sysmocom/asterisk
11
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity
8953 commits 141 branches 1481 tags 406 MiB
Commit graph

65 commits

Author SHA1 Message Date
Christian Richter
baad1c266c * fixed the busy state 
* fixed tone handling after ast_hangup was called
* optimized the tone_indication function
* removed warnings in favour of log debugs
* improved the round_robin method
* added logs for channel setting/emptying
* fixed channel forgot to set bug





git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-26 17:37:11 +00:00
Christian Richter
70a4c98a5a * added support for PID export/import to identify bridging in misdn_call already 
* fixed a few inband Alerting issues, sometimes we need to create alerting, some
times it's inband
* beautified the state debugging of misdn_hangup
* removed "real"  bchannel activating/deactivating in chan_misdn.c
* fixed "round_robin" bug when there's only 1 port
* added more informative prints when channel could not be created
* changed some warnings to notices
* reworked the whole bchannel state machine stuff, 
  it is now like in the examples of mISDNuser and therefore a lot easier, 
  and it is now harder to create bugs
* bchannel_activate/deactivate is now only called in setup/cleanup bc, 
  they may merge sometime
* it is very important to setup/cleanup the bchannels under the correct 
  conditions, especially in the NT Side we can only setup the bchannels 
  when we send a Message! 
  In the TE side we can only setup the bchannel when we received the channel 
  of course
	      


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-21 15:21:46 +00:00
Christian Richter
2302fa58d7 when we get a disconnect, we should stop the tones on te side.. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-19 09:44:04 +00:00
Christian Richter
cdbddf5892 we need to send DISCONNECT on Timeout in the ALERTING/PROCEEDING cases too 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-19 07:36:18 +00:00
Christian Richter
829583a3dc added TONE_HANGUP, also added Tone generation by mISDN_dsp for post hangup tones, therefore the TONE_X defines are in the isdn_lib.h now. changed a REALEASE to a DISCONNECT in NT Stack, to make tones available in that state. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-18 19:09:32 +00:00
Christian Richter
d86993b90b * fixed bchannel cleaning race condition with new bchannel state (CLEAN_REQUEST) 
* added early bridge-hook, so we know if we need to generate ringing or 
  can take it from the far end chan_misdn channel (if available)
* fixed the issue, that we may not activate the bchannel on PTMP, 
  when we receive ALERTING/PROCEEDING/PROGRESS, only on CONNECT. There might
  be other PTMP devices and we might disturb their bchannel. 



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-17 10:37:35 +00:00
Christian Richter
05eed16044 Fixed detection of dtmfs with bridged channels. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-08 09:51:13 +00:00
Kevin P. Fleming
472c1ca282 simplify autoconfig include mechanism (make tholo happy he can use lint again :-) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-07 18:54:56 +00:00
Christian Richter
7924c21e33 changed the clean_up_bc function to work proper again. fixed the print with the conference nr. when bridged 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-07 13:01:02 +00:00
Christian Richter
24818d53b1 added select before write to avoid deadlock on full buffer. added some defines for deadlock debugging. added code snippet for generating silence if we don't have data to write. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-06 09:40:41 +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
eefc9193e7 added EVENT_NEW_CHANNEL. We change the channel name now when we got the real channel, also changed name generation to new stringfield api 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-24 23:21:03 +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
39ac1a5b83 added a l1watcher timeout, therefore removed the old behaviour of guessing the l1state. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-23 19:40:16 +00:00
Christian Richter
adfae0960f * export_ies uses now _VAR else the vars are not copied to the dest chan 
* when receiving a connect from the NT Side we wait until we have the final
  l3id until we queue the answer to asterisk to avoid bridging conflicts
* when not bridged to misdn we had a segfault after receiving the connect 
  due to a strcasecmp bug.. this didn't happen before, cause we hadn't had
  the bridge before
* cleanup of the bchannels is queued now, due to possible race conditions
* added mISDN_clear_stack when cleaning the bchannel



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-23 12:38:06 +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
3fb070853b fixed bug that we've lost the orginator of the call.. now we can hear to disconnect indications again 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-17 15:29:01 +00:00
Christian Richter
a309a2db9f fixed very bad memory overwrite which leaded to segfaults on some systems. removed redundant buffer betweend mISDN reading thread and ast_read in favour of the already existing pipe, this clarifies the way a voice frame takes between mISDN and asterisk a lot. centralized debugging of NumberPlan. removed a compiler warning. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-16 14:34:21 +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
Christian Richter
7f58902bf3 use set_callerid to update callerid with prefix 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-02 14:26:40 +00:00
Christian Richter
63b0baa757 returning 128 emtpy Frames instead of null frame. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-02 14:00:37 +00:00
Tilghman Lesher
dfdf6e1f76 Don't return stack pointers from functions (especially since the code will attempt to free() them later) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-01 03:58:13 +00:00
Christian Richter
835dfe67b3 added an up-queue message mechanism to avoid buffer fillups in the kernel, also changed some strdups to ast_strdupa 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-29 22:56:00 +00:00
Kevin P. Fleming
574e14cbea Thanks to the fine work of Russell Bryant and Dancho Lazarov, we now have autoconf and menuselect tools for Asterisk! 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-24 17:11:45 +00:00
Kevin P. Fleming
2229586d71 more module loader related fixes 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-17 16:42:21 +00:00
Christian Richter
52cf26255a fixed some issues, that appear at higher load 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-10 16:50:20 +00:00
Christian Richter
cb6f278393 we send nearly everytime a RELEASE, only if we for sure know, that it's a TE and we did create the call we don't to hear the Inband Info 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-10 14:09:17 +00:00
Kevin P. Fleming
f10f427d49 since the module API is changing, it's a good time to const-ify the description() and key() return values 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-08 22:01:19 +00:00
Christian Richter
ece21c3019 sorry litle mistake 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-07 12:03:51 +00:00
Christian Richter
47b8610f70 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18192 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2006-04-07 09:46:38 +00:00
Christian Richter
7788d5061c fixed a bridging-endless-loop also fixed the rdnis is not exported issue 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-07 09:31:27 +00:00
Christian Richter
7028946605 some final fixes for cpn 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-05 15:54:15 +00:00
Christian Richter
ce8fc47ffc * advanced the error Handling routine for wrong frames from mISDN 
* added more code for connected party number handling
* fixed the portinfo program, it can now be used to test mISDN again



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-05 14:51:48 +00:00
Christian Richter
d7a839a299 fixed some essential segfaults happening under heavy load. Added a bit more error handling if we receive wrong data form mISDN (e.g. if we crashed before => handle_err() ). Added first steps to implement the parseing of the connected party number. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-04 19:09:26 +00:00
Christian Richter
a096fbd87d * removed unneeded bc->state field 
* added statefullness for bchannel activation/deactivation
* fixed a lot PCM bridging issues
* some debugging logs are now on a higher loglevel



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-03 19:17:59 +00:00
Matt O'Gorman
a5ece3388a Janitor work converting !ast_strlen_zero(a)?a:b 
to S_OR functions. from bug note 6805 with minor
modifications.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-03-27 19:31:54 +00:00
Christian Richter
bd4715f3bc fixed hicom busy bug, we now see if the systemphone is busy, also fixed the immediate=yes does not work anymore issue 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-03-27 10:13:11 +00:00
Christian Richter
939dda2d6c removed unneeded debugs in level=0 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-03-21 09:47:51 +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
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
2c1cd943d0 fixed #6642 in trunk as well, sorry i really didn't want to copy this issue ... grrrr 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-03-07 19:34:23 +00:00
Christian Richter
5cd1828467 when receiving disconnect with inband info, we don't send release immediately, we also don't inform asterisk with hangup or congestion, since we want to hear the inband message. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-03-07 12:17:35 +00:00
Christian Richter
bcbcdaa67c added misdn port down to send a L1 Deactivate to a NT Port. changed some default values regarding jitterbuffer. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-03-07 11:16:56 +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
61f1bf14fa added export and import of RDNIS channel variable 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11017 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-02-24 17:38:43 +00:00
Kevin P. Fleming
302c943001 add 'consumed' argument to ast_get_time_t, so callers can know how many characters were used in the parser 
update pbx_dundi to use ast_get_time_t
eliminate some compiler warnings


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-02-23 17:13:57 +00:00
Christian Richter
fdd33866b4 * changed some strncpys to ast_copy_str 
* added copying of some IEs (user,rate,urate) with channel vars
* added #6251 



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-02-22 21:59:46 +00:00
Christian Richter
c7e0abdfed fixed a ETSI violation (after RELEASE we need to RELEASE_COMPLETE (network side) one needs to upgread mISDNuser for that fix as well. also fixed the reload issue #6547 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-02-22 16:48:25 +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