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  1. Dec 23, 2011
    • Jonathan Rose's avatar
      INFO/Record request configurable to use dynamic features · 19a4928f
      Jonathan Rose authored 
      Adds two new options to SIP peers allowing them to specify features (dynamic or builtin)
to use when sending INFO/record requests. Recordonfeature activates whatever feature
is specified when recieving a record: on request while recordofffeature activates
whatever feature is specified when receiving a record: off request. Both of these
features can be disabled by setting the feature to an empty string.

(closes issue ASTERISK-16507)
Reported by: Jon Bright
Review: https://reviewboard.asterisk.org/r/1634/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      19a4928f
    • Jonathan Rose's avatar
      chan_sip autocreatepeer=persist option for auto-created peers to survive reload · 03596bcb
      Jonathan Rose authored 
      This patch moves destruction of sip peers to immediately after the general section of
sip.conf is read so that autocreatepeer setting can be read before deletion of peers.
If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting
will be skipped when purging the current SIP peer list.

(closes ASTERISK-16508)
Reported by: Kirill Katsnelson
Patches:
	017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill Katsnelson (license 5845)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      03596bcb
    • Sean Bright's avatar
      Merged revisions 349045 via svnmerge from · 35a64c2e
      Sean Bright authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r349045 | seanbright | 2011-12-23 12:32:33 -0500 (Fri, 23 Dec 2011) | 25 lines
  
  Merged revisions 349044 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec 2011) | 18 lines
    
    In ChanSpy, don't create audiohooks that will never be used.
    
    When ChanSpy is initialized it creates and attaches 3 audiohooks:
    
      1) Read audio off of the channel that we are spying on
      2) Write audio to the channel that we are spying on
      3) Write audio to the channel that is bridged to the channel that we are
         spying on.
    
    The first is always necessary, but the others are used only when specific
    options are passed to the ChanSpy application (B, d, w, and W to be specific).
    
    When those flags are not passed, neither of those audiohooks are ever sent
    frames, but we still try to process the hooks for each voice frame that we
    recieve on the channel.
    
    So in short - only create and attach audiohooks that we actually need.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      35a64c2e
    • Kinsey Moore's avatar
      Fix missing doc tags found while fixing ASTERISK-18689 · 011843e3
      Kinsey Moore authored 
      Add missing <variable></variable> tags in app_dial documentation.
........

Merged revisions 348992 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 348993 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      011843e3
    • Richard Mudgett's avatar
      Fix extension state callback references in chan_sip. · 32e35e5f
      Richard Mudgett authored 
      Chan_sip gives a dialog reference to the extension state callback and
assumes that when ast_extension_state_del() returns, the callback cannot
happen anymore.  Chan_sip then reduces the dialog reference count
associated with the callback.  Recent changes (ASTERISK-17760) have
resulted in the potential for the callback to happen after
ast_extension_state_del() has returned.  For chan_sip, this could be very
bad because the dialog pointer could have already been destroyed.

* Added ast_extension_state_add_destroy() so chan_sip can account for the
sip_pvt reference given to the extension state callback when the extension
state callback is deleted.

* Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy()
and handle_statechange() now that the struct ast_state_cb has a destructor
to call.

* Ensure that ast_extension_state_add_destroy() will never return -1 or 0
for a successful registration.

* Fixed pbx.c statecbs_cmp() to compare the correct information.  The
passed in value to compare is a change_cb function pointer not an object
pointer.

* Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with
AST_EXTENSION_REMOVED with locks held.  Chan_sip is notorious for
deadlocking when those locks are held during the callback.

* Removed unused lock declaration for the pbx.c store_hints list.

(closes issue ASTERISK-18844)
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/1635/
........

Merged revisions 348940 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 348952 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      32e35e5f
  3. Dec 22, 2011
  5. Dec 21, 2011
  7. Dec 20, 2011
  9. Dec 19, 2011
  11. Dec 18, 2011
  13. Dec 16, 2011
  15. Dec 14, 2011
  17. Dec 13, 2011
  19. Dec 12, 2011
  21. Dec 09, 2011
    • Jonathan Rose's avatar
      Adds MixMonitor and StopMixMonitor AMI commands to the manager · e8181c22
      Jonathan Rose authored 
      These commands work much like the dialplan applications that would otherwise invoke them.
A nice benefit of these is that they can be invoked on a call remotely and at any time
during a call. They work much like the Monitor and StopMonitor ami commands.

(closes issue ASTERISK-17726)
Reported by: Sergio González Martín
Patches:
	mixmonitor_actions.diff uploaded by Sergio González Martín (license 5644)
Review: https://reviewboard.asterisk.org/r/1193/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e8181c22
    • Jonathan Rose's avatar
      Remove autojump extensions from SayUnixTime, make an option to perform automatic jumps. · 518ccb67
      Jonathan Rose authored 
      When a caller sends DTMF while the SayUnixTime application is saying the time, The call
would jump to the next extension much like it does during Background(). This patch adds
option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch
allows arguments to sayunixtime to not be used as empty strings in the case of something
like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern).

(closes issue ASTERISK-16675)
Reported by: jlpedrosa
Patches:
	patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959)
Review: https://reviewboard.asterisk.org/r/956/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      518ccb67
    • Richard Mudgett's avatar
      Fix some parsing issues in add_exten_to_pattern_tree(). · 01d3fd21
      Richard Mudgett authored 
      * Simplify compare_char() and avoid potential sign extension issue.

* Fix infinite loop in add_exten_to_pattern_tree() handling of character
set escape handling.

* Added buffer overflow checks in add_exten_to_pattern_tree() character
set collection.

* Made ignore empty character sets.

* Added escape character handling to end-of-range character in character
sets.  This has a slight change in behavior if the end-of-range character
is an escape character.  You must now escape it.

* Fix potential sign extension issue when expanding character set ranges.

* Made remove duplicated characters from character sets.  The duplicate
characters lower extension matching priority and prevent duplicate
extension detection.

* Fix escape character handling when the escape character is trying to
escape the end-of-string.  We could have continued processing characters
after the end of the exten string.  We could have added the previous
character to the pattern matching tree incorrectly.

(closes issue ASTERISK-18909)
Reported by: Luke-Jr
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Merged revisions 347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 347812 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      01d3fd21
  23. Dec 08, 2011