diff --git a/CHANGES b/CHANGES index bd4b5f9..1b50269 100644 --- a/CHANGES +++ b/CHANGES @@ -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. diff --git a/INSTALL b/INSTALL index 81783b2..6a4b2ce 100644 --- a/INSTALL +++ b/INSTALL @@ -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. diff --git a/capi.conf b/capi.conf index 35ea459..4238cdd 100644 --- a/capi.conf +++ b/capi.conf @@ -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.