[asterisk-commits] lmadsen: tag 1.6.2.0-rc3 r222236 - /tags/1.6.2.0-rc3/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Oct 6 11:17:30 CDT 2009


Author: lmadsen
Date: Tue Oct  6 11:17:28 2009
New Revision: 222236

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=222236
Log:
Importing files for 1.6.2.0-rc3 release.

Added:
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    tags/1.6.2.0-rc3/.version   (with props)
    tags/1.6.2.0-rc3/ChangeLog   (with props)

Added: tags/1.6.2.0-rc3/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.0-rc3/.lastclean?view=auto&rev=222236
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Added: tags/1.6.2.0-rc3/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.0-rc3/ChangeLog?view=auto&rev=222236
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--- tags/1.6.2.0-rc3/ChangeLog (added)
+++ tags/1.6.2.0-rc3/ChangeLog Tue Oct  6 11:17:28 2009
@@ -1,0 +1,19079 @@
+2009-10-06  Leif Madsen <lmadsen at digium.com>
+
+	* Released Asterisk 1.6.2.0-rc3
+
+2009-10-06 01:39 +0000 [r222113-222187]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c,
+	  channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c,
+	  res/res_clialiases.c, /, channels/chan_sip.c,
+	  funcs/func_dialgroup.c, include/asterisk/astobj2.h,
+	  res/res_phoneprov.c: Merged revisions 222176 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct
+	  2009) | 27 lines Recorded merge of revisions 222152 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05
+	  Oct 2009) | 20 lines Fix ao2_iterator API to hold references to
+	  containers being iterated. See Mantis issue for details of what
+	  prompted this change. Additional notes: This patch changes the
+	  ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum
+	  instead of a macro, with a name that fits our naming policy;
+	  also, it is now necessary to call ao2_iterator_destroy() on any
+	  iterator that has been created. Currently this only releases the
+	  reference to the container being iterated, but in the future this
+	  could also release other resources used by the iterator, if the
+	  iterator implementation changes to use additional resources.
+	  (closes issue #15987) Reported by: kpfleming Review:
+	  https://reviewboard.asterisk.org/r/383/ ........ ................
+
+	* configs/sip.conf.sample, main/udptl.c, /, channels/chan_sip.c,
+	  configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 222110
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05
+	  Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be
+	  supportable via configuration option. Many T.38 endpoints
+	  incorrectly send the maximum IFP frame size they can accept as
+	  the T38FaxMaxDatagram value in their SDP, when in fact this value
+	  is supposed to be the maximum UDPTL payload size (datagram size)
+	  they can accept. If the value they supply is small enough (a
+	  commonly supplied value is '72'), T.38 UDPTL transmissions will
+	  likely fail completely because the UDPTL packets will not have
+	  enough room for a primary IFP frame and the redundancy used for
+	  error correction. If this occurs, the Asterisk UDPTL stack will
+	  emit log messages warning that data loss may occur, and that the
+	  value may need to be overridden. This patch extends the
+	  't38pt_udptl' configuration option in sip.conf to allow the
+	  administrator to override the value supplied by the remote
+	  endpoint and supply a value that allows T.38 FAX transmissions to
+	  be successful with that endpoint. In addition, in any SIP call
+	  where the override takes effect, a debug message will be printed
+	  to that effect. This patch also removes the T38FaxMaxDatagram
+	  configuration option from udptl.conf.sample, since it has not
+	  actually had any effect for a number of releases. In addition,
+	  this patch cleans up the T.38 documentation in sip.conf.sample
+	  (which incorrectly documented that T.38 support was passthrough
+	  only). (issue #15586) Reported by: globalnetinc ........
+
+2009-10-02 17:35 +0000 [r222032]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 222030 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500
+	  (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02
+	  Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a
+	  memcpy. ........ ................
+
+2009-10-02 17:01 +0000 [r221923-221974]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/astobj2.c, /: Merged revisions 221971 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009)
+	  | 9 lines Merged revisions 221970 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009)
+	  | 2 lines Ensure the result of the hash function is positive.
+	  Negative array offsets suck. ........ ................
+
+	* /, main/logger.c: Merged revisions 221920 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 |
+	  tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines
+	  Initialize a variable that we check immediately upon startup.
+	  (closes issue #15973) Reported by: atis ........
+
+2009-10-02 01:35 +0000 [r221879]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
+	  Merged revisions 221844 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009)
+	  | 33 lines Merged revisions 221769 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009)
+	  | 26 lines Occasionally losing use of B channels in chan_misdn. I
+	  have not been able to reproduce the problem of losing channels.
+	  However, I have seen in the code a reentrancy problem that might
+	  give these symptoms. The reentrancy patch does several things: 1)
+	  Guards B channel and B channel structure allocation. 2) Makes the
+	  B channel structure find routines more precise in locating
+	  records. 3) Never leave a B channel allocated if we received
+	  cause 44. The last item may cause temporary outgoing call
+	  problems, but they should clear when the line becomes idle.
+	  (closes issue #15490) Reported by: slutec18 Patches:
+	  issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
+	  (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
+	  Reported by: FabienToune Patches:
+	  issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
+	  (license 664) Tested by: FabienToune, rmudgett, slutec18 ........
+	  ................
+
+2009-10-02 00:07 +0000 [r221744-221780]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions
+	  221777 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009)
+	  | 9 lines Merged revisions 221776 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009)
+	  | 2 lines Fix a bunch of off-by-one errors ........
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 |
+	  tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines
+	  Revision 220906 (a merge from 1.4) was not merged correctly,
+	  causing a problem with non-dynamic peers. ........
+
+2009-10-01 19:35 +0000 [r221698]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 |
+	  dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines
+	  outbound tls connections were not defaulting to port 5061 (closes
+	  issue #15854) Reported by: dvossel Patches:
+	  sip_port_config_trunk.diff uploaded by dvossel (license 671)
+	  Tested by: dvossel ........
+
+2009-10-01 16:57 +0000 [r221660]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 221554,221589 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu,
+	  01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE
+	  constructs when it's just TRUE or FALSE. ................ r221589
+	  | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9
+	  lines Merged revisions 221588 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct
+	  2009) | 2 lines Use unsigned ints for portinuri flags. ........
+	  ................
+
+2009-10-01 16:25 +0000 [r221622]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged
+	  revisions 221592 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 |
+	  kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12
+	  lines Remove ability to control T.38 FAX error correction from
+	  udptl.conf. chan_sip has had the ability to control T.38 FAX
+	  error correction mode on a per-peer (or global) basis for a
+	  couple of releases now, which is where it should have been all
+	  along. This patch removes the ability to configure it in
+	  udptl.conf, but issues a warning if the user tries to do, telling
+	  them to look at sip.conf.sample for how to configure it now. For
+	  any SIP peers that are T.38 enabled in sip.conf, there is already
+	  a default for FEC error correction even if the user does not
+	  specify any mode, so this change will not turn off error
+	  correction by default, it will have the same default value that
+	  has been in the udptl.conf sample file. ........
+
+2009-09-30 23:07 +0000 [r221477-221485]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 |
+	  mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2
+	  lines Cleaned up merge from r221432 ........
+
+	* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+	  221432 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep
+	  2009) | 17 lines Merged revisions 221360 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep
+	  2009) | 10 lines Fix SRV lookup and Request-URI generation in
+	  chan_sip. This patch adds a new field "portinuri" to the sip
+	  dialog struct and the sip peer struct. That field is used during
+	  RURI generation to determine if the port should be included in
+	  the RURI. It is also used in some places to determine if an SRV
+	  lookup should occur. (closes issue #14418) Reported by: klaus3000
+	  Tested by: klaus3000, mnicholson Review:
+	  https://reviewboard.asterisk.org/r/369/ ........ ................
+
+2009-09-30 21:46 +0000 [r221371-221472]  Matthias Nick <mnick at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 221436 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 |
+	  mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines
+	  Prevents from division by zero ........
+
+	* configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged
+	  revisions 221368 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) |
+	  23 lines Merged revisions 221153,221157,221303 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) |
+	  2 lines check bounds - prevents for buffer overflow ........
+	  r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) |
+	  8 lines added a new dialplan function 'CSV_QUOTE' and changed the
+	  cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
+	  Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
+	  mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed,
+	  30 Sep 2009) | 2 lines changed the prototype definition of
+	  csv_quote ........ ................
+
+2009-09-30 19:15 +0000 [r221304]  Terry Wilson <twilson at digium.com>
+
+	* configs/sip.conf.sample, main/rtp.c, /, channels/chan_sip.c,
+	  include/asterisk/rtp.h: Merged revisions 221266 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009)
+	  | 32 lines Merged revisions 221086 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009)
+	  | 25 lines Change the SSRC by default when our media stream
+	  changes Be default, change SSRC when doing an audio stream
+	  changes Asterisk doesn't honor marker bit when reinvited to
+	  already-bridged RTP streams,resulting in far-end stack discarding
+	  packets with "old" timestamps that areactually part of a new
+	  stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is
+	  a reinvite, unless the 'constantssrc' is set to true in sip.conf.
+	  The original issue reported to Digium support detailed the
+	  following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
+	  Application Server Call comes in fromITSP, Asterisk dials the app
+	  server which sends a re-invite back toAsterisk--not to negotiate
+	  to send media directly to the ITSP, but to indicatethat it's
+	  changing the stream it's sending to Asterisk. The app
+	  servergenerates a new SSRC, sequence numbers, timestamps, and
+	  sets the marker bit on the new stream. Asterisk passes through
+	  the teimstamp of the new stream, butdoes not reset the SSRC,
+	  sequence numbers, or set the marker bit. When the timestamp on
+	  the new stream is older than the timestamp on the originalstream,
+	  the ITSP (which doesn't know there has been any change) discards
+	  the newframes because it thinks they are too old. This patch
+	  addresses this by changing the SSRC on a stream update unless
+	  constantssrc=true is set in sip.conf. Review:
+	  https://reviewboard.asterisk.org/r/374/ ........ ................
+
+2009-09-30 16:57 +0000 [r221204]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/channel.c, /: Merged revisions 221201 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009)
+	  | 14 lines Merged revisions 221200 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009)
+	  | 7 lines Avoid a potential NULL dereference. (closes issue
+	  #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
+	  uploaded by tilghman (license 14) Tested by: kobaz ........
+	  ................
+
+2009-09-30 14:57 +0000 [r221089]  Sean Bright <sean at malleable.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep
+	  2009) | 9 lines Clarify documentation for VoiceMailMain()'s a()
+	  option. We require box numbers, not names as the documentation
+	  implies. (issue #14740) Reported by: pj Patches:
+	  __20090729-app_voicemail-documentation.patch uploaded by lmadsen
+	  (license 10) Tested by: seanbright, lmadsen ........
+
+2009-09-30 04:41 +0000 [r221027-221047]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, funcs/func_lock.c: Recorded merge of revisions 221044 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29
+	  Sep 2009) | 8 lines Allow locks to be inherited through a
+	  masquerade without causing starvation. (closes issue #14859)
+	  Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded
+	  by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt
+	  uploaded by tilghman (license 14) Tested by: atis, tilghman
+	  ........
+
+	* include/asterisk/smdi.h, include/asterisk/optional_api.h
+	  (removed), apps/app_voicemail.c, include/asterisk/agi.h,
+	  include/asterisk/monitor.h: Remove optional_api from 1.6.2
+	  branch, since it is not currently working. This is a blocking
+	  issue for the 1.6.2 release. (closes issue #15914) Reported by:
+	  mbeckwell Branch:
+	  http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162
+	  Tested by: mbeckwell
+
+	* /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009)
+	  | 16 lines Merged revisions 220873 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009)
+	  | 9 lines Reduce CPU usage related to building a peer merely for
+	  devicestates. This fixes a 100% CPU problem in the SIP driver,
+	  found by profiling the driver while the problem was occurring.
+	  (closes issue #14309) Reported by: pkempgen Patches:
+	  20090924__issue14309.diff.txt uploaded by tilghman (license 14)
+	  Tested by: pkempgen, vrban ........ ................
+
+2009-09-29 20:24 +0000 [r220905-220934]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the
+	  spyee is masqueraded and chanspy_ds_chan_fixup() is called with
+	  the channel locked. (closes issue #15965) Reported by: atis
+	  Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson
+	  (license 96) Tested by: atis
+
+	* /, apps/app_confbridge.c: Merged revisions 220904 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep
+	  2009) | 5 lines Fix options 'm' and 's'. They were swapped in the
+	  code. Also document the fact that app_confbridge does not
+	  automatically answer the channel. (closes issue #15964) Reported
+	  by: shrift ........
+
+2009-09-29 17:06 +0000 [r220836]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009)
+	  | 12 lines Make deletion of temporary greetings work properly
+	  with IMAP_STORAGE When imapgreetings was set to yes, the message
+	  was being deleted but wasn't actually being expunged. When
+	  imapgreetings was set to no, the file based message was not being
+	  deleted at all. All good now! (closes issue #14949) Reported by:
+	  noahisaac Patches: vm_tempgreeting_removal.patch uploaded by
+	  noahisaac (license 748), modified by me ........
+
+2009-09-28 19:13 +0000 [r220725]  Sean Bright <sean at malleable.com>
+
+	* /, Makefile.rules: Merged revisions 220721 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep
+	  2009) | 10 lines Merged revisions 220717 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep
+	  2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect,
+	  explicitly pass -O0 to the compiler so we override any default
+	  optimization levels for a particular install. ........
+	  ................
+
+2009-09-28 19:11 +0000 [r220722]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 220718 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 |
+	  jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines
+	  Fix building of registration entry in build_peer when using
+	  callbackextension Check for remotesecret option was
+	  unintentionally always true, which therefore caused the secret
+	  option to never be used. Thanks to dvossel for pointing out the
+	  exact fix. (closes issue #15943) Reported by: tpsast ........
+
+2009-09-27 20:45 +0000 [r220632]  Michiel van Baak <michiel at vanbaak.info>
+
+	* funcs/func_callerid.c, /: Merged revisions 220629 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009)
+	  | 3 lines add name argument for the CALLERID dialplan function to
+	  the xml documentation. Pointed out to me on IRC by snuff-home.
+	  Thanks ........
+
+2009-09-26 15:12 +0000 [r220589]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009)
+	  | 2 lines Allow AES to compile, when OpenSSL is not present.
+	  ........
+
+2009-09-24 20:38 +0000 [r220369]  David Vossel <dvossel at digium.com>
+
+	* main/tcptls.c, /: Merged revisions 220365 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 |
+	  dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines
+	  fixes tcptls_session memory leak caused by ref count error
+	  (closes issue #15939) Reported by: dvossel Review:
+	  https://reviewboard.asterisk.org/r/375/ ........
+
+2009-09-24 19:42 +0000 [r220292]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged
+	  revisions 220289 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009)
+	  | 13 lines Merged revisions 220288 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009)
+	  | 6 lines Implicitly sending a progress signal breaks some
+	  applications. Call Progress() in your dialplan if you explicitly
+	  want progress to be sent. (Reverts change 216430, closes issue
+	  #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing
+	  list
+	  http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
+	  ........ ................
+
+2009-09-24 18:22 +0000 [r220103-220221]  Sean Bright <sean at malleable.com>
+
+	* Makefile, /: Merged revisions 220217 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep
+	  2009) | 9 lines Merged revisions 220213 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep
+	  2009) | 1 line Resolve parallel build warnings. Reported by Klaus
+	  Darilion on the asterisk-dev mailing list. ........
+	  ................
+
+	* Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400
+	  (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu,
+	  24 Sep 2009) | 2 lines Remove the remaining bashisms in the
+	  Makefile/mkpkgconfig ........ ................
+
+2009-09-24 08:43 +0000 [r220031]  Michiel van Baak <michiel at vanbaak.info>
+
+	* build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200
+	  (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009)
+	  | 7 lines mkpkgconfig does not need bash so make it use /bin/sh
+	  This fixes building on all systems that don't have bash at
+	  /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on
+	  #asterisk-dev ........ ................
+
+2009-09-24 07:45 +0000 [r219989]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_directory.c, /: Merged revisions 219987 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009)
+	  | 8 lines Fix two possible crashes, one only in 1.6.1 and one in
+	  1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg
+	  Patches: 20090914__issue15739.diff.txt uploaded by tilghman
+	  (license 14) 20090922__issue15739.diff.txt uploaded by tilghman
+	  (license 14) Tested by: DLNoah, jeffg ........
+
+2009-09-22 21:48 +0000 [r219821]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500
+	  (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009)
+	  | 10 lines When IMAP variables were changed during a reload,
+	  Voicemail did not use the new values. This change introduces a
+	  configuration version variable, which ensures that connections
+	  with the old values are not reused but are allowed to expire
+	  normally. (closes issue #15934) Reported by: viniciusfontes
+	  Patches: 20090922__issue15934.diff.txt uploaded by tilghman
+	  (license 14) Tested by: viniciusfontes ........ ................
+
+2009-09-21 17:01 +0000 [r219722]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 219721 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500
+	  (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21
+	  Sep 2009) | 3 lines Reverting merge 219520. This change was not
+	  necessary. ........ ................
+
+2009-09-20 18:21 +0000 [r219669]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, main/file.c: Merged revisions 219654 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009)
+	  | 15 lines Merged revisions 219653 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009)
+	  | 8 lines Really stop the stream, when ast_closestream() is
+	  called. (closes issue #15129) Reported by: bmh Patches:
+	  20090918__issue15129.diff.txt uploaded by tilghman (license 14)
+	  Review: https://reviewboard.asterisk.org/r/372/ ........
+	  ................
+
+2009-09-19 03:14 +0000 [r219590]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 219587 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r219587 | russell | 2009-09-18 21:59:52 -0500
+	  (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009)
+	  | 6 lines Make sure the iax_pvt exists before dereferencing it.
+	  This fixes the latest crash posted on issue 15609. (issue #15609)
+	  ........ ................
+
+2009-09-18 23:21 +0000 [r219452-219521]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 219520 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500
+	  (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009)
+	  | 9 lines iax2 frame double free The iax frame's retrans sched id
+	  was written over right before iax2_frame_free was called. In
+	  iax2_frame_free that retrans id is used to delete the sched item.
+	  By writing over the retrans field before the sched item could be
+	  deleted, it was possible for a retransmit to occur on a freed
+	  frame. ........ ................
+
+	* /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009)
+	  | 20 lines Merged revisions 219450 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009)
+	  | 14 lines via-header branches not updated correctly on INVITE
+	  INVITE requests must always contain a new unique branch id. When
+	  a new branch id is created for an INVITE, the dialog's
+	  invite_branch variable must be updated so CANCEL requests use the
+	  correct branch id. (closes issue #15262) Reported by: maniax
+	  Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
+	  (license 608) invite_new_branch_trunk.diff uploaded by dvossel
+	  (license 671) Tested by: maniax, dvossel ........
+	  ................
+
+2009-09-18 13:57 +0000 [r219415]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009)
+	  | 6 lines Missing value setting line for maxsecs/maxmessage
+	  (closes issue #15696) Reported by: fhackenberger Patches:
+	  maxsecs.patch uploaded by fhackenberger (license 592) ........
+
+2009-09-17 22:38 +0000 [r219376]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 219371 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 |
+	  dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines
+	  fixes deadlock when performing directed pickup w Invite/replaces
+	  (closes issue #15340) Reported by: lmsteffan Patches:
+	  deadlock.patch uploaded by lmsteffan (license 779) Tested by:
+	  lmsteffan ........
+
+2009-09-17 22:37 +0000 [r219370]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep
+	  2009) | 12 lines Merged revisions 219320 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep
+	  2009) | 6 lines Send a 100 Trying response when we detect a
+	  spiral. This was problematic during spiral tests at SIPit...
+	  along with some other things as well. ........ ................
+
+2009-09-17 22:06 +0000 [r219307]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009)
+	  | 27 lines Merged revisions 219303 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009)
+	  | 21 lines INVITE w/Replaces deadlock fix This patch cleans up
+	  the locking logic in chan_sip.c's handle_invite_replaces()
+	  function as well as making use of ast_do_masquerade() rather than
+	  forcing the masquerade on an ast_read(). The code had several
+	  redundant unlocks that would result in 'freed more times than
+	  we've locked!' errors. I cleaned these up as well as moving all
+	  the unlock logic to the end of the function. This patch should
+	  also resolve the issue people were having with the replacecall
+	  channel never being unlocked with one legged calls. (closes issue
+	  #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff
+	  uploaded by dvossel (license 671) Tested by: irroot, dvossel
+	  Review: https://reviewboard.asterisk.org/r/371/ ........
+	  ................
+
+2009-09-17 19:58 +0000 [r219267]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 |
+	  file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines
+	  Ensure no spaces exist before "refresher=" when doing the
+	  comparison. ........
+
+2009-09-17  Leif Madsen <lmadsen at digium.com>
+
+	* Released Asterisk 1.6.2.0-rc2
+
+2009-09-17 15:38 +0000 [r219194]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/channel.c, /, include/asterisk/cdr.h,
+	  include/asterisk/channel.h: Merged revisions 219139 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500
+	  (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
+	  2009) | 10 lines Prevent a potential race condition and crash
+	  when hanging up a channel by removing the channel from the
+	  channel list before begining channel tear down. This fix may
+	  potentially cause problems with CDR backends that access the
+	  channel a CDR is associated with via the channel list. This fix
+	  makes the channel unavabile at the time when the CDR backend is
+	  invoked. This has been documented in include/asterisk/cdr.h.
+	  (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+	  Review: https://reviewboard.asterisk.org/r/362/ ........
+	  ................
+
+2009-09-16 23:52 +0000 [r219063]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/config.c, configs/extensions.conf.sample, /: Merged
+	  revisions 219061 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009)
+	  | 15 lines Merged revisions 219023 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
+	  | 8 lines Properly deal with quotes in the arguments of '#exec'
+	  includes. (closes issue #15583) Reported by: pkempgen Patches:
+	  20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+	  20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+	  169) Tested by: pkempgen ........ ................
+
+2009-09-16 19:40 +0000 [r218938]  David Brooks <dbrooks at digium.com>
+
+	* main/pbx.c, /: Merged revisions 218868 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009)
+	  | 20 lines Merged revisions 218867 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
+	  | 13 lines Fixes CID pattern matching behavior to mirror that of
+	  extension pattern matching. Pattern matching for extensions uses
+	  a type of scoring system, giving values for specificity to each
+	  character in the pattern. Unfortunately, this is done character
+	  by character, in order. This does lead to some less specific
+	  patterns being first in line for matching, but it will usually
+	  get the job done. This patch merely brings CID matching to the
+	  same level as extension matching. This patch does not attempt to
+	  tackle the problem shared by extension matching. (closes issue
+	  #14708) Reported by: klaus3000 ........ ................
+
+2009-09-16 19:29 +0000 [r218937]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 |
+	  mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12
+	  lines Reverse order of args to fread. This way, we don't always
+	  write a null byte into byte 1 of the buffer (closes issue #15905)
+	  Reported by: ebroad Patches: freadfix.patch uploaded by ebroad
+	  (license 878) Tested by: ebroad ........
+
+2009-09-16 19:25 +0000 [r218934]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 |
+	  file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On
+	  TCP and TLS connections do not attempt to stop retransmission of
+	  the packet internally. This was preventing responses from being
+	  properly processed because the packet was not being found causing
+	  handle_response to return prematurely. ........
+
+2009-09-16 13:38 +0000 [r218802]  Russell Bryant <russell at digium.com>
+
+	* contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
+	  revisions 218799 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009)
+	  | 16 lines Merged revisions 218798 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
+	  | 9 lines Remove the IAXy firmware from Asterisk. The firmware
+	  can now be found on downloads.digium.com, where the rest of our
+	  binary downloads live. This was the last part of our Asterisk

[... 18402 lines stripped ...]



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