https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines
Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor.
After talking to rmudgett about some of his recent iflist locking changes, it
was determined that the only place that would destroy a channel without being
explicitly to do so was in handle_init_event. The loop to walk the interface
list has been modified to wait to destroy the channel until the dahdi_pvt of
the channel to be destroyed is no longer needed.
(closes issue #15378)
Reported by: samy
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@218430 f38db490-d61c-443f-a65b-d21fe96a405b
* Fixed walking the iflist so it is always done with the iflock locked.
* Simplified iflist walking routines.
* Created chan_dahdi iflist insertion and extraction routines.
* Fixed duplicate_pseudo() malloc fail handling.
* Fixed infinite loop in action_dahdishowchannels() when showing a single channel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217912 f38db490-d61c-443f-a65b-d21fe96a405b
This fixes the problem that would occur if an analog phone was picked up while
the caller id was being sent. The caller id before sent the whole spill even
after pickup and is now corrected.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217744 f38db490-d61c-443f-a65b-d21fe96a405b
Added detection of DTMF tone energy levels on FXO channels in chan_dahdi
monitoring loop so DTMF CID can be detected without the need of a polarity
change precursor.
(closes issue #9096)
Reported by: fleed
Patches:
9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819)
Tested by: cyberplant, sum, maturs
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@216094 f38db490-d61c-443f-a65b-d21fe96a405b
made analog_set_linear_mode return back the mode that was being overwritten
so it could be restored later.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@215608 f38db490-d61c-443f-a65b-d21fe96a405b
* confirmanswer option now respected
* check and set waiting for dialtone timer
* unneeded needcallerid flag removed from analog_subchannel
* ss_astchan does not need to be a void pointer
* swap_channels callback updated to trunk
* analog_hangup now resets channel to default law
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212287 f38db490-d61c-443f-a65b-d21fe96a405b
dahdi_read relies on the dahdi_pvt copy of ringt which was not getting set
in sig_analog. This patch adds a callback to do so.
(closes issue #15288)
Reported by: alecdavis
Patches:
chan_dahdi.ringtimeout.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211908 f38db490-d61c-443f-a65b-d21fe96a405b
Created the dahdi_sig_pri_lib_handles() function and
SIG_PRI_LIB_HANDLE_CASES macro to simplify testing for which signaling
styles are handled by sig_pri.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211675 f38db490-d61c-443f-a65b-d21fe96a405b
Putting some DSP code back into sig_pri that was removed by the
chan_dahdi/sig_pri reorganization.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211392 f38db490-d61c-443f-a65b-d21fe96a405b
You cannot cast the struct dahdi_pvt.sig_pvt pointer to a specific
signaling private pointer without first checking that it is in fact
pointing to the correct signaling private structure.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211197 f38db490-d61c-443f-a65b-d21fe96a405b
* Moved SUPPORT_USERUSER to sig_pri.c
* Fix PRI_DEADLOCK_AVOIDANCE parameter.
* Whitespace changes.
* Added missing unlock in pri_dchannel():PRI_EVENT_RING case.
* Balanced curly braces.
* ast_debug/ast_log changes from chan_dahdi.
* sig_pri_indicate() should default to return -1 if the indication is not
handled.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210154 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines
Minor changes inspired by testing with latest GCC.
The latest GCC (what will become 4.5.x) has a few new warnings, that in these
cases found some either downright buggy code, or at least seriously poorly
designed code that could be improved.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209760 f38db490-d61c-443f-a65b-d21fe96a405b
The wrong logic was being used in chan_dahdi to convert a sig_pri_chan
to the proper libpri channel number. The most significant bit must only
be set only when trunk groups are being used.
(closes issue #15452)
Reported by: alecdavis
Patches:
bug15452.patch uploaded by jpeeler (license 325)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208267 f38db490-d61c-443f-a65b-d21fe96a405b
Pointed out by Matt F.
Also in the case of not using a signaling module, set the law back to the
default as well.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208155 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines
Wait for wink before dialing when using E&M wink signaling
There was already code for other signaling types in dahdi_handle_event to
handle dialing if a dial operation dial string was present. Simply add
SIG_EMWINK to the list.
(closes issue #14434)
Reported by: araasch
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207854 f38db490-d61c-443f-a65b-d21fe96a405b
Sig_analog handles allocating the sub channel for callwaiting, so no longer try
to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as
allocated upon success of the alloc_sub callback, which was responsible for the
segfault. Also, the callwaiting and callwaitingcallerid options were being
unconditionally set to true. Now, the options are properly set from
chan_dahdi.conf.
(closes issue #15508)
Reported by: elguero
Tested by: elguero
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206998 f38db490-d61c-443f-a65b-d21fe96a405b
This readds the proper setting of the flag and is really a continuation of
r205731. The flag was being set properly in sig_analog, but use of the
newly added set_dialing callback allowed for some simplification in
chan_dahdi.
(closes issue #15486)
Reported by: rmudgett
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206767 f38db490-d61c-443f-a65b-d21fe96a405b
The main purpose of this commit is to restore missing functionality present in
the ss_thread before all the sig related work was done. Two of the biggest
missing things were distinctive ring detection and cid handling for V23.
fxsoffhookstate and associated mwi variables have been moved inside sig_analog
as they were not being set properly as well.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206566 f38db490-d61c-443f-a65b-d21fe96a405b
Also go back and wrap all of the places that use the specific reverse charge
APIs with preprocessor conditionals.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204919 f38db490-d61c-443f-a65b-d21fe96a405b
This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse
Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse
Charging Indication in LibPRI. This patch adds the ability to specify RCI on
the outbound leg of a PRI call from within Asterisk, by prefixing the dialed
number with a capital 'C' like:
...,Dial(DAHDI/g1/C4445556666)
And to read it off an inbound channel:
exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
Thanks again to rmudgett for the thorough review.
(closes issue #13760)
Reported by: mrgabu
Review: https://reviewboard.asterisk.org/r/303/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204749 f38db490-d61c-443f-a65b-d21fe96a405b
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
tzafrir