[asterisk-commits] bebuild: tag 12.7.0-rc1 r427147 - in /tags/12.7.0-rc1: ./ contrib/realtime/my...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 3 12:53:39 CST 2014
Author: bebuild
Date: Mon Nov 3 12:53:33 2014
New Revision: 427147
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=427147
Log:
Importing files for 12.7.0-rc1 release.
Added:
tags/12.7.0-rc1/.lastclean (with props)
tags/12.7.0-rc1/.version (with props)
tags/12.7.0-rc1/ChangeLog (with props)
tags/12.7.0-rc1/contrib/realtime/mysql/mysql_cdr.sql (with props)
tags/12.7.0-rc1/contrib/realtime/mysql/mysql_config.sql (with props)
tags/12.7.0-rc1/contrib/realtime/mysql/mysql_voicemail.sql (with props)
tags/12.7.0-rc1/contrib/realtime/oracle/oracle_cdr.sql (with props)
tags/12.7.0-rc1/contrib/realtime/oracle/oracle_config.sql (with props)
tags/12.7.0-rc1/contrib/realtime/oracle/oracle_voicemail.sql (with props)
tags/12.7.0-rc1/contrib/realtime/postgresql/postgresql_cdr.sql (with props)
tags/12.7.0-rc1/contrib/realtime/postgresql/postgresql_config.sql (with props)
tags/12.7.0-rc1/contrib/realtime/postgresql/postgresql_voicemail.sql (with props)
tags/12.7.0-rc1/contrib/realtime/sqlserver/mssql_cdr.sql (with props)
tags/12.7.0-rc1/contrib/realtime/sqlserver/mssql_config.sql (with props)
tags/12.7.0-rc1/contrib/realtime/sqlserver/mssql_voicemail.sql (with props)
Added: tags/12.7.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/12.7.0-rc1/.lastclean?view=auto&rev=427147
==============================================================================
--- tags/12.7.0-rc1/.lastclean (added)
+++ tags/12.7.0-rc1/.lastclean Mon Nov 3 12:53:33 2014
@@ -1,0 +1,1 @@
+40
Propchange: tags/12.7.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/12.7.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/12.7.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/12.7.0-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/12.7.0-rc1/.version?view=auto&rev=427147
==============================================================================
--- tags/12.7.0-rc1/.version (added)
+++ tags/12.7.0-rc1/.version Mon Nov 3 12:53:33 2014
@@ -1,0 +1,1 @@
+12.7.0-rc1
Propchange: tags/12.7.0-rc1/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/12.7.0-rc1/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/12.7.0-rc1/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/12.7.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/12.7.0-rc1/ChangeLog?view=auto&rev=427147
==============================================================================
--- tags/12.7.0-rc1/ChangeLog (added)
+++ tags/12.7.0-rc1/ChangeLog Mon Nov 3 12:53:33 2014
@@ -1,0 +1,31985 @@
+2014-11-03 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 12.7.0-rc1 Released.
+
+2014-11-03 17:54 +0000 [r427129] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_pjsip.c, configs/pjsip.conf.sample,
+ res/res_pjsip/config_system.c, UPGRADE.txt: res_pjsip: Add
+ disable_tcp_switch option. When a packet exceeds the MTU,
+ pjproject will switch from UDP to TCP. In some circumstances (on
+ some networks), this can cause some issues with messages not
+ getting sent to the correct destination - and can also cause
+ connections to get dropped due to quirks in pjproject deciding to
+ terminate TCP connections with no messages. While fixing the
+ routing/messaging issues is important, having a configuration
+ option in Asterisk that tells pjproject to not switch over to TCP
+ would be useful. That way, if some glitch is discovered on some
+ other network/site, we can at least disable the behavior until a
+ fix is put into place. AFS-197 #close Review:
+ https://reviewboard.asterisk.org/r/4137/
+
+2014-11-03 02:33 +0000 [r427020-427088] Corey Farrell <git at cfware.com>
+
+ * apps/app_voicemail.c, /: Fix compile error caused by review 4138
+ There is no procedure called ast_closeframe, fix code to use
+ ast_closestream. Reported By: Matt Jordan ........ Merged
+ revisions 427087 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_voicemail.c, /, main/app.c: Fix ast_writestream leaks
+ Fix cleanup in __ast_play_and_record where others[x] may be
+ leaked. This was caught where prepend != NULL && outmsg != NULL,
+ once realfile[x] == NULL any further others[x] would be leaked. A
+ cleanup block was also added for prepend != NULL && outmsg ==
+ NULL. 11+: Fix leak of ast_writestream recording_fs in
+ app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by:
+ Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/
+ ........ Merged revisions 427023 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 427024 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, main/abstract_jb.c: func_jitterbuffer: fix frame leaks. Fix
+ code paths where it is possible for frames to leak. Fix
+ uninitialized variable in jb_get_fixed and jb_get_adaptive.
+ ASTERISK-22409 #related Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4128/ ........ Merged
+ revisions 427019 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-11-02 01:01 +0000 [r426995] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_stasis.c: res/res_stasis: Fix crash on module unload
+ while performing operation When the res_stasis module is
+ unloaded, it will dispose of the apps_registry container. This is
+ a problem if an ARI operation is in flight that attempts to use
+ the registry, as the shutdown occurs in a separate thread. This
+ patch adds some sanity checks to the various routines that access
+ the registry which cause the operations to fail if the
+ apps_registry does not exist. Crash caught by the Asterisk Test
+ Suite.
+
+2014-10-31 16:46 +0000 [r426933] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * Makefile, /: install init.d files on GNU/kFreeBSD Review:
+ https://reviewboard.asterisk.org/r/4118/ ........ Merged
+ revisions 426926 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 426927 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-31 16:33 +0000 [r426923-426928] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * configs/pjsip.conf.sample, res/res_pjsip.c: pjsip: clarify tls
+ cert and key file usage A question arose as to whether a .pem
+ file could be provided in place of the .crt and .key files in a
+ PJSIP TLS configuration. I tested this and discovered that
+ although a cert will be read from the pem file, a key will not,
+ and thus the priv_key_file entry is still required. This update
+ to the fine documentation clarifies the option usage. AST-1448
+ #close Review: https://reviewboard.asterisk.org/r/4129/ Reported
+ by: John Bigelow
+
+ * res/res_pjsip_outbound_registration.c: pjsip: Handle outbound
+ unregister correctly This updates the status of the outbound
+ registration to reflect when it has been unregistered. Since the
+ registration is unregistered but is not stopped, the registration
+ schedule remains active as before. The patch also updates the
+ documentation of both the AMI and CLI commands. ASTERISK-24411
+ #close Review: https://reviewboard.asterisk.org/r/4119/ Reported
+ by: John Bigelow patches: unregister-patch1.txt uploaded by John
+ Bigelow (License 5091)
+
+2014-10-31 03:25 +0000 [r426863] Matthew Jordan <mjordan at digium.com>
+
+ * channels/sip/include/reqresp_parser.h, /,
+ channels/sip/reqresp_parser.c: channels/sip/reqresp_parser: Fix
+ unit tests for r426594 When r426594 was made, it did not take
+ into account a unit test that verified that the function properly
+ populated the unsupported buffer. The function would previously
+ memset the buffer if it detected it had any contents; since this
+ function can now be called iteratively on successive headers, the
+ unit tests would now fail. This patch updates the unit tests to
+ reset the buffer themselves between successive calls, and updates
+ the documentation of the function to note that this is now
+ required. ........ Merged revisions 426858 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 426860 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-31 03:06 +0000 [r426806-426832] Corey Farrell <git at cfware.com>
+
+ * contrib/Makefile (added), Makefile, /: REF_DEBUG: Install
+ refcounter.py to $(ASTDATADIR)/scripts This change ensures
+ refcounter.py is installed to a place where it can be found by
+ the Asterisk testsuite if REF_DEBUG is enabled. ASTERISK-24432
+ #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4094/ ........ Merged
+ revisions 426830 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 426831 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * apps/app_queue.c, /: app_queue: fix a couple leaks to struct
+ call_queue in set_member_value set_member_value has a couple
+ leaks to references in the variable q found through testsuite
+ tests/queues/set_penalty. Also remove the REF_DEBUG_ONLY_QUEUES
+ compiler declaration, this is no longer possible with the updated
+ REF_DEBUG code. ASTERISK-24466 #close Reported by: Corey Farrell
+ Review: https://reviewboard.asterisk.org/r/4125/ ........ Merged
+ revisions 426805 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-30 21:12 +0000 [r426755-426779] Kevin Harwell <kharwell at digium.com>
+
+ * res/res_pjsip_exten_state.c: res_pjsip_exten_state:
+ PJSIPShowSubscriptionsInbound causes crash Currently, it is
+ possible for some subscriptions to get into a NULL state. When
+ this occurs and the PJSIPShowSubscriptionsInbound ami action is
+ issued and a device is subscribed for extension state then the
+ associated subscription state object can't be located. The code
+ then attempts to dereference a NULL object. Added a NULL check to
+ avoid the problem. Reported by: John Bigelow
+
+ * res/res_pjsip/pjsip_options.c: res_pjsip: incorrect qualify
+ statistics after disabling for contact When removing the
+ qualify_frequency from an AoR or a contact the statistics shown
+ when issuing "pjsip show aors" from the CLI are incorrect. This
+ patch deletes the contact's status object from sorcery,
+ disassociating it from the contact, if the qualify_freqency is
+ removed from configuration. ASTERISK-24462 #close Reported by:
+ Mark Michelson Review: https://reviewboard.asterisk.org/r/4116/
+
+2014-10-30 09:18 +0000 [r426696] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * apps/app_voicemail.c, /: app_voicemail: Fix unchecked bounds of
+ myArray in IMAP_STORAGE. In update_messages_by_imapuser(),
+ messages were appended to a finite array which resulted in a
+ crash when an IMAP mailbox contained more than 256 entries. This
+ memory is now dynamically increased as needed. Observe that this
+ patch adds a bunch of XXX's to questionable code. See the review
+ (url below) for more information. ASTERISK-24190 #close Reported
+ by: Nick Adams Tested by: Nick Adams Review:
+ https://reviewboard.asterisk.org/r/4126/ ........ Merged
+ revisions 426691 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 426692 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-30 06:02 +0000 [r426667] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c, /: Add additional checks for NULL
+ pointers to fix several crashes reported. ASTERISK-24304 #close
+ Reported by: dhanapathy sathya ........ Merged revisions 426666
+ from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-30 01:58 +0000 [r426596-426601] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: channels/chan_sip: Add improved support
+ for 4xx error codes This patch adds support for 414, 493, 479,
+ and a stray 400 response in REGISTER response handling. This
+ helps interoperability in a number of scenarios. Review:
+ https://reviewboard.asterisk.org/r/3437 patches: rb3437.patch
+ uploaded by oej (License 5267) ........ Merged revisions 426599
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Merged revisions 426600 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * channels/sip/reqresp_parser.c, /, channels/chan_sip.c:
+ channels/chan_sip: Support mutltiple Supported and Required
+ headers A SIP request may contain multiple Supported: and
+ Required: headers. Currently, chan_sip only parses the first
+ Supported/Required header it finds. This patch adds support for
+ multiple Supported/Required headers for INVITE requests. Review:
+ https://reviewboard.asterisk.org/r/2478 ASTERISK-21721 #close
+ Reported by: Olle Johansson patches: rb2478.patch uploaded by oej
+ (License 5267) ........ Merged revisions 426594 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 426595 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-28 21:16 +0000 [r426531] Richard Mudgett <rmudgett at digium.com>
+
+ * bridges/bridge_builtin_features.c: bridge_builtin_features: Add
+ missing channel locks around
+ ast_get_chan_features_general_config(). The feature_automonitor()
+ and feature_automixmonitor() functions were not locking the
+ channel around ast_get_chan_features_general_config(). Accessing
+ the channel datastore list without the channel locked is a good
+ way to corrupt the list or follow the pointer chain into
+ oblivion.
+
+2014-10-28 20:55 +0000 [r426524-426528] Corey Farrell <git at cfware.com>
+
+ * /, res/res_fax.c: res_fax: Resolve T38 gateway frame leak. When
+ frames are translated by a fax gateway they need to be freed. The
+ existing call to ast_frfree was unreachable. This change
+ reorganizes fax_gateway_framehook to ensure that ast_frfree is
+ called when needed. ASTERISK-24457 #close Reported by: Corey
+ Farrell Review: https://reviewboard.asterisk.org/r/4115/ ........
+ Merged revisions 426527 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * main/manager.c: manager: Unsubscribe from acl_change_sub at
+ shutdown. ASTERISK-24453 #close Reported by: Corey Farrell
+ Review: https://reviewboard.asterisk.org/r/4110/
+
+2014-10-28 18:08 +0000 [r426458] mdavenport <mdavenport at localhost>:
+
+ * configs/manager.conf.sample: ASTERISK-23512, correct inaccurate
+ comment in manager.conf.sample
+
+2014-10-28 16:40 +0000 [r426367-426431] Matthew Jordan <mjordan at digium.com>
+
+ * main/bridge.c: main/bridge: Destroy features struct on off
+ nominal path during bridge impart When a channel is imparted to a
+ bridge, the invocation of the function may provide an
+ ast_bridge_features struct. Upon passing this to
+ ast_bridge_impart, the caller must assume that ownership has
+ passed to the function, as in all paths the function destroys the
+ struct prior to returning (as its purpose is to configure the
+ behavior of the channel while in the bridge). On one off nominal
+ path - where the channel already has a PBX thread - the struct
+ was not being destroyed. This patch fixes that glitch.
+ ASTERISK-24437 #close Reported by: Scott Griepentrog
+
+ * main/manager.c, /: main/manager: Fix typo in AMI event
+ documentation of "OriginateResponse" The parameter name is
+ "Response", not "Resonse". ASTERISK-24430 #close Reported by:
+ Dafi Ni ........ Merged revisions 426366 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-28 14:56 +0000 [r426293-426361] mdavenport <mdavenport at localhost>:
+
+ * res/res_agi.c: ASTERISK-24323, fix bug in documentation of AGI
+ STREAM FILE CONTROL
+
+ * configs/extensions.conf.sample: ASTERISK-24419, fix incorrect
+ syntax for setting language in extensions.conf.sample
+
+2014-10-28 11:19 +0000 [r426260] Corey Farrell <git at cfware.com>
+
+ * /, apps/app_queue.c: app_queue: Cleanup ao2_iterator Clean
+ ao2_iterator, resolving reference leak to queue members.
+ ASTERISK-24454 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4111/ ........ Merged
+ revisions 426255 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-27 02:45 +0000 [r426142-426210] Matthew Jordan <mjordan at digium.com>
+
+ * /, res/res_http_websocket.c: res/res_http_websocket: Fix minor
+ nits found by wdoekes on r409681 When Moises committed the fixes
+ for WSS (which was a great patch), wdoekes had a few style nits
+ that were on the review that got missed. This patch resolves what
+ I *think* were all of the ones that were still on the review.
+ Thanks to both moy for the patch, and wdoekes for the reviews.
+ Review: https://reviewboard.asterisk.org/r/3248/ ........ Merged
+ revisions 426209 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * res/res_phoneprov.c: res/res_phoneprov: Fix crash on shutdown
+ caused by container cleanup In res_phoneprov, unloading the
+ module first destroys the http_routes container, followed by the
+ users. However, users may have a route in the http_routes
+ container; the validity of this container is not checked in the
+ users destructor. Hence, we hit an assert as the container has
+ already been set to NULL. This patch does two things: (1) It adds
+ a sanity check in the user destructor (because why not) (2) It
+ switches the order of destruction, so that users are disposed of
+ prior to the HTTP routes they may hold a reference to. Note that
+ this crash was caught by the Test Suite (go go testing!)
+
+ * /, res/res_srtp.c: res/res_srtp: Fix include issue for libsrtp
+ 1.5.0 In libsrtp 1.5.0, crypto_get_random is no longer resolved
+ simply by including srtp.h. Now, one must include crypto_kernel.h
+ as well. As it turns out, this header file has been provided by
+ the library since 2006, so this is a relatively benign change.
+ ASTERISK-24436 #close Reported by: Patrick Laimbock ........
+ Merged revisions 426140 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 426141 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-20 14:10 +0000 [r425987] Matthew Jordan <mjordan at digium.com>
+
+ * UPGRADE.txt, res/res_xmpp.c, res/res_jabber.c, main/tcptls.c:
+ AST-2014-011: Fix POODLE security issues There are two aspects to
+ the vulnerability: (1) res_jabber/res_xmpp use SSLv3 only. This
+ patch updates the module to use TLSv1+. At this time, it does not
+ refactor res_jabber/res_xmpp to use the TCP/TLS core, which
+ should be done as an improvement at a latter date. (2) The
+ TCP/TLS core, when tlsclientmethod/sslclientmethod is left
+ unspecified, will default to the OpenSSL SSLv23_method. This
+ method allows for all encryption methods, including SSLv2/SSLv3.
+ A MITM can exploit this by forcing a fallback to SSLv3, which
+ leaves the server vulnerable to POODLE. This patch adds WARNINGS
+ if a user uses SSLv2/SSLv3 in their configuration, and explicitly
+ disables SSLv2/SSLv3 if using SSLv23_method. For TLS clients,
+ Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is
+ explicitly chosen. For TLS servers, Asterisk will no longer
+ support SSLv2 or SSLv3. Much thanks to abelbeck for reporting the
+ vulnerability and providing a patch for the res_jabber/res_xmpp
+ modules. Review: https://reviewboard.asterisk.org/r/4096/
+ ASTERISK-24425 #close Reported by: abelbeck Tested by: abelbeck,
+ opsmonitor, gtjoseph patches: asterisk-1.8-jabber-tls.patch
+ uploaded by abelbeck (License 5903)
+ asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License
+ 5903) AST-2014-011-1.8.diff uploaded by mjordan (License 6283)
+ AST-2014-011-11.diff uploaded by mjordan (License 6283)
+
+2014-10-19 17:03 +0000 [r425964] George Joseph <george.joseph at fairview5.com>
+
+ * configure.ac, makeopts.in, Makefile, configure: build: Force
+ -fsigned-char on platforms where the default for char is unsigned
+ gcc on the ARM platform defaults 'char' to 'unsigned char'
+ whereas Intel and SPARC default to 'signed char'. This is only an
+ issue in the rare cases where negative values are assigned to a
+ 'char' but this this patch insures compatibility by detecting
+ platforms that default to 'unsigned' and adding an
+ '-fsigned-char' flag to _ASTCFLAGS. If compiling for ARM (native
+ or cross-compile) be sure to run ./bootstrap.sh and ./configure
+ to regenerate the build files. You shouldn't have to do this for
+ Intel or SPARC. Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4091/
+
+2014-10-19 03:58 +0000 [r425921-425943] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Undo 425921 This
+ patch for r425921 introduced a different bug, wherein sending an
+ INVITE request with no SDP would cause Asterisk to not send an
+ SDP Offer in the 200 OK. The current structure of
+ res_pjsip_sdp_rtp is a bit hard to deal with to fix this,
+ particularly in 12: (1) The format capabilities structures and
+ how they are used are a bit harder to manipulate than they are in
+ 13 (2) create_outgoing_sdp has no knowledge of whether or not it
+ is creating an SDP as a new Offer or an Answer. This is something
+ of an oversight in the callback definition, as the caller of it
+ does have this information.
+
+ * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Check joint caps
+ when looking to decline outgoing media When constructing an
+ outgoing media stream for an SDP offer/answer, the current code
+ checks the override preferences (set by the PJSIP_MEDIA_OFFER
+ function) as well as what is configured on the endpoint to
+ determine if a codec is available for the media stream.
+ Unfortunately, this isn't good enough: we must also look at the
+ negotiated (joint) format capabilities. Otherwise, we'll
+ construct a media stream offer/answer with no codecs. Note that
+ this isn't an issue in 13, which already looks at the joint
+ capabilities thanks to the media re-work done there.
+
+2014-10-17 13:32 +0000 [r425820-425868] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_pjsip.c, res/res_pjsip_session.c,
+ res/res_pjsip_sdp_rtp.c: res_pjsip_session/res_pjsip_sdp_rtp: Be
+ more tolerant of offers When an inbound SDP offer is received,
+ Asterisk currently makes a few incorrection assumptions: (1) If
+ the offer contains more than a single audio/video stream,
+ Asterisk will reject the entire stream with a 488. This is an
+ overly strict response; generally, Asterisk should accept the
+ media streams that it can accept and decline the others. (2) If
+ the offer contains a declined media stream, Asterisk will attempt
+ to process it anyway. This can result in attempting to match
+ format capabilities on a declined media stream, leading to a 488.
+ Asterisk should simply ignore declined media streams. (3)
+ Asterisk will currently attempt to handle offers with AVPF with
+ use_avpf=No/AVP with use_avpf=Yes. This mismatch results in
+ invalid SDP answers being sent in response. If there is a
+ mismatch between the media type being offered and the
+ configuration, Asterisk must reject the offer with a 488. This
+ patch does the following: * Asterisk will accept SDP offers with
+ at least one media stream that it can use. Some WARNING messages
+ have been dropped to NOTICEs as a result. * Asterisk will not
+ accept an offer with a media type that doesn't match its
+ configuration. * Asterisk will ignore declined media streams
+ properly. #SIPit31 Review:
+ https://reviewboard.asterisk.org/r/4063/ ASTERISK-24122 #close
+ Reported by: James Van Vleet ASTERISK-24381 #close Reported by:
+ Matt Jordan
+
+ * /, channels/chan_sip.c: channels/chan_sip: Respect outboundproxy
+ setting when sending qualify requests The outboundproxy setting
+ is currently ignored when sending OPTIONS requests as a result of
+ the qualify setting. This means that if an Asterisk server is
+ unable to send the packet directly to a peer, it is unable to
+ qualify any non-inbound registered peer (e.g. a peer SIP Trunk).
+ This patch grabs the outboundproxy information for a peer when a
+ qualify attempt is being constructed and, if it finds the
+ information, uses it when sending the OPTIONS request. Review:
+ https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close
+ Reported by: Damian Ivereigh patches: outboundproxy-dai.patch
+ uploaded by Damian Ivereigh (License 6632) ........ Merged
+ revisions 425818 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425819 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-17 02:32 +0000 [r425782] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, main/core_unreal.c: AMI: Add missing VarSet
+ events when a channel inherits variables. There should be AMI
+ VarSet events when channel variables are inherited by an outgoing
+ channel. Also local;2 should generate VarSet events when it gets
+ all of its channel variables from channel local;1. ASTERISK-24415
+ #close Reported by: Richard Mudgett Patches:
+ jira_asterisk_24415_v12.patch (license #5621) patch uploaded by
+ Richard Mudgett Review: https://reviewboard.asterisk.org/r/4074/
+
+2014-10-17 01:55 +0000 [r425735-425760] Matthew Jordan <mjordan at digium.com>
+
+ * bridges/bridge_native_rtp.c: bridge_native_rtp: Fix audio issues
+ when moving from remote bridge to softmix When a native RTP
+ bridge that is remotely bridging its participants switches to a
+ softmix bridge, it may not properly re-INVITE the media for one
+ or both participants back to Asterisk. This is due to the current
+ bridge_native_rtp code only re-INVITEs if it believes the channel
+ will survive the bridge operation. Currently, that code is
+ failing, as it expects the channels to have a soft hangup flag
+ set on it indicating that a redirect has occurred or that the
+ channel is going to leave the bridge. (The code did not take into
+ account a smart bridge operation). This patch also renames a few
+ things to be more reflective of the underlying types. Review:
+ https://reviewboard.asterisk.org/r/3997/ ASTERISK-24327 #close
+
+ * tests/test_cel.c: test_cel: Update pickup test to expect CANCEL
+ instead of ANSWSER The CEL pickup test previously looked for a
+ disposition of ANSWER between the original caller/peer when the
+ call is picked up. This is actually incorrect: the disposition
+ should, at the very least, not be ANSWER as the call was never
+ ANSWERed. The disposition is now CANCEL; this patch updates the
+ test accordingly.
+
+ * main/cdr.c: main/cdr: Use 'time' when rescheduling batched CDRs
+ as opposed to 'size' When refactoring CDRs to use the
+ configuration framework, a 'whoops' was introduced where the CDR
+ batch size was used when rescheduling a batch, as opposed to the
+ time duration. This patch corrects that obvious mistake.
+ ASTERISK-24426 #close Reported by: Shane Blaser
+
+2014-10-16 17:29 +0000 [r425713] George Joseph <george.joseph at fairview5.com>
+
+ * tests/test_config.c, main/config.c, include/asterisk/config.h:
+ config: Fix inf loop using ast_category_browse and
+ ast_variable_retrieve Fix infinite loop when calling
+ ast_variable_retrieve inside an ast_category_browse loop when
+ there is more than 1 category with the same name. Tested-by:
+ George Joseph Review: https://reviewboard.asterisk.org/r/4089/
+
+2014-10-16 14:24 +0000 [r425690] Kinsey Moore <kmoore at digium.com>
+
+ * include/asterisk/res_pjsip_session.h, res/res_pjsip_notify.c,
+ res/res_pjsip_pidf_digium_body_supplement.c,
+ res/res_pjsip_endpoint_identifier_ip.c,
+ res/res_pjsip_registrar_expire.c, res/res_pjsip_t38.c,
+ res/res_pjsip_mwi_body_generator.c,
+ res/res_pjsip_endpoint_identifier_user.c,
+ res/res_pjsip_send_to_voicemail.c,
+ include/asterisk/res_pjsip_pubsub.h,
+ res/res_pjsip_outbound_authenticator_digest.c,
+ res/res_pjsip_outbound_registration.c,
+ res/res_pjsip_endpoint_identifier_anonymous.c,
+ res/res_pjsip_path.c, res/res_pjsip_one_touch_record_info.c,
+ res/res_pjsip_acl.c, res/res_pjsip_pubsub.c,
+ res/res_pjsip_diversion.c, res/res_pjsip_refer.c,
+ include/asterisk/res_pjsip.h,
+ res/res_pjsip_pidf_body_generator.c, res/res_pjsip_dtmf_info.c,
+ res/res_pjsip_multihomed.c, res/res_pjsip_authenticator_digest.c,
+ res/res_pjsip_sdp_rtp.c, res/res_hep_pjsip.c,
+ res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c,
+ res/res_pjsip_logger.c, res/res_pjsip_nat.c,
+ res/res_pjsip_exten_state.c, res/res_pjsip_session.c,
+ res/res_pjsip_header_funcs.c, res/res_pjsip_rfc3326.c,
+ res/res_pjsip_phoneprov_provider.c, res/res_pjsip_mwi.c,
+ res/res_pjsip_xpidf_body_generator.c,
+ res/res_pjsip_dialog_info_body_generator.c,
+ res/res_pjsip_pidf_eyebeam_body_supplement.c,
+ channels/chan_pjsip.c, res/res_pjsip_registrar.c,
+ res/res_pjsip_transport_websocket.c: PJSIP: Enforce module load
+ dependencies This enforces that res_pjsip, res_pjsip_session, and
+ res_pjsip_pubsub have loaded properly before attempting to load
+ any modules that depend on them since the module loader system is
+ not currently capable of resolving module dependencies on its
+ own. ASTERISK-24312 #close Reported by: Dafi Ni Review:
+ https://reviewboard.asterisk.org/r/4062/
+
+2014-10-16 06:07 +0000 [r425668] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c, /: Fix loss of voice after second call
+ drops (on a second line) in case using multiple lines on unistim
+ phones. There is regression was introduced in r391379. Reported
+ by: Rustam Khankishyiev (closes issue ASTERISK-23846) ........
+ Merged revisions 425667 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-16 01:25 +0000 [r425645] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix a bug where ICE
+ state would get reset when it shouldn't. In the case where the
+ ICE negotiation had not yet started current state would get wiped
+ when it shouldn't. This also removes channel binding as in
+ practice this does not work well with other implementations.
+ ........ Merged revisions 425644 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-15 09:45 +0000 [r425589] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: chan_ooh323: fix rtptimeout general
+ value checking correct condition to check rtptimeout in [general]
+ config section ASTERISK-24393 #close Reported by: Dmitry Melekhov
+ Tested by: Dmitry Melekhov Patches: ASTERISK-24393.patch ........
+ Merged revisions 425547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425548 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-14 20:45 +0000 [r425525] George Joseph <george.joseph at fairview5.com>
+
+ * main/config.c, include/asterisk/config.h, tests/test_config.c:
+ config: Fix SEGV in unit test with MALLOC_DEBUG With MALLOC_DEBUG
+ the /main/config config_basic_ops test was causing a SEGV while
+ doing an ast_category_delete in an ast_category_browse loop.
+ Apparently this never worked but was also never tested. I removed
+ the test, added 2 notes to config.h indicating that it's not
+ supported and added a few lines of code to ast_category_delete to
+ prevent the SEGV should someone attempt it in the future.
+ Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4078/
+
+2014-10-14 18:49 +0000 [r425503] Jonathan Rose <jrose at digium.com>
+
+ * main/sched.c: Scheduler: Fix a nasty scheduler caching bug which
+ makes new tasks not execute Tasks that were marked for pending
+ deletion in the scheduler would be moved to the cache for later
+ reuse, but after being recycled the deleted mark wouldn't be
+ removed resulting in fresh tasks being deleted without reason...
+ and immediately moved back into the cache where they could be
+ reused again. This could cause horrendous things to happen in
+ just about anything that used a scheduler. ASTERISK-24321 #close
+ Reported by: Steve Pitts Review:
+ https://reviewboard.asterisk.org/r/4071/
+
+2014-10-14 18:11 +0000 [r425480] George Joseph <george.joseph at fairview5.com>
+
+ * include/asterisk/phoneprov.h, res/res_pjsip_phoneprov_provider.c,
+ res/res_phoneprov.c: res_phoneprov: Create accessor for
+ ast_phoneprov_std_variable_lookup Based on feedback from Richard,
+ I created an accessor for
+ res_phoneprov/ast_phoneprov_std_variable_lookup and added load
+ priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by:
+ Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/
+
+2014-10-14 16:45 +0000 [r425458] Corey Farrell <git at cfware.com>
+
+ * /, res/res_fax.c: res_fax: Fix reference leak caused by gateway
+ sessions Fax gateway session objects can be re-used, causing the
+ same gateway session to be added to faxregistry.container more
+ than once. This change causes fax_session_new to remove the
+ reserved session from the container before it's id is changed,
+ ensuring it's possible for the session to be freed.
+ ASTERISK-24392 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4049/ ........ Merged
+ revisions 425457 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-14 16:24 +0000 [r425430] Richard Mudgett <rmudgett at digium.com>
+
+ * main/stasis_channels.c: stasis_channels.c: Resolve unfinished
+ Dials when doing masquerades (Part 2) Masquerades into and out of
+ channels that are involved in a dial operation don't create the
+ expected dial end event. The missing dial end event goes against
+ the model for things like CDRs and generating Dial end manager
+ actions and such. There are four cases: 1) A channel masquerades
+ into the caller channel. The case happens when performing a
+ blonde transfer using the channel driver's protocol. 2) A channel
+ masquerades into a callee channel. The case happens when
+ performing a directed call pickup. 3) The caller channel
+ masquerades out of dial. The case happens when using the Bridge
+ application on the caller channel. 4) A callee channel
+ masquerades out of dial. The case happens when using the Bridge
+ application on a peer channel. As it turned out, all four cases
+ need to be handled instead of just the first one. ASTERISK-24237
+ Reported by: Richard Mudgett ASTERISK-24394 #close Reported by:
+ Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/
+
+2014-10-14 16:18 +0000 [r425411] Corey Farrell <git at cfware.com>
+
+ * /, res/res_fax.c: res_fax: Resolve module reference leak caused
+ by reserved sessions Remove reference to module providing
+ reserved session after adding a reference to the final module.
+ This re-reference is done to ensure that module references are
+ correct even if the final session selects a different module than
+ the reserved session. ASTERISK-18923 #close Reported by: Grigoriy
+ Puzankin Review: https://reviewboard.asterisk.org/r/4048/
+ ........ Merged revisions 425405 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425407 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-13 16:07 +0000 [r425383] George Joseph <george.joseph at fairview5.com>
+
+ * apps/app_directory.c, tests/test_sorcery.c, main/config.c,
+ tests/test_sorcery_realtime.c, res/res_sorcery_realtime.c,
+ apps/app_voicemail.c, res/res_sorcery_config.c, main/manager.c,
+ include/asterisk/config.h, pbx/pbx_realtime.c,
+ tests/test_config.c: manager/config: Support templates and
+ non-unique category names via AMI This patch provides the
+ capability to manipulate templates and categories with non-unique
+ names via AMI. Summary of changes: GetConfig and GetConfigJSON:
+ Added "Filter" parameter: A comma separated list of
+ name_regex=value_regex expressions which will cause only
+ categories whose variables match all expressions to be
+ considered. The special variable name TEMPLATES can be used to
+ control whether templates are included. Passing 'include' as the
+ value will include templates along with normal categories.
+ Passing 'restrict' as the value will restrict the operation to
+ ONLY templates. Not specifying a TEMPLATES expression results in
+ the current default behavior which is to not include templates.
+ UpdateConfig: NewCat now includes options for allowing duplicate
+ category names, indicating if the category should be created as a
+ template, and specifying templates the category should inherit
+ from. The rest of the actions now accept a filter string as
+ defined above. If there are non-unique category names, you can
+ now update specific ones based on variable values. To facilitate
+ the new capabilities in manager, corresponding changes had to be
+ made to config, most notably the addition of filter criteria to
+ many of the APIs. In some cases it was easy to change the
+ references to use the new prototype but others would have
+ required touching too many files for this patch so a wrapper with
+ the original prototype was created. Macros couldn't be used in
+ this case because it would break binary compatibility with
+ modules such as res_digium_phone that are linked to real symbols.
+ Tested-by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/4033/
+
+2014-10-12 21:08 +0000 [r425361] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Make the ICE
+ transport check case insensitive as some implementations use
+ 'udp'. ........ Merged revisions 425360 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-10-12 08:14 +0000 [r425288-425298] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: chan_sip: Fix so asterisk won't send
+ reINVITE after a BYE. After a reINVITE glare situation, Asterisk
+ would re-send the reINVITE even though the call had been hung up
+ in the mean time. This patch unschedules the reinvite when
+ handling the BYE. ASTERISK-22791 #close Reported by: Paolo
+ Compagnini Tested by: Paolo Compagnini Review:
+ https://reviewboard.asterisk.org/r/4056/ (testcase is in review
+ r4055) ........ Merged revisions 425296 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 425297 from
+ http://svn.asterisk.org/svn/asterisk/branches/11
+
+ * /, Makefile: build: Relax badshell tilde test to allow for ~ in
+ middle of DESTDIR. The main Makefile has a target test called
+ 'badshell' that tests if DESTDIR does not happen to have an
+ an-expanded tilde (~). This might be the case if you run: make
+ install DESTDIR=~/somewhere/ That test also disallowed valid
+ tildes in directory names. The test is now changed to only
+ trigger on a tilde at the start of the path. ASTERISK-13797
+ #close Reported by: Tzafrir Cohen Review:
+ https://reviewboard.asterisk.org/r/4064/ ........ Merged
+ revisions 425291 from
[... 35222 lines stripped ...]
More information about the asterisk-commits
mailing list