Author: bebuild
Date: Fri Jan 30 15:48:16 2015
New Revision: 431516
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=431516
Log:
Importing files for 13.2.0-rc1 release.
Added:
tags/13.2.0-rc1/.lastclean (with props)
tags/13.2.0-rc1/.version (with props)
tags/13.2.0-rc1/ChangeLog (with props)
tags/13.2.0-rc1/contrib/realtime/mysql/mysql_cdr.sql (with props)
tags/13.2.0-rc1/contrib/realtime/mysql/mysql_config.sql (with props)
tags/13.2.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/13.2.0-rc1/contrib/realtime/oracle/oracle_cdr.sql (with props)
tags/13.2.0-rc1/contrib/realtime/oracle/oracle_config.sql (with props)
tags/13.2.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/13.2.0-rc1/contrib/realtime/postgresql/postgresql_cdr.sql (with
props)
tags/13.2.0-rc1/contrib/realtime/postgresql/postgresql_config.sql (with
props)
tags/13.2.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql
(with props)
tags/13.2.0-rc1/contrib/realtime/sqlserver/mssql_cdr.sql (with props)
tags/13.2.0-rc1/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/13.2.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql (with
props)
Added: tags/13.2.0-rc1/.lastclean
URL:
http://svnview.digium.com/svn/asterisk/tags/13.2.0-rc1/.lastclean?view=auto&rev=431516
==============================================================================
--- tags/13.2.0-rc1/.lastclean (added)
+++ tags/13.2.0-rc1/.lastclean Fri Jan 30 15:48:16 2015
@@ -1,0 +1,1 @@
+40
Propchange: tags/13.2.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/13.2.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/13.2.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/13.2.0-rc1/.version
URL:
http://svnview.digium.com/svn/asterisk/tags/13.2.0-rc1/.version?view=auto&rev=431516
==============================================================================
--- tags/13.2.0-rc1/.version (added)
+++ tags/13.2.0-rc1/.version Fri Jan 30 15:48:16 2015
@@ -1,0 +1,1 @@
+13.2.0-rc1
Propchange: tags/13.2.0-rc1/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/13.2.0-rc1/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/13.2.0-rc1/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/13.2.0-rc1/ChangeLog
URL:
http://svnview.digium.com/svn/asterisk/tags/13.2.0-rc1/ChangeLog?view=auto&rev=431516
==============================================================================
--- tags/13.2.0-rc1/ChangeLog (added)
+++ tags/13.2.0-rc1/ChangeLog Fri Jan 30 15:48:16 2015
@@ -1,0 +1,22801 @@
+2015-01-30 Asterisk Development Team <asteriskt...@digium.com>
+
+ * Asterisk 13.2.0-rc1 Released.
+
+2015-01-30 17:44 +0000 [r431492] Richard Mudgett <rmudg...@digium.com>
+
+ * apps/app_agent_pool.c: app_agent_pool: Fix initial module load
+ agent device state reporting. When the app_agent_pool module
+ initially loads there is a race condition between the thread
+ loading agents.conf and the device state internal processing
+ thread. If the device state internal processing thread handles
+ the agent creation state updates before the thread that loaded
+ agents.conf registers the device state provider callback then the
+ cached agent state is "Invalid". When a consumer module like
+ app_queue asks for the agent state it gets the cached "Invalid"
+ state instead of the real state from the provider. * Moved
+ loading the agents.conf configuration to the last thing setup by
+ app_agent_pool in load_module(). Now the device state provider
+ callback is registered before the config is loaded so the agent
+ creation state updates are guaranteed to get the initial device
+ state. * Removed some now redundant config cleanup on error in
+ load_config(). * Added lock protection when accessing the device
+ state in agent_pvt_devstate_get() and eliminated the RAII_VAR()
+ usage. ASTERISK-24737 #close Reported by: Steve Pitts Review:
+ https://reviewboard.asterisk.org/r/4390/
+
+2015-01-30 17:38 +0000 [r431490] Kevin Harwell <kharw...@digium.com>
+
+ * res/res_pjsip_outbound_publish.c: res_pjsip_outbound_publish:
+ eventually crashes when no response is ever received When
+ Asterisk attempts to send SIP outbound publish information and no
+ response is ever received (no 200 okay, 412, 423) the system
+ eventually crashes. A response is never received because the
+ system Asterisk is attempting to send publish information to is
+ not available. The underlying pjsip framework attempts to send
+ publish information. After several attempts it calls back into
+ the Asterisk outbound publish code. At this point if the
+ "client->queue" is empty Asterisk attempts to schedule a refresh
+ which utilizes "rdata" and since no response was received the
+ given "rdata" struture is NULL. Attempting to dereference a NULL
+ object of course results in a crash. The fix here removes the
+ dependency on rdata for schedule_publish_refresh. Instead
+ param->expiration is now passed to it as this is set to -1 if no
+ response is received. Also added a notification when no response
+ is received. ASTERISK-24635 #close Reported by: Marco Paland
+ Review: https://reviewboard.asterisk.org/r/4384/
+
+2015-01-30 16:52 +0000 [r431471] asanders <asanders@localhost>:
+
+ * include/asterisk/http.h, configs/samples/http.conf.sample,
+ main/http.c: HTTP: For httpd server, need option to define server
+ name for security purposes Added a new config property
+ [servername] to the http.conf file; updated the http server to
+ use the new property when sending responses, for showing http
+ status through the CLI and when reporting status through the
+ 'httpstatus' webpage. ASTERISK-24316 #close Reported By: Andrew
+ Nagy Review: https://reviewboard.asterisk.org/r/4374/
+
+2015-01-30 16:47 +0000 [r431468] Mark Michelson <mmichel...@digium.com>
+
+ * main/stasis_channels.c, channels/chan_pjsip.c, main/xmldoc.c,
+ res/res_pjsip_refer.c, main/pbx.c, main/manager.c,
+ pbx/pbx_spool.c, main/bridge_after.c: Fix some memory leaks.
+ These memory leaks were found and fixed by John Hardin. I'm just
+ committing them for him. ASTERISK-24736 #close Reported by Mark
+ Michelson Review: https://reviewboard.asterisk.org/r/4389
+
+2015-01-29 23:02 +0000 [r431450] Scott Griepentrog <sgriepent...@digium.com>
+
+ * include/asterisk/bridge.h, main/bridge.c,
+ res/stasis/stasis_bridge.c: stasis transfer: fix stasis bridge
+ push race part two When swapping a Local channel in place of one
+ already in a bridge (to complete a bridge attended transfer), the
+ channel that was swapped out can actually be hung up before the
+ stasis bridge push callback executes on the independant transfer
+ thread. This results in the stasis app loop dropping out and
+ removing the control that has the the app name which the local
+ replacement channel needs so it can re-enter stasis. To avoid
+ this race condition a new push_peek callback has been added, and
+ called from the ast_bridge_impart thread before it launches the
+ independant thread that will complete the transfer. Now the
+ stasis push_peek callback can copy the stasis app name before the
+ swap channel can hang up. ASTERISK-24649 Review:
+ https://reviewboard.asterisk.org/r/4382/
+
+2015-01-29 20:58 +0000 [r431420-431426] Mark Michelson <mmichel...@digium.com>
+
+ * res/res_pjsip.c, res/res_pjsip_sips_contact.c (added): Use SIPS
+ URIs in Contact headers when appropriate. RFC 3261 sections
+ 8.1.1.8 and 12.1.1 dictate specific scenarios when we are
+ required to use SIPS URIs in Contact headers. Asterisk's
+ non-compliance with this could actually cause calls to get
+ dropped when communicating with clients that are strict about
+ checking the Contact header. Both of the SIP stacks in Asterisk
+ suffered from this issue. This changeset corrects the behavior in
+ res_pjsip/chan_pjsip.c Review:
+ https://reviewboard.asterisk.org/r/4345
+
+ * /, channels/chan_sip.c: Use SIPS URIs in Contact headers when
+ appropriate. RFC 3261 sections 8.1.1.8 and 12.1.1 dictate
+ specific scenarios when we are required to use SIPS URIs in
+ Contact headers. Asterisk's non-compliance with this could
+ actually cause calls to get dropped when communicating with
+ clients that are strict about checking the Contact header. Both
+ of the SIP stacks in Asterisk suffered from this issue. This
+ changeset corrects the behavior in chan_sip. ASTERISK-24646
+ #close Reported by Stephan Eisvogel Review:
+ https://reviewboard.asterisk.org/r/4346 ........ Merged revisions
+ 431423 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_pjsip/pjsip_configuration.c: Allow disabling of 100rel
+ support on PJSIP endpoints. Due to an inversion error, setting
+ 100rel=no would not actually change the current value of the
+ setting (which defaulted to "yes"). With this fix, the inversion
+ is corrected.
+
+2015-01-29 16:46 +0000 [r431403] George Joseph <george.jos...@fairview5.com>
+
+ * res/res_pjsip_exten_state.c: res_pjsip_exten_state: Reduce log
+ clutter... change a WARNING to a VERBOSE/2 Reduce log clutter by
+ changing the "Watcher for hint %s (removed|deactivated)" message
+ from WARNING to VERBOSE/2. Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4387/
+
+2015-01-29 12:09 +0000 [r431385] Joshua Colp <jc...@digium.com>
+
+ * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix DTLS when used
+ with OpenSSL 1.0.1k A recent security fix for OpenSSL broke DTLS
+ negotiation for many applications. This was caused by read ahead
+ not being enabled when it should be. While a commit has gone into
+ OpenSSL to force read ahead on for DTLS it may take some time for
+ a release to be made and the change to be present in
+ distributions (if at all). As enabling read ahead is a simple one
+ line change this commit does that and fixes the issue.
+ ASTERISK-24711 #close Reported by: Jared Biel ........ Merged
+ revisions 431384 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-01-28 17:37 +0000 [r431301-431303] Mark Michelson <mmichel...@digium.com>
+
+ * /, res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c,
+ res/res_pjsip_session.c: Fix file descriptor leak in RTP code.
+ SIP requests that offered codecs incompatible with configured
+ values could result in the allocation of RTP and RTCP ports that
+ would not get reclaimed later. ASTERISK-24666 #close Reported by
+ Y Ateya Review: https://reviewboard.asterisk.org/r/4323
+ AST-2015-001 ........ Merged revisions 431300 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+ * funcs/func_curl.c, /: Multiple revisions 431297-431298 ........
+ r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan
+ 2015) | 17 lines Mitigate possible HTTP injection attacks using
+ CURL() function in Asterisk. CVE-2014-8150 disclosed a
+ vulnerability in libcURL where HTTP request injection can be
+ performed given properly-crafted URLs. Since Asterisk makes use
+ of libcURL, and it is possible that users of Asterisk may get
+ cURL URLs from user input or remote sources, we have made a patch
+ to Asterisk to prevent such HTTP injection attacks from
+ originating from Asterisk. ASTERISK-24676 #close Reported by Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/4364
+ AST-2015-002 ........ r431298 | mmichelson | 2015-01-28 11:12:49
+ -0600 (Wed, 28 Jan 2015) | 3 lines Fix compilation error from
+ previous patch. ........ Merged revisions 431297-431298 from
+ http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+ revisions 431299 from
+ http://svn.asterisk.org/svn/asterisk/branches/12
+
+2015-01-28 12:18 +0000 [r431267] Sean Bright <s...@malleable.com>
+
+ * res/res_format_attr_silk.c, res/res_format_attr_opus.c: media
+ formats: update res_format_attr_opus & silk In r419044, we
+ changed how formats were handled, but the return value of the
+ format_parse_sdp_fmtp functions in res_format_attr_opus and
+ res_format_attr_silk were not updated, causing calls to fail. Ran
+ into this when getting codec_opus working with Asterisk 13. Once
+ the return value was corrected, we were crashing in opus_getjoint
+ because of NULL format attributes. I've fixed this as well in
+ this patch. Review: https://reviewboard.asterisk.org/r/4371/
+
+2015-01-28 04:09 +0000 [r431243] Richard Mudgett <rmudg...@digium.com>
+
+ * main/sorcery.c, res/res_pjsip_outbound_registration.c,
+ res/res_pjsip.c: res_pjsip_outbound_registration: Fix reload race
+ condition. Performing a CLI "module reload" command when there
+ are new pjsip.conf registration objects defined frequently failed
+ to load them correctly. What happens is a race condition between
+ res_pjsip pushing its reload into an asynchronous task processor
+ task and the thread that does the rest of the reloads when it
+ gets to reloading the res_pjsip_outbound_registration module. A
+ similar race condition happens between a reload and the CLI/AMI
+ show registrations commands. The reload updates the
+ current_states container and the CLI/AMI commands call
+ get_registrations() which builds a new current_states container.
+ * Made res_pjsip.c reload_module() use
+ ast_sip_push_task_synchronous() instead of ast_sip_push_task() to
+ eliminate two threads processing config reloads at the same time.
+ * Made get_registrations() not replace the global current_states
+ container so the CLI/AMI show registrations command cannot
+ interfere with reloading. You could never add/remove objects in
+ the container without the possibility of the container being
+ replaced out from under you by get_registrations(). * Added a
+ registration loaded sorcery instance observer to purge any dead
+ registration objects since get_registrations() cannot do this job
+ anymore. The struct ast_sorcery_instance_observer callbacks must
+ be used because the callback happens inline with the load
+ process. The struct ast_sorcery_observer callbacks are pushed to
+ a different thread. * Added some global current_states NULL
+ pointer checks in case the container disappears because of
+ unload_module(). * Made sorcery's struct
+ ast_sorcery_instance_observer.object_type_loaded callbacks
+ guaranteed to be called before any struct
+ ast_sorcery_observer.loaded callbacks will be called. * Moved the
+ check for non-reloadable objects to before the sorcery instance
+ loading callbacks happen to short circuit unnecessary work.
+ Previously with non-reloadable objects, the sorcery instance
+ loading/loaded callbacks would always happen, the individual
+ wizard loading/loaded would be prevented, and the non-reloadable
+ type logging message would be logged for each associated wizard.
+ ASTERISK-24729 #close Review:
+ https://reviewboard.asterisk.org/r/4381/
+
+2015-01-27 22:56 +0000 [r431179-431219] Kevin Harwell <kharw...@digium.com>
+
+ * /, main/tcptls.c: tcptls: Bad file descriptor error when
+ reloading chan_sip While running through some scenarios using
+ chan_sip and tcp a problem would occur that resulted in a flood
+ of bad file descriptor messages on the cli: tcptls.c:712
+ ast_tcptls_server_root: Accept failed: Bad file descriptor The
+ message is received because the underlying socket has been
+ closed, so is valid. This is probably happening because unloading
+ of chan_sip is not atomic. That however is outside the scope of
+ this patch. This patch simply stops the logging of multiple
+ occurrences of that message. ASTERISK-24728 #close Reported by:
+ Thomas Thompson Review: https://reviewboard.asterisk.org/r/4380/
+ ........ Merged revisions 431218 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: chan_sip: stale nonce causes failure When
+ refreshing (with a small expiration) a registration that was sent
+ to chan_sip the nonce would be considered stale and reject the
+ registration. What was happening was that the initial
+ registration's "dialog" still existed in the dialogs container
+ and upon refresh the dialog match algorithm would choose that as
+ the "dialog" instead of the newly created one. This occurred
+ because the algorithm did not check to see if the from tag
+ matched if authentication info was available after the 401. So,
+ it ended up assuming the original "dialog" was a match and
+ stopped the search. The old "dialog" of course had an old nonce,
+ thus the stale nonce message. This fix attempts to leave the
+ original functionality alone except in the case of a REGISTER. If
+ a REGISTER is received if searches for an existing "dialog"
+ matching only on the callid. If the expires value is low enough
+ it will reuse dialog that is there, otherwise it will create a
+ new one. ASTERISK-24715 #close Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/4367/ ........ Merged
+ revisions 431187 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_pjsip/pjsip_outbound_auth.c, res/res_pjsip/config_auth.c,
+ main/stasis_message_router.c, res/res_pjsip/location.c,
+ res/res_pjsip/pjsip_configuration.c,
+ res/res_pjsip/pjsip_distributor.c,
+ res/res_pjsip/include/res_pjsip_private.h,
+ res/res_pjsip/pjsip_global_headers.c,
+ res/res_pjsip/pjsip_options.c, res/res_pjsip.c,
+ res/res_pjsip/config_transport.c: res_pjsip: make it unloadable
+ (take 2) Due to the original patch causing memory corruptions it
+ was removed until the problem could be resolved. This patch is
+ the original patch plus some added locking around stasis router
+ subcription that was needed to avoid the memory corruption.
+ Description of the original problem and patch (still applicable):
+ The res_pjsip module was previously unloadable. With this patch
+ it can now be unloaded. This patch is based off the original
+ patch on the issue (listed below) by Corey Farrell with a few
+ modifications. Namely, removed a few changes not required to make
+ the module unloadable and also fixed a bug that would cause
+ asterisk to crash on unloading. This patch is the first step
+ (should hopefully be followed by another/others at some point) in
+ allowing res_pjsip and the modules that depend on it to be
+ unloadable. At this time, res_pjsip and some of the modules that
+ depend on res_pjsip cannot be unloaded without causing problems
+ of some sort. The goal of this patch is to get res_pjsip and only
+ res_pjsip to be able to unload successfully and/or shutdown
+ without incident (crashes, leaks, etc...). Other dependent
+ modules may still cause problems on unload. Basically made sure,
+ with the patch applied, that res_pjsip (with no other dependent
+ modules loaded) could be succesfully unloaded and Asterisk could
+ shutdown without any leaks or crashes that pertained directly to
+ res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell
+ Review: https://reviewboard.asterisk.org/r/4363/ patches:
+ pjsip_unload-broken-r1.patch submitted by Corey Farrell (license
+ 5909)
+
+2015-01-27 17:36 +0000 [r431160] Richard Mudgett <rmudg...@digium.com>
+
+ * /, apps/confbridge/include/confbridge.h, apps/app_confbridge.c:
+ app_confbridge: Repeatedly starting and stopping recording ref
+ leaks the recording channel. Starting and stopping conference
+ recording more than once causes the recording channels to be
+ leaked. For v13 the channels also show up in the CLI "core show
+ channels" output. * Reworked and simplified the recording channel
+ code to use ast_bridge_impart() instead of managing the recording
+ thread in the ConfBridge code. The recording channel's ref
+ handling easily falls into place and other off nominal code paths
+ get handled better as a result. ASTERISK-24719 #close Reported
+ by: John Bigelow Review: https://reviewboard.asterisk.org/r/4368/
+ Review: https://reviewboard.asterisk.org/r/4369/ ........ Merged
+ revisions 431135 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-01-27 17:32 +0000 [r431157] Joshua Colp <jc...@digium.com>
+
+ * main/bridge_channel.c, res/res_pjsip_sdp_rtp.c: bridge /
+ res_pjsip_sdp_rtp: Fix issues with media not being reinvited
+ during direct media. This change fixes two issues: 1. During a
+ swap operation bridging added the new channel before having the
+ swap channel leave. This was not handled in bridge_native_rtp and
+ could result in a channel not getting reinvited back to Asterisk.
+ After this change the swap channel will leave first and the new
+ channel will then join. 2. If a re-invite was received after a
+ session had been established any upstream elements (such as
+ bridge_native_rtp) were not notified that they may want to
+ re-evaluate things. After this change an UPDATE_RTP_PEER control
+ frame is queued when this situation occurs and upstream can
+ react. AST-1524 #close Review:
+ https://reviewboard.asterisk.org/r/4378/
+
+2015-01-27 17:22 +0000 [r431153] Jonathan Rose <jr...@digium.com>
+
+ * main/manager.c: Manager: Fix Manager Action ModuleLoad to give
+ correct response when reloading Prior to this patch, ModuleLoad
+ would respond with an error indicating that the requested module
+ wasn't found in spite of finding and reloading the module.
+ Review: https://reviewboard.asterisk.org/r/4373/ ASTERISK-24721
+ #close
+
+2015-01-27 17:20 +0000 [r431134-431145] Matthew Jordan <mjor...@digium.com>
+
+ * res/ari/resource_bridges.c,
+ rest-api-templates/asterisk_processor.py,
+ res/ari/resource_channels.h, res/res_ari_bridges.c,
+ res/ari/resource_bridges.h, rest-api-templates/api.wiki.mustache,
+ rest-api/api-docs/channels.json,
+ rest-api-templates/swagger_model.py,
+ rest-api/api-docs/bridges.json: ARI: Improve wiki documentation
+ This patch improves the documentation of ARI on the wiki.
+ Specifically, it addresses the following: * Allowed values and
+ allowed ranges weren't documented. This was particularly
+ frustrating, as Asterisk would reject query parameters with
+ disallowed values - but we didn't tell anyone what the allowed
+ values were. * The /play/id operation on /channels and /bridges
+ failed to document all of the added media resource types. *
+ Documentation for creating a channel into a Stasis application
+ failed to note when it occurred, and that creating a channel into
+ Stasis conflicts with creating a channel into the dialplan. *
+ Some other minor tweaks in the mustache templates, including
+ italicizing the parameter type, putting the default value on its
+ own sub-bullet, and some other nicities. Review:
+ https://reviewboard.asterisk.org/r/4351
+
+ * apps/confbridge/conf_config_parser.c,
+ apps/confbridge/include/confbridge.h: app_confbridge: Restore
+ user's menu name to CLI output of 'confbridge list' When issuing
+ a 'confbridge list XXXX' CLI command, the resulting output no
+ longer displays the menu associated with a ConfBridge
+ participant. The issue was caused by ASTERISK-22760. When that
+ patch was done, it removed the copying of the menu name
+ associated with the user from the actual user profile. This patch
+ fixes the issue by copying the menu name over to the user profile
+ when the menu hooks are applied to the user. Since that function
+ now does a little bit more than just apply the hooks, the name of
+ the function has been changed to cover the copying of the menu
+ name over as well. In addition, there is a disparity between the
+ menu name length as it is stored on the conf_menu structure and
+ the confbridge_user structure; this patch makes the lengths match
+ so that a strcpy can be used. Review:
+ https://reviewboard.asterisk.org/r/4372/ ASTERISK-24723 #close
+ Reported by: Steve Pitts
+
+2015-01-27 11:47 +0000 [r431114] Joshua Colp <jc...@digium.com>
+
+ * res/parking/parking_manager.c: res_parking: Fix crash due to race
+ condition when unloading. There is currently a race condition
+ when unloading the res_parking module. Depending on the will of
+ the universe the subscription invocation may occur AFTER the
+ module is unloaded. This is because the module does NOT use
+ stasis_unsubscribe_and_join when terminating the subscription. It
+ merely uses stasis_unsubscribe. This change makes it use
+ stasis_unsubscribe_and_join which is documented for usage in this
+ exact scenario. AST-1520 #close Review:
+ https://reviewboard.asterisk.org/r/4375/
+
+2015-01-26 14:49 +0000 [r431092] David M. Lee <d...@digium.com>
+
+ * channels/sip/include/route.h, funcs/func_presencestate.c,
+ main/rtp_engine.c, configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/sem.h, configure.ac, main/app.c,
+ main/bridge_channel.c, main/sem.c, res/res_timing_kqueue.c,
+ main/asterisk.c: Various fixes for OS X This patch addresses
+ compilation errors on OS X. It's been a while, so there's quite a
+ few things. * Fixed __attribute__ decls in route.h to be
+ portable. * Fixed htonll and ntohll to work when they are defined
+ as macros. * Replaced sem_t usage with our ast_sem wrapper. *
+ Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC
+ 4.9 warnings using sig*set() functions. * Fixed some format
+ strings for portability. * Fixed compilation issues with
+ res_timing_kqueue (although tests still fail on OS X). * Fixed
+ menuconfig /sbin/launchd detection, which disables
+ res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by:
+ George Joseph ASTERISK-24544 #close Reported by: George Joseph
+ Review: https://reviewboard.asterisk.org/r/4327/
+
+2015-01-25 13:42 +0000 [r431072] Matthew Jordan <mjor...@digium.com>
+
+ * main/config.c: dynamic realtime: Updates fail to work due to
+ update fields being passed over When a crash was fixed due to
+ usage of the REALTIME function in r423003, a regression was
+ introduced into ast_update2_realtime where the update fields
+ passed to the function would be skipped and the lookup field
+ processed twice. The use of this function is a bit interesting: A
+ variable argument list is used with two sentinel values - the
+ first marks the end of the lookup fields/values; the second marks
+ the end of the update fields/values. Unfortunately,
+ ast_update2_realtime parses over the lookup fields twice, as
+ opposed to parsing over the update fields. This causes the
+ lookups to succeed, but the updates itself to have no effect.
+ Note that the most common instance of this problem occurred in
+ app_voicemail during the updating of a mailbox password. Thanks
+ to the issue reporter, Paddy Grice, for pointing out the problem.
+ Review: https://reviewboard.asterisk.org/r/4356/ ASTERISK-24231
+ ASTERISK-24626 #close Reported by: Paddy Grice
+
+2015-01-23 20:13 +0000 [r431050-431052] Richard Mudgett <rmudg...@digium.com>
+
+ * apps/confbridge/conf_chan_record.c: app_confbridge: Make CBRec
+ channel names more unique. Channel names should be different from
+ other channels in the system while the channel exists. * Use a
+ sequence number for CBRec channels instead of a random number
+ because the same random number could be picked again for the next
+ CBRec channel.
+
+ * /, apps/app_confbridge.c: app_confbridge: Whitespace Because
+ there is sometimes no sence to any whitespace. ........ Merged
+ revisions 431049 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-01-23 17:08 +0000 [r431030] David M. Lee <d...@digium.com>
+
+ * res/res_pjsip_config_wizard.c: Add depend on pjproject to
+ res_pjsip_config_wizard.c
+
+2015-01-23 15:12 +0000 [r430999] Kevin Harwell <kharw...@digium.com>
+
+ * res/parking/parking_applications.c, channels/chan_iax2.c,
+ res/res_pjsip/pjsip_global_headers.c, res/res_pjsip_pubsub.c,
+ res/res_ari_channels.c, res/res_stasis.c,
+ rest-api-templates/param_parsing.mustache,
+ res/res_ari_endpoints.c, res/res_ari_events.c,
+ include/asterisk/stasis_app.h, res/res_pjsip_mwi.c: Investigate
+ and fix memory leaks in Asterisk Fixed memory leaks that were
+ found in Asterisk. ASTERISK-24693 #close Reported by: Kevin
+ Harwell Review: https://reviewboard.asterisk.org/r/4347/
+
+2015-01-23 15:03 +0000 [r430994-430998] Walter Doekes <walter+aster...@wjd.nu>
+
+ * apps/app_voicemail.c, channels/chan_unistim.c,
+ funcs/func_hangupcause.c, main/manager_bridges.c,
+ channels/chan_misdn.c, funcs/func_groupcount.c, /,
+ addons/ooh323c/src/ooh245.c, channels/chan_sip.c, res/res_fax.c,
+ res/res_pjsip_outbound_registration.c, apps/app_minivm.c,
+ apps/app_alarmreceiver.c, include/asterisk/channel.h,
+ contrib/utils/eagi_proxy.c: Fix typo's (retrieve, specified,
+ address). ........ Merged revisions 430996 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, channels/chan_sip.c: chan_sip: Case insensitive comparison of
+ "defaultuser" parameter. All the other configuration options are
+ case insensitive, so this one should be too. ASTERISK-24355
+ #close Reported by: HZMI8gkCvPpom0tM patches: ast.patch uploaded
+ by HZMI8gkCvPpom0tM (License 6658) ........ Merged revisions
+ 430993 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-01-22 19:24 +0000 [r430957-430975] Richard Mudgett <rmudg...@digium.com>
+
+ * include/asterisk/bridge.h,
+ include/asterisk/bridge_channel_internal.h, main/bridge.c,
+ include/asterisk/bridge_internal.h, main/bridge_channel.c: Bridge
+ core: Pass a ref with the swap channel when joining a bridge.
+ When code imparts a channel into a bridge to swap with another
+ channel, a ref needs to be held on the swap channel to ensure
+ that it cannot dissapear before finding it in the bridge. * The
+ ast_bridge_join() swap channel parameter now always steals a ref
+ for the swap channel. This is the only change to the bridge
+ framework's public API semantics. *
+ bridge_channel_internal_join() now requires the
+ bridge_channel->swap channel to pass in a ref. ASTERISK-24649
+ Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/4354/
+
+ * res/res_pjsip_outbound_registration.c:
+ res_pjsip_outbound_registration.c: Minor code cleanup. * Add an
+ allocation failure check and assert in
+ sip_outbound_registration_response_cb(). * Made
+ sip_outbound_registration_state_destroy() handle partially
+ created state objects from
+ sip_outbound_registration_state_alloc(). Review:
+ https://reviewboard.asterisk.org/r/4366/
+
+2015-01-22 18:09 +0000 [r430939] Scott Griepentrog <sgriepent...@digium.com>
+
+ * res/stasis/app.c, res/stasis/stasis_bridge.c: stasis transfer:
+ fix a race condition on stasis bridge push After a bridge
+ transfer completes where a local replacement channel is used, a
+ stasis transfer message with the details of the transfer is sent.
+ This is processed by stasis which then sets the stasis app name
+ and replaced channel snapshot on the replacement channel.
+ However, since a separate thread was already started to run
+ stasis on the new replacement channel, a race was on to see if
+ the message processing would be completed before the app name was
+ needed, otherwise the channel would be hung up. This change moves
+ the calls used to set the stasis app name and the replace
+ snapshot to the bridge_stasis_push function callback from the
+ bridge transfer logic, allowing the steps to be completed earlier
+ and more deterministically, and the race elimianted. NOTE: the
+ swap channel parameter to bridge_stasis_push (and thus all bridge
+ push callbacks) must always be present when performing a swap
+ with another channel. ASTERISK-24649 #close Reported by: John
+ Bigelow Review: https://reviewboard.asterisk.org/r/4341/
+
+2015-01-22 14:23 +0000 [r430921] Matthew Jordan <mjor...@digium.com>
+
+ * /, apps/app_voicemail.c: apps/app_voicemail: Trigger MWI
+ notification with MixMonitor m() option The MixMonitor m() option
+ allows a recording to be pushed to a specific voicemail mailbox.
+ If the message is delivered to the mailbox's INBOX, however, no
+ MWI notification is currently raised. This patch corrects the
+ issue by properly calling notify_new_state from the
+ msg_create_from_file function. This will cause MWI to be
+ triggered if the message was placed in the mailbox's INBOX.
+ ASTERISK-24709 #close Reported by: Gareth Palmer patches:
+ app_voicemail-430919.patch uploaded by Gareth Palmer (License
+ 5169) ........ Merged revisions 430920 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2015-01-21 21:53 +0000 [r430902] Richard Mudgett <rmudg...@digium.com>
+
+ * res/res_pjsip_outbound_registration.c:
+ res_pjsip_outbound_registration.c: Move unref to a better place.
+ Move an unconditional unref of client_state so it doesn't look
+ like it could be used after the last ref has destroyed it.
+
+2015-01-21 13:33 +0000 [r430840-430864] Matthew Jordan <mjor...@digium.com>
+
+ * channels/chan_sip.c: channels/chan_sip: Fix registration leak
+ during reload When the SIP registrations were migrated to using
+ ao2 in what was then trunk, the explicit destruction of the
+ registrations on module reload was removed and not replaced with
+ an ao2 equivalent. Debugging done by Stefan Engström, the issue
+ reporter, on ASTERISK-24673 confirmed that the reference in the
+ registry_list container was being leaked. Since the purpose of
+ cleanup_all_regs is to prep a registration for destruction, this
+ function now calls an ao2_callback function callback with the
+ OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the
+ registrations. This cleans up each registration, and also removes
+ it from the registration container registry_list. Review:
+ https://reviewboard.asterisk.org/r/4355/ ASTERISK-24640 #close
+ Reported by: Max Man ASTERISK-24673 #close Reported by: Stefan
+ Engström Tested by: Stefan Engström
+
+ * cdr/cdr_manager.c, cel/cel_manager.c: AMI: Add documentation for
+ the missing Cdr/CEL events. This patch adds AMI event
+ documentation for the Cdr and CEL AMI events. Note that while
+ these events do share fields with each other and with other
+ channel related events, they do not contain all of the fields in
+ a standard channel snapshot, nor is the description of the fields
+ identical. As such, the patch opts for documentation for each
+ field, for each event. Review:
+ https://reviewboard.asterisk.org/r/4350/ ASTERISK-24671 #close
+ Reported by: Dan Jenkins
+
+ * apps/app_dial.c: apps/app_dial: Don't publish DialEnd twice on
+ unexpected GoSub/Macro values The Dial application has some
+ interesting options with the mid-call Macro (M) and GoSub (U)
+ options. If the MACRO_RESULT/GOSUB_RESULT returns specific
+ values, the Dial application will take some action upon the
+ channels involved in the dial operation (such as hanging up a
+ particular party, etc.) The Dial application ensures that a
+ Stasis message is published in the event that
+ MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial
+ operation, so that there is a corresponding DialEnd event
+ published in AMI/ARI for the DialBegin event that preceeded it. A
+ bug exists where that same DialEnd event will be published on
+ Stasis even if the value returned in MACRO_RESULT/GOSUB_RESULT is
+ not one that the Dial application cares about. This causes two
+ DialEnd events to be published - one with the
+ MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is
+ all sorts of wrong. This patch fixes the bug by ensuring that we
+ only publish a DialEnd message to Stasis if the Dial
+ application's mid-call Macro/GoSub returns something that Dial
+ cares about. Review: https://reviewboard.asterisk.org/r/4336
+ ASTERISK-24682 #close Reported by: Matt Jordan
+
+ * main/rtp_engine.c: main/rtp_engine: Format NTP timestamps as
+ unsigned longs When the RTCP reports are created, the NTP
+ timestamps are stored as strings, as JSON does not have an
+ integer type long enough to store the value. However, on 32-bit
+ systems, a signed long may overflow for some portion of the
+ timestamp. This patch corrects the overflow by formatting the
+ timestamps as unsigned longs.
+
+2015-01-20 16:51 +0000 [r430818] asanders <asanders@localhost>:
+
+ * res/ari/resource_bridges.c: ARI: Fixed crash that occurred when
+ updating a bridge when the optional query parameter 'name' was
+ not supplied. Prior to this changeset, posting to the:
+ /ari/bridges/{bridgeId} endpoint without specifying a value for
+ the [name] query parameter, would crash Asterisk if the bridge
+ you are attempting to create (or update) had the same ID as an
+ existing bridge. The internal mechanism of the POST operation
+ interpreted a null value for name, thus resulting in an error
+ condition that crashed Asterisk. ASTERISK-24560 #close Reported
+ By: Kinsey Moore Review: https://reviewboard.asterisk.org/r/4349/
+
+2015-01-20 16:46 +0000 [r430817] Richard Mudgett <rmudg...@digium.com>
+
+ * configs/samples/iax.conf.sample, res/res_fax.c,
+ funcs/func_channel.c, UPGRADE.txt, res/snmp/agent.c,
+ channels/chan_iax2.c: CHANNEL(peer), chan_iax2, res_fax, SNMP
+ agent: Fix deadlock from reaching across a bridge. Calling
+ ast_channel_bridge_peer() cannot be done while holding any
+ channel locks. The reported issue hit the deadlock in chan_iax2,
+ but an audit of the ast_channel_bridge_peer() calls found three
+ more locations where the same deadlock can occur. * Made
+ CHANNEL(peer), res_fax, and the SNMP agent not call
+ ast_channel_bridge_peer() with any channel locked. For
+ CHANNEL(peer) I had to rework the logic to not hold the channel
+ lock. * Made chan_iax2 no longer call ast_channel_bridge_peer().
+ It was done for legacy reasons that no longer apply. * Removed
+ the iax.conf forcejitterbuffer option. It is now always enabled
+ when the jitterbuffer option is enabled. If you put a jitter
+ buffer on a channel it will be on the channel. ASTERISK-24600
+ #close Reported by: Jeff Collell Review:
+ https://reviewboard.asterisk.org/r/4342/
+
+2015-01-20 02:39 +0000 [r430796-430799] Matthew Jordan <mjor...@digium.com>
+
+ * contrib/scripts/install_prereq, /:
+ contrib/scripts/install_prereq: Don't install 32-bit packages on
+ 64-bit hosts On Debian based systems, the install_prereq tool
+ uses a search command on Debian that results in selecting both
+ 64-bit and 32-bit packages. Besides the waste of disk space, this
+ can actually cause aptitude use 100% of memory on a VM with 1GB
+ of RAM as it tried to work out all of the 32-bit package
+ dependencies. This patch filters out the 32-bit packages on a
+ 64-bit machine, and leaves 32-bit machines alone. ASTERISK-24048
+ #close Reported by: Ben Klang Tested by: Ben Klang, Matt Jordan
+ patches: install_prereq_64-bit_compat.patch uploaded by Ben Klang
+ (License 5876) ........ Merged revisions 430798 from
[... 26168 lines stripped ...]
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
svn-commits mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/svn-commits
