[asterisk-commits] lmadsen: tag 1.8.0-beta1 r279203 - /tags/1.8.0-beta1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Jul 23 15:43:17 CDT 2010


Author: lmadsen
Date: Fri Jul 23 15:43:13 2010
New Revision: 279203

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=279203
Log:
Importing files for 1.8.0-beta1 release.

Added:
    tags/1.8.0-beta1/.lastclean   (with props)
    tags/1.8.0-beta1/.version   (with props)
    tags/1.8.0-beta1/ChangeLog   (with props)

Added: tags/1.8.0-beta1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-beta1/.lastclean?view=auto&rev=279203
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Added: tags/1.8.0-beta1/ChangeLog
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==============================================================================
--- tags/1.8.0-beta1/ChangeLog (added)
+++ tags/1.8.0-beta1/ChangeLog Fri Jul 23 15:43:13 2010
@@ -1,0 +1,21960 @@
+2010-07-12  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.8.0-beta1 Released.
+
+2010-07-23 18:56 +0000 [r279113]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_odbc.c: Silly 64-bit compilers (who uses 64-bit anyway?)
+
+2010-07-23 18:23 +0000 [r279056-279094]  Russell Bryant <russell at digium.com>
+
+	* /: fix up properties on 1.8 branch
+
+	* / (added): Create a branch for Asterisk 1.8.
+
+	  ___      _            _     _      _   ___
+	 / _ \ ___| |_ ___ _ __(_)___| | __ / | ( _ )
+	| |_| / __| __/ _ \ '__| / __| |/ / | | / _ \
+	|  _  \__ \ ||  __/ |  | \__ \   <  | || (_) |
+	|_| |_|___/\__\___|_|  |_|___/_|\_\ |_(_)___/
+
+2010-07-23 17:05 +0000 [r278982-278985]  Tilghman Lesher <tlesher at digium.com>
+
+	* autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
+	  revisions 278984 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
+	  | 5 lines Establish a maximum version for openh323 (i.e. not
+	  opal), because chan_h323 will fail to load, even if it links.
+	  (issue #17679) Reported by: am ........
+
+	* /, main/asterisk.c: Merged revisions 278981 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
+	  | 8 lines Avoid race with consolethread on shutdown (on parallel
+	  processors). (closes issue #17080) Reported by: sybasesql
+	  Patches: 20100721__issue17080.diff.txt uploaded by tilghman
+	  (license 14) Tested by: sybasesql ........
+
+2010-07-23 16:33 +0000 [r278980]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c, channels/sip/reqresp_parser.c,
+	  channels/sip/include/reqresp_parser.h: SIP URI comparison fixes.
+	  This initially was created to work around the issue of using a
+	  string comparison instead of a binary comparison for IP
+	  addresses. It evolved a bit when test cases were created and it
+	  was discovered that comparison of URI parameters was not working
+	  exactly as it should. sip_uri_cmp() and its helpers have been
+	  moved to reqresp_parser.c and a new test has been added. (closes
+	  issue #17662) Reported by: oej Review:
+	  https://reviewboard.asterisk.org/r/792
+
+2010-07-23 16:19 +0000 [r278957]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/res_odbc.h, res/res_config_odbc.c,
+	  configs/extconfig.conf.sample, CHANGES, main/config.c,
+	  res/res_odbc.c, configs/res_odbc.conf.sample: Merge the realtime
+	  failover branch
+
+2010-07-23 16:07 +0000 [r278947]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* doc/asterisk.8: Some left-over hyphen-minus fixes in the man page
+
+2010-07-23 15:57 +0000 [r278944-278945]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: ... just kidding. Enable SIP by default. :-)
+
+	* channels/chan_sip.c: Disable SIP support by default for Asterisk
+	  1.8.
+
+2010-07-23 15:52 +0000 [r278943]  Mark Michelson <mmichelson at digium.com>
+
+	* addons/chan_ooh323.c: Well, who knew chan_ooh323 used udptl? I
+	  sure didn't!
+
+2010-07-23 15:41 +0000 [r278942]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+	  Rename sig_pri_pri to sig_pri_span. More descriptive of concept.
+
+2010-07-23 15:16 +0000 [r278908]  Mark Michelson <mmichelson at digium.com>
+
+	* main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h,
+	  channels/sip/include/sip.h: Allow IPv6 addresses for UDPTL
+	  streams. Review: https://reviewboard.asterisk.org/r/795
+
+2010-07-23 13:37 +0000 [r278875]  Olle Johansson <oej at edvina.net>
+
+	* res/res_config_ldap.c: Minor corrections to the LDAP realtime
+	  driver Review: https://reviewboard.asterisk.org/r/798/ Thanks
+	  Mark for a quick review!
+
+2010-07-23 13:26 +0000 [r278873]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* Makefile, agi/Makefile, sounds/Makefile: Portability updates for
+	  Makefiles. When possible, use $(INSTALL). This allows us to use
+	  the functionality within install for setting directory / file
+	  permissions, a requirement for unprivileged installation. Also
+	  move any directory we plan to create within the installdirs
+	  macro. Plus various other formatting issues. (issue #17436)
+	  Reported by: pabelanger Patches: non-root.patch.v8 uploaded by
+	  pabelanger (license 224) Tested by: pabelanger Review:
+	  https://reviewboard.asterisk.org/r/654/
+
+2010-07-23 11:01 +0000 [r278809-278841]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c: missed FXS kewl
+	  start polarityswitch when finally on hook. (issue #17318)
+
+	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
+	  channels/sig_analog.c, channels/sig_analog.h: Support FXS module
+	  Polarity Reversal on remote party Answer and Hangup FXS lines
+	  normally connect to a telephone. However, when FXS lines are
+	  routed to an external PBX or Key System to act as "external" or
+	  "CO" lines, it is extremely difficult, if not impossible for the
+	  external PBX to know when the call has been disconnected without
+	  receiving a polarity reversal on the line. Now using
+	  answeronpolarityswitch and hanguponpolarityswitch keywords that
+	  previously were used only for FXO ports, now applies like
+	  functionality for an FXS port, but from the connected equipment's
+	  point of view. (closes issue #17318) Reported by: armeniki
+	  Patches: fxs_linepolarity.diff5.txt uploaded by alecdavis
+	  (license 585) Tested by: alecdavis Review:
+	  https://reviewboard.asterisk.org/r/797/
+
+2010-07-22 21:16 +0000 [r278777]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: DNID not cleared when channel hang up
+	  (Affects PRI and SS7) The "dahdi show channels" CLI command still
+	  reports the DNID of the previous call even if the call is already
+	  hang up. The "dahdi show channels" command of older releases
+	  clear the DNID once the channel is hang up. Regression from the
+	  sig_analog/sig_pri extraction from chan_dahdi. (closes issue
+	  #17623) Reported by: klaus3000 Patches: issue17623.patch uploaded
+	  by rmudgett (license 664) Tested by: rmudgett
+
+2010-07-22 19:45 +0000 [r278708]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/xmldoc.c: Add method for finding XML doc files for systems
+	  that don't support GLOB_BRACE. In particular, Solaris and perhaps
+	  others do not support the above mentioned GNU extension. In this
+	  case the paths are simply expanded without the braces and the
+	  calls to glob are made separately. Note: I could not explain
+	  memory allocation failures that were being reported from within
+	  libxml itself when making calls to glob without using
+	  GLOB_NOCHECK. This is the only reason why that flag is being
+	  used. (closes issue #15402) Reported by: snuffy Patches:
+	  bug_xmlpatt-v3.diff uploaded by snuffy (license 35), modified by
+	  me
+
+2010-07-22 14:58 +0000 [r278620]  Mark Michelson <mmichelson at digium.com>
+
+	* main/channel.c, /: Merged revisions 278618 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul
+	  2010) | 13 lines Allow PLC to function properly when channels use
+	  SLIN for audio. If a channel involved in a bridge was using SLIN
+	  audio, then translation paths were not guaranteed to be set up
+	  properly since in all likelihood the number of translation steps
+	  was only 1. This patch enforces the transcode_via_slin behavior
+	  if transcode_via_slin or generic_plc is enabled and one of the
+	  formats to make compatible is SLIN. AST-352 ........
+
+2010-07-22 14:56 +0000 [r278619]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: update sip subscription debug message to a
+	  warning message If the Expire header of a SUBSCRIBE is less that
+	  our expiremin, a log warning will be displayed.
+
+2010-07-22 05:29 +0000 [r278579]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/doxyref.h: Add the full current set of CDR
+	  drivers
+
+2010-07-21 19:16 +0000 [r278539]  David Vossel <dvossel at digium.com>
+
+	* tests/test_func_file.c: make func_file unit test's category
+	  consistent with other tests
+
+2010-07-21 19:11 +0000 [r278538]  Terry Wilson <twilson at digium.com>
+
+	* channels/iax2-parser.h, include/asterisk/crypto.h,
+	  main/aescrypt.c (removed), include/asterisk/aes_internal.h
+	  (removed), funcs/func_aes.c, res/res_crypto.c, main/aestab.c
+	  (removed), main/aesopt.h (removed), include/asterisk/aes.h
+	  (removed), main/aeskey.c (removed), pbx/pbx_dundi.c,
+	  channels/chan_iax2.c, res/res_crypto.exports.in,
+	  pbx/dundi-parser.h: Remove built-in AES code and use optional_api
+	  instead Review: https://reviewboard.asterisk.org/r/793/
+
+2010-07-21 18:52 +0000 [r278536]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: send "423 Interval too small" Response to
+	  Subscribe with Expires less that min allowed [RFC3265]3.1.6.1....
+	  The notifier MAY also check that the duration in the "Expires"
+	  header is not too small. If and only if the expiration interval
+	  is greater than zero AND smaller than one hour AND less than a
+	  notifier- configured minimum, the notifier MAY return a "423
+	  Interval too small" error which contains a "Min-Expires" header
+	  field. The "Min- Expires" header field is described in SIP [1].
+
+2010-07-21 17:44 +0000 [r278501]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c: Fix invalid test
+	  for rxisoffhook in FXO channels This fixes some cases of no
+	  outgoing calls on FXO before an incoming call. Remove an
+	  unnecessary testing of an "off-hook" bit from DAHDI for FXO
+	  (KS/GS) channels.In some cases the bit would not be initialized
+	  properly before the first inbound call and thus prevent an
+	  outgoing call. If those tests are actually required by anybody,
+	  they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c
+	  . (closes issue #14577) Reported by: jkroon Patches:
+	  asterisk_chan_dahdi_hookstate_fix_trunk_new.diff uploaded by
+	  frawd (license 610) Tested by: frawd Review:
+	  https://reviewboard.asterisk.org/r/699/
+
+2010-07-21 16:15 +0000 [r278465]  Russell Bryant <russell at digium.com>
+
+	* res/res_timing_pthread.c: Use poll() instead of select() in
+	  res_timing_pthread to avoid stack corruption. This code did not
+	  properly check FD_SETSIZE to ensure that it did not try to
+	  select() on fds that were too large. Switching to poll() removes
+	  the limitation on the maximum fd value. (closes issue #15915)
+	  Reported by: keiron (closes issue #17187) Reported by: Eddie
+	  Edwards (closes issue #16494) Reported by: Hubguru (closes issue
+	  #15731) Reported by: flop (closes issue #12917) Reported by:
+	  falves11 (closes issue #14920) Reported by: vrban (closes issue
+	  #17199) Reported by: aleksey2000 (closes issue #15406) Reported
+	  by: kowalma (closes issue #17438) Reported by: dcabot (closes
+	  issue #17325) Reported by: glwgoes (closes issue #17118) Reported
+	  by: erikje possibly other issues, too ...
+
+2010-07-21 15:56 +0000 [r278463]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_meetme.c: Ensure realtime conferences are treated the
+	  same as static conferences when trying to find an empty one.
+	  Also, parse the useropts properly, when retrieving from realtime,
+	  and add them to the existing flags. (closes issue #17502)
+	  Reported by: kenji Patches: 20100720__issue17502.diff.txt
+	  uploaded by tilghman (license 14) Tested by: kenji
+
+2010-07-21 15:54 +0000 [r278426-278462]  Matthew Nicholson <mnicholson at digium.com>
+
+	* res/res_fax_spandsp.c: Properly show the current page being
+	  transfered for 'fax show session'
+
+	* channels/chan_sip.c: Properly set the port number for UDPTL media
+	  sessions.
+
+	* res/res_fax.c: Don't print failure status when the remote end
+	  hangs up, it may not be an actual failure.
+
+2010-07-21 13:02 +0000 [r278425]  Russell Bryant <russell at digium.com>
+
+	* main/features.c, UPGRADE.txt, configs/features.conf.sample:
+	  Update documentation for 'comebacktoorigin' in featuers.conf. The
+	  documentation for this option did not match the code. Fix that
+	  along with some minor cleanups to the code along the way.
+	  Document a slight change in behavior (to something that was
+	  previously undocumented) in UPGRADE.txt.
+
+2010-07-21 06:45 +0000 [r278393]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_iax2.c: Change order so that it more closely
+	  matches the related SIP command. (closes issue #17648) Reported
+	  by: GMLudo Review: https://reviewboard.asterisk.org/r/789/
+
+2010-07-21 03:53 +0000 [r278361]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: include stat.h for everybody, needed for
+	  device2chan
+
+2010-07-20 23:23 +0000 [r278275-278307]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_config_pgsql.c, main/logger.c, CHANGES,
+	  contrib/realtime/mysql/queue_log.sql (added),
+	  configs/logger.conf.sample: Separate queue_log arguments into
+	  separate fields, and allow the text file to be used, even when
+	  realtime is used. (closes issue #17082) Reported by: coolmig
+	  Patches: 20100720__issue17082.diff.txt uploaded by tilghman
+	  (license 14) Tested by: coolmig
+
+	* /, apps/app_voicemail.c: Merged revisions 278261 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20
+	  Jul 2010) | 7 lines Delete IMAP messages in reverse order, to
+	  ensure reordering after each expunge does not cause deletion of
+	  the wrong message. (closes issue #16350) Reported by: noahisaac
+	  Patches: 20100623__issue16350.diff.txt uploaded by tilghman
+	  (license 14) ........
+
+2010-07-20 22:38 +0000 [r278274]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c: Reference correct struct member for unlikely
+	  event PRI_EVENT_CONFIG_ERR.
+
+2010-07-20 22:26 +0000 [r278272]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/autoservice.c, /, main/features.c,
+	  include/asterisk/channel.h: Merged revisions 278167 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20
+	  Jul 2010) | 4 lines Do not queue up DTMF frames while a call is
+	  on hold. (Fixes ABE-2110) ........
+
+2010-07-20 21:41 +0000 [r278234]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: fixes sip CANCEL race condition If Asterisk
+	  sends a 4xx error and the other side sends a CANCEl before
+	  receiving the 4xx and responding with the ACK, Asterisk will
+	  process the CANCEL and send a 487 Request Terminated as a new
+	  final response to the INVITE. Since we are issuing a new final
+	  response to the INVITE, the old one must be pretend_acked else it
+	  will keep retransmitting.
+
+2010-07-20 21:01 +0000 [r278168]  Matthew Nicholson <mnicholson at digium.com>
+
+	* res/res_fax.c: This commit contains several changes to the way
+	  output channel variables are handled. FAX output channel
+	  variables will now match the values reported by FAXOPT() and
+	  should be set in all failure and success cases. This commit also
+	  contains a few modifications to the way FAXOPT() variables are
+	  populated in a few spots and fixes for some reference count leaks
+	  of the session details structure in some failure cases. Also
+	  found and fixed more cases where FAXOPT(status) may not have
+	  gotten set. FAX-214 FAX-203
+
+2010-07-20 19:35 +0000 [r278132]  Tilghman Lesher <tlesher at digium.com>
+
+	* cel/cel_pgsql.c, cdr/cdr_sqlite3_custom.c, channels/chan_local.c,
+	  res/res_timing_dahdi.c, cdr/cdr_adaptive_odbc.c,
+	  res/res_calendar_caldav.c, formats/format_sln16.c,
+	  formats/format_wav_gsm.c, channels/chan_iax2.c, main/config.c,
+	  main/loader.c, res/res_rtp_multicast.c, channels/chan_dahdi.c,
+	  res/res_smdi.c, channels/chan_skinny.c,
+	  include/asterisk/module.h, formats/format_pcm.c,
+	  channels/chan_alsa.c, formats/format_h263.c, res/res_curl.c,
+	  cdr/cdr_odbc.c, formats/format_jpeg.c, res/res_speech.c,
+	  formats/format_gsm.c, cdr/cdr_manager.c, formats/format_g719.c,
+	  res/res_calendar_exchange.c, cel/cel_tds.c, formats/format_wav.c,
+	  channels/chan_bridge.c, channels/chan_agent.c,
+	  formats/format_ogg_vorbis.c, res/res_monitor.c,
+	  res/res_calendar_ews.c, res/res_config_curl.c,
+	  channels/chan_misdn.c, funcs/func_curl.c,
+	  res/res_timing_kqueue.c, formats/format_g726.c, main/asterisk.c,
+	  res/res_odbc.c, cel/cel_adaptive_odbc.c, res/res_calendar.c,
+	  cel/cel_radius.c, channels/chan_multicast_rtp.c,
+	  apps/app_meetme.c, formats/format_sln.c, res/res_musiconhold.c,
+	  channels/chan_gtalk.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
+	  res/res_jabber.c, res/res_config_sqlite.c,
+	  formats/format_siren7.c, cdr/cdr_csv.c, formats/format_ilbc.c,
+	  res/res_config_odbc.c, cel/cel_manager.c, cel/cel_custom.c,
+	  cdr/cdr_sqlite.c, res/res_agi.c, res/res_timing_timerfd.c,
+	  apps/app_confbridge.c, formats/format_h264.c,
+	  res/res_config_ldap.c, addons/chan_mobile.c,
+	  formats/format_siren14.c, cdr/cdr_custom.c, channels/chan_mgcp.c,
+	  res/res_rtp_asterisk.c, res/res_config_pgsql.c,
+	  res/res_calendar_icalendar.c, channels/chan_sip.c,
+	  cdr/cdr_syslog.c, res/res_fax.c, res/res_crypto.c,
+	  res/res_adsi.c, include/asterisk/config.h, pbx/pbx_lua.c,
+	  channels/chan_console.c, apps/app_queue.c, cdr/cdr_tds.c,
+	  res/res_srtp.c, channels/chan_jingle.c, formats/format_vox.c,
+	  res/res_timing_pthread.c, channels/chan_h323.c,
+	  cel/cel_sqlite3_custom.c, formats/format_g723.c,
+	  funcs/func_devstate.c, formats/format_g729.c,
+	  addons/res_config_mysql.c: Add load priority order, such that
+	  preload becomes unnecessary in most cases
+
+2010-07-20 18:11 +0000 [r278051-278096]  Russell Bryant <russell at digium.com>
+
+	* contrib/scripts/install_prereq: Add a package to install_prereq.
+
+	* channels/chan_local.c: Only call ast_channel_cc_params_init() if
+	  allocating a channel succeeds.
+
+2010-07-20 16:50 +0000 [r278024]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/manager.c, /: Merged revisions 278023 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010)
+	  | 7 lines Off-by-one error (closes issue #16506) Reported by:
+	  nik600 Patches: 20100629__issue16506.diff.txt uploaded by
+	  tilghman (license 14) ........
+
+2010-07-19 21:07 +0000 [r277945]  Jean Galarneau <jgalarneau at digium.com>
+
+	* /, main/features.c: Merged revisions 277906 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) |
+	  7 lines Avoid trying to pickup a parked extension before the park
+	  operation is completed. A crash could occur if the extension is
+	  picked up while the parking extension is being announced. Testing
+	  pu->notquiteyet while searching for a parked extension resolves
+	  this crash. (ABE-2418) ........
+
+2010-07-19 17:16 +0000 [r277872-277873]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c, configs/sip.conf.sample,
+	  channels/sip/include/sip.h: Fix port setting of external address
+	  in SIP. There are two changes here: 1. Since the externip setting
+	  can now have a port attached to it, calling it "externip" is
+	  misleading. The option is now documented and parsed as
+	  "externaddr." This also extends to the "matchexterniplocally"
+	  setting. It is now documented and parsed as
+	  "matchexternaddrlocally." The old names for the options may still
+	  be used, but they are no longer used in the sip.conf.sample file.
+	  2. If no port is set for the externaddr, and UDP is the transport
+	  to be used, then we will set the port of the externaddr to that
+	  of the udpbindaddr. This was how things worked prior to the IPv6
+	  merge, so this is a regression fix. (closes issue #17665)
+	  Reported by: mmichelson Patches: 17665.diff#2 uploaded by
+	  pprindeville (license 347) Tested by: pprindeville
+
+	* tests/test_acl.c: Remove the fe80:1234::1234 test case from
+	  test_acl.c The ACL test was failing on Mac OS X because it would
+	  convert the above invalid link-local address into fe80::1234
+	  while reporting no error from getaddrinfo(). Linux does not do
+	  this.
+
+2010-07-19 14:39 +0000 [r277837]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c,
+	  channels/sig_analog.h: Fix regression with distinctive ring
+	  detection. The issue here is that passing an array to a function
+	  prohibits the ARRAY_LEN macro from returning the real size. To
+	  avoid this the size is now defined and use of ARRAY_LEN is
+	  avoided. (closes issue #15718) Reported by: alecdavis Patches:
+	  bug15718.patch uploaded by jpeeler (license 325)
+
+2010-07-19 14:17 +0000 [r277814]  Mark Michelson <mmichelson at digium.com>
+
+	* include/asterisk/acl.h, main/netsock2.c, main/manager.c,
+	  channels/chan_sip.c, channels/chan_skinny.c, tests/test_acl.c,
+	  main/acl.c, include/asterisk/netsock2.h, configs/sip.conf.sample,
+	  channels/chan_iax2.c: Make ACLs IPv6-capable. ACLs can now be
+	  configured to match IPv6 networks. This is only relevant for ACLs
+	  in chan_sip for now since other channel drivers do not support
+	  IPv6 addressing. However, once those channel drivers are
+	  outfitted to support IPv6 addressing, the ACLs will already be
+	  ready for IPv6 support. https://reviewboard.asterisk.org/r/791
+
+2010-07-17 17:42 +0000 [r277773-277775]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, autoconf/ast_func_fork.m4, configure,
+	  include/asterisk/autoconfig.h.in: Merged revisions 277738 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010)
+	  | 5 lines Remove uclibc cross-compile triplet, as uclibc has a
+	  working fork()... it's only uclinux that does not. (closes issue
+	  #17616) Reported by: pprindeville ........
+
+	* res/res_config_pgsql.c, res/res_config_odbc.c, /,
+	  include/asterisk/config.h, main/config.c,
+	  addons/res_config_mysql.c: Merged revisions 277568 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16
+	  Jul 2010) | 8 lines Since we split values at the semicolon, we
+	  should store values with a semicolon as an encoded value. (closes
+	  issue #17369) Reported by: gkservice Patches:
+	  20100625__issue17369.diff.txt uploaded by tilghman (license 14)
+	  Tested by: tilghman ........
+
+2010-07-17 13:10 +0000 [r277703]  Russell Bryant <russell at digium.com>
+
+	* Makefile, configure, include/asterisk/autoconfig.h.in,
+	  configure.ac, makeopts.in: Allow xmllint to be used for XML docs
+	  validation. xmllint seems to be more commonly available since it
+	  comes with libxml2.
+
+2010-07-17 00:03 +0000 [r277667]  Bradley Latus <brad.latus at gmail.com>
+
+	* res/res_fax.c: Update res_fax.c to be a good xml citizen. (closes
+	  issues #17667) Reported by: snuffy
+
+2010-07-16 23:23 +0000 [r277657]  Tim Ringenbach <tim.ringenbach at gmail.com>
+
+	* main/features.c: Merged revisions 277625 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul
+	  2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on
+	  attended transfer. ast_bridge_call() clears
+	  AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
+	  ast_bridge_call() is called for a second bridge on the same
+	  channel, and it clears that flag, which still needs to get set
+	  for when the original ast_bridge_call() gets control back and
+	  checks it. Review: https://reviewboard.asterisk.org/r/741
+	  ........
+
+2010-07-16 21:24 +0000 [r277530]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 277497 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul
+	  2010) | 4 lines Default to no udptl error correction so that
+	  error correction will be disabled in the event that the remote
+	  end indicates that they do not support the error correction mode
+	  we requested. FAX-128 ........
+
+2010-07-16 21:16 +0000 [r277488]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_queue.c: Fix reporting estimated queue hold time. Just
+	  say the number of seconds (after minutes) rather than doing some
+	  incorrect calculation with respect to minutes. (closes issue
+	  #17498) Reported by: corruptor Patches: holdesecs_bug.diff
+	  uploaded by corruptor (license 253)
+
+2010-07-16 20:35 +0000 [r277484]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/sched.h, main/sched.c: Finally, a method that
+	  really fixes the assertions in chan_iax2.c related to cancelling
+	  lagid. No, replacing usleep(1) with sched_yield() did not have an
+	  effect.
+
+2010-07-16 20:27 +0000 [r277467]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 277419 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16
+	  Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when
+	  reloading dahdi module During a reload, the priexclusive and
+	  outsignalling parameters are not read in from the config file as
+	  intended. Unfortunately, they get set to defaults as a result.
+	  This patch makes sure that they do not get set to defaults during
+	  a reload. (closes issue #17441) Reported by: mtryfoss Patches:
+	  issue17441_v1.4.patch uploaded by rmudgett (license 664)
+	  issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
+	  issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
+	  by: rmudgett ........
+
+2010-07-16 20:25 +0000 [r277452]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
+	  (added): Add documentation for MOH realtime fields
+
+2010-07-16 19:32 +0000 [r277409]  Matthew Nicholson <mnicholson at digium.com>
+
+	* tests/test_devicestate.c: updated devicestate test for device
+	  state changes
+
+2010-07-16 19:22 +0000 [r277366]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_queue.c: Add missing handling for ringing state for use
+	  with queue empty options. (closes issue #17471) Reported by:
+	  jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056)
+
+2010-07-16 18:31 +0000 [r277331]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c, /: Merged revisions 277327 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul
+	  2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as
+	  extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
+	  Reported by: francesco_r Patches: pbx.c.patch uploaded by
+	  viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar
+	  ........
+
+2010-07-16 18:14 +0000 [r277263]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/manager.c, /: Merged revisions 277261 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010)
+	  | 5 lines If variable gotten is not set, will segfault on
+	  Solaris. (closes issue #17636) Reported by: bklang ........
+
+2010-07-16 18:05 +0000 [r277250-277262]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/channel.c: Print f->subclass.integer instead of f->subclass.
+	  (fix build breakage introduced in r277250)
+
+	* main/channel.c, /: Merged revisions 277247 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul
+	  2010) | 4 lines For pass through DTMF tones, measure the actual
+	  duration between the begin and end packets on the wire. If it is
+	  detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
+	  emulation. AST-362 ........
+
+2010-07-16 17:13 +0000 [r277183]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* /, apps/app_amd.c: Merged revisions 277182 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul
+	  2010) | 8 lines Total analysis time error with SIP and silence
+	  suppression When using app_amd with SIP providers that have
+	  silence suppression on, the iTotalTime count increases
+	  exponentially. (closes issue #17656) Reported by: juls ........
+
+2010-07-16 16:25 +0000 [r277175]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/sip/reqresp_parser.c: Fix up some weird indentation
+	  problems in reqresp_parser.c
+
+2010-07-16 15:20 +0000 [r277143]  Sean Bright <sean at malleable.com>
+
+	* main/translate.c: Avoid crashing when installing a duplicate
+	  translation path with a lower cost. (closes issue #17092)
+	  Reported by: moy Patches: translate.rev254273.patch uploaded by
+	  moy (license 222) Tested by: moy
+
+2010-07-16 13:40 +0000 [r277103]  Eliel C. Sardanons <eliels at gmail.com>
+
+	* CREDITS: Add Despegar.com (my main sponsor) to the CREDITS file.
+
+2010-07-16 13:32 +0000 [r276950-277102]  Olle Johansson <oej at edvina.net>
+
+	* main/dnsmgr.c, main/srv.c: Formatting changes
+
+	* channels/chan_sip.c: Formatting fixes
+
+	* configs/sip.conf.sample: Clarify syntax changes
+
+	* CREDITS: Adding a few more to the list of CREDITS
+
+	* channels/chan_sip.c: Formatting changes (guideline corrections)
+	  Found a unused bag of curly brackets under my table. I always
+	  wondered where they had gone. They where indeed needed in
+	  chan_sip.c
+
+	* CREDITS: Adding a few more credits
+
+	* channels/chan_sip.c, doc/tex/channelvariables.tex,
+	  configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Add
+	  ability to configure the Max-Forwards header in the dialplan, as
+	  well as in sip.conf configuration for the channel and for
+	  devices. The Max-Forwards header is used to prevent loops in a
+	  SIP network. Each intermediary, like SIP proxys and SBCs,
+	  decrement this counter and detects when it reaches zero, at which
+	  point the SIP request is nicely killed in a SIP-friendly way.
+	  Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel
+	  for the review and good advice.
+
+	* CHANGES, apps/app_queue.c: Add a dialplan function to check if a
+	  queue exists: QUEUE_EXISTS Review:
+	  https://reviewboard.asterisk.org/r/777/
+
+2010-07-16 06:04 +0000 [r276910-276911]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_jabber.c: And yet one more
+
+	* res/res_jabber.c: "Item may be used uninitialized in this
+	  function."
+
+2010-07-16 05:42 +0000 [r276909]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Fix reversed logic of if statement. Found
+	  based on message from Philip Prindeville on the Asterisk
+	  Developers mailing list.
+
+2010-07-16 05:38 +0000 [r276830-276908]  Tilghman Lesher <tlesher at digium.com>
+
+	* configure, configure.ac: Detect the --dynamic-list flag a bit
+	  better
+
+	* configure, main/Makefile, configure.ac, makeopts.in: Fix build on
+	  FreeBSD
+
+	* tests/test_utils.c: Fix trunk build for Mac OS X 10.6
+
+	* contrib/realtime/mysql/iaxfriends.sql,
+	  contrib/realtime/mysql/meetme.sql,
+	  contrib/realtime/postgresql/realtime.sql,
+	  contrib/realtime/mysql/sipfriends.sql: Allow ipaddress to contain
+	  the maximum IPv6 address. Also, update meetme to the full list of
+	  supported fields.
+
+	* configure, autoconf/ast_gcc_attribute.m4: Quote AC_SUBST within
+	  m4_ifval, so it does not get prematurely expanded. (closes issue
+	  #17654) Reported by: pprindeville Patches: issue17654.diff
+	  uploaded by qwell (license 4) Tested by: qwell, pprindeville
+
+2010-07-15 20:21 +0000 [r276788]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_sip.c: Correct not setting the bindport before
+	  attempting to open the socket. Related to changes from 276571, I
+	  was accidentally testing with a port set in my configuration
+	  causing me to miss this. Also moved the TCP handling as well to
+	  occur before build_peer is called.
+
+2010-07-15 19:46 +0000 [r276731-276769]  Tilghman Lesher <tlesher at digium.com>
+
+	* configure, include/asterisk/autoconfig.h.in,
+	  include/asterisk/compat.h, configure.ac: Define LLONG_MAX on
+	  systems that do not have it. (closes issue #17644) Reported by:
+	  pprindeville
+
+	* configure, main/Makefile, autoconf/ast_gcc_attribute.m4,
+	  configure.ac, makeopts.in: Fix linking asterisk on CentOS 5,
+	  which is using gcc 4.1.1. Gcc 4.1.2 has the real fix. Review:
+	  https://reviewboard.asterisk.org/r/790/
+
+2010-07-15 13:51 +0000 [r276653]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c, /: Merged revisions 276652 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010)
+	  | 2 lines In a perfect world, the frame source would never be
+	  NULL. In the meantime, don't crash when it is. ........
+
+2010-07-15 12:21 +0000 [r276616]  Russell Bryant <russell at digium.com>
+
+	* contrib/scripts/install_prereq: Add lua5.1 to the handy dandy
+	  list of packages.
+
+2010-07-14 22:58 +0000 [r276571]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_sip.c: Fix MWI notification transmission problems
+	  over SIP. MWI updates were not being sent if no messages were
+	  found in the event cache. This was corrected since a phone may
+	  need to clear its MWI status configured previously from another
+	  mailbox. Upon module or sip reload, MWI updates could not be sent
+	  due to the sipsock socket not being set early enough in
+	  reload_config. The code handling the descriptor assignment and
+	  such has simply been moved before the call to build_peer. Issuing
+	  a sip reload cleared the IP address of the peer, but skipped
+	  checking the database for registration information. The database
+	  is now checked both for sip reload and actually reloading the
+	  module. If a transmission occurs before the do_monitor thread has
+	  started, do not attempt to send a signal to it. (closes issue
+	  #17398) Reported by: ip-rob
+
+2010-07-14 22:32 +0000 [r276570]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
+	  main/acl.c: Fix errors where incorrect address information was
+	  printed. ast_sockaddr_stringiy_fmt (which is call by all
+	  ast_sockaddr_stringify* functions) uses thread-local storage for
+	  storing the string that it creates. In cases where
+	  ast_sockaddr_stringify_fmt was being called twice within the same
+	  statement, the result of one call would be overwritten by the
+	  result of the other call. This usually was happening in
+	  printf-like statements and was resulting in the same stringified
+	  addressed being printed twice instead of two separate addresses.
+	  I have fixed this by using ast_strdupa on the result of stringify
+	  functions if they are used twice within the same statement. As
+	  far as I could tell, there were no instances where a pointer to
+	  the result of such a call were saved anywhere, so this is the
+	  only situation I could see where this error could occur.
+
+2010-07-14 21:29 +0000 [r276531]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_h323.c: Make compile again.
+
+2010-07-14 21:11 +0000 [r276490-276493]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/loader.c: Oops, merge reverted this fix.
+
+	* include/asterisk/adsi.h, include/asterisk/agi.h,
+	  include/asterisk/crypto.h, main/asterisk.dynamics, main/Makefile,
+	  tests/test_utils.c, main/adsistub.c (removed), main/cryptostub.c
+	  (removed), res/res_adsi.c, res/res_crypto.c,
+	  res/res_crypto.exports.in (added), res/res_adsi.exports.in,
+	  main/loader.c, include/asterisk/optional_api.h: Remove the old
+	  stub files, preferring the optional_api method. (closes issue
+	  #17475) Reported by: tilghman Review:
+	  https://reviewboard.asterisk.org/r/695/
+
+2010-07-14 20:15 +0000 [r276441]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/loader.c: Don't try to call an embedded module's
+	  backup_globals() function until after confirming it exists.
+
+2010-07-14 19:51 +0000 [r276439]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: handle special case were "200 Ok" to pending
+	  INVITE never receives ACK Unlike most responses, the 200 Ok to a
+	  pending INVITE Request is acknowledged by an ACK Request. If the
+	  ACK Request for this Response is not received the previous
+	  behavior was to immediately destroy the dialog and hangup the
+	  channel. Now in an effort to be more RFC compliant, instead of
+	  immediately destroying the dialog during this special case,
+	  termination is done with a BYE Request as the dialog is
+	  technically confirmed when the 200 Ok is sent even if the ACK is
+	  never received. The behavior of immediately hanging up the
+	  channel remains. This only affects how dialog termination
+	  proceeds for this one special case. RFC 3261 section 13.3.1.4 "If
+	  the server retransmits the 2xx response for 64*T1 seconds without
+	  receiving an ACK, the dialog is confirmed, but the session SHOULD
+	  be terminated. This is accomplished with a BYE, as described in
+	  Section 15."
+
+2010-07-14 16:58 +0000 [r276393]  Richard Mudgett <rmudgett at digium.com>
+

[... 21191 lines stripped ...]



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