(closes issue #16293)
Reported by: malcolmd
Patches:
      g719.passthrough.patch.7 uploaded by malcolmd (license 924)
      format_g719.c uploaded by malcolmd (license 924)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270940 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Per Tilghman's request on IRC (#asterisk-bugs).

(closes issue #17506)
Reported by: brycebaril
Tested by: pabelanger, tilghman


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270936 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines
  
  fixes chan_iax2 race condition
  
  There is code in chan_iax2.c that attempts to guarantee that only a single
  active thread will handle a call number at a time.  This code works once
  the thread is added to an active_list of threads, but we are not currently
  guaranteed that a newly activated thread will enter the active_list immediately
  because it is left up to the thread to add itself after frames have been
  queued to it.  This means that if two frames come in for the same call number
  at the same time, it is possible for them to grab two separate threads because
  the first thread did not add itself to the active_list fast enough.  This
  causes some pretty complex problems.
  
  This patch resolves this race condition by immediately adding an activated
  thread to the active_list within the network thread and only depending on
  the thread to remove itself once it is done processing the frames queued to
  it.  By doing this we are guaranteed that if another frame for the same call
  number comes in at the same time, that this thread will immediately be found
  in the active_list of threads.
  
  Review: https://reviewboard.asterisk.org/r/720/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270867 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Clearing the callwaitcas flag in analog_call was causing the incoming D digit
to be ignored which triggers sending the caller ID.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270836 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17511)
Reported by: klaus3000
Patches:
      channelvariables.tex-patch.txt uploaded by klaus3000 (license 65)
Tested by: pabelanger



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270801 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If there is a problem with a firmware file, Polycom phones will close the
connection. We were continuing to send the file anyway. There should be no
reason to continue sending a file if there is an error writing it.

(closes issue #16682)
Reported by: lmadsen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270692 65c4cc65-6c06-0410-ace0-fbb531ad65f3

After the manager http auth changes, we forgot to remove the manual
sending of the file. Also, ast_http_send adds two \r\n to the header that
is passed to it, so a trailing \r\n is removed from the Content-type
header. It might be better to change ast_http_send, but I don't like changing
the behavior of an API function.

(closes issue #17239)
Reported by: cjacobsen
Patches: 
      patch2.diff uploaded by cjacobsen (license 1029)
Tested by: lathama, cjacobsen



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270660 65c4cc65-6c06-0410-ace0-fbb531ad65f3

chan_sip's "contactdeny" feature screens the "to be registered contact".
In case of nat=yes it should not use the address information from the
Contact header (which is not used at all for routing), but the source
IP address of the request.

Thus, if nat=yes and a client sends a request from a denied IP address
(e.g. by spoofing the src-IP address) it can bypass the screening.

This commit makes contactdeny apply to the src ip when nat=yes instead.

(closes issue #17276)
Reported by: klaus3000
Patches: 
      patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270658 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010) | 5 lines
  
  Variables have always been case-sensitive, so we should not be removing case-insensitive matches.
  
  Bug reported via the -dev list.  See
  http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270584 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270552 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15757)
 Reported by: Marquis
 Patches: 
       distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
 Tested by: Marquis, lmadsen, marcelloceschia
 
Review: https://reviewboard.asterisk.org/r/351/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010) | 1 line
  
  Move information about zonemessages into the [zonemessages] section.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270443 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon, 14 Jun 2010) | 14 lines
  
  Properly play first file in sort list.
  
  When using sort=alpha we would always skip the first file
  in the list first time through.  We now check for that
  properly. 
  
  (closes issue #17470)
  Reported by: pabelanger
  Patches:
        sort.aplha.patch uploaded by pabelanger (license 224)
  Tested by: lmadsen
  
  Review: https://reviewboard.asterisk.org/r/703/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270332 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also found a place where sig_pri_init_pri() was inlined and called it
instead.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270298 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The "channel" option would chop the channel name at the last '-', which made
it useless for something like a channel transfer from the dialplan.  The
"fullchannel" option will return the channel name as-is.

ABE-2218


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270260 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn.

Review:	https://reviewboard.asterisk.org/r/696/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Theoretically the ./configure script is a pure bourne-shell script.
Practically it may be run by bash if /bin/sh is not good enough. But we should not count on it. See bug report for the gory details.

(closes issue #17485)
Patches:
      0001-remove-bashism-from-ast_check_pwlib.m4.patch uploaded by tzafrir (license 46)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270184 65c4cc65-6c06-0410-ace0-fbb531ad65f3

FreeBSD / OSX builds.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270151 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun 2010) | 2 lines
  
  Fix typo in example
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270079 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This way the libraries can be found even if they are in
non-standard locations. 

(closes issue #16155)
Reported by: jcollie
Patches:
      0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch uploaded by jcollie (license 412)
Tested by: jsmith, tilghman, pabelanger


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270042 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010) | 8 lines
  
  For SpeeX, 0 bits remaining is valid and does not need an emitted warning.
  
  (closes issue #15762)
   Reported by: nblasgen
   Patches: 
         issue15672.patch uploaded by pabelanger (license 224)
   Tested by: nblasgen
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269976 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #16965)
 Reported by: rrb3942
 Patches: 
       DBGetComplete.patch uploaded by rrb3942 (license 1003)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269938 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269936 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17031)
Reported by: pabelanger
Patches:
      Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224)
Tested by: pabelanger, tilghman


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269889 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun 2010) | 19 lines
  
  Fix potential crash when writing raw SLIN audio on a PLC-enabled channel.
  
  The issue here was that the frame created when adjusting for PLC had no offset
  to its audio data. If this frame were translated to another format prior to
  being sent out an RTP socket, all went well because the translation code would
  put an appropriate offset into the frame. However, if the SLIN audio were not
  translated before being sent out the RTP socket, bad things would happen.
  Specifically, the ast_rtp_raw_write makes the assumption that the frame has
  at least enough of an offset that it can accommodate an RTP header. This was
  not the case. As such, data was being written prior to the allocation, likely
  corrupting the data the memory allocator had written. Thus when the time came
  to free the data, all hell broke loose. ....Well, Asterisk crashed at least.
  
  The fix was just what one would expect. Offset the data in the frame by a reasonable
  amount. The method I used is a bit odd since the data in the frame is 16 bit integers
  and not bytes. I left a big ol' comment about it. This can be improved on if someone
  is interested. I was more interested in getting the crash resolved.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269822 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/688/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269749 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269711 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010) | 9 lines
  
  Ensure restartable system calls can restart (BSD signal semantics).
  
  This eliminates the annoying <beep> on the console.
  
  (closes issue #17477)
   Reported by: jvandal
   Patches: 
         20100610__issue17477.diff.txt uploaded by tilghman (license 14)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269602 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269569 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010) | 2 lines
  
  Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it).
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269497 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r269426 | qwell | 2010-06-09 16:19:17 -0500 (Wed, 09 Jun 2010) | 6 lines
  
  Let systems without a working fork() use res_musiconhold.
  
  Files mode doesn't require anything special, so that can still be used just fine.
  
  AST-357
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269486 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Valgrind pointed out that attempting to get an IE value from an event that has
no IEs produces an invalid memory read past the end of the event.  Thanks to
mmichelson for pointing the problem out to me and then testing the fix.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269417 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun 2010) | 12 lines
  
  Fix Debian init script to not use -c.
  
  When using the init script as-is currently, it could cause issues on Debian
  such as high CPU usage. This fix has worked for several people so I'm
  implementing the change.  We now handle color displays properly.
  
  (closes issue #16784)
  Reported by: pabelanger
  Patches:
        20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
  Tested by: pabelanger, tilghman
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269346 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269308 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are
in a potentially inconsistent state is a potentially bad thing in
principle.

However, calling dahdi_indicate() when the channel private lock is already
held can cause a deadlock if the PRI lock is needed because
dahdi_indicate() will also get the channel private lock.  The pri_grab()
function assumes that the channel private lock is held once to avoid
deadlock.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269307 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The result for moh_register was not verified to guarantee
the mohclass as added to the container.


(closes issue #16993)
Reported by: dmitri
Patches:
      res_musiconhold_rtclass2.patch uploaded by dmitri (license 1001)
      moh_crash2.diff uploaded by dvossel (license 671)
Tested by: dmitri




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269271 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* chan_dahdi supports dialing configuring and dialing by device file name.
  DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
  it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
* A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
  False by default. If set, chan_dahdi will ignore failed 'channel' entries.
  Handy for the above name-based syntax as it does not depend on
  initialization order.
* have my_pri_make_cc_dialstring() only manupulate dial-strings of group
  (gGrR) dialing, which make it lsightly more complicated.

https://reviewboard.asterisk.org/r/535/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269238 65c4cc65-6c06-0410-ace0-fbb531ad65f3