Commit Graph

107 Commits

Author SHA1 Message Date
Corey Farrell
e04607f8a3 res_smdi: convert to astobj2
Remove functions:
	ast_smdi_interface_unref
	ast_smdi_md_message_putback
	ast_smdi_mwi_message_putback
	ast_smdi_md_message destructor
	ast_smdi_mwi_message destructor

Includes for astobj.h are removed everywhere it's possible.

ASTERISK-24066 #close
Review: https://reviewboard.asterisk.org/r/3758/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-21 08:41:29 +00:00
Jonathan Rose
d00882108f res_pjsip_refer: Fix bugs involving Parking/PJSIP/transfers
PJSIP would never send the final 200 Notify for a blind transfer
when transferring to parking. This patch fixes that. In addition,
it fixes a reference leak when performing blind transfers to
non-bridging extensions.

Review: https://reviewboard.asterisk.org/r/3485/
........

Merged revisions 414400 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22 15:52:30 +00:00
Kinsey Moore
abd3e4040b Allow Asterisk to compile under GCC 4.10
This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
........

Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-09 22:49:26 +00:00
Kevin Harwell
28c0cb28d0 channel locking: Add locking for channel snapshot creation
Original commit message by mmichelson (asterisk 12 r403311):

"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."

The above was initially committed and then reverted at r403398.  The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed.  Fixed by unreffing the channels.

Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel.  Fixed by
unlocking "other->chan"

(closes issue ASTERISK-22709)
Reported by: John Bigelow
........

Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-18 20:33:37 +00:00
David M. Lee
1212906351 Reverting r403311. It's causing ARI tests to hang.
........

Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-05 22:10:20 +00:00
Mark Michelson
8e8b329e14 Add channel locking for channel snapshot creation.
This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
........

Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03 17:07:29 +00:00
Richard Mudgett
7d17d5fb04 chan_dahdi: Fix analog parking using flash-hook.
Transferring an analog call using a flash-hook to parking would fail to
park the call and result in an invalid ao2 object unref.

* Park the correct bridged channel.
........

Merged revisions 400236 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-01 21:19:13 +00:00
Richard Mudgett
868be02a2f Fix uninitialized value in struct ast_control_pvt_cause_code usage.
........

Merged revisions 397744 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 397745 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-27 16:51:08 +00:00
Kinsey Moore
56aea1c030 Allow res_parking to be unloadable
This change protects accesses of res_parking such that it can unload
safely once transient uses of its registered functions are complete.
The parking API has been restructured such that its consumers do not
have access to the vtable exposed by the parking provider, but instead
route through stubs to prevent consumers from holding on to function
pointers.

This adds calls to all the parking unload functions and moves
application loading and unloading into functions in
parking_applications.c similar to the rest of the parts of res_parking.

Review: https://reviewboard.asterisk.org/r/2763/
(closes issue ASTERISK-22142)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 15:01:54 +00:00
David M. Lee
4f07502000 Fixed chan_dahdi compilation failure
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 04:48:12 +00:00
Matthew Jordan
5c4b482471 Support externally initiated parking requests; remove some dead code
This patch does the following:
 * It adds support for externally initiated parking requests. In particular,
   chan_skinny has a protocol level message that initiates a call park.
   This patch now supports that option, as well as the protocol specific
   mechanisms in chan_dahdi/sig_analog and chan_mgcp.
 * A parking bridge features virtual table has been added that provides
   access to the parking functionality that the Bridging API needs. This
   includes requests to park an entire 'call' (with little or no additional
   information, thank you chan_skinny), perform a blind transfer to a parking
   extension, determine if an extension is a parking extension, as well as the
   actual "do the parking" request from the Bridging API.
 * Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new
   functions
 * The removal of some - but not all - dead parking code from features.c

This also fixed blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the parking features
kK might not have worked)

Review: https://reviewboard.asterisk.org/r/2710

(closes issue ASTERISK-22134)
Reported by: Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01 20:55:17 +00:00
Matthew Jordan
cafc115896 A great big renaming patch
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.

A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.

(closes issue ASTERISK-22130)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-25 04:06:32 +00:00
Kinsey Moore
684c83b29b Add transfer support to CEL
This adds CEL support for blind and attended transfers and call pickup.
During the course of adding this functionality I noticed that
CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly
useless without a bridge identifier, so I added that as well.

This adds tests for blind transfers, several types of attended
transfers, and call pickup.

The extra field in CEL records now consists of a JSON blob whose fields
are defined on a per-event basis.

Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-20 13:10:22 +00:00
Kinsey Moore
909ee4bfb9 Refactor extraneous channel events
This change removes JitterBufStats, ChannelReload, and ChannelUpdate
and refactors the following events to travel over Stasis-Core:
* LocalBridge
* DAHDIChannel
* AlarmClear
* SpanAlarmClear
* Alarm
* SpanAlarm
* DNDState
* MCID
* SIPQualifyPeerDone
* SessionTimeout

Review: https://reviewboard.asterisk.org/r/2627/
(closes issue ASTERISK-21476)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-01 13:16:09 +00:00
Richard Mudgett
6114166237 Refactor chan_dahdi/sig_analog/sig_pri and chan_misdn to use the common transfer functions.
(closes issue ASTERISK-21523)
Reported by: Matt Jordan

(closes issue ASTERISK-21524)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2600/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-07 01:06:49 +00:00
Mark Michelson
2dc8a06006 Refactor the features configuration scheme.
Features configuration is handled in its own API in
features_config.h and features_config.c. This way, features
configuration is accessible to anything that needs it.

In addition, features configuration has been altered to
be more channel-oriented. Most callers of features API
code will be supplying a channel so that the individual
channel's settings will be acquired rather than the global
setting.

Missing from this commit is XML documentation for the
features configuration. That will be handled in a separate
commit.

Review: https://reviewboard.asterisk.org/r/2578/

(issue ASTERISK-21542)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-06 21:40:35 +00:00
Jason Parker
154fbf8cae Split Hold event into Hold/Unhold, and move it into core.
(closes issue ASTERISK-21487)
Review: https://reviewboard.asterisk.org/r/2565/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 21:21:25 +00:00
Richard Mudgett
5ab5715646 Remove some unnecessary calls to ast_bridged_channel() in chan_dahdi.c/sig_analog.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01 20:00:31 +00:00
Mark Michelson
f2bb9afe17 Multiple revisions 375993-375994
........
  r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
  
  Fix misuses of timeouts throughout the code.
  
  Prior to this change, a common method for determining if a timeout
  was reached was to call a function such as ast_waitfor_n() and inspect
  the out parameter that told how many milliseconds were left, then use
  that as the input to ast_waitfor_n() on the next go-around.
  
  The problem with this is that in some cases, submillisecond timeouts
  can occur, resulting in the out parameter not decreasing any. When this
  happens thousands of times, the result is that the timeout takes much
  longer than intended to be reached. As an example, I had a situation where
  a 3 second timeout took multiple days to finally end since most wakeups
  from ast_waitfor_n() were under a millisecond.
  
  This patch seeks to fix this pattern throughout the code. Now we log the
  time when an operation began and find the difference in wall clock time
  between now and when the event started. This means that sub-millisecond timeouts
  now cannot play havoc when trying to determine if something has timed out.
  
  Part of this fix also includes changing the function ast_waitfor() so that it
  is possible for it to return less than zero when a negative timeout is given
  to it. This makes it actually possible to detect errors in ast_waitfor() when
  there is no timeout.
  
  (closes issue ASTERISK-20414)
  reported by David M. Lee
  
  Review: https://reviewboard.asterisk.org/r/2135/
........
  r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
  
  Remove some debugging that accidentally made it in the last commit.
........

Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10
........

Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-07 19:15:26 +00:00
Richard Mudgett
b5138fccf4 Add pause one second W dial modifier.
* The following dialplan applications now recognize 'W' to pause sending
DTMF for one second in addition to the previously existing 'w' that paused
sending DTMF for half a second.  Dial, ExternalIVR, and SendDTMF.

* The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now
distinguishes between 'w' and 'W'.  The 'w' pauses dialing for half a
second.  The 'W' pauses dialing for one second.

* Created dahdi_dial_str() in chan_dahdi that eliminated a lot of
duplicated dialing code and diagnostic messages for the channel driver.

(closes issue ASTERISK-20039)
Reported by: Jeremiah Gowdy
Patches:
      jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy
      Expanded patch to add support in chan_dahdi.
Tested by: rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-28 18:27:02 +00:00
Kinsey Moore
76d642ff69 Add HANGUPCAUSE information to callee channels
This adds HANGUPCAUSE information to called channels so that hangup
handlers can, in conjunction with predial dialplan execution, access
the hangupcause information when the dialed channel hangs up on a
one-to-one basis instead of a many-to-one basis as with HANGUPCAUSE
usage on the caller channel.

Review: https://reviewboard.asterisk.org/r/2069/
(closes issue ASTERISK-20198)
........

Merged revisions 371258 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-15 17:56:04 +00:00
Richard Mudgett
062becab80 Convert sig_analog to use a global callback table.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-08 20:32:53 +00:00
Richard Mudgett
f1dce57742 Fix the analog dial *0 flash-hook of bridged peer feature.
The flash-hook the bridged peer feature now correctly determines if the
bridged peer is another chan_dahdi channel, that it is an analog channel,
and that it has the correct signaling for an FXO port.  It now also
flash-hooks the correct channel.
........

Merged revisions 370900 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370901 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-08 20:17:02 +00:00
Kinsey Moore
9b16c8b0f6 Clean up and ensure proper usage of alloca()
This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().

(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
........

Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31 20:21:43 +00:00
Kinsey Moore
cb9756daa2 Add hangupcause translation support
The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now
been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan
functions to better facilitate access to the AST_CAUSE translations
for technology-specific cause codes. The HangupCauseClear application
has also been added to remove this data from the channel.

(closes issue SWP-4738)
Review: https://reviewboard.asterisk.org/r/2025/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-20 15:48:55 +00:00
Richard Mudgett
9773d2351b Add missing ast_hangup() calls on some analog exception paths.
Make starting analog_ss_thread() or __analog_ss_thread() failure paths
hangup the channel.
........

Merged revisions 370017 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370025 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12 20:28:07 +00:00
Matthew Jordan
9bc2127d7b Fix validation errors when producing documentation using default build script
The awk script parses out the first instance of the DOCUMENTATION tag that it
finds within a file.  If a file did not previously have a DOCUMENTATION tag
but received one due to it having an AMI event, then the XML fragment
associated with the AMI event was erroneously placed in the resulting XML
file.  Without the python scripts, these XML fragments will not validate.

This patch adds DOCUMENTATION tags at the top of those files that did
not previously have them to prevent the awk script from pulling AMI event
documentation.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11 02:06:05 +00:00
Matthew Jordan
2ffae5745d Add some additional documentation for core AMI events
This patch adds some basic documentation for a number of modules.  This
includes core source files in Asterisk (those in main), as well as
chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri.  The DTD
has also been updated to allow referencing of AMI commands.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10 22:26:27 +00:00
Kinsey Moore
db59a3f123 Remove unnecessary generation of informational cause frames
It is not necessary to generate information cause code frames on every
protocol event that occurs.  This removes all the instances where the
frame was not conveying a cause code and was instead just conveying a
protocol-specific message.  This also corrects the generation of the
message associated with disconnects for MFC/R2 to use the MFC/R2
specific text for the disconnect cause.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-06 22:03:44 +00:00
Jonathan Rose
5eb94d7ebb Unique Call ID logging Phases III and IV
Adds call ID logging changes to specific channel drivers that weren't handled
handled in phase II of Call ID Logging. Also covers logging for threads for
threads created by systems that may be involved with many different calls.
Extra special thanks to Richard for rigorous review of chan_dahdi and its
various signalling modules.

review: https://reviewboard.asterisk.org/r/1927/
review: https://reviewboard.asterisk.org/r/1950/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-26 21:45:22 +00:00
Kinsey Moore
1ab47ac137 Add HANGUPCAUSE hash support for analog and PRI DAHDI subtechs
This is part of the DAHDI support for the Asterisk 11 "Who Hung Up?"
project and covers the implementation for the technologies implemented
in sig_analog.c and sig_pri.c. Tested on a local machine to verify
protocol and cause information is available.

Review: https://reviewboard.asterisk.org/r/1953/
(issue SWP-4222)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22 15:10:38 +00:00
Kevin P. Fleming
166b4e2b30 Multiple revisions 369001-369002
........
  r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
  
  Add support-level indications to many more source files.
  
  Since we now have tools that scan through the source tree looking for files
  with specific support levels, we need to ensure that every file that is
  a component of a 'core' or 'extended' module (or the main Asterisk binary)
  is explicitly marked with its support level. This patch adds support-level
  indications to many more source files in tree, but avoids adding them to
  third-party libraries that are included in the tree and to source files
  that don't end up involved in Asterisk itself.
........
  r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
  
  Add a script to enable finding source files without support-levels defined.
........

Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15 16:20:16 +00:00
Richard Mudgett
72eb8eb1e7 Fix deadlock potential with ast_set_hangupsource() calls.
Calling ast_set_hangupsource() with the channel lock held can result in a
deadlock because the function also locks the bridged channel.

(issue ASTERISK-19537)

(closes issue AST-891)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter

(closes issue ASTERISK-19801)
Reported by: Alec Davis
........

Merged revisions 368759 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 368760 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-11 17:34:08 +00:00
Richard Mudgett
0f71b29e2f Fix POTS flash hook to orignate a second call deadlock.
A deadlock can occur when a POTS phone tries to flash hook to originate a
second call for 3-way or transfer.  If another process is scanning the
channels container when the POTS line flash hooks then a deadlock will
occur.

* Release the channel and private locks when creating a new channel as a
result of a flash hook.

(closes issue ASTERISK-19842)
Reported by: rmudgett
Tested by: rmudgett
........

Merged revisions 368644 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 368645 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-06 21:34:10 +00:00
Jonathan Rose
8227f70cd7 Coverity Report: Fix issues for error type CHECKED_RETURN for core
(issue ASTERISK-19658)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1905/
........

Merged revisions 366094 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 366106 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-10 18:35:14 +00:00
Terry Wilson
786f5898d1 Finalize ast_channel opaquification
Review: https://reviewboard.asterisk.org/r/1786/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13 18:20:34 +00:00
Terry Wilson
0e5c761c28 Opaquify ast_channel typedefs, fd arrays, and softhangup flag
Review: https://reviewboard.asterisk.org/r/1784/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-01 22:09:18 +00:00
Terry Wilson
a9d607a357 Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-29 16:52:47 +00:00
Terry Wilson
57f42bd74f ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20 23:43:27 +00:00
Alec L Davis
a4f6d96b2e push 'outgoing' flag from sig_XXX up to chan_dahdi
'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag.
sig_pri and sig_ss7 were keeping 'outgoing' flag insync.

Now provides a callback for all the low level sig_XXX modules.

(issue ASTERISK-19316)

alecdavis (license 585)
Reported by: Jeremy Pepper
Tested by: alecdavis
 
Review: https://reviewboard.asterisk.org/r/1747/
........

Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 355851 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-18 08:02:08 +00:00
Terry Wilson
34c55e8e7c Opaquify char * and char[] in ast_channel
Review: https://reviewboard.asterisk.org/r/1733/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-13 17:27:06 +00:00
Richard Mudgett
23bc964e1c Constify some more channel driver technology callback parameters.
Review: https://reviewboard.asterisk.org/r/1707/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-01 19:53:38 +00:00
Terry Wilson
99cae5b750 Opaquify channel stringfields
Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.

Review: https://reviewboard.asterisk.org/r/1661/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24 20:12:09 +00:00
Terry Wilson
04da92c379 Replace direct access to channel name with accessor functions
There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.

This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.

The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.

The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).

Review: https://reviewboard.asterisk.org/r/1655/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
Richard Mudgett
938b642245 Restore call progress code for analog ports.
Extracting sig_analog from chan_dahdi lost call progress detection
functionality.

* Fix analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.

(closes issue ASTERISK-18841)
Reported by: Richard Miller
Patches:
      chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
      sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
      sig_analog.h.diff (license #5685) patch uploaded by Richard Miller
........

Merged revisions 347006 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 347007 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-05 17:44:15 +00:00
Richard Mudgett
10de040b6e More parking issues.
* Fix potential deadlocks in SIP and IAX blind transfer to parking.

* Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the
parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
parameter).  Created ast_park_call_exten() and ast_masq_park_call_exten()
to maintian API compatibility.

* Made masq_park_call() handle a failed ast_channel_masquerade() setup.

* Reduced excessive struct parkeduser.peername[] size.
........

Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-18 21:15:45 +00:00
Kinsey Moore
db3d113414 Merged revisions 330706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r330706 | kmoore | 2011-08-03 08:39:06 -0500 (Wed, 03 Aug 2011) | 17 lines
  
  Merged revisions 330705 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) | 10 lines
    
    Call pickup broken for DAHDI channels when beginning with #
    
    The call pickup feature did not work on DAHDI devices for anything other than
    feature codes beginning with * since all feature codes in chan_dahdi were
    originally hard-coded to begin with *.  This patch is also applied to
    chan_dahdi.c to fix this bug with radio modes.
    
    (closes issue AST-621)
    Review: https://reviewboard.asterisk.org/r/1336/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-03 13:40:22 +00:00
Richard Mudgett
810b9c8879 Merged revisions 316224 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r316224 | rmudgett | 2011-05-03 14:18:30 -0500 (Tue, 03 May 2011) | 16 lines
  
  The dahdi_hangup() call does not clean up the channel fully.
  
  After dahdi_hangup() has supposedly hungup an ISDN channel there is still
  traffic on the S0-bus because the channel was not cleaned up fully.
  
  Shuffled the hangup code to include some missing cleanup.  Also fixed some
  code formatting in the area.  I think the primary missing clean up code
  was the call to tone_zone_play_tone() to turn off any active tones on the
  channel.
  
  (closes issue #19188)
  Reported by: jg1234
  Patches:
        issue19188_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: jg1234
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-03 19:22:29 +00:00
Richard Mudgett
bc907695bf Merged revisions 313190 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r313190 | rmudgett | 2011-04-11 10:40:30 -0500 (Mon, 11 Apr 2011) | 39 lines
  
  Merged revisions 313189 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r313189 | rmudgett | 2011-04-11 10:32:53 -0500 (Mon, 11 Apr 2011) | 32 lines
    
    Merged revisions 313188 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) | 25 lines
      
      Stuck channel using FEATD_MF if caller hangs up at the right time.
      
      The cause was actually a caller hanging up just at the end of the Feature
      Group D DTMF tones that setup the call.  The reason for this is a "guard
      timer" that's implemented using ast_safe_sleep(100).  If the caller
      happens to hang up AFTER the final tone of the DTMF string but BEFORE the
      end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
      This causes the code to bounce to the end of ss_thread(), but it does NOT
      tear down the call properly.
      
      This should be a rare occurrence because the caller has to hang up at
      EXACTLY the right time.  Nonetheless, it was happening quite regularly on
      the reporter's system.  It's not easily reproducible, unless you purposely
      increase the guard-time to 2000 or more.  Once you do that, you can
      reproduce it every time by watching the DTMF debug and hanging up just as
      it ends.
      
      Simply add an ast_hangup() before goto quit.
      
      (closes issue #15671)
      Reported by: jcromes
      Patches:
            issue15671.patch uploaded by pabelanger (license 224)
      Tested by: jcromes
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-11 15:47:17 +00:00
Richard Mudgett
ce17f956dc Add private lock deadlock avoidance callback to PRI and SS7.
Factor out the equivalent function for analog.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-08 16:17:32 +00:00