(closes issue #15269)
Reported by: contactmayankjain
Patches:
      patch.txt uploaded by contactmayankjain (license 740)
      memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
Tested by: contactmayankjain, dvossel




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201678 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines
  
  Fix memory corruption and leakage related reloads of non files mode MoH classes.
  
  For Music on Hold classes that are not files mode, meaning that we are executing
  an application that will feed us audio data, we use a thread to monitor the
  external application and read audio from it.  This thread also makes use of the
  MoH class object.  In the MoH class destructor, we used pthread_cancel() to ask
  the thread to exit.  Unfortunately, the code did not wait to ensure that the
  thread actually went away.  What needed to be done is a pthread_join() to ensure
  that the thread fully cleans up before we proceed.  By adding this one line, we
  resolve two significant problems:
  
    1) Since the thread was never joined, it never fully goes away.  So, on every
       reload of non-files mode MoH, an unused thread was sticking around.
  
    2) There was a race condition here where the application monitoring thread
       could still try to access the MoH class, even though the thread executing
       the MoH reload has already destroyed it.
  
  (issue #15109)
  Reported by: jvandal
  
  (issue #15123)
  Reported by: axisinternet
  
  (issue #15195)
  Reported by: amorsen
  
  (issue AST-208)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201610 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This contains the interface by which we can let an rtp instance know
that it might start receiving audio from a new source. This is similar
in nature to revision 197588 of Asterisk 1.4.

Review: https://reviewboard.asterisk.org/r/276



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'.

(closes issue #15111)
Reported by: ffs
Patches:
      chan_sip.c_register-parser.patch uploaded by ffs (license 730)
Tested by: ffs, dvossel



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201570 65c4cc65-6c06-0410-ace0-fbb531ad65f3

chan_sip has an option to save the sysname on rtupdate.  This patch copies that same logic to chan_iax.

(closes issue #14837)
Reported by: barthpbx
Patches:
      iax2-rtsavesysname.patch uploaded by barthpbx (license 744)
      rt_iax.diff uploaded by dvossel (license 671)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201534 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15186)
 Reported by: ajohnson
 Patches: 
       20090528__issue15186.diff.txt uploaded by tilghman (license 14)
 Tested by: ajohnson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201531 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A recent change to our SDP version comparison made audio not function
on some calls. This was because of a test wherein we were trying to
see if an unsigned value was less than 0. This is a dumb comparison
and arguably the compiler should have warned about it. Alas, though,
it slipped past. Now it's fixed by changing the variable to be a
signed type.

Found by several developers. Tested by mnicholson and dbrooks.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines
  
  Change the datastore traversal in ast_do_masquerade to use a safe list traversal.
  
  It is possible for datastore fixup functions to remove the datastore from the list
  and free it. In particular, the queue_transfer_fixup in app_queue does this. While
  I don't yet know of this causing any crashes, it certainly could.
  
  Found while discussing a separate issue with Brian Degenhardt.
........


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201453 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines
  
  StopMixMonitor race condition (not giving up file immediately)
  
  StopMixMonitor only indicates to the MixMonitor thread to stop
  writing to the file.  It does not guarantee that the recording's
  file handle is available to the dialplan immediately after execution.
  This results in a race condition.  To resolve this, the filestream
  pointer is placed in a datastore on the channel. When StopMixMonitor
  is called, the datastore is retrieved from the channel and the
  filestream is closed immediately before returning to the dialplan.
  Documentation indicating the use of StopMixMonitor to free files
  has been updated as well.
  
  (closes issue #15259)
  Reported by: travisghansen
  Tested by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/283/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201445 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines
  
  Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read()
  
  Zombie channels could be passed, and chan_sip.c wasn't checking for it.
  Could crash Asterisk. Now checking for NULL pointer.
  
  (closes issue #15330)
  Reported by: okrief
  Tested by: dbrooks
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201381 65c4cc65-6c06-0410-ace0-fbb531ad65f3

During a sip reload, the list of sip_registry objects are
supposed to be traversed, unlinked, and destroyed, but
destruction never takes place due to a ref counting error.
This causes a memory leak when registry items are removed
from sip.conf and reloaded.  While the registries are removed
from the global list, they are not removed from the scheduler.
Because of this, SIP register attempts continue to be sent
out for the item even though it may no longer be in the .conf.

(closes issue #15295)
Reported by: amorsen

Review: https://reviewboard.asterisk.org/r/282/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15335)
Reported by: lmadsen

Review: https://reviewboard.asterisk.org/r/284/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines
  
  Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty.
  
  When the list to be appended is empty, and the list to be appended to is *not*,
  AST_LIST_APPEND_LIST would actually cause the target list to become broken,
  and no longer have a pointer to its last entry. This patch fixes the problem.
  
  (reported by Stanislaw Pitucha on the asterisk-dev mailing list)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201262 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201223 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201190 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

Since we use 'static' weakref symbols, and not all GCC versions support them,
test for that combination explicitly.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201137 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change ensures that Emacs can find the proper source files when parsing
compiler error messages, since it uses the 'make' output including directory
names to do it.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201135 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Defaulting to 'static' for the function scope was bad... so remove it.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
  
  Improve support for media paths that can generate multiple frames at once.
  
  There are various media paths in Asterisk (codec translators and UDPTL, primarily)
  that can generate more than one frame to be generated when the application calling
  them expects only a single frame. This patch addresses a number of those cases,
  at least the primary ones to solve the known problems. In addition it removes the
  broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
  functions, and cleans up various code paths affected by these changes.
  
  https://reviewboard.asterisk.org/r/175/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201056 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The last changes to ast_gcc_attribute.m4 caused some problems checking for
various attributes, because the scope of the symbol the attribute is applied
to can be important; this patch allows the scope to be specified for the check.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200985 65c4cc65-6c06-0410-ace0-fbb531ad65f3

What this patch addresses:
1. ast_sip_ouraddrfor() by default binds to the UDP address/port
reguardless if the sip->pvt is of type UDP or not.  Now when no
remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
transport type, attempting to set the address and port to the
correct TCP/TLS bindings if necessary.
2.  It is not necessary to send the port number in the Contact
header unless the port is non-standard for the transport type.
This patch fixes this and removes the todo note.
3.  In sip_alloc(), the default dialog built always uses transport
type UDP.  Now sip_alloc() looks at the sip_request (if present)
and determines what transport type to use by default.
4.  When changing the transport type of a sip_socket, the file
descriptor must be set to -1 and in some cases the tcptls_session's
ref count must be decremented and set to NULL.  I've encountered
several issues associated with this process and have created a function,
set_socket_transport(), to handle the setting of the socket type.


(closes issue #13865)
Reported by: st
Patches:
      dont_add_port_if_tls.patch uploaded by Kristijan (license 753)
      13865.patch uploaded by mmichelson (license 60)
      tls_port_v5.patch uploaded by vrban (license 756)
      transport_issues.diff uploaded by dvossel (license 671)
Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel

Review: https://reviewboard.asterisk.org/r/278/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200946 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Voicemail can only use one storage module at the moment.
Because it's unclear that selecting one of the storage modules
in menuselect will disable filesystem storage we now have
a FILE_STORAGE option that conflicts with the other modules.

(closes issue #15333)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200943 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200942 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r200875 | eliel | 2009-06-16 09:25:51 -0400 (Tue, 16 Jun 2009) | 5 lines
  
  Show the interface name on error, if it is not found.
  
  If the smdiport specified is not found, show the interface name
  instead of '(null)'.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200878 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the smdiport specified is not found, show the interface name
instead of '(null)'.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200841 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since there was only 1 bucket, and no hash function was specified, the code
actually worked perfectly fine.  However, in theory, this was invalid use of
the OBJ_POINTER flag, so remove it so the code provides a better usage example.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200805 65c4cc65-6c06-0410-ace0-fbb531ad65f3

keep backwards compatible chan_dahdi with older openr2 versions by not using the new skip category feature unless supported


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200799 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The configure script tests for compiler attributes didn't actually enable
enough warnings or provide a proper test harness to determine whether the 
compiler supports the attribute in question or not; this caused gcc 4.1 to
report that it supports 'weakref', but it doesn't actually support it in the
way that is needed for our optional API mechanism. The new configure script
test will properly distinguish between full support and partial support
for this attribute, among others.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200762 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk will now automatically ignore incorrect incoming SDP version numbers
when necessary to complete a T.38 re-INVITE operation.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This commit changes the 'incoming SDP version' check logic a bit more; when
'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to
switch to T.38, we'll always accept the peer's SDP response, even if they
don't properly increment the SDP version number as they should. If this situation
occurs, a warning message will be generated suggesting that the peer's
configuration be changed to include the 'ignoresdpversion' configuration option
(although ideally they'd fix their SIP implementation to be RFC compliant).

AST-221


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix up modules in the 'apps' directory, and also correct the bad example of
enum definitions in include/asterisk/app.h, which many developers followed
(thanks for reading the documentation!). In addition, add some basic usage
examples of the 'pahole' and 'pglobal' tools to the coding guidelines.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The 'pglobal' tool is quite handy indeed :-)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 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

Using the 'pahole' tool, it is now quite easy to see where structure fields
could be organized differently to keep the compiler from having to add
padding to satisfy alignment requirements. These changes reduced the sizes of
sip_pvt and sip_peer by a few bytes each (on 64-bit platforms), and also fixed
a spelling error in a field name.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200584 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch provides a new implementation of the optional API support defined
in asterisk/optional_api.h; this new version provides solves compatibility
issues with the use of linker version scripts for suppressing global symbols.
In addition, there is now a functional (and tested!) implementation for Mac OS/X,
so module writers no longer need to use special tests before calling optional
API functions. All future implementations must provide these same semantics,
so that module writers can rely on them.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200519 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines
  
  Add INFO to our allowed methods so that endpoints know they may send it to us.
  
  AST-223
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200514 65c4cc65-6c06-0410-ace0-fbb531ad65f3

added openr2 to menuselect-deps.in, recent commit in menuselect made me realize this was never done but was working anyways
also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200477 65c4cc65-6c06-0410-ace0-fbb531ad65f3