Fix: Asterisk DTMF detection works now To enable, use option "a". -> for calls from LCR use lcr_config(a) in extensions.conf -> for calls to LCR use Dial(LCR/pbx/<number>/a)
Define prload of mISDN buffer by chan_lcr (required for fax) Use q<ms> option to peload.
Removed complete bchannel handling from chan_lcr The remote application interface does not allow any bchannel to be exported or imported. Audio traffic via socket interface is used instead. The joinremote instance became obsolete and is removed. The remote action (routing) became obsolete, use interface.conf instead. The handling of loopback device became obsolete and was removed The chan_lcr does not rely on mISDN anymore, that means: - can be used with GSM and without mISDN at all. - chan_lcr can be used as internal extension of LCR (e.g. SIP phone) (chan_lcr can be handled as any other interface) - no loopback device to be used anymore.
[chan_lcr] Fixed caller ID for calls from AST->LCR
Improved and applied Wimpy's Asterisk 1.8.x support. configure script will automatically detect new asterisk API, so there is no need for different chan_lcr.c source codes. 'type', 'presentation', and 'screening indicator' are now transcoded.
Added queue buffer for chan_lcr sending faxes without interruption. Use options "t:q250" for disabling mISDN_dsp and adding a 250ms delay. modified: README modified: bchannel.c modified: bchannel.h modified: chan_lcr.c modified: chan_lcr.h modified: select.c
Replaced polling loop for LCR and chan_lcr with select based event loop. Now LCR and chan_lcr will not use any CPU until there is work to do.
Added keypad forwarding, keypad parameter, chan_lcr keypad option 'k'. modified: README modified: action.cpp modified: apppbx.cpp modified: chan_lcr.c modified: chan_lcr.h modified: dss1.cpp modified: joinpbx.cpp modified: joinpbx.h modified: message.h modified: route.c modified: route.h
Fixed disabling of DTMF using 'n' option of chan_lcr. Please enter the commit message for your changes. modified: README modified: apppbx.cpp modified: bchannel.c modified: chan_lcr.c modified: chan_lcr.h
Added PROGRESS indication in both directions, so early audio is possible. -> Tones and announcements shall be forwarded. modified: README modified: chan_lcr.c modified: chan_lcr.h
simplified rebuffer-mode to make large block sizes work better Thanks to Kristijan Vrban for the patch!
Added fax detection patch by gregory. modified: README modified: chan_lcr.c modified: chan_lcr.h
Merge branch 'master' of ssh://schlaile@git.misdn.org/var/git/lcr
added inband dtmf support to chan_lcr Use option 's' in lcr_config or within the dial-command options, just like in misdn.
Completed documentation about instance creation/destruction proceedure. modified: chan_lcr.c modified: chan_lcr.h
chan_lcr: forgotten commit for new ref fix
work around broken HOLD/UNHOLD handling on some SIP phones Some SIP phones don't send RETRIEVE before they send TRANSFER. So we RETRIEVE if we bridge two channels, if calls are still on hold. Also: handle CONTROL_SRCUPDATE with a debug message in recent versions of asterisk.
make compile with gcc 4.* without warnings. (hopefully with all versions) modified: Makefile modified: bchannel.c modified: chan_lcr.h modified: extension.c modified: gentones.c modified: genwave.c modified: joinpbx.cpp modified: tones.c
many fixes on HDLC issues many fixes on briding issues -> briding will work with dsp and directly via chan_lcr -> hdlc will work with dsp and directly with chan_lcr modified: bchannel.c modified: chan_lcr.c modified: chan_lcr.h
rebuffer option for chan_lcr (160 bytes per frame) l1-link state "unknown" if not known yet. removed root user check. modified: bchannel.c modified: bchannel.h modified: chan_lcr.c modified: chan_lcr.h modified: dss1.cpp modified: lcradmin.c modified: mISDN.cpp modified: main.c