[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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--- 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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