https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines
  
  Make the structure of the table specified before match the queries and results.
  
  (closes issue #17557)
   Reported by: cmaj
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272568 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272558 65c4cc65-6c06-0410-ace0-fbb531ad65f3

RFC3261 states that Timer A should start at 500ms (T1) by default.
In chan_sip this value initially started at 1000ms and I changed
it to 500ms recently. After doing that I noticed in my packet
captures that it still occasionally retransmitted starting at
1000ms instead of 500ms like I told it to.  This occurs because
the scheduler runs in the do_monitor thread.  If a new retransmission
is added while the do_monitor thread is sleeping then it may not
detect that retransmission for nearly 1000ms.  To fix this I just
poke the do_monitor thread to wake up when a new packet is sent
reliably requiring retransmits.  The thread then detects the new
scheduler entry and adjusts its sleep time to account for it.

Review: https://reviewboard.asterisk.org/r/747



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272557 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272533 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines
  
  ss_thread calls pri_grab without lock during overlap dial
  
  Recent changes to chan_dahdi with relation to overlap dialing call
  pri_grab without first obtaining a lock.
  
  (closes issue #17414)
  Reported by: pdf
  Patches:
        bug17414.patch uploaded by jpeeler (license 325)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272447 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The external test suite stops Asterisk using the "core stop gracefully" command.
The logs from the tests show that there are a number of problems with Asterisk
trying to cleanly shut down.  This patch addresses the following type of error
that comes from chan_iax2:

[Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy:
                chan_iax2.c line 11371 (iax2_process_thread_cleanup):
                Error destroying mutex &thread->lock: Device or resource busy

For an example in the context of a build, see:

http://bamboo.asterisk.org/browse/AST-TRUNK-739/log

The primary purpose of this patch is to change the thread pool shutdown
procedure to be more explicit to ensure that the thread exits from a point
where it is not holding a lock.  While testing that, I encountered various
crashes due to the order of operations in unload_module() being problematic.
I reordered some things there, as well.

Review: https://reviewboard.asterisk.org/r/736/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272370 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

This version of the patch only adds AgentComplete for attended transfers.  It was already present for blind transfers.

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  r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines
  
  Send AgentComplete manager events in the event of blind and attended transfers.
  
  (closes issue #16819)
  Reported by: elbriga
  Patches:
        app_queue.diff uploaded by elbriga (license 482)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272368 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If there is realtime configuration, it does not get re-read on reload unless the config file also changes.

(closes issue #16982)
 Reported by: dmitri
 Patches: 
       res_musiconhold.patch uploaded by dmitri (license 1001)
 Tested by: atis


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272332 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17215)
 Reported by: vazir
 Patches: 
       20100518__issue17215.diff.txt uploaded by tilghman (license 14)
 Tested by: tilghman


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272260 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272259 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines
  
  First caller into a dynamic conference now enter pin once.
  
  If MeetMe is configured to use dynamic conference
  numbers, then the first caller (which creates the
  conference) had to enter the PIN number twice.
  
  (closes issue #15878)
  Reported by: shawkris
  Patches:
        issue15878.patch uploaded by pabelanger (license 224)
  Tested by: pabelanger
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272257 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272256 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #16991)
Reported by: pprindeville
Patches: 
      with_netsnmp.patch.txt uploaded by twilson (license 396)
Tested by: twilson

Review: https://reviewboard.asterisk.org/r/739/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272254 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17520)
Reported by: kobaz
Patches:
      manager.patch uploaded by kobaz (license 834)
Tested by: lmadsen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272252 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17548)
Reported by: cjacobsen
Patches:
      say.conf.sample.diff uploaded by cjacobsen (license 1029)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272243 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This command lets you request a "/n" local channel
optimize itself out of the way anyway.

Review: https://reviewboard.asterisk.org/r/732/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272218 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272150 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272147 | tilghman | 2010-06-23 13:40:28 -0500 (Wed, 23 Jun 2010) | 5 lines
  
  Backport part of revision 136715 to fix callerid in voicemail text files (IMAP only).
  
  (closes issue #16945)
   Reported by: mneuhauser
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272148 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272146 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17144)
 Reported by: nahuelgreco
 Patches: 
       20100513__issue17144__trunk.diff.txt uploaded by tilghman (license 14)
 Tested by: tilghman


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272145 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Even if there are no stations or trunks defined, we need to start the sla
thread to make sure we get the reload event. Also, when doing a reload we need
to remove the existing trunks and stations or they end up hanging around.

(closes issue #16818)
Reported by: mbonin
Patches: 
      sla_reload.patch uploaded by twilson (license 396)
Tested by: twilson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272109 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Testing proved that if Asterisk sent a connected line reinvite, and
the endpoint to which the reinvite were being sent sent a reinvite, Asterisk
would not properly respond with a 491 response.

The reason is that on connected line reinvites, we set the dialog's invitestate
to INV_CALLING to prevent Asterisk from sending a rapid flurry of connected line
reinvites. For other reinvites we do not do this. Because of the current invitestate,
when Asterisk received the reinvite, we interpreted this as a spiraled INVITE, and thus
did not behave properly.

The fix for this is to not enter the loop detection or spiral logic in handle_request_invite
if the channel state is currently up. This way, no mid-call reinvites will be misinterpreted,
no matter what the nature of the reinvite may have been.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This small changes prevents destroy_all_channels() from accessing a lock on an
unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when
shutting Asterisk down gracefully.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272052 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17440)
Reported by: kobaz


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272014 65c4cc65-6c06-0410-ace0-fbb531ad65f3

RFC 3261 section 9 states that a CANCEL has no effect on a
request to a UAS that has already given a final response.  This
patch checks to make sure there is a pending invite before
allowing a CANCEL request to be processed, otherwise it responds
to the CANCEL with a "481 Call/Transaction Does Not Exist".

Review: https://reviewboard.asterisk.org/r/697/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271977 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes a ref count leak in event filters and checks for
a filter container allocation failure during session creation.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271905 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines
  
  Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set.  This is necessary to keep the ref count correct.
  
  (closes issue #16815)
  Reported by: rain
  Patches:
        chan_sip-unref-fix.diff uploaded by rain (license 327) (modified)
  Tested by: rain
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271903 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch as documented in the sample config allows one to optionally apply
white, black, or both types of filtering to manager events. The new
'eventfilter' option is set per user.

(closes issue #14861)
Reported by: fnordian
Patches: 
      eventfilter3.patch uploaded by fnordian (license 110),
      modified by me

Review: https://reviewboard.asterisk.org/r/673/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Don't Finalize() if Initialize() did not succeed.  This resulted in an error
about trying to Finalize() an invalid handle.

Also trim some trailing whitespace while in the area.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271867 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Using this method makes it so res_fax doesn't have to be rebuilt on every
svn update.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271833 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If both the transferer and transferee of a attended transfer hangup before
the new channel picks up, the new channel should be hung up as well as it
has no endpoint to talk to.  This mirrors the expected behavior used in 1.4. 

(closes issue #17444)
Reported by: corruptor



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271831 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines
  
  Allow users to specify a port for dundi peers.
  
  (closes issue #17056)
  Reported by: klaus3000
  Patches:
        dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
  Tested by: klaus3000
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271762 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines
  
  Modify chan_sip's packet generation api to automatically calculate the Content-Length.  This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated.  This change was made to ensure that the Content-Length is always correct.
  
  (closes issue #17326)
  Reported by: kenner
  Tested by: mnicholson, kenner
  
  Review: https://reviewboard.asterisk.org/r/693/
........


This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271690 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271657 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17534)
Reported by: fabled
Patches:
      speex-wb-sample.diff uploaded by fabled (license 448)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271625 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) | 7 lines
  
  Do not use sizeof to calculate size of a heap allocated character array.
  
  Change left out from 271399.
  
  (closes issue #16053)
  Reported by: diLLec
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271554 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17437)
Reported by: klaus3000
Patches:
      sip_crash uploaded by dvossel (license 671)
Tested by: klaus3000



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271553 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #16869)
 Reported by: chappell
 Patches: 
       app_say_counted-20100317.c uploaded by chappell (license 8)
 Tested by: chappell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271520 65c4cc65-6c06-0410-ace0-fbb531ad65f3