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  r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines
  
  Misc minor changes in chan_sip.
  
  * Add load failure exit if primary SIP container(s) could not get created
  in chan_sip.c:load_module().
  
  * Removed a redundant static prototype.
  
  * Some typos.
  
  * Some whitespace.
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Review: https://reviewboard.asterisk.org/r/1288/



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  r325877 | mjordan | 2011-06-30 15:09:48 -0500 (Thu, 30 Jun 2011) | 9 lines
  
  Patched voicemail user option for emailbody / emailsubject
  
  Incorporated changes per ASTERISK-16795; updated unit tests to check for vmu->emailbody / vmu->emailsubject
  
  (closes issue ASTERISK-16795)
  Reported by: mdeneen
  Tested by: mjordan
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  r325821 | jrose | 2011-06-30 14:17:32 -0500 (Thu, 30 Jun 2011) | 10 lines
  
  Fixes an issue with Music on Hold classes losing files in playlist when realtime is used.
  
  The bug occurs rather intermittently and I relied on the reporters to test the patch.
  After a sanity check and some testing, I'm giving it an OK.
  
  (closes issue ASTERISK-17875)
  Reported by: David Cunningham
  Patches: 
        res_musiconhold.c.mohrt17875_v1 uploaded by Igor Goncharovsky (license #5009)
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terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the
dialplan.

Big thanks to irroot for porting this code to use the framehooks api.


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  r325740 | kmoore | 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) | 7 lines
  
  chan_sip: cleanup from the introduction of ast_str
  
  Remove the length field from sip_req and sip_pkt in chan_sip since they are
  redundant (ast_str holds its own length) and refactor the necessary functions.
  
  Review: https://reviewboard.asterisk.org/r/1281/
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  r325673 | dvossel | 2011-06-29 13:59:33 -0500 (Wed, 29 Jun 2011) | 6 lines
  
  Fixes timerfd locking issue.
  
  (closes ASTERISK-17867, ASTERISK-17415)
  Patches:
       fix uploaded by kobaz
  Review: https://reviewboard.asterisk.org/r/1255/
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  r325614 | rmudgett | 2011-06-29 13:16:45 -0500 (Wed, 29 Jun 2011) | 5 lines
  
  Fixed some error exit cleanup in app_queue.c.
  
  * Fixed error exit cleanup in app_queue.c copy_rules() and
  reload_queue_rules().
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  r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) | 18 lines
  
  Response to QueueRule manager command does not contain ActionID if it was specified.
  
  * Add ActionID support as documented for the QueueRule AMI action.
  
  * Remove documentation for ActionID with the Queues AMI action.  The
  output does not follow normal AMI response output and there is no place to
  put an ActionID header.
  
  (closes issue AST-602)
  Reported by: Vlad Povorozniuc
  Patches:
        jira_ast_602_v1.8.patch (license #5621) patch uploaded by rmudgett
  Tested by: Vlad Povorozniuc, rmudgett
  
  Review: https://reviewboard.asterisk.org/r/1295/
  
  JIRA SWP-3575
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  r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun 2011) | 2 lines
  
  make framehooks prevent native bridging (for real this time)
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  r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) | 2 lines
  
  don't do native/remote bridging if a framehook is active on the channel
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Commit "distrotech" app_queue changes to Trunk

 * Added general option negative_penalty_invalid default off. when set
   members are seen as invalid/logged out when there penalty is negative.  
   for realtime members when set remove from queue will set penalty to -1.  
 * Added queue option autopausedelay when autopause is enabled it will be
   delayed for this number of seconds since last successful call if there
   was no prior call the agent will be autopaused immediately.
 * Added member option ignorebusy this when set and ringinuse is not   
   will allow per member control of multiple calls as ringinuse does for
   the Queue.
  
 - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
 - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.

(closes issue ASTERISK-17421)
(closes issue ASTERISK-17391)
Reported by: irroot
Tested by: irroot, jrose
Review: https://reviewboard.asterisk.org/r/1119/


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  r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) | 3 lines
  
  Fix random misspelling noticed on asterisk-users.
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  r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) | 4 lines
  
  Fixes locking inversion caused by holding sip pvt lock during async_goto.
  
  (closes ASTERISK-17352)
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  r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) | 7 lines
  
  Use the device name and not the channel name to initialize the device state.
  
  Correct ASTERISK-11323 implementation as I don't see how it ever worked as
  claimed when it used the channel name and not the device name.
  
  (issue ASTERISK-11323)
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  r325152 | jrose | 2011-06-28 10:46:29 -0500 (Tue, 28 Jun 2011) | 5 lines
  
  Fixes moh reload breaking custom mode moh classes when the config file is untouched
  
  (closes issue ASTERISK-17730)
  Reported by: sdolloff
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If a SDP does not modify the session, we ignore it.  However, we were defaulting
no text and video support to true before checking to see if the sdp modified
anything or not.  This would result in process_sdp ignoring an sdp but removing
video and text from the call during direct media reinvites.


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  r325091 | lmadsen | 2011-06-28 10:12:00 -0500 (Tue, 28 Jun 2011) | 1 line
  
  Remove line from prep_tarball that kills mkrelease.
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  r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) | 5 lines
  
  Save and restore errno from within signal handlers.
  
  This is recommended by the POSIX standard, as well as by the sigaction(2) manpage
  for various platforms that we support (e.g. Mac OS X).
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  r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) | 21 lines
  
  When subscribing MWI to an unsolicited mailbox the first notification is incorrect.
  
  A remote peer subscribed to MWI with the unsolicited option and a local
  phone subscribed to the remote mailbox.  The notify message-summary events
  are sent correctly except for the first one when subscribing, which will
  always be 0.  This means the phone MWI indicator will be wrong until the
  mailbox read/unread count changes and the event is fired.
  
  Looks like this is a regression from ASTERISK-16149.
  
  * Fix the logic to check the cache and if allowed then fallback to
  manually counting mailbox messages.
  
  (closes issue ASTERISK-17997)
  Reported by: rsw686
  Patches:
        jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett
  Tested by: rsw686
  
  JIRA SWP-3551
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  r324849 | rmudgett | 2011-06-24 15:46:01 -0500 (Fri, 24 Jun 2011) | 15 lines
  
  Syntax errors in dialplan do not display the file name.
  
  When issuing the CLI command "dialplan reload" syntax errors and warnings
  are displayed on the console.  The offending line number is displayed on
  the console, but the file name is not displayed.  Errors caught in
  main/config.c do display the file name.
  
  (closes issue ASTERISK-17985)
  Reported by: ulogic
  Patches:
        pbx_config.patch uploaded by ulogic (License #5685) modified format
  Tested by: rmudgett
  
  JIRA SWP-3554
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  r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) | 11 lines
  
  DTMF wasn't being logged on connected consoles when enabled in logger.conf
  
  Previously in order for DTMF to be logged in a connected console session, the user would
  have to do logger set channel DTMF on.  This corrects that so that it is on by default.
  This issue was caused by an off by one error incurred by a logger level count of 6 in
  logger.h where it should have been 7.
  
  (closes issue: ASTERISK-17974)
  Reported by: Luke H
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There is no reason to clean up features twice.

Review: https://reviewboard.asterisk.org/r/1279/


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  r324678 | kmoore | 2011-06-23 13:29:17 -0500 (Thu, 23 Jun 2011) | 11 lines
  
  Merged revisions 324643 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) | 4 lines
    
    Addresses AST-2011-008, memory corruption and remote crash in SIP driver.
    
    AST-2011-008
  ........
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  r324685 | dvossel | 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) | 8 lines
  
  Fixes sip crash when calling remove_uri_parameters with NULL
  
  AST-2011-009
  
  (closes issue ASTERISK-18017)
  Reported by: jaredmauch
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  r324652 | dvossel | 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) | 20 lines
  
  Merged revisions 324634 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines
    
    Merged revisions 324627 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines
      
      Addresses AST-2011-010, remote crash in IAX2 driver
      
      Thanks to twilson for identifying the issue and providing the patches.
      
      AST-2011-010
    ........
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  r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22 Jun 2011) | 5 lines
  
  Remove tests for parsing address with invalid port
  
  getaddrinfo on OS X returns with EAI_NONAME error when passed a port
  greater than 65535. Linux throws no error, so remove the tests for now.
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  r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) | 1 line
  
  Use correct variable for text SRTP media.
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  r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines
  
  Stop sending IPv6 link-local scope-ids in SIP messages
  
  The idea behind the patch listed below was used, but in a more targeted manner.
  There are now address stringification functions for addresses that are meant to
  be sent to a remote party. Link-local scope-ids only make sense on the machine
  from which they originate and so are stripped in the new functions.
  
  There is also a host sanitization function added to chan_sip which is used
  for when peer and dialog tohost fields or sip_registry hostnames are used to
  craft a SIP message.
  
  Also added are some basic unit tests for netsock2 address parsing.
  
  (closes issue ASTERISK-17711)
  Reported by: ch_djalel
  Patches:
        asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)
  
  Review: https://reviewboard.asterisk.org/r/1278/
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Also fixed a reference leak in an error path in sip_msg_send().

........
  r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines

  Timout or error on INFO or MESSAGE transaction causes call to be lost.

  When exchanging INFO messages within a call, 4xx error causes the call to
  be disconnected although RFC 2976 explicitly states that such transactions
  do not modify the state of the dialog.

  When exchanging MESSAGE messages within a call, 4xx error causes the call
  to be disconnected.  To provide least surprise, we should not disconnect
  the call since a MESSAGE is like INFO in this case.  (Implied by RFC 3428
  Section 2)

  (closes issue ASTERISK-17901)
  Reported by: neutrino88

  Review: https://reviewboard.asterisk.org/r/1257/
  Review: https://reviewboard.asterisk.org/r/1258/

  JIRA SWP-3486
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  r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) | 1 line
  
  Comments and whitespace in chan_sip.c
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  r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) | 10 lines
  
  Fixes locking inversion issue in ast_async_goto()
  
  During this function we can not hold the "chan" lock while
  doing the masquerade, the explicit goto on the tmp chan, or
  the channel alloc.  Instead we need to get the channel lock,
  store off information about the channel that we need, and
  then let the channel lock go for the remainder of the function.
  
  Review: https://reviewboard.asterisk.org/r/1275/
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When playing back a prompt to a channel, confbridge neglects to check for
hangup events causing lockup condititions for hangups that occur before
actually joining the conference.  This change ensures that the user is removed
from the conference in the event of a premature hangup.

Review: https://reviewboard.asterisk.org/r/1277/


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  r324241 | lmadsen | 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011) | 2 lines
  
  Remove extra 'the'.
  Reported by Vlad Povorozniuc
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