Christian Richter
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39ac1a5b83
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added a l1watcher timeout, therefore removed the old behaviour of guessing the l1state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-05-23 19:40:16 +00:00 |
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Christian Richter
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19d46333bf
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added callcounters for incoming and outgoing calls
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-05-22 15:02:03 +00:00 |
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Christian Richter
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efccf89eae
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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..
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-05-05 16:38:15 +00:00 |
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Christian Richter
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0b6bd0073b
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put the default misdn.trace to /var/log/asterisk/misdn.log for better integration of existing log structure
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-04-27 08:23:53 +00:00 |
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Christian Richter
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52eb1ad9d1
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-03-20 18:04:05 +00:00 |
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Christian Richter
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8e7dd52695
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added option to change the connected party number dialplan (ton)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-03-09 18:01:27 +00:00 |
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Christian Richter
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21735de56d
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-03-07 11:08:09 +00:00 |
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Christian Richter
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bd9c89a710
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better default values for jitterbuffer in code and config
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-02-28 11:46:55 +00:00 |
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Christian Richter
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c7e0abdfed
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fixed a ETSI violation (after RELEASE we need to RELEASE_COMPLETE (network side) one needs to upgread mISDNuser for that fix as well. also fixed the reload issue #6547
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-02-22 16:48:25 +00:00 |
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Christian Richter
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afaf8e4c04
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-02-15 19:51:33 +00:00 |
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Christian Richter
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f6bd1b8559
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added pmp_l1_check option, to avoid l1 checking for group calls on PMP ports
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-02-15 19:32:45 +00:00 |
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Christian Richter
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8d3f63f467
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fixed the occasional no audio issue, still need deeper investigation .. echotraining is off by default
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-02-14 10:44:00 +00:00 |
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Russell Bryant
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1a23f4d092
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rename chan_misdn_config.c to misdn_config.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2006-02-11 22:08:12 +00:00 |
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