[svn-commits] bebuild: tag certified-1.8.11-cert5-rc1 r369850 - /certified/tags/1.8.11-cert...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jul 9 15:01:04 CDT 2012
Author: bebuild
Date: Mon Jul 9 15:01:00 2012
New Revision: 369850
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369850
Log:
Importing files for 1.8.11-cert5-rc1 release.
Added:
certified/tags/1.8.11-cert5-rc1/.lastclean (with props)
certified/tags/1.8.11-cert5-rc1/.version (with props)
certified/tags/1.8.11-cert5-rc1/ChangeLog (with props)
Added: certified/tags/1.8.11-cert5-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/certified/tags/1.8.11-cert5-rc1/.lastclean?view=auto&rev=369850
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URL: http://svnview.digium.com/svn/asterisk/certified/tags/1.8.11-cert5-rc1/ChangeLog?view=auto&rev=369850
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--- certified/tags/1.8.11-cert5-rc1/ChangeLog (added)
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@@ -1,0 +1,38423 @@
+2012-07-09 Asterisk Development Team
+
+ * Certified Asterisk 1.8.11-cert5-rc1 Released.
+
+2012-07-09 19:59 +0000 [r369848] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
+ channels/sig_analog.c, /, channels/chan_sip.c,
+ include/asterisk/channel.h: Fix deadlock between bridged channels
+ that attempt to set the hangup source Calling
+ ast_set_hangupsource with the channel lock held can result in a
+ deadlock because the function also locks the bridged channel.
+ (issue AST-891)
+
+2012-07-09 19:50 +0000 [r369845] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Add support for exposing the received
+ contact URI and also for setting the request URI in messages.
+ (closes issue AST-911)
+
+2012-07-09 19:06 +0000 [r369839-369840] Jason Parker <jparker at digium.com>
+
+ * include/asterisk/app_voicemail.h (removed): Remove file that
+ should no longer exist.
+
+ * apps/app_mixmonitor.c, apps/app_voicemail.c,
+ include/asterisk/callerid.h, include/asterisk/app.h,
+ channels/chan_sip.c, apps/app_voicemail.exports.in,
+ tests/test_voicemail_api.c, main/callerid.c, main/app.c,
+ include/asterisk/app_voicemail.h: Re-merge changes that were
+ reverted.
+ ------------------------------------------------------------------------
+ r365395 | qwell | 2012-05-04 16:17:08 -0500 (Fri, 04 May 2012) |
+ 7 lines Add support for folders in MixMonitor 'm' option.
+ Backport manager actions. The manager actions are needed, so
+ MixMonitor can be executed on existing channels. (issue DPMA-68)
+ ------------------------------------------------------------------------
+ r364761 | qwell | 2012-05-01 12:25:14 -0500 (Tue, 01 May 2012) |
+ 6 lines Remove folder_dir from voicemail snapshots API. It was
+ both unused (except in tests, where it was fudged) and
+ unnecessary. (closes issue AST-842)
+ ------------------------------------------------------------------------
+ r367161 | mmichelson | 2012-05-21 14:05:52 -0500 (Mon, 21 May
+ 2012) | 21 lines Add "send to voicemail" Digium phone
+ functionality to Asterisk. This change accommodates two methods
+ by which calls can be directed to a user's voicemail. * Incoming
+ calls can be redirected to any user's voicemail. * Established
+ calls can be blind transferred to any user's voicemail. Digium
+ phones indicate the desire to direct a call to voicemail by using
+ a Diversion header with a reason parameter of "send_to_vm". This
+ patch adds the "send_to_vm" reason as a valid redirecting reason.
+ In addition, chan_sip.c has been modified to update redirecting
+ information on the transferred channel by reading a Diversion
+ header on a REFER request. (closes issue AST-871) Reported by
+ Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925
+ ------------------------------------------------------------------------
+ r368790 | mjordan | 2012-06-12 08:44:36 -0500 (Tue, 12 Jun 2012)
+ | 18 lines Fix deadlock in SIP transfers that involve a REFER
+ request In r367163, "send to voicemail" functionality was added
+ to the SIP channel driver. This required updating the party
+ redirecting information for the channel based on the headers
+ provided in the REFER request. When the redirecting party
+ information is updated on the channel, a call to
+ ast_indicate_data occurs. Because handle_request_refer still had
+ the sip_pvt locked, a deadlock could occur between the pbx_thread
+ and the do_monitor thread servicing the REFER request. This patch
+ preserves the proper locking order between the channel and the
+ sip_pvt by ensuring that the sip_pvt is unlocked prior to
+ updating the party redirecting information on the channel.
+ (closes issue AST-903) Reported by: Matt Jordan patches:
+ jira_ast_903_trunk.patch by rmudgett (license 5621)
+ ------------------------------------------------------------------------
+ r368962 | qwell | 2012-06-14 13:38:48 -0500 (Thu, 14 Jun 2012) |
+ 11 lines Remove global symbol requirement from app_voicemail.
+ This uses the existing "function installation" stuff that already
+ existed for other functions, like getting message counts. (closes
+ issue AST-807) (issue AST-901) (issue AST-908) Review:
+ https://reviewboard.asterisk.org/r/1965/
+ ------------------------------------------------------------------------
+ r368964 | qwell | 2012-06-14 14:03:24 -0500 (Thu, 14 Jun 2012) |
+ 8 lines These functions that were moved need to be static. Also
+ wrap test functions in a #ifdef. (issue AST-807) (issue AST-901)
+ (issue AST-908)
+ ------------------------------------------------------------------------
+ r368998 | qwell | 2012-06-15 10:31:43 -0500 (Fri, 15 Jun 2012) |
+ 6 lines Remove some symbol exports that got missed in the removal
+ of global symbols. (issue AST-807) (issue AST-901) (issue
+ AST-908)
+ ------------------------------------------------------------------------
+ r369024 | qwell | 2012-06-15 11:29:40 -0500 (Fri, 15 Jun 2012) |
+ 2 lines Fix voicemail API tests by using the correct argument
+ order for create/destroy.
+ ------------------------------------------------------------------------
+
+2012-07-05 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.11-cert4 Released.
+
+ * AST-2012-010
+
+ * AST-2012-011
+
+2012-05-29 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.11-cert2 Released.
+
+ * AST-2012-007
+
+ * AST-2012-008
+
+2012-04-25 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.11-cert1 Released.
+
+2012-04-25 16:53 +0000 [r363674] Jason Parker <jparker at digium.com>
+
+ * / (added): Asterisk 1.8-digiumphones branch has become Certified
+ Asterisk 1.8.11. For more details about Certified Asterisk, see
+ http://tinyurl.com/7pfp639
+
+2012-04-24 20:57 +0000 [r363374] Jason Parker <jparker at digium.com>
+
+ * /res/res_smdi.c,
+ /apps/app_osplookup.c,
+ /channels/chan_misdn.c,
+ /channels/chan_skinny.c,
+ /funcs/func_frame_trace.c,
+ /cdr/cdr_sqlite.c,
+ /pbx/pbx_realtime.c,
+ /apps/app_amd.c,
+ /pbx/pbx_dundi.c,
+ /apps/app_url.c,
+ /channels/chan_nbs.c,
+ /apps/app_externalivr.c,
+ /apps/app_zapateller.c,
+ /cdr/cdr_odbc.c,
+ /res/res_fax_spandsp.c,
+ /channels/chan_mgcp.c,
+ /cel/cel_pgsql.c,
+ /apps/app_readfile.c,
+ /apps/app_test.c,
+ /apps/app_ices.c,
+ /channels/chan_gtalk.c,
+ /cdr/cdr_csv.c,
+ /channels/chan_phone.c,
+ /funcs/func_pitchshift.c,
+ /apps/app_waitforring.c,
+ /formats/format_vox.c,
+ /res/res_timing_pthread.c,
+ /apps/app_minivm.c,
+ /channels/chan_h323.c,
+ /cel/cel_sqlite3_custom.c,
+ /apps/app_confbridge.c,
+ /res/res_config_ldap.c,
+ /apps/app_nbscat.c,
+ /cdr/cdr_sqlite3_custom.c,
+ /res/res_snmp.c,
+ /apps/app_dictate.c,
+ /apps/app_waitforsilence.c,
+ /apps/app_dahdiras.c,
+ /pbx/pbx_lua.c,
+ /apps/app_alarmreceiver.c,
+ /apps/app_image.c,
+ /res/res_ael_share.c,
+ /cdr/cdr_tds.c,
+ /apps/app_setcallerid.c,
+ /apps/app_mp3.c,
+ /channels/chan_alsa.c,
+ /res/res_timing_kqueue.c,
+ /channels/chan_unistim.c,
+ /apps/app_dahdibarge.c,
+ /res/res_config_pgsql.c,
+ /res/res_adsi.c,
+ /res/res_phoneprov.c,
+ /apps/app_morsecode.c,
+ /cdr/cdr_pgsql.c,
+ /res/res_config_sqlite.c,
+ /channels/chan_jingle.c,
+ /pbx/pbx_ael.c,
+ /apps/app_sms.c,
+ /formats/format_jpeg.c,
+ /apps/app_jack.c,
+ /apps/app_adsiprog.c,
+ /cel/cel_radius.c,
+ /res/res_ais.c,
+ /cel/cel_tds.c,
+ /apps/app_festival.c,
+ /apps/app_chanisavail.c,
+ /channels/chan_console.c,
+ /apps/app_talkdetect.c,
+ /res/res_jabber.c,
+ /cdr/cdr_radius.c,
+ /apps/app_getcpeid.c,
+ /channels/chan_oss.c: Disable extended
+ and deprecated modules by default. Users can still enable any of
+ these using menuselect if they so choose. (closes issue AST-873)
+
+2012-04-23 15:17 +0000 [r363161] Jason Parker <jparker at digium.com>
+
+ * /main/manager.c,
+ ,
+ /channels/chan_sip.c,
+ /channels/chan_skinny.c: Multiple
+ revisions 363102,363106,363141 ........ r363102 | mjordan |
+ 2012-04-23 08:37:55 -0500 (Mon, 23 Apr 2012) | 16 lines
+ AST-2012-005: Fix remotely exploitable heap overflow in keypad
+ button handling When handling a keypad button message event, the
+ received digit is placed into a fixed length buffer that acts as
+ a queue. When a new message event is received, the length of that
+ buffer is not checked before placing the new digit on the end of
+ the queue. The situation exists where sufficient keypad button
+ message events would occur that would cause the buffer to be
+ overrun. This patch explicitly checks that there is sufficient
+ room in the buffer before appending a new digit. (closes issue
+ ASTERISK-19592) Reported by: Russell Bryant ........ Merged
+ revisions 363100 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ r363106 | mjordan | 2012-04-23 09:05:02 -0500 (Mon, 23 Apr 2012)
+ | 17 lines AST-2012-006: Fix crash in UPDATE handling when no
+ channel owner exists If Asterisk receives a SIP UPDATE request
+ after a call has been terminated and the channel has been
+ destroyed but before the SIP dialog has been destroyed, a
+ condition exists where a connected line update would be attempted
+ on a non-existing channel. This would cause Asterisk to crash.
+ The patch resolves this by first ensuring that the SIP dialog has
+ an owning channel before attempting a connected line update. If
+ an UPDATE request is received and no channel is associated with
+ the dialog, a 481 response is sent. (closes issue ASTERISK-19770)
+ Reported by: Thomas Arimont Tested by: Matt Jordan Patches:
+ ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license
+ 6283) ........ r363141 | jrose | 2012-04-23 09:33:16 -0500 (Mon,
+ 23 Apr 2012) | 20 lines AST-2012-004: Fix an error that allows
+ AMI users to run shell commands sans authorization. As detailed
+ in the advisory, AMI users without write authorization for SYSTEM
+ class AMI actions were able to run system commands by going
+ through other AMI commands which did not require that
+ authorization. Specifically, GetVar and Status allowed users to
+ do this by setting their variable/s options to the SHELL or EVAL
+ functions. Also, within 1.8, 10, and trunk there was a similar
+ flaw with the Originate action that allowed users with originate
+ permission to run MixMonitor and supply a shell command in the
+ Data argument. That flaw is fixed in those versions of this
+ patch. (closes issue ASTERISK-17465) Reported By: David Woolley
+ Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) ........ Merged revisions 363117 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 363102,363106,363141 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-19 20:31 +0000 [r362673] Mark Michelson <mmichelson at digium.com>
+
+ * /channels/chan_sip.c: Add a test
+ application for sending custom SIP INFO messages. When
+ TEST_FRAMEWORK is enabled, SIPSendCustomInfo is available to test
+ sending custom INFO requests. Review:
+ https://reviewboard.asterisk.org/r/1866
+
+2012-04-13 17:19 +0000 [r362042-362132] Matthew Jordan <mjordan at digium.com>
+
+ * : Rename property branches-1.8-merged to
+ branch-1.8-merged
+
+ * : Update properties on 1.8-digiumphones
+ Change the merge property tag from svnmerge-integrated to
+ branches-1.8-merged. Added merged revisions from r362042.
+
+ * ,
+ /channels/chan_sip.c,
+ /main/features.c: Merge of several
+ needed fixes for 1.8-digiumphones This merges fixes for the
+ following issues into the 1.8-digiumphones branch: *
+ ASTERISK-19355 - Call transfer with consultation frequently fails
+ in cross- linked Asterisk scenario (directmedia & sendrpid
+ active) * ASTERISK 19365 - Remote SIP Call legs are frequently
+ not released in a cross-linked Asterisk scenario (directmedia &
+ sendrpid) * ASTERISK-19183 - Sporadically missing connectedline
+ event to caller channel in directed pickup app
+
+2012-04-09 20:40 +0000 [r361704] Mark Michelson <mmichelson at digium.com>
+
+ * /apps/app_voicemail.c,
+ /apps/app_voicemail.exports.in,
+ /tests/test_voicemail_api.c (added),
+ /include/asterisk/app_voicemail.h: Fix
+ bugs in voicemail APIs and add unit tests. There were several
+ crashes that could occur due to NULL inputs, invalid inputs, and
+ the like. This fixes all known ones and adds unit tests to
+ exercise the APIs.
+
+2012-04-06 19:08 +0000 [r361502] Richard Mudgett <rmudgett at digium.com>
+
+ * /main/message.c: Update Func MESSAGE()
+ and AMI MessageSend documentation. * Document
+ MESSAGE(custom_data) * Update AMI MessageSend documentation *
+ Eliminate a shadowed variable name in msg_func_write() for
+ custom_data.
+
+2012-04-05 17:24 +0000 [r361283] Mark Michelson <mmichelson at digium.com>
+
+ * /funcs/func_presence_state.c,
+ /tests/test_config.c: Add additional
+ configuration and presence unit tests. These were originally
+ written while merging features into trunk, but these tests apply
+ just as much for the 1.8 version of Digium phones, so might as
+ well have them here, too.
+
+2012-04-03 21:03 +0000 [r361088] Jonathan Rose <jrose at digium.com>
+
+ * /apps/app_mixmonitor.c: Make m option
+ for mixmonitor delete the source file once it is finished copying
+ to vm. Review: https://reviewboard.asterisk.org/r/1842/
+
+2012-03-29 21:49 +0000 [r360826] Jason Parker <jparker at digium.com>
+
+ * /main/manager.c,
+ ,
+ /main/utils.c,
+ /include/asterisk/manager.h,
+ /apps/app_milliwatt.c: Multiple
+ revisions 359656,359706,359979 ........ r359656 | mjordan |
+ 2012-03-15 13:35:59 -0500 (Thu, 15 Mar 2012) | 22 lines Fix
+ remotely exploitable stack overrun in Milliwatt Milliwatt is
+ vulnerable to a remotely exploitable stack overrun when using the
+ 'o' option. This occurs due to the milliwatt_generate function
+ not accounting for AST_FRIENDLY_OFFSET when calculating the
+ maximum number of samples it can put in the output buffer. This
+ patch resolves this issue by taking into account
+ AST_FRIENDLY_OFFSET when determining the maximum number of
+ samples allowed. Note that at no point is remote code execution
+ possible. The data that is written into the buffer is the
+ pre-defined Milliwatt data, and not custom data. (closes issue
+ ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt
+ Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell
+ Bryant (license 6283) Note that this patch was written by
+ Russell, even though Matt uploaded it ........ Merged revisions
+ 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+ ........ r359706 | mjordan | 2012-03-15 14:01:22 -0500 (Thu, 15
+ Mar 2012) | 16 lines Fix remotely exploitable stack overflow in
+ HTTP manager There exists a remotely exploitable stack buffer
+ overflow in HTTP digest authentication handling in Asterisk. The
+ particular method in question is only utilized by HTTP AMI. When
+ parsing the digest information, the length of the string is not
+ checked when it is copied into temporary buffers allocated on the
+ stack. This patch fixes this behavior by parsing out pre-defined
+ key/value pairs and avoiding unnecessary copies to the stack.
+ (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
+ by: Matt Jordan ........ r359979 | rmudgett | 2012-03-20 12:21:16
+ -0500 (Tue, 20 Mar 2012) | 28 lines Allow AMI action callback to
+ be reentrant. Fix AMI module reload deadlock regression from
+ ASTERISK-18479 when it tried to fix the race between calling an
+ AMI action callback and unregistering that action. Refixes
+ ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2
+ object guaranteed that there were no active callbacks that
+ mattered when ast_manager_unregister() was called. Unfortunately,
+ this causes the deadlock situation. The patch stops locking the
+ ao2 object to allow multiple threads to invoke the callback
+ re-entrantly. There is no way to guarantee a module unload will
+ not crash because of an active callback. The code attempts to
+ minimize the chance with the registered flag and the maximum 5
+ second delay before ast_manager_unregister() returns. The trunk
+ version of the patch changes the API to fix the race condition
+ correctly to prevent the module code from unloading from memory
+ while an action callback is active. * Don't hold the lock while
+ calling the AMI action callback. (closes issue ASTERISK-19487)
+ Reported by: Philippe Lindheimer Review:
+ https://reviewboard.asterisk.org/r/1818/ Review:
+ https://reviewboard.asterisk.org/r/1820/ ........ Merged
+ revisions 359656,359706,359979 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-21 15:44 +0000 [r360031-360188] Mark Michelson <mmichelson at digium.com>
+
+ * /main/pbx.c: Prevent potentially passing
+ a NULL pointer to strcasecmp()
+
+ * /main/pbx.c: Fix one more "(null)"
+ string. If a hint with no presence portion were added, it would
+ result in another "(null)" string warning.
+
+ * /main/pbx.c: Fix another "Possible
+ programming error" bug. Similar to the previous commit, don't
+ pass a printf-generated string to ast_strlen_zero.
+
+ * /main/pbx.c: Get rid of an annoying
+ "Possible programming error" message. If an extension's 'app'
+ field is NULL, then a "(null)" string would be written into an
+ ast_str due to the way that snprintf works. When this is passed
+ to ast_strlen_zero(), it fires up a big warning indicating
+ something is probably wrong. There indeed was a problem, but
+ luckily it wasn't a very big problem. After the failed
+ ast_strlen_zero() check and big warning message, the very next if
+ statement, checking to see if the "(null)" matched a presence
+ provider, would fail, so no harm was done.
+
+2012-03-08 18:40 +0000 [r358725] Jonathan Rose <jrose at digium.com>
+
+ * /apps/app_mixmonitor.c: Fixes
+ unitialized variable use warning introduced by addition of
+ mixmonitor forward to vm
+
+2012-03-08 18:02 +0000 [r358692] Jason Parker <jparker at digium.com>
+
+ * , /main/acl.c:
+ Prevent outbound SIP NOTIFY packets from displaying a port of 0
+ In the change from 1.6.2 to 1.8, ast_sockaddr was introduced
+ which changed the behavior of ast_find_ourip such that port
+ number was wiped out. This caused the port in internip (which is
+ used for Contact and Call-ID on NOTIFYs) to be 0. This change
+ causes ast_find_ourip to be port-preserving again. (closes issue
+ ASTERISK-19430) ........ Merged revisions 357665 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-02 15:18 +0000 [r357808] Paul Belanger <pabelanger at digium.com>
+
+ * /apps/app_mixmonitor.c: Fixed xmldoc
+ formatting error for 'm' option
+
+2012-02-28 21:52 +0000 [r357456-357459] Jason Parker <jparker at digium.com>
+
+ * /main/channel.c,
+ /funcs/func_presence_state.c (added),
+ /main/manager.c,
+ /channels/chan_skinny.c,
+ /funcs/func_frame_trace.c,
+ /include/asterisk/jabber.h,
+ /main/file.c,
+ /main/app.c,
+ /tests/test_config.c (added),
+ /include/asterisk/frame.h,
+ /main/custom_control_frame.c (added),
+ /main/message.c (added),
+ /apps/app_mixmonitor.c,
+ /channels/sip/include/sip.h,
+ /main/asterisk.c,
+ /tests/test_custom_control.c (added),
+ /main/pbx.c,
+ /include/asterisk/presencestate.h
+ (added),
+ /include/asterisk/app_voicemail.h
+ (added), /include/asterisk/channel.h,
+ /include/asterisk/manager.h,
+ /apps/app_queue.c,
+ /main/config.c,
+ /include/asterisk/file.h,
+ /include/asterisk/app.h,
+ /include/asterisk/event_defs.h,
+ /configs/jabber.conf.sample,
+ /include/asterisk/custom_control_frame.h
+ (added), /include/asterisk/message.h
+ (added), /main/features.c,
+ /apps/app_voicemail.exports.in,
+ /main/event.c,
+ /include/asterisk/pbx.h,
+ /configs/sip.conf.sample,
+ /apps/app_voicemail.c,
+ /channels/chan_sip.c,
+ /include/asterisk/config.h,
+ /configs/manager.conf.sample,
+ /include/asterisk/_private.h,
+ /res/res_jabber.c,
+ /main/presencestate.c (added): Add
+ support for Digium Phones.
+
+
+2012-03-29 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.11.0 Released.
+
+2012-03-26 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.11.0-rc3 Released.
+
+ * AST-2012-003
+
+ * AST-2012-002
+
+ * /main/manager.c, /include/asterisk/manager.h: Fix AMI deadlock
+ regression by allowing AMI action callback to be reentrant
+
+ Fix AMI module reload deadlock from ASTERISK-18479 when it tried
+ to fix the race between calling an AMI action callback and
+ unregistering that action. Refixes ASTERISK-13784 broken by
+ ASTERISK-17785 change.
+
+ Locking the ao2 object guaranteed that there were no active
+ callbacks that mattered when ast_manager_unregister() was called.
+ Unfortunately, this causes the deadlock situation. The patch stops
+ locking the ao2 object to allow multiple threads to invoke the
+ callback re-entrantly. There is no way to guarantee a module unload
+ will not crash because of an active callback. The code attempts to
+ minimize the chance with the registered flag and the maximum 5
+ second delay before ast_manager_unregister() returns.
+
+ The trunk version of the patch changes the API to fix the race
+ condition correctly to prevent the module code from unloading from
+ memory while an action callback is active.
+
+ * Don't hold the lock while calling the AMI action callback.
+
+ (closes issue ASTERISK-19487)
+ Reported by: Philippe Lindheimer
+
+ Review: https://reviewboard.asterisk.org/r/1818/
+
+2012-03-06 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.11.0-rc2 Released.
+
+ * main/acl.c: Prevent outbound SIP NOTIFY packets from displaying
+ a port of 0.
+
+ In the change from 1.6.2 to 1.8, ast_sockaddr was
+ introduced which changed the behavior of ast_find_ourip such
+ that port number was wiped out. This caused the port in
+ internip (which is used for Contact and Call-ID on NOTIFYs) to be
+ 0. This change causes ast_find_ourip to be port-preserving again.
+
+2012-01-30 21:57 +0000 [r353368-353320] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/sip/include/sip.h, channels/sip/include/dialog.h,
+ channels/chan_sip.c: RFC3261 Section 8.1.1.5. The sequence number
+ value MUST be expressible as a 32-bit unsigned integer * fix: use
+ %u instead of %d when dealing with CSeq numbers - to remove
+ possibility of -ve numbers. * fix: change all uses of seqno and
+ friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
+ Summary of CSeq numbers. An initial CSeq number must be less than
+ 2^31 A CSeq number can increase in value up to 2^32-1 An
+ incrementing CSeq number must not wrap around to 0. Tested with
+ Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
+ Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1699/
+
+ * channels/chan_sip.c: prevent debug messsges displaying -ve Cseq
+ numbers. Missed in R353320
+
+2012-01-30 23:17 +0000 [r353371] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/dnsmgr.h, main/dnsmgr.c, channels/chan_sip.c:
+ Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr
+ currently takes a pointer to an ast_sockaddr and updates it
+ anytime an address resolves to something different. There are a
+ couple of issues with this. First, the ast_sockaddr is usually
+ the address of an ast_sockaddr inside a refcounted struct and we
+ never bump the refcount of those structs when using dnsmgr. This
+ makes it possible that a refresh could happen after the
+ destructor for that object is called (despite ast_dnsmgr_release
+ being called in that destructor). Second, the module using dnsmgr
+ cannot be aware of an address changing without polling for it in
+ the code. If an action needs to be taken on address update (like
+ re-linking a SIP peer in the peers_by_ip table), then polling for
+ this change negates many of the benefits of having dnsmgr in the
+ first place. This patch adds a function to the dnsmgr API that
+ calls an update callback instead of blindly updating the address
+ itself. It also moves calls to ast_dnsmgr_release outside of the
+ destructor functions and into cleanup functions that are called
+ when we no longer need the objects and increments the refcount of
+ the objects using dnsmgr since those objects are stored on the
+ ast_dnsmgr_entry struct. A helper function for returning the
+ proper default SIP port (non-tls vs tls) is also added and used.
+ This patch also incorporates changes from a patch posted by Timo
+ Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue
+ ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/
+
+2012-01-31 16:51 +0000 [r353454] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/channel.h, main/manager.c: Fix memory leak in
+ error paths for action_originate(). * Fix memory leak of vars in
+ error paths for action_originate(). * Moved struct
+ fast_originate_helper tech and data members to stringfields. *
+ Simplified ActionID header handling for fast_originate(). * Added
+ doxygen note to ast_request() and ast_call() and the associated
+ channel callbacks that the data/addr parameters should be treated
+ as const char *. Review: https://reviewboard.asterisk.org/r/1690/
+
+2012-01-31 23:41 +0000 [r353502] Terry Wilson <twilson at digium.com>
+
+ * res/res_calendar.c: Allow res_calendar to be unloaded The
+ calendaring tech modules depend on res_calendar and initially
+ res_calendar just bumped the use count so that it couldn't be
+ unloaded. res_calendar can potentially create many threads and
+ I've seen issues where the Asterisk shutdown has failed where it
+ looked like these threads could be the culprit. This patch adds
+ unload support for res_calendar. Unloading res_calendar will also
+ unload the dependant tech modules as well. (closes issue
+ ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/
+
+2012-02-01 15:02 +0000 [r353550] Matthew Jordan <mjordan at digium.com>
+
+ * contrib/init.d/etc_default_asterisk: Added clarification for the
+ VERBOSITY setting to etc_default_asterisk Clarified that using
+ the VERBOSITY setting in etc_default_asterisk is the same as
+ using the -v command line switch, which causes Asterisk to launch
+ in console mode. (closes issue ASTERISK-17030) Reported by: Jonas
+
+2012-02-01 15:50 +0000 [r353598] Sean Bright <sean at malleable.com>
+
+ * include/asterisk/audiohook.h: Resolve an overlap in the
+ ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
+ AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
+ unintended side effects. This patch moves
+ AST_AUDIOHOOK_TRIGGER_WRITE, and updates
+ AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
+ This will affect existing modules that use these flags, so be
+ sure to recompile as necessary. (closes issue ASTERISK-19246)
+ Reported by: feyfre
+
+2012-02-01 21:05 +0000 [r353769-353720] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers for
+ various functions in chan_sip There are a number of cleaner
+ looking wrappers for ast_sockaddr_stringify_fmt available which
+ are slightly more readable than using a direct call to
+ ast_sockaddr_stringify_fmt. This patch switches a number of those
+ calls in chan_sip to use those wrappers and is generally
+ harmless. (Closes issue ASTERISK-16930) Reported by: Michael L.
+ Young Patches: chan_sip-broken-registration-1.8.diff uploaded by
+ Michael L. Young (license 5026)
+
+ * channels/chan_sip.c: Fix sip show peers port output, align
+ columns, and fix ami port output. A previous patch I committed
+ from ASTERISK-16930 unexpectedly changed some output for the AMI
+ action "sippeers" which this patch changes back. Also, this
+ aligns the output for the cli command "sip show peers" and fixes
+ another issue that patch introduced by using
+ ast_sockaddr_stringify calls multiple times without immediately
+ using the pointer. I also went ahead and did a little janitorial
+ work to clean up whitespace in _sip_show_peers. (issue
+ ASTERISK-16930) (closes issue ASTERISK-19281) Reported by:
+ Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by
+ Walter Doekes (license 5674)
+
+2012-02-02 16:58 +0000 [r353770] Mark Michelson <mmichelson at digium.com>
+
+ * UPGRADE.txt, configs/manager.conf.sample,
+ include/asterisk/manager.h, configs/http.conf.sample,
+ main/manager.c, main/http.c: Fix TLS port binding behavior as
+ well as reload behavior: * Removes references to tlsbindport from
+ http.conf.sample and manager.conf.sample * Properly bind to port
+ specified in tlsbindaddr, using the default port if specified. *
+ On a reload, properly close socket if the service has been
+ disabled. A note has been added to UPGRADE.txt to indicate how
+ ports must be set for TLS. (closes issue ASTERISK-16959) reported
+ by Olaf Holthausen (closes issue ASTERISK-19201) reported by
+ Chris Mylonas (closes issue ASTERISK-19204) reported by Chris
+ Mylonas Review: https://reviewboard.asterisk.org/r/1709
+
+2012-02-02 18:31 +0000 [r353818] Jonathan Rose <jrose at digium.com>
+
+ * funcs/func_curl.c: Backports some documentation for func_curl
+ from 10 to 1.8 For some reason this function was completely
+ undocumented in 1.8. I copied the 10 docs over to 1.8 and removed
+ references to an enumerator that was added in the Asterisk 10
+ version of func_curl. That was the only change I noted. (closes
+ issue ASTERISK-19186) Reported by: Olivier Krief
+
+2012-02-02 20:01 +0000 [r353867] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
+ Restore the 'w' modifier support for ISDN spans.
+ Dial(DAHDI/g0/1234w888) This feature also causes the sending
+ complete ie to be sent for switch types that do not automatically
+ send the ie. (EuroISDN/ETSI) The main difference between dialing
+ Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the
+ sending of the sending complete ie. (closes issue ASTERISK-19176)
+ Reported by: rmudgett Tested by: rmudgett
+
+2012-02-02 22:26 +0000 [r353915] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Ensure entering T.38 passthrough does not
+ cause an infinite loop After R340970 Asterisk was still polling
+ the RTCP file descriptor after RTCP is shut down and removed. If
+ the descriptor happened to have data ready when the removal
+ occured then Asterisk would go into an infinite loop trying to
+ read data that it can never actually access. This change disables
+ the audio RTCP file descriptor for the duration of the T.38
+ transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
+ Vrban
+
+2012-02-03 21:24 +0000 [r353999] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due
+ to r335976 Bad locking order was added to chan_agent to prevent
+ segfaults from having no locking in a patch by irroot. This patch
+ addresses the bad locking order by releasing locks before getting
+ the right locking order to stop deadlocks from occuring when
+ doing multiple interactions with agents. (closes issue
+ ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
+ https://reviewboard.asterisk.org/r/1708/
+
+2012-02-06 17:28 +0000 [r354216-354116] Richard Mudgett <rmudgett at digium.com>
+
+ * main/features.c: Add missing headers to AMI UnParkedCall event to
+ uniquely identify the call. The AMI UnParkedCall event was
+ missing the Parkinglot and Uniqueid headers that the AMI
+ ParkedCall event contains. (closes issue ASTERISK-19240) Reported
+ by: Michael Yara
+
+ * pbx/pbx_config.c: Improved documentation of CLI "dialplan add
+ extension" command. * Documented dialplan add extension
+ <exten>,<priority>,<app(<app-data>)> format. * Allow acceptance
+ of command without the app-data value. There are many
+ applications that do no need any parameters so it is silly to
+ require that field for all commands. * Fixed a couple
+ ast_malloc/ast_free mismatches with ast_add_extension2() calls.
+ (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
+ by: rmudgett
+
+2012-02-07 15:04 +0000 [r354263] Jonathan Rose <jrose at digium.com>
+
+ * cdr/cdr_pgsql.c: Fix column duplication bug in module reload for
+ cdr_pgsql. Prior to this patch, attempts to reload cdr_pgsql.so
+ would cause the column list to keep its current data and then add
+ a second copy during the reload. This would cause attempts to log
+ the CDR to the database to fail. This patch also cleans up some
+ unnecessary null checks for ast_free and deals with a few
+ potential locking problems. (closes issue ASTERISK-19216)
+ Reported by: Jacek Konieczny Review:
+ https://reviewboard.asterisk.org/r/1711/
+
+2012-02-07 20:53 +0000 [r354348] Terry Wilson <twilson at digium.com>
+
+ * contrib/realtime/postgresql/realtime.sql, channels/chan_sip.c:
+ Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing
+ instead of "" 2. Don't set ipaddr or port to the string "(null)"
+ when they are empty 3. Add missing required fields, set default
+ for lastms to 0, and modify the length of the ipaddr field to 45
+ in the Postgresql realtime.sql file. (closes issue
+ ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/
+
+2012-02-09 02:23 +0000 [r354492] Russell Bryant <russell at russellbryant.com>
+
+ * main/channel.c: Remove some unnecessary locking from
+ ast_hangup(). This patch removes some unnecessary locking of the
+ channels container in ast_hangup(). The reason this came up is
+ that this lock can very quickly block the entire system. If any
+ of the channel cleanup code decides to block, it causes a problem
+ for the whole system. For example, when audiohooks get destroyed,
+ if that blocks for a while waiting on the mixmonitor thread to
+ exit because it's busy blocking on some I/O, it causes a problem
+ for many other threads in the meantime. Review:
+ https://reviewboard.asterisk.org/r/1712/
+
+2012-02-09 02:52 +0000 [r354495] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce. Well,
+ thats embarrasing. I forgot to initialize the caller_id storage.
+ (closes issue ASTERISK-19311) Reported by: tootai Tested by:
+ rmudgett
+
+2012-02-09 16:30 +0000 [r354542] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
+ codes In ASTERISK-18924, SIP INFO DTMF handlingw as changed to
+ account for both lowercase alphatbetic DTMF events, as well as
+ uppercase alphabetic DTMF events. When this occurred, the
+ comparison of the character buffer containing the event code was
+ changed such that the buffer was first compared again '0' and '9'
+ to determine if it was numeric. Unfortunately, since the first
+ character in the buffer will typically be '1' in the case of
+ non-numeric event codes (10-16), this caused those codes to be
+ converted to a DTMF event of '1'. This patch fixes that, and
+ cleans up handling of both application/dtmf-relay and
+ application/dtmf content types. Review:
+ https://reviewboard.asterisk.org/r/1722/ (closes issue
+ ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan
+
+2012-02-09 16:56 +0000 [r354545] Mark Michelson <mmichelson at digium.com>
+
+ * CHANGES, res/res_fax.c: Adding reload support to res_fax.so
+ (closes issue ASTERISK-16712) reported by Frank DiGennaro Review:
+ https://reviewboard.asterisk.org/r/1713
+
+2012-02-09 17:07 +0000 [r354547] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: Clean-up of minor formatting issues in
+ r354542/3/4 rmudgett pointed out some formatting issues in the
+ check-in for ASTERISK-19290. This cleans those up. Review:
+ https://reviewboards.asterisk.org/r/1722/
+
+2012-02-09 17:32 +0000 [r354640-354594] Mark Michelson <mmichelson at digium.com>
+
+ * main/translate.c: Fix translation path choices. This change makes
+ it so computational cost is not taken into account when deciding
+ if a multistep path is better than a single-step path. This means
+ that the only time a multistep path will be chosen is if no
+ single-step path exists. This ensures a better quality
+ translation even if it turns out to be slightly slower. (closes
+ issue ASTERISK-16821) reported by Andrew Lindh Review:
+ https://reviewboard.asterisk.org/r/1715
+
+ * main/translate.c: Remove outdated comment.
+
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