Skip to content
Snippets Groups Projects
Select Git revision
  • pjsip_debug_early_media
  • dev_fxs_no_wb
  • fix_re_register_authorization_header
  • 14600-wenpeng
  • asterisk_rdkb
  • lk_channel_joined_bridge_indication
  • fix_fallback
  • 14666_fxs_no_wideband_codec
  • fix_srv_records
  • wenpeng-registration-test
  • fix_deadlock_in_bridge_peer_functions
  • devel default protected
  • securesip_pjproject_part1
  • PJ_Example_Review
  • fix_internal_calls_auth
  • fix_internal_calls
  • Secure_SIPtelephony_part1
  • SIPtelephony_DECT_syslog
  • syslog_develop_phase1
  • merge-asterisk-20.8.1
  • 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
  • Filip Strelec Filip Strelec filip.strelec
  • Fredrik Östling Fredrik Östling fredrik.ostling
  • Geoff Elliott Geoff Elliott gdelliott
  • George Yang George Yang gyang
20 authors
  1. Jun 04, 2012
    • Mark Michelson's avatar
      Remove automerge properties. · 458f6c4b
      Mark Michelson authored 
      
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      458f6c4b
    • Mark Michelson's avatar
      Merge changes dealing with support for Digium phones. · 14a98556
      Mark Michelson authored 
      Presence support has been added. This is accomplished by
allowing for presence hints in addition to device state
hints. A dialplan function called PRESENCE_STATE has been
added to allow for setting and reading presence. Presence
can be transmitted to Digium phones using custom XML
elements in a PIDF presence document.

Voicemail has new APIs that allow for moving, removing,
forwarding, and playing messages. Messages have had a new
unique message ID added to them so that the APIs will work
reliably. The state of a voicemail mailbox can be obtained
using an API that allows one to get a snapshot of the mailbox.
A voicemail Dialplan App called VoiceMailPlayMsg has been
added to be able to play back a specific message.

Configuration hooks have been added. Configuration hooks
allow for a piece of code to be executed when a specific
configuration file is loaded by a specific module. This is
useful for modules that are dependent on the configuration
of other modules.

chan_sip now has a public method that allows for a custom
SIP INFO request to be sent mid-dialog. Digium phones use
this in order to display progress bars when files are played.

Messaging support has been expanded a bit. The main
visible difference is the addition of an AMI action
MessageSend.

Finally, a ParkingLots manager action has been added in order
to get a list of parking lots.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      14a98556
    • Richard Mudgett's avatar
      Fix potential deadlock between masquerade and chan_local. · c1bbe797
      Richard Mudgett authored 
      * Restructure ast_do_masquerade() to not hold channel locks while it calls
ast_indicate().

* Simplify many calls to ast_do_masquerade() since it will never return a
failure now.  If it does fail internally because a channel driver callback
operation failed, the only thing ast_do_masquerade() can do is generate a
warning message about strange things may happen and press on.

* Fixed the call to ast_bridged_channel() in ast_do_masquerade().  This
change fixes half of the deadlock reported in ASTERISK-19801 between
masquerades and chan_iax.

(closes issue ASTERISK-19537)
Reported by: rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1915/
........

Merged revisions 368405 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 368407 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c1bbe797
  3. Jun 02, 2012
  5. Jun 01, 2012
  7. May 31, 2012
  9. May 30, 2012
  11. May 29, 2012
  13. May 25, 2012
  15. May 24, 2012
  17. May 23, 2012
  19. May 22, 2012
  21. May 21, 2012
  23. May 18, 2012
    • Jonathan Rose's avatar
      app_queue: Per Member ringinuse option and deprecation of ignorebusy · ec3b8a1f
      Jonathan Rose authored 
      Adds a number of methods for controlling the setting of 'ringinuse'
which is basically the same concept as the old ignorebusy setting,
only now the per member setting always controls whether or not the
member is actually ringed while in use. A CLI command and a manager
action have been added to change a given queue member's ringinuse
option while Asterisk is running and the an argument has been added
for adding members with deliberately set ringinuse in queues.conf
Some effort has been made to ensure compatability with dialplans and
databases still referring to 'ignorebusy'.

(issue ASTERISK-19536)
reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1919/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ec3b8a1f
    • Mark Michelson's avatar
      Address MISSING_BREAK static analysis reports some more. · 11348736
      Mark Michelson authored 
      This addresses core findings 4 and 6.

Moises Silva helped me by stating that a break could be
safely added to the case where it is added in chan_dahdi.c

In say.c, I have added a comment indicating that static analysis
complains but that it is currently unknown if this is correct.

This fixes all core findings of this type.

(closes issue ASTERISK-19662)
reported by Matthew Jordan
........

Merged revisions 367027 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 367028 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      11348736
    • Mark Michelson's avatar
      Fix memory leak of SSL_CTX structures in TLS core. · 5c576aa3
      Mark Michelson authored 
      SSL_CTX structures were allocated but never freed. This was a bigger
issue for clients than servers since new SSL_CTX structures could be
allocated for each connection. Servers, on the other hand, typically
set up a single SSL_CTX for their lifetime.

This is solved in two ways:

1. In __ssl_setup(), if a tcptls_cfg has an ssl_ctx on it, it is
freed so that a new one can take its place.
2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
been added so that servers can properly free their SSL_CTXs.

(issue ASTERISK-19278)
........

Merged revisions 367002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 367003 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      5c576aa3
    • Matthew Jordan's avatar
      Fix more memory leaks · 6eb4e810
      Matthew Jordan authored 
      This patch adds to what was fixed in r366880.  Specifically, it addresses the
following:

* chan_sip:   dispose of an allocated frame in off nominal code paths in
              sip_rtp_read
* func_odbc:  when disposing of an allocated resultset, ensure that any rows
              that were appended to that resultset are also disposed of
* cli:        free the created return string buffer in another off nominal code
              path
* chan_dahdi: free a frame that was allocated by the dsp layer if we choose
              not to process that frame

(issue ASTERISK-19665)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1922/
........

Merged revisions 366944 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 366948 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      6eb4e810
    • Matthew Jordan's avatar
      Fix a variety of memory leaks · 7b513206
      Matthew Jordan authored 
      This patch addresses a number of memory leaks in a variety of modules that were
found by a static analysis tool.  A brief summary of the changes:

* app_minivm:       free ast_str objects on off nominal paths
* app_page:         free the ast_dial object if the requested channel technology
                    cannot be appended to the dialing structure
* app_queue:        if a penalty rule failed to match any existing rule list
                    names, the created rule would not be inserted and its memory
                    would be leaked
* app_read:         dispose of the created silence detector in the presence of
                    off nominal circumstances
* app_voicemail:    dispose of an allocated unique ID field for MWI event
                    un-subscribe requests in off nominal paths; dispose of
                    configuration objects when using the secret.conf option
* chan_dahdi:       dispose of the allocated frame produced by ast_dsp_process
* chan_iax2:        properly unref peer in CLI command "iax2 unregister"
* chan_sip:         dispose of the allocated frame produced by sip_rtp_read's
                    call of ast_dsp_process; free memory in parse unit tests
* func_dialgroup:   properly deref ao2 object grhead in nominal path of
                    dialgroup_read
* func_odbc:        free resultset in off nominal paths of odbc_read
* cli:              free match_list in off nominal paths of CLI match completion
* config:           free comment_buffer/list_buffer when configuration file load
                    is unchanged; free the same buffers any time they were
                    created and config files were processed
* data:             free XML nodes in various places
* enum:             free context buffer in off nominal paths
* features:         free ast_call_feature in off nominal paths of applicationmap
                    config processing
* netsock2:         users of ast_sockaddr_resolve pass in an ast_sockaddr struct
                    that is allocated by the method.  Failures in
                    ast_sockaddr_resolve could result in the users of the method
                    not knowing whether or not the buffer was allocated.  The
                    method will now not allocate the ast_sockaddr struct if it
                    will return failure.
* pbx:              cleanup hash table traversals in off nominal paths; free
                    ignore pattern buffer if it already exists for the specified
                    context
* xmldoc:           cleanup various nodes when we no longer need them
* main/editline:    various cleanup of pointers not being freed before being
                    assigned to other memory, cleanup along off nominal paths
* menuselect/mxml:  cleanup of value buffer for an attribute when that attribute
                    did not specify a value
* res_calendar*:    responses are allocated via the various *_request method
                    returns and should not be allocated in the various
                    write_event methods; ensure attendee buffer is freed if no
                    data exists in the parsed node; ensure that calendar objects
                    are de-ref'd appropriately
* res_jabber:       free buffer in off nominal path
* res_musiconhold:  close the DIR* object in off nominal paths
* res_rtp_asterisk: if we run out of ports, close the rtp socket object and free
                    the rtp object
* res_srtp:         if we fail to create the session in libsrtp, destroy the
                    temporary ast_srtp object

(issue ASTERISK-19665)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1922
........

Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7b513206
    • Jonathan Rose's avatar
      chan_sip: Fix a small TEST_FRAMEWORK related error that prevents compiling · 6fc8e992
      Jonathan Rose authored 
      Introduced with r366842, a function call made only with TEST_FRAMEWORK enabled
was missing an argument since the function arguments were changed.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      6fc8e992