330 lines
14 KiB

- refuse to reload on active channels
- ast_devstate_changed() changes with new cache argument
- check for bchannel information element on incoming call
- added 't' option to select in-band tones available indication as Q.931
(thanks to Maciej S. Szmigiero <>)
- support for Asterisk 13 (thanks to Michael Kuron <>)
- fixed check of asterisk version with asterisk binary if another install path is used.
- remove codec 'none' from translation path which prevents bridge.
- Asterisk 1.8 changes
- added dialplan variables ISDNPI1 and ISDNPI2
- Support for 'divastreaming' available (Diva 4PRI PCI, Diva 2PRI PCI,
Diva 4PRI HS PCIe, Diva 2PRI HS PCIe, Diva 1PRI HS PCIe,
Diva 8PRI PCIe FS, Diva 4PRI PCIe FS)
- Fixed memory leak in chat_play if using with 's' option
- Add 'econtransitconn' configurations parameter. Allows to preserve
EC for transit connections
- added variable DISCONNECT_IND_REASON to get real reason code from CAPI
- added capicommand(keypad)
- added log of FACILITY_IND for call deflect suplementary service
- malloc,strdup,free -> ast_(malloc,strdup,free)
- Use Diva streaming for NULL PLCI and for resource PLCI
- Resolved
- Use Diva QSIG CAPI extensions for processing of Calling Party Name
- Add Slinear
- Add HD voice using G.722, Siren7, Siren14 and Slinear16
- added faxdestination= to capi.conf to configure custom context,exten,prio.
- allow capicommand option separator "|" and ",".
- Add MWI server
- Add MWI client
- Improved call distribution algorithm for outgoing calls
- Add 'capi ifcstate', 'capi exec', 'capi show resources' and
'chat manage remove' CLI commands
- Add AMI 'Capichat[List,Mute,Unmute,Remove]' and 'CapiCommand' actions and
Capichat[List,ListComplete,Join,Leave,End] events
- Add device state events for char rooms (capichat:[roomname]) and for
interface state (ISDN/I[N]/congestion)
- Add access to Diva trace driver
- corrected check for Info value on CONNECT_B3_CONF
- fixed NULL-pointer in cid_name (asterisk 1.6 pickup revealed this)
- changed action on prodding in .write function to ignore it
- API adaptions to new asterisk trunk
- use extended fax if available only and have fine resolution on receive activated by default
- fixed chat error on creating third chat-room
- 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(R) Diva(R) Media Boards
- added variable CAPI_CIP for full access to all bearer capabilities.
- fixed CAPI 'chat' for using more than 8 controllers.
- fixed CAPI 'chat' for not exceeding maximal size of CAPI message
- add color fax
- add autodetection of fax file format for sendfax command (SFF, CFF JPEG, CFF T.43, TEXT)
- add extension for FAX paper formats and resolutions
- adjust NULL PLCI LI path
- add resource PLCI
- add echocancelpath configuration option
- fixed possible race condition while waiting for DISCONNECT_CONF
- add static half duplex conference
- add dynamic half duplex conference
- add clear channel fax
- add DTMF detection for NULL PLCI
- use direct access to vocoders without RTP framing
- play message to conference and music on hold to caller
- add commands to remove users from chat
- allow to specity the 'ETS 300 102-1' called party number octet 3
- fixed possible buffer overflow in 'deflect'
- added 'h' option to chat to auto-hangup caller if alone in conference too long
- 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 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.
- fixed controller count when creating NULL-PLCI interface.
- simplified check for unused controller.
- send local DATE/TIME with CONNECT when in NT-mode.
- added transfergroup config option to set controllers allowed for transfer.
- send Sending-Complete if overlap dialing is not used.
- fixed numberingplan for 'connected number'.
- added devicestate option.
- capi-chat can play music-on-hold for first caller.
- adapt to new asterisk 1.6 API
- fixed reading capi profile on big-endian
- increased maximum number of CAPI controllers to 64 (needed for big PBX).
- if immediate=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).
- possibly fixed ECT channel hang
- 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.
- fixed ton-display in 'show capi channels' on outgoing line.
- fix for 64bit support
- Asterisk 1.4.4 adaptions
- send 'In-band info available' for progress in NT-mode
- detect KEYPAD digits in NT-mode and send call to 'K...' extension
- fixed possible deadlock when answering a just disconnected call
- added qsig caller-name patch by Mario Goegel
- recognize asterisk 1.4.1
- register at CAPI with needed maxLogicalConnections only
- 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 variable setting REDIRECTIONNUMBER on outgoing call
- fixed deadlock with ast_async_goto on fax tone detection
- listen to CAPI supplementary information
- added support for different qsig variants
- added support for rerouting informations on incoming calls
- added implementation of Simple Call Transfer
- Added preliminary handling of MANUFACTURER_IND.
- Check for fax connection when in receivefax mode before writing data
to file.
- Try to detect for generic jitter buffer patch
- Adapted to new Asterisk version 1.4 (Bug PR#20)
- Adapted to new ast_channel_alloc from asterisk trunk.
- Make capicommand(echocancel) setting non-permanent.
Setting is restored after hangup.
- Better check for asterisk version 1.4 BUG#23
- Added information and changed example for tx/rx gains.
- 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)
- 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
- 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.
- 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.
- Fixed compiler warning with Asterisk 1.4.0
- Added music-on-hold on HOLD request for Asterisk 1.4
- Set FAXREASONTEXT to "OK" instead of empty string.
- Disconnect reason 0x3400 is treated as 'successful' in faxmode.
- use CIP speech as default if transfercapability is unkown.
- set correct channel/cdr status in fax mode too.
- Use correct interface to wait on for ECT facility.
- added amaflags
- added pickupgroup (internal pickup with pickupexten)
- added capicommand(sendfax)
- dropped compatiblity to versions before Asterisk-1.2.0
- added capicommand(echocancel)
- added CAPI RTP (Eicon DIVA Server cards can do that)
- added new cli 'capi show channels'
- a lot of fixes
- fixed compilation with Asterisk 1.0.7
- fixed call deflect if number is longer
- removed function VANITYNUMBER, this does not belong into channel drivers
- 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
- use safer copy_string function
- removed unneeded lockB3q
- avoid capi message number zero
- added locking rules
- fix ECT (use implicit ECT)
- fixed setting language for each interface
- fix wait for b-channel to go down on ECT.
- fixed buffer overflow in overlap dial
- remove some compiler warnings
- set some info variables when receiving fax
- added language support
- prepared devicestate(hint) support
- do not change early-B3 setting on conf error for CONNECT_B3
- small fixes in Makefile targets (thanks to Karsten Keil)
- 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
- 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
- removed deadlock in faxreceive.
- initialize variable ocid
- correct use of timeoutms in native bridge
- use more common defines for mutex_lock/unlock and _log()
- clean up and OpenPBX portability.
- adapted to latest asterisk bridge function prototype.
- fixed setting of redirecting number rdnis instead of cid.
- allow setting a callerid in the dial() command without changing
the original channel callerid.
- added config 'defaultcid' to set a default caller id which will be
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, 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
- 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
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.
- do not send audio data, if in fax receive mode
- disconnect on finished fax immediately
- do not run through gain list, if gain is 1.0.
- use correct A-law idle value.
- removed old example from capi.conf
- added 'relaxdtmf'.
- more BSD compatibility
- correct PROGRESS handling
- start PBX on SETUP/SENDING-COMPLETE for PtP only.
- added verbose text for capi info/reason error messages.
- fixed echo-cancel setup structure
- use correct facility-selector for echo-cancel
- use capi.conf option 'echocancelold' for old facility-selector (6)
- changed isdnmode configuration from ptp/ptmp to msn/did
- added ntmode configuration
- added application capicommand() for CAPI based applications
(removed standalone applications)
- capicommand(RETRIEVE) can now be called from other channels
- support ISDN hold (holdtype in capi.conf)
- added HOLD/RETRIEVE for Asterisk indications.
- added custom function VANITYNUMBER to convert letters into digits.
- added CAPI Line Interconnect (native bridging)
- use variable CONNECTEDNUMBER on Answer().
- set variable REDIRECTINGNUMBER on incomming call if it was diverted.
- added variable REDIRECTREASON
- fixed unload
- removed obsolete thread mutex
- fixed dnid/exten/immediate handling on PtP.
- 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.
- 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.
- allow using interface name in Dial().
- on hangup, use hangupcause from other channel or from var PRI_CAUSE.
- improved DID handling on PtP connections.
- capi.conf structure changes: one own section for each interface,
no global 'interfaces' any more. Section name will be interface name.
- restructured module loading and init.
- dial string changed: parameters like 'b' not as part of number any more.
- send alert on alerting only (busy() and congestion() work now).
- better overlap sending (new parameter 'o' for dialstring to
send only the first two digits with CONNECT_REQ only, the remaining
digits and even digits following the dial() command, will be send
as INFO_REQ/Overlap).
- further fixups