (no library for capi is needed any more).
- use own libcapi20 implementation by default (no library for CAPI is needed any more).
chan_capi-1.0.2
------------------
@ -51,7 +51,7 @@ chan_capi-1.0.2
- added 'x' option to capicommand(ect) to have real 'explicit call transfer'
(needed by some ISDN lines)
- support CCBS (call completion on busy subscriber)
- added capicommand(chat) for CAPI based MeetMe/Conference using onboard DSPs.
- added capicommand(chat) for CAPI-based MeetMe/Conference using onboard DSPs.
- fixed ton-display in 'show capi channels' on outgoing line.
- fix for 64bit support
- Asterisk 1.4.4 adaptions
@ -63,12 +63,12 @@ chan_capi-1.0.2
chan_capi-1.0.1
------------------
- added qsig caller-name patch by Mario Goegel
- recognise asterisk 1.4.1
- recognize asterisk 1.4.1
- register at CAPI with needed maxLogicalConnections only
- don't send SELECT_B_PROTOCOL more than once
- don't send more DATA_B3 messages than allowed by CAPI_MAX_B3_BLOCKS
- do not send SELECT_B_PROTOCOL more than once
- do not send more DATA_B3 messages than allowed by CAPI_MAX_B3_BLOCKS
- added 'k' option to capi receivefax command for not deleting bad faxes
- added b-channel number to channel name for better identification
- added B-channel number to channel name for better identification
- added variable setting REDIRECTIONNUMBER on outgoing call
- fixed deadlock with ast_async_goto on fax tone detection
- listen to CAPI supplementary information
@ -92,15 +92,15 @@ chan_capi-1.0.0
- Read the channel frames during wait for fax finish.
- Added progress when in faxmode to wakeup asterisk-1.2.
(needed for e.g. auto-hangup on timeout)
- Don't error on invalid controller in capi.conf, just ignore it.
- Do not error on invalid controller in capi.conf, just ignore it.
- Added 3PTY patch by Simon Peter.
- Allow echo-cancel even with old capi configuration bit for
- Allow echo-cancel even with old CAPI configuration bit for
echo-cancel.
- Fix compiler warnings.
- Fixed callerid on incoming call with Asterisk 1.4 (PR#25)
- Remove possible race condition in with hangup and DISCONNECT_IND.
- Rixed gain and echosquelch use according to transfercapability.
- Don't wait for DISCONNECT_B3_CONF in activehangup.
- Do not wait for DISCONNECT_B3_CONF in activehangup.
- Reset PLCI on DISCONNECT_IND to avoid race if asterisk is too slow
with hangup command.
- Added Asterisk 1.4 jitterbuffer usage.
@ -134,7 +134,7 @@ chan_capi-cm-0.6.5
chan_capi-cm-0.6.4
------------------
- don't do echo-squelch or gain if transfercapability is non-voice
- do not do echo-squelch or gain if transfercapability is non-voice
- fix deadlock when changing to fax mode
- better capi message handling
- removed double memset to zero
@ -156,15 +156,15 @@ chan_capi-cm-0.6.2
- set some info variables when receiving fax
- added language support
- prepared devicestate(hint) support
- don't change early-B3 setting on conf error for CONNECT_B3
- do not change early-B3 setting on conf error for CONNECT_B3
- small fixes in Makefile targets (thanks to Karsten Keil)
- don't send audio to local exchange when in TE mode.
- do not send audio to local exchange when in TE-mode.
- fix missing CONF messages when no interface is found
- small transfercap and overlapdial fix
- don't forward DTMF if in NT-mode and the line is not connected yet.
- do not forward DTMF if in NT-mode and the line is not yet connected
- fixed line interconnect
- b-channel handling better
- NT mode progress
- B-channel handling better
- NT-mode progress
- removed deadlock in faxreceive.
- initialize variable ocid
- correct use of timeoutms in native bridge
@ -181,26 +181,26 @@ chan_capi-cm-0.6.1
used if dial option 'd' is specified.
- moved ast_softhangup() out of interface lock
- use correct mutex_init call for interface lock
- when 'o' option is used for overlap dialing, don't send any digits
- when 'o' option is used for overlap dialing, do not send any digits
with the CONNECT_REQ. This gives better progress together with 'b'.
- create a pseudo channel for each interface for incoming signalling
without B-channel.
- added channel locks
- fixed capi init order (thanks to Hans Petter Selasky)
- fixed did handling
- fixed CAPI init order (thanks to Hans Petter Selasky)
- fixed DID handling
- set RDNIS if redirecting number was received.
- simplified call to ast_exists_extension()
- when check for valid extension, check the callerid as well
- changed call-waiting and deflect handling in CONNECT_IND
- use 'immediate' config in MSN mode, if pbx shall be started on
- use 'immediate' config in MSN mode, if PBX shall be started on
CONNECT_IND and shall not wait until SETUP/SENDING-COMPLETE was received.
Since info like REDIRECTINGNUMBER will come after CONNECT_IND, this may
be lost then. But for some drivers/telcos/pbx, this setting is needed.
- fix start of line interconnect in old mode.
- start early-b3 on PROCEEDING too.
- don't send audio data, if in fax receive mode
- do not send audio data, if in fax receive mode
- disconnect on finished fax immediately
- don't run through gain list, if gain is 1.0.
- do not run through gain list, if gain is 1.0.
- use correct A-law idle value.
- removed old example from capi.conf
@ -232,12 +232,12 @@ chan_capi-cm-0.6
- receive a fax via CAPI is now done with capicommand(receivefax|...) and added stationid...
- fixed call-deflection and moved this feature from separate application
to capicommand().
- added config option 'immediate' to start pbx if no dnid has been received yet.
- added config option 'immediate' to start PBX if no dnid has been received yet.
- endian fixes
- compile fixes with newer Asterisk
- update channel name on did changes.
- support 'type of number' (numbering-plan).
- U-Law setting is now done in capi.conf instead of Makefile define.
- u-Law setting is now done in capi.conf instead of Makefile define.
- allow using interface name in Dial().
- on hangup, use hangupcause from other channel or from var PRI_CAUSE.
- improved DID handling on PtP connections.

@ -4,12 +4,12 @@ INSTALL
Modify the Makefile to fit your system, especially the path to the Asterisk
include files.
To build the driver you will need an installed Capi system, including header
To build the driver you will need an installed CAPI system, including header
files.
By default an internal version of libcapi20 is used (you don't need libcapi20 to
be installed on your system). If you don't want this and the installed libcapi20
shall be used, add the option
By default, an internal version of libcapi20 is used (you do not need libcapi20 to
be installed on your system). If you do not want this and the installed libcapi20
should be used, add the option
USE_OWN_LIBCAPI=no
to the 'make' command.

@ -12,7 +12,7 @@ internationalprefix=00 ; or for example "+"
rxgain=1.0 ;linear receive gain (1.0 = no change)
txgain=1.0 ;linear transmit gain (1.0 = no change)
language=de ;set default language
;ulaw=yes ;set this, if you live in u-law world instead of a-law
;ulaw=yes ;set this, if you use u-law world instead of a-law
;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.
@ -23,20 +23,20 @@ language=de ;set default language
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;Use one interface section for each isdn port!
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
;Use one interface section for each ISDN port!
;ntmode=yes ;if the ISDN card operates in NT-mode, set this to 'yes'
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * = any
;defaultcid=123 ;set a default caller id to that interface for dial-out,
;this caller id will be used when dial option 'd' is set.
;defaultcid=123 ;set a default caller ID to that interface for dial-out,
;this caller ID will be used when the dial option 'd' is set.
;controller=0 ;ISDN4BSD default
;controller=7 ;ISDN4BSD USB default
controller=1 ;capi controller number of this interface/port
controller=1 ;CAPI controller number of this interface/port
group=1 ;dialout group
;prefix=0 ;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection
;prefix=0 ;set a prefix to the calling number on incoming calls
softdtmf=on ;enable/disable software DTMF detection, recommended for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed DTMF detection
faxdetect=off ;enable faxdetection and redirection to EXTEN 'fax' for incoming and/or
;outgoing calls. (default='off', possible values: 'incoming','outgoing','both')
faxdetecttime=0 ;Only detect faxes during the first 'n' seconds of the call.
@ -47,14 +47,14 @@ context=isdn-in ;context for incoming calls
;holdtype=hold ;when the PBX puts the call on hold, ISDN HOLD will be used. If
;set to 'local' (default value), no hold is done and the PBX may
;play MOH.
;immediate=yes ;DID: immediate start of pbx with extension 's' if no digits were
;immediate=yes ;DID: immediate start of PBX with extension 's' if no digits were
; received on incoming call (no destination number yet)
;MSN: start pbx on CONNECT_IND and don't wait for SETUP/SENDING-COMPLETE.
;MSN: start PBX on CONNECT_IND and do not wait for SETUP/SENDING-COMPLETE.
; info like REDIRECTINGNUMBER may be lost, but this is necessary for
; drivers/pbx/telco which does not send SETUP or SENDING-COMPLETE.
;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression. Disable this before you start recording voicemail
;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression. Disable it before you start recording voicemail
;or your files may get choppy. (you can use capicommand(echosquelch|no) for this)
;echocancel=yes ;Dialogic Diva (Capi) echo cancelation (yes=g165)
;echocancel=yes ;Dialogic(R) Diva(R) (CAPI) echo cancellation (yes=g165)
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)
;echotail=64 ;echo cancel tail setting (default=0 for maximum)
@ -65,9 +65,9 @@ echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for ol
;pickupgroup=1 ;PBX pickup group (which call groups are we allowed to pickup)
;transfergroup=1 ;Controller(s) where a transfer on native bridge is allowed to.
;language=de ;set language for this device (overwrites default language)
;disallow=all ;RTP codec selection (valid with Dialogic Diva only)
;allow=all ;RTP codec selection (valid with Dialogic Diva only)
devices=2 ;number of concurrent calls (b-channels) on this controller
;disallow=all ;RTP codec selection (valid with Dialogic(R) Diva(R) Media Boards only)
;allow=all ;RTP codec selection (valid with Dialogic(R) Diva(R) Media Boards only)
devices=2 ;number of concurrent calls (B-Channels) on this controller
;(2 makes sense for single BRI, 30/23 for PRI/T1)
;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.

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