Commit Graph

38 Commits

Author SHA1 Message Date
Sean Bright f22962a0c1 Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.
Let's try that again, this time removing trailing whitespace and not leading
whitespace.  I can't believe no one noticed.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:39:21 +00:00
Sean Bright a7d813cae7 Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:32:03 +00:00
Richard Mudgett 7872538b83 Add outgoing_colp misdn.conf port parameter.
Select what to do with outgoing COLP information on this port.
0 - Send out COLP information unaltered. (default)
1 - Force COLP to restricted on all outgoing COLP information.
2 - Do not send COLP information.
outgoing_colp=0

Also fixed sending the EctInform message so it always has the
required redirectionNumber parameter when the status is active.

JIRA ABE-1853


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14 22:03:49 +00:00
Richard Mudgett 6bb2b6c096 Added CCBS/CCNR Party A support and enhanced COLP support.
This change adds the following features to chan_misdn:
* CCBS/CCNR Party A support for PTMP and PTP modes.
* Enhances COLP support for call diversion and explicit call transfer.

These enhanced features require a modified version of mISDN.

The latest modified mISDN v1.1.x based version is available at:
http://svn.digium.com/svn/thirdparty/mISDN/trunk
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk

Taged versions of the modified mISDN code are available under:
http://svn.digium.com/svn/thirdparty/mISDN/tags
http://svn.digium.com/svn/thirdparty/mISDNuser/tags

Review: http://reviewboard.digium.com/r/218/

Merged from team/rmudgett/misdn_facility branch.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-21 17:44:01 +00:00
Mark Michelson 6f53ed4c67 This commit introduces COLP/CONP and Redirecting party information into Asterisk.
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.

A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.

This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.

(closes issue #8824)
Reported by: gareth

Review: http://reviewboard.digium.com/r/201



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03 22:41:46 +00:00
Richard Mudgett 9fd753a30e Merged revisions 185121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line
  
  Update the channel allocation method documentation.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-30 20:42:14 +00:00
Richard Mudgett 1678a005b6 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/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/trunk@138738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-18 21:07:28 +00:00
Richard Mudgett b92df4dc1e Merged revisions 136241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008) | 5 lines

*  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/trunk@136594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-07 19:01:03 +00:00
Christian Richter 2a0b16b663 Merged revisions 89173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | 1 line

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/trunk@89179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12 13:36:45 +00:00
Christian Richter c9b8afb447 Merged revisions 89169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | 1 line

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/trunk@89174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12 12:49:19 +00:00
Mark Michelson 6ed072cb5a Merged revisions 82091 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r82091 | mmichelson | 2007-09-10 10:02:12 -0500 (Mon, 10 Sep 2007) | 5 lines

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/trunk@82092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-10 15:05:13 +00:00
Christian Richter 090cbd2945 added general Jitterbuffer Implementation. #9960
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-05 07:45:21 +00:00
Christian Richter 1fe0e3d192 Merged revisions 49313 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines

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/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
Christian Richter f19300635f Merged revisions 46351-46353 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r46351 | crichter | 2006-10-27 11:49:20 +0200 (Fr, 27 Okt 2006) | 9 lines

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
........

................
r46352 | crichter | 2006-10-27 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line

fixed not compile issue, which was just introduced
................
r46353 | crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines

Merged revisions 46350 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line

fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c
........

................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-27 11:18:32 +00:00
Christian Richter e09ad744af Merged revisions 44561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r44561 | crichter | 2006-10-06 14:50:25 +0200 (Fr, 06 Okt 2006) | 9 lines

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/trunk@44841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-11 08:34:03 +00:00
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
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