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20 authors
  1. Jul 17, 2014
  3. Jul 04, 2014
  5. Jul 03, 2014
  7. Jun 16, 2014
    • Richard Mudgett's avatar
      chan_dahdi: Adds support for major update to libss7. · 0c896d8b
      Richard Mudgett authored 
      * SS7 support now requires libss7 v2.0 or later.  The new libss7 is not
backwards compatible.

* Added SS7 support for connected line and redirecting.

* Most SS7 CLI commands are reworked as well as new SS7 commands added.
See online CLI help.

* Added several SS7 config option parameters described in
chan_dahdi.conf.sample.

* ISUP timer support reworked and now requires explicit configuration.
See ss7.timers.sample.

Special thanks to Kaloyan Kovachev for his support and persistence in
getting the original patch by adomjan updated and ready for release.

SS7-27 #close
Reported by: adomjan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      0c896d8b
  9. May 13, 2014
    • Richard Mudgett's avatar
      chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing is enabled. · 8b6ab478
      Richard Mudgett authored 
      When overlap dialing is enabled, the lack of inband audio available
information in the SETUP_ACKNOWLEDGE events causes an interoperability
problem with SIP.  sig_pri doesn't know if there is dialtone present when
a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an
AST_CONTROL_PROGRESS frame.  The SIP channel driver then sends out a 183
Session Progress and blocks the desired 180 Ringing message when the
ALERTING message comes in.

* Made the configure script detect if the installed version of libpri
supports the SETUP_ACKNOWLEDGE enhancements.

* Using the new API, made generate an AST_CONTROL_PROGRESS frame on an
incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio
is present instead of assuming that dialtone is present.

* Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio
available indication only if dialtone is expected.  The change also makes
the fallback behaviour of sending the PROGRESS message better by sending
it only if dialtone is expected.

* Changed receiving a PROCEEDING message to not generate an
AST_CONTROL_PROGRESS frame if the progress indication ie indicates
non-end-to-end-ISDN.  This helps interoperability with SIP.

* Changed sending a PROCEEDING message in response to an
AST_CONTROL_PROCEEDING frame to not indicate inband audio available.  It
was silly to do so anyway because the channel driver doesn't know if
inband audio is even available.  This helps interoperability with SIP.

This patch and a corresponding change in libpri work together to allow
Asterisk to control the inband audio available progress indication ie on
the SETUP_ACKNOWLEDGE message when dialtone is present.

AST-1338 #close
Reported by: Tyler Stewart

Review: https://reviewboard.asterisk.org/r/3521/
........

Merged revisions 413714 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 413765 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 413771 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8b6ab478
  11. Apr 23, 2014
    • George Joseph's avatar
      This patch adds support for spinlocks in Asterisk. · 64045f0b
      George Joseph authored 
      There are cases in Asterisk where it might be desirable to lock
a short critical code section but not incur the context switch
and yield penalty of a mutex or rwlock.  The primary spinlock
implementations execute exclusively in userspace and therefore
don't incur those penalties.  Spinlocks are NOT meant to be a
general replacement for mutexes.  They should be used only for
protecting short blocks of critical code such as simple compares
and assignments.  Operations that may block, hold a lock, or
cause the thread to give up it's timeslice should NEVER be
attempted in a spinlock.

The first use case for spinlocks is in astobj2 - internal_ao2_ref.
Currently the manipulation of the reference counter is done with
an ast_atomic_fetchadd_int which works fine.  When weak reference
containers are introduced however, there's an additional comparison
and assignment that'll need to be done while the lock is held.
A mutex would be way too expensive here, hence the spinlock.
Given that lock contention in this situation would be infrequent,
the overhead of the spinlock is only a few more machine instructions
than the current ast_atomic_fetchadd_int call.

ASTERISK-23553 #close
Review: https://reviewboard.asterisk.org/r/3405/
........

Merged revisions 412976 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      64045f0b
  13. Mar 05, 2014
  15. Jan 28, 2014
  17. Jan 08, 2014
  19. Dec 03, 2013
  21. Nov 26, 2013
  23. Nov 16, 2013
  25. Oct 08, 2013
  27. Oct 03, 2013
  29. Sep 30, 2013
    • David M. Lee's avatar
      Multiple revisions 399887,400138,400178,400180-400181 · 2de42c2a
      David M. Lee authored 
      ........
  r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
  
  Minor performance bump by not allocate manager variable struct if we don't need it
........
  r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
  
  Stasis performance improvements
  
  This patch addresses several performance problems that were found in
  the initial performance testing of Asterisk 12.
  
  The Stasis dispatch object was allocated as an AO2 object, even though
  it has a very confined lifecycle. This was replaced with a straight
  ast_malloc().
  
  The Stasis message router was spending an inordinate amount of time
  searching hash tables. In this case, most of our routers had 6 or
  fewer routes in them to begin with. This was replaced with an array
  that's searched linearly for the route.
  
  We more heavily rely on AO2 objects in Asterisk 12, and the memset()
  in ao2_ref() actually became noticeable on the profile. This was
  #ifdef'ed to only run when AO2_DEBUG was enabled.
  
  After being misled by an erroneous comment in taskprocessor.c during
  profiling, the wrong comment was removed.
  
  Review: https://reviewboard.asterisk.org/r/2873/
........
  r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
  
  Taskprocessor optimization; switch Stasis to use taskprocessors
  
  This patch optimizes taskprocessor to use a semaphore for signaling,
  which the OS can do a better job at managing contention and waiting
  that we can with a mutex and condition.
  
  The taskprocessor execution was also slightly optimized to reduce the
  number of locks taken.
  
  The only observable difference in the taskprocessor implementation is
  that when the final reference to the taskprocessor goes away, it will
  execute all tasks to completion instead of discarding the unexecuted
  tasks.
  
  For systems where unnamed semaphores are not supported, a really
  simple semaphore implementation is provided. (Which gives identical
  performance as the original taskprocessor implementation).
  
  The way we ended up implementing Stasis caused the threadpool to be a
  burden instead of a boost to performance. This was switched to just
  use taskprocessors directly for subscriptions.
  
  Review: https://reviewboard.asterisk.org/r/2881/
........
  r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
  
  Optimize how Stasis forwards are dispatched
  
  This patch optimizes how forwards are dispatched in Stasis.
  
  Originally, forwards were dispatched as subscriptions that are invoked
  on the publishing thread. This did not account for the vast number of
  forwards we would end up having in the system, and the amount of work it
  would take to walk though the forward subscriptions.
  
  This patch modifies Stasis so that rather than walking the tree of
  forwards on every dispatch, when forwards and subscriptions are changed,
  the subscriber list for every topic in the tree is changed.
  
  This has a couple of benefits. First, this reduces the workload of
  dispatching messages. It also reduces contention when dispatching to
  different topics that happen to forward to the same aggregation topic
  (as happens with all of the channel, bridge and endpoint topics).
  
  Since forwards are no longer subscriptions, the bulk of this patch is
  simply changing stasis_subscription objects to stasis_forward objects
  (which, admittedly, I should have done in the first place.)
  
  Since this required me to yet again put in a growing array, I finally
  abstracted that out into a set of ast_vector macros in
  asterisk/vector.h.
  
  Review: https://reviewboard.asterisk.org/r/2883/
........
  r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
  
  Remove dispatch object allocation from Stasis publishing
  
  While looking for areas for performance improvement, I realized that an
  unused feature in Stasis was negatively impacting performance.
  
  When a message is sent to a subscriber, a dispatch object is allocated
  for the dispatch, containing the topic the message was published to, the
  subscriber the message is being sent to, and the message itself.
  
  The topic is actually unused by any subscriber in Asterisk today. And
  the subscriber is associated with the taskprocessor the message is being
  dispatched to.
  
  First, this patch removes the unused topic parameter from Stasis
  subscription callbacks.
  
  Second, this patch introduces the concept of taskprocessor local data,
  data that may be set on a taskprocessor and provided along with the data
  pointer when a task is pushed using the ast_taskprocessor_push_local()
  call. This allows the task to have both data specific to that
  taskprocessor, in addition to data specific to that invocation.
  
  With those two changes, the dispatch object can be removed completely,
  and the message is simply refcounted and sent directly to the
  taskprocessor.
  
  Review: https://reviewboard.asterisk.org/r/2884/
........

Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2de42c2a
  31. Sep 28, 2013
  33. Aug 30, 2013
    • David M. Lee's avatar
      optional_api: Fix linking problems between modules that export global symbols · 9bed50db
      David M. Lee authored 
      With the new work in Asterisk 12, there are some uses of the
optional_api that are prone to failure. The details are rather involved,
and captured on [the wiki][1].

This patch addresses the issue by removing almost all of the magic from
the optional API implementation. Instead of relying on weak symbol
resolution, a new optional_api.c module was added to Asterisk core.

For modules providing an optional API, the pointer to the implementation
function is registered with the core. For modules that use an optional
API, a pointer to a stub function, along with a optional_ref function
pointer are registered with the core. The optional_ref function pointers
is set to the implementation function when it's provided, or the stub
function when it's now.

Since the implementation no longer relies on magic, it is now supported
on all platforms. In the spirit of choice, an OPTIONAL_API flag was
added, so we can disable the optional_api if needed (maybe it's buggy on
some bizarre platform I haven't tested on)

The AST_OPTIONAL_API*() macros themselves remained unchanged, so
existing code could remain unchanged. But to help with debugging the
optional_api, the patch limits the #include of optional API's to just
the modules using the API. This also reduces resource waste maintaining
optional_ref pointers that aren't used.

Other changes made as a part of this patch:
 * The stubs for http_websocket that wrap system calls set errno to
   ENOSYS.

 * res_http_websocket now properly increments module use count.

 * In loader.c, the while() wrappers around dlclose() were removed. The
   while(!dlclose()) is actually an anti-pattern, which can lead to
   infinite loops if the module you're attempting to unload exports a
   symbol that was directly linked to.

 * The special handling of nonoptreq on systems without weak symbol
   support was removed, since we no longer rely on weak symbols for
   optional_api.

 [1]: https://wiki.asterisk.org/wiki/x/wACUAQ

(closes issue ASTERISK-22296)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2797/
........

Merged revisions 397989 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      9bed50db
  35. Aug 23, 2013
  37. Aug 01, 2013
  39. Jul 03, 2013
    • David M. Lee's avatar
      Shuffle RESTful URL's around. · dcf03554
      David M. Lee authored 
      This patch moves the RESTful URL's around to more appropriate
locations for release.

The /stasis URL's are moved to /ari, since Asterisk REST Interface was
a more appropriate name than Stasis-HTTP. (Most of the code still has
stasis_http references, but they will be cleaned up after there are no
more outstanding branches that would have merge conflicts with such a
change).

A larger change was moving the ARI events WebSocket off of the shared
/ws URL to its permanent home on /ari/events. The Swagger code
generator was extended to handle "upgrade: websocket" and
"websocketProtocol:" attributes on an operation.

The WebSocket module was modified to better handle WebSocket servers
that have a single registered protocol handler. If a client
connections does not specify the Sec-WebSocket-Protocol header, and
the server has a single protocol handler registered, the WebSocket
server will go ahead and accept the client for that subprotocol.

(closes issue ASTERISK-21857)
Review: https://reviewboard.asterisk.org/r/2621/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      dcf03554
  41. Jun 03, 2013
    • David M. Lee's avatar
      Correct autoconf script for finding UUID support. · 6d805dc0
      David M. Lee authored 
      The library that provides UUID support varies greatly from system to
system. On most Linux distros, it's in libuuid. On OpenBSD, it's in
libe2fs-uuid. On OS X, it is in libsystem.

This patch plays hide-and-seek with UUID support, looking for it in the
three places we know about. It also corrects the Makefile so that it uses
the configured library name and include path.

(closes issue ASTERISK-21816)
Reported by: Brad Latus (snuffy)
Tested by: Brad Latus (snuffy)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      6d805dc0
  43. May 15, 2013
  45. Mar 22, 2013
    • David M. Lee's avatar
      Move more channel events to Stasis; move res_json.c to main/json.c. · cf9324b2
      David M. Lee authored 
      This patch started out simply as fixing the bouncing tests introduced
in r382685, but required some other changes to give it a decent
implementation.

To fix the bouncing tests, the UserEvent and Newexten AMI events
needed to be refactored to dispatch via Stasis. Dispatching directly
to AMI resulted in those events sometimes getting ahead of the
associated Newchannel events, which would understandably confuse anyone.

I found that instead of creating a zillion different message types and
structures associated with them, it would be preferable to define a
message type that has a channel snapshot and a blob of structured data
with a small bit of additional information. The JSON object model
provides a very nice way of representing structured data, so I went
with that.

 * Move JSON support from res_json.c to main/json.c
   * Made libjansson-dev a required dependency
 * Added an ast_channel_blob message type, which has a channel
   snapshot and JSON blob of data.
 * Changed UserEvent and Newexten events so that they are dispatched
   via ast_channel_blob messages on the channel's topic.
 * Got rid of the ast_channel_varset message; used ast_channel_blob
   instead.
 * Extracted the manager functions converting Stasis channel events to
   AMI events into manager_channel.c.

(issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      cf9324b2
  47. Mar 12, 2013
  49. Feb 26, 2013
  51. Feb 22, 2013
  53. Jan 30, 2013
  55. Jan 19, 2013
  57. Jan 18, 2013
  59. Jan 11, 2013
    • Automerge script's avatar
      Merged revisions 378915,378918 via svnmerge from · 2c1720b4
      Automerge script authored 
      file:///srv/subversion/repos/asterisk/trunk

................
  r378915 | dlee | 2013-01-11 16:31:42 -0600 (Fri, 11 Jan 2013) | 21 lines
  
  Add JSON API for Asterisk.
  
  This provides a JSON API by pulling in and wrapping the Jansson JSON
  library[1]. The Asterisk API basically mirrors the Jansson
  functionality, with a few minor tweaks.
  
   * Some names have been asteriskified to protect the innocent.
   * Jansson provides both reference-stealing and reference-borrowing
     versions of several API's. The Asterisk API is exclusively
     reference-stealing for operations that put elements into arrays and
     objects.
   * No support for doubles, since we usually don't need that.
   * Coming along for the ride is the ast_test_validate macro, which made
     the unit tests much easier to write.
  
   [1]: http://www.digip.org/jansson/
  
  (issue ASTERISK-20887)
  (closes issue ASTERISK-20888)
  Review: https://reviewboard.asterisk.org/r/2264/
................
  r378918 | file | 2013-01-11 17:05:38 -0600 (Fri, 11 Jan 2013) | 11 lines
  
  Retain XMPP filters across reconnections so external modules continue to function as expected.
  
  Previously if an XMPP client reconnected any filters added by an external module were lost.
  This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling.
  
  (closes issue ASTERISK-20916)
  Reported by: kuj
  ........
  
  Merged revisions 378917 from http://svn.asterisk.org/svn/asterisk/branches/11
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378927 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2c1720b4
    • David M. Lee's avatar
      Add JSON API for Asterisk. · 7695ea26
      David M. Lee authored 
      This provides a JSON API by pulling in and wrapping the Jansson JSON
library[1]. The Asterisk API basically mirrors the Jansson
functionality, with a few minor tweaks.

 * Some names have been asteriskified to protect the innocent.
 * Jansson provides both reference-stealing and reference-borrowing
   versions of several API's. The Asterisk API is exclusively
   reference-stealing for operations that put elements into arrays and
   objects.
 * No support for doubles, since we usually don't need that.
 * Coming along for the ride is the ast_test_validate macro, which made
   the unit tests much easier to write.

 [1]: http://www.digip.org/jansson/

(issue ASTERISK-20887)
(closes issue ASTERISK-20888)
Review: https://reviewboard.asterisk.org/r/2264/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7695ea26
  61. Dec 13, 2012
  63. Dec 11, 2012