Christian Richter
94274cc26b
* Introducing a new way for the l1watcher thread using the ast_sched way. Now l1watcher timeouts can be configured separately for every portgroup.
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* added a signal handler to allow waking up the misdn task thread (that may sleep in a poll call) via misdn_tasks_wakeup().
* overlap_dial functionality implemented.
* fixes a bug which leads to a segfault after reordering config elements in the enum or struct
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-11 19:30:35 +00:00
Christian Richter
bd0b801a0d
* removed tone_indicate, we genrate only the dialtone by ourself (and the hanguptone of course)
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* removed the state handling from release_chan, and simplified the ast_hangup/ast_queue_hangup stuff
* added pp_l2_check option, for pp lines where the pbx does not initially gets the L2 up
* simplified and fixed a bug in the pid generation code
* fixed a bug in empty_chan, which might cause segfaults and memorry corruptions
* added prepare_bc function, which is sort of the opposite of empty_bc
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-06 15:11:40 +00:00
Christian Richter
f629ae1872
added misdn show config description[s] to show all the possible misdn.conf settings with a description in the CLI
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-03 16:41:43 +00:00
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-29 20:12:19 +00:00
Kevin P. Fleming
472c1ca282
simplify autoconfig include mechanism (make tholo happy he can use lint again :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-07 18:54:56 +00:00
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01 12:51:41 +00:00
Christian Richter
39ac1a5b83
added a l1watcher timeout, therefore removed the old behaviour of guessing the l1state.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-23 19:40:16 +00:00
Christian Richter
19d46333bf
added callcounters for incoming and outgoing calls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 15:02:03 +00:00
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..
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-05 16:38:15 +00:00
Christian Richter
0b6bd0073b
put the default misdn.trace to /var/log/asterisk/misdn.log for better integration of existing log structure
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-27 08:23:53 +00:00
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-20 18:04:05 +00:00
Christian Richter
8e7dd52695
added option to change the connected party number dialplan (ton)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-09 18:01:27 +00:00
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-07 11:08:09 +00:00
Christian Richter
bd9c89a710
better default values for jitterbuffer in code and config
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-28 11:46:55 +00:00
Christian Richter
c7e0abdfed
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-22 16:48:25 +00:00
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-15 19:51:33 +00:00
Christian Richter
f6bd1b8559
added pmp_l1_check option, to avoid l1 checking for group calls on PMP ports
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-15 19:32:45 +00:00
Christian Richter
8d3f63f467
fixed the occasional no audio issue, still need deeper investigation .. echotraining is off by default
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-14 10:44:00 +00:00
Russell Bryant
1a23f4d092
rename chan_misdn_config.c to misdn_config.c
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-11 22:08:12 +00:00
