[asterisk-commits] tag 1.2.10-netsec r37642 - in /tags/1.2.10-netsec: .lastclean .version ChangeLog

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Fri Jul 14 14:32:20 MST 2006


Author: kpfleming
Date: Fri Jul 14 16:32:20 2006
New Revision: 37642

URL: http://svn.digium.com/view/asterisk?rev=37642&view=rev
Log:
importing files for 1.2.10-netsec release

Added:
    tags/1.2.10-netsec/.lastclean   (with props)
    tags/1.2.10-netsec/.version   (with props)
    tags/1.2.10-netsec/ChangeLog   (with props)

Added: tags/1.2.10-netsec/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.2.10-netsec/.lastclean?rev=37642&view=auto
==============================================================================
--- tags/1.2.10-netsec/.lastclean (added)
+++ tags/1.2.10-netsec/.lastclean Fri Jul 14 16:32:20 2006
@@ -1,0 +1,1 @@
+8

Propchange: tags/1.2.10-netsec/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/1.2.10-netsec/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/1.2.10-netsec/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/1.2.10-netsec/.version
URL: http://svn.digium.com/view/asterisk/tags/1.2.10-netsec/.version?rev=37642&view=auto
==============================================================================
--- tags/1.2.10-netsec/.version (added)
+++ tags/1.2.10-netsec/.version Fri Jul 14 16:32:20 2006
@@ -1,0 +1,1 @@
+1.2.10-netsec

Propchange: tags/1.2.10-netsec/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/1.2.10-netsec/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/1.2.10-netsec/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/1.2.10-netsec/ChangeLog
URL: http://svn.digium.com/view/asterisk/tags/1.2.10-netsec/ChangeLog?rev=37642&view=auto
==============================================================================
--- tags/1.2.10-netsec/ChangeLog (added)
+++ tags/1.2.10-netsec/ChangeLog Fri Jul 14 16:32:20 2006
@@ -1,0 +1,2804 @@
+2006-07-14 Kevin P. Fleming <kpfleming at digium.com>
+
+	* Asterisk 1.2.10 released
+
+2006-07-14 13:31 +0000 [r37612]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_sms.c: Bug 7526 - previous commit broke app_sms
+
+2006-07-13 21:22 +0000 [r37571]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_voicemail.c: don't fail/abort if the message category
+	  sound file cannot be played, just generate a warning message and
+	  continue message playback
+
+2006-07-13 18:44 +0000 [r37546]  Russell Bryant <russell at digium.com>
+
+	* rtp.c: yeah, ummm... This frame pointer should not be static.
+	  This situation only exists in 1.2 (pointed out by Constantine
+	  Filin on the asterisk-dev mailing list)
+
+2006-07-13 16:44 +0000 [r37531]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: report address of peer trying to subscribe
+	  to unknown hint
+
+2006-07-13 15:45 +0000 [r37458-37516]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* doc/README.enum: Bug 7532 - Typo in enum example
+
+	* contrib/init.d/rc.mandrake.zaptel: Merge fixup for asterisk
+	  startup script to zaptel startup script
+
+2006-07-12 15:53 +0000 [r37441-37442]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_voicemail.c: fix a weird case where a lock file could be
+	  left (but would happen almost never)
+
+	* app.c: fix a case where ast_lock_path() could leave a
+	  randomly-named lock file hanging around make ast_unlock_path
+	  actually report when unlocking fails
+
+2006-07-12 15:23 +0000 [r37439]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_iax2.c: Add support to have maxauthreq as a global
+	  option
+
+2006-07-12 13:54 +0000 [r37417-37419]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_zap.c, utils.c, res/res_agi.c, apps/app_zapras.c,
+	  asterisk.c, channels/chan_modem.c, channels/chan_iax2.c: remove
+	  some more bad examples of using printf
+
+	* enum.c, pbx/pbx_config.c: get rid of some more printf's (although
+	  most of these were ifdef-ed out)
+
+2006-07-12 03:55 +0000 [r37402]  Matt O'Gorman <mogorman at digium.com>
+
+	* app.c: GRRR no fprintf!
+
+2006-07-11 19:00 +0000 [r37378]  Joshua Colp <jcolp at digium.com>
+
+	* configs/iax.conf.sample, channels/chan_iax2.c: Add configuration
+	  option for IAX2 users that will limit the amount of outstanding
+	  AUTHREQs we are waiting for replies on.
+
+2006-07-10 21:01 +0000 [r37361]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channel.c: do masquerade-behind-proxy checking with better
+	  control over locks
+
+2006-07-07 23:57 +0000 [r37307]  Joshua Colp <jcolp at digium.com>
+
+	* rtp.c: Change message regarding marker bit forcing when SSRC
+	  changes to be shown only during debug so it doesn't overload high
+	  capacity systems
+
+2006-07-06 21:41 +0000 [r37224]  Matt O'Gorman <mogorman at digium.com>
+
+	* channel.c: patch resolves issue with when to decide if its right
+	  time to native bridge, feature redirect was not being checked.
+	  patch from bug #7296
+
+2006-07-06 20:38 +0000 [r37212]  BJ Weschke <bweschke at btwtech.com>
+
+	* channels/chan_agent.c: Don't do weird things on a callback agent
+	  that has attempted logoff while still on the phone.
+
+2006-07-06 15:48 +0000 [r37173]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Instead of giving the scheduled item ID on a
+	  peer expiration, give the time until they expire (issue #7455
+	  reported by slavon)
+
+2006-07-06 13:47 +0000 [r37143]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* funcs/func_db.c: Fix spelling/grammar (issue 7493)
+
+2006-07-05 15:31 +0000 [r36998]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_oss.c: Spell extension correctly in documentation
+	  for chan_oss dial (issue #7487 reported by flefoll)
+
+2006-07-04 14:45 +0000 [r36838-36911]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Tell clients based on old SIP standard that
+	  we only support MD5 digest authentication...
+
+	* channels/chan_sip.c: issue #7470 - Need larger buffer for
+	  record-route headers...
+
+2006-07-03 05:12 +0000 [r36697-36751]  Russell Bryant <russell at digium.com>
+
+	* asterisk.c: fix a race condition that caused asterisk to log a
+	  *ton* of warnings on mac osx about poll returning an error
+	  because the polled file descriptor was bad.
+
+	* channels/chan_mgcp.c, channels/chan_phone.c,
+	  channels/chan_local.c, channels/chan_misdn.c,
+	  channels/chan_sip.c, channels/chan_skinny.c,
+	  channels/chan_agent.c, channels/chan_features.c,
+	  channels/chan_h323.c, channels/chan_modem.c,
+	  channels/chan_iax2.c: use ast_set_callerid to be more consistent
+	  and to make sure that the "callerid" option in the conf files is
+	  always handled the same way and sets ANI (issue #7285, gkloepfer)
+
+	* dsp.c: fix the build with BUSYDETECT_TONEONLY defined (issue
+	  #7414)
+
+2006-06-30 14:05 +0000 [r36290-36377]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_directory.c: Bug 7349 - Directory did not work correctly
+	  when USE_ODBC_STORAGE was defined.
+
+	* Makefile: Bug 7388 - compatibility changes for Solaris
+
+2006-06-29 07:19 +0000 [r36253-36254]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* configs/queues.conf.sample: clarify documentation for
+	  'persistentmembers' setting
+
+	* configs/sip.conf.sample: add documentation for peer-specific
+	  'outboundproxy' setting
+
+2006-06-28 14:12 +0000 [r36187]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Don't delete scheduled item twice in
+	  sip_destroy (already fixed in svn trunk)
+
+2006-06-26 17:10 +0000 [r36078]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: ensure that two SIP channels that exist at
+	  the same moment will not have the same channel names (issue
+	  #7245, different fix)
+
+2006-06-26 15:27 +0000 [r36043]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Issue 6997 maybe, but anyway - don't
+	  retransmit responses to NON-invite requests.
+
+2006-06-25 15:10 +0000 [r35915]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* channels/chan_sip.c: Bug 7425 - Size of buffer is passed in by
+	  len
+
+2006-06-23 11:30 +0000 [r35669]  BJ Weschke <bweschke at btwtech.com>
+
+	* apps/app_queue.c: We should lock the queue before we go making
+	  changes to member interface statuses.
+
+2006-06-21 19:25 +0000 [r35334]  Joshua Colp <jcolp at digium.com>
+
+	* configs/indications.conf.sample: Add Venezuelan indications
+	  (issue #7402 reported by palillo)
+
+2006-06-20 15:05 +0000 [r35121]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* stdtime/private.h: Bug 7398 - Solaris puts its zoneinfo files in
+	  a nonstandard place
+
+2006-06-20 10:27 +0000 [r35058]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Issue #6820 - Possible fix (already
+	  implemented in trunk)
+
+2006-06-19 20:27 +0000 [r34911]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_voicemail.c: Call reset_user_pw upon changing the
+	  password using externpass (issue #7395 reported by Ryan Cumming)
+
+2006-06-19 18:07 +0000 [r34875]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Issue 7357 - txt file left behind when
+	  going to operator. Also, fix a possible file descriptor leak.
+
+2006-06-18 21:03 +0000 [r34627-34655]  Russell Bryant <russell at digium.com>
+
+	* pbx.c: don't set state to BUSY if the channel is already in the
+	  UP state (issue #7376, backported from trunk)
+
+	* configs/iax.conf.sample, channels/chan_iax2.c: don't store
+	  multiple secrets delimited with semicolons for peers because this
+	  is only valid for users. Instead, only keep the last specified
+	  secret for a peer entry. Also, document how multiple secrets are
+	  handled in the sample config. (Reported by PCadach on
+	  #asterisk-bugs)
+
+2006-06-16 03:37 +0000 [r34400]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_iax2.c: Zero out a declared structure so as to not
+	  crash if it contains invalid data (reported by Qwell on
+	  #asterisk-dev)
+
+2006-06-15 14:11 +0000 [r34306]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Issue 7294 - patch by phsultan - Asterisk
+	  sends Invite instead of BYE in some cases.
+
+2006-06-15 13:30 +0000 [r34274]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_queue.c: don't use prefixed structure names for internal
+	  structures don't use a plural structure name for a singular
+	  object
+
+2006-06-15 12:40 +0000 [r34242]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: VoicemailMain exits on any key, when the
+	  language is set to Italian, instead of properly handling the key
+	  (issue 7353).
+
+2006-06-14 22:22 +0000 [r33841-34160]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_queue.c: coding style cleanups on queue interface
+	  handling code that was committed for the last release
+
+	* channels/chan_iax2.c: use existing dial string parser for strings
+	  supplied to iax2_devicestate, because they can be complete dial
+	  strings, not just device names
+
+	* include/asterisk/plc.h, jitterbuf.c, plc.c, apps/app_dumpchan.c,
+	  apps/app_chanspy.c: clarify file headers that mention disclaimer
+	  usage
+
+	* file.c: don't output 'no format found' when we _did_ find the
+	  format but couldn't open the desired file for some other reason
+
+	* apps/app_mixmonitor.c: memory allocation optimizations
+
+2006-06-13 12:40 +0000 [r33753-33813]  Russell Bryant <russell at digium.com>
+
+	* pbx.c: remove duplicate mutex_unlock
+
+	* apps/app_voicemail.c: fix various places where the code returns
+	  without unlocking vmlock or destroying loaded configuration
+
+	* apps/app_festival.c: add a missing close of an open fd, destroy
+	  of open config, and removal of the calling channel from the
+	  localusers list
+
+	* asterisk.c: revert a change that caused more problems than it
+	  fixed and fix the real problem in this code. fds was declared as
+	  an array of zero size which caused some weird problems, some of
+	  which would only be seen when compiling without optimizations.
+	  (fixes issues #7071, #7326, and #7305)
+
+2006-06-12 21:34 +0000 [r33724]  Joshua Colp <jcolp at digium.com>
+
+	* include/asterisk/chanspy.h, apps/app_mixmonitor.c, channel.c:
+	  Greatly simply the mixmonitor thread, and move channel reference
+	  directly to spy structure so that the core can modify it.
+
+2006-06-12 20:40 +0000 [r33693]  Russell Bryant <russell at digium.com>
+
+	* res/res_agi.c: fix a place where a frame would be free'd twice
+
+2006-06-12 16:03 +0000 [r33638]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_local.c: only allow chan_local to masquerade the
+	  outbound channel onto its owner, instead of the other way around
+	  (this will ensure that group variables on the outbound channel are
+	  preserved)
+
+2006-06-12 15:27 +0000 [r33615]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* res/res_agi.c: Move set priority up, because at this point in the
+	  code, stdout is no longer the console. If we're unable to set
+	  priority, the error goes to Asterisk as if it were an AGI command
+	  (issue 7335).
+
+2006-06-11 21:21 +0000 [r33449-33548]  Russell Bryant <russell at digium.com>
+
+	* pbx.c: fix another place where a frame does not get free'd
+
+	* apps/app_meetme.c: fix up five little places where frames would
+	  not be free'd and remove an unnecessary mutex_unlock where there
+	  is no way for it to be locked at that time
+
+	* apps/app_ices.c: fix a place that would leak a frame (all of
+	  these fixes are in applications that call ast_read() on a channel
+	  but have code paths in them that would not free the frame)
+
+	* apps/app_festival.c: fix a couple places that would leak a frame
+
+	* apps/app_alarmreceiver.c: fix two places that would cause a frame
+	  to be leaked
+
+	* apps/app_url.c: fix a case where an HTML frame would be leaked
+
+	* apps/app_test.c: Free frames read from the channel when measuring
+	  noise. This resulted in about 9 or 10 seconds of leaked frames in
+	  both the TestClient and TestServer applications
+
+	* apps/app_zapbarge.c, apps/app_zapscan.c: backport a couple of
+	  frame leak fixes from the trunk (revisions 33446, 33447)
+
+2006-06-09 18:52 +0000 [r33264-33300]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_meetme.c: Allow the format outputted by meetme list to
+	  be used for meetme commands (like kick) (issue #7322 reported by
+	  darkskiez)
+
+	* channels/chan_iax2.c: Remove an unneeded double lock (issue #7310
+	  reported by arkadia)
+
+	* apps/app_dial.c: Handle hangup during recording of screened name
+	  (issue #7304 reported by kulldominique)
+
+	* apps/app_meetme.c: Add missing newlines (issue #7323 reported by
+	  darkskiez)
+
+2006-06-09 15:53 +0000 [r33235]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Do not require a context on a domain=
+	  setting
+
+2006-06-08 16:57 +0000 [r33036]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* frame.c: handle out-of-memory conditions properly in
+	  ast_frisolate() (reported by Slav Kenov on asterisk-dev mailing
+	  list)
+
+2006-06-07 17:53 +0000 [r32818]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: fix some broken code with
+	  BRIDGE_OPTIMIZATION defined (issue #7292)
+
+2006-06-06 16:55 +0000 [r32605]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Bug 7287 - A too short voicemail with
+	  ODBC_STORAGE will cause the first voicemail to be deleted
+	  erroneously
+
+2006-06-06 Kevin P. Fleming <kpfleming at digium.com>
+
+	* Asterisk 1.2.9.1 released
+
+2006-06-06 16:02 +0000 [r32582]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* callerid.c: Bug 7268 - Callerid leaks memory on error
+
+2006-06-06 15:48 +0000 [r32566]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_iax2.c: clean up yesterday's security fix to not
+	  cause breakage when video mini frames are received
+
+2006-06-03 Kevin P. Fleming <kpfleming at digium.com>
+
+	* Asterisk 1.2.9 released
+
+2006-06-05 19:53 +0000 [r32373]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_iax2.c: ensure that the received number of bytes is
+	  included in all IAX2 incoming frame analysis checks (fixes a
+	  known vulnerability)
+
+2006-06-04 03:43 +0000 [r31921]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_queue.c: return bridge exit logic to what it was before
+	  i broke it :-(
+
+2006-06-03 17:02 +0000 [r31775]  Russell Bryant <russell at digium.com>
+
+	* res/res_musiconhold.c: when using moh files mode, don't look for
+	  a file past the number of files that have been loaded, or worse,
+	  past the size of the files array
+
+2006-06-01 21:46 +0000 [r31321-31555]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* res/res_musiconhold.c: remove pointless forcing of the channel
+	  into SLINEAR mode; the write format will be set later based on
+	  the file that is chosen to be played to the channel
+
+	* include/asterisk/channel.h, channel.c: handle Zap transfers
+	  behind chan_agent properly so the agent doesn't get stuck waiting
+	  for the call to hang up
+
+	* configs/sip.conf.sample: remove a sample entry that never should
+	  have been added (code to support it was not merged)
+
+2006-05-31 23:50 +0000 [r31194]  Russell Bryant <russell at digium.com>
+
+	* res/res_agi.c: if the connection to a FastAGI server fails
+	  because of a timeout, log a more informative log message
+
+2006-05-31 22:26 +0000 [r31161]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* rtp.c: silence a warning message that is not a warning
+
+2006-05-31 20:26 +0000 [r31127]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_zap.c: fix misplaced manager event (issue #6866,
+	  flefoll)
+
+2006-05-30 Kevin P. Fleming <kpfleming at digium.com>
+
+	* Asterisk 1.2.8 released
+
+2006-05-30 14:55 +0000 [r30770]  BJ Weschke <bweschke at btwtech.com>
+
+	* apps/app_queue.c: Fix infinite loop scenario and add some sanity
+	  checking to prevent segfault on a NULL parameter coming in (which
+	  probably shouldn't happen, but just to be safe...)
+
+2006-05-26 17:09 +0000 [r30424-30546]  BJ Weschke <bweschke at btwtech.com>
+
+	* apps/app_queue.c: A new way to try and deal with deadlocks that
+	  occur in app_queue at present. Using this approach, we only
+	  manipulate the main queue mutexes when we get a dev state change
+	  on a device that is actually a member of a queue. Backported from
+	  /trunk for the "bug fix".
+
+2006-05-25 20:03 +0000 [r30373]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_meetme.c: Don't play the enter sound twice when a person
+	  joins a conference after the leader has joined it. (issue #6138
+	  reported by shanermn)
+
+2006-05-25 17:39 +0000 [r30293-30296]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* codecs/gsm/Makefile: don't try to use -march=s390 when building
+	  on S/390 systems (reported via asterisk-users mailing list)
+
+	* channels/chan_sip.c: allow SIPCHANINFO(peername) to work for
+	  calls from users as well (issue #7215)
+
+2006-05-25 15:27 +0000 [r30239]  Joshua Colp <jcolp at digium.com>
+
+	* configs/extensions.conf.sample: Get rid of an incorrect SIP dial
+	  string in the sample extensions.conf - I even tried variations...
+	  no go (issue #7222 reported by arkadia)
+
+2006-05-24 21:24 +0000 [r30069-30098]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: oops... make sure to stop processing a
+	  request once we have sent an authentication challenge (issue
+	  #7220)
+
+	* channels/chan_sip.c: don't send CANCEL on a pending INVITE if we
+	  haven't received a provisional response yet... mark it pending
+	  until the first response is received (issue #7079)
+
+2006-05-24 19:55 +0000 [r30037]  Matt O'Gorman <mogorman at digium.com>
+
+	* apps/app_meetme.c: app_meetme used the ast_max_exten instead of
+	  path_max solves bug 6822
+
+2006-05-24 19:44 +0000 [r30033-30035]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_dial.c: Merge branch for bug 6264 (Privacy option 2
+	  returns dial-status ANSWER / option_priority_jumping not
+	  respected) (reported by jkoopmann and branch by murf)
+
+	* logger.c: Fix deadlock caused by a race condition in the logger
+	  when reloading (issue #7195 reported and fixed by softins)
+
+2006-05-24 16:59 +0000 [r29904-29973]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* res/res_agi.c: support video recording via AGI 'RECORD FILE'
+	  command (issue #7068)
+
+	* apps/app_queue.c: fix various bugs related to exiting from queue
+	  via keypress and moh handling (issue #6776, different fix)
+
+	* channels/chan_zap.c: respect 'usecallingpres' in zapata.conf even
+	  if CLID has not been set for the channel (issue #7123)
+
+	* channels/chan_sip.c, configs/sip.conf.sample: add an option to
+	  allow the admin to 'hide' SIP user/peer names from systems trying
+	  to 'fish' names
+
+2006-05-23 21:44 +0000 [r29849]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: fix the sourceaddress option (issue #7213,
+	  alphaque)
+
+2006-05-23 18:16 +0000 [r29764]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: simplify/fix lock retry, and fix comment
+
+2006-05-23 17:17 +0000 [r29733]  BJ Weschke <bweschke at btwtech.com>
+
+	* channels/chan_sip.c: Sanity check code for an extended failure in
+	  trying to obtain a channel lock that may have been obtained
+	  elsewhere. Prevents the monitor thread of the SIP module from
+	  going into an infinite loop, effectively, breaking SIP until you
+	  restart Asterisk or the mutex is unlocked, whichever comes first.
+
+2006-05-23 17:15 +0000 [r29732]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* dnsmgr.c, res/res_features.c, include/asterisk/linkedlists.h,
+	  include/asterisk/lock.h, apps/app_sql_postgres.c, pbx.c: backport
+	  some mutex initialization and linked list handling fixes from
+	  trunk
+
+2006-05-23 15:58 +0000 [r29696]  BJ Weschke <bweschke at btwtech.com>
+
+	* res/res_features.c: Fix a potential leak and correct (hopefully)
+	  a segfault under certain conditions. #6784 (vovan and perry
+	  testing)
+
+2006-05-22 21:27 +0000 [r29464-29555]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_waitforsilence.c: Increase the silence threshold to 128
+	  to "fix" it, so I'm told. (issue #6595 reported by davetroy fixed
+	  by casper)
+
+	* res/res_features.c: Use the correct language when playing the
+	  transfer sound (issue #7109 reported by casper)
+
+	* channels/chan_local.c: Preserve presentation bit when going
+	  through chan_local (issue #7002 reported by acunningham)
+
+2006-05-22 14:59 +0000 [r29394-29398]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_meetme.c: Bug 7194 - spelling fix
+
+	* pbx.c: Bug 7196 - month range did not work
+
+2006-05-21 15:16 +0000 [r29196]  BJ Weschke <bweschke at btwtech.com>
+
+	* res/res_features.c: When an application that is executed via
+	  applicationmap and exits non-zero, make sure that we pass through
+	  the correct return value from the application to make sure a
+	  segfault doesn't occur by a bridge trying to continue when it
+	  should not. Also, when executing applications via applicationmap,
+	  make sure that the application is executed against the channel
+	  whose DTMF caused it to be fired off in the first place. (part
+	  1/2 of #7090 - this is the only fix that will be applied to both
+	  1.2 and /trunk) acunningham and blitzrage on testing...
+
+2006-05-20 19:50 +0000 [r29052]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: fix the possibility of writing one byte past
+	  the end of a buffer. (issue #7189, Mithraen)
+
+2006-05-20 02:35 +0000 [r28968]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_queue.c: don't allow queue member devices to ring longer
+	  than the total queue timeout (issue #6423, reported and patched
+	  by bcnit)
+
+2006-05-20 02:31 +0000 [r28966]  Russell Bryant <russell at digium.com>
+
+	* apps/app_sms.c: fix a case where code made assumptions about how
+	  memory for variables is allocatted on the stack - this patch is
+	  slightly different than the one that went in for the trunk
+
+2006-05-20 00:55 +0000 [r28794-28896]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_iax2.c: don't try to predict where the compiler
+	  will place things on the stack... put them in the right place
+	  explicitly (issues #7029 and #7100, maybe others)
+
+	* channels/chan_sip.c: use the specified 'subscribecontext' for a
+	  peer rather than the context found via the target domain (domain
+	  contexts are for calls, not for subscriptions) (issue #7122,
+	  reported by raarts)
+
+2006-05-19 19:18 +0000 [r28754-28790]  Russell Bryant <russell at digium.com>
+
+	* utils/smsq.c: fix the build of smsq with -Werror. I learned
+	  something new about format strings from this patch! (issue #7141,
+	  Mithraen)
+
+	* asterisk.c: This explicit poll is only needed on mac. In fact, it
+	  breaks some systems such as some versions of Fedora, causing
+	  'asterisk -rx' to never exit. This has been tested on systems
+	  showing the asterisk -rx problem, as well as other unaffected
+	  versions of linux, mac osx 10.4, and FreeBSD 6. (issue #7071)
+
+2006-05-19 17:04 +0000 [r28627-28698]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_zap.c: Make the minidle option actually exist as
+	  documented (issue #7159 reported by imran)
+
+	* apps/app_voicemail.c: When forwarding messages use the context
+	  that the active voicemail user was found in. (issue #7010)
+
+	* enum.c: Backport of fix for issue #6654 that was fixed in trunk
+	  but not here
+
+	* apps/app_queue.c: Treat paused queue members as unreachable
+	  (issue #7127 reported by peterh)
+
+2006-05-18 20:43 +0000 [r28335-28384]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: fix up a few more places to find the SDP
+	  properly (fallout from fix for #7124)
+
+	* channels/chan_sip.c: handle incoming multipart/mixed message
+	  bodies in SIP and find the SDP, if present (issue #7124 reported
+	  and patched by eborgstrom, but very different fix)
+
+	* enum.c: use unsigned counters for handling answer/IE lengths
+	  while processing DNS results (issue #7174)
+
+	* channels/chan_sip.c: support 'inactive' tag for SDP media streams
+	  (simple fix, proper fix will appear in 1.4 release) (issue #7130)
+
+2006-05-18 17:27 +0000 [r28257]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_hasnewvoicemail.c: Bug 7167 - HasNewVoicemail and
+	  VMCOUNT() didn't work when USE_ODBC_STORAGE was defined
+
+2006-05-18 16:31 +0000 [r28169-28212]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_voicemail.c: Return -1 on error in ODBC messagecount and
+	  0 on success (issue #7133 reported by cfieldmtm)
+
+	* apps/app_voicemail.c: Fix endless looping message by checking
+	  value of res before doing retries stuff. (issue #7140 reported by
+	  tanischen)
+
+2006-05-18 12:13 +0000 [r28125]  Olle Johansson <oej at edvina.net>
+
+	* apps/app_meetme.c: Video in meetme? Hmmm. Removed until we do
+	  have some code for it.
+
+2006-05-17 22:34 +0000 [r27973]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_iax2.c: Fix codec priority stuff during
+	  authentication (issue #6194 reported by jkoopmann)
+
+2006-05-17 19:27 +0000 [r27927]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Issue #7176 - Crash in expire_register (We
+	  need to find out what's causing peer to be undefined, so this is
+	  just a bandaid, not a real fix)
+
+2006-05-17 17:07 +0000 [r27767-27847]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_voicemail.c: Priority jumping not working on VoiceMail
+	  app with new syntax (issue #7164 reported and fixed by
+	  alvaro_palma_aste)
+
+	* apps/app_osplookup.c: OSPNext does not handle success/failure
+	  correctly (issue #7147 reported and fixed by eborgstrom)
+
+2006-05-17 09:21 +0000 [r27723]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: chan_sip did not use the TRANSFER_CONTEXT
+	  for transfers, like res_features. Now fixed.
+
+2006-05-17 02:19 +0000 [r27636]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Bug 7125 - Fix race condition between
+	  resequencing and leaving a message
+
+2006-05-16 23:31 +0000 [r27594]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_dial.c: Inherit channel variables during call forwards
+	  when going through chan_local (issue #7095 reported by raarts)
+
+2006-05-16 20:05 +0000 [r27468]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channel.c: don't leak frames when deferring DTMF or dropping
+	  duplicate ANSWER frames (issue #7041, slightly different fix,
+	  reported/patched by clausf)
+
+2006-05-13 04:08 +0000 [r27093]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Bug 7134 - File descriptor leak with ODBC
+	  storage of voicemail
+
+2006-05-11 23:02 +0000 [r27051]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* funcs/func_logic.c: Bug 7086 - pbx_checkcondition substitution,
+	  so that arbitrary strings are true (for regex)
+
+2006-05-11 09:05 +0000 [r26760-26773]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* rtp.c: backport fix from trunk for bug #6934, ensuring that RTP
+	  mark bit is changed when SSRC changes
+
+	* channels/chan_sip.c: ensure that we send a response to REGISTER
+	  requests that are successfully authenticated but contain invalid
+	  Contact URIs
+
+2006-05-09 14:18 +0000 [r26050-26090]  BJ Weschke <bweschke at btwtech.com>
+
+	* channels/chan_sip.c, doc/README.variables: Add the appropriate
+	  jumping behavior that is the standard for 1.2.X to SIPGetHeader
+	  that is now deprecated in /trunk. #7111 (blitzrage!!!)
+
+	* apps/app_voicemail.c: Correct memory leak in find_user_realtime
+	  #7118 (fnordian)
+
+2006-05-08 15:09 +0000 [r25608]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Issue 7103 - mikma - The header is named
+	  "Require" - Don't reply to ACK (Not using patch against trunk)
+
+2006-05-08 14:12 +0000 [r25518-25563]  BJ Weschke <bweschke at btwtech.com>
+
+	* channels/chan_agent.c: Don't show agents as available when they
+	  are in wrap-up time. #6726 (ZX81)
+
+	* apps/app_queue.c: Make QueueStatusComplete event thread safe by
+	  wrapping it inside the queue lock clause already there. #7013
+	  (bziherl reporting)
+
+	* apps/app_queue.c: Don't recheck valid_exit() after getting the
+	  result from say_position (which already checks it). Should
+	  prevent another loop if the caller hits digits during the
+	  position announcement. #6776 (tgj reporting)
+
+2006-05-08 11:16 +0000 [r25442]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_features.c: Incorrect log statement when playing transfer
+	  sounds (issue #7008 reported and fixed by nathan)
+
+2006-05-07 13:38 +0000 [r25288-25322]  BJ Weschke <bweschke at btwtech.com>
+
+	* apps/app_meetme.c: Fix playback behavior to exit correctly when
+	  we receive a hangup during playback of the invalid pin message.
+	  #7091 (AntD reporting)
+
+	* asterisk.c: Reset the value of ast_mainpid if we fork so future
+	  remote unix connections display the correct PID. #7098 (tzafrir
+	  reporting)
+
+2006-05-06 02:32 +0000 [r25015-25165]  Russell Bryant <russell at digium.com>
+
+	* frame.c: fix a problem where the frame's data pointer is
+	  overwritten by the newly allocated data buffer before the data
+	  can be copied from it. This is in the ast_frisolate() function
+	  which is rarely used. (issue #6732, stefankroon)
+
+	* channels/chan_zap.c: ensure that the appropriate manager events
+	  are sent in all of the places where alarms are detected or
+	  cleared (issue #6866, flefoll)
+
+	* channels/chan_h323.c: update chan_h323 to reflect the new
+	  prototype for rtp_set_peer (issue #6560, casper) This was fixed a
+	  couple months ago in the trunk, but never in 1.2.
+
+2006-05-05 20:44 +0000 [r25014]  BJ Weschke <bweschke at btwtech.com>
+
+	* apps/app_voicemail.c, include/asterisk/app.h, app.c: Voicemail
+	  fixes along with an API change approved by russellb to fix the
+	  bug(s). (jcollie and supczinskib) #7064
+
+2006-05-05 17:39 +0000 [r24837-24911]  Russell Bryant <russell at digium.com>
+
+	* apps/app_while.c, apps/app_macro.c: use pbx_checkcondition()
+	  instead of ast_true() to evaluate the condition for MacroIf and
+	  WhileIf (issue #7086)
+
+2006-05-04 16:27 +0000 [r24706]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_queue.c: Bug 7023 - reload should not unpause members
+
+2006-05-04 11:17 +0000 [r24567-24669]  BJ Weschke <bweschke at btwtech.com>
+
+	* apps/app_verbose.c: Make sure that only the "|" is a recognized
+	  delimiter for Verbose(), as the app documentation already
+	  specifies. #7080 (alessiof reporting)
+
+	* apps/app_dial.c: Correct application documentation to make users
+	  aware that certain options cannot be used in conjunction with
+	  others. #6666 (chotaire)
+
+2006-05-03 18:31 +0000 [r24496]  Russell Bryant <russell at digium.com>
+
+	* redhat/asterisk.spec: fix up "make rpm" - don't reference the
+	  gzipped man page, because we don't store them compressed anymore
+	  - add some files that currently were not listed (issue #6837)
+
+2006-05-03 12:39 +0000 [r24381]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Issue #7074 - Problem with long contact
+	  lines
+
+2006-05-02 19:39 +0000 [r24295]  BJ Weschke <bweschke at btwtech.com>
+
+	* file.c: Make certain ast_stopstream() sets the channel's stream
+	  members to NULL after closing them. #7067 (jcomellas)
+
+2006-05-02 02:12 +0000 [r24019-24097]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_privacy.c: Prompt does not request '#' to end input, so
+	  the application should not require it
+
+	* apps/app_nbscat.c, apps/app_festival.c, apps/app_mp3.c,
+	  apps/app_zapras.c, asterisk.c, apps/app_externalivr.c,
+	  apps/app_ices.c, res/res_musiconhold.c,
+	  include/asterisk/options.h: Bug 6864 - drop realtime priority on
+	  ALL external processes
+
+2006-05-01 19:34 +0000 [r23985-23988]  BJ Weschke <bweschke at btwtech.com>
+
+	* apps/app_voicemail.c: Make sure that when someone 0's out while
+	  recording a msg and then chooses to DELETE the recorded file, the
+	  .txt file isn't left around by itself to cause problems later.
+	  #7061 (dimitripietro reporting, blitzrage confirmed)
+
+2006-05-01 15:12 +0000 [r23951]  Russell Bryant <russell at digium.com>
+
+	* pbx.c: add missing locking of the dialplan functions list in the
+	  "show functions" CLI command
+
+2006-05-01 10:45 +0000 [r23305-23899]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_skel.c: fix this to actually compile so people can learn
+	  from it
+
+	* cdr/cdr_sqlite.c: eliminate compiler warning
+
+	* channels/chan_iax2.c: remove a pointless comparison, since the
+	  buffer is smaller than the length being checked for
+
+	* Makefile, editline/configure, cdr/Makefile, channels/Makefile,
+	  db1-ast/Makefile: allow top-level OPTIMIZE setting to affect
+	  builds in these subdirectories too
+
+	* Makefile: let the compiler determine whether hardware or software
+	  floating point should be used (like we do in the editline
+	  subdirectory)
+
+	* Makefile, apps/Makefile: remove extraneous -m64 flag that is not
+	  needed remove old 'look' target which is no longer needed (these
+	  are coming from Debian patches <G>)
+
+	* editline/makelist: ensure that the script output is correctly
+	  generated when the system's character set does not use the
+	  English lowercase/uppercase character groups
+
+	* Makefile: do installation in subdirs as a separate target (so
+	  external modules can use the Makefile more easily) generate final
+	  messages -after- running any post-install script that may be
+	  present
+
+2006-04-28 16:40 +0000 [r23176]  Russell Bryant <russell at digium.com>
+
+	* configs/zapata.conf.sample, configs/mgcp.conf.sample,
+	  configs/sip.conf.sample: note that group assignments must be from
+	  0 to 63 (issue #7048)
+
+2006-04-27 19:11 +0000 [r22954]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_queue.c: Queue(somequeue,,,,) -> interpreted as
+	  Queue(somequeue,,,,0) (issue #7044 reported nathan fixed by
+	  jsmith - sort of)
+
+2006-04-27 16:12 +0000 [r22866]  Matt Frederickson <creslin at digium.com>
+
+	* channels/chan_zap.c: Fix buglet in channel reassignment on
+	  SETUP_ACK
+
+2006-04-26 19:18 +0000 [r22596]  Matt O'Gorman <mogorman at digium.com>
+
+	* apps/app_voicemail.c: do not allow for users to forward voicemail
+	  to themselves, patch from 7001
+
+2006-04-21 22:39 +0000 [r22112-22113]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* channel.c: Bug 7004 - release all threads waiting on a condition
+	  prior to freeing it
+
+2006-04-19 21:10 +0000 [r21638]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* contrib/scripts/safe_asterisk.8, contrib/scripts/safe_asterisk:
+	  support system-specific scripts in safe_asterisk, before starting
+	  Asterisk proper
+
+2006-04-19 18:43 +0000 [r21597]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* cdr/cdr_odbc.c: Bug 6553 - plug memory leaks when ODBC connection
+	  is down
+
+2006-04-18 23:31 +0000 [r21237]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* pbx.c: properly handle brace-wrapped strings in variable/function
+	  references in the dialplan
+
+2006-04-18 06:26 +0000 [r20966-21037]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_random.c: Bug 6984 - off by one error in Random()
+
+	* res/res_musiconhold.c: Bug 6544 - when we remove a music class,
+	  the thread servicing it should die
+
+2006-04-14 17:21 +0000 [r20034-20037]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* sounds.txt: uncomment files that actually do exist (oops)
+
+	* sounds.txt: update text to match actual prompts being distributed
+	  (thanks to Kinsey in the support department for reviewing all the
+	  prompts!)
+
+2006-04-13 20:37 +0000 [r19891]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Bug 6947 - Allow vm broadcasts to more than
+	  256 characters worth of mailboxes
+
+2006-04-13 Kevin P. Fleming <kpfleming at digium.com>
+
+	* Asterisk 1.2.7.1 released
+
+2006-04-13 17:40 +0000 [r19812]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_page.c: oops... let's not set a variable and then
+	  immediately overwrite it while assuming its old value will
+	  magically return
+
+2006-04-13 15:56 +0000 [r19768]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* pbx.c: Bug 6957 - variable names beginning with CALLERID weren't
+	  substituted correctly
+
+2006-04-12 Kevin P. Fleming <kpfleming at digium.com>
+
+	* Asterisk 1.2.7 released
+
+2006-04-11 22:39 +0000 [r19394-19397]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_dial.c: Bug 6490 - telco intercept should report
+	  NOANSWER instead of CHANUNAVAIL
+
+	* apps/app_voicemail.c: Bug 6061 - Fix ODBC storage of VM on PGSQL
+	  and MSSQL
+
+2006-04-11 21:58 +0000 [r19353]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* Makefile: don't create a 'voicemail' symlink in the sounds
+	  directory; app_voicemail has not needed it since January of 2005
+	  (issue #6613)
+
+2006-04-11 21:55 +0000 [r19351]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* asterisk.c: Bug 6097 - possible descriptor leak
+
+2006-04-11 21:50 +0000 [r19345-19348]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_page.c: don't call the originating device as part of the
+	  Page() operation (issue #6932)
+
+	* channel.c: simplify spy queue flushing logic, and always force a
+	  flush when one side gets full, even if the other side is not
+	  empty (issue #6457)
+
+	* pbx/pbx_config.c: don't destroy the entire dialplan during
+	  'reload', just atomically replace it like 'extensions reload'
+	  does (issue #6047)
+

[... 1853 lines stripped ...]


More information about the asterisk-commits mailing list