Asterisk
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asterisk
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22 Commits (f60ada0be22cd646454278e102d384a5dbaf7d59)

Author SHA1 Message Date
Christian Richter 54ce0f0a22 added even more statefulness for sending out disconnect/release/release_complete messages. added support for incoming presentation/screening. fixed a bug that we generate TONE_EVENTS on hanguptone_indicatem, which caused asterisk to write blocking thread messages. added nodialtone option to prevent dialtone for always_immediate
20 years ago
Christian Richter bd0b801a0d * removed tone_indicate, we genrate only the dialtone by ourself (and the hanguptone of course)
20 years ago
Christian Richter f5c0cd2ddc added better L2 handling for ptp, if it's down we don't try to call on that port in groupdial anymore, also we try to get it up then. Additionally added the configoptions ntdebugflags and ntdebugfile to debug the mISDNuser NT Stack (should have done that ages before..). isdn_lib.c compiles again.
20 years ago
Christian Richter 4be235a974 added bearer capability reject support. we send release instead of disconnect in case we have no real channel yet. added support for Restarting channels added support for sending complete decoding. changed some log levels.
20 years ago
Christian Richter 8122c35675 fixed to early connect bug which came in yesterday.., also added the transmit of progress indicators through channel vars
20 years ago
Christian Richter 19d46333bf added callcounters for incoming and outgoing calls
20 years ago
Christian Richter efccf89eae Added option far_alerting. This option makes it possible to generate a Ringing on other channels if they feel that they should have inband ringing, but there is non in reality. I need this due to the fact that asterisk has not the possibility to transmit progress indicators thus chan_sip and others do not know wether they should generate a Rining tone themselves if they receive AST_CONTROL_RINGING..
20 years ago
Russell Bryant c38fbd246e note that group assignments must be from 0 to 63 (issue #7048)
20 years ago
Christian Richter 0b6bd0073b put the default misdn.trace to /var/log/asterisk/misdn.log for better integration of existing log structure
20 years ago
Christian Richter 52eb1ad9d1 removed dynamic switching from transparent to hdlc mode. Instead we've got a config option hdlc=yes now which enables the hdlc controller for a data call
20 years ago
Christian Richter a0800bd179 these traceing option do not exist anymore
20 years ago
Christian Richter 8e7dd52695 added option to change the connected party number dialplan (ton)
20 years ago
Christian Richter 21735de56d added a bit more detailed description for the echotraining parameter, also changed the default from 1 to 2000. The default for the upper_threshold is now 0
20 years ago
Christian Richter bd9c89a710 better default values for jitterbuffer in code and config
20 years ago
Christian Richter afaf8e4c04 adde incoming_early_audio option, to avoid sending tone indications to the remote party on incoming calls from the pstn, this shouldnt be enabled, only if the provider allows it
20 years ago
Christian Richter f6bd1b8559 added pmp_l1_check option, to avoid l1 checking for group calls on PMP ports
20 years ago
Christian Richter b42dd639ee default values of jitterbuffer and jitterbuffer_upper_threshold should be > 0, this fixes the tv_fix warnings, because we use ast_read to transmit frames to asterisk in jitterbuffer mode, instead of queueing the audio data with ast_queue_frame.
20 years ago
Christian Richter 7133d1b006 * removed unnecessary struct elements and functions
20 years ago
Christian Richter d37857c208 updated the documentation and the sample config to meet the present
20 years ago
Kevin P. Fleming 2c65582b66 remove extraneous svn:executable properties
20 years ago
Kevin P. Fleming 986a8ca089 issue #5566
20 years ago
Kevin P. Fleming 0ac988acaa add experimental mISDN channel driver (issue #4077)
20 years ago