asterisk

Commit Graph

Author	SHA1	Message	Date
Christian Richter	fb52698667	Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 \| crichter \| 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) \| 1 line changed a few debugs to higher debug levels ........ r48321 \| crichter \| 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) \| 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 \| crichter \| 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) \| 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 \| crichter \| 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) \| 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 \| crichter \| 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) \| 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 \| crichter \| 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) \| 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 \| crichter \| 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) \| 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Christian Richter	6964f148ba	Merged revisions 46176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46176 \| crichter \| 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) \| 1 line added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Christian Richter	13825dab85	Merged revisions 44334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44334 \| crichter \| 2006-10-04 17:13:58 +0200 (Mi, 04 Okt 2006) \| 1 line added the option 'reject_cause' to make it possible to set the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is automatically rejected because chan_misdn does not support that kind of callwaiting. Therefore chan_misdn supports now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the info if the requested channel is incoming or outgoing to make the 3. channel possible ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	091e1aed8d	Merged revisions 42716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r42716 \| tilghman \| 2006-09-11 11:39:06 -0500 (Mon, 11 Sep 2006) \| 2 lines Spelling/grammar fixes (Issue 7929) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Christian Richter	bd0b801a0d	* removed tone_indicate, we genrate only the dialtone by ourself (and the hanguptone of course) * 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	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Christian Richter	19d46333bf	added callcounters for incoming and outgoing calls git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	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.. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Russell Bryant	c38fbd246e	note that group assignments must be from 0 to 63 (issue #7048 ) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Christian Richter	0b6bd0073b	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	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Christian Richter	a0800bd179	these traceing option do not exist anymore git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Christian Richter	8e7dd52695	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	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Christian Richter	bd9c89a710	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	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Christian Richter	f6bd1b8559	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	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Christian Richter	7133d1b006	* removed unnecessary struct elements and functions * 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	20 years ago
Christian Richter	d37857c208	updated the documentation and the sample config to meet the present git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	2c65582b66	remove extraneous svn:executable properties git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	986a8ca089	issue #5566 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	0ac988acaa	add experimental mISDN channel driver (issue #4077 ) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago

26 Commits (368777176077aca2737952c6320e3494a9686bf4)