asterisk/asterisk.git
11 years agoSeveral changes to codec_dahdi to play nice with G723.
Shaun Ruffell [Tue, 17 Feb 2009 22:28:41 +0000 (22:28 +0000)]
Several changes to codec_dahdi to play nice with G723.

This commit brings in the changes that were living out on the
svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch.  codec_dahdi.c now
always uses signed linear as the simple codec so that a soft g729 codec will
not end up being preferred to the hardware codec.  There are also changes to
allow codec_dahdi.c to feed packets to the hardware in the native sample size of
the codec.  This solves problems with choppy audio when using G723.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176760 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 176701 via svnmerge from
Jeff Peeler [Tue, 17 Feb 2009 22:08:00 +0000 (22:08 +0000)]
Merged revisions 176701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines

  Modify bridging to properly evaluate DTMF after first warning is played

  The main problem is currently if the Dial flag L is used with a warning sound,
  DTMF is not evaluated after the first warning sound. To fix this, a flag has
  been added in ast_generic_bridge for playing the warning which ensures that if
  a scheduled warning is missed, multiple warrnings are not played back (due to a
  feature evaluation or waiting for digits). ast_channel_bridge was modified to
  store the nexteventts in the ast_bridge_config structure as that information
  was lost every time ast_channel_bridge was reentered, causing a hangup due to
  incorrect time calculations.

  (closes issue #14315)
  Reported by: tim_ringenbach

  Reviewed on reviewboard:
  http://reviewboard.digium.com/r/163/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176708 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoUse constants from inttypes.h to clear up 32-bit compilation errors
Mark Michelson [Tue, 17 Feb 2009 22:02:42 +0000 (22:02 +0000)]
Use constants from inttypes.h to clear up 32-bit compilation errors

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176706 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agocreate a UDPTL structure in create_addr_from_peer() if it does not already exist...
Dwayne M. Hubbard [Tue, 17 Feb 2009 21:59:38 +0000 (21:59 +0000)]
create a UDPTL structure in create_addr_from_peer() if it does not already exist for T38

This is required to create a UDPTL structure in create_addr_from_peer() to handle the
scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but
is defined the peer's context.  I tested this patch by enabling t38pt_udptl in the
[general] section on one system and only enabling t38pt_udptl in a peer's context on
the system sending a fax.  Without the patch, the sending system will fail to initiate
T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure".
When this patch is applied the sending side will successfully initiate T38 negotiation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176705 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoClear up documentation of AST_FRIENDLY_OFFSET in frame.h
Mark Michelson [Tue, 17 Feb 2009 21:40:09 +0000 (21:40 +0000)]
Clear up documentation of AST_FRIENDLY_OFFSET in frame.h

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176697 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRecorded merge of revisions 176661 via svnmerge from
Tilghman Lesher [Tue, 17 Feb 2009 21:23:10 +0000 (21:23 +0000)]
Recorded merge of revisions 176661 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r176661 | tilghman | 2009-02-17 15:21:41 -0600 (Tue, 17 Feb 2009) | 9 lines

  Backport change to 1.4:
    Prior to masquerade, move the group definitions to the channel performing the
    masq, so that the group count lingers past the bridge.
    (closes issue #14275)
     Reported by: kowalma
     Patches:
           20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
     Tested by: kowalma
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176669 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoUpdate the timing API to have better support for multiple timing interfaces.
Russell Bryant [Tue, 17 Feb 2009 21:22:40 +0000 (21:22 +0000)]
Update the timing API to have better support for multiple timing interfaces.

1) Add module use count handling so that timing modules can be unloaded.

2) Implement unload_module() functions for the timing interface modules.

3) Allow multiple timing modules to be loaded, and use the one with the
   highest priority value.

4) Report which timing module is being use in the "timing test" CLI command.

(closes issue #14489)
Reported by: russell

Review: http://reviewboard.digium.com/r/162/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176666 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoPrior to masquerade, move the group definitions to the channel performing the
Tilghman Lesher [Tue, 17 Feb 2009 21:14:18 +0000 (21:14 +0000)]
Prior to masquerade, move the group definitions to the channel performing the
masq, so that the group count lingers past the bridge.
(closes issue #14275)
 Reported by: kowalma
 Patches:
       20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
 Tested by: kowalma

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176642 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoSignificantly improve scheduler performance under high load.
Russell Bryant [Tue, 17 Feb 2009 21:04:08 +0000 (21:04 +0000)]
Significantly improve scheduler performance under high load.

This patch changes the scheduler to use a max-heap to store pending scheduler
entries instead of a fully sorted doubly linked list.  When the number of
entries in the scheduler gets large, this will perform much better.  For much
more detailed information on this change, see the review request.

Review: http://reviewboard.digium.com/r/160/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176639 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd a test module for the heap implementation.
Russell Bryant [Tue, 17 Feb 2009 20:56:26 +0000 (20:56 +0000)]
Add a test module for the heap implementation.

Review: http://reviewboard.digium.com/r/160/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176635 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd an implementation of the heap data structure.
Russell Bryant [Tue, 17 Feb 2009 20:51:10 +0000 (20:51 +0000)]
Add an implementation of the heap data structure.

A heap is a convenient data structure for implementing a priority queue.

Code from svn/asterisk/team/russell/heap/.

Review: http://reviewboard.digium.com/r/160/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176632 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoTypo
Olle Johansson [Tue, 17 Feb 2009 20:50:03 +0000 (20:50 +0000)]
Typo

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176631 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerge a large set of updates to the Asterisk indications API.
Russell Bryant [Tue, 17 Feb 2009 20:41:24 +0000 (20:41 +0000)]
Merge a large set of updates to the Asterisk indications API.

This patch includes a number of changes to the indications API.  The primary
motivation for this work was to improve stability.  The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.

The changes included are:

1) Remove the module res_indications.  This included the critical functionality
   that actually loaded the indications configuration.  I have seen many people
   have Asterisk problems because they accidentally did not have an
   indications.conf present and loaded.  Now, this code is in the core,
   and Asterisk will fail to start without indications configuration.

   There was one part of res_indications, the dialplan applications, which did
   belong in a module, and have been moved to a new module, app_playtones.

2) Object management has been significantly changed.  Tone zones are now
   managed using astobj2, and it is no longer possible to crash Asterisk by
   issuing a reload that destroys tone zones while they are in use.

3) The API documentation has been filled out.

4) The API has been updated to follow our naming conventions.

5) Various bits of code throughout the tree have been updated to account
   for the API update.

6) Configuration parsing has been mostly re-written.

7) "Code cleanup"

The code is from svn/asterisk/team/russell/indications/.

Review: http://reviewboard.digium.com/r/149/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd assertions in the quest to track down a refcount leak.
Tilghman Lesher [Tue, 17 Feb 2009 18:49:20 +0000 (18:49 +0000)]
Add assertions in the quest to track down a refcount leak.
(closes issue #14485)
 Reported by: davevg

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176592 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix a race condition that caused device states to become incorrect for hints.
Russell Bryant [Tue, 17 Feb 2009 17:33:38 +0000 (17:33 +0000)]
Fix a race condition that caused device states to become incorrect for hints.

The problem here is that the hint processing code was subscribed to the wrong
event type.  So, it started processing state for a hint too soon, before the
device state cache had been updated.

Also, fix a similar bug in app_queue, as it was also subscribed to the wrong
event type.

(closes issue #14461)
Reported by: alecdavis

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176557 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoTypo
Olle Johansson [Tue, 17 Feb 2009 17:28:21 +0000 (17:28 +0000)]
Typo

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176556 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoIf there are no realtime engines, there's no reason to check for realtime families
Olle Johansson [Tue, 17 Feb 2009 15:18:55 +0000 (15:18 +0000)]
If there are no realtime engines, there's no reason to check for realtime families

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176513 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoIn this version, we can combine the queries, because we support dropping
Tilghman Lesher [Tue, 17 Feb 2009 14:39:36 +0000 (14:39 +0000)]
In this version, we can combine the queries, because we support dropping
nonexistent columns.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176501 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 176426 via svnmerge from
Tilghman Lesher [Tue, 17 Feb 2009 01:58:39 +0000 (01:58 +0000)]
Merged revisions 176426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines

  After a 'sip reload', qualifies for realtime peers weren't immediately
  restarted, instead waiting until the next registration.  We're now
  caching the qualify across a reload/restart and starting the qualify
  immediately upon loading the peer.
  (closes issue #14196)
   Reported by: pdf
   Patches:
         20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
   Tested by: pdf
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176459 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMight want to update the buffer pointer after a realloc (or we crash)
Tilghman Lesher [Mon, 16 Feb 2009 23:48:54 +0000 (23:48 +0000)]
Might want to update the buffer pointer after a realloc (or we crash)
(closes issue #14485)
 Reported by: davevg

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176360 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoadd support for Siren7 and Siren14 flavors of prompts and music on hold
Kevin P. Fleming [Mon, 16 Feb 2009 23:37:37 +0000 (23:37 +0000)]
add support for Siren7 and Siren14 flavors of prompts and music on hold

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176356 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 176354 via svnmerge from
David Vossel [Mon, 16 Feb 2009 23:33:55 +0000 (23:33 +0000)]
Merged revisions 176354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines

  Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging

  This should have been committed with rev176247, but I missed it.  srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either.  This fixs that.

  issue #13749
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176355 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoUse the correct list macros for deleting an item from the middle of a list.
Tilghman Lesher [Mon, 16 Feb 2009 23:14:08 +0000 (23:14 +0000)]
Use the correct list macros for deleting an item from the middle of a list.
(issue #13777)
 Reported by: pj
 Patches:
       20090203__bug13777.diff.txt uploaded by Corydon76 (license 14)
 Tested by: pj

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176320 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 176216 via svnmerge from
Kevin P. Fleming [Mon, 16 Feb 2009 21:45:54 +0000 (21:45 +0000)]
Merged revisions 176216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines

  fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak.
........
  r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines

  correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176255 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 176249,176252 via svnmerge from
Mark Michelson [Mon, 16 Feb 2009 21:40:40 +0000 (21:40 +0000)]
Merged revisions 176249,176252 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines

  Open the DAHDI pseudo device and set it to be nonblocking atomically

  Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately
  from opening the file was causing an "inappropriate ioctl for device" error.
  While I cannot fathom why this would be happening, I certainly am not opposed
  to making the code a bit more compact/efficient if it also fixes a bug.

  (closes issue #14482)
  Reported by: ys
  Patches:
        meetme.patch uploaded by ys (license 281)
  Tested by: ys
........
  r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines

  Remove unused variable and make dev-mode compilation happy
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176253 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175597 via svnmerge from
David Vossel [Mon, 16 Feb 2009 21:30:17 +0000 (21:30 +0000)]
Merged revisions 175597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines

  Fixed iax2 key rotation backwards compatibility

  Turns key rotation back on by default.  Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility.  This eliminates the need for the keyrotate option in iax.conf, so it has been removed.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176248 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAssist proper thread synchronization when stopping the logger thread.
Mark Michelson [Mon, 16 Feb 2009 18:25:57 +0000 (18:25 +0000)]
Assist proper thread synchronization when stopping the logger thread.

I was finding that on my dev box, occasionally attempting to "stop now" in
trunk would cause Asterisk to hang. I traced this to the fact that the logger
thread was waiting on a condition which had already been signalled. The logger
thread also need to be sure to check the value of the close_logger_thread variable.

The close_logger_thread variable is only checked when the list of logmessages is empty.
This allows for the logger thread to print and free any pending messages before exiting.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176174 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoCan't set debug level 2 (intense debugging) unless the syntax matches
Tilghman Lesher [Mon, 16 Feb 2009 17:44:51 +0000 (17:44 +0000)]
Can't set debug level 2 (intense debugging) unless the syntax matches

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176138 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRemove chan_features.
Russell Bryant [Mon, 16 Feb 2009 17:09:24 +0000 (17:09 +0000)]
Remove chan_features.

Review: http://reviewboard.digium.com/r/161/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176100 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 176029 via svnmerge from
Joshua Colp [Mon, 16 Feb 2009 15:36:19 +0000 (15:36 +0000)]
Merged revisions 176029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines

  Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog.

  This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value
  is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the
  pool was used for the value while the old was left untouched/unused. If the current pool was full a new
  pool was created. This would cause memory usage to increase steadily.

  (issue #AA50-2332)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMake the causes array static, and remove the type name as it is not needed.
Russell Bryant [Mon, 16 Feb 2009 02:54:42 +0000 (02:54 +0000)]
Make the causes array static, and remove the type name as it is not needed.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175983 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175921 via svnmerge from
Michiel van Baak [Mon, 16 Feb 2009 00:26:59 +0000 (00:26 +0000)]
Merged revisions 175921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines

  fix mis-spelling of the word registered.
  Reported by De_Mon on #asterisk-dev.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175952 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMake ast_sched_report() and ast_sched_dump() thread safe.
Russell Bryant [Sun, 15 Feb 2009 21:27:33 +0000 (21:27 +0000)]
Make ast_sched_report() and ast_sched_dump() thread safe.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175882 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix a number of problems with ast_sched_report().
Russell Bryant [Sun, 15 Feb 2009 20:56:27 +0000 (20:56 +0000)]
Fix a number of problems with ast_sched_report().

1) It had numerous coding guidelines violations with regards to formatting.

2) It allocated memory using ast_calloc() that was never freed.

3) It didn't check for failure from the allocation.

4) It used sprintf() and strcat() to build the result, doing zero checking to
   prevent writing past the end of the provided buffer.

The function also lacks API documentation, but that has not been addressed in
this commit.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175829 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175825 via svnmerge from
Olle Johansson [Sun, 15 Feb 2009 20:39:55 +0000 (20:39 +0000)]
Merged revisions 175825 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb 2009) | 2 lines

format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta!

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175827 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175792 via svnmerge from
Olle Johansson [Sun, 15 Feb 2009 20:22:12 +0000 (20:22 +0000)]
Merged revisions 175792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb 2009) | 2 lines

Disable format_ilbc.so by default, like codec_ilbc.so

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175801 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175777 via svnmerge from
Olle Johansson [Sun, 15 Feb 2009 20:18:27 +0000 (20:18 +0000)]
Merged revisions 175777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2 lines

Make sure that the debug line is not printed on debug level 0

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175783 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoBlocked revisions 175698 via svnmerge
Jason Parker [Fri, 13 Feb 2009 21:54:34 +0000 (21:54 +0000)]
Blocked revisions 175698 via svnmerge

........
  r175698 | qwell | 2009-02-13 15:53:16 -0600 (Fri, 13 Feb 2009) | 1 line

  Zaptel is not DAHDI.  Rather, Zaptel is actually Zaptel.  (in case you're confused, DAHDI is still DAHDI)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175699 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerge queue-reset branch to Asterisk
Mark Michelson [Fri, 13 Feb 2009 20:57:37 +0000 (20:57 +0000)]
Merge queue-reset branch to Asterisk

From a user point-of-view, this adds new CLI commands and Manager Actions to
better facilitate the reloading of queues and the resetting of their statistics.

The new CLI commands are the "queue reload" and "queue reset stats" commands.

The new manager actions are the QueueReload and QueueReset commands.

Review: http://reviewboard.digium.com/r/115

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175663 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd manager events for chanspy starting or stopping
Mark Michelson [Fri, 13 Feb 2009 20:35:26 +0000 (20:35 +0000)]
Add manager events for chanspy starting or stopping

(closes issue #14469)
Reported by: caio1982
Patches:
      chanspy_events2.diff uploaded by caio1982 (license 22)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175655 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agofix a few more XML documentation problems
Russell Bryant [Fri, 13 Feb 2009 20:26:49 +0000 (20:26 +0000)]
fix a few more XML documentation problems

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoadd missing </para>
Russell Bryant [Fri, 13 Feb 2009 20:23:39 +0000 (20:23 +0000)]
add missing </para>

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFixed iax2 key rotation backwards compatibility
David Vossel [Fri, 13 Feb 2009 20:11:55 +0000 (20:11 +0000)]
Fixed iax2 key rotation backwards compatibility

Turns key rotation back on by default.  Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility.  This eliminates the need for the keyrotate option in iax.conf, so it has been removed.

Review: http://reviewboard.digium.com/r/159/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175597 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175590 via svnmerge from
Mark Michelson [Fri, 13 Feb 2009 19:49:38 +0000 (19:49 +0000)]
Merged revisions 175590 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines

  Fix a potential crash situation when using IMAP voicemail

  If calling into VoiceMailMain when using IMAP storage, it was
  possible to crash Asterisk by hanging up the phone when prompted
  for a voicemail mailbox. This patch fixes the issue.

  While it may appear that this patch is superficial, it allows code
  execution to continue to the failure case just below the IMAP_STORAGE
  code block where this patch has been applied

  (closes issue #14473)
  Reported by: dwpaul
  Patches:
        voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175591 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd an option to keep the recorded file upon hangup.
Joshua Colp [Fri, 13 Feb 2009 16:41:15 +0000 (16:41 +0000)]
Add an option to keep the recorded file upon hangup.
(closes issue #14341)
Reported by: fnordian

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175549 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agodocument G.722.1/.1C support
Kevin P. Fleming [Fri, 13 Feb 2009 13:41:52 +0000 (13:41 +0000)]
document G.722.1/.1C support

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175512 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also...
Kevin P. Fleming [Fri, 13 Feb 2009 13:35:24 +0000 (13:35 +0000)]
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)

This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.

Along the way, some related work was done:

1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.

2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.

3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).

Review: http://reviewboard.digium.com/r/158/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoadd 'faxbuffers' configuration option information to CHANGES
Dwayne M. Hubbard [Fri, 13 Feb 2009 04:22:35 +0000 (04:22 +0000)]
add 'faxbuffers' configuration option information to CHANGES

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175475 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd dynamic fax buffer configuration option to chan_dahdi.conf
Dwayne M. Hubbard [Fri, 13 Feb 2009 00:13:38 +0000 (00:13 +0000)]
Add dynamic fax buffer configuration option to chan_dahdi.conf

When the 'faxdetect' configuration option is used, one may also want to use
the 'faxbuffers' configuration option in chan_dahdi.conf.  This option will
dynamically use the configured 'faxbuffers' buffer policy on a channel for
the life of the call following the detection of fax tones.  The faxbuffers
buffer policy will be reverted during call teardown.

An example use of 'faxbuffers' is below.  This example would switch to using
6 buffers with a full buffer policy.

faxbuffers=>6,full

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175411 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoBlocked revisions 175407 via svnmerge
Mark Michelson [Thu, 12 Feb 2009 23:23:47 +0000 (23:23 +0000)]
Blocked revisions 175407 via svnmerge

........
  r175407 | mmichelson | 2009-02-12 17:22:44 -0600 (Thu, 12 Feb 2009) | 12 lines

  Fix a place where filestreams were not refcounted properly

  This section was already present in trunk and other branches,
  but did not exist in 1.4.

  (closes issue #14395)
  Reported by: ZX81
  Patches:
        14395.patch uploaded by putnopvut (license 60)
  Tested by: ZX81
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175408 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRemove useless string copy, and make sscanf safe again
Russell Bryant [Thu, 12 Feb 2009 21:41:01 +0000 (21:41 +0000)]
Remove useless string copy, and make sscanf safe again

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175368 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdds force encryption option to iax.conf
David Vossel [Thu, 12 Feb 2009 21:27:11 +0000 (21:27 +0000)]
Adds force encryption option to iax.conf

This patch adds forceencryption=yes as an iax.conf option.  When force encryption is enabled, no unencrypted connections are allowed.  This insures all connections are encrypted.  This is a new feature, so CHANGES and iax.conf.sample are updated as well.

(closes issue #13285)
Reported by: sgofferj
Tested by: russell
Review: http://reviewboard.digium.com/r/150/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175311 via svnmerge from
Tilghman Lesher [Thu, 12 Feb 2009 21:25:14 +0000 (21:25 +0000)]
Merged revisions 175311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines

  Fix crashes when receiving certain T.38 packets.  Also, increase the maximum
  size of T.38 packets and warn users when they try to set the limits above those
  maximums.
  (closes issue #13050)
   Reported by: schern
   Patches:
         20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
   Tested by: schern
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175334 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175294 via svnmerge from
Jeff Peeler [Thu, 12 Feb 2009 20:48:56 +0000 (20:48 +0000)]
Merged revisions 175294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines

  Fix ParkedCall event information for From field in the case of a blind transfer

  If the parker information can not be obtained from the peer, try and see if
  the BLINDTRANSFER channel variable has been set. Previously, a blind transfer
  to the ParkAndAnnounce app would return nothing for the From.

  Closes AST-189
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175298 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAvoid using ast_strdupa() in a loop.
Russell Bryant [Thu, 12 Feb 2009 20:45:47 +0000 (20:45 +0000)]
Avoid using ast_strdupa() in a loop.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175295 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoDon't enable something by default that has a dependency on something _not_ enabled...
Russell Bryant [Thu, 12 Feb 2009 19:11:08 +0000 (19:11 +0000)]
Don't enable something by default that has a dependency on something _not_ enabled by default.

menuselect was not happy with this.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175255 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agocorrect warning message to not refer specifically to DAHDI
Kevin P. Fleming [Thu, 12 Feb 2009 18:48:52 +0000 (18:48 +0000)]
correct warning message to not refer specifically to DAHDI

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175250 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175187 via svnmerge from
Jeff Peeler [Thu, 12 Feb 2009 18:00:11 +0000 (18:00 +0000)]
Merged revisions 175187 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines

  Fix crash in event of failed attempt to transfer to parking

  The peer may not necessarily exist, such as in the case of a transfer to
  ParkAndAnnounce. In this case don't try to play a sound to it.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175188 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoSetting key rotation to be off by default
David Vossel [Thu, 12 Feb 2009 17:07:17 +0000 (17:07 +0000)]
Setting key rotation to be off by default

Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0).  As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175127 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175124 via svnmerge from
Russell Bryant [Thu, 12 Feb 2009 16:57:25 +0000 (16:57 +0000)]
Merged revisions 175124 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines

Don't send DTMF for infinite time if we do not receive an END event.

I thought that this was going to end up being a pretty gnarly fix, but it turns
out that there was actually already a configuration option in rtp.conf,
dtmftimeout, that was intended to handle this situation.  However, in between
Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost.
So, this commit brings it back to life.

The default timeout is 3 seconds.  However, it is worth noting that having
this be configurable at all is not really the recommended behavior in RFC 2833.
From Section 3.5 of RFC 2833:

      Limiting the time period of extending the tone is necessary
      to avoid that a tone "gets stuck". Regardless of the
      algorithm used, the tone SHOULD NOT be extended by more than
      three packet interarrival times. A slight extension of tone
      durations and shortening of pauses is generally harmless.

Three seconds will pretty much _always_ be far more than three packet
interarrival times.  However, that behavior is not required, so I'm going to
leave it with our legacy behavior for now.

Code from svn/asterisk/team/russell/issue_14460

(closes issue #14460)
Reported by: moliveras

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175125 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMake lock information for ao2_trylock be more useful and gnarly
Mark Michelson [Thu, 12 Feb 2009 16:28:06 +0000 (16:28 +0000)]
Make lock information for ao2_trylock be more useful and gnarly

Core show locks information involving an ao2_trylock did not
show the function that called ao2_trylock, but would instead
show ao2_trylock as the source of the lock. This is not useful
when trying to debug locking issues.

One bizarre note is that this logic is already in 1.4 but somehow
did not get merged to trunk or the 1.6.X branches.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175121 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoIssue a warning message if our candidate's IP is the loopback address.
Philippe Sultan [Thu, 12 Feb 2009 14:25:03 +0000 (14:25 +0000)]
Issue a warning message if our candidate's IP is the loopback address.

(closes issue #13985)
Reported by: jcovert
Tested by: phsultan

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 175029 via svnmerge from
Philippe Sultan [Thu, 12 Feb 2009 10:31:36 +0000 (10:31 +0000)]
Merged revisions 175029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines

Set the initiator attribute to lowercase in our replies when receiving calls.

This attribute contains a JID that identifies the initiator of the GoogleTalk
voice session. The GoogleTalk client discards Asterisk's replies if the
initiator attribute contains uppercase characters.

(closes issue #13984)
Reported by: jcovert
Patches:
      chan_gtalk.2.patch uploaded by jcovert (license 551)
Tested by: jcovert

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175058 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix a bit of odd logic for announcing position. Sync with 1.6.0's logic
Mark Michelson [Wed, 11 Feb 2009 23:12:57 +0000 (23:12 +0000)]
Fix a bit of odd logic for announcing position. Sync with 1.6.0's logic

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174951 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix odd "thank you" sound playing behavior in app_queue.c
Mark Michelson [Wed, 11 Feb 2009 23:03:08 +0000 (23:03 +0000)]
Fix odd "thank you" sound playing behavior in app_queue.c

If someone has configured the queue to play an position or holdtime
announcement, then it is odd and potentially unexpected to hear a
"Thank you for your patience" sound when no position or holdtime
was actually announced.

This fixes the announcement so that the "thanks" sound is only played
in the case that a position or holdtime was actually announced.

There is a way that the "thank you" sound can be played without a
position or holdtime, and that is to set announce-frequency to a value
but keep announce-position and announce-holdtime both turned off.

(closes issue #14227)
Reported by: caspy
Patches:
      14227_v3.patch uploaded by putnopvut (license 60)
Tested by: caspy

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174948 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix 'd' option for app_dial and add new option to Answer application
Mark Michelson [Wed, 11 Feb 2009 22:41:01 +0000 (22:41 +0000)]
Fix 'd' option for app_dial and add new option to Answer application

The 'd' option would not work for channel types which use RTP to transport
DTMF digits. The only way to allow for this to work was to answer the channel
if we saw that this option was enabled.

I realized that this may cause issues with CDRs, specifically with giving false
dispositions and answer times. I therefore modified ast_answer to take another
parameter which would tell if the CDR should be marked answered.

I also extended this to the Answer application so that the channel may be answered
but not CDRified if desired.

I also modified app_dictate and app_waitforsilence to only answer the channel if it
is not already up, to help not allow for faulty CDR answer times.

All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
the changes except for the change to the Answer application will go in since we do
not introduce new features into stable branches

(closes issue #14164)
Reported by: DennisD
Patches:
      14164.patch uploaded by putnopvut (license 60)
Tested by: putnopvut

Review: http://reviewboard.digium.com/r/145

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoBlocked revisions 174885 via svnmerge
Tilghman Lesher [Wed, 11 Feb 2009 20:55:46 +0000 (20:55 +0000)]
Blocked revisions 174885 via svnmerge

........
  r174885 | tilghman | 2009-02-11 14:54:18 -0600 (Wed, 11 Feb 2009) | 13 lines

  Restore a behavior that was recently changed, when we fixed issue #13962 and
  issue #13363 (related to issue #6176).  When a hangup occurs during a Macro
  execution in earlier 1.4, the h extension would execute within the Macro
  context, whereas it was always supposed to execute only within the main context
  (where Macro was called).  So this fix checks for an "h" extension in the
  deepest macro context where a hangup occurred; if it exists, that "h" extension
  executes, otherwise the main context "h" is executed.
  (closes issue #14122)
   Reported by: wetwired
   Patches:
         20090210__bug14122.diff.txt uploaded by Corydon76 (license 14)
   Tested by: andrew
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174886 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoTell the device state core a change happened when a channel is freed but not a specif...
Joshua Colp [Wed, 11 Feb 2009 14:44:47 +0000 (14:44 +0000)]
Tell the device state core a change happened when a channel is freed but not a specific state.
We need to do this because while we know that the freeing of the channel may cause something to become
not in use we do not know this for sure. There may be another channel that is still up which would cause
it to be in use.
(closes issue #13238)
Reported by: kowalma
Patches:
      20090121__bug13238.diff.txt uploaded by Corydon76 (license 14)
Tested by: alecdavis

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix potential for stack overflows in app_chanspy.c
Mark Michelson [Tue, 10 Feb 2009 23:17:03 +0000 (23:17 +0000)]
Fix potential for stack overflows in app_chanspy.c

When using the 'g' or 'e' options, the stack allocations that
were used could cause a stack overflow if a spyer stayed on the
line long enough without actually successfully spying on anyone.

The problem has been corrected by using static buffers and copying
the contents of the appropriate strings into them instead of using
functions like alloca or ast_strdupa

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174805 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix an fd leak that would occur in HTTP AMI sessions
Mark Michelson [Tue, 10 Feb 2009 21:45:14 +0000 (21:45 +0000)]
Fix an fd leak that would occur in HTTP AMI sessions

The explanation behind this fix is a bit complicated, and I've already
typed it up in the code as a huge comment inside of manager.c, so I'll
give the abridged version here.

We needed a way to separate action-specific data from session-specific data.
Unfortunately, the only way to maintain API compatibility and to not have to
change every single manager action was to rename the current mansession structure
and wrap it inside a new mansession structure which actually contains action-
specific data.

(closes issue #14364)
Reported by: awk
Patches:
      14364_better.patch uploaded by putnopvut (license 60)
Tested by: putnopvut

Review: http://reviewboard.digium.com/r/148/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoOnly decrease inringing count if above zero.
Joshua Colp [Tue, 10 Feb 2009 20:15:43 +0000 (20:15 +0000)]
Only decrease inringing count if above zero.
(issue #13238)
Reported by: kowalma

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174710 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoimprove slinfactory API to remove implicit sample rate and require explicit sample...
Kevin P. Fleming [Tue, 10 Feb 2009 19:38:26 +0000 (19:38 +0000)]
improve slinfactory API to remove implicit sample rate and require explicit sample rate selection by creator of the slinfactory

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174705 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoBlocked revisions 174644 via svnmerge
Joshua Colp [Tue, 10 Feb 2009 18:57:30 +0000 (18:57 +0000)]
Blocked revisions 174644 via svnmerge

........
  r174644 | file | 2009-02-10 14:50:50 -0400 (Tue, 10 Feb 2009) | 6 lines

  Go off hold when we get an empty reinvite telling us to.
  (closes issue #14448)
  Reported by: frawd
  Patches:
        hold_invite_nosdp.patch uploaded by frawd (license 610)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174645 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 174583 via svnmerge from
Matthew Nicholson [Tue, 10 Feb 2009 18:16:31 +0000 (18:16 +0000)]
Merged revisions 174583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines

  Improve behavior of jitterbuffer when maxjitterbuffer is set.

  This change improves the way the jitterbuffer handles maxjitterbuffer and
  dramatically reduces the number of frames dropped when maxjitterbuffer is
  exceeded.  In the previous jitterbuffer, when maxjitterbuffer was exceeded, all
  new frames were dropped until the jitterbuffer is empty.  This change modifies
  the code to only drop frames until maxjitterbuffer is no longer exceeded.

  Also, previously when maxjitterbuffer was exceeded, dropped frames were not
  tracked causing stats for dropped frames to be incorrect, this change also
  addresses that problem.

  (closes issue #14044)
  Patches:
        bug14044-1.diff uploaded by mnicholson (license 96)
  Tested by: mnicholson
  Review: http://reviewboard.digium.com/r/144/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174584 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoSet the type for the peer structure to be a peer as the default.
Joshua Colp [Tue, 10 Feb 2009 17:48:29 +0000 (17:48 +0000)]
Set the type for the peer structure to be a peer as the default.
(closes issue #14447)
Reported by: triccyx

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174580 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMake the logic for inuse and inringing manipluation match that of 1.4. The old broken...
Joshua Colp [Tue, 10 Feb 2009 15:37:07 +0000 (15:37 +0000)]
Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported.
(closes issue #14399)
Reported by: caspy
(issue #13238)
Reported by: kowalma

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix0ring build
Tilghman Lesher [Tue, 10 Feb 2009 07:06:29 +0000 (07:06 +0000)]
Fix0ring build

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174503 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRemove the usage of the KeepAlive app, as it no longer exists.
Tilghman Lesher [Tue, 10 Feb 2009 05:39:33 +0000 (05:39 +0000)]
Remove the usage of the KeepAlive app, as it no longer exists.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174470 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoThis patch removes the use of AST_PBX_KEEPALIVE
Steve Murphy [Tue, 10 Feb 2009 04:49:02 +0000 (04:49 +0000)]
This patch removes the use of AST_PBX_KEEPALIVE
from app_rpt.c.

(closes issue #14435)
Reported by: D_McNaul

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174435 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMore intptr_t work.
Steve Murphy [Tue, 10 Feb 2009 04:36:22 +0000 (04:36 +0000)]
More intptr_t work.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 174369 via svnmerge from
Steve Murphy [Tue, 10 Feb 2009 02:45:56 +0000 (02:45 +0000)]
Merged revisions 174369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines

  This patch solves some compiler complaints
  in both 32 and 64-bit environments.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174370 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix something I messed up in the merge I just did
Mark Michelson [Mon, 9 Feb 2009 17:27:32 +0000 (17:27 +0000)]
Fix something I messed up in the merge I just did

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174327 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFixes issue with hangups not being sent and external process never terminating.
David Vossel [Mon, 9 Feb 2009 17:26:02 +0000 (17:26 +0000)]
Fixes issue with hangups not being sent and external process never terminating.

The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued.  If the external script expects to be notified of a hangup and never receives one, it runs indefinitely.

(closes issue #14251)
Reported by: chris-mac
Tested by: dvossel

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174325 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 174282 via svnmerge from
Mark Michelson [Mon, 9 Feb 2009 17:20:55 +0000 (17:20 +0000)]
Merged revisions 174282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines

Don't do an SRV lookup if a port is specified

RFC 3263 says to do A record lookups on a hostname
if a port has been specified, so that's what we're
going to do. See section 4.2.

(closes issue #14419)
Reported by: klaus3000
Patches:
      patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65)

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174301 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 174218 via svnmerge from
Joshua Colp [Mon, 9 Feb 2009 14:49:24 +0000 (14:49 +0000)]
Merged revisions 174218 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines

  Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off.
  (closes issue #14407)
  Reported by: mostyn
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 174148 via svnmerge from
Russell Bryant [Sat, 7 Feb 2009 16:16:50 +0000 (16:16 +0000)]
Merged revisions 174148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines

Fix a race condition that could cause a crash.

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174149 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 174082 via svnmerge from
Dwayne M. Hubbard [Fri, 6 Feb 2009 23:51:56 +0000 (23:51 +0000)]
Merged revisions 174082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines

check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp()
and sip_uri_params_cmp()

The reporter didn't actually upload a properly-formed patch, instead a
modified chan_sip.c file was uploaded.  I created a patch to determine the
changes, then modified the suggested changes to create a proper fix.  The
summary above is a complete description of the changes.

(closes issue #13547)
Reported by: tecnoxarxa
Patches:
      chan_sip.c.gz uploaded by tecnoxarxa (license 258)
Tested by: tecnoxarxa

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174084 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdds immediate yes/no option to iax.conf
David Vossel [Fri, 6 Feb 2009 20:12:33 +0000 (20:12 +0000)]
Adds immediate yes/no option to iax.conf

This is very similar to the DAHDI immediate=yes option.  When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension.  Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled.  Examples explaining its use are added to iax2.conf.sample.  CHANGES has been updated as well.

(closes issue #14266)
Reported by: jcovert
Patches:
      chan_iax2.c.patch-trunk uploaded by jcovert (license 551)
      iax.conf.sample.patch uploaded by jcovert (license 551)
Tested by: jcovert, dvossel
Review: http://reviewboard.digium.com/r/143/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174046 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoDon't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue...
Joshua Colp [Fri, 6 Feb 2009 19:28:53 +0000 (19:28 +0000)]
Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription.
(closes issue #14322)
Reported by: amessina

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174041 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 173967-173968 via svnmerge from
Joshua Colp [Fri, 6 Feb 2009 17:18:35 +0000 (17:18 +0000)]
Merged revisions 173967-173968 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines

  Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string.
  (closes issue #14350)
  Reported by: fhackenberger
........
  r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines

  Remove a debug message I put in by accident.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 173917 via svnmerge from
Matthew Nicholson [Fri, 6 Feb 2009 16:28:19 +0000 (16:28 +0000)]
Merged revisions 173917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines

  Limit the addition of the Contact header in SIP responses according to various
  SIP RFCs.

  (closes issue #13602)
  Reported by: hjourdain
  Tested by: mnicholson
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173952 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAlways detach and destroy the whisper and barge audiohooks. Additionally also allow...
Joshua Colp [Fri, 6 Feb 2009 15:59:17 +0000 (15:59 +0000)]
Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached.
(closes issue #14414)
Reported by: bluecrow76

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173902 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoBlocked revisions 173900 via svnmerge
Tilghman Lesher [Fri, 6 Feb 2009 15:44:23 +0000 (15:44 +0000)]
Blocked revisions 173900 via svnmerge

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  r173900 | tilghman | 2009-02-06 09:43:32 -0600 (Fri, 06 Feb 2009) | 3 lines

  Backport OS X fix from trunk
  (AGAIN, closes issue #14360)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173901 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd a common implementation of a scheduler context with a dedicated thread.
Russell Bryant [Fri, 6 Feb 2009 10:55:35 +0000 (10:55 +0000)]
Add a common implementation of a scheduler context with a dedicated thread.

This commit expands the Asterisk scheduler API to include a common implementation
of a scheduler context being processed by a dedicated thread.  chan_iax2 has been
updated to use this new code.  Also, as a result, this resolves some race
conditions related to the previous chan_iax2 scheduler handling.

Related to rev 171452 which resolved the same issues in 1.4.

Code from team/russell/sched_thread2

Review: http://reviewboard.digium.com/r/129/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoResolve a memory leak that would occur on an invalid channel given to Action: Status
Russell Bryant [Fri, 6 Feb 2009 10:25:09 +0000 (10:25 +0000)]
Resolve a memory leak that would occur on an invalid channel given to Action: Status

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173848 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoUpdate extensions.conf.sample to be correct.
Mark Michelson [Thu, 5 Feb 2009 23:48:48 +0000 (23:48 +0000)]
Update extensions.conf.sample to be correct.

In trunk, the only necessary change pointed out was that the call
to ChanIsAvail uses an option that has been removed.

For the 1.6.1 branch, however, it appears that the sample file is
badly in need of updating since there are |'s used all over the place
there. My tentative plan is just to copy trunk's sample config file
to those branches since the info there is most up-to-date and should
be correct for use in 1.6.1

Thanks to macli in #asterisk-dev for bringing this up

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173776 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoProperly set "seen" and "unseen" flags when moving messages from the new to the old...
Mark Michelson [Thu, 5 Feb 2009 23:28:19 +0000 (23:28 +0000)]
Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage

(closes issue #13905)
Reported by: jaroth
Patches:
      foldermove_v2.patch uploaded by jaroth (license 50)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173773 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoBlocked revisions 173770 via svnmerge
Mark Michelson [Thu, 5 Feb 2009 23:19:51 +0000 (23:19 +0000)]
Blocked revisions 173770 via svnmerge

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r173770 | mmichelson | 2009-02-05 17:19:16 -0600 (Thu, 05 Feb 2009) | 20 lines

Fix logic regarding when to perform an SRV lookup for outgoing REGISTER requests

With this fix, we only will perform an SRV lookup at the following times:

* The first time we register with a remote registrar
* If we send a REGISTER but do not receive a response
* If the sendto() function returns an error

While I wrote the patch that fixes this issue, a huge amount of credit is due
to Brett Bryant, who wrote the initial patch on which I based this one.

(closes issue #12312)
Reported by: jrast
Patches:
      12312.patch uploaded by putnopvut (license 60)
Tested by: blitzrage

Review: http://reviewboard.digium.com/r/132/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173771 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 173696 via svnmerge from
Jeff Peeler [Thu, 5 Feb 2009 21:00:26 +0000 (21:00 +0000)]
Merged revisions 173696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines

  Add new configuration option to make shared IMAP mailboxes function as expected.

  The new option is "imapvmshareid" which is an ID to tag multiple mailboxes
  using the same IMAP storage location to function as one mailbox. This allows
  all messages to be retrieved for any user in the group. The patch alters the
  'X-Asterisk-VM-Extension' header that is responsible for matching voicemails
  for a given user.

  (closes issue #13673)
  Reported by: howardwilkinson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173697 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 173692 via svnmerge from
Mark Michelson [Thu, 5 Feb 2009 20:30:45 +0000 (20:30 +0000)]
Merged revisions 173692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines

Fix situations where queue members could be autopaused unexpectedly

Specifically, this patch prevents us from autopausing members when
we receive a busy or congestion frame from them.

(closes issue #14376)
Reported by: fiddur
Patches:
      14376.patch uploaded by putnopvut (license 60)
Tested by: fiddur

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173693 65c4cc65-6c06-0410-ace0-fbb531ad65f3