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* Miscellaneous doxygen comments added.
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Intrepid, so all developers will see the same warnings and errors
since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
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https://origsvn.digium.com/svn/asterisk/team/group/issue8824
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Fixes JIRA ABE-1726
The dial extension could be empty if you are using MISDN_KEYPAD
to control ISDN provider features.
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mISDN sets the channel to the wrong state when it receives
the indication AST_CONTROL_RINGING.
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channels/misdn/isdn_lib.c
* Miscellaneous other fixes from trunk to make merging easier later.
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r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines
* Miscellaneous formatting changes to make v1.4 and trunk
more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
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r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
improved helptext of misdn_set_opt.
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r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
Cleaned up comment
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r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
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/misdn.txt (doc/tex/misdn.tex)
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
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of its options: digital_restricted.
* Fixed some other spelling errors and typos.
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should continue working. Release announcement to follow.
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(closes issue #11958)
Reported by: norman
Patches:
20080209__bug11958.diff.txt uploaded by Corydon76 (license 14)
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possibly cause memory to be accessed after it is freed, which causes all
sorts of random memory corruption. Instead, if a deletion fails, wait a
bit and try again (noting that another thread could change our taskid
value).
(closes issue #11386)
Reported by: flujan
Patches:
20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, flujan, stuarth`
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If we don't make it clean, it happens that one way audio occurs..
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now we just jump into the s extension, which makes a lot more sense.
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starting.
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ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option.
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RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89172 f38db490-d61c-443f-a65b-d21fe96a405b
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removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89171 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
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not match after an overlap call. Also added out_cause=1
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with --enable-dev-mode so this doesn't happen again ...)
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channels)
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changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)
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instead of the mISDN_dsp ones. also added the patch from irroot #10190, so that dtmf tones detected by the asterisk detector are passed outofband to asterisk, to make any use of dtmf tones at all.
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not return because it blocks. later chan_misdn would try to queue a frame like
a AST_CONTROL_ANSWER which could result in a deadlock situation. misdn_read
will now not block forever anymore, and we don't queue the ANSWER frame at all
when we already was called with misdn_answer -> answer would be called twice.
Also we don't explicitly send a RELEASE_COMPLETE on receiption of a RELEASE
anymore, because mISDN does that for us, this resulted in a problem on some
switches, which would block our port after some calls for a short while.
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to go through as inband. Otherwise they might be generated by the other channel partner and then there is a double tone.
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Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79174 f38db490-d61c-443f-a65b-d21fe96a405b
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the disconnect message.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04 Jul 2007) | 1 line
bad bug in overlapdial case, we called start_pbx multiple times, because the state wasn't changed..
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03 Jul 2007) | 1 line
fixed issue, that misdn_l2l1_check could only be called from mISDN Source channels.. #9449
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) | 1 line
check if the bchannel stack id is already used, if so don't use it a second time. Also added a release_chan lock, so that the same chan_list object cannot be freed twice. chan_misdn does not crash anymore on heavy load with these changes.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) | 1 line
simplified generation for dummy bchannels, also we mark them as dummies, so they are not used later as real-bchannels, optimized the RESTART mechanisms, we block a channel now on cause:44, and send out a RESTART automatically, then on reception of RESTART_ACKNOWLEDGE we unblock the channel again.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20 Jun 2007) | 1 line
fixed a bug that was introduced by copy and paste in the last commit ..bchannels weren't cleaned properly.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) | 1 line
we activate the bchannels in TE mode on incoming calls only when we want to connect the call.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20 Jun 2007) | 1 line
on receiption of cause:44 we mark the channel as in use and inform the user about the situation, we need to test the RESTART stuff then. Also shuffled the empty_chan_in_stack function after the bchannel cleaning functions, to avoid race conditions.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19 Jun 2007) | 1 line
when we send out a SETUP, but get no response, we should cleanup everything after reception of a hangup.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12 Jun 2007) | 1 line
if the bridged partner is mISDN too we should not send dtmf tones, they are transmitted inband always
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12 Jun 2007) | 1 line
if we have already some digits, we just stop the tones.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) | 1 line
added check for NULL Pointer when calling misdn_new. Asterisk does not allow us to create channels anymore when stop gracefully is used :). also modified the restart_indicator to 0
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) | 1 line
fixed problem that the dummybc chanels had no lock, checking for the lock now. Also fixed the channel restart stuff, we can now specify and restart particular channels too.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 Jun 2007) | 1 line
simplified the EVENT_SETUP handling in the cb_events function a lot. Commented the different possibilities a bit and made functions of shared code. When the dialed extension does not exist in the extensions.conf we'll jump into the 'i' extension if this does exist, else we disconnect the call with the cause:1 = No Route to Destination.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | 1 line
added possibility to deactivate bridging per port
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