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Author SHA1 Message Date
mmichelson
992c26d776 Merged revisions 82091 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82092 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-10 15:05:13 +00:00
crichter
177aff2f30 added general Jitterbuffer Implementation. #9960
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@73298 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-05 07:45:21 +00:00
crichter
85c5dfacde Merged revisions 49313 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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: http://svn.digium.com/svn/asterisk/trunk@49321 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-03 11:15:02 +00:00
crichter
0131a92301 Merged revisions 46351-46353 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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

fixed not compile issue, which was just introduced
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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

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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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46354 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-27 11:18:32 +00:00
crichter
9f4655a71a Merged revisions 44561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44841 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-11 08:34:03 +00:00
tilghman
9b9ca6cac5 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: http://svn.digium.com/svn/asterisk/trunk@42717 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-11 16:41:49 +00:00
crichter
64ca850133 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: http://svn.digium.com/svn/asterisk/trunk@37508 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-13 14:13:24 +00:00
crichter
cb375e9152 * 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: http://svn.digium.com/svn/asterisk/trunk@37172 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-06 15:11:40 +00:00
crichter
0423e3b72b 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: http://svn.digium.com/svn/asterisk/trunk@36298 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-29 20:12:19 +00:00
crichter
f6aed90150 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: http://svn.digium.com/svn/asterisk/trunk@31324 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-01 12:51:41 +00:00
crichter
3f167cb7d7 fixed to early connect bug which came in yesterday.., also added the transmit of progress indicators through channel vars
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29938 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-24 07:58:52 +00:00
crichter
0430160715 added callcounters for incoming and outgoing calls
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29411 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-22 15:02:03 +00:00
crichter
d84eb5f102 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: http://svn.digium.com/svn/asterisk/trunk@24879 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-05 16:38:15 +00:00
russell
3aec7a9bd5 note that group assignments must be from 0 to 63 (issue #7048)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@23177 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-28 16:42:42 +00:00
crichter
7b26d95886 put the default misdn.trace to /var/log/asterisk/misdn.log for better integration of existing log structure
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22795 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-27 08:23:53 +00:00
crichter
24baf4cf91 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: http://svn.digium.com/svn/asterisk/trunk@13637 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-20 18:04:05 +00:00
crichter
2bf55bda4d these traceing option do not exist anymore
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13633 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-20 10:00:34 +00:00
crichter
7b5c69e01b added option to change the connected party number dialplan (ton)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@12481 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-09 18:01:27 +00:00
crichter
f320a81034 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: http://svn.digium.com/svn/asterisk/trunk@12287 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-07 11:08:09 +00:00
crichter
df18d5fb04 better default values for jitterbuffer in code and config
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11334 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-28 11:46:55 +00:00
crichter
8e9b1bb203 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: http://svn.digium.com/svn/asterisk/trunk@10227 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15 19:51:33 +00:00
crichter
2913414dba added pmp_l1_check option, to avoid l1 checking for group calls on PMP ports
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10225 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15 19:32:45 +00:00
crichter
8e35b8224f 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: http://svn.digium.com/svn/asterisk/trunk@9186 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-07 13:34:59 +00:00
crichter
870f55a59b * 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: http://svn.digium.com/svn/asterisk/trunk@9114 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-02 21:15:34 +00:00
crichter
ff7c3adedd updated the documentation and the sample config to meet the present
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7446 f38db490-d61c-443f-a65b-d21fe96a405b
2005-12-12 22:26:35 +00:00
kpfleming
24c1e3c222 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7221 f38db490-d61c-443f-a65b-d21fe96a405b 2005-11-29 18:24:39 +00:00
kpfleming
733158c357 issue #5566
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6938 f38db490-d61c-443f-a65b-d21fe96a405b
2005-11-01 22:04:14 +00:00
kpfleming
3021b77d12 add experimental mISDN channel driver (issue #4077)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6910 f38db490-d61c-443f-a65b-d21fe96a405b
2005-10-31 22:51:12 +00:00