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2010-05-17Enhancements to connected line and redirecting work.mmichelson1-0/+13
From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02fixes adaptive jitterbuffer configurationdvossel1-0/+7
When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249893 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.seanbright1-12/+12
Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197535 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.seanbright1-12/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197528 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-14Add outgoing_colp misdn.conf port parameter.rmudgett1-0/+7
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: http://svn.digium.com/svn/asterisk/trunk@194479 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-21Added CCBS/CCNR Party A support and enhanced COLP support.rmudgett1-3/+17
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: http://svn.digium.com/svn/asterisk/trunk@189735 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03This commit introduces COLP/CONP and Redirecting party information into ↵mmichelson1-85/+109
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: http://svn.digium.com/svn/asterisk/trunk@186525 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 185121 via svnmerge from rmudgett1-4/+5
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: http://svn.digium.com/svn/asterisk/trunk@185123 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-18channels/chan_misdn.crmudgett1-4/+4
* 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: http://svn.digium.com/svn/asterisk/trunk@138738 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Merged revisions 136241 via svnmerge from rmudgett1-4/+13
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: http://svn.digium.com/svn/asterisk/trunk@136594 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Merged revisions 89173 via svnmerge from crichter1-0/+7
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: http://svn.digium.com/svn/asterisk/trunk@89179 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Merged revisions 89169 via svnmerge from crichter1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@89174 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-10Merged revisions 82091 via svnmerge from mmichelson1-10/+0
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: http://svn.digium.com/svn/asterisk/trunk@82092 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-05added general Jitterbuffer Implementation. #9960crichter1-0/+26
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@73298 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-03Merged revisions 49313 via svnmerge from crichter1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@49321 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-27Merged revisions 46351-46353 via svnmerge from crichter1-0/+7
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: http://svn.digium.com/svn/asterisk/trunk@46354 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-11Merged revisions 44561 via svnmerge from crichter1-0/+11
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: http://svn.digium.com/svn/asterisk/trunk@44841 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-11Merged revisions 42716 via svnmerge from tilghman1-11/+11
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: http://svn.digium.com/svn/asterisk/trunk@42717 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-13added even more statefulness for sending out ↵crichter1-0/+9
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: http://svn.digium.com/svn/asterisk/trunk@37508 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-06* removed tone_indicate, we genrate only the dialtone by ourself (and the ↵crichter1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@37172 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-29added better L2 handling for ptp, if it's down we don't try to call on that ↵crichter1-0/+31
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: http://svn.digium.com/svn/asterisk/trunk@36298 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-01added bearer capability reject support. we send release instead of ↵crichter1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@31324 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-24fixed to early connect bug which came in yesterday.., also added the ↵crichter1-0/+14
transmit of progress indicators through channel vars git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29938 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-22added callcounters for incoming and outgoing callscrichter1-0/+15
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29411 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-05Added option far_alerting. This option makes it possible to generate a ↵crichter1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@24879 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-28note that group assignments must be from 0 to 63 (issue #7048)russell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@23177 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-27put the default misdn.trace to /var/log/asterisk/misdn.log for better ↵crichter1-1/+1
integration of existing log structure git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22795 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-20removed dynamic switching from transparent to hdlc mode. Instead we've got a ↵crichter1-0/+6
config option hdlc=yes now which enables the hdlc controller for a data call git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13637 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-20these traceing option do not exist anymorecrichter1-9/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13633 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-09added option to change the connected party number dialplan (ton)crichter1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@12481 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-07added a bit more detailed description for the echotraining parameter, also ↵crichter1-5/+9
changed the default from 1 to 2000. The default for the upper_threshold is now 0 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@12287 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-28better default values for jitterbuffer in code and configcrichter1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11334 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15adde incoming_early_audio option, to avoid sending tone indications to the ↵crichter1-0/+11
remote party on incoming calls from the pstn, this shouldnt be enabled, only if the provider allows it git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10227 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15added pmp_l1_check option, to avoid l1 checking for group calls on PMP portscrichter1-0/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10225 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-07default values of jitterbuffer and jitterbuffer_upper_threshold should be > ↵crichter1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@9186 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-02* removed unnecessary struct elements and functionscrichter1-36/+30
* 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: http://svn.digium.com/svn/asterisk/trunk@9114 f38db490-d61c-443f-a65b-d21fe96a405b
2005-12-12updated the documentation and the sample config to meet the presentcrichter1-12/+45
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7446 f38db490-d61c-443f-a65b-d21fe96a405b
2005-11-29git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7221 ↵kpfleming1-0/+0
f38db490-d61c-443f-a65b-d21fe96a405b
2005-11-01issue #5566kpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6938 f38db490-d61c-443f-a65b-d21fe96a405b
2005-10-31add experimental mISDN channel driver (issue #4077)kpfleming1-0/+267
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6910 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-23 14:24:31 +0000