[svn-commits] lmadsen: tag 1.4.35-rc1 r278698 - /tags/1.4.35-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jul 22 14:27:59 CDT 2010
Author: lmadsen
Date: Thu Jul 22 14:27:54 2010
New Revision: 278698
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=278698
Log:
Importing files for 1.4.35-rc1 release.
Added:
tags/1.4.35-rc1/.lastclean (with props)
tags/1.4.35-rc1/.version (with props)
tags/1.4.35-rc1/ChangeLog (with props)
Added: tags/1.4.35-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.4.35-rc1/.lastclean?view=auto&rev=278698
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--- tags/1.4.35-rc1/ChangeLog (added)
+++ tags/1.4.35-rc1/ChangeLog Thu Jul 22 14:27:54 2010
@@ -1,0 +1,29429 @@
+2010-07-22 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.35-rc1 Released.
+
+2010-07-22 14:55 +0000 [r278618] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: 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
+
+2010-07-20 22:23 +0000 [r278023-278261] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: 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)
+
+ * main/autoservice.c, res/res_features.c,
+ include/asterisk/channel.h: Do not queue up DTMF frames while a
+ call is on hold. (Fixes ABE-2110)
+
+ * main/manager.c: Off-by-one error (closes issue #16506) Reported
+ by: nik600 Patches: 20100629__issue16506.diff.txt uploaded by
+ tilghman (license 14)
+
+2010-07-19 20:56 +0000 [r277944] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_sip.c: Regression with T.38 negotiation Prior to
+ 1.4.26.3 T.38 negotiation worked properly, in the case of the
+ reporter. (issue #16852) Reported by: cfc (closes issue #16705)
+ Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
+ by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
+ samdell3 Review: https://reviewboard.asterisk.org/r/754/
+
+2010-07-19 20:16 +0000 [r277906] Jean Galarneau <jgalarneau at digium.com>
+
+ * res/res_features.c: 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)
+
+2010-07-17 16:59 +0000 [r277738] Tilghman Lesher <tlesher at digium.com>
+
+ * autoconf/ast_func_fork.m4, configure: Remove uclibc cross-compile
+ triplet, as uclibc has a working fork()... it's only uclinux that
+ does not. (closes issue #17616) Reported by: pprindeville
+
+2010-07-16 22:43 +0000 [r277625] Tim Ringenbach <tim.ringenbach at gmail.com>
+
+ * res/res_features.c: 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 21:54 +0000 [r277568] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c: 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
+
+2010-07-16 21:18 +0000 [r277497] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: 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 20:18 +0000 [r277419] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: 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 18:30 +0000 [r277327] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c: 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 18:04 +0000 [r277261] Tilghman Lesher <tlesher at digium.com>
+
+ * main/manager.c: If variable gotten is not set, will segfault on
+ Solaris. (closes issue #17636) Reported by: bklang
+
+2010-07-16 17:29 +0000 [r277247] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c: 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 17:10 +0000 [r277182] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * apps/app_amd.c: 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
+
+2010-07-15 13:48 +0000 [r276652] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c: In a perfect world, the frame source would never
+ be NULL. In the meantime, don't crash when it is.
+
+2010-07-14 11:49 +0000 [r276267] Leif Madsen <lmadsen at digium.com>
+
+ * configs/voicemail.conf.sample: Update documentation for
+ voicemail.conf externpass option.
+
+2010-07-13 19:14 +0000 [r275994-276126] Russell Bryant <russell at digium.com>
+
+ * res/res_features.c: Only reset a CDR that exists.
+
+ * res/res_features.c: Use chan->cdr instead of chan_cdr (just like
+ peer->cdr instead of peer_cdr in the last commit).
+
+ * res/res_features.c: Access peer->cdr directly instead of through
+ a saved off reference. At this point in the code, it is possible
+ that peer_cdr may be invalid. Specifically, in the blind transfer
+ code, CDRs are swapped between channels. So, peer_cdr is no
+ longer == peer->cdr. The scenario that exposed a crash in this
+ code was a blind transfer that hit the system call limit, causing
+ the transferee channel to get destroyed after the transfer
+ attempt failed. Even if it succeeds and this code doesn't crash,
+ this code was still trying to reset a CDR on a channel that was
+ now owned by a different thread, which is a BadThing(tm).
+ (ABE-2417)
+
+2010-07-13 14:47 +0000 [r275909] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/realtime/mysql/sipfriends.sql,
+ contrib/realtime/mysql/voicemail.sql,
+ contrib/scripts/realtime_pgsql.sql (removed),
+ contrib/scripts/vmdb.sql (removed),
+ contrib/scripts/iax-friends.sql (removed),
+ contrib/realtime/mysql/iaxfriends.sql,
+ contrib/realtime/mysql/meetme.sql, contrib/scripts/meetme.sql
+ (removed), contrib/realtime (added), contrib/realtime/postgresql,
+ contrib/realtime/postgresql/realtime.sql, contrib/realtime/mysql,
+ contrib/realtime/oracle, contrib/realtime/sqlserver,
+ contrib/scripts/sip-friends.sql (removed): Move SQL scripts into
+ their own database-specific directories.
+
+2010-07-12 20:34 +0000 [r275665-275773] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_meetme.c: Make user removals and traversals thread safe
+ in meetme. Race conditions present in meetme involving the user
+ list where a lack of locking has the potential for a user to be
+ removed during a traversal or as in the case of the reporter
+ after checking if the list is empty could cause a crash. Fixing
+ this was done by convering the userlist to an ao2 container.
+ (closes issue #17390) Reported by: Vince Review:
+ https://reviewboard.asterisk.org/r/746/
+
+ * main/channel.c: Change ast_write to not stop generator when
+ called from ast_prod. For SIP channels configured with the
+ progressinband option on, the ringback was being immediately
+ stopped. This problem was due to ast_prod being moved for a
+ deadlock fix in 259858. Prodding the channel after setting up the
+ generator triggered the check in ast_write to stop the generator.
+ The fix here should write the frame the same as was done before
+ the call to ast_prod was moved. (closes issue #17372) Reported
+ by: tech_admin
+
+2010-07-09 19:28 +0000 [r275241-275290] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * main/cli.c: fix tab-completion for unload command. (closes issue
+ #17536) Reported by: junky Patches: unload_vs_mod_unload.diff
+ uploaded by junky (license 177) Tested by: pabelanger
+
+ * channels/chan_sip.c: Fix logging message for stale nonce. (closes
+ issue #17582) Reported by: kenner Patches: chan_sip.c.diff
+ uploaded by kenner (license 1040) Tested by: lmadsen
+
+2010-07-09 18:23 +0000 [r275027-275182] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/loader.c: give a better error message when attempting to
+ unload a module that is not loaded
+
+ * main/loader.c: don't unload modules that returned
+ AST_MODULE_LOAD_DECLINE when they were loaded
+
+ * apps/app_dial.c: Clear the AST_CDR_FLAG_DIALED flag for channels
+ going into the pbx via the G option in app_dial (closes issue
+ #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: jamicque,
+ mnicholson
+
+2010-07-09 15:33 +0000 [r275021] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/test.h, main/test.c: Document that a leading and
+ trailing slash is expected for test categories. Also, emit a
+ warning if a test is registered without one of these.
+
+2010-07-07 18:12 +0000 [r274579] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Close the DAHDI FD on error when
+ processing chan_dahdi toneduration config parameter.
+
+2010-07-07 06:13 +0000 [r274417] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/say.conf.sample: Correct how 100, 200, 300, etc. is said.
+ Also add the crazy British numbers. (closes issue #16102)
+ Reported by: Delvar Patches: say.conf.fix.patch uploaded by
+ Delvar (license 908) (plus a few additional fixes and
+ simplifications by me)
+
+2010-07-06 22:46 +0000 [r274283-274359] Jeff Peeler <jpeeler at digium.com>
+
+ * main/Makefile: Ensure file.o is built correctly. (related to
+ issue #15250)
+
+ * configs/sip.conf.sample: Correct sip.conf.sample comments for
+ prematuremedia option. (closes issue #17513) Reported by: festr
+ Patches: patch uploaded by festr (license 443)
+
+2010-07-06 22:08 +0000 [r274280] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Add option to not
+ do a call forward on 482 Loop Detected Asterisk has always set up
+ a forwarded call when receiving a 482 Loop Detected. This
+ prevents handling the call failure by just continuing on in the
+ dialplan. Since this would be a change in behavior, the new
+ option to disable this behavior is forwardloopdetected which
+ defaults to 'yes'. Review:
+ https://reviewboard.asterisk.org/r/764/
+
+2010-07-06 14:29 +0000 [r274157] Mark Michelson <mmichelson at digium.com>
+
+ * main/rtp.c: Fix problem with RFC 2833 DTMF not being accepted. A
+ recent check was added to ensure that we did not erroneously
+ detect duplicate DTMF when we received packets out of order. The
+ problem was that the check did not account for the fact that the
+ seqno of an RTP stream will roll over back to 0 after hitting
+ 65535. Now, we have a secondary check that will ensure that the
+ seqno rolling over will not cause us to stop accepting DTMF.
+ (closes issue #17571) Reported by: mdeneen Patches:
+ rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
+ Tested by: richardf, maxochoa, JJCinAZ
+
+2010-07-06 13:52 +0000 [r274093] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c: Make get_member_status return QUEUE_NO_MEMBERS
+ instead of QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work
+ again. This regression was introduced in 273639. Also fixed
+ whitespace.
+
+2010-07-05 19:48 +0000 [r273981] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_oss.c, channels/chan_iax2.c: Command 'stop
+ gracefully' doesn't.
+
+2010-07-05 13:51 +0000 [r273884] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * main/config.c: Remove extra line breaks from 'core show config
+ mappings' (closes issue #17583) Reported by: pabelanger Patches:
+ issue17583.patch uploaded by pabelanger (license 224) Tested by:
+ lmadsen
+
+2010-07-02 21:36 +0000 [r273717-273793] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_local.c, configure,
+ include/asterisk/autoconfig.h.in, channels/chan_agent.c,
+ configure.ac, channels/chan_h323.c, include/asterisk/lock.h,
+ include/asterisk/compiler.h: Have the DEADLOCK_AVOIDANCE macro
+ warn when an unlock fails, to help catch potentially large
+ software bugs. (closes issue #17407) Reported by: pdf Patches:
+ 20100527__issue17407.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/751/
+
+ * main/autoservice.c: Autoservice loop optimization causes a busy
+ loop, when channels are serviced while in hangup. (closes issue
+ #17564) Reported by: ramonpeek Patches:
+ 20100630__issue17564.diff.txt uploaded by tilghman (license 14)
+ Tested by: ramonpeek
+
+2010-07-02 15:54 +0000 [r273640] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * apps/app_voicemail.c, channels/chan_dahdi.c,
+ channels/chan_misdn.c, channels/chan_sip.c, res/res_agi.c,
+ res/res_jabber.c: Fix various typos, reported by Lintian
+
+2010-07-02 15:46 +0000 [r273639] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: If all members are paused, the wrong status is
+ indicated. (closes issue #17576) Reported by: ramonpeek Patches:
+ diff.txt uploaded by ramonpeek (license 266) Tested by: ramonpeek
+
+2010-07-01 22:09 +0000 [r273565] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Don't return a partially initialized datastore.
+ If memory allocation fails in ast_strdup(), don't return a
+ partially initialized datastore. Bad things may happen. (related
+ to ABE-2415)
+
+2010-07-01 20:19 +0000 [r273354-273474] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_meetme.c: Allow admin user to join conference without
+ using admin mode and no user pin. Configuring the conference in
+ meetme.conf like the following: conf => 2345,,6666 did not prompt
+ for pin when used without admin mode. This meant that the
+ conference could not be joined as an admin even if the user knew
+ the correct pin. The original bug report was submitted claiming
+ that the blank user pin should deny entry into the conference. I
+ think a better way to handle this would be with a feature
+ enhancement that used the following syntax: conf => 2345,X,6666 -
+ where X denotes no acceptable pin allowed (closes issue #15704)
+ Reported by: modelnine
+
+ * apps/app_meetme.c: Ensure channel placed in meetme in ringing
+ state is properly hung up. An outgoing channel placed in meetme
+ while still ringing which was then hung up would not exit meetme
+ and the channel was not properly destroyed. Specifically checking
+ for this scenario by looking at the appropriate control frames
+ resolves the issue. (closes issue #15871) Reported by: Ivan
+ Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
+ (license 229)
+
+2010-06-29 23:15 +0000 [r273057-273060] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Allow the "useragent" value to be restored
+ into memory from the realtime backend. This value is purely
+ informational. It does not alter configuration at all. (closes
+ issue #16029) Reported by: Guggemand Patches:
+ realtime-useragent.patch uploaded by Guggemand (license 897)
+ Tested by: Guggemand
+
+ * main/channel.c: _Really_ skip the channel... don't just retry for
+ another 200 cycles. (Closes issue SWP-1652, ABE-2240)
+
+2010-06-29 21:36 +0000 [r273017] Russell Bryant <russell at digium.com>
+
+ * /: Remove properties that were erroneously merged to 1.4 from one
+ of my branches.
+
+2010-07-22 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.34 Released.
+
+2010-07-07 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.34-rc2 Released.
+
+ * Fix problem with RFC 2833 DTMF not being accepted.
+
+ A recent check was added to ensure that we did not erroneously
+ detect duplicate DTMF when we received packets out of order.
+ The problem was that the check did not account for the fact that
+ the seqno of an RTP stream will roll over back to 0 after hitting
+ 65535. Now, we have a secondary check that will ensure that the
+ seqno rolling over will not cause us to stop accepting DTMF.
+
+ (closes issue 0017571)
+ Reported by: mdeneen
+ Patches:
+ rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
+ Tested by: richardf, maxochoa, JJCinAZ
+
+ * Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx
+ via the G option in app_dial
+
+ (closes issue 0017592)
+ Reported by: jamicque
+ Patches:
+ G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: jamicque, mnicholson
+
+2010-06-29 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.34-rc1 Released.
+
+2010-06-28 21:50 +0000 [r272921-272925] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Don't change ownership/group/permissions on run
+ directory, if it already exists. (closes issue #17076) Reported
+ by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
+ tilghman (license 14) Tested by: stuarth
+
+ * main/config.c: Also trim trailing blanks on #includes
+
+ * main/config.c: Change the way that we read include files, to
+ accommodate for changes in GCC 4.4. (closes issue #17472)
+ Reported by: seandarcy Patches: config2.patch uploaded by nivan
+ (license 1066) Tested by: nivan
+
+2010-06-28 18:47 +0000 [r272878-272881] Russell Bryant <russell at digium.com>
+
+ * tests/test_astobj2.c (added): Backport applicable parts of
+ test_astobj2.
+
+ * main/asterisk.c, Makefile, include/asterisk/test.h (added),
+ build_tools/cflags-devmode.xml, include/asterisk.h,
+ tests/Makefile, tests/test_skel.c, /, main/Makefile, tests
+ (added), include/asterisk/linkedlists.h, main/test.c (added):
+ Backport unit test API to 1.4. Review:
+ https://reviewboard.asterisk.org/r/750/
+
+2010-06-28 17:31 +0000 [r272804] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Decode URI in contact header of 302
+ response. ABE-2352
+
+2010-06-28 17:11 +0000 [r272688-272763] Russell Bryant <russell at digium.com>
+
+ * Makefile: Force SILENTMAKE where it is needed.
+
+ * Makefile: Backport method of setting SUBMAKE from trunk. By
+ setting the PRINT_DIR variable, SUBMAKE will print out the
+ directories it descends into, which is important for editors
+ (like vim) that watch the build output so that they can take you
+ to the file where an error occurred.
+
+2010-06-25 20:17 +0000 [r272562] Tilghman Lesher <tlesher at digium.com>
+
+ * doc/voicemail_odbc_postgresql.txt: Make the structure of the
+ table specified before match the queries and results. (closes
+ issue #17557) Reported by: cmaj
+
+2010-06-24 21:58 +0000 [r272446] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: ss_thread calls pri_grab without lock
+ during overlap dial Recent changes to chan_dahdi with relation to
+ overlap dialing call pri_grab without first obtaining a lock.
+ (closes issue #17414) Reported by: pdf Patches: bug17414.patch
+ uploaded by jpeeler (license 325)
+
+2010-06-23 22:33 +0000 [r272367] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c: Send AgentComplete manager events in the event
+ of blind and attended transfers. (closes issue #16819) Reported
+ by: elbriga Patches: app_queue.diff uploaded by elbriga (license
+ 482)
+
+2010-06-23 20:57 +0000 [r272255] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * apps/app_meetme.c: First caller into a dynamic conference now
+ enter pin once. If MeetMe is configured to use dynamic conference
+ numbers, then the first caller (which creates the conference) had
+ to enter the PIN number twice. (closes issue #15878) Reported by:
+ shawkris Patches: issue15878.patch uploaded by pabelanger
+ (license 224) Tested by: pabelanger
+
+2010-06-23 18:40 +0000 [r272147] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Backport part of revision 136715 to fix
+ callerid in voicemail text files (IMAP only). (closes issue
+ #16945) Reported by: mneuhauser
+
+2010-06-22 17:31 +0000 [r271689-271902] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Decrease the module ref count in sip_hangup
+ when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep
+ the ref count correct. (closes issue #16815) Reported by: rain
+ Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
+ (modified) Tested by: rain
+
+ * pbx/pbx_dundi.c: Allow users to specify a port for dundi peers.
+ (closes issue #17056) Reported by: klaus3000 Patches:
+ dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
+ Tested by: klaus3000
+
+ * configs/sip_notify.conf.sample, channels/chan_sip.c: Modify
+ chan_sip's packet generation api to automatically calculate the
+ Content-Length. This is done by storing packet content in a
+ buffer until it is actually time to send the packet, at which
+ time the size of the packet is calculated. This change was made
+ to ensure that the Content-Length is always correct. (closes
+ issue #17326) Reported by: kenner Tested by: mnicholson, kenner
+ Review: https://reviewboard.asterisk.org/r/693/
+
+2010-06-21 20:37 +0000 [r271552] Jeff Peeler <jpeeler at digium.com>
+
+ * pbx/pbx_ael.c: Do not use sizeof to calculate size of a heap
+ allocated character array. Change left out from 271399. (closes
+ issue #16053) Reported by: diLLec
+
+2010-06-18 20:52 +0000 [r271399-271444] Jeff Peeler <jpeeler at digium.com>
+
+ * pbx/pbx_ael.c: Check for newly added memory allocation failures
+ gracefully during AEL2 parsing.
+
+ * pbx/pbx_ael.c: Fix crash when parsing some heavily nested
+ statements in AEL on reload. Due to the recursion used when
+ compiling AEL in gen_prios, all the stack space was being
+ consumed when parsing some AEL that contained nesting 13 levels
+ deep. Changing a few large buffers to be heap allocated fixed the
+ crash, although I did not test how many more levels can now be
+ safely used. (closes issue #16053) Reported by: diLLec Tested by:
+ jpeeler
+
+2010-06-18 18:54 +0000 [r271339-271340] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/lock.h: Remove an unnecessary assignment that
+ causes a DEBUG_THREADS build failure on mac os x.
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Fix a build problem on Mac OS X with
+ DEBUG_THREADS enabled. This set of changes was already in trunk.
+
+2010-06-18 18:33 +0000 [r271335] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Eliminate deadlock potential in
+ dahdi_fixup(). (This is a backport of 269307, committed to trunk
+ by rmudgett.) Calling dahdi_indicate() when the channel private
+ lock is already held can cause a deadlock if the PRI lock is
+ needed because dahdi_indicate() will also get the channel private
+ lock. The pri_grab() function assumes that the channel private
+ lock is held once to avoid deadlock. (closes issue #17261)
+ Reported by: aragon
+
+2010-06-22 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.33.1 Released.
+
+ * channels/chan_dahdi.c: Merge revision 270404 from the 1.4 branch.
+
+ fixes FXS port still ringing when answered, as reported by Tzafrir
+ on dev-list.
+
+ (issue #17067)
+ Reported by: tzafrir
+ Tested by: alecdavis
+
+2010-06-17 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.33 Released.
+
+2010-06-10 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.33-rc2 Released.
+
+2010-06-10 Tilghman Lesher <tlesher at digium.com>
+
+ * Ensure signals are not blocked inside other signal handlers.
+
+ This eliminates the annoying <beep> on the console.
+
+ (closes issue 0017477)
+ Reported by: jvandal
+ Patches:
+ 20100610__issue17477.diff.txt uploaded by tilghman (license 14)
+
+2010-06-09 Paul Belanger <paul.belanger at polybeacon.com>
+
+ * Fix Debian init script to not use -c.
+
+ When using the init script as-is currently, it could cause issues on Debian
+ such as high CPU usage. This fix has worked for several people so I'm
+ implementing the change. We now handle color displays properly.
+
+ (closes issue 0016784)
+ Reported by: pabelanger
+ Patches:
+ 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: pabelanger, tilghman
+
+2010-06-01 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.33-rc1 Released.
+
+2010-06-01 15:17 +0000 [r266585] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Prevent CLI prompt from distorting output of
+ lines shorter than the prompt. Uses the VT100 method of clearing
+ the line from the cursor position to the end of the line: Esc-0K
+ (closes issue #17160) Reported by: coolmig Patches:
+ 20100531__issue17160.diff.txt uploaded by tilghman (license 14)
+ Tested by: coolmig
+
+2010-06-01 14:57 +0000 [r266579-266580] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_sip.c: Fix formatting issue with previous patch.
+
+ * channels/chan_sip.c: Missing fallback to audio fax feature when
+ T.38 re-INVITE failed When a T.38 re-INVITE failed with an 488 or
+ 606 answer, we should fallback to audio fax by send a
+ re-re-INVITE without T.38. The function is backported from 1.6
+ asterisk. (closes issue #16795) Reported by: vrban (closes issue
+ #16692) Reported by: vrban Patches:
+ t38_fallback_to_audio_v3.patch uploaded by vrban (license 756)
+ Tested by: lmadsen, vrban, haggard
+ https://reviewboard.asterisk.org/r/514/
+
+2010-05-30 04:43 +0000 [r266437] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Reverting patch and reopening
+ issue #16784, as patch breaks color display.
+
+2010-05-26 21:11 +0000 [r266142] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, main/logger.c: Use sigaction for signals which
+ should persist past the initial trigger, not signal. If you call
+ signal() in a Solaris signal handler, instead of just resetting
+ the signal handler, it causes the signal to refire, because the
+ signal is not marked as handled prior to the signal handler being
+ called. This effectively causes Solaris to immediately exceed the
+ threadstack in recursive signal handlers and crash. (closes issue
+ #17000) Reported by: rmcgilvr Patches:
+ 20100526__issue17000.diff.txt uploaded by tilghman (license 14)
+ Tested by: rmcgilvr
+
+2010-05-26 20:33 +0000 [r266140] David Vossel <dvossel at digium.com>
+
+ * channels/chan_dahdi.c: add dahdi_func_write to zap_tech structure
+ This was supposed to be committed with r263292, the back-port of
+ teh DAHDI buffer policy dial string option
+
+2010-05-26 18:21 +0000 [r266004] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Make AgentComplete message more consistent. At
+ times, the "Member" field was not specified during the event.
+ It's there now. (closes issue #15638) Reported by: elbriga
+ Patches: patchAppQueueAgentComplete.diff uploaded by elbriga
+ (license 482)
+
+2010-05-26 16:21 +0000 [r265910] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_pgsql.c: Not finding rows in the DB does not rise
+ to the level of a warning. (closes issue #17062) Reported by:
+ drookie Patches: 20100525__issue17062.diff.txt uploaded by
+ tilghman (license 14)
+
+2010-05-25 17:11 +0000 [r265613] David Vossel <dvossel at digium.com>
+
+ * channels/chan_dahdi.c: fixes build issue with zaptel (closes
+ issue #17394) Reported by: aragon Patches: half_buffer_fix.diff
+ uploaded by dvossel (license 671) Tested by: aragon
+
+2010-05-25 16:48 +0000 [r265610] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c: Don't mark the cdr records of unanswered queue
+ calls with "NOANSWER". This restores the behavior prior to
+ r258670. (closes issue #17334) Reported by: jvandal Patches:
+ queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
+ by: aragon, jvandal
+
+2010-05-25 13:33 +0000 [r265570] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c, Makefile,
+ doc/manager.txt, main/manager.c: Merged revisions 265320,265467
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24
+ May 2010) | 14 lines Add the FullyBooted AMI event It is possible
+ to connect to the manager interface before all Asterisk modules
+ are loaded. To ensure that an application does not send AMI
+ actions that might require a module that has not yet loaded, the
+ application can listen for the FullyBooted manager event. It will
+ be sent upon connection if all modules have been loaded, or as
+ soon as loading is complete. The event: Event: FullyBooted
+ Privilege: system,all Status: Fully Booted Review:
+ https://reviewboard.asterisk.org/r/639/ ........ r265467 |
+ twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
+ Merge the rest of the FullyBooted patch ........
+
+2010-05-24 19:37 +0000 [r265365] David Vossel <dvossel at digium.com>
+
+ * main/channel.c: fixes segfault when using generic plc
+
+2010-05-21 20:59 +0000 [r264996-265089] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/file.h, apps/app_queue.c: Don't hang up on a
+ queue caller if the file we attempt to play does not exist. This
+ also fixes a documentation mistake in file.h that made my
+ original attempt to correct this problem not work correctly.
+ (closes issue #17061) Reported by: RoadKill
+
+ * include/asterisk/channel.h: Fix grammatical error in comment.
+
+ * main/channel.c, main/autoservice.c, include/asterisk/channel.h:
+ Allow ast_safe_sleep to defer specific frames until after the
+ sleep has concluded. From reviewboard Background: A Digium
+ customer discovered a somewhat odd bug. The setup is that parties
+ A and B are bridged, and party A places party B on hold. While
+ party B is listening to hold music, he mashes a bunch of DTMF.
+ Party A takes party B off hold while this is happening, but party
+ B continues to hear hold music. I could reproduce this about 1 in
+ 5 times. The issue: When DTMF features are enabled and a user
+ presses keys, the channel that the DTMF is streamed to is placed
+ in an ast_safe_sleep for 100 ms, the duration of the emulated
+ tone. If an AST_CONTROL_UNHOLD frame is read from the channel
+ during the sleep, the frame is dropped. Thus the unhold
+ indication is never made to the channel that was originally
+ placed on hold. The fix: Originally, I discussed with Kevin
+ possible ways of fixing the specific problem reported. However,
+ we determined that the same type of problem could happen in other
+ situations where ast_safe_sleep() is used. Using autoservice as a
+ model, I modified ast_safe_sleep_conditional() to defer specific
+ frame types so they can be re-queued once the sleep has finished.
+ I made a common function for determining if a frame should be
+ deferred so that there are not two identical switch blocks to
+ maintain. Review: https://reviewboard.asterisk.org/r/674/
+
+2010-05-20 23:23 +0000 [r264820] Richard Mudgett <rmudgett at digium.com>
+
+ * main/callerid.c: ast_callerid_parse() had a path that left name
+ uninitialized. Several callers of ast_callerid_parse() do not
+ initialize the name parameter before calling thus there is the
+ potential to use an uninitialized pointer.
+
+2010-05-20 15:59 +0000 [r264541] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/options.h, main/loader.c, main/channel.c,
+ include/asterisk/channel.h: 1.4 version of PLC fix. Analogous to
+ trunk revision 264452, but without the change to chan_sip since
+ it is not necessary in this branch.
+
+2010-05-19 20:01 +0000 [r264334] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_speech_utils.c: Set quieted flag when receiving a dtmf
+ tone during playback in speechbackground. (closes issue #16966)
+ Reported by: asackheim
+
+2010-05-19 17:41 +0000 [r264248] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/options.h, configure, configure.ac: Internal
+ timing is now on by default, if you're using DAHDI 2.3 or above.
+ The reason for ensuring DAHDI 2.3 or above is that this version
+ ensures that a timer is always available, whereas in previous
+ versions, it was possible for DAHDI to be loaded, but have no
+ drivers to actually generate timing. If internal_timing was
+ turned on in this circumstance, a complete lack of audio would
+ result. This is the reason why internal_timing was not on by
+ default. However, now that DAHDI ensures the availability of a
+ timer, there is no reason for this setting to be off (and in
+ fact, it solves a great many initial user problems). (closes
+ issue #15932) Reported by: dimas Patches:
+ 20100519__issue15932.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2010-05-19 08:23 +0000 [r264056] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * configs/indications.conf.sample: fix incorrectly typed
+ indications for [nz] stutter and dialrecall (closes issue #17359)
+ Reported by: alecdavis Patches: bug17359.diff.txt uploaded by
+ alecdavis (license 585)
+
+2010-05-19 06:32 +0000 [r263949] Tilghman Lesher <tlesher at digium.com>
+
+ * main/dsp.c: Because progress is called multiple times, across
+ several frames, we must persist states when detecting multitone
+ sequences. (closes issue #16749) Reported by: dant Patches:
+ dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
+ dant
+
+2010-05-18 18:54 +0000 [r263769] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_directory.c: Modify directory name reading to be
+ interrupted with operator or pound escape. In the case of
+ accidentally entering the wrong first three letters for the
+ reading, users could be very frustrated if the name listing is
+ very long. This allows interrupting the reading by pressing 0 or
+ #. 0 will attempt to execute a configured operator (o) extension
+ and # will exit and proceed in the dialplan. ABE-2200
+
+2010-05-17 22:00 +0000 [r263637-263639] Mark Michelson <mmichelson at digium.com>
+
+ * main/devicestate.c: Fix logic error when checking for a devstate
+ provider. When using strsep, if one of the list of specified
+ separators is not found, it is the first parameter to strsep
+ which is now NULL, not the pointer returned by strsep. This issue
+ isn't especially severe in that the worst it is likely to do is
+ waste some cycles when a device with no '/' and no ':' is passed
+ to ast_device_state.
+
+ * main/pbx.c: Remove arbitrary size limitation for hints. (closes
+ issue #17257) Reported by: tim_ringenbach Patches:
+ hints_crash_fix.diff uploaded by tim ringenbach (license 540)
+
+2010-05-17 14:35 +0000 [r263374-263456] Leif Madsen <lmadsen at digium.com>
+
+ * main/http.c: Manager cookies are not compatible with RFC2109. The
+ Version field in the cookies we're setting contain quotes around
+ the version number which is not compatible with RFC2109 and
+ breaks some implementations. (closes issue #17231) Reported by:
+ ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
+ ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
+ ecarruda (license 559) Tested by: ecarruda, russell
+
+ * sounds/Makefile: Update link to new version of core sounds. The
+ latest version of the core sounds files 1.4.19 now includes the
+ missing queue-minute sound file which is called by app_queue but
+ which has been missing. (closes issue #17123) Reported by:
+ n8ideas
+
+2010-05-17 13:01 +0000 [r263292] David Vossel <dvossel at digium.com>
+
+ * channels/chan_dahdi.c: backport of DAHDI buffer policy dial
+ string option
+
+2010-05-13 23:08 +0000 [r263112] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, main/file.c: Fix internal timing not working with
+ Zaptel dahdi_compat.h was not being included in channel.c when
+ used with Zaptel and wasn't in file.c at all. (closes issue
+ #15250) Reported by: mneuhauser Patches: dahdi_compat.patch
+ uploaded by mneuhauser (license 425) Tested by: IgorG
+
+2010-05-12 17:00 +0000 [r262662] David Vossel <dvossel at digium.com>
+
+ * apps/app_meetme.c: fixes app_meetme dsp error We attempted to
+ detect silence after translating a frame from signed linear. This
+ caused a flooding of errors. To resolve this the code to detect
+ silence was moved before the translation. (closes issue #17133)
+ Reported by: jsdyer
+
+2010-05-11 19:55 +0000 [r262421] Jason Parker <jparker at digium.com>
+
[... 28617 lines stripped ...]
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