Commit graph

23134 commits

Author SHA1 Message Date
Matthew Jordan
7cf42e90b6 Fix uninitialized in this function error
With some versions of gcc, n_buckets will be flagged as being uninitialized
before use.  While its technically impossible (since the switch statement,
even without a default, accounts for all possibilities), we'll initialize the
variable to 0 anyway.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-19 02:14:54 +00:00
Matthew Jordan
530ce21b5d Reorder startup sequence to prevent lockups when process is sent to background
Although it is very rare and timing dependent, the potential exists for the
call to 'daemon' to cause what appears to be a deadlock in Asterisk during
startup.  This can occur when a recursive mutex is obtained prior to the
daemon call executing.  Since daemon uses fork to send the process into the
background, any threading primitives are unsafe to re-use after the call.
Implementations of pthread recursive mutexes are highly likely to store the
thread identifier of the thread that previously obtained the mutex.  If
the mutex was locked prior to the fork, a subsequent unlock operation will
potentially fail as the thread identifier is no longer valid.  Since the
mutex is still locked, all subsequent attempts to grab the mutex by other
threads will block.

This behavior exhibited itself most often when DEBUG_THREADS was enabled, as
this compile time option surrounds the mutexes in Asterisk with another
recursive mutex that protects the storage of thread related information.  This
made it much more likely that a recursive mutex would be obtained prior to
daemon and unlocked after the call.

This patch does the following:
a) It backports a patch from Asterisk 11 that prevents the spawning of the
   localtime monitoring thread.  This thread is now spawned after Asterisk has
   fully booted.
b) It re-orders the startup sequence to call daemon earlier during Asterisk
   startup.  This limits the potential of threading primitives being accessed
   by initialization calls before daemon is called.
c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called.
   Developers should send error messages directly to stderr prior to daemon,
   as calls to ast_log may access recursive mutexes that store thread related
   information.
d) It reorganizes when thread local storage is created for storing lock
   information during the creation of threads.  Prior to this patch, the
   read/write lock protecting the list of threads in ast_register_thread would
   utilize the lock in the thread local storage prior to it being initialized;
   this patch prevents that.

On a very related note, this patch will *greatly* improve the stability of the
Asterisk Test Suite.

Review: https://reviewboard.asterisk.org/r/2197

(closes issue ASTERISK-19463)
Reported by: mjordan
Tested by: mjordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
Matthew Jordan
99af7789dd Add a test event that reports changes in ConfBridge state
This patch adds a test event to ConfBridge that reports transitions between
states in ConfBridge.  This is used by tests in the Asterisk Test Suite
that verify state changes based on the entering/leaving of conference
participants.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 14:31:32 +00:00
David M. Lee
8e07045eb0 Fixed extconf.c breakage introduced in r376306.
To quote wdoekes:
> Note that I'm not confirming legitimacy of having that file in tree at
> all. Is anyone using aelparse/conf2ael?
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 00:15:30 +00:00
David M. Lee
00736a5e5d Somehow I put in svn-1.6 merge information. Oops.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 00:14:00 +00:00
David M. Lee
752ffc69d4 Migrate hashtest/hashtest2 to be unit tests.
Both hashtest and hashtest2 are manual testing apps that thrash hash
tables (hashtab and ao2 containers, respectively), by spinning up
several threads that randomly insert, delete, lookup and iterate over
the hash table. If the app doesn't crash, the hash table probably passes
the test. Those utils are not a part of the typical Asterisk build, so
they do not usually get compiled. This all makes them less that useful.

This patch removes those manual test programs and replaces them with
Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It also
attempts to make the tests more deterministic.

* Rather than spinning up some number of threads that operate on the
  hash table randomly, spin up four threads that concurrenly add,
  remove, lookup and iterate over the hash table.
* Each thread checks the state of the hash table both during and after
  execution, and indicates a test failure if things are not as expected.
* Each thread times out after 60 seconds to prevent deadlocking the unit
  test run.

(closes issue ASTERISK-20505)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2189/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 00:08:00 +00:00
Jonathan Rose
7ec134a8d7 app_meetme: Fix channels lingering when hung up under certain conditions
Channels would get stuck and MeetMe would repeatedly display an Unable
to write frame to channel error in the conf_run function if hung up
during certain sound prompts such as during user count announcements.
This patch fixes that by reintroducing a hangup check in the meetme's
main loop (also in conf_run).

(closes issue ASTERISK-20486)
Reported by: Michael Cargile
Review: https://reviewboard.asterisk.org/r/2187/
Patches:
    meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan Rose (license 6182)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-15 23:10:13 +00:00
Brent Eagles
6b07dd60cc Patch to prevent stopping the active generator when it is not the silence
generator.

This patch introduces an internal helper function to safely check whether the
current generator is the one that is expected before deactivating it. The
current externally accessible ast_channel_stop_generator() function has been
modified to be implemented in terms of the new function.

(closes issue ASTERISK-19918)
Reported by: Eduardo Abad
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-15 14:35:01 +00:00
Rusty Newton
21a0dadac0 Patch to play correct sound file when a voicemail's urgent status is removed
We were attempting to play "vm-urgent-removed", which didn't exist. Now we play "vm-marked-nonurgent" which exists      
and is the correct sound file. Previous behavior was silence and a warning on the CLI.

(issue ASTERISK-20280)
(closes issue ASTERISK-20280)
Reported by: Tomo Takebe
Tested by: Rusty Newton
Patches:
    asterisk20280.patch uploaded by Rusty Newton (license 5829)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-15 02:29:40 +00:00
Richard Mudgett
ee3927be17 Fix call files when astspooldir is relative.
Future dated call files are ignored when astspooldir is relative to the
current directory.  The queue_file() assumed that the qdir needed to be
prepended if the given filename did not start with a '/'.  If astspooldir
is relative it is not going to start from the root directory obviously so
it will not start with a '/'.  The filename used in queue_file()
ultimately results in qdir prepended multiple times.

* Made queue_file() not prepend qdir if the filename contains a '/'.

(closes issue ASTERISK-20593)
Reported by: James Le Cuirot
Patches:
      0004-Fix-future-call-files-from-relative-directories.patch (license #6439) patch uploaded by James Le Cuirot
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-14 19:55:39 +00:00
Jonathan Rose
e62bab8131 chan_sip: Add SubscribeContext field to SIPshowpeer AMI response
The new field is will show up within the response if the requested peer has a
subscribe context set.

(closes issue ASTERISK-20626)
Reported by: Jaco Kroon
Patches:
    asterisk-sip-ami-SubscrContext.patch uploaded by jkroon (license 5671)
        -with modifications by jrose to conform to style guidelines
Review: https://reviewboard.asterisk.org/r/2195/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-13 19:42:13 +00:00
Joshua Colp
c42d9d0d62 Properly check if the "Context" and "Extension" headers are empty in a ShowDialPlan action.
The code which handles the ShowDialPlan action wrongly assumed that a non-NULL return value
from the function which retrieves headers from an action indicates that the header has a
value. This is incorrect and the contents must be checked to see if they are blank.

(closes issue ASTERISK-20628)
Reported by: jkroon
Patches:
     asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-12 20:46:51 +00:00
Michael L. Young
d04bf30215 Fix Dynamic Hints Variable Substition - Underscore Problem
When adding a dynamic hint, if an extension contains an underscore no variable
subsitution is being performed.

This patch changes from checking if the extension contains an underscore to
checking if the extension begins with an underscore.

(closes issue ASTERISK-20639)
Reported by: Steven T. Wheeler
Tested by: Steven T. Wheeler, Michael L. Young
Patches:
  asterisk-20639-dynamic-hint-underscore.diff 
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2188/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-12 20:18:47 +00:00
Joshua Colp
866d968149 Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default.
With ICE support enabled in chan_sip and a large number of interfaces on the system it was
possible for the produced SDP to be truncated due to some fixed size buffers. These buffers
have now been changed so they will dynamically grow as needed.

ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience
for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is
no longer enabled by default there.

(closes issue ASTERISK-20643)
Reported by: coopvr
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376131 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-11 17:15:47 +00:00
Mark Michelson
e773bbdd10 Fix a "set but not used" warning on newer gccs.
Turns out the "helpful" setting of ms and res in this
macro is completely useless after the timeout antipattern
fix.

If you're a new guy looking to write code, don't write
a macro like this one.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-08 22:10:29 +00:00
Richard Mudgett
735f5c5059 chan_dahdi/SS7: Made reject incoming call for an in-alarm or blocked channel.
If a SS7 call comes in requesting a CIC that is in-alarm, the call is
accepted and connects if the extension exists in the dialplan.  The call
does not have any audio.

* Made release the call immediately with circuit congestion cause.

(closes issue ASTERISK-20204)
Reported by: Tuan Le
Patches:
      jira_asterisk_20204_v1.8.patch (license #5621) patch uploaded by rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-08 21:12:35 +00:00
Richard Mudgett
3a9640b605 Add MALLOC_DEBUG enhancements.
* Makes malloc() behave like calloc().  It will return a memory block
filled with 0x55.  A nonzero value.

* Makes free() fill the released memory block and boundary fence's with
0xdeaddead.  Any pointer use after free is going to have a pointer
pointing to 0xdeaddead.  The 0xdeaddead pointer is usually an invalid
memory address so a crash is expected.

* Puts the freed memory block into a circular array so it is not reused
immediately.

* When the circular array rotates out a memory block to the heap it checks
that the memory has not been altered from 0xdeaddead.

* Made the astmm_log message wording better.

* Made crash if the DO_CRASH menuselect option is enabled and something is
found.

* Fixed a potential alignment issue on 64 bit systems.
struct ast_region.data[] should now be aligned correctly for all
platforms.

* Extracted region_check_fences() from __ast_free_region() and
handle_memory_show().

* Updated handle_memory_show() CLI usage help.

Review: https://reviewboard.asterisk.org/r/2182/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-08 17:38:31 +00:00
Mark Michelson
f2bb9afe17 Multiple revisions 375993-375994
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  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/
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  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.
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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
6ad0126425 Fix stuck DTMF when bridge is broken.
When a bridge is broken by an AMI Redirect action or the ChannelRedirect
application, an in progress DTMF digit could be stuck sending forever.

* Made simulate a DTMF end event when a bridge is broken and a DTMF digit
was in progress.

(closes issue ASTERISK-20492)
Reported by: Jeremiah Gowdy
Patches:
      bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by Jeremiah Gowdy
      Modified to jira_asterisk_20492_v1.8.patch
      jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2169/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-06 19:05:11 +00:00
Joshua Colp
82dc21e0e1 Fix a bug where our Motif ICE candidates were not quite proper, and make us more forgiving.
An issue was reported on the mailing list where calling would result in an "Incomplete
ICE-UDP candidate received on session" error message. This is the result of the ICE-UDP
candidate code not placing a "network" attribute within the candidates. This is now done.
To increase compatibility though I have removed the requirement for the "network" attribute
to exist within ICE-UDP candidates that are received since we don't actually require the
value.

Reported on the mailing list by Jean-Denis Girard.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-06 12:15:31 +00:00
Matthew Jordan
a0c363e227 Refactor ast_timer_ack to return an error and handle the error in timer users
Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor.  This can lead to situations where errors stream to the
CLI/log file.  This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.

This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures.  It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.

Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.

Review: https://reviewboard.asterisk.org/r/2178/

(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
  jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-05 23:10:14 +00:00
Richard Mudgett
5336a97f25 Add safety NULL pointer check in module user references.
Made __ast_module_user_remove() check for NULL pointers.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-05 21:42:49 +00:00
Jonathan Rose
ff09fa5ac5 chan_sip: Document a change to user-field encoding introduced with r303509
The change in question was added to improve compliance with RFC3261, but at
the time of commit, it wasn't adequately documented in the UPGRADE notes.

(closes issue ASTERISK-20561)
Reported by: Deniz
Review: https://reviewboard.asterisk.org/r/2177/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-05 18:00:39 +00:00
Matthew Jordan
b6bac916f0 Don't attempt to purge sessions when no sessions exist
Manager's tcp/tls objects have a periodic function that purge old manager
sessions periodically.  During shutdown, the underlying container holding
those sessions can be disposed of and set to NULL before the tcp/tls periodic
function is stopped.  If the periodic function fires, it will attempt to
iterate over a NULL container.

This patch checks for whether or not the sessions container exists before
attempting to purge sessions out of it.  If the sessions container is NULL,
we simply return.

Note that this error was also caught by the Asterisk Test Suite.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-04 03:10:21 +00:00
Matthew Jordan
069f5f8b93 Only deref a reserved gateway session if we actually reserved one
Its perfectly acceptable to have a gateway session unreserved when we go to
first allocate one.  Unreffing the reserved gateway session - when its NULL -
will result in an assertion error.

This problem was caught by the Asterisk Test Suite (once we had enough of the
debugging flags enabled)
........

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2012-11-04 02:44:35 +00:00
Matthew Jordan
4bd66cb96b Properly clean up manager resources on exit
This patch does two things:
1) It properly unregisters the manager CLI commands
2) It cleans up AMI users on exit.  Prior to this patch, the AMI users
   were not being disposed of properly, resulting in a memory leak.

(closes issue ASTERISK-20646)
Reported by: Corey Farrell
patches:
  manager_shutdown.patch uploaded by Corey Farrell (license 5909)
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2012-11-04 02:38:19 +00:00
Matthew Jordan
a243f4f153 Properly finalize prepared SQLite3 statements to prevent memory leak
The AstDB uses prepared SQLite3 statements to retrieve data from the SQLite3
database.  These statements should be finalized during Asterisk shutdown so
that the SQLite3 database can be properly closed.  Failure to finalize the
statements results in a memory leak and a failure when closing the database.

This patch fixes those issues by ensuring that all prepared statements are
properly finalized at shutdown.

(closes issue ASTERISK-20647)
Reported by: Corey Farrell
patches:
  astdb-sqlite3_close.patch uploaded by Corey Farrell (license 5909)
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2012-11-04 01:19:43 +00:00
Matthew Jordan
ff469e9d5d Fix memory leaks in XML documentation
This patch fixes two memory leaks:
1) When building XML documentation items, the 'name' attribute was extracted
   from XML elements but not properly freed after being copied into the item
   being built.
2) When unloading XML documentation, the doctree container objects were not
   properly freed.

This patch corrects these memory leaks.  Note that this patch was modified
slightly for this commmit, as the case where the 'name' attribute doesn't
exist also wasn't handled in the item construction.  This patch also checks
for that attribute not existing.

(closes issue ASTERISK-20648)
Reported by: Corey Farrell
Tested by: mjordan
patches:
  xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909)
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2012-11-04 00:48:24 +00:00
Matthew Jordan
19282f682e Prevent multiple CDR batches from conflicting when scheduling the CDR write
The Asterisk Test Suite caught an error condition where a scheduled CDR batch
write can be deleted twice if two channels attempt to post their CDRs at the
same time.  The batch CDR mutex is locked while the CDRs are appended to the
current batch list; however, it is unlocked prior to actually scheduling the
CDR write.  As such, two threads can attempt to remove the currently scheduled
batch write at the same time, resulting in an assertion error.

This patch extends the time that the mutex is locked to encompass actually
scheduling the write.  This prevents two threads from unscheduling the
currently scheduled write at the same time.
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2012-11-04 00:02:06 +00:00
Andrew Latham
27a9bab706 Blocked revisions 375702
........
Doxygen Updates

Replace links to missing text files removed in the 1.6.x series with links to the wiki.  Doxygen can handle URLs fine, don't atempt to quote them.  Also update the wiki link in the Readme to get everyone on the same page.

(issue ASTERISK-20259)
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2012-11-03 03:21:44 +00:00
Damien Wedhorn
732767f230 Fix for chan_skinny leaving RTP ports open
Skinny wasn't closing RTP sockets. This patch includes ast_rtp_instance_stop before 
ast_rtp_instance_destroy which fixes the problem. Also add destroy for VRTP (which 
I believe is unused, but exists).

Review: https://reviewboard.asterisk.org/r/2176/
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2012-11-02 21:03:56 +00:00
Richard Mudgett
f85db0e34d Things don't need to be that const.
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2012-11-02 21:01:33 +00:00
Richard Mudgett
e950086daf Multiple revisions 375519-375524
........
  r375519 | rmudgett | 2012-10-30 16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines

  chan_misdn: Timer primitives must be handled first.

  The frm->addr is a different "address space" than the stack/instance
  address of other Lx primitives.  The test for B channel instance address
  could fail.

  Patches:
	patch01_timers.diff (license #6372) patch uploaded by Guenther Kelleter

  JIRA ABE-2888

  ........
  r375520 | rmudgett | 2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines

  chan_misdn: Free memory in error paths and other memory leaks.

  The one line commented with BUG is not easily fixable because there is no
  de-init function one can call.

  Patches:
	patch02_memory.diff (license #6372) patch uploaded by Guenther Kelleter

  JIRA ABE-2888

  ........
  r375521 | rmudgett | 2012-10-30 16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines

  chan_misdn: ISDN NT L2 de-establish/establish

  * An NT-PTMP cannot de/establish L2 since it doesn't know the TEIs.
  * On NT-PTP L2 is started when L1 is finally active in handle_l1.
  * L2 deactivation logging cleanup.
  * L2 aggregate link status is unknown for NT-PTMP, show as "UNKN".
  * Removed unused functions and code for L2 handling.

  Patches:
	patch03_L2estab.diff (license #6372) patch uploaded by Guenther Kelleter
	Modified

  JIRA ABE-2888

  ........
  r375522 | rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22 lines

  chan_misdn: Fix broken upper_id/lower_id usage.

  Sending PH prim via lower_id layer (3 or 1) simply does not work.  For TE
  (3) it returns an error (len=-6) which is not evaluated by handle_l1(), so
  the L1 layer status ends up wrong.  Instead PH must be sent via L4, only
  then does it reach L1 without an error message.

  And NT PH prims only reach L1 when they are sent to layer 2 id.
  --> use upper_id to send PH primitives.

  * Check for errors in PH_(DE)ACTIVATE | CONFIRM.
  * Debug messages are improved.

  * The lower_id is now not used for anything, except: Why is lower_id layer
  deleted when it wasn't created?  I removed this code since it looks very
  wrong.

  Patches:
	patch04_l1activation.diff (license #6372) patch uploaded by Guenther Kelleter

  JIRA ABE-2888

  ........
  r375523 | rmudgett | 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines

  chan_misdn: Fix loss of B channels if L1 is down.

  If you make 2 calls out an NT PTMP port which is not connected to any
  phone, the B channel associated with that call becomes unusable until
  Asterisk is restarted.

  The problem is the EVENT_SETUP is queued when L1 is not up in
  misdn_lib_send_event().  If L1 cannot be activated the event won't be
  dequeued.  It gets even worse when the call is hung up.  The queued
  EVENT_SETUP will be overwritten by an EVENT_DISCONNECT.  The reserved B
  channel then will never be freed.  If later someone connects a phone to
  the port, L1 will eventually activate and the queued EVENT_DISCONNECT is
  sent down the stack.  However, it is ignored because it is the wrong call
  state.

  The real fix would be that activation and queueing for a new SETUP is done
  by the NT stack.  But since it doesn't, the workaround must be removed
  because it doesn't always work.

  Fix: The event is no longer queued but immediately sent to the stack.  If
  L1 cannot be activated, the L3 state machine that was started by the
  EVENT_SETUP will do its work, i.e.  a timeout will release the B channel
  properly.  The SETUP possibly cannot be sent the first time but is resent
  by T303 in case L1 could be activated.

  Patches:
	patch05_bchan-loss.diff (license #6372) patch uploaded by Guenther Kelleter
	Modified

  JIRA ABE-2888

........
  r375524 | rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13 lines

  chan_misdn: Remove some calls to exit().

  Try proper cleanup when something goes wrong in misdn_lib_init().
  Especially do not call exit()!

  * Fix memory leak because stack_destroy() does not free the stack struct.

  Patches:
	patch06_cleanup-init.diff (license #6372) patch uploaded by Guenther Kelleter
	Modified

  JIRA ABE-2888
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2012-11-02 18:46:58 +00:00
Michael L. Young
01526b2c3c Fix Wrong Result In Debug Message For SDP Origin Processing
While looking at some debug logs, I noticed that it was being reported that the
SDP origin line was unsupported or failed.  Upon looking into this on my local
machine, I found that I too was getting this debug message yet everything seemed
to be getting processed properly.  What was discovered is, that, the variable to
determine what is displayed in the debug message for the SDP line that was
processed, was not being set for the origin line when the result was successful.

This patch fixes this and was tested on local machine.
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2012-11-02 17:27:24 +00:00
Jonathan Rose
d4a357b82f chan_sip: Fix a bug causing SIP reloads to remove all entries from the registry
A regression was introduced in chan_sip by changes to sip reload introduced by
r349097. That patch moved peer purging from the beginning of the reload to
after the general configuration was finished. This patch fixes that by undoing
the repositioning of the original peer purging code and using a similar
function after performing general configuration that purges only autocreated
peers that were created when persist mode isn't enabled.

(closes issue ASTERISK-20611)
Reported by: Alisher
Review: https://reviewboard.asterisk.org/r/2171/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-01 15:03:04 +00:00
Joshua Colp
6de0b18b3b Fix an issue with res_http_websocket where the chan_sip WebSocket handler could not be registered.
On some systems the optional API support uses the GCC compiler attribute "weakref" to provide its
functionality. This code changes the function names and prefixes "__" to the front. The
res_http_websocket exports file did not take this into account, thereby not allowing those functions
to be global and ultimately found.

(closes issue ASTERISK-20631)
Reported by: danjenkins
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2012-10-31 18:01:09 +00:00
Matthew Jordan
05cee7b717 Properly extract the Body information of an EWS calendar item
Unlike all other calendar modules, res_calendar_ews fails to extract the Body
information for a calendar item.  This is due, in part, to a quirk in the
schema in the XML - not only does a CalendarItem contain a Body element, but
the CalendarItem exists as a descendant of a different Body element.  The neon
parser was erroneously skipping all Body elements.

This patch fixes that by bypassing Body elements that are not a child of
CalendarItem, and parsing the Body element out if it is a child.

Note that the original patch by Terry Wilson only needed slight modifications
to make it properly pull the Body information out; as such, while I've linked
to the patch that I uploaded for Dmitry, I've attributed the patch to Terry.

(closes issue ASTERISK-19738)
Reported by: Dmitry Burilov
Tested by: Dmitry Burilov
patches:
  calendar_ews_body_2012_10_29.diff uploaded by Terry Wilson (license 6283)
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2012-10-31 14:58:44 +00:00
Richard Mudgett
9240971cd4 Fix ConfBridge crash if no timing module loaded.
(closes issue ASTERISK-19448)
Reported by: feyfre
Patches:
      smfix.patch (license #6099) patch uploaded by feyfre
      Modified for coding guidelines.
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2012-10-30 19:31:02 +00:00
Jonathan Rose
1e59b210af mixmonitor: Add a test event
This test event is being used to fix the  mixmonitor_audiohook_inherit
test.
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2012-10-30 19:20:33 +00:00
Jonathan Rose
9a80b5da22 confbridge: Fix a bug which made conferences not record with AMI/CLI commands
When confbridge was changed to handle conference status with a state machine in
r374658. The function responsible for starting recording for a conference was
refactored with the function actually responsible for launching the recording
thread being split into a function with another name. The old function name was
still used for manually started recordings through AMI or CLI. This patch fixes
that by switching which function is used to start recording the conference.

(closes issue ASTERISK-20601)
Reported by: Vilius
Patches:
    confbridge_mixmonitor.diff uploaded by Jonathan Rose (license 6182)
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2012-10-30 15:10:38 +00:00
Mark Michelson
5f3f32c494 Prevent resetting of NATted realtime peer address on reload.
If a "sip reload" is issued for a SIP peer, then his
IP address will be cleared, thus resulting in forgetting the
public IP address. Asterisk will then attempt to route SIP
traffic to the private IP address.

The fix here is to make "sip reload" ignore realtime peers
when "host = dynamic" is spotted. Realtime peers can now only
have their IP address reset if they have gone from being not
dynamic to being dynamic.

(closes issue ASTERISK-18203)
reported by daren ferreira

(closes issue ASTERISK-20572)
reported by JoshE
Patches:
	fix_nat_realtime.diff uploaded by JoshE (license #6075)
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2012-10-29 21:38:40 +00:00
Mark Michelson
da85f8489f Make evaluation of channel variables consistently case-sensitive.
Due to inconsistencies in how variable names were evaluated, the
decision was made to make all evaluations case-sensitive. See the
UPGRADE.txt file or https://wiki.asterisk.org/wiki/display/AST/Case+Sensitivity
for more details.

(closes issue ASTERISK-20163)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2160


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-29 21:27:09 +00:00
Matthew Jordan
1eb14dbff8 Ensure that CDRs for a caller in a Queue that is not answered is NO ANSWER.
When a caller enters a queue and no queue member answers the call, the current
behaviour can be a little odd depending on the paused status of the queue
members.  If any queue member is paused, but not all, the CDR disposition
will be BUSY.  If all queue members are paused, then the CDR disposition is
based instead on the disposition of the call prior to entering the Queue.

This patch modifies the behaviour in the following ways:
* If no queue members are paused, the CDR disposition is whatever the
  disposition was prior to going into Queue.  If the call was answered this
  will be ANSWERED; otherwise, it is NO ANSWER.
* If some queue members are pused, the CDR result is NO ANSWER. (This is a
  change in behaviour, as the result would previously have been BUSY)
* If all queue members are paused, the CDR result is whatever the result was
  prior to going into Queue.  This is the same as the behaviour prior to this
  patch.
* If the caller hangs up, times out, or presses '*' with the 'h' option, the
  CDR disposition is again not set and is dependent on whether or not the
  caller was Answered prior to entering Queue.

This patch was based on one provided by Thomas Arimont, but has been modified
to accomodate findings by the reviewers.

Review: https://reviewboard.asterisk.org/r/2064/

(closes issue AST-906)
Reported by: Thomas Arimont

(closes issue ASTERISK-17776)
Reported by: Attila Megyeri



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-29 21:02:20 +00:00
Richard Mudgett
8d65c777c8 Fix the Park 'r' option when a channel parks itself.
When a channel uses the Park appliation to park itself with the 'r'
option, the channel hears music-on-hold instead of the requested ringing.

* Added a missing check for the 'r' option when a channel parks itself.

(closes issue ASTERISK-19382)
Reported by: James Stocks
Patches by: dsessions

Review: https://reviewboard.asterisk.org/r/2148/
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2012-10-29 19:31:36 +00:00
Richard Mudgett
e2702177a4 chan_dahdi: Fix segfault dereferencing a NULL tech_pvt.
The tech support customer was using the AMI Redirect action shortly after
a call was placed.  While the channel tried to do an ast_read(), the
masquerade resulting from the channel redirect took place.  The masquerade
in the middle of the ast_read() resulted in the segfault.

(closes issue AST-1025)
Reported by: Trey Blancher
Patches:
      jira_ast_1025_v1.8_v2.patch (license #5621) patch uploaded by rmudgett
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2012-10-29 15:56:13 +00:00
Jonathan Rose
2c3638df98 ast_tls_cert script: Better response for various exit conditions to openssl
(closes issue ASTERISK-20260)
Reported by: Daniel O'Connor
Patches:
	ast_tls_cert-update.diff uploaded by Daniel O'Connor (license 6419)
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2012-10-23 16:22:44 +00:00
Jonathan Rose
3d540ef218 core: Fix a memory leak in app.c from an early return
ast_app_group_match_get_count allocates memory with the regcomp
function and we previously forgot to free it when bailing out
due to a regex compilation failure against category.

(closes issue AST-1018)
Reported by: Guenther Kelleter
Patches:
	regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372)
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2012-10-22 20:19:51 +00:00
Jonathan Rose
31f1881ceb GSM: Fix encoding problems with GSM
(closes issue ASTERISK-20457)
Reported by: Richard Miller
Patches:
	code.patch uploaded by Richard Miller (license 5685)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-22 17:31:20 +00:00
Jonathan Rose
987a5d58c3 app_queue: add upgrade notes for 375216
Adds UPGRADE notes describing behavioral changes to rrmemory strategy caused by
375216

(issue AST-989)
Reported by: Thomas Arimont


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-18 21:49:24 +00:00
Jonathan Rose
5873e06626 Blocked revisions 375247
........
pp_queue: add upgrade notes for 375216

Adds UPGRADE notes describing behavioral changes to rrmemory strategy caused by
375216

(issue AST-989)
Reported by: Thomas Arimont


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-18 21:47:46 +00:00