[asterisk-commits] bebuild: tag 10.3.0-rc2 r358428 - /tags/10.3.0-rc2/ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Mar 6 14:45:20 CST 2012
Author: bebuild
Date: Tue Mar 6 14:45:16 2012
New Revision: 358428
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=358428
Log:
Add ChangeLog for 10.3.0-rc2
Added:
tags/10.3.0-rc2/ChangeLog (with props)
Added: tags/10.3.0-rc2/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/10.3.0-rc2/ChangeLog?view=auto&rev=358428
==============================================================================
--- tags/10.3.0-rc2/ChangeLog (added)
+++ tags/10.3.0-rc2/ChangeLog Tue Mar 6 14:45:16 2012
@@ -1,0 +1,22285 @@
+2012-03-06 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.3.0-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 22:16 +0000 [r353369-353321] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/sip/include/dialog.h, /, channels/chan_sip.c,
+ channels/sip/include/sip.h: Merged revisions 353320 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31
+ Jan 2012) | 18 lines 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: Merged revisions 353368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan
+ 2012) | 2 lines prevent debug messsges displaying -ve Cseq
+ numbers. Missed in R353320 ........
+
+2012-01-30 23:28 +0000 [r353397] Terry Wilson <twilson at digium.com>
+
+ * main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h:
+ 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/
+ ........ Merged revisions 353371 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-31 17:21 +0000 [r353463] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c, /, include/asterisk/channel.h: 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/
+ ........ Merged revisions 353454 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 00:00 +0000 [r353503] 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/
+ ........ Merged revisions 353502 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 15:05 +0000 [r353551] 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 ........ Merged revisions 353550 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 15:51 +0000 [r353599] 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 ........ Merged revisions 353598 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 21:16 +0000 [r353771-353721] 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) ........ Merged revisions 353720
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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) ........ Merged revisions 353769
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-02 18:48 +0000 [r353820] Mark Michelson <mmichelson at digium.com>
+
+ * configs/http.conf.sample, main/manager.c, /, main/http.c,
+ configs/manager.conf.sample, include/asterisk/manager.h: 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 ........ Merged
+ revisions 353770 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-02 20:11 +0000 [r353868] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.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 ........ Merged
+ revisions 353867 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-02 22:27 +0000 [r353916] 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 ........ Merged revisions 353915 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-03 16:22 +0000 [r354000-353962] Jonathan Rose <jrose at digium.com>
+
+ * res/res_fax.c: Fixes a segfault occuring when performing attended
+ transfer with FAXOPT(gateway)=yes (closes issue ASTERISK-19184)
+ Reported by: Alexandr
+
+ * /, 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/ ........ Merged
+ revisions 353999 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-06 17:31 +0000 [r354217-354119] 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 ........ Merged revisions 354116 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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 ........ Merged revisions 354216 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-07 15:19 +0000 [r354270] 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/ ........ Merged
+ revisions 354263 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-07 21:17 +0000 [r354349] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql:
+ 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/
+ ........ Merged revisions 354348 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 02:25 +0000 [r354493] 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/ ........ Merged
+ revisions 354492 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 02:54 +0000 [r354496] 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 ........ Merged revisions 354495 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 16:35 +0000 [r354543] 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 ........
+ Merged revisions 354542 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 17:04 +0000 [r354546] Mark Michelson <mmichelson at digium.com>
+
+ * /, 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 ........ Merged revisions
+ 354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 17:08 +0000 [r354548] 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/ ........ Merged
+ revisions 354547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 19:54 +0000 [r354703-354656] Kinsey Moore <kmoore at digium.com>
+
+ * /, main/config.c: Make the config parser remove escaping
+ backslashes The config parser in Asterisk does not currently
+ remove a backslash that is used to escape a semicolon which would
+ otherwise be interpreted as the start of a comment. The change
+ here causes that backslash to be removed, but does not create a
+ real escape system in the config parser. The biggest complication
+ with a real escape system would be breaking existing configs
+ everywhere (parsing \\ as \ and breaking on escaped non-semicolon
+ characters) even though it would be the "right" way to do things.
+ (closes issue ASTERISK-17121) Review:
+ https://reviewboard.asterisk.org/r/1724/ ........ Merged
+ revisions 354655 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Fix parsing of SIP headers where compact
+ and non-compact headers are mixed Change parsing of SIP headers
+ so that compactness of the header no longer influences which
+ header will be chosen. Previously, a non-compact header would be
+ chosen instead of a preceeding compact-form header. (closes issue
+ ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/
+ ........ Merged revisions 354702 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 22:03 +0000 [r354750] Terry Wilson <twilson at digium.com>
+
+ * /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge
+ is torn down CDRs cannot be modified after a bridge is torn down,
+ (e.g. after Dial() returns) even though the CDR() function may be
+ called. Since modifying the CDR code to change this behavior
+ could very easily break all kinds of things, this patch just
+ documents this limitation. (closes issues ASTERISK-16923) Review:
+ https://reviewboard.asterisk.org/r/1720/ ........ Merged
+ revisions 354749 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-10 18:05 +0000 [r354836] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting
+ to the same exten and context. The astman_get_header() never
+ returns NULL so the check by the code for NULL would never fail.
+ (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches:
+ 0018325.patch (license #6116) patch uploaded by Nuno Borges
+ (modified) ........ Merged revisions 354835 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-10 22:00 +0000 [r354890] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.c, /: Fix a voicemail memory leak with
+ heard/deleted messages. open_mailbox() was changed quite a long
+ time ago to allocate this memory. close_mailbox() should have
+ been changed to be responsible for freeing it. ........ Merged
+ revisions 354889 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-13 16:41 +0000 [r354938] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_confbridge.c: Don't try to play sound files that do not
+ exist. (closes issue ASTERISK-19188) Reported by: slesru
+
+2012-02-13 17:24 +0000 [r354959] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix
+ reconnecting to pgsql database after connection loss. There can
+ only be one database connection in res_config_pgsql just like
+ res_config_sqlite. If the connection is lost, the connection may
+ not get reestablished to the same database if the res_pgsql.conf
+ and extconfig.conf files are inconsistent. * Made only use the
+ configured database from res_pgsql.conf. * Fixed potential buffer
+ overwrite of last[] in config_pgsql(). (closes issue
+ ASTERISK-16982) Reported by: german aracil boned Review:
+ https://reviewboard.asterisk.org/r/1731/ ........ Merged
+ revisions 354953 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-13 19:51 +0000 [r355010] Joshua Colp <jcolp at digium.com>
+
+ * /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute
+ at a time as otherwise they would share the same common local
+ context list. (closes issue AST-758) ........ Merged revisions
+ 355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-13 22:03 +0000 [r355057] Richard Mudgett <rmudgett at digium.com>
+
+ * pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file
+ execution. Since the dir timestamp is available at one second
+ resolution, we cannot know if it was updated within the same
+ second after we scanned it. Therefore, we will force another scan
+ if the dir was just modified. * Changed to force another scan if
+ the directory was just modified. (closes issue ASTERISK-19081)
+ Reported by: Knut Bakke Review:
+ https://reviewboard.asterisk.org/r/1688/ ........ Merged
+ revisions 355056 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 09:49 +0000 [r355137] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: call manager_event only if there is not
+ null channel structure (Closes issue ASTERISK-19298) Reported by:
+ robinfood Patches: issue19298.patch uploaded by may213 (License
+ #5415) ........ Merged revisions 355136 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 13:33 +0000 [r355183] Sean Bright <sean at malleable.com>
+
+ * /, channels/chan_iax2.c: Clear the high order bit from the
+ destination call number before sending. send_apathetic_reply
+ takes the incoming frame's source call number as the destination
+ call number for the outgoing frame. If the incoming frame was a
+ full frame, then the high order bit of the source call number is
+ set and will be interpreted as a retransmit when sent back out as
+ the destination call number. ........ Merged revisions 355182
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 15:53 +0000 [r355229] Jason Parker <jparker at digium.com>
+
+ * /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3
+ CDRs by default in sample configs. ........ Merged revisions
+ 355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 16:27 +0000 [r355271] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Properly invert the return of a strncmp
+ call. This was causing identification that should have been made
+ private to be public. (closes issue AST-814) reported by Patrick
+ Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
+ (license 5430) ........ Merged revisions 355268 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 18:14 +0000 [r355375-355320] Richard Mudgett <rmudgett at digium.com>
+
+ * /, cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock
+ in cel_sqlite_custom reload. (closes issue ASTERISK-19356)
+ Reported by: Alex Villacis Lasso Patches:
+ asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch
+ (license #5617) patch uploaded by Alex Villacis Lasso Review:
+ https://reviewboard.asterisk.org/r/1740/ ........ Merged
+ revisions 355319 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ formats/format_ogg_vorbis.c: Fix voicemail problems when using
+ ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file
+ format because it did not implement the seek and tell format
+ callbacks among other problems. Since we were already using the
+ libvorbis and libvorbisenc libraries we can use libvorbisfile as
+ it is also part of the vorbis library package. * Made use the
+ libvorbisfile to handle the ogg/vorbis file stream. The
+ format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
+ (closes issue ASTERISK-16926) Reported by: sque Patches:
+ ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded
+ by sque ........ Merged revisions 355365 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-15 17:25 +0000 [r355530-355449] Sean Bright <sean at malleable.com>
+
+ * /, channels/chan_iax2.c: Use TRUNK_CALL_START as originally
+ intended. Back in r646, TRUNK_CALL_START was added and defined as
+ 0x4000. That same value was also hard-coded in one part of the
+ IAX2 code instead of using the #define. TRUNK_CALL_START has
+ changed over the years (for dealing with LOW_MEMORY), but the
+ hard-coded usage was never updated to match. This patch fixes
+ that. ........ Merged revisions 355448 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: Only use maxtrunkcall and
+ maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified. These
+ variables are only accessed from the IAX_OLD_FIND path, so there
+ is no reason to keep them updated otherwise. ........ Merged
+ revisions 355458 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: When IAX2 debugging is enabled, make
+ sure to log 'apathetic' messages too. ........ Merged revisions
+ 355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-16 18:32 +0000 [r355620-355575] Richard Mudgett <rmudgett at digium.com>
+
+ * /, res/res_monitor.c: Fix AMI Monitor action without File header
+ converting channel name into filename. * Fix potential Solaris
+ crash if Monitor application has a urlbase and no fname_base
+ option. ........ Merged revisions 355574 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ autoconf/ast_c_declare_check.m4 (added), configure.ac,
+ formats/format_ogg_vorbis.c: Fix compile problem when old version
+ of libvorbisfile v1.1.2 is used. The principle difference between
+ libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition
+ of the predefined callbacks OV_CALLBACKS_xxx in
+ vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the
+ configure script to detect if libvorbisfile.h declares
+ OV_CALLBACKS_NOCLOSE. * Copied the declaration of
+ OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile.
+ (closes issue ASTERISK-19370) Reported by: Jonn Taylor ........
+ Merged revisions 355608 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-16 20:01 +0000 [r355623] Sean Bright <sean at malleable.com>
+
+ * /, main/audiohook.c: Revert a change to
+ audio_audiohook_write_list that had no affect. When I made this
+ change initially, I was under the false impression that the
+ audiohooks structure remained on the channel after all of the
+ hooks had been detached. This is not the case, ast ast_read takes
+ care of removing the audiohooks structure if the lists are empty.
+ ........ Merged revisions 355622 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-17 19:06 +0000 [r355733] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Fix regressions with regards to route-set
+ creation on early dialogs. This fixes two main issues: 1.
+ Asterisk would send a CANCEL to the route created by the
+ provisional response instead of using the same destination it did
+ in the initial INVITE. 2. If a new route set arrives in a 200 OK
+ than was in the 1XX response (perfectly possible if our outbound
+ INVITE gets forked), then the route set in the 200 OK needs to
+ overwrite the route set in the 1XX response. (closes issue
+ ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten
+ Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson
+ (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt
+ (license 6034) Review: https://reviewboard.asterisk.org/r/1749
+ ........ Merged revisions 355732 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-17 19:34 +0000 [r355794-355747] Sean Bright <sean at malleable.com>
+
+ * /, channels/chan_iax2.c: Pass the correct value to
+ ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq
+ variable to determine how often to send trunk packets, but this
+ value is in milliseconds while ast_timer_set_rate() expects the
+ rate argument to be ticks per second. So we divide 1000 by
+ trunkfreq and pass that in instead. With a default of 20ms, this
+ change makes IAX2 send trunk packets every 20ms instead of every
+ 50ms. Tracked down by myself and Bob Wienholt. ........ Merged
+ revisions 355746 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configs/iax.conf.sample, /, channels/chan_iax2.c: Don't allow
+ trunkfreq to be greater than 1000ms. ........ Merged revisions
+ 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-18 07:58 +0000 [r355851] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_ss7.h, /, channels/sig_analog.h, channels/sig_pri.c,
+ channels/sig_ss7.c: push 'outgoing' flag from sig_XXX up to
+ chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept
+ in sync, particulary FXS ast_hangup didn't clear the 'outgoing'
+ flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
+ Now provides a callback for all the low level sig_XXX modules.
+ (issue ASTERISK-19316) alecdavis (license 585) Reported by:
+ Jeremy Pepper Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1747/ ........ Merged
+ revisions 355850 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-18 17:02 +0000 [r355896-355895] Paul Belanger <pabelanger at digium.com>
+
+ * /: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
+ ........ Merged revisions 355839 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /: Revert commit
+
+2012-02-19 17:50 +0000 [r355998-355902] Sean Bright <sean at malleable.com>
+
+ * /, channels/chan_iax2.c: Set the length of the ast_sockaddr, so
+ that we can set it's port later. Without this, the call to
+ ast_sockaddr_set_port a few lines later is a noop. ........
+ Merged revisions 355901 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: Add some boilerplate documentation for
+ IAXVAR and IAXPEER. ........ Merged revisions 355904 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, /: Change some debug messages from
+ LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, /: This was a LOG_NOTICE, so roll it back.
+ ........ Merged revisions 355952 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_iax2.c: Remove spurious warning when
+ 'qualifyfreqnotok' is set successfully. (closes issue
+ ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright
+ Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John
+ Covert (license 5512) ........ Merged revisions 355997 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-21 04:30 +0000 [r356074] Kinsey Moore <kmoore at digium.com>
+
+ * main/ccss.c: Add missing newline to ccss state change
+ notification Move along, nothing to see here...
+
+2012-02-21 11:17 +0000 [r356108] Sean Bright <sean at malleable.com>
+
+ * /, channels/chan_iax2.c: Make 'iax2 show callnumber usage' output
+ make sense when an IP is passed in. ........ Merged revisions
+ 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-22 14:53 +0000 [r356215] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 356214 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012)
+ | 27 lines Fix potential buffer overrun and memory leak when
+ executing "sip show peers" The "sip show peers" command uses a
+ fix sized array to sort the current peers in the peers
+ ao2_container. The size of the array is based on the current
+ number of peers in the container. However, once the size of the
+ array is determined, the number of peers in the container can
+ change, as the peers container is not locked. This could cause a
+ buffer overrun when populating the array, if peers were added to
+ the container after the array was created. Additionally, a memory
+ leak of the allocated array would occur if a user caused the
+ _show_peers method to return CLI_SHOWUSAGE. We now create a
+ snapshot of the current peers using an ao2_callback with the
+ OBJ_MULTIPLE flag. This size of the array is set to the number of
+ peers that the iterator will iterate over; hence, if peers are
+ added or removed from the peers container it will not affect the
+ execution of the "sip show peers" command. Review:
+ https://reviewboard.asterisk.org/r/1738/ (closes issue
+ ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas
+ Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey
+ Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan
+ (license 6283) ........
+
+2012-02-22 21:18 +0000 [r356297] Terry Wilson <twilson at digium.com>
+
+ * main/loader.c, res/res_calendar.c, /,
+ include/asterisk/calendar.h: Track module use count for
+ res_calendar If the res_calendar module was followed immediately
+ by one of the calendar tech modules and "core stop gracefully"
+ was run, Asterisk would crash. This patch adds use count tracking
+ for res_calendar so that it is unloaded after the tech modules
+ when shutting down gracefully. It is now not possible to unload
+ all the of the calendar modules via "module unload
+ res_calednar.so", but it is still possible to unload them all via
+ "module unload -h res_calendar.so". Review:
+ https://reviewboard.asterisk.org/r/1752/ ........ Merged
+ revisions 356291 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-23 03:23 +0000 [r356431-356428] Paul Belanger <pabelanger at digium.com>
+
+ * /, apps/app_rpt.c: Multiple revisions 356290,356335,356337
+ ........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed,
+ 22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2) Review:
+ https://reviewboard.asterisk.org/r/1763/ ........ r356335 |
+ pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2
+ lines Add back strsep() function for previous commit ........
+ r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb
+ 2012) | 2 lines Missed one strsep() function ........ Merged
+ revisions 356290,356335,356337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * addons/chan_ooh323.c, /: Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2) ........ Merged revisions 356430 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-23 15:40 +0000 [r356476] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Fix ACK routing for non-2xx responses.
+ When we send an ACK for a 2xx response to an INVITE, we are
+ supposed to use the learned route set. However, when we receive a
+ non-2xx final response to an INVITE, we are supposed to send the
+ ACK to the same place we initially sent the INVITE. We had been
+ doing this up until the changes went in that would build a route
+ set from provisional responses. That introduced a regression
+ where we would use the learned route set under all circumstances.
+ With this change, we now will set the destination of our ACK
+ based on the invitestate. If it is INV_COMPLETED then that means
+ that we have received a non-2xx final response (INV_TERMINATED
+ indicates a 2xx response was received). If it is INV_CANCELLED,
+ then that means the call is being canceled, which means that we
+ should be ACKing a 487 response. The other change introduced here
+ is setting the invitestate to INV_CONFIRMED when we send an ACK
+ *after* the reqprep instead of before. This way, we can tell in
+ reqprep more easily what the invitestate is prior to sending the
+ ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer
+ patches: ASTERISK-19389v2.patch uploaded by Mark Michelson
+ (license #5049) (with some slight modifications prior to commit)
+ ........ Merged revisions 356475 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-23 19:52 +0000 [r356522] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c, main/features.c: Fix blind transfer
+ parking issues if the dialed extension is not recognized as a
+ parking extension. Custom parking extensions may not be coded
+ such that the first and only extension priority is the Park
+ application. These custom parking extensions will not be
+ recognized as parking extensions. When a call is blind
+ transferred to an extension that is not recognized as a parking
+ extension, the normal blind transfer code causes the transferred
+ channel to start executing dialplan. Calls that get parked in
+ this manner do not know the original channel name that parked the
+ call so the original parker could never be called back if the
+ parked call is not retrieved before the timeout time. The parking
+ space is also announced to the call being parked as a side effect
+ of not knowing the original parking channel. * Fix handling of
+ BLINDTRANSFER channel variable for call parking. * Fixed SIP
+ blind transfer using the wrong dialplan context variable to check
+ for the parking extension. (closes issue ASTERISK-19322) Reported
+ by: aragon Tested by: rmudgett, jparker Review:
+ https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........
+ Merged revisions 356521 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-24 15:07 +0000 [r356651-356605] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_srtp.c, channels/sip/sdp_crypto.c,
+ include/asterisk/res_srtp.h, main/rtp_engine.c, /,
+ include/asterisk/rtp_engine.h: Allow SRTP policies to be reloaded
+ Currently, when using res_srtp, once the SRTP policy has been
+ added to the current session the policy is locked into place. Any
+ attempt to replace an existing policy, which would be needed if
+ the remote endpoint negotiated a new cryptographic key, is
+ instead rejected in res_srtp. This happens in particular in
+ transfer scenarios, where the endpoint that Asterisk is
+ communicating with changes but uses the same RTP session. This
+ patch modifies res_srtp to allow remote and local policies to be
+ reloaded in the underlying SRTP library. From the perspective of
+ users of the SRTP API, the only change is that the adding of
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