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20 authors
  1. Apr 15, 2011
  3. Apr 14, 2011
    • Terry Wilson's avatar
      Sets video mark bit on format field correctly · e9ba0cba
      Terry Wilson authored 
      This fixes a regression in the media architecture change
where video frames did not have their video mark set
correctly.  dvossel wrote this. twilson kindly committed
this, mmichelson found the bug.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e9ba0cba
    • Richard Mudgett's avatar
      Merged revisions 313780 via svnmerge from · 4f8d56a8
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14 Apr 2011) | 20 lines
  
  Leftover debug messages unconditionally sent to the console.
  
  Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config
  option enabled outputs the following debug messages unconditionally:
  
  Dialing T1847555121 on 1
  Dialing www2w on 1
  
  * Made debug messages in my_dial_digits() normal debug messages that do
  not get output unless enabled.
  
  * Reworded some debug messages in my_dial_digits() to be clearer.
  
  * Replace strncpy() with ast_copy_string() in my_dial_digits() which does
  the same job better.
  
  (closes issue #18847)
  Reported by: vmikhelson
  Tested by: rmudgett
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4f8d56a8
    • Richard Mudgett's avatar
      Add Device State Information CCSS for Generic Devices. · ae2926b5
      Richard Mudgett authored 
      Add Asterisk Device State information and callbacks to the Call Completion
Supplemental Services for generic agents.

There are currently not many devices that have native support for CCSS.
Even as the devices become available there may be other reasons why one
may choose to not take advantage of the native abilities and stick with
the generic implementation.  The generic implementation is quite capable
and could be greatly enhanced by adding device state capabilities.  A
phone could then subscribe to the device state with a BLF key in
conjunction with Asterisk hints.

The advantages of the device state information would allow a single button
to: request CCSS, cancel a CCSS request, and display the current state of
a CCSS request.

For example, you may have a single button that when not lit, there is no
active CCSS request.  When you press that button, the dialplan can query
the DEVICE_STATE() associated with that caller to determine whether they
should be calling CallCompletionRequest() or CallCompletionCancel().  If
there is currently a pending request, then the dialplan would cancel it.
This also has the advantage of showing the true state of a request, which
is an asynchronous call, even when CallCompletionRequest() thinks it was
successful.  The actual request could ultimately fail.  Once lit, further
feedback can be provided to the caller about the current state of their
request since it will be updated by the CCSS State Machine as appropriate.

The DEVICE_STATE mapping is configurable since the BLF being used on a
given phone type may vary.  The idea is to allow some level of
customization as to the phone's behavior.

As an example, you may want the BLF key to go solid once you have
requested a callback.  You may then want the LED to blink (typically
ringing) when either the callback is in process, which is a visual
indication that the incoming call is the desired callback.  You may want
it to blink when the callee is ready but you are busy, giving you a visual
indication that the target is available as you may want to get off the
line so that the callback can be successful.

Device state information is sent back via the ast_devstate_prov_add()
callback for any generic CCSS device as it traverses through the state
machine.  You simply provide a map between CC_STATE values and the
corresponding AST_DEVICE state values.

You could then generate hints against these states similar to what is
possible today with Custom Devstates or MeetMe states.  For example, you
may have an extension 3000 that is currently associated with device
SIP/3000.  You could then create a feature code for that extension that
may look something like:

exten => *823000,hint,ccss:sip/3000

You would then subscribe a BLF button to *823000 which would point to the
dialplan that handled CCSS requests/cancels using the available
DEVICE_STATE() information about ccss:sip/3000 to make the decision about
what to do.

(closes issue #18788)
Reported by: p_lindheimer
Patches:
      ccss.trunk.18788.patch uploaded by p lindheimer (license 558)
      Modified with final reviewboard comments.
Tested by: p_lindheimer, loloski

Review: https://reviewboard.asterisk.org/r/1105/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ae2926b5
  5. Apr 13, 2011
  7. Apr 12, 2011
  9. Apr 11, 2011
  11. Apr 09, 2011
  13. Apr 08, 2011
  15. Apr 07, 2011
  17. Apr 05, 2011
  19. Apr 04, 2011
  21. Apr 01, 2011
    • Richard Mudgett's avatar
      Merged revisions 312509 via svnmerge from · 6826b083
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r312509 | rmudgett | 2011-04-01 18:15:42 -0500 (Fri, 01 Apr 2011) | 22 lines
  
  When a call going out an NT-PTMP port gets rejected, Asterisk crashes.
  
  If a call is sent to an ISDN phone that rejects the call with
  RELEASE_COMPLETE(cause: call reject(21), or busy(17)) Asterisk crashes.
  
  I could not get my setup to crash.  However, I could see the possibility
  from a race condition between queuing an AST_CONTROL_BUSY to the core and
  then queueing an AST_CONTROL_HANGUP.  If the AST_CONTROL_BUSY is processed
  before the AST_CONTROL_HANGUP is queued, the ast_channel could be
  destroyed out from under chan_misdn.
  
  Avoid this particular crash scenario by not queueing the
  AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued.
  
  (closes issue #18408)
  Reported by: wimpy
  Patches:
        issue18408_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: rmudgett, wimpy
  
  JIRA SWP-2679
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      6826b083
    • Richard Mudgett's avatar
      Merged revisions 312461 via svnmerge from · 75594e6e
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines
  
  CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail.
  
  The CallCompletionRequest()/CallCompletionCancel() dialplan applications
  exit nonzero on normal failure conditions.  The nonzero exit causes the
  dialplan to hangup immediately.  The dialplan author has no opportunity to
  report success/failure to the user.
  
  * Made always return zero so the dialplan can continue.
  
  * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
  CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively.  Also
  documented the values set.
  
  * Reduced the warning about no core instance in CallCompletionCancel() to
  a debug message.  It is a normal event and should not be output at the
  WARNING level.
  
  (closes issue #18763)
  Reported by: p_lindheimer
  Patches:
        ccss.patch uploaded by p lindheimer (license 558) Modified
  Tested by: p_lindheimer, rmudgett
  
  JIRA SWP-3042
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      75594e6e
    • Jonathan Rose's avatar
      Fixing bad line break from 312384 · 759bf6b8
      Jonathan Rose authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      759bf6b8
    • Jonathan Rose's avatar
      New Feature for chan_dahdi. 4 length pattern matching. · 846cfa0e
      Jonathan Rose authored 
      In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns.  The s
ntax remains the same and the method used to track the pattern history will only change when using the length
 4 patterns.

(closes issue SWP-3250)
Code:
        jrose
        rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      846cfa0e
    • Tilghman Lesher's avatar
      Merged revisions 312286,312288 via svnmerge from · 3731fd9c
      Tilghman Lesher authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r312286 | tilghman | 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) | 2 lines
  
  Reload must react correctly against a possibly changed table, so dropping the conditional reload flag.
................
  r312288 | tilghman | 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) | 21 lines
  
  Merged revisions 312287 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines
    
    Merged revisions 312285 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines
      
      Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code.
      
      (issue #18969)
       Reported by: oej
       Patches: 
             20110315__issue18969__14.diff.txt uploaded by tilghman (license 14)
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      3731fd9c
    • Alec L Davis's avatar
      Merged revisions 312211 via svnmerge from · e59a051c
      Alec L Davis authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r312211 | alecdavis | 2011-04-01 22:03:11 +1300 (Fri, 01 Apr 2011) | 36 lines
  
  Merged revisions 312210 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312210 | alecdavis | 2011-04-01 21:47:29 +1300 (Fri, 01 Apr 2011) | 29 lines
    
    Merged revisions 312174 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr 2011) | 23 lines
      
      voicemail: get real last_message_index and count_messages, ODBC resequence
      
      change last_message_index to read the max msgnum stored in the database
      change count_messages to actually count the number of messages.
      
      last_message_index change:
        This fixed overwriting of the last message if msgnum=0 was missing.
        Previously every incoming message would overwrite msgnum=1.
      count_messages change:
        allows us to detect when requencing is required in opneA_mailbox.
      resequence enabled for ODBC storage:
        Assists with fixing up corrupt databases with gaps, but only when
        a user actively opens there mailboxes.
      
      (closes issue #18692,#18582,#19032)
      Reported by: elguero
      Patches: 
            based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37)
      Tested by: elguero, nivek, alecdavis
      
      Review: https://reviewboard.asterisk.org/r/1153/
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e59a051c
    • Alec L Davis's avatar
      Merged revisions 312117 via svnmerge from · d07fb85b
      Alec L Davis authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r312117 | alecdavis | 2011-04-01 20:32:12 +1300 (Fri, 01 Apr 2011) | 29 lines
  
  Merged revisions 312103 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312103 | alecdavis | 2011-04-01 20:25:54 +1300 (Fri, 01 Apr 2011) | 22 lines
    
    Merged revisions 312070 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr 2011) | 16 lines
      
      app_voicemail: close_mailbox needs to respect additional messages while mailbox is open.
      
      close_mailbox leave gaps in message sequence if messages are deleted and new messages
      arrive during this time, this is because the shuffle down to slot 0, only shuffles
      the number of pre-existing messages when mailbox is opened, ignoring new arrivals.
      
      Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals.
      
      Happens on filebased or ODBC storage.
      
      (issues #19032,#18582,#18692,#18998)
      Reported by: alecdavis,tootai,afosorio
      
      Review: https://reviewboard.asterisk.org/r/1153/
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d07fb85b
  23. Mar 31, 2011