[svn-commits] lmadsen: tag 1.8.1-rc1 r295161 - /tags/1.8.1-rc1/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Tue Nov 16 12:49:40 CST 2010


Author: lmadsen
Date: Tue Nov 16 12:49:36 2010
New Revision: 295161

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=295161
Log:
Importing files for 1.8.1-rc1 release.

Added:
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    tags/1.8.1-rc1/.version   (with props)
    tags/1.8.1-rc1/ChangeLog   (with props)

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+2010-11-16  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.8.1-rc1 Released.
+
+2010-11-15 18:30 +0000 [r294989-295078]  Tilghman Lesher <tlesher at digium.com>
+
+	* tests/test_expr.c (added), /: Merged revisions 295062 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r295062 | tilghman | 2010-11-15 12:24:02 -0600
+	  (Mon, 15 Nov 2010) | 9 lines Merged revisions 295026 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15
+	  Nov 2010) | 2 lines Create test verifying results of expression
+	  parser ........ ................
+
+	* funcs/func_curl.c, /: Merged revisions 294988 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010)
+	  | 8 lines It is possible to crash Asterisk by feeding the curl
+	  engine invalid data. (closes issue #18161) Reported by: wdoekes
+	  Patches: 20101029__issue18161.diff.txt uploaded by tilghman
+	  (license 14) Tested by: tilghman ........
+
+2010-11-12 21:14 +0000 [r294905-294911]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 294910 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12
+	  Nov 2010) | 4 lines Return correct error code if lock path fails.
+	  The recent changes to open_mailbox actually caused it to be
+	  fixed, but let's be consistent. Reported by alecdavis in
+	  asterisk-dev. ........
+
+	* apps/app_voicemail.c, /: Merged revisions 294904 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600
+	  (Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010)
+	  | 16 lines Fix regression causing abort in voicemail after
+	  opening a mailbox with no mesgs. In order to be more safe, some
+	  error handling code was changed to respect more error conditions
+	  including the potential memory allocation failure for deleted and
+	  heard message tracking introduced in 293004. However,
+	  last_message_index returns -1 for zero messages (perhaps as
+	  expected) and was triggering the stricter error checking. Because
+	  last_message_index is only called directly in one place, just
+	  return 0 from open_mailbox (for file based storage) when no
+	  messages are detected unless a real error has occurred. (closes
+	  issue #18240) Reported by: leobrown Patches:
+	  bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
+	  Tested by: pabelanger ........ ................
+
+2010-11-12 02:45 +0000 [r294823]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c, channels/sig_pri.h, /: Merged revisions
+	  294822 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600
+	  (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010)
+	  | 11 lines Asterisk is getting a "No D-channels available!"
+	  warning message every 4 seconds. Asterisk is just whining too
+	  much with this message: "No D-channels available! Using Primary
+	  channel XXX as D-channel anyway!". Filtered the message so it
+	  only comes out once if there is no D channel available without an
+	  intervening D channel available period. (closes issue #17270)
+	  Reported by: jmls ........ ................
+
+2010-11-11 22:17 +0000 [r294740-294745]  Russell Bryant <russell at digium.com>
+
+	* doc/CCSS_architecture.pdf (removed): Remove CCSS architecture
+	  PDF. It has been moved to:
+	  https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture
+
+	* doc/digium-mib.txt (removed), doc/followme.txt (removed),
+	  doc/building_queues.txt (removed), doc/timing.txt (removed),
+	  doc/advice_of_charge.txt (removed), doc/unistim.txt (removed),
+	  doc/video_console.txt (removed), doc/macroexclusive.txt
+	  (removed), doc/google-soc2009-ideas.txt (removed), doc/README.txt
+	  (added), doc/callfiles.txt (removed), doc/externalivr.txt
+	  (removed), doc/codec-64bit.txt (removed),
+	  build_tools/prep_tarball, doc/video.txt (removed), doc/jingle.txt
+	  (removed), doc/modules.txt (removed), doc/manager_1_1.txt
+	  (removed), doc/PEERING (removed), doc/snmp.txt (removed),
+	  doc/siptls.txt (removed), doc/HOWTO_collect_debug_information.txt
+	  (removed), doc/ldap.txt (removed), doc/sip-retransmit.txt
+	  (removed), doc/distributed_devstate.txt (removed),
+	  doc/voicemail_odbc_postgresql.txt (removed), doc/tex (removed),
+	  doc/queue.txt (removed), doc/jabber.txt (removed),
+	  doc/chan_sip-perf-testing.txt (removed), Makefile,
+	  doc/asterisk-mib.txt (removed), doc/database_transactions.txt
+	  (removed), doc/smdi.txt (removed), doc/janitor-projects.txt
+	  (removed), doc/hoard.txt (removed), doc/res_config_sqlite.txt
+	  (removed), doc/osp.txt (removed), doc/speechrec.txt (removed),
+	  doc/sms.txt (removed), doc/distributed_devstate-XMPP.txt
+	  (removed), doc/valgrind.txt (removed), doc/realtimetext.txt
+	  (removed), doc/cli.txt (removed), doc/rtp-packetization.txt
+	  (removed), doc/datastores.txt (removed), doc/CODING-GUIDELINES
+	  (removed), doc/ss7.txt (removed), doc/backtrace.txt (removed),
+	  doc/India-CID.txt (removed): Remove most of the contents of the
+	  doc dir in favor of the wiki content. This merge does the
+	  following things: * Removes most of the contents from the doc/
+	  directory in favor of the wiki - http://wiki.asterisk.org/ *
+	  Updates the build_tools/prep_tarball script to know how to export
+	  the contents of the wiki in both PDF and plain text formats so
+	  that the documentation is still included in Asterisk release
+	  tarballs.
+
+2010-11-11 21:58 +0000 [r294640-294734]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 294733 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600
+	  (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
+	  | 18 lines Fix problem with qualify option packets for realtime
+	  peers never stopping. The option packets not only never stopped,
+	  but if a realtime peer was not in the peer list multiple options
+	  dialogs could accumulate over time. This scenario has the
+	  potential to progress to the point of saturating a link just from
+	  options packets. The fix was to ensure that the poke scheduler
+	  checks to see if a peer is in the peer list before continuing to
+	  poke. The reason a peer must be in the peer list to be able to
+	  properly manage an options dialog is because otherwise the call
+	  pointer is lost when the peer is regenerated from the database,
+	  which is how existing qualify dialogs are detected. (closes issue
+	  #16382) (closes issue #17779) Reported by: lftsy Patches:
+	  bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
+	  zerohalo ........ ................
+
+	* main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
+	  revisions 294639 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r294639 | jpeeler | 2010-11-11 13:31:00 -0600
+	  (Thu, 11 Nov 2010) | 53 lines Merged revisions 294384 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
+	  | 47 lines Fix a deadlock in device state change processing.
+	  Copied from some notes from the original author (Russell):
+	  Deadlock scenario: Thread 1: device state change thread Holds -
+	  rdlock on contexts Holds - hints lock Waiting on channels
+	  container lock Thread 2: SIP monitor thread Holds the "iflock"
+	  Holds a sip_pvt lock Holds channel container lock Waiting for a
+	  channel lock Thread 3: A channel thread (chan_local in this case)
+	  Holds 2 channel locks acquired within app_dial Holds a 3rd
+	  channel lock it got inside of chan_local Holds a local_pvt lock
+	  Waiting on a rdlock of the contexts lock A bunch of other threads
+	  waiting on a wrlock of the contexts lock To address this
+	  deadlock, some locking order rules must be put in place and
+	  enforced. Existing relevant rules: 1) channel lock before a pvt
+	  lock 2) contexts lock before hints lock 3) channels container
+	  before a channel What's missing is some enforcement of the order
+	  when you involve more than any two. To fix this problem, I put in
+	  some code that ensures that (at least in the code paths involved
+	  in this bug) the locks in (3) come before the locks in (2). To
+	  change the operation of thread 1 to comply, I converted the
+	  storage of hints to an astobj2 container. This allows processing
+	  of hints without holding the hints container lock. So, in the
+	  code path that led to thread 1's state, it no longer holds either
+	  the contexts or hints lock while it attempts to lock the channels
+	  container. (closes issue #18165) Reported by: antonio ABE-2583
+	  ........ ................
+
+2010-11-10 23:26 +0000 [r294569-294605]  Tilghman Lesher <tlesher at digium.com>
+
+	* pbx/pbx_spool.c: Fixing the Mac OS X build (bamboo warning)
+
+	* pbx/pbx_spool.c: Properly queue files with inotify(7). (closes
+	  issue #18089) Reported by: abelbeck Patches:
+	  20101021__issue18089.diff.txt uploaded by tilghman (license 14)
+	  Tested by: tilghman
+
+2010-11-10 14:14 +0000 [r294501-294535]  Russell Bryant <russell at digium.com>
+
+	* UPGRADE.txt, res/ais/clm.c, res/ais/evt.c: Tweak a couple of CLI
+	  commands back to their original form. The "module" in this case
+	  is two parts, so there are two words before the verb of the CLI
+	  command.
+
+	* main/devicestate.c, /: Merged revisions 294500 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010)
+	  | 7 lines Improve a debug message to be more readable and
+	  consistent. (closes issue #18282) Reported by: klaus3000 Patches:
+	  ast_devstate2str-patch.txt uploaded by klaus3000 (license 65)
+	  ........
+
+2010-11-09 22:46 +0000 [r294466]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c: Allow ast_do_masquerade() failure to be reported
+	  again.
+
+2010-11-09 20:33 +0000 [r294430]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+	  Merged revisions 294429 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010)
+	  | 8 lines Detect GMime properly on systems where gmime flags and
+	  libs are configured with pkg-config. (closes issue #16155)
+	  Reported by: jcollie Patches: 20100917__issue16155.diff.txt
+	  uploaded by tilghman (license 14) Tested by: tilghman ........
+
+2010-11-09 16:55 +0000 [r294349]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/channel.h, channels/sig_pri.c, main/channel.c,
+	  channels/chan_misdn.c, channels/sig_analog.c: Analog lines do not
+	  transfer CONNECTED LINE or execute the interception macros. Add
+	  connected line update for sig_analog transfers and simplify the
+	  corresponding sig_pri and chan_misdn transfer code. Note that if
+	  you create a three-way call in sig_analog before transferring the
+	  call, the distinction of the caller/callee interception macros
+	  make little sense. The interception macro writer needs to be
+	  prepared for either caller/callee macro to be executed. The
+	  current implementation swaps which caller/callee interception
+	  macro is executed after a three-way call is created. Review:
+	  https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA
+	  SWP-2372
+
+2010-11-08 22:32 +0000 [r294278-294313]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, res/res_timing_timerfd.c: Merged revisions 294312 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08
+	  Nov 2010) | 1 line add missing unlock not present in 294277
+	  ........
+
+	* include/asterisk/timing.h, main/timing.c, main/channel.c, /,
+	  res/res_timing_timerfd.c: Merged revisions 294277 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08
+	  Nov 2010) | 16 lines Fix playback failure when using IAX with the
+	  timerfd module. To fix this issue the alert pipe will now be used
+	  when the timerfd module is in use. There appeared to be a race
+	  that was not solved by adding locking in the timerfd module, but
+	  needed to be there anyway. The race was between the timer being
+	  put in non-continuous mode in ast_read on the channel thread and
+	  the IAX frame scheduler queuing a frame which would enable
+	  continuous mode before the non-continuous mode event was read.
+	  This race for now is simply avoided. (closes issue #18110)
+	  Reported by: tpanton Tested by: tpanton I put tested by tpanton
+	  because it was tested on his hardware. Thanks for the remote
+	  access to debug this issue! ........
+
+2010-11-08 20:56 +0000 [r294243]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 294242 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov
+	  2010) | 8 lines Go off hold when we get an empty reinvite telling
+	  us to. (closes issue 0014448) Reported by: frawd (closes issue
+	  #17878) Reported by: frawd ........
+
+2010-11-08 19:56 +0000 [r294207]  Terry Wilson <twilson at digium.com>
+
+	* configs/calendar.conf.sample, res/res_calendar.c: Set a default
+	  waittime, and make sure to convert it to milliseconds
+
+2010-11-08 17:16 +0000 [r294125]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_misdn.c: valgrind reported references to freed
+	  memory during a mISDN hangup collision. Bad things have been
+	  happening in chan_misdn because the chan_misdn channel private
+	  struct chan_list is not protected from reentrancy. Hangup
+	  collisions have be causing read and write accesses to freed
+	  memory. Converted chan_misdn struct chan_list to an ao2 object
+	  for its reference counting feature. ********** Removed an
+	  impediment to converting chan_list to an ao2 object. The use of
+	  the other_ch member in chan_list is shaky at best. It is set if
+	  the incoming and outgoing call legs are mISDN. The use of the
+	  other_ch member goes against the Asterisk architecture and can
+	  even cause problems. 1) It is used to disable echo cancellation.
+	  This could be bad if the call is forked and the winning call leg
+	  is not mISDN or the winning call leg is not the last mISDN
+	  channel called by the fork. The other_ch would become a dangling
+	  pointer. 2) It is used when the far end is alerting to hear the
+	  far end's inband audio instead of Asterisk's generated ringback
+	  tone. This is bad if the call is forked. You would only hear the
+	  last forked mISDN channel and it may not be ringing yet. The
+	  other_ch would become a dangling pointer if the call is later
+	  transferred. ********** JIRA SWP-2423 JIRA ABE-2614
+
+2010-11-05 22:03 +0000 [r294084]  Brett Bryant <bbryant at digium.com>
+
+	* channels/chan_sip.c: Fixed deadlock avoidance issues while
+	  locking channel when adding the Max-Forwards header to a request.
+	  (closes issue #17949) (closes issue #18200) Reported by: bwg
+	  Review: https://reviewboard.asterisk.org/r/997/
+
+2010-11-05 16:05 +0000 [r294047-294049]  Terry Wilson <twilson at digium.com>
+
+	* contrib/scripts/ast_tls_cert: Corret spelling and example
+
+	* contrib/scripts/ast_tls_cert: Tell people to use the correct
+	  common name for clients as well
+
+2010-11-05 00:07 +0000 [r293970]  Shaun Ruffell <sruffell at digium.com>
+
+	* codecs/codec_dahdi.c, /: Merged revisions 293969 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r293969 | sruffell | 2010-11-04 19:06:02 -0500
+	  (Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010)
+	  | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when
+	  receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically
+	  commit 9034) added the capability for the wctc4xxp to return more
+	  than a single packet of data in response to a read. However, when
+	  decoding packets, codec_dahdi was still assuming that the default
+	  number of samples was in each read. In other words, each packet
+	  your provider sent you, regardless of size, would result in 20 ms
+	  of decoded data (30 ms if decoding G723). If your provider was
+	  sending 60 ms packets then codec_dahdi would end up stripping 40
+	  ms of data from each transcoded frame resulting in "choppy"
+	  audio. This would only affect systems where G729 packets are
+	  arriving in sizes greater than 20ms or G723 packets arriving in
+	  sizes greater than 30ms. DAHDI-744. ........ ................
+
+2010-11-04 21:39 +0000 [r293924]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: Fixes ringback tone on sip semi-attended
+	  transfer. ABE-2168
+
+2010-11-04 13:27 +0000 [r293887]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_sip.c: Do not output port in IPaddress for AMI
+	  sippeers. (closes issue #18248) Reported by: orn Patches:
+	  ami_sippeers.patch uploaded by pabelanger (license 224) Tested
+	  by: orn
+
+2010-11-03 18:35 +0000 [r293807]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+	  293806 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500
+	  (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010)
+	  | 20 lines Party A in an analog 3-way call would continue to hear
+	  ringback after party C answers. All parties are analog FXS ports.
+	  1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to
+	  bring C into 3-way call before C answers. (A and B hear ringback)
+	  4) C answers 5) A continues to hear ringback during the 3-way
+	  call. (All parties can hear each other.) * Fixed use of wrong
+	  variable in dahdi_bridge() that stopped ringback on the wrong
+	  subchannel. * Made several debug messages have more information.
+	  A similar issue happens if B and C are SIP channels. B continues
+	  to hear ringback. For some reason this only affects v1.8 and
+	  trunk. * Don't start ringback on the real and 3-way subchannels
+	  when creating the 3-way conference. Removing this code is benign
+	  on v1.6.2 and earlier. ........ ................
+
+2010-11-03 18:05 +0000 [r293803]  Terry Wilson <twilson at digium.com>
+
+	* include/asterisk/rtp_engine.h, main/rtp_engine.c,
+	  channels/chan_sip.c: Avoid valgrind warnings for
+	  ast_rtp_instance_get_xxx_address The documentation for
+	  ast_rtp_instance_get_(local/remote)_address stated that they
+	  returned 0 for success and -1 on failure. Instead, they returned
+	  0 if the address structure passed in was already equivalent to
+	  the address instance local/remote address or 1 otherwise. 90% of
+	  the calls to these functions completely ignored the return
+	  address and passed in an uninitialized struct, which would make
+	  valgrind complain even though the operation was technically safe.
+	  This patch fixes the documentation and converts the
+	  get_xxx_address functions to void since all they really do is
+	  copy the address and cannot fail. Additionally two new functions
+	  (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created
+	  for the 3 times where the return value was actually checked. The
+	  get_and_cmp_local_address function is currently unused, but
+	  exists for the sake of symmetry. The only functional change as a
+	  result of this change is that we will not do an
+	  ast_sockaddr_cmp() on (mostly uninitialized) addresses before
+	  doing the ast_sockaddr_copy() in the get_*_address functions. So,
+	  even though it is an API change, it shouldn't have a noticeable
+	  change in behavior. Review:
+	  https://reviewboard.asterisk.org/r/995/
+
+2010-11-02 23:09 +0000 [r293724]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 293723 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500
+	  (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
+	  | 8 lines Add enabled/disabled information for rtautoclear sip
+	  show settings output. When setting to zero/"no", the numeric
+	  default was shown making it not obvious the disabled setting was
+	  respected. (closes issue #18123) Reported by: zerohalo ........
+	  ................
+
+2010-11-02 21:29 +0000 [r293648]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+	  293647 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500
+	  (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010)
+	  | 6 lines Make warning message have more useful information in
+	  it. Change "Unable to get index, and nullok is not asserted" to
+	  "Unable to get index for '<channel-name>' on channel <number>
+	  (<function>(), line <number>)". ........ ................
+
+2010-11-02 20:45 +0000 [r293611]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* main/manager.c: If manager and tls are disabled, do not display
+	  TCP/TLS Bindaddress.
+
+2010-11-01 17:29 +0000 [r293530]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c,
+	  channels/sig_analog.h: Analog 3-way call would not connect all
+	  parties if one was using sig_pri. Also the "dahdi show channel"
+	  would not show the correct 3-way call status. * Synchronized the
+	  inthreeway flag between chan_dahdi and sig_analog. * Fixed a
+	  my_set_linear_mode() sign error and made take an analog sub
+	  channel enum.
+
+2010-11-01 16:09 +0000 [r293496]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_iax2.c: Use ast_sockaddr_from_sin function not
+	  memcpy This resolves some IAX2 registration issue report on the
+	  asterisk-users mailing list. (closes issue #18202) Reported by:
+	  pabelanger Patches: update_registry.patch.v2 uploaded by
+	  pabelanger (license 224) Tested by: pabelanger, Nic Colledge
+	  (mailing list) Review: https://reviewboard.asterisk.org/r/993
+
+2010-11-01 14:58 +0000 [r293493]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Only offer codecs both sides support for
+	  directmedia When using directmedia, Asterisk needs to limit the
+	  codecs offered to just the ones that both sides recognize,
+	  otherwise they may end up sending audio that the other side
+	  doesn't understand. (closes issue #17403) Reported by: one47
+	  Patches: sip_codecs_simplified4 uploaded by one47 (license 23)
+	  Tested by: one47, falves11 Review:
+	  https://reviewboard.asterisk.org/r/967/
+
+2010-10-30 01:53 +0000 [r293341-293418]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+	  293417 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500
+	  (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
+	  Oct 2010) | 1 line Remove some more code that serves no purpose.
+	  ........ ................
+
+	* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
+	  293340 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500
+	  (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
+	  Oct 2010) | 1 line Remove some code that serves no purpose.
+	  ........ ................
+
+2010-10-29 21:48 +0000 [r293305]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_sip.c: Modify sip_setoption to not complain about
+	  unknown options. This now behaves just like the other setoption
+	  callbacks. For the curious the offending option for the reporter
+	  was AST_OPTION_CHANNEL_WRITE which was getting passed due to a
+	  fix for chan_local in 286189. (closes issue #17985) Reported by:
+	  globalnetinc
+
+2010-10-28 20:00 +0000 [r293197]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/ael/ael.tab.h, main/ast_expr2.c, /, main/ast_expr2.h,
+	  res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c: Merged
+	  revisions 293195-293196 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r293195 | tilghman | 2010-10-28 14:52:52 -0500
+	  (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+	  | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+	  Reported (though the reporter did not understand he was reporting
+	  a bug) on the asterisk-users list:
+	  http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+	  ........ ................ r293196 | tilghman | 2010-10-28
+	  14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions
+	  293194 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+	  | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+	  Reported (though the reporter did not understand he was reporting
+	  a bug) on the asterisk-users list:
+	  http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+	  ........ ................
+
+2010-10-28 16:11 +0000 [r293159]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, funcs/func_strings.c: Merged revisions 293158 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28
+	  Oct 2010) | 11 lines Fix infinite loop in FILTER(). Specifically
+	  when you're using characters above \x7f or invalid character
+	  escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes
+	  Patches: issue18060_func_strings_filter_infinite_loop.patch
+	  uploaded by wdoekes (license 717) Tested by: wdoekes ........
+
+2010-10-26 18:49 +0000 [r293119]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 293118 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500
+	  (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010)
+	  | 29 lines Fix inprocess_container in voicemail to correctly
+	  restrict max messages. The comparison function logic was off, so
+	  the number of sessions for a given mailbox were not being
+	  incremented properly. This problem caused the maximum number of
+	  messages per folder to not be respected when simultaneously
+	  leaving multiple voicemails just below the threshold. These
+	  problems should be fixed by the above, but just in case: Fixed
+	  resequence_mailbox to rely on the actual number of detected
+	  number of files in a directory rather than just assuming only 10
+	  messages more than the maximum had been left. Also if more
+	  messages than the maximum are deleted they are actually removed
+	  now. The second purpose of this commit should have been separated
+	  out probably, but is related to the above. Again, if the number
+	  of messages in a given voicemail folder exceeds the maximum set
+	  limit make sure to allocate enough space for the deleted and
+	  heard index tracking array. A few random fixes: There was a
+	  forgotten decrement of the inprocess count in imap_store_file.
+	  When using IMAP storage, do not look in the directory where file
+	  based storage messages may still reside and influence the message
+	  count. Ensure to use only the first format in sendmail. ABE-2516
+	  ........ ................
+
+2010-10-26 16:32 +0000 [r293046-293081]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c: No need to define the struct if there are no
+	  users.
+
+	* channels/sig_pri.c, configure, include/asterisk/autoconfig.h.in,
+	  configure.ac: Allow the DAHDI driver to compile, even with a
+	  sufficiently older version of libpri. Fixes our Bamboo builds.
+
+2010-10-25 21:15 +0000 [r292906-292969]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/sig_pri.c: Several more defines that need to be altered
+	  for compiling against an older version of libpri
+
+	* channels/sig_pri.c, configure, include/asterisk/autoconfig.h.in,
+	  configure.ac: Allow the DAHDI driver to compile, even with a
+	  sufficiently older version of libpri. Fixes our Bamboo builds.
+
+2010-10-25 19:07 +0000 [r292868]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_local.c, /: Merged revisions 292867 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500
+	  (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010)
+	  | 27 lines This patch turns chan_local pvts into astobj2 objects.
+	  chan_local does some dangerous things involving deadlock
+	  avoidance. tech_pvt functions like hangup and queue_frame are
+	  provided with a locked channel upon entry. Those functions are
+	  completely safe as long as you don't attempt to give up that
+	  channel lock, but that is impossible to guarantee due to the
+	  required deadlock avoidance necessary to lock both the tech_pvt
+	  and both channels involved. In the past, we have tried to account
+	  for this by doing things like setting a "glare" flag that
+	  indicates what function should destroy the pvt. This was used in
+	  local_hangup and local_queue_frame to decided who should destroy
+	  the pvt if they collided in separate threads. I have removed the
+	  need to do this by converting all chan_local tech_pvts to
+	  astobj2. This means we can ref a pvt before deadlock avoidance
+	  and not have to worry about that pvt possibly getting destroyed
+	  under us. It also cleans up where we destroy the tech_pvt. The
+	  only unlink from the tech_pvt container occurs in local_hangup
+	  now, which is where it should occur. Since there still may be
+	  thread collisions on some functions like local_hangup after
+	  deadlock avoidance, I have added some checks to detect those
+	  collisions and exit appropriately. I think this patch is going to
+	  solve quite a bit of weirdness we have had with local channels in
+	  the past. ........ ................
+
+2010-10-22 22:35 +0000 [r292794-292825]  Terry Wilson <twilson at digium.com>
+
+	* contrib/scripts/ast_tls_cert: Don't create directories without at
+	  least o+x Also, making files that you are going to modify
+	  read-only is dumb.
+
+	* contrib/scripts/ast_tls_cert: Make files readable only by the
+	  owner
+
+2010-10-22 21:28 +0000 [r292787]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/res_ldap.conf.sample, contrib/scripts/asterisk.ldif, /,
+	  channels/chan_sip.c: Merged revisions 292786 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010)
+	  | 13 lines Update the LDIF file for LDAP. The LDIF file
+	  asterisk.ldif was quite a bit out of date from the
+	  asterisk.ldap-schema file, so I've now updated that to be in
+	  sync. The asterisk.ldif file being out of sync was a problem on
+	  my systems where I was doing an ldapadd to import the schema into
+	  the LDAP database, and the existing file would cause problems and
+	  ERROR messages when registering. Additional documention has been
+	  added based on feedback in the issue I'm closing. (closes issue
+	  #13861) Reported by: scramatte Patches: ldap-update.txt uploaded
+	  by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec,
+	  rgenthner ........
+
+2010-10-22 17:09 +0000 [r292741]  Mark Michelson <mmichelson at digium.com>
+
+	* tests/test_event.c: Prevent multiple runs of event_sub_test from
+	  producing false failure results. The array of test subscriptions
+	  was declared "static," meaning that the data.count field would
+	  retain its value between runs of the test. After the first test
+	  run, this would result in false reports of test failures. I chose
+	  to just remove the "static" keyword from the structure since it's
+	  not a huge deal to construct this structure during each run of
+	  the test. Another alternative would have been to zero out the
+	  data.count fields of each test subscription instead.
+
+2010-10-22 16:49 +0000 [r292740]  Terry Wilson <twilson at digium.com>
+
+	* contrib/scripts/ast_tls_cert (added): Add TLS cert helper script
+	  This script is useful for quickly generating self-signed CA,
+	  server, and client certificates for use with Asterisk. It is
+	  still recommended to obtain certificates from a recognized
+	  Certificate Authority and to develop an understanding how SSL
+	  certificates work. Real security is hard work. OPTIONS: -h Show
+	  this message -m Type of cert "client" or "server". Defaults to
+	  server. -f Config filename (openssl config file format) -c CA
+	  cert filename (creates new CA cert/key as ca.crt/ca.key if not
+	  passed) -k CA key filename -C Common name (cert field) For a
+	  server cert, this should be the same address that clients attempt
+	  to connect to. Usually this will be the Fully Qualified Domain
+	  Name, but might be the IP of the server. For a CA or client cert,
+	  it is merely informational. Make sure your certs have unique
+	  common names. -O Org name (cert field) An informational string
+	  (company name) -o Output filename base (defaults to asterisk) -d
+	  Output directory (defaults to the current directory) Example: To
+	  create a CA and a server (pbx.mycompany.com) cert with output in
+	  /tmp: ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp
+	  This will create a CA cert and key as well as asterisk.pem and
+	  the the two files that it is made from: asterisk.crt and
+	  asterisk.key. Copy asterisk.pem and ca.crt somewhere (like
+	  /etc/asterisk) and set tlscertfile=/etc/asterisk.pem and
+	  tlscafile=/etc/ca.crt. Since this is a self-signed key, many
+	  devices will require you to import the ca.crt file as a trusted
+	  cert. To create a client cert using the CA cert created by the
+	  example above: ast_tls_cert -m client -c /tmp/ca.crt -k
+	  /tmp/ca.key -C "Joe User" -O \ "My Company" -d /tmp -o joe_user
+	  This will create client.crt/key/pem in /tmp. Use this if your
+	  device supports a client certificate. Make sure that you have the
+	  ca.crt file set up as a tlscafile in the necessary Asterisk
+	  configs. Make backups of all .key files in case you need them
+	  later.
+
+2010-10-22 15:47 +0000 [r292704]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c, main/channel.c, channels/chan_misdn.c:
+	  Connected line is not updated when chan_dahdi/sig_pri or
+	  chan_misdn transfers a call. When a call is transfered by ECT or
+	  implicitly by disconnect in sig_pri or implicitly by disconnect
+	  in chan_misdn, the connected line information is not exchanged.
+	  The connected line interception macros also need to be executed
+	  if defined. The CALLER interception macro is executed for the
+	  held call. The CALLEE interception macro is executed for the
+	  active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches:
+	  abe_2589_c3bier.patch uploaded by rmudgett (license 664)
+	  abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review:
+	  https://reviewboard.asterisk.org/r/958/
+
+2010-10-21 22:09 +0000 [r292667]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/misdn/ie.c: Compile correctly on Linux
+	  (asterisk/localtime.h depends upon asterisk/autoconfig.h loading
+	  first).
+

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