Skip to content
Snippets Groups Projects
Select Git revision
  • fix_fallback_from_tls
  • asterisk_rdkb
  • fix_fallback
  • 14666_fxs_no_wideband_codec
  • fix_srv_records
  • wenpeng-test-250121
  • wenpeng-registration-test
  • fix_deadlock_in_bridge_peer_functions
  • devel default protected
  • securesip_pjproject_part1
  • PJ_Example_Review
  • fix_re_register_authorization_header
  • fix_internal_calls_auth
  • fix_internal_calls
  • Secure_SIPtelephony_part1
  • SIPtelephony_DECT_syslog
  • syslog_develop_phase1
  • merge-asterisk-20.8.1
  • asterisk_rdkb_temp
  • bug_14682_voicemail
  • 22.0.0-pre1
  • 21.4.2
  • 20.9.2
  • 18.24.2
  • certified-20.7-cert2
  • certified-18.9-cert11
  • 21.4.1
  • 20.9.1
  • 18.24.1
  • 21.4.0
  • 20.9.0
  • 18.24.0
  • certified-20.7-cert1
  • certified-18.9-cert10
  • 21.4.0-rc1
  • 20.9.0-rc1
  • 18.24.0-rc1
  • 21.3.1
  • 20.8.1
  • 18.23.1
40 results
Search by author
  • Any Author
  • Adam Porteous Adam Porteous adam.porteous
  • Alex Hopkins Alex Hopkins 59hopper
  • Amin Ben Romdhane Amin Ben Romdhane amin.benromdhane
  • Andreas Gnau Andreas Gnau AGnau
  • Andrej Kucharik Andrej Kucharik andrej.kucharik
  • Andrii Pypa Andrii Pypa andrii.pypa
  • Anjan Chanda Anjan Chanda anjancx
  • Balalakshmi Arunachalam Rajendran Balalakshmi Arunachalam Rajendran arbala
  • Benjamin Larsson Benjamin Larsson benjamin
  • Charles Foster Charles Foster charles.foster_bt.com
  • Dariusz Iwanoczko Dariusz Iwanoczko dariusz.iwanoczko
  • Divya Chandrasekaran Divya Chandrasekaran divya.c
  • Dominic Lake Dominic Lake dominic.lake
  • Elena Vengerova Elena Vengerova helen
  • Erik Karlsson Erik Karlsson erik.karlsson
  • Filip Matusiak Filip Matusiak filmat
  • Fredrik Östling Fredrik Östling fredrik.ostling
  • Geoff Elliott Geoff Elliott gdelliott
  • George Yang George Yang gyang
  • Gouri Natarajan Gouri Natarajan gouri.natarajan
20 authors
  1. Feb 08, 2011
  3. Nov 11, 2010
  5. Oct 14, 2010
  7. Sep 21, 2010
  9. Sep 14, 2010
  11. Sep 10, 2010
  13. Sep 02, 2010
  15. Aug 16, 2010
  17. Aug 03, 2010
  19. Jul 23, 2010
  21. Jul 16, 2010
  23. Jul 14, 2010
  25. Jul 07, 2010
  27. Jun 28, 2010
  29. Jun 25, 2010
  31. Jun 16, 2010
  33. Jun 15, 2010
  35. Jun 12, 2010
  37. Jun 10, 2010
  39. 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
  41. Jun 02, 2010
    • David Vossel's avatar
      Update CHANGES and aoc help doc to reflect AOC additions · 3280a5c0
      David Vossel authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      3280a5c0
    • Richard Mudgett's avatar
      Generic Advice of Charge. · afd4454c
      Richard Mudgett authored 
      Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
  (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages

Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup

AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.

SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
  snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
  'snom_aoc_enabled' sip.conf option.

IAX AOC Support
- Natively supports AOC pass-through through the use of the new
  AST_CONTROL_AOC frame type

DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
  pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
  example usage:
  ;requests AOC-S, AOC-D, and AOC-E on call setup
  exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))

Review:	https://reviewboard.asterisk.org/r/552/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      afd4454c
  43. May 24, 2010
  45. May 03, 2010
  47. Apr 22, 2010
  49. Apr 21, 2010
  51. Apr 15, 2010
  53. Apr 12, 2010
    • Leif Madsen's avatar
      Merged revisions 256900 via svnmerge from · 47d8fdae
      Leif Madsen authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) | 15 lines
  
  Add How-To document on collecting debugging info for issues.asterisk.org
  
  Paul Belanger has been helping a lot with bug tracking recently and created
  this document that we can now point to when additional debugging information
  is required. This document will help those filing issues to know how to get
  the information required when filing their issues. This will make things
  easier on the developers.
  
  Initial text and changes by pabelanger. Tweaks and editing by myself.
  
  (closes issue #17159)
  Reported by: pabelanger
  Patches: 
        HOWTO_collect_debug_information.txt.patch uploaded by lmadsen (license 10)
  Tested by: tzafrir, pabelanger, lmadsen
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      47d8fdae
  55. Apr 10, 2010
  57. Apr 09, 2010
    • Richard Mudgett's avatar
      Merge CCSS architecture document from CCSS branch. · d66b44b4
      Richard Mudgett authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d66b44b4
    • 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