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path: root/configs/misdn.conf.sample
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2006-03-20removed dynamic switching from transparent to hdlc mode. Instead we've got a ↵Christian Richter
config option hdlc=yes now which enables the hdlc controller for a data call git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-20these traceing option do not exist anymoreChristian Richter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-09added option to change the connected party number dialplan (ton)Christian Richter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-07added a bit more detailed description for the echotraining parameter, also ↵Christian Richter
changed the default from 1 to 2000. The default for the upper_threshold is now 0 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-28better default values for jitterbuffer in code and configChristian Richter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-15adde incoming_early_audio option, to avoid sending tone indications to the ↵Christian Richter
remote party on incoming calls from the pstn, this shouldnt be enabled, only if the provider allows it git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-15added pmp_l1_check option, to avoid l1 checking for group calls on PMP portsChristian Richter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-07default values of jitterbuffer and jitterbuffer_upper_threshold should be > ↵Christian Richter
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-02* removed unnecessary struct elements and functionsChristian Richter
* fixed "RETRIEVE does not work" bug * fixed some NT Mode bugs * removed some // comments * added configureable jitterbuffer * removed own tone-generator, and use asterisks instead, to support asterisks indications * added more support for hw-bridging, we bridge now every possible call * fixed some hdlc mode issues, with a patch for chan_zap we can make data calls between chan_zap and chan_misdn now * completely reworked the config engine, works like a charm now * fixed SetCallerPres - bug * added Progress and Proceeding passing * optimized Ringing Indication handling * added full ast_send_text support (you can setup nice menus with the dialplan now) * added support to read /etc/misdn-init.conf to clarify the NT+PTP Problem * we compile now channels/misdn if mISDNuser is installed systemwide git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-12updated the documentation and the sample config to meet the presentChristian Richter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29remove extraneous svn:executable propertiesKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-01issue #5566Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-10-31add experimental mISDN channel driver (issue #4077)Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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