[asterisk-commits] lmadsen: tag 1.8.5-rc1 r324779 - /tags/1.8.5-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jun 24 12:03:23 CDT 2011
Author: lmadsen
Date: Fri Jun 24 12:03:19 2011
New Revision: 324779
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=324779
Log:
Importing files for 1.8.5-rc1 release.
Added:
tags/1.8.5-rc1/.lastclean (with props)
tags/1.8.5-rc1/.version (with props)
tags/1.8.5-rc1/ChangeLog (with props)
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--- tags/1.8.5-rc1/ChangeLog (added)
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+2011-06-25 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.5-rc1 released.
+
+2011-06-24 16:48 +0000 [r324768] Jonathan Rose <jrose at digium.com>
+
+ * include/asterisk/logger.h: DTMF wasn't being logged on connected
+ consoles when enabled in logger.conf Previously in order for DTMF
+ to be logged in a connected console session, the user would have
+ to do logger set channel DTMF on. This corrects that so that it
+ is on by default. This issue was caused by an off by one error
+ incurred by a logger level count of 6 in logger.h where it should
+ have been 7. (closes issue: ASTERISK-17974) Reported by: Luke H
+
+2011-06-23 18:31 +0000 [r324685] David Vossel <dvossel at digium.com>
+
+ * channels/sip/reqresp_parser.c: Fixes sip crash when calling
+ remove_uri_parameters with NULL AST-2011-009 (closes issue
+ ASTERISK-18017) Reported by: jaredmauch
+
+2011-06-23 18:29 +0000 [r324678] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 324643 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) |
+ 4 lines Addresses AST-2011-008, memory corruption and remote
+ crash in SIP driver. AST-2011-008 ........
+
+2011-06-23 18:23 +0000 [r324652] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/frame.h, /,
+ main/features.c: Merged revisions 324634 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500
+ (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011)
+ | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver
+ Thanks to twilson for identifying the issue and providing the
+ patches. AST-2011-010 ........ ................
+
+2011-06-23 03:10 +0000 [r324557] Terry Wilson <twilson at digium.com>
+
+ * tests/test_netsock2.c: Remove tests for parsing address with
+ invalid port getaddrinfo on OS X returns with EAI_NONAME error
+ when passed a port greater than 65535. Linux throws no error, so
+ remove the tests for now.
+
+2011-06-22 19:16 +0000 [r324491] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Use correct variable for text SRTP media.
+
+2011-06-22 18:52 +0000 [r324484] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/netsock2.h, tests/test_netsock2.c (added),
+ main/netsock2.c, channels/chan_sip.c: Stop sending IPv6
+ link-local scope-ids in SIP messages The idea behind the patch
+ listed below was used, but in a more targeted manner. There are
+ now address stringification functions for addresses that are
+ meant to be sent to a remote party. Link-local scope-ids only
+ make sense on the machine from which they originate and so are
+ stripped in the new functions. There is also a host sanitization
+ function added to chan_sip which is used for when peer and dialog
+ tohost fields or sip_registry hostnames are used to craft a SIP
+ message. Also added are some basic unit tests for netsock2
+ address parsing. (closes issue ASTERISK-17711) Reported by:
+ ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded
+ by ch_djalel (license 1251) Review:
+ https://reviewboard.asterisk.org/r/1278/
+
+2011-06-22 18:41 +0000 [r324479-324481] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Timout or error on INFO or MESSAGE
+ transaction causes call to be lost. When exchanging INFO messages
+ within a call, 4xx error causes the call to be disconnected
+ although RFC 2976 explicitly states that such transactions do not
+ modify the state of the dialog. When exchanging MESSAGE messages
+ within a call, 4xx error causes the call to be disconnected. To
+ provide least surprise, we should not disconnect the call since a
+ MESSAGE is like INFO in this case. (Implied by RFC 3428 Section
+ 2) (closes issue ASTERISK-17901) Reported by: neutrino88 Review:
+ https://reviewboard.asterisk.org/r/1257/ Review:
+ https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486
+
+ * channels/chan_sip.c: Comments and whitespace in chan_sip.c
+
+2011-06-21 20:11 +0000 [r324364] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/pbx.h, main/pbx.c: Fixes locking inversion issue
+ in ast_async_goto() During this function we can not hold the
+ "chan" lock while doing the masquerade, the explicit goto on the
+ tmp chan, or the channel alloc. Instead we need to get the
+ channel lock, store off information about the channel that we
+ need, and then let the channel lock go for the remainder of the
+ function. Review: https://reviewboard.asterisk.org/r/1275/
+
+2011-06-21 16:09 +0000 [r324305] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_confbridge.c: ConfBridge does not handle hangup properly
+ When playing back a prompt to a channel, confbridge neglects to
+ check for hangup events causing lockup condititions for hangups
+ that occur before actually joining the conference. This change
+ ensures that the user is removed from the conference in the event
+ of a premature hangup. Review:
+ https://reviewboard.asterisk.org/r/1277/
+
+2011-06-20 18:12 +0000 [r324239-324241] Leif Madsen <lmadsen at digium.com>
+
+ * configs/queuerules.conf.sample: Remove extra 'the'. Reported by
+ Vlad Povorozniuc
+
+ * configs/queuerules.conf.sample,
+ contrib/scripts/asterisk.logrotate: Revert previous merge which
+ had extra changes.
+
+ * configs/queuerules.conf.sample,
+ contrib/scripts/asterisk.logrotate: Remove extra 'the'. Reported
+ by Vlad Povorozniuc
+
+2011-06-20 17:33 +0000 [r324237] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Ignore media offers with a port of 0 Section
+ 5.1 of RFC3264 states: A port number of zero in the offer
+ indicates that the stream is offered but MUST NOT be used.
+ (closes issue ASTERISK-17845) Reported by: jacco Patches:
+ issue19281_2.patch uploaded by jacco (license 1277) Tested by:
+ jacco, twilson
+
+2011-06-17 18:51 +0000 [r324176-324178] Leif Madsen <lmadsen at digium.com>
+
+ * main/manager.c: Add Username and Secret fields to manager Login
+ action. Pointed out by Vlad Povorozniuc
+
+ * apps/app_meetme.c: Fix typo in documentation. Pointed out by Vlad
+ Povorozniuc
+
+2011-06-17 18:23 +0000 [r324174] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Add header string to libpri debug output.
+ Add header string to libpri debug output so the libpri output can
+ be found/extracted easier from huge debug trace files.
+
+2011-06-17 15:14 +0000 [r324115] Leif Madsen <lmadsen at digium.com>
+
+ * main/pbx.c: Fix grammar in documentation for Goto() and GotoIf()
+ (closes issue ASTERISK-18023) Reported by: Tim Osman
+
+2011-06-16 22:41 +0000 [r324048-324049] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c: Shame on me
+
+ * include/asterisk/channel.h, main/channel.c,
+ channels/chan_local.c, channels/chan_sip.c: Lock the channel
+ before calling the setoption callback The channel needs to be
+ locked before calling these callback functions. Also,
+ sip_setoption needs to lock the pvt and a check p->rtp is
+ non-null before using it. Review:
+ https://reviewboard.asterisk.org/r/1220/
+
+2011-06-16 18:12 +0000 [r323990] Richard Mudgett <rmudgett at digium.com>
+
+ * tests/test_event.c: The test_event unit test is occasionally
+ failing. Wait for the special posted event to process before
+ adding a new subscription.
+
+2011-06-16 15:58 +0000 [r323754-323932] Terry Wilson <twilson at digium.com>
+
+ * Makefile: Don't assume ASTDBDIR exists It most likely doesn't on
+ FreeBSD
+
+ * tests/test_db.c: Remove now-useless cast of ARRAY_LEN
+
+ * include/asterisk/utils.h: Make ARRAY_LEN() return the same type
+ on x86 and x86_64 systems
+
+ * tests/test_db.c: Fix more ARRAY_LEN format string issues
+
+ * /, main/features.c: Merged revisions 323733 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r323733 | twilson | 2011-06-15 13:13:00 -0500
+ (Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011)
+ | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a
+ recent DTMF change. This patch makes sure that dynamic features
+ are also checked when deciding whether or not to pass DTMF
+ through or store it for interpreting. (closes issue
+ ASTERISK-17914) Reported by: vrban ........ ................
+
+2011-06-15 17:42 +0000 [r323730] Jonathan Rose <jrose at digium.com>
+
+ * res/res_config_pgsql.c: Adds locking to find_table in
+ res_configure_pgsql to prevent a crash. Bryonclark described the
+ problem as occuring during this function because of multiple
+ simultaneous database operations causing corruption against a
+ pgsqlConn object. (closes issue ASTERISK-17811) Reported by:
+ byronclark Patches: pgsql_find_table_locking.patch uploaded by
+ byronclark (license 1200)
+
+2011-06-15 17:09 +0000 [r323672] Terry Wilson <twilson at digium.com>
+
+ * tests/test_db.c: Cast ARRAY_LEN to size_t for ast_logging 32-bit
+ and 64-bit machines return different types for ARRAY_LEN(), so
+ cast it before using in a format string.
+
+2011-06-15 16:43 +0000 [r323669-323670] Richard Mudgett <rmudgett at digium.com>
+
+ * tests/test_event.c: Add a test to the event unit tests to catch
+ ASTERISK-18002. The new tests check to see if there are ANY
+ subscribers to the event type when ast_event_check_subscriber()
+ is not passed any specific ie values. (issue ASTERISK-18002)
+
+ * main/event.c: [regression] Voicemail MWI is no longer sent. When
+ leaving a voicemail, the MWI message is never sent. The same
+ thing happens when checking a voicemail and marking it as read.
+ If you restart Asterisk, everything comes up at that state
+ correctly, but changes to the messages in voicemail causes the
+ light to not be set appropriately. Very easy to reproduce. * Made
+ ast_event_check_subscriber() return TRUE if there are ANY
+ subscribers to an event type when there are no restricting ie
+ values passed. This allows an event being queued to be queued.
+ (closes issue ASTERISK-18002) Reported by: lmadsen Tested by:
+ lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded
+ by rmudgett (License #5621) (closes issue ASTERISK-18019)
+
+2011-06-15 16:09 +0000 [r323610] Jonathan Rose <jrose at digium.com>
+
+ * res/res_config_pgsql.c: Adds PQclear calls on result to various
+ parts of res_conf_pgsql (closes issue ASTERISK-17812) Reported
+ by: byronclark Patches: pgsql_pqclear.patch uploaded by
+ byronclark (license 1200)
+
+2011-06-15 15:31 +0000 [r323608] Sean Bright <sean at malleable.com>
+
+ * main/manager.c, /: Merged revisions 323579 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400
+ (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun
+ 2011) | 25 lines Resolve a segfault/bus error when we try to map
+ memory that falls on a page boundary. The fix for ASTERISK-15359
+ was incorrect in that it added 1 to the length of the mmap'd
+ region. The problem with this is that reading/writing to that
+ extra byte outside of the bounds of the underlying fd causes a
+ bus error. The real issue is that we are working with both a FILE
+ * and the raw fd underneath it and not synchronizing between
+ them. The code that was removed in ASTERISK-15359 was correct,
+ but we weren't flushing the FILE * before mapping the fd. Looking
+ at the manager code in 1.4 reveals that the FILE * in 'struct
+ mansession' is never used except to create a temporary file that
+ we immediately fdopen. This means we just need to write a 0 byte
+ to the fd and everything will just work. The other branches
+ require a call to fflush() which, while not a guaranteed fix,
+ should reduce the likelihood of a crash. This all makes sense in
+ my head. (closes issue ASTERISK-16460) Reported by:
+ Ravelomanantsoa Hoby (hoby) Patches:
+ issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license
+ #5060) ........ ................
+
+2011-06-15 00:50 +0000 [r323392-323456] Richard Mudgett <rmudgett at digium.com>
+
+ * main/event.c: Add missing break in ast_event_get_cached().
+
+ * main/netsock2.c: Made ast_sockaddr_split_hostport() port warning
+ msgs more meaningful.
+
+ * main/dnsmgr.c: Add more strict hostname checking to
+ ast_dnsmgr_lookup(). Change suggested in review. Review:
+ https://reviewboard.asterisk.org/r/1240/
+
+2011-06-14 16:38 +0000 [r323371] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: Changes contact use in build_peer to use the
+ FORCE_RPORT flag instead of RPORT_PRESENT It turned out that this
+ was causing NAT=Yes to always use rport when present which was
+ against 1.6.2 behavior and the check itself was redundant since
+ the only way this segment of code could be reached was if
+ RPORT_PRESENT was already evaluated as true earlier. (closes
+ issue ASTERISK-17789) Reported by: byronclark Patches:
+ use_sip_nat_force_rport.patch uploaded by byronclark (license
+ 1200)
+
+2011-06-14 16:33 +0000 [r323370] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, channels/chan_sip.c: Add rtpkeepalives back to
+ 1.8 The RTP-engine conversion left out support for handling
+ rtpkeepalives. This patch adds them back. (closes issue
+ ASTERISK-17304) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/1226/
+
+2011-06-13 20:22 +0000 [r323154-323234] Leif Madsen <lmadsen at digium.com>
+
+ * configs/sip.conf.sample: Additional documentation for bindaddr.
+ Note that bindaddr will only enable UDP instead of both UDP and
+ TCP which is what I would expect for backwards compatibility with
+ systems being upgraded which only support UDP transportation.
+ (closes issue ASTERISK-17976) Reported by: Sean Darcy
+
+ * main/channel.c: Avoid dividing by zero with L() option to Dial()
+ Reported by: nicolasom Patches: issue-17995.patch - nicolasom
+ (License #5994)
+
+ * res/res_agi.c: Tweak documentation for AGI Hangup command.
+ (closes issue ASTERISK-17999) Reported by: Ben Klang Patches:
+ hangup-doc.diff - uploaded by Ben Klang (License #5876)
+
+2011-06-10 19:20 +0000 [r323040] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Unlock the sip channel during fax detection
+ like chan_dahdi does to prevent a deadlock with
+ ast_autoservice_stop. (closes issue ASTERISK-17798) tested by
+ mnicholson
+
+2011-06-10 15:29 +0000 [r322865-322981] Terry Wilson <twilson at digium.com>
+
+ * main/db.c: Avoid a DB1 infinite loop bug Explicity check the last
+ entry in the DB and make sure that we don't iterate past it.
+ Since there can be no duplicates, this just makes sure that we
+ stop after matching the last key. This patch also refactors the
+ code to get away from some code duplication. A previous patch
+ added many astdb tests and this patch passed them. Review:
+ https://reviewboard.asterisk.org/r/1259/
+
+ * tests/test_db.c (added): Add some astdb unit tests
+
+ * include/asterisk/astdb.h: Correct ast_db_deltree documentation
+ ast_db_deltree returns -1 on error, otherwise the number of
+ deletions
+
+2011-06-09 17:37 +0000 [r322807] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: don't drop any voice frames when checking
+ for T.38 during early media (closes issue ASTERISK-17705) Review:
+ https://reviewboard.asterisk.org/r/1186/ patch by oej reported by
+ oej
+
+2011-06-09 16:31 +0000 [r322749] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/features.h, apps/app_directed_pickup.c,
+ main/features.c: Remove potential deadlock in call pickup race.
+ Deadlock is possible in ast_do_pickup() when holding the target
+ channel lock and trying to get the chan channel lock. Also,
+ holding the target lock when calling ast_channel_masquerade() is
+ not a good idea because that routine does deadlock avoidance. *
+ Removed the need to hold the target lock after marking the target
+ with a datastore and getting the connected line data off of the
+ target channel. * Moved can_pickup() to ast_can_pickup() in
+ features.c. Now all the call pickup methods use the same basic
+ call pickup availability check. Review:
+ https://reviewboard.asterisk.org/r/1234/
+
+2011-06-09 14:06 +0000 [r322585] Jonathan Rose <jrose at digium.com>
+
+ * main/utils.c, include/asterisk/utils.h, channels/chan_sip.c,
+ tests/test_utils.c: Adds ast_escape_encoded utility to properly
+ handle escaping of quoted field before uri. This commit backports
+ a feature in trunk affecting initreqprep so that display name
+ won't be encoded improperly. Also includes unit tests for the
+ ast_escape_quoted function. This patch gives 1.8 a much improved
+ outlook in countries which don't use standard ASCII characters.
+ (closes issue ASTERISK-16949) Reported by: Örn Arnarson Review:
+ https://reviewboard.asterisk.org/r/1235/
+
+2011-06-08 20:46 +0000 [r322425-322484] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c: Ring all queue with more than 255 agents will
+ cause crash. 1. Create a ring-all queue with 500 permanent
+ agents. 2. Call it. 3. Asterisk will crash. The watchers array in
+ app_queue.c has a hard limit of 255. Bounds checking is not done
+ on this array. No sane person should put 255 people in a ring-all
+ queue, but we should not crash anyway. * Added bounds checking to
+ the watchers array. JIRA AST-464 JIRA SWP-2903
+
+ * main/dnsmgr.c: SRV lookup attempted for SIP peers listed as an IP
+ address. Asterisk attempts to SRV lookup a host name even if the
+ host name is an IP address. Regression introduced when IPv6
+ support was added. * Restored the check in ast_dnsmgr_lookup() to
+ see if the given host name is an IP address. The IP address could
+ be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815)
+ Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett
+ Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett
+ (License #5621) Review: https://reviewboard.asterisk.org/r/1240/
+
+2011-06-08 06:18 +0000 [r322322] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * channels/chan_sip.c: Make handle_request_publish do dialog
+ expiration and destruction. This patch fixes
+ handle_request_publish so that it does dialog expiration and
+ destruction. Without this patch the incoming PUBLISH requests
+ will get stuck in the dialog list. Restarting asterisk is the
+ only way to remove them. Personal observation on one system the
+ server hung up while looping through the channels rendering
+ asterisk unusable and all sip phones unregisterd when they try
+ reregister more requests are added. (closes issue #18898)
+ Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj,
+ irroot Jira:
+ https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review:
+ https://reviewboard.asterisk.org/r/1253
+
+2011-06-07 17:59 +0000 [r322189] Paul Belanger <pabelanger at digium.com>
+
+ * configs/sip_notify.conf.sample: Use correct syntax for 'sip
+ notify snom-reboot' (closes issue ASTERISK-17915)
+
+2011-06-06 19:07 +0000 [r322069] Jonathan Rose <jrose at digium.com>
+
+ * main/asterisk.c, include/asterisk/logger.h: Fixes level toggling
+ for logger set levels since it was reversed (closes issue
+ ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H
+ Review: https://reviewboard.asterisk.org/r/1244/
+
+2011-06-03 22:09 +0000 [r321812-321926] Richard Mudgett <rmudgett at digium.com>
+
+ * cdr/cdr_radius.c, cel/cel_radius.c: Asterisk crash when unloading
+ cdr_radius/cel_radius. The rc_openlog() API call is passed a
+ string that is used by openlog() to format log messages. The
+ openlog() does not copy the string it just keeps a pointer to it.
+ When the module is unloaded, the string is gone from memory.
+ Depending upon module load order and if the other module then has
+ an error, a crash happens. * Pass rc_openlog() a strdup'd string
+ with the understanding that there will be a small memory leak if
+ the cdr_radius/cel_radius modules are unloaded. * Call
+ rc_destroy() to free the rc handle memory when the module is
+ unloaded. JIRA AST-483 JIRA SWP-3062
+
+ * main/ccss.c: Be more explicit for CCSS generic device state event
+ subscription. Make CCSS generic device state event subscription
+ specify the AST_EVENT_IE_STATE ie exists to be safe.
+
+ * main/event.c, tests/test_event.c: Event subscription fixes. Must
+ commit the subscription fixes together with the integration
+ subscription tests. The subscription fixes cause an erroneously
+ passing test to fail. The new subscription tests detect errors
+ without the subscription fixes. * Added missing event_names[]
+ table entry. * Reworked
+ ast_event_check_subscriber()/match_sub_ie_val_to_event() to
+ correctly detect if a subscriber exists for the proposed event. *
+ Made match_ie_val() and match_sub_ie_val_to_event() check the
+ buffer length for RAW payload types. * Fixed error handling
+ memory leak in ast_event_sub_activate(), ast_event_unsubscribe(),
+ and ast_event_queue(). * Made ast_event_new() and
+ ast_event_check_subscriber() better protect themselves from an
+ invalid payload type. * Added container lock protection between
+ removing old cache events and adding the new cached event in
+ ast_event_queue_and_cache()/event_update_cache(). * Added new
+ event subscription tests.
+
+ * main/event.c, include/asterisk/event.h: Constify subscription
+ description parameter string.
+
+ * channels/chan_iax2.c, channels/chan_sip.c: Correct IAX2 and SIP
+ event subscription description string.
+
+2011-06-03 18:32 +0000 [r321753] Russell Bryant <russell at digium.com>
+
+ * tests/test_astobj2.c: Backport an astobj2 unit test so that it
+ runs on 1.8 as well.
+
+2011-06-03 13:17 +0000 [r321685] Leif Madsen <lmadsen at digium.com>
+
+ * configs/queues.conf.sample: Also document the 'queue-minute'
+ option. (closes issue #19386) Reported by: juanmol
+
+2011-06-01 23:11 +0000 [r321547] Richard Mudgett <rmudgett at digium.com>
+
+ * main/cdr.c: CDR comment tweaks.
+
+2011-06-01 20:10 +0000 [r321537] Brett Bryant <bbryant at digium.com>
+
+ * apps/app_voicemail.c: This patch fixes an issue with using the
+ wrong voicemail folders with greetings. (closes issue #17871)
+ Reported by: edhorton Patches: digium_bug_17871_2 uploaded by
+ fhackenberger (license 592) Tested by: edhorton, fhackenberger
+
+2011-06-01 10:40 +0000 [r321528] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c: Fix double alerting, add forced
+ alerting before answer Fix double alerting (it wasn't fixed here
+ by issue #18542) Add forced alerting before connect (if it wasn't
+ before) Try to send all packets from outgoing queue rather than
+ one only Call goes into clearing state when disconnect command is
+ received (closes issue #19361) Reported by: vmikhelson Patches:
+ issue19361-3.patch uploaded by may213 (license 454) Tested by:
+ vmikhelson
+
+2011-05-31 20:54 +0000 [r321517] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/dnsmgr.h, include/asterisk/acl.h: Update some
+ comments.
+
+2011-05-31 18:52 +0000 [r321515] David Vossel <dvossel at digium.com>
+
+ * channels/chan_local.c: Chan_local locking cleanup. This patch
+ removes all of the unnecessary deadlock avoidance loops that
+ occur in chan_local. It also resolves an issue with a deadlock
+ triggered by local channel optimizations. (issue #18028) Review:
+ https://reviewboard.asterisk.org/r/1231/
+
+2011-05-31 16:04 +0000 [r321511] Leif Madsen <lmadsen at digium.com>
+
+ * channels/chan_sip.c: Enhance NOTICE message to know who couldn't
+ access the dialplan. (closes issue #19390) Reported by: lmadsen
+ Patches: __20110531-sip-notice-tweak.txt uploaded by lmadsen
+ (license 10) Tested by: russell
+
+2011-05-28 00:27 +0000 [r321337-321436] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_agi.c: Some hagi launch cleanup. Inspired by issue 19256.
+ This patch would also fix the crash.
+
+ * main/srv.c: Crash when using hagi and no servers are available.
+ When none of the servers returned by the SRV querey respond,
+ asterisk crashes. The problem is that if the loop over all the
+ SRV entries finishes then the srv_context has already been
+ cleaned up. * Make ast_srv_cleanup() check to see if the context
+ is already cleaned up. (closes issue #19256) Reported by:
+ byronclark
+
+ * apps/app_privacy.c: The app_privacy args have undocumented
+ "options" position, interferes with "context" position. * Add
+ documention for unused "options" position to match existing code.
+ (closes issue #19273) Reported by: mdavenport
+
+2011-05-27 21:54 +0000 [r321333-321335] Leif Madsen <lmadsen at digium.com>
+
+ * include/asterisk/frame.h, main/file.c: Fix issue with playback of
+ H.261 video. (closes issue #19379) Reported by: neutrino88
+ Patches: videoprompt.patch uploaded by neutrino88 (license 297)
+ (changes by russell)
+
+ * main/features.c: Allow parking lot hints and musicclass to be
+ set. (closes issue #19378) Reported by: sboily_proformatique
+ Patches: pf_parkinghint_music_fix uploaded by sboily
+ proformatique (license 206) Tested by: russell
+
+2011-05-27 21:31 +0000 [r321330] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_privacy.c: The app_privacy args have undocumented
+ "options" position, interferes with "context" position. * Add
+ documention for unused "options" position to match existing code.
+ The trunk(v1.10) version will remove the unused options position.
+ (closes issue #19273) Reported by: mdavenport
+
+2011-05-27 14:59 +0000 [r321273] Jonathan Rose <jrose at digium.com>
+
+ * channels/sip/reqresp_parser.c: markm committed a patch I was
+ working on yesterday, this fixes it to mesh up with suggestions
+ by mnicholson.
+
+2011-05-27 08:31 +0000 [r321211] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/features.c: Fix *8 directed pickup locks system during
+ pickupsound play out move playout from sip_pickup_thread to
+ bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2
+ threads trying to write audio to same channel. In addition fixes
+ choppy audio beep in issue 19177. (issue #18654) (issue #19177)
+ Reported by: Docent Patches: review1232-1.88888888 alecdavis
+ (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1232/
+
+2011-05-26 21:48 +0000 [r321100-321155] Mark Murawki <markm at intellasoft.net>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c: Fixed build
+ problem with dev mode enabled, which was caused by commit 321100.
+ Reformulated patch to be more generic. Moved the sip uri parse
+ variable initalization to parse_uri_full in reqresp_parser.c.
+ This will ensure that any use of parse uri will have null output
+ variables if the parse fails. (closes issue #19346) Reported by:
+ kobaz Tested by: kobaz,JonathanRose Review: [full review board
+ URL with trailing slash]
+
+ * main/netsock2.c, channels/chan_sip.c: ast_sockaddr_resolve() in
+ netsock2.c may deref a null pointer Added a null check in
+ netsock2 ast_sockaddr_resolve() as well as added default
+ initalizers in chan_sip parse_uri_legacy_check() to make sure
+ that invalid uris will make null (and not undefined)
+ user,pass,domain,transport variables (closes issue #19346)
+ Reported by: kobaz Patches: netsock2.patch uploaded by kobaz
+ (license 834) Tested by: kobaz, Marquis
+
+2011-05-26 18:10 +0000 [r321044] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/netsock2.h: Update ast_sockaddr comment with an
+ important note.
+
+2011-05-26 17:29 +0000 [r321042] Terry Wilson <twilson at digium.com>
+
+ * main/rtp_engine.c: Initialize stack-allocated ast_sockaddrs
+ before use It is important to always initialize ast_sockaddrs
+ before use--even if they are passed to ast_sockaddr_copy as the
+ underlying storage could be bigger than what ends up being
+ copied--leaving part of the data unitialized.
+
+2011-05-26 15:57 +0000 [r320947] Russell Bryant <russell at digium.com>
+
+ * channels/chan_alsa.c, channels/chan_mgcp.c: Remove some variables
+ that were set but unused.
+
+2011-05-25 22:25 +0000 [r320796-320883] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Native SIP CCSS sends bad CC cancel
+ SUBSCRIBE message. The SUBSCRIBE message used to cancel a CC
+ request has incorrect To/From SIP headers. They are reversed and
+ the dialog tags are the same when they should not be. If pedantic
+ mode was disabled, then the cancel would have succeeded despite
+ the incorrect message. * The SIP_OUTGOING flag was not set
+ correctly for the dialog and I had to move some CC subscribe
+ handling code as a result. * Initialized the dialog subscribed
+ type to CALL_COMPLETION earlier. If a CC request SUBSCRIBE
+ message comes in and the CC instance is not found, the 404
+ response was duplicated. JIRA AST-568 JIRA SWP-3493
+
+ * UPGRADE.txt, CHANGES, apps/app_queue.c, apps/app_dial.c,
+ main/channel.c, main/manager.c, apps/app_meetme.c,
+ apps/app_fax.c, main/features.c: The AMI Newstate event contains
+ different information between v1.4 and v1.8. The addition of
+ connected line support in v1.8 changes the behavior of the
+ channel caller ID somewhat. The channel caller ID value no longer
+ time shares with the connected line ID on outgoing call legs. The
+ timing of some AMI events/responses output the connected line ID
+ as caller ID. These party ID's are now separate. * The
+ ConnectedLineNum and ConnectedLineName headers were added to many
+ AMI events/responses if the CallerIDNum/CallerIDName headers were
+ also present. (closes issue #18252) Reported by: gje Tested by:
+ rmudgett Review: https://reviewboard.asterisk.org/r/1227/
+
+ * include/asterisk/channel.h, main/channel.c, main/features.c: Give
+ zombies a safe channel driver to use. Recent crashes from zombie
+ channels suggests that they need a safe home to goto. When a
+ masquerade happens, the physical part of the zombie channel is
+ hungup. The hangup normally sets the channel private pointer to
+ NULL. If someone then blindly does a callback to the channel
+ driver, a crash is likely because the private pointer is NULL.
+ The masquerade now sets the channel technology of zombie channels
+ to the kill channel driver. Related to the following issues:
+ (issue #19116) (issue #19310) Review:
+ https://reviewboard.asterisk.org/r/1224/
+
+2011-05-25 00:49 +0000 [r320716] Terry Wilson <twilson at digium.com>
+
+ * addons/chan_mobile.c: Cast data as char * before using S_OR This
+ is required for compiling successfully under dev mode
+
+2011-05-23 17:53 +0000 [r320650] Richard Mudgett <rmudgett at digium.com>
+
+ * CHANGES, main/manager.c: Add ConnectedLineNum/Name headers to
+ output of AMI action Status. * Add ConnectedLineNum and
+ ConnectedLineName headers to the output of the AMI action Status.
+ This makes it easier to find out who the channel is connected to
+ without having to lookup BridgedChannel or when they are
+ connected to an application (e.g.: VoiceMail) which has no
+ bridged channel. * Bridged channels with no CallerID had ""
+ instead of "<unknown>" output, that might be a bug as "<unknown>"
+ was what older versions used. (closes issue #18158) Reported by:
+ gareth Patches: svn-292308.diff uploaded by gareth (license 208)
+
+2011-05-23 16:19 +0000 [r320573] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * configure, configure.ac: GNU libiconv uses symbol "libiconv_open"
+ instead of "iconv_open". (closes issue #19344) Reported by:
+ rohanl Patches: iconv-check.patch uploaded by rohanl (license
+ 1284)
+
+2011-05-23 16:18 +0000 [r320568] David Vossel <dvossel at digium.com>
+
+ * main/tcptls.c, /: Merged revisions 320562 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r320562 | dvossel | 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011)
+ | 9 lines Adds missing part to the ast_tcptls_server_start fails
+ second attempt to bind patch. (closes issue #19289) Reported by:
+ wdoekes Patches:
+ issue19289_delay_old_address_setting_tcptls_2.patch uploaded by
+ wdoekes (license 717) ........
+
+2011-05-23 15:47 +0000 [r320560] Kevin P. Fleming <kpfleming at digium.com>
+
+ * configure, configure.ac: Don't generate spurious "No: command not
+ found" messages when running the configure script on a system
+ that has neither gmime-config nor pkg-config.
+
+2011-05-23 14:33 +0000 [r320504] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: Fixes segfault occuring in chan_sip.c at
+ __set_address_from_contact Checks to see if domain contains
+ anything before sending it off to ast_sockaddr_resolve which is
+ where the segfault was occuring due to null str. (closes issue
+ #18857) Reported by: sybasesql Review:
+ https://reviewboard.asterisk.org/r/1225/
+
+2011-05-22 23:34 +0000 [r320445] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * res/res_odbc.c, /: Merged revisions 320444 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r320444 | tilghman | 2011-05-22 18:25:51 -0500 (Sun, 22 May 2011)
+ | 8 lines Don't crash when the connection fails. (closes issue
+ #19250) Reported by: seadweller Patches:
+ 20110514__issue19250.diff.txt uploaded by tilghman (license 14)
+ Tested by: seadweller, sum ........
+
+2011-05-20 21:39 +0000 [r320338] David Vossel <dvossel at digium.com>
+
+ * main/tcptls.c, /: Merged revisions 320271 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r320271 | dvossel | 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011)
+ | 8 lines Fixes issue with ast_tcptls_server_start failing on
+ second attempt to bind. (closes issue #19289) Reported by:
+ wdoekes Patches:
+ issue19289_delay_old_address_setting_tcptls.patch uploaded by
+ wdoekes (license 717) ........
+
+2011-05-20 20:49 +0000 [r320237] Richard Mudgett <rmudgett at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 320236 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500
+ (Fri, 20 May 2011) | 20 lines Merged revisions 320235 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011)
+ | 13 lines The meetme CLI command completion leaves conferences
+ mutex locked. When issuing a meetme kick CLI command and an
+ invalid (non-existent) conference number is specified, pressing
+ Tab leaves the conferences mutex locked and, therefore, all
+ conferences deadlock. Add missing unlock. (closes issue #19336)
+ Reported by: zvision Patches: app_meetme.diff uploaded by zvision
+ (license 798) ........ ................
+
+2011-05-20 18:48 +0000 [r320180] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: This commit modifies the way polling is done
+ on TLS sockets. Because of the buffering the TLS layer does,
+ polling is unreliable. If poll is called while there is data
+ waiting to be read in the TLS layer but not at the network layer,
+ the messaging processing engine will not proceed until something
+ else writes data to the socket, which may not occur. This change
+ modifies the logic around TLS sockets to only poll after a failed
+ read on a non-blocking socket. This way we know that there is no
+ data waiting to be read from the buffering layer. (closes issue
+ #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson
+
+2011-05-20 18:12 +0000 [r320162] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_voicemail.c: Fixes an imapfolder related crash
+ imapfolders being set in the general section of voicemail would
+ cause the inbox folder name to change. Since sound file names are
+ made based on the names of the folders, this would cause the
+ audio related to that folder name to change and if Asterisk
+ attempted to play it, the channel would instantly hang up when
+ the audio file couldn't be found. This patch searches for the
+ name of the folder first to leave existing behavior in tact and
+ if that fails, it uses the normal inbox name to get the sound
+ file instead. (closes issue #16104) Reported by: blkline Review:
+ https://reviewboard.asterisk.org/r/1215/
+
+2011-05-20 17:03 +0000 [r319997-320059] Richard Mudgett <rmudgett at digium.com>
+
+ * main/features.c: Misc comment cleanup in features.c.
+
+ * main/channel.c, main/features.c: Crash while transferring a call
+ during DTMF feature timeout. When a call is being attended
+ transferred during the time between AST_FRAME_DTMF_BEGIN and
[... 31376 lines stripped ...]
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