Commit Graph

31 Commits

Author SHA1 Message Date
Richard Mudgett
27b54f4c1c channels/chan_misdn.c
channels/misdn/isdn_lib.c
*  Miscellaneous other fixes from trunk to make merging easier later.

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r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines

*  Miscellaneous formatting changes to make v1.4 and trunk
more merge compatible in the mISDN area.

channels/chan_misdn.c
*  Eliminated redundant code in cb_events() EVENT_SETUP

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r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines

improved helptext of misdn_set_opt.
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r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line

Cleaned up comment

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r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines

channels/chan_misdn.c
*  Made bearer2str() use allowed_bearers_array[]
*  Made use the causes.h defines instead of hardcoded numbers.
*  Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
*  Updated the misdn_set_opt application option descriptions.
*  Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.

channels/misdn/isdn_lib.c
*  Made use the causes.h defines instead of hardcoded numbers.
*  Fixed some spelling errors and typos.
*  Added all defined facility code strings to fac2str().

channels/misdn/isdn_lib.h
*  Added doxygen comments to struct misdn_bchannel.

channels/misdn/isdn_lib_intern.h
*  Added doxygen comments to struct misdn_stack.

channels/misdn_config.c
configs/misdn.conf.sample
*  Updated the mISDN presentation and screen parameter descriptions.

doc/misdn.txt (doc/tex/misdn.tex)
*  Updated the misdn_set_opt application option descriptions.
*  Fixed some spelling errors and typos.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@145293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-30 23:55:24 +00:00
Richard Mudgett
57e3570b62 * The allowed_bearers setting in misdn.conf misspelled one
of its options: digital_restricted.
*  Fixed some other spelling errors and typos.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@136241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-06 21:18:53 +00:00
Christian Richter
c2c1e68238 if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12 11:26:48 +00:00
Christian Richter
57ccb76df1 aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12 09:45:36 +00:00
Mark Michelson
a13b91f49b Removing non-existent options from misdn configuration sample.
(closes issue #10678, reported and patched by IgorG)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@82091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-10 15:02:12 +00:00
Christian Richter
fb52698667 Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line

changed a few debugs to higher debug levels
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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.
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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.
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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.
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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
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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. 
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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


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 09:06:50 +00:00
Christian Richter
6964f148ba Merged revisions 46176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-27 09:49:20 +00:00
Christian Richter
13825dab85 Merged revisions 44334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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
2006-10-06 12:50:25 +00:00
Tilghman Lesher
091e1aed8d Merged revisions 42716 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r42716 | tilghman | 2006-09-11 11:39:06 -0500 (Mon, 11 Sep 2006) | 2 lines

Spelling/grammar fixes (Issue 7929)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-11 16:41:49 +00:00
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
2006-07-13 14:13:24 +00:00
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
2006-07-06 15:11:40 +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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-29 20:12:19 +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.
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
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
2006-05-24 07:58:52 +00:00
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
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..
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-05 16:38:15 +00:00
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
2006-04-28 16:42:42 +00:00
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
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
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
a0800bd179 these traceing option do not exist anymore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-20 10:00:34 +00:00
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
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
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
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
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
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
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
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
2006-02-07 13:34:59 +00:00
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
2006-02-02 21:15:34 +00:00
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
2005-12-12 22:26:35 +00:00
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
2005-11-29 18:24:39 +00:00
Kevin P. Fleming
986a8ca089 issue #5566
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-01 22:04:14 +00:00
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
2005-10-31 22:51:12 +00:00