https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines
  
  Remove potential deadlock in call pickup race.
  
  Deadlock is possible in ast_do_pickup() when holding the target channel
  lock and trying to get the chan channel lock.  Also, holding the target
  lock when calling ast_channel_masquerade() is not a good idea because that
  routine does deadlock avoidance.
  
  * Removed the need to hold the target lock after marking the target with a
  datastore and getting the connected line data off of the target channel.
  
  * Moved can_pickup() to ast_can_pickup() in features.c.  Now all the call
  pickup methods use the same basic call pickup availability check.
  
  Review: https://reviewboard.asterisk.org/r/1234/
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  r322585 | jrose | 2011-06-09 09:06:42 -0500 (Thu, 09 Jun 2011) | 11 lines
  
  Adds ast_escape_encoded utility to properly handle escaping of quoted field before uri.
  
  This commit backports a feature in trunk affecting initreqprep so that display name won't
  be encoded improperly. Also includes unit tests for the ast_escape_quoted function.
  This patch gives 1.8 a much improved outlook in countries which don't use standard
  ASCII characters.
  
  (closes issue ASTERISK-16949)
  Reported by: Örn Arnarson
  Review: https://reviewboard.asterisk.org/r/1235/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322586 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER.
Initial value must be the time to autoanswer in ms, then optionally :BEEP
to play a tone when answered and :MUTE to mute the mic when answering. 
eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and 
play a beep. just 3000 would answer afer 3 secs of ringing with no 
beep and full two way audio. 



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  r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) | 15 lines
  
  Ring all queue with more than 255 agents will cause crash.
  
  1. Create a ring-all queue with 500 permanent agents.
  2. Call it.
  3. Asterisk will crash.
  
  The watchers array in app_queue.c has a hard limit of 255.  Bounds
  checking is not done on this array.  No sane person should put 255 people
  in a ring-all queue, but we should not crash anyway.
  
  * Added bounds checking to the watchers array.
  
  JIRA AST-464
  JIRA SWP-2903
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  r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) | 16 lines
  
  SRV lookup attempted for SIP peers listed as an IP address.
  
  Asterisk attempts to SRV lookup a host name even if the host name is an IP
  address.  Regression introduced when IPv6 support was added.
  
  * Restored the check in ast_dnsmgr_lookup() to see if the given host name
  is an IP address.  The IP address could be in either IPv4 or IPv6 formats.
  
  (closes issue ASTERISK-17815)
  Reported by: Byron Clark
  Tested by: Byron Clark, Richard Mudgett
  Patches:
       issue19248_v1.8.patch - uploaded by Richard Mudgett (License #5621)
  
  Review: https://reviewboard.asterisk.org/r/1240/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322426 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything.

Review: https://reviewboard.asterisk.org/r/1256/


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  r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines
  
    Make handle_request_publish do dialog expiration and destruction.
  
    This patch fixes handle_request_publish so that it does dialog expiration and destruction.
  
    Without this patch the incoming PUBLISH requests will get stuck in the dialog list.
    Restarting asterisk is the only way to remove them.
  
    Personal observation on one system the server hung up while looping through the channels
    rendering asterisk unusable and all sip phones unregisterd when they try reregister
    more requests are added.
  
    (closes issue #18898)
    Reported by: gareth
    Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot
  
    Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915
    Review: https://reviewboard.asterisk.org/r/1253
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(closes issue ASTERISK-17978)
Reported by: elguero
Patches:
    stop_messages_going_to_dialplan.patch (license #5026)


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  r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun 2011) | 4 lines
  
  Use correct syntax for 'sip notify snom-reboot'
  
  (closes issue ASTERISK-17915)
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Review: https://reviewboard.asterisk.org/r/1172



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  r322069 | jrose | 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) | 8 lines
  
  Fixes level toggling for logger set levels since it was reversed
   
  (closes issue ASTERISK-17850)
  Reported by: Luke H
  Tested by: jrose, Luke H
    
  Review: https://reviewboard.asterisk.org/r/1244/
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  r321926 | rmudgett | 2011-06-03 17:09:36 -0500 (Fri, 03 Jun 2011) | 18 lines
  
  Asterisk crash when unloading cdr_radius/cel_radius.
  
  The rc_openlog() API call is passed a string that is used by openlog() to
  format log messages.  The openlog() does not copy the string it just keeps
  a pointer to it.  When the module is unloaded, the string is gone from
  memory.  Depending upon module load order and if the other module then has
  an error, a crash happens.
  
  * Pass rc_openlog() a strdup'd string with the understanding that there
  will be a small memory leak if the cdr_radius/cel_radius modules are
  unloaded.
  
  * Call rc_destroy() to free the rc handle memory when the module is
  unloaded.
  
  JIRA AST-483
  JIRA SWP-3062
........


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  r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) | 5 lines
  
  Be more explicit for CCSS generic device state event subscription.
  
  Make CCSS generic device state event subscription specify the
  AST_EVENT_IE_STATE ie exists to be safe.
........


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  r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) | 27 lines
  
  Event subscription fixes.
  
  Must commit the subscription fixes together with the integration
  subscription tests.  The subscription fixes cause an erroneously passing
  test to fail.  The new subscription tests detect errors without the
  subscription fixes.
  
  * Added missing event_names[] table entry.
  
  * Reworked ast_event_check_subscriber()/match_sub_ie_val_to_event() to
  correctly detect if a subscriber exists for the proposed event.
  
  * Made match_ie_val() and match_sub_ie_val_to_event() check the buffer
  length for RAW payload types.
  
  * Fixed error handling memory leak in ast_event_sub_activate(),
  ast_event_unsubscribe(), and ast_event_queue().
  
  * Made ast_event_new() and ast_event_check_subscriber() better protect
  themselves from an invalid payload type.
  
  * Added container lock protection between removing old cache events and
  adding the new cached event in
  ast_event_queue_and_cache()/event_update_cache().
  
  * Added new event subscription tests.
........


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  r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) | 1 line
  
  Correct IAX2 and SIP event subscription description string.
........
  r321813 | rmudgett | 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line
  
  Constify subscription description parameter string.
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  r321753 | russell | 2011-06-03 13:32:45 -0500 (Fri, 03 Jun 2011) | 2 lines
  
  Backport an astobj2 unit test so that it runs on 1.8 as well.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321754 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1254/


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  r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) | 5 lines
  
  Also document the 'queue-minute' option.
  
  (closes issue #19386)
  Reported by: juanmol
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Don't send all messages to 's'.  Get the destination from the request URI.
(Found using automated test cases).


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  r321547 | rmudgett | 2011-06-01 18:11:55 -0500 (Wed, 01 Jun 2011) | 1 line
  
  CDR comment tweaks.
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Asterisk now has protocol independent support for processing text messages
outside of a call.  Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported.  There are options in sip.conf
and jabber.conf that enable these features.

There is a new application, MessageSend().  There are two new functions,
MESSAGE() and MESSAGE_DATA().  Documentation will be available on
the project wiki, wiki.asterisk.org.

Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.

Review: https://reviewboard.asterisk.org/r/1042/


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  r321537 | bbryant | 2011-06-01 16:10:02 -0400 (Wed, 01 Jun 2011) | 8 lines
  
  This patch fixes an issue with using the wrong voicemail folders with greetings.
  
  (closes issue #17871)
  Reported by: edhorton
  Patches: 
        digium_bug_17871_2 uploaded by fhackenberger (license 592)
  Tested by: edhorton, fhackenberger
........


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  r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines
  
  Fix double alerting, add forced alerting before answer
  
  Fix double alerting (it wasn't fixed here by issue #18542)
  Add forced alerting before connect (if it wasn't before)
  Try to send all packets from outgoing queue rather than one only
  Call goes into clearing state when disconnect command is received
  
  (closes issue #19361)
  Reported by: vmikhelson
  Patches: 
        issue19361-3.patch uploaded by may213 (license 454)
  Tested by: vmikhelson
........


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  r321517 | rmudgett | 2011-05-31 15:54:35 -0500 (Tue, 31 May 2011) | 1 line
  
  Update some comments.
........


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  r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31 May 2011) | 12 lines
  
  Chan_local locking cleanup.
  
  This patch removes all of the unnecessary deadlock
  avoidance loops that occur in chan_local.  It also
  resolves an issue with a deadlock triggered by
  local channel optimizations.
  
  (issue #18028)
  
  Review: https://reviewboard.asterisk.org/r/1231/
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  r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) | 8 lines
  
  Enhance NOTICE message to know who couldn't access the dialplan.
  
  (closes issue #19390)
  Reported by: lmadsen
  Patches: 
        __20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10)
  Tested by: russell
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  r321436 | rmudgett | 2011-05-27 19:27:52 -0500 (Fri, 27 May 2011) | 4 lines
  
  Some hagi launch cleanup.
  
  Inspired by issue 19256.  This patch would also fix the crash.
........


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  r321392 | rmudgett | 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011) | 12 lines

  Crash when using hagi and no servers are available.

  When none of the servers returned by the SRV querey respond, asterisk
  crashes.  The problem is that if the loop over all the SRV entries
  finishes then the srv_context has already been cleaned up.

  * Make ast_srv_cleanup() check to see if the context is already cleaned
  up.

  (closes issue #19256)
  Reported by: byronclark
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Also revert -r321331 and -r321332.

........
  r321337 | rmudgett | 2011-05-27 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines
  
  The app_privacy args have undocumented "options" position, interferes with "context" position.
  
  * Add documention for unused "options" position to match existing code.
  
  (closes issue #19273)
  Reported by: mdavenport
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  r321335 | lmadsen | 2011-05-27 17:54:54 -0400 (Fri, 27 May 2011) | 7 lines
  
  Fix issue with playback of H.261 video.
  
  (closes issue #19379)
  Reported by: neutrino88
  Patches:
        videoprompt.patch uploaded by neutrino88 (license 297)
  (changes by russell)
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  r321333 | lmadsen | 2011-05-27 17:40:23 -0400 (Fri, 27 May 2011) | 7 lines
  
  Allow parking lot hints and musicclass to be set.
  
  (closes issue #19378)
  Reported by: sboily_proformatique
  Patches:
        pf_parkinghint_music_fix uploaded by sboily proformatique (license 206)
  Tested by: russell
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  r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines
  
  The app_privacy args have undocumented "options" position, interferes with "context" position.
  
  * Add documention for unused "options" position to match existing code.
  The trunk(v1.10) version will remove the unused options position.
  
  (closes issue #19273)
  Reported by: mdavenport
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  r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) | 3 lines
  
  markm committed a patch I was working on yesterday, this fixes it to mesh up with suggestions by mnicholson.
........


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  r321211 | alecdavis | 2011-05-27 20:31:15 +1200 (Fri, 27 May 2011) | 16 lines
  
  Fix *8 directed pickup locks system during pickupsound play out
  
  move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method,
  This stop the clash of 2 threads trying to write audio to same channel.
  In addition fixes choppy audio beep in issue 19177.
   
   (issue #18654)
   (issue #19177)
   Reported by: Docent
   Patches: 
        review1232-1.8.diff.txt alecdavis (license 585)
   Tested by: alecdavis
   
  Review: https://reviewboard.asterisk.org/r/1232/
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  r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | 10 lines
  
  Fixed build problem with dev mode enabled, which was caused by commit 321100.  Reformulated patch to be more generic.
  
  Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c.  This will ensure that any use of parse uri will have null output variables if the parse fails.
  
  (closes issue #19346)
  Reported by: kobaz
  Tested by: kobaz,JonathanRose
  
  Review: [full review board URL with trailing slash]
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  r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | 11 lines
  
  ast_sockaddr_resolve() in netsock2.c may deref a null pointer
  
  Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables
  
  (closes issue #19346)
  Reported by: kobaz
  Patches: 
        netsock2.patch uploaded by kobaz (license 834)
  Tested by: kobaz, Marquis
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  r321044 | rmudgett | 2011-05-26 13:10:17 -0500 (Thu, 26 May 2011) | 1 line
  
  Update ast_sockaddr comment with an important note.
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