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20 authors
  1. Sep 21, 2010
  3. Sep 14, 2010
  5. Sep 02, 2010
  7. Aug 16, 2010
  9. Jul 16, 2010
  11. Jul 14, 2010
  13. Jun 28, 2010
  15. Jun 16, 2010
  17. Jun 10, 2010
  19. Jun 08, 2010
    • Terry Wilson's avatar
      Add SRTP support for Asterisk · 857814f4
      Terry Wilson authored 
      After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.

Original patch by mikma, updated for trunk and revised by me.

(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel

Review: https://reviewboard.asterisk.org/r/191/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      857814f4
  21. May 24, 2010
    • Terry Wilson's avatar
      Add the FullyBooted AMI event · 3c9a8eba
      Terry Wilson authored 
      It is possible to connect to the manager interface before all Asterisk modules
are loaded. To ensure that an application does not send AMI actions that might
require a module that has not yet loaded, the application can listen for the
FullyBooted manager event. It will be sent upon connection if all modules have
been loaded, or as soon as loading is complete. The event:

   Event: FullyBooted
   Privilege: system,all
   Status: Fully Booted

Review: https://reviewboard.asterisk.org/r/639/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      3c9a8eba
  23. Apr 22, 2010
  25. Apr 21, 2010
  27. Apr 09, 2010
    • Mark Michelson's avatar
      Merge Call completion support into trunk. · e24661fd
      Mark Michelson authored 
      From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:

1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
   multiple calls to the same device. This proved to not be such a good idea
   when implementing protocol-specific monitors, and so we ended up using one
   monitor per-device per-call.
3. There are some configuration options which were conceived after the document
   was written. These are documented in the ccss.conf.sample that is on this
   review request.
		      
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.

This implements CCBS and CCNR in several flavors.

First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.

Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:

* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
  what is defined in the referenced draft.

* Implementation of the draft required support for SIP PUBLISH. I attempted to write
  this in a generic-enough fashion such that if someone were to want to write PUBLISH
  support for other event packages, such as dialog-state or presence, most of the effort
  would be in writing callbacks specific to the event package.

* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
  parser. The PIDF support added is a bit minimal. I first wrote a validation
  routine to ensure that the PIDF document is formatted properly. The rest of the
  PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
  code. In other words, while there is PIDF support here, it is not in any state
  where it could easily be applied to other event packages as is.

Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.

Review: https://reviewboard.asterisk.org/r/523


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e24661fd
  29. Apr 05, 2010
  31. Mar 18, 2010
  33. Mar 03, 2010
    • Leif Madsen's avatar
      Update existing Local channel documentation. · 06ff72a7
      Leif Madsen authored 
      A complete re-write of the Local channel documentation has been performed, with
the existing information from localchannel.txt and localchannel.tex merged in.

(closes issue #16637)
Reported by: kobaz
Patches: 
      localchannel.tex uploaded by lmadsen (license 10)
      localchannel.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, jsmith, mmichelson

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      06ff72a7
  35. Mar 02, 2010
  37. Jan 13, 2010
  39. Dec 14, 2009
  41. Nov 06, 2009
  43. Oct 29, 2009
  45. Oct 28, 2009
  47. Oct 21, 2009
  49. Aug 27, 2009
  51. Aug 18, 2009
  53. Aug 05, 2009
  55. Aug 03, 2009
    • Kevin P. Fleming's avatar
      Rename 'canreinvite' option to 'directmedia', with backwards compatibility. · e9d22f80
      Kevin P. Fleming authored 
      It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e9d22f80
  57. Aug 01, 2009
  59. Jul 17, 2009
  61. Jul 11, 2009
    • Russell Bryant's avatar
      Add an API for reporting security events, and a security event logging module. · 4cf8a968
      Russell Bryant authored 
      This commit introduces the security events API.  This API is to be used by
Asterisk components to report events that have security implications.
A simple example is when a connection is made but fails authentication.  These
events can be used by external tools manipulate firewall rules or something
similar after detecting unusual activity based on security events.

Inside of Asterisk, the events go through the ast_event API.  This means that
they have a binary encoding, and it is easy to write code to subscribe to these
events and do something with them.

One module is provided that is a subscriber to these events - res_security_log.
This module turns security events into a parseable text format and sends them
to the "security" logger level.  Using logger.conf, these log entries may be
sent to a file, or to syslog.

One service, AMI, has been fully updated for reporting security events.
AMI was chosen as it was a fairly straight forward service to convert.
The next target will be chan_sip.  That will be more complicated and will
be done as its own project as the next phase of security events work.

For more information on the security events framework, see the documentation
generated from doc/tex/.  "make asterisk.pdf"

Review: https://reviewboard.asterisk.org/r/273/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4cf8a968
  63. Jun 30, 2009
    • Russell Bryant's avatar
      Move Asterisk-addons modules into the main Asterisk source tree. · c511a267
      Russell Bryant authored 
      Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?".  After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.

For more information about why a module goes in addons, see README-addons.txt.

chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c511a267
  65. Jun 26, 2009
    • Russell Bryant's avatar
      Merge the new Channel Event Logging (CEL) subsystem. · 0264eef1
      Russell Bryant authored 
      CEL is the new system for logging channel events.  This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records.  For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.

Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code.  Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.

Review: https://reviewboard.asterisk.org/r/239/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      0264eef1
  67. Jun 23, 2009