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2006-11-24Merged revisions 47968 via svnmerge from crichter1-2/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE. beatufied some logs, changed some loglevels. changed the default value of block_on_alarm ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47989 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-27fixed not compile issue, which was just introducedcrichter1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@46352 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-27Merged revisions 46176 via svnmerge from crichter1-0/+3
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: http://svn.digium.com/svn/asterisk/branches/1.4@46351 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-06Merged revisions 44334 via svnmerge from crichter1-0/+4
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/branches/1.4@44561 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-09fixing compile warnings, renaming config option "overlap_dial" to "overlapdial"nadi1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39479 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-08* first bits of decoding facility information elementsnadi1-13/+41
* fail on misdn_cfg_init() if elements in the config enum don't match with the config structs in misdn_config.c * implemented first bits for encoding ISDN facility information elements via ASN.1 descriptions * using unnamed semaphore for syncing in misdn_thread * advanced fax detection: configurable detect timeout and context to jump into git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39378 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-03* removed pp_l2_check (fixed L2 bug in mISDNuser)crichter1-10/+17
* added blocking flag to stack object. A port can be blocked/unblocked from the cli * added EVENT_PORT_ALARM to send alarm infos to the chan_misdn.c layer (later we can add a manager event for that) * added block_on_alarm option, to block the port whenever a ALARM occurs * added need_busy flag to indicate if we've sended a CONTROL_BUSY already * changed a bunch of cb_log(-1,..) to cb_log(0,..) due to funny behaviour in recent asterisk ast_log messages.. * fixed a few ETSI state violations, especially when finishing calls in different seldom states * changed debug levels a lot to make the log more readable in low debuglevels * some first fixes for the HOLD/RETRIEVE stuff (doesn't work totally still) * removed the PRECONNECTED state stuff * added cause 27 when we get a CLEANUP directly after a outgoing SETUP, this creates a CHANISUNAVAIL instead of a NOANSWER * removed the addr pointer from "misdn show stacks" that's not needed anymore and makes the output more unreadable * added cause saving on RELEASE/RELEASE_COMPLETE * set cause to 16 on prepare_bc * removed stack getting from ph_control functions, we don't really need it there * added beroec api git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38801 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-13added even more statefulness for sending out ↵crichter1-0/+4
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
2006-07-11* Introducing a new way for the l1watcher thread using the ast_sched way. ↵crichter1-13/+20
Now l1watcher timeouts can be configured separately for every portgroup. * added a signal handler to allow waking up the misdn task thread (that may sleep in a poll call) via misdn_tasks_wakeup(). * overlap_dial functionality implemented. * fixes a bug which leads to a segfault after reordering config elements in the enum or struct git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37382 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-06* removed tone_indicate, we genrate only the dialtone by ourself (and the ↵crichter1-0/+4
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
2006-07-03added misdn show config description[s] to show all the possible misdn.conf ↵crichter1-50/+296
settings with a description in the CLI git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36865 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 ↵crichter1-3/+11
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
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵kpfleming1-7/+11
again :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32846 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-01added bearer capability reject support. we send release instead of ↵crichter1-0/+1
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
2006-05-23added a l1watcher timeout, therefore removed the old behaviour of guessing ↵crichter1-2/+3
the l1state. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29803 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-22added callcounters for incoming and outgoing callscrichter1-0/+5
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/+1
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
2006-04-27put 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/+1
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
2006-03-09added option to change the connected party number dialplan (ton)crichter1-0/+1
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 ↵crichter1-1/+1
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 configcrichter1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11334 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-22fixed a ETSI violation (after RELEASE we need to RELEASE_COMPLETE (network ↵crichter1-1/+1
side) one needs to upgread mISDNuser for that fix as well. also fixed the reload issue #6547 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10713 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15adde incoming_early_audio option, to avoid sending tone indications to the ↵crichter1-0/+1
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/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10225 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-14fixed the occasional no audio issue, still need deeper investigation .. ↵crichter1-1/+1
echotraining is off by default git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9882 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-11rename chan_misdn_config.c to misdn_config.crussell1-0/+769
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9643 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-05 05:46:04 +0000