Reviewed by doc control
MelwareDE
parent
4c42d3fe26
commit
35137fc921
60
CHANGES
60
CHANGES
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@ -5,7 +5,7 @@ HEAD
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------------------
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- fixed buffer length error with internal libcapi debug code.
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- performance optimizations, use debug code when needed only.
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- added commands for media features supported by Dialogic Diva cards.
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- added commands for media features supported by Dialogic(R) Diva(R) Media Boards
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- added variable CAPI_CIP for full access to all bearer capabilities.
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@ -14,15 +14,15 @@ chan_capi-1.1.2
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- added config setting 'faxdetecttime' to limit the fax detection for a given amount of seconds.
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- added config option for subscriber prefix. Some lines may show local calls without area code
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signaled as subscriber-number. Here the complete prefix including area code must be added.
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- better counting of active b-channels.
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- make capicommand(progress) "early-B3" usable for non NT-mode incoming channels too.
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- support early Line-Interconnect (bridge) as soon as both b-channels are up. This bridges b-channels
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from beginning of call-establishment, even before calls are connected and the bridge command is received.
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Dial() option 'G' is used to activate this feature.
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- better counting of active B-channels.
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- make capicommand(progress) "early-B3" usable for non NT-mode incoming channels as well.
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- support early Line-Interconnect (bridge) as soon as both B-channels are up. This bridges B-channels
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from beginning of the call-establishment, even before calls are connected and the bridge command is received.
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The dial() option 'G' is used to activate this feature.
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- fixed big-endian issue for DATA_B3 messages in internal libcapi code.
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- fixed NULL-pointer when no digits are signaled in DID mode.
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- adapt to new Asterisk 1.6.1 changes.
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- added capicommand to set CAPI application id into an Asterisk dialplan variable.
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- added capicommand to set CAPI application ID into an Asterisk dialplan variable.
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chan_capi-1.1.1
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@ -43,7 +43,7 @@ chan_capi-1.1.0
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- fixed reading capi profile on big-endian
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- increased maximum number of CAPI controllers to 64 (needed for big PBX).
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- if immedіate=yes is set in DID mode, accept calls to empty DNID (needed for Austrian lines).
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- use own libcapi20 implementation by default (no library for capi is needed any more).
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- use own libcapi20 implementation by default (no library for CAPI is needed any more).
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chan_capi-1.0.2
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------------------
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@ -51,7 +51,7 @@ chan_capi-1.0.2
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- added 'x' option to capicommand(ect) to have real 'explicit call transfer'
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(needed by some ISDN lines)
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- support CCBS (call completion on busy subscriber)
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- added capicommand(chat) for CAPI based MeetMe/Conference using onboard DSPs.
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- added capicommand(chat) for CAPI-based MeetMe/Conference using onboard DSPs.
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- fixed ton-display in 'show capi channels' on outgoing line.
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- fix for 64bit support
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- Asterisk 1.4.4 adaptions
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@ -63,12 +63,12 @@ chan_capi-1.0.2
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chan_capi-1.0.1
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------------------
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- added qsig caller-name patch by Mario Goegel
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- recognise asterisk 1.4.1
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- recognize asterisk 1.4.1
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- register at CAPI with needed maxLogicalConnections only
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- don't send SELECT_B_PROTOCOL more than once
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- don't send more DATA_B3 messages than allowed by CAPI_MAX_B3_BLOCKS
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- do not send SELECT_B_PROTOCOL more than once
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- do not send more DATA_B3 messages than allowed by CAPI_MAX_B3_BLOCKS
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- added 'k' option to capi receivefax command for not deleting bad faxes
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- added b-channel number to channel name for better identification
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- added B-channel number to channel name for better identification
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- added variable setting REDIRECTIONNUMBER on outgoing call
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- fixed deadlock with ast_async_goto on fax tone detection
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- listen to CAPI supplementary information
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@ -92,15 +92,15 @@ chan_capi-1.0.0
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- Read the channel frames during wait for fax finish.
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- Added progress when in faxmode to wakeup asterisk-1.2.
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(needed for e.g. auto-hangup on timeout)
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- Don't error on invalid controller in capi.conf, just ignore it.
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- Do not error on invalid controller in capi.conf, just ignore it.
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- Added 3PTY patch by Simon Peter.
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- Allow echo-cancel even with old capi configuration bit for
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- Allow echo-cancel even with old CAPI configuration bit for
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echo-cancel.
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- Fix compiler warnings.
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- Fixed callerid on incoming call with Asterisk 1.4 (PR#25)
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- Remove possible race condition in with hangup and DISCONNECT_IND.
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- Rixed gain and echosquelch use according to transfercapability.
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- Don't wait for DISCONNECT_B3_CONF in activehangup.
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- Do not wait for DISCONNECT_B3_CONF in activehangup.
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- Reset PLCI on DISCONNECT_IND to avoid race if asterisk is too slow
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with hangup command.
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- Added Asterisk 1.4 jitterbuffer usage.
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@ -134,7 +134,7 @@ chan_capi-cm-0.6.5
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chan_capi-cm-0.6.4
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------------------
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- don't do echo-squelch or gain if transfercapability is non-voice
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- do not do echo-squelch or gain if transfercapability is non-voice
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- fix deadlock when changing to fax mode
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- better capi message handling
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- removed double memset to zero
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@ -156,15 +156,15 @@ chan_capi-cm-0.6.2
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- set some info variables when receiving fax
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- added language support
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- prepared devicestate(hint) support
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- don't change early-B3 setting on conf error for CONNECT_B3
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- do not change early-B3 setting on conf error for CONNECT_B3
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- small fixes in Makefile targets (thanks to Karsten Keil)
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- don't send audio to local exchange when in TE mode.
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- do not send audio to local exchange when in TE-mode.
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- fix missing CONF messages when no interface is found
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- small transfercap and overlapdial fix
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- don't forward DTMF if in NT-mode and the line is not connected yet.
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- do not forward DTMF if in NT-mode and the line is not yet connected
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- fixed line interconnect
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- b-channel handling better
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- NT mode progress
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- B-channel handling better
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- NT-mode progress
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- removed deadlock in faxreceive.
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- initialize variable ocid
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- correct use of timeoutms in native bridge
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@ -181,26 +181,26 @@ chan_capi-cm-0.6.1
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used if dial option 'd' is specified.
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- moved ast_softhangup() out of interface lock
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- use correct mutex_init call for interface lock
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- when 'o' option is used for overlap dialing, don't send any digits
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- when 'o' option is used for overlap dialing, do not send any digits
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with the CONNECT_REQ. This gives better progress together with 'b'.
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- create a pseudo channel for each interface for incoming signalling
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without B-channel.
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- added channel locks
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- fixed capi init order (thanks to Hans Petter Selasky)
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- fixed did handling
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- fixed CAPI init order (thanks to Hans Petter Selasky)
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- fixed DID handling
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- set RDNIS if redirecting number was received.
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- simplified call to ast_exists_extension()
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- when check for valid extension, check the callerid as well
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- changed call-waiting and deflect handling in CONNECT_IND
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- use 'immediate' config in MSN mode, if pbx shall be started on
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- use 'immediate' config in MSN mode, if PBX shall be started on
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CONNECT_IND and shall not wait until SETUP/SENDING-COMPLETE was received.
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Since info like REDIRECTINGNUMBER will come after CONNECT_IND, this may
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be lost then. But for some drivers/telcos/pbx, this setting is needed.
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- fix start of line interconnect in old mode.
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- start early-b3 on PROCEEDING too.
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- don't send audio data, if in fax receive mode
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- do not send audio data, if in fax receive mode
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- disconnect on finished fax immediately
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- don't run through gain list, if gain is 1.0.
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- do not run through gain list, if gain is 1.0.
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- use correct A-law idle value.
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- removed old example from capi.conf
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@ -232,12 +232,12 @@ chan_capi-cm-0.6
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- receive a fax via CAPI is now done with capicommand(receivefax|...) and added stationid...
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- fixed call-deflection and moved this feature from separate application
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to capicommand().
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- added config option 'immediate' to start pbx if no dnid has been received yet.
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- added config option 'immediate' to start PBX if no dnid has been received yet.
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- endian fixes
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- compile fixes with newer Asterisk
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- update channel name on did changes.
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- support 'type of number' (numbering-plan).
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- U-Law setting is now done in capi.conf instead of Makefile define.
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- u-Law setting is now done in capi.conf instead of Makefile define.
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- allow using interface name in Dial().
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- on hangup, use hangupcause from other channel or from var PRI_CAUSE.
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- improved DID handling on PtP connections.
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8
INSTALL
8
INSTALL
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@ -4,12 +4,12 @@ INSTALL
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Modify the Makefile to fit your system, especially the path to the Asterisk
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include files.
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To build the driver you will need an installed Capi system, including header
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To build the driver you will need an installed CAPI system, including header
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files.
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By default an internal version of libcapi20 is used (you don't need libcapi20 to
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be installed on your system). If you don't want this and the installed libcapi20
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shall be used, add the option
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By default, an internal version of libcapi20 is used (you do not need libcapi20 to
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be installed on your system). If you do not want this and the installed libcapi20
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should be used, add the option
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USE_OWN_LIBCAPI=no
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to the 'make' command.
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32
capi.conf
32
capi.conf
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@ -12,7 +12,7 @@ internationalprefix=00 ; or for example "+"
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rxgain=1.0 ;linear receive gain (1.0 = no change)
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txgain=1.0 ;linear transmit gain (1.0 = no change)
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language=de ;set default language
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;ulaw=yes ;set this, if you live in u-law world instead of a-law
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;ulaw=yes ;set this, if you use u-law world instead of a-law
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;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,
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;see Asterisk documentation for all jb* setting available.
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@ -23,20 +23,20 @@ language=de ;set default language
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[ISDN1] ;this example interface gets name 'ISDN1' and may be any
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;name not starting with 'g' or 'contr'.
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;Use one interface section for each isdn port!
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;ntmode=yes ;if isdn card operates in nt mode, set this to yes
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;Use one interface section for each ISDN port!
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;ntmode=yes ;if the ISDN card operates in NT-mode, set this to 'yes'
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isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
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;when using NT-mode, 'DID' should be set in any case
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incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * = any
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;defaultcid=123 ;set a default caller id to that interface for dial-out,
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;this caller id will be used when dial option 'd' is set.
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;defaultcid=123 ;set a default caller ID to that interface for dial-out,
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;this caller ID will be used when the dial option 'd' is set.
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;controller=0 ;ISDN4BSD default
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;controller=7 ;ISDN4BSD USB default
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controller=1 ;capi controller number of this interface/port
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controller=1 ;CAPI controller number of this interface/port
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group=1 ;dialout group
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;prefix=0 ;set a prefix to calling number on incoming calls
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softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards
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relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection
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;prefix=0 ;set a prefix to the calling number on incoming calls
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softdtmf=on ;enable/disable software DTMF detection, recommended for AVM cards
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relaxdtmf=on ;in addition to softdtmf, you can use relaxed DTMF detection
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faxdetect=off ;enable faxdetection and redirection to EXTEN 'fax' for incoming and/or
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;outgoing calls. (default='off', possible values: 'incoming','outgoing','both')
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faxdetecttime=0 ;Only detect faxes during the first 'n' seconds of the call.
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@ -47,14 +47,14 @@ context=isdn-in ;context for incoming calls
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;holdtype=hold ;when the PBX puts the call on hold, ISDN HOLD will be used. If
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;set to 'local' (default value), no hold is done and the PBX may
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;play MOH.
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;immediate=yes ;DID: immediate start of pbx with extension 's' if no digits were
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;immediate=yes ;DID: immediate start of PBX with extension 's' if no digits were
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; received on incoming call (no destination number yet)
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;MSN: start pbx on CONNECT_IND and don't wait for SETUP/SENDING-COMPLETE.
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;MSN: start PBX on CONNECT_IND and do not wait for SETUP/SENDING-COMPLETE.
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; info like REDIRECTINGNUMBER may be lost, but this is necessary for
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; drivers/pbx/telco which does not send SETUP or SENDING-COMPLETE.
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;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression. Disable this before you start recording voicemail
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;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression. Disable it before you start recording voicemail
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;or your files may get choppy. (you can use capicommand(echosquelch|no) for this)
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;echocancel=yes ;Dialogic Diva (Capi) echo cancelation (yes=g165)
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;echocancel=yes ;Dialogic(R) Diva(R) (CAPI) echo cancellation (yes=g165)
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;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
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echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)
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;echotail=64 ;echo cancel tail setting (default=0 for maximum)
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@ -65,9 +65,9 @@ echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for ol
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;pickupgroup=1 ;PBX pickup group (which call groups are we allowed to pickup)
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;transfergroup=1 ;Controller(s) where a transfer on native bridge is allowed to.
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;language=de ;set language for this device (overwrites default language)
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;disallow=all ;RTP codec selection (valid with Dialogic Diva only)
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;allow=all ;RTP codec selection (valid with Dialogic Diva only)
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devices=2 ;number of concurrent calls (b-channels) on this controller
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;disallow=all ;RTP codec selection (valid with Dialogic(R) Diva(R) Media Boards only)
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;allow=all ;RTP codec selection (valid with Dialogic(R) Diva(R) Media Boards only)
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devices=2 ;number of concurrent calls (B-Channels) on this controller
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;(2 makes sense for single BRI, 30/23 for PRI/T1)
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;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,
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;see Asterisk documentation for all jb* setting available.
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