[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:
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    tags/1.8.5-rc1/ChangeLog   (with props)

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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

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