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: https://origsvn.digium.com/svn/asterisk/trunk@208267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@208155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@206998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@206767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@206566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also go back and wrap all of the places that use the specific reverse charge
APIs with preprocessor conditionals.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@204749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into
sig_pri.c. Functionality in theory should not change (mostly). A few trivial
changes were made in sig_analog with verbose messages and commenting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Moved SETCADENCE ioctl call to before call into new analog signal module
to insure that it gets set.
(closes issue #15381)
Reported by: alecdavis
Patches:
fix15381.diff uploaded by dbailey (license 819)
Tested by: dbailey
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines
Improved chan_dahdi.conf pritimer error checking.
Valid format is: pritimer=timer_name,timer_value
* Fixed segfault if the ',' is missing.
* Completely check the range returned by pri_timer2idx() to prevent
possible access outside array bounds.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some applications (notably app_fax) do not need digit detection nor FAX tone
detection while they are running, and if Asterisk is using software DSPs to provide
the detection, this consumes extra CPU cycles that could be better spent on the
actual application. This patch allows applications to query and control the state
of digit and tone detection on a channel, and modifies app_fax to disable them
while the FAX operations are occurring (and re-enable digit detection afterwards).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These modules all contained variables that are module-global but not system-global,
but were not marked 'static'.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since a DAHDI channel may belong to multiple groups, we need to use
a bitwise and instead of equivalence to determine whether to display
the channel information.
(closes issue #15248)
Reported by: gentian
Patches:
15248.patch uploaded by mmichelson (license 60)
Tested by: gentian
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This branch splits all the analog signaling logic out of chan_dahdi.c into
sig_analog.c. Functionality in theory should not change at all. As noted
in the code, there is still some unused code remaining that will be cleaned
up in a later commit.
Review: https://reviewboard.asterisk.org/r/253/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This feels like a sane change (wouldn't compile without this addition), but I'm
not intimately familiar with this code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191739 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds ability for CHANNEL() dialplan function, when used on DAHDI channels,
to temporarily change the number of buffers and/or the buffer policy, and also
to enable, disable, or switch the echo canceller between FAX/data and voice
modes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This function incorrectly reported success even if the option was
unsupported. This was exposed by the options to change the underlying
channel format. The function now returns a failure if the option
is unsupported.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines
National prefix inserted even when caller ID not available
When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank.
(closes issue #13207)
Reported by: shawkris
Patches:
national_prefix.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/220/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is the companion commit to libpri r732. Service messages are now supported
for switch types 4ess/5ess. A new option service_message_support has been added
to chan_dahdi.conf and is noted in the sample config file. The service message
support is turned off by default. The current implementation relies on AstDB
to keep track of channel state, which allows the statuses to be preserved
across Asterisk restarts. Below is a description of the storage format.
The state and reason for the service state are in the form <state>:<reason>,
where:
<state> ::= { 'O' } // 'O' – Out Of Service
<reason> ::= { '0' | '1' | '2' | '3' }, where:
'0' – No reason (backwards compatibility)
'1' – NEAR END
'2' – FAR END
'3' – both NEAR and FAR END
The new CLI commands to handle channel service state are:
pri service disable channel <chan>
pri service enable channel <chan>
Many people contributed to the development of this functionality. Because I
entered at the very end I do not know the exact history. Special thanks to
all who moved the bug forward one way or another:
cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7,
tilghman, lmadsen, and especially dhubbard (he answered lots of my questions
and did a large portion of the work)
(closes issue #3450)
Reported by: cmaj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
