Convert all release tags to Opsound music-on-hold.

For more details:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/



git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.2.4@212958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

.cvsignore

@@ -1,16 +0,0 @@
-asterisk
-defaults.h
-ast_expr2.output
-.version
-.depend
-.applied
-mpg123-0.59r
-mpg123-0.59r.tar.gz
-update.out
-.lastclean
-.cleancount
-.tags-depend
-.tags-sources
-tags
-TAGS
-testexpr2

.lastclean

@@ -0,0 +1 @@
+8

.version

@@ -0,0 +1 @@
+1.2.4

CREDITS

@@ -85,7 +85,7 @@ Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
 	available under a Creative Commons License at http://www.asteriskdocs.org
 
 === HOLD MUSIC ===
-Music provided by www.freeplaymusic.com
+Music provided by www.opsound.org
 
 === OTHER SOURCE CODE IN ASTERISK ===
 

ChangeLog

@@ -1,3 +1,831 @@
+2006-01-30 Russell Bryant <russell@digium.com>
+
+	* Asterisk 1.2.4 Released
+
+2006-01-30 17:08 +0000 [r8905]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_zap.c: disable buggy PRI user-user code until it
+	  can be fixed
+
+2006-01-28 13:52 +0000 [r8808]  Olle Johansson <oej@edvina.net>
+
+	* channels/chan_sip.c: Issue 6182 - Don't remove scheduled event
+	  until it's really done. (reported by malverian)
+
+2006-01-27 08:02 +0000 [r8785]  Olle Johansson <oej@edvina.net>
+
+	* channels/chan_sip.c: Issue 6362 - Register without Contact: and
+	  Expires: fails (reporter: op)
+
+2006-01-27 00:52 +0000 [r8758]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* ast_expr2.h, ast_expr2f.c, ast_expr2.c: Bug 6072 - Revisions to
+	  the source bison and flex files don't auto-regenerate these files
+
+2006-01-26 19:42 +0000 [r8729]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_zap.c: fix problem with dtmf on e&m (issue #6364)
+
+2006-01-26 14:39 +0000 [r8710]  Olle Johansson <oej@edvina.net>
+
+	* channels/chan_sip.c: Issue 5898: Registrations does not get
+	  deleted if there's an active SIP dialog
+
+2006-01-25 19:14 +0000 [r8666-8677]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_sip.c: don't call ast_update_realtime with
+	  uninitialized variables if we get a registration with an expirey
+	  of 0 seconds (issue #6173)
+
+	* channels/chan_features.c: fix memory leak (inspired by issue
+	  #6351)
+
+2006-01-25 Russell Bryant <russell@digium.com>
+
+	* Asterisk 1.2.3 Released
+
+2006-01-25 09:46 +0000 [r8632]  Olle Johansson <oej@edvina.net>
+
+	* channel.c: Issue #6439 - the "timebomb" bug. Patch by Markster
+	  over GPRS
+
+2006-01-25 05:38 +0000 [r8619]  Russell Bryant <russell@digium.com>
+
+	* utils/astman.c: don't leak almost 200 bytes for each new channel
+	  (issue #6330)
+
+2006-01-25 01:50 +0000 [r8608]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* apps/app_dial.c: ensure hangup cause code is handled properly
+	  when channel does not return a frame (issue #6346)
+
+2006-01-24 22:55 +0000 [r8600]  Russell Bryant <russell@digium.com>
+
+	* asterisk.c: completely arbitrary whitespace change for testing
+	  something with svnmerge ...
+
+2006-01-24 22:32 +0000 [r8588]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channel.c: ensure that channel cannot become zombie after we
+	  check but before we try to start indications
+
+2006-01-24 20:37 +0000 [r8573]  Matt Frederickson <creslin@digium.com>
+
+	* channels/chan_zap.c: Backport fix for #6229, hangup on polarity
+	  reversal
+
+2006-01-24 19:21 +0000 [r8537-8562]  Olle Johansson <oej@edvina.net>
+
+	* channels/chan_sip.c: Issue 6114: Don't hangup on BYE/ALSO with no
+	  channel.
+
+	* channels/chan_sip.c: Issue #6308 - never send response to ACK.
+	  (Reported by whiskerp)
+
+2006-01-22 19:03 +0000 [r8437-8445]  Russell Bryant <russell@digium.com>
+
+	* apps/app_queue.c: fix memory leak from not freeing the queue
+	  member list when freeing an old queue
+
+	* channel.c: fix MixMonitor crash (issue #6321, probably others)
+
+2006-01-22 15:13 +0000 [r8433]  BJ Weschke <bweschke@btwtech.com>
+
+	* channels/chan_sip.c: Bug fix: Correct some scenarios where
+	  CALL_LIMIT could not be getting adjusted properly allowing
+	  chan_sip to send calls when it really shouldn't. Bug #6111
+
+2006-01-22 08:52 +0000 [r8429]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* channels/chan_sip.c: Bug 6281 - Cannot set more than a single
+	  header with SIPAddHeader
+
+2006-01-22 02:05 +0000 [r8412-8418]  Russell Bryant <russell@digium.com>
+
+	* pbx.c: add a modified fix to prevent writing outside of the
+	  provided workspace when calculating a substring (issue #6271)
+
+	* pbx.c: temporarily revert substring fix pending the result of the
+	  discussion in issue #6271
+
+	* pbx.c: prevent the possibility of writing outside of the
+	  available workspace (issue #6271)
+
+2006-01-21 18:29 +0000 [r8394]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_queue.c: Bug 5936 - AddQueueMember fails on realtime
+	  queue, if queue not yet loaded
+
+2006-01-20 18:34 +0000 [r8347]  Russell Bryant <russell@digium.com>
+
+	* apps/app_queue.c: fix invalid value of prev_q (issue #6302)
+
+2006-01-20 01:00 +0000 [r8320]  Matt O'Gorman <mogorman@digium.com>
+
+	* channels/chan_iax2.c: solved problem with delayreject and iax
+	  trunking bug 4291
+
+2006-01-19 19:40 +0000 [r8281]  Olle Johansson <oej@edvina.net>
+
+	* channels/chan_sip.c: Enable "musicclass" setting for sip peers as
+	  per the config sample.
+
+2006-01-19 19:14 +0000 [r8276]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* ast_expr2.y, ast_expr2.fl: Bug 6072 - Memory leaks in the
+	  expression parser
+
+2006-01-19 04:56 +0000 [r8232-8242]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_sip.c: fix Message-Account header to use the ip
+	  address if the fromdomain isn't set (issue #6278)
+
+	* apps/app_milliwatt.c: fix a seg fault due to assuming that space
+	  gets allocatted on the stack in the same order that we declare
+	  the variables (issue #6290)
+
+2006-01-18 21:02 +0000 [r8194]  Matt O'Gorman <mogorman@digium.com>
+
+	* apps/app_meetme.c: Solves issue with the login proccess in meetme
+	  patch from 6136
+
+2006-01-18 02:49 +0000 [r8173]  Russell Bryant <russell@digium.com>
+
+	* ChangeLog (removed): remove ChangeLog from the 1.2 branch. It
+	  will only be present in the tags.
+
+2006-01-18 Russell Bryant <russell@digium.com>
+
+	* Asterisk 1.2.2 Released
+
+2006-01-18 00:47 +0000 [r8140-8162]  Matt O'Gorman <mogorman@digium.com>
+
+	* loader.c: Changed order of autoload so that pbx_ comes before
+	  channels, and in doing so cause bug 6002 to not be an issue
+
+	* apps/app_festival.c: Stop any generators running on a channel
+	  when festival is called as described in 5996
+
+2006-01-17 18:29 +0000 [r8134]  Matt Frederickson <creslin@digium.com>
+
+	* res/res_features.c: Backport of fix for #6094
+
+2006-01-17 16:55 +0000 [r8124]  Matt O'Gorman <mogorman@digium.com>
+
+	* logger.c: Fixed code ordering of logger_init and queue_log_init
+	  bug 6263
+
+2006-01-17 13:11 +0000 [r8112-8122]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* asterisk.c: update CLI copyright notice
+
+	* asterisk.c: do rlimit check _after_ reading config file, in case
+	  'dumpcore' is specified there
+
+2006-01-14 19:06 +0000 [r8074]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* funcs/func_strings.c: Bug 6238 - Fix segfault when delimiter not
+	  specified
+
+2006-01-13 06:07 +0000 [r8047]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_agent.c: fix spelling errors (issue #6227)
+
+2006-01-12 06:14 +0000 [r7999]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c, configs/voicemail.conf.sample: Bug 6211 -
+	  Add option deletevoicemail as equivalent to option delete for
+	  Realtime
+
+2006-01-11 19:08 +0000 [r7965-7986]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_agent.c: move variable to correct scope (issue
+	  #6197)
+
+	* apps/app_voicemail.c: fix temp greetings with ODBC storage (issue
+	  #6078)
+
+	* channels/chan_sip.c: fix mem leak on module unload (issue #6190)
+
+	* app.c: don't override an error condition that occurred when
+	  acting on the primary channel when stopping the autoservice on
+	  the peer channel. (from issue #6087)
+
+	* translate.c: lock list of translators *before* recalculating the
+	  translation matrix
+
+2006-01-11 04:38 +0000 [r7963]  Matt O'Gorman <mogorman@digium.com>
+
+	* channel.c: Minor typo refrenced in 6191
+
+2006-01-11 04:19 +0000 [r7957-7960]  Russell Bryant <russell@digium.com>
+
+	* pbx.c: fix locking error - lock instead of unlock
+
+	* apps/app_dial.c: fix a little typo
+
+2006-01-11 01:30 +0000 [r7955]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Bug 6192 - behave correctly when mailbox is
+	  specified as argument
+
+2006-01-10 08:48 +0000 [r7939]  Olle Johansson <oej@edvina.net>
+
+	* doc/README.cdr: - Adding reference to README.tds - Reformatting
+	  table
+
+2006-01-09 22:48 +0000 [r7917]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_iax2.c: re-initialize _all_ sequence numbers when
+	  transfer completes
+
+2006-01-09 22:07 +0000 [r7915]  Russell Bryant <russell@digium.com>
+
+	* file.c: add missing unlock (issue #6112)
+
+2006-01-09 20:08 +0000 [r7904-7908]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* pbx/pbx_spool.c: Bug 6157 - Memory leak
+
+	* doc/README.variables: Update variable documentation to match the
+	  code
+
+2006-01-09 18:11 +0000 [r7898-7900]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* asterisk.c: commit user/group-related changes from trunk
+
+	* db.c: backport fix from revision 7856 of trunk
+
+	* apps/app_voicemail.c: fix breakage introduced in revision 7871
+
+2006-01-09 05:11 +0000 [r7870-7871]  Russell Bryant <russell@digium.com>
+
+	* apps/app_voicemail.c: fix seg fault when using greek syntax in
+	  VoicemMailMain (issue #6142)
+
+	* manager.c: backport fix for unnecessary unlock (issue #6171)
+
+2006-01-07 07:27 +0000 [r7848]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* pbx/pbx_spool.c: Bug 6156 - catch all threading errors, not just
+	  simple failure
+
+2006-01-06 00:34 +0000 [r7831]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* pbx/pbx_config.c: Dumb error messages - "Context 'context'
+	  already included in 'in' context"
+
+2006-01-06 00:21 +0000 [r7829]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_agent.c: update agent persistence when an agent
+	  gets logged off by autologoff
+
+2006-01-05 23:53 +0000 [r7827]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* include/asterisk/strings.h: Bug 6076 - Fix documentation of
+	  ast_trim_blank return value
+
+2006-01-05 23:49 +0000 [r7825]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channel.c: eliminate rounding errors that caused call time limits
+	  to be inaccurate (issue #5913) round 'time left' reported during
+	  call limit warnings up to sound more accurate
+
+2006-01-05 23:07 +0000 [r7823]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* res/res_features.c: Bug 6081 - fix for memory leak, formatting
+	  fixes
+
+2006-01-05 20:52 +0000 [r7819]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that
+	  variable is initialized
+
+2006-01-05 09:13 +0000 [r7812]  Olle Johansson <oej@edvina.net>
+
+	* res/res_features.c: Fix copyright of changed file
+
+2006-01-05 00:58 +0000 [r7799-7809]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_agent.c: send device state updates for auto-logoff
+	  of agents as well
+
+	* formats/format_pcm.c, formats/format_pcm_alaw.c: doh... fseek()
+	  has no useful return value
+
+	* formats/format_pcm.c, formats/format_pcm_alaw.c: use proper
+	  fwrite() parameters and return value
+
+	* formats/format_pcm.c, formats/format_pcm_alaw.c: return properly
+	  after extending file
+
+	* formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that
+	  ulaw/alaw sound files are filled with silence when extended (not
+	  zeroes)
+
+	* channel.c: make monitoring more tolerant of peers that deliver
+	  frames in bursts
+
+2006-01-04 21:46 +0000 [r7792-7795]  Olle Johansson <oej@edvina.net>
+
+	* res/res_features.c: Issue #5980: Removing extra CR+LF in manager
+	  events - needs port to trunk
+
+	* channels/chan_sip.c: Fixing typo in XML for video updates.
+
+2006-01-04 07:06 +0000 [r7773]  Russell Bryant <russell@digium.com>
+
+	* funcs/func_moh.c: use a more correct way of determining the size
+	  of the destination buffer
+
+2006-01-04 05:27 +0000 [r7771]  BJ Weschke <bweschke@btwtech.com>
+
+	* apps/app_privacy.c: Fix the 'if' clause to be true under the
+	  right conditions. Bug #6126
+
+2006-01-03 20:22 +0000 [r7746]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* ast_expr.y (removed): remove unused 'old' expression parser
+
+2006-01-03 18:15 +0000 [r7743]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_stack.c: Bug 6121 - typo in application description
+
+2006-01-03 17:24 +0000 [r7736-7740]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* include/asterisk/chanspy.h, apps/app_chanspy.c,
+	  apps/app_mixmonitor.c, channel.c: revert incorrect fix for bug
+	  #6048 from revision 7709 put in correct (simpler) fix add doxygen
+	  docs for channel spy 'state' values
+
+	* channels/chan_sip.c: backport rport scanning fix from trunk (bug
+	  #6071)
+
+	* ast_expr2f.c, ast_expr2.fl: don't leak memory for (most)
+	  expression evaluations
+
+2006-01-02 07:31 +0000 [r7709]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_mixmonitor.c: Bug 6084 - MixMonitor after a 'cli stop
+	  monitor' deadlocks
+
+2006-01-02 02:04 +0000 [r7706]  BJ Weschke <bweschke@btwtech.com>
+
+	* channels/chan_sip.c, channels/chan_iax2.c: Fix compiler warnings.
+
+2005-12-30 14:54 +0000 [r7677]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* channel.c: Bug 6091 - Fix race condition around uniqueid
+
+2005-12-28 17:35 +0000 [r7663-7665]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_sip.c: fix memory leak in build_rpid (issue #6070)
+
+	* apps/app_chanspy.c: backport fix for permissions of created
+	  recordings (issue #6067)
+
+2005-12-27 00:07 +0000 [r7641]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_zap.c: backport fix to ensure that DSP is never
+	  enabled on pseudo channels
+
+2005-12-26 20:32 +0000 [r7637]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* cdr/cdr_tds.c: Remove copy of code in libc, preferring code in
+	  utils.c (public domain code)
+
+2005-12-26 18:19 +0000 [r7634]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_sip.c, channels/chan_agent.c, apps/app_sms.c,
+	  asterisk.c, config.c, pbx/pbx_dundi.c, apps/app_externalivr.c,
+	  apps/app_queue.c, channels/chan_iax2.c, cli.c,
+	  apps/app_chanspy.c, res/res_monitor.c: cast time_t to an int in
+	  printf/scanf (issue #5635)
+
+2005-12-23 06:38 +0000 [r7608]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_hasnewvoicemail.c: Bug 6051 - VMCOUNT should work as
+	  documented and count all, not quit after finding 1
+
+2005-12-23 03:01 +0000 [r7606]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* asterisk.c: add license reference to copyright notice displayed
+	  when CLI session begins add 'show warranty' and 'show license'
+	  CLI commands (still need a complete list of non-GPL components
+	  included in Asterisk)
+
+2005-12-23 00:00 +0000 [r7605]  BJ Weschke <bweschke@btwtech.com>
+
+	* apps/app_waitforsilence.c: Another app documentation tweak.
+
+2005-12-22 22:04 +0000 [r7601]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Bug 6050 SQL requires the use of single
+	  ticks to delimit values, not quotes
+
+2005-12-22 20:36 +0000 [r7595-7599]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_sip.c, configs/sip.conf.sample: revert changes to
+	  videosupport to allow per-peer setting, since it isn't quite
+	  complete and there is not an obvious fix at this point
+
+	* channels/chan_sip.c: remove stray unlock (issue #5955)
+
+2005-12-21 22:23 +0000 [r7586]  Josh Roberson <josh@asteriasgi.com>
+
+	* channels/chan_sip.c: Actually put in the per-peer settings for
+	  sip video, as they didn't make it in at astricon somehow, and
+	  I've been too busy up until now to redo it.
+
+2005-12-21 20:01 +0000 [r7582]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* channels/chan_alsa.c: Allow a chan_alsa that failed to open sound
+	  devices to be unloaded.
+
+2005-12-21 19:53 +0000 [r7580]  BJ Weschke <bweschke@btwtech.com>
+
+	* channels/chan_agent.c: Bug #6040 - Documentation correction
+
+2005-12-21 19:23 +0000 [r7577]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* pbx/pbx_ael.c: Bug 5777 - Remove parentheses on Goto in AEL, so
+	  that it parses correctly
+
+2005-12-20 20:21 +0000 [r7550-7557]  Russell Bryant <russell@digium.com>
+
+	* res/res_agi.c: check array bounds when parsing arguments to AGI
+	  (issue #5868)
+
+	* channels/chan_iax2.c: backport fix for reloading peer context
+	  (issue #6007)
+
+	* apps/app_directed_pickup.c: backport fix for segfault on directed
+	  pickup when no CDR is available (issue #5998)
+
+2005-12-20 12:58 +0000 [r7546]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* apps/app_meetme.c: backport fix for larger-than-20ms-frames from
+	  trunk (bug #5697)
+
+2005-12-19 23:47 +0000 [r7529]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_sip.c: I messed up and accidently committed this to
+	  the trunk first ... - add note on required values of sip_methods
+	  struct - remove duplicate function prototype - remove duplicate
+	  ast_mutex_lock (issue #6025)
+
+2005-12-19 19:06 +0000 [r7521-7523]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* file.c: Bug 5988 - record append option not working
+
+	* cdr.c: Bug 6026 - segfault for the sequence NoCDR(),
+	  SetAMAFlags()
+
+2005-12-17 18:55 +0000 [r7517-7519]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* doc/README.ael: Document that curley braces must be on the same
+	  line as the keyword.
+
+	* apps/app_chanspy.c: Bug 6009 - off by one error
+
+2005-12-17 03:59 +0000 [r7510-7515]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_sip.c: Max-Forwards headers must only be present on
+	  requests, not responses
+
+	* channels/chan_sip.c: forcibly expire previous subscriptions from
+	  a peer when they resubscribe (keeps them from building up and
+	  waiting for expiration, and stops us sending unwanted NOTIFY
+	  messages to devices)
+
+	* build_tools/make_svn_branch_name: fix some buglet when building
+	  team branch version strings
+
+2005-12-17 01:02 +0000 [r7508]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* include/asterisk/linkedlists.h: We want to check the previous
+	  value, not the current value (which was just changed).
+
+2005-12-16 00:49 +0000 [r7497]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_cut.c: First field is truncated
+
+2005-12-15 10:52 +0000 [r7490]  Christian Richter <christian.richter@beronet.com>
+
+	* doc/README.misdn, channels/misdn/mISDNuser.patch (added),
+	  channels/misdn/isdn_lib_intern.h, channels/misdn/mISDN.patch
+	  (added), channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+	  channels/misdn/Makefile, channels/misdn/chan_misdn_config.h,
+	  channels/misdn/ie.c, channels/chan_misdn_config.c,
+	  channels/misdn/isdn_msg_parser.c, channels/Makefile,
+	  channels/misdn/isdn_lib.c: * Added mISDN/mISDNuser Echo cancel
+	  Patch * Fixed Makefiles so that chan_misdn can be compiled again
+	  * added some hints, that mISDN cannot be compiled against gcc-4,
+	  SMP, Spinlock Debug * fixed some Minor issues in chan_misdn,
+	  regarding Type Of Number and Presentation
+
+2005-12-15 02:51 +0000 [r7482]  BJ Weschke <bweschke@btwtech.com>
+
+	* channel.c: Bug #6003 - Don't free the channel structure until
+	  after having sent the manager event.
+
+2005-12-13 18:54 +0000 [r7435-7470]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* doc/README.variables: clarify substring documentation
+
+	* utils.c: correct broken math in tvfix() for timestamp values over
+	  one million
+
+	* apps/app_dial.c: restore ability of caller to hangup calls that
+	  are still ringing (issue #5839)
+
+	* channels/chan_sip.c, pbx.c: ensure that hangups while incoming
+	  calls are in early state are handled properly (issue #5919)
+
+	* channels/chan_agent.c: only report AGENT_IDLE for callback mode
+	  agents when they are actually idle (issue #5902)
+
+	* app.c: use the stream's current point when pausing/unpausing,
+	  instead of elapsed time (which doesn't work when the stream has
+	  been skipped forward or backward) (issue #5897)
+
+	* apps/app_externalivr.c: set all the child file descriptors to
+	  non-blocking so that we don't hang if the child fails to send a
+	  newline-terminated command or error message
+
+2005-12-12 17:19 +0000 [r7433]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* include/asterisk/linkedlists.h: Typo
+
+2005-12-11 06:08 +0000 [r7430]  Russell Bryant <russell@digium.com>
+
+	* utils/astman.c: silence a couple of compiler warnings about
+	  pointer signedness
+
+2005-12-11 01:26 +0000 [r7427-7429]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* include/asterisk/linkedlists.h: Bug 5965 - major bug in
+	  AST_LIST_REMOVE
+
+	* apps/app_voicemail.c: Bug 5967
+
+2005-12-10 18:10 +0000 [r7425]  BJ Weschke <bweschke@btwtech.com>
+
+	* channels/chan_zap.c: Bug #5877 Make sure the digit string from
+	  E&M wink DNIS collection is properly null terminated as it grows.
+
+2005-12-08 23:45 +0000 [r7404-7406]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Bug 5960
+
+	* configs/res_odbc.conf.sample: Documenting two keywords that were
+	  previously missing
+
+2005-12-08 01:05 +0000 [r7382-7386]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* pbx.c: initialize the buffer before using it...
+
+	* pbx.c: ensure that hints are allowed to use global variable
+	  references
+
+2005-12-06 Kevin P. Fleming <kpfleming@digium.com>
+
+	* Asterisk 1.2.1 Released
+
+2005-12-05 06:47 +0000 [r7335-7340]  Russell Bryant <russell@digium.com>
+
+	* Makefile: remove ASTERISKVERSIONNUM from the version string given
+	  to doxygen
+
+	* apps/app_queue.c: don't delete dynamic queue members when
+	  reloading the static members from a realtime database (issue
+	  #5922)
+
+	* channels/chan_sip.c: fix the order of arguments to an error
+	  message (issue #5927)
+
+2005-12-04 18:03 +0000 [r7329]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* build_tools/make_svn_branch_name: use a more efficient way to get
+	  the revision number, that will also report if the working copy
+	  contains uncommitted modifications
+
+2005-12-03 19:55 +0000 [r7310]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Bug 5925: check for "Unknown", as that's
+	  what app_voicemail puts into the field for Unknown callerid Also,
+	  remove useless res checks (initialized to 0; never set)
+
+2005-12-03 01:24 +0000 [r7299]  Olle Johansson <oej@edvina.net>
+
+	* configs/sip.conf.sample: Documenting the default registerattempts
+	  setting as 0, continue hammering the server for ever and ever ;-)
+
+2005-12-02 21:12 +0000 [r7285]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* contrib/init.d/rc.debian.asterisk,
+	  contrib/init.d/rc.mandrake.asterisk,
+	  contrib/init.d/rc.redhat.asterisk,
+	  contrib/init.d/rc.gentoo.asterisk,
+	  contrib/init.d/rc.mandrake.zaptel,
+	  contrib/init.d/rc.slackware.asterisk: Turn on executable bits for
+	  startup scripts, and fix bash var interpolation for Mandrake
+
+2005-12-02 00:52 +0000 [r7275]  Olle Johansson <oej@edvina.net>
+
+	* channels/chan_sip.c: Bug #5907. Improve SIP INFO DTMF debugging
+	  output. (1.2 & Trunk)
+
+2005-12-02 00:51 +0000 [r7266-7274]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* apps/app_page.c, pbx.c: inherit channel variables into channels
+	  created by Page() application (issue #5888)
+
+	* apps/app_voicemail.c, configs/voicemail.conf.sample, UPGRADE.txt:
+	  allow previous context-searching behavior to be used if desired
+	  (issue #5899)
+
+	* apps/app_voicemail.c: properly handle password changes when
+	  mailbox is last line of config file and not followed by a newline
+	  (issue #5870) reformat password changing code to conform to
+	  coding guidelines (issue #5870)
+
+	* channels/chan_agent.c: protect agent_bridgedchannel() from
+	  segfaulting when there is no bridged channel (issue #5879)
+
+	* channels/chan_local.c: allow variables to exist on both 'halves'
+	  of the Local channel (issue #5810)
+
+	* apps/app_festival.c: don't block waiting for the Festival server
+	  forever when it goes away (issue #5882)
+
+	* channel.c: ensure channel's scheduling context is freed (issue
+	  #5788)
+
+	* Makefile, patches (removed): Makefile 'update' target now
+	  supports updating from Subversion repositories (issue #5875)
+	  remove support for 'patches' subdirectory, it's no longer useful
+
+2005-12-01 23:18 +0000 [r7261-7265]  Olle Johansson <oej@edvina.net>
+
+	* doc/README.misdn: Changing bug report address to the Asterisk
+	  issue tracker
+
+	* doc/README.jitterbuffer, doc/README.realtime: Removing references
+	  to 1.1dev, replacing with 1.2, in documentation files.
+
+	* doc/README.misdn: Fixing some spelling errors, as well as
+	  changing "cvs" to "subversion" in misdn documentation.
+
+2005-12-01 19:25 +0000 [r7257]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* build_tools/make_svn_branch_name: ensure that 'svn info' output
+	  is in the expected language for the script to parse (issue #5880)
+
+2005-12-01 02:33 +0000 [r7228-7251]  Russell Bryant <russell@digium.com>
+
+	* apps/app_externalivr.c: use ast_app_separate_args to split
+	  arguments (issue #5686)
+
+	* apps/app_queue.c: fix queue weight feature - compare member
+	  interfaces instead of pointers to the members, since each queue
+	  has its own list of members. (issue #5863)
+
+	* build_tools/make_svn_branch_name: use '=' instead of '==' for
+	  string comparisons. /bin/bash is ok with this, but /bin/sh is
+	  not. (issue #5885)
+
+	* redhat/asterisk (removed), Makefile: remove outdated redhat init
+	  script and provide the updated one in 'make rpm' (issue #5786)
+
+	* contrib/init.d/rc.debian.asterisk,
+	  contrib/init.d/rc.redhat.asterisk: Comment out LD_ASSUME_KERNEL
+	  by default. Print error messages if the asterisk executable or
+	  the asterisk configuration directory are not found. (issue #5785,
+	  #5708)
+
+	* apps/app_dial.c: fix DIALEDTIME when call has not been answered
+	  (issue #5862)
+
+	* rtp.c: do not allow an rtp message with zero type (issue #5749)
+
+	* pbx.c: fix hint case sensitivity (issue #5856)
+
+	* configs/sip.conf.sample: add description of the "fromdomain"
+	  option (issue #5874)
+
+2005-11-30 03:52 +0000 [r7227]  Josh Roberson <josh@asteriasgi.com>
+
+	* apps/app_voicemail.c, UPGRADE.txt, ChangeLog: backport fix from
+	  trunk
+
+2005-11-30 03:37 +0000 [r7219-7226]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* doc/cdr.txt, doc/CODING-GUIDELINES, include/asterisk.h,
+	  doc/README.mp3: remove remaining CVS references
+
+	* channel.c: port memory leak fix from rev 7223 in trunk
+
+	* include/asterisk/lock.h: do the multiple-lock check for cond_wait
+	  properly...
+
+2005-11-29 06:12 +0000 [r7216-7218]  Russell Bryant <russell@digium.com>
+
+	* apps/app_cut.c: print an error message if invalid arguments are
+	  specified
+
+	* apps/app_skel.c: fix a couple of typos and a buglet
+
+2005-11-29 01:25 +0000 [r7199-7213]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* include/asterisk/lock.h: if the lock protected a pthread_cond is
+	  held recursively, warn before waiting onthe condition
+
+	* Makefile, build_tools/make_svn_branch_name (added): port version
+	  string computation from trunk
+
+	* / (added): branch renames remove unneeded branches
+
+2005-11-29  Josh Roberson <josh@asteriasgi.com>
+	
+	* apps/app_voicemail.c: Only look in 'default' context when no context defined to VoiceMailMain(). (issue #5887)
+
+2005-11-25  Russell Bryant  <russell@digium.com>
+
+	* apps/app_dial.c: Properly duplicate the string for ANI (issue #5850)
+
+2005-11-23  Russell Bryant  <russell@digium.com>
+
+	* configs/voicemail.conf.sample: Add note to indicate that #include should not be used for this file. (issue #5828)
+
+	* indications.c: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
+	* configs/indications.conf.sample: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
+	* include/asterisk/indications.h: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
+	* res/res_indications.c: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
+
+	* apps/app_voicemail.c: Remove left over "yay!" debugging message. (issue #5829)
+
+2005-11-21  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* apps/app_cut.c: remove unnecessary include that causes spurious rebuilding
+
+	* channels/chan_sip.c (build_peer): ensure that case changes made to peer names are not ignored during reload operations
+ 	(build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified
+
+	* channels/chan_iax2.c (build_peer and build_user): ensure that case changes made to peer/user names are not ignored during reload operations
+ 	(build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified
+
+2005-11-21  Russell Bryant  <russell@digium.com>
+
+	* Makefile: Revert previous change for Darwin.
+
+	* apps/app_osplookup.c: Properly populate the number of results. (issue #5789)
+
+	* Makefile: Don't hard-code that poll functionality needs to be provided on Darwin.
+	* apps/Makefile: Fix incorrect portion of the patch to fix 'make install' on Solaris.
+
+	* channels/chan_iax2.c (iax2_getpeername): Return non-zero to indicate that a peer was found when using realtime (issue #5815)
+
+2005-11-20  Russell Bryant  <russell@digium.com>
+
+	* Makefile apps/Makefile: Fix 'make install' for Solaris. (issue #5775)
+
+	* apps/app_record.c: Don't leak a frame if writing it to the file fails. (issue #5787)
+
+	* Makefile: Create the monitor spool directory when the other spool directories are created.
+
+	* channels/chan_sip.c channels/chan_iax2.c: Change warning messages about the number of scheduled events happening all at once to debug messages. (issue #5794)
+
+	* pbx/pbx_spool.c: Fix crash when a value is not specified with a variable on a Set: line in a call file. (issue #5806)
+
+	* apps/app_meetme.c: Fix the 'X' option to the MeetMe application. (issue #5773)
+
+	* apps/app_voicemail.c: Correct the use of a mailbox entered by the calling party instead of indicated as an argument to the Voicemail application. (issue #5774)
+
+	* apps/app_controlplayback.c: Fix logic in checking for success when jumping to priority n+101.
+	* apps/app_md5.c: Fix logic in checking for success when jumping to priority n+101.
+
+	* apps/app_hasnewvoicemail.c: Fix a typo in the application description.  Also, fix the logic in checking for success when jumping to priority n+101. (issue #5795) 
+
+	* UPGRADE.txt: Add a note on a second way that the IAX2 channel naming convention has changed. (issue #5792)
+	* channels/chan_iax2.c: Fix alignment of the output for the "iax2 show peer <peer>" CLI command (issue #5792)
+
+	* channels/Makefile: Re-add chan_oss to the default build. (issue #5799)
+
+	* res/res_musiconhold.c: Fix incorrect argument for the buffer size to an ast_copy_string call (issue #5803)
+
+	* funcs/func_enum.c: Shorten the module description (issue #5791)
+
+2005-11-17  Russell Bryant  <russell@digium.com>
+
+	* Makefile: Fix the output of Makefile generated variables to doxygen
+
+	* channels/chan_sip.c: Add missing carriage return and line feed to the SDP line indicating that we don't support VAD (issue #5780)
+
+2005-11-16  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* Asterisk 1.2.0 released.
+	
 2005-11-16  Jeremy McNamara <jj@nufone.net>
 
 	* apps/app_voicemail.c (load_config): do not terminate asterisk if no voicemail config file
@@ -10,7 +838,8 @@
 2005-11-16  Kevin P. Fleming  <kpfleming@digium.com>
 
 	* channels/Makefile: don't build chan_modem and sub-modules by default
-
+	* configs/modules.conf.sample: explicitly 'noload' chan_modem.so and submodules, in case old versions exist
+	
 	* res/Makefile: issue mpg123 not-installed warning at 'make install' time, not 'make'
 
 	* apps/app_forkcdr.c (forkcdr_exec): issue warning (and don't segfault) if ForkCDR is called on a channel that doesn't have a CDR (issue #5763)

Makefile

@@ -305,22 +305,26 @@ endif # WITHOUT_ZAPTEL
 LIBEDIT=editline/libedit.a
 
 ifneq ($(wildcard .version),)
-  ASTERISKVERSION=$(shell cat .version)
-  ASTERISKVERSIONNUM=$(shell awk -F. '{printf "%02d%02d%02d", $$1, $$2, $$3}' .version)
-  RPMVERSION=$(shell sed 's/[-\/:]/_/g' .version)
+  ASTERISKVERSION:=$(shell cat .version)
+  ASTERISKVERSIONNUM:=$(shell awk -F. '{printf "%02d%02d%02d", $$1, $$2, $$3}' .version)
+  RPMVERSION:=$(shell sed 's/[-\/:]/_/g' .version)
 else
   RPMVERSION=unknown
 endif
 
-ifneq ($(wildcard CVS),)
+# CVS mirrors of SVN have .svnrevision files showing
+# which SVN revision they are based on, and .svnbranch
+# showing the branch they are made from
+ifneq ($(wildcard .svnrevision),)
   ASTERISKVERSIONNUM=999999
-  ifneq ($(wildcard CVS/Tag),)
-    ASTERISKVERSION=$(shell echo "CVS-`sed 's/^T//g' CVS/Tag`-`date +"%D-%T"`")
-  else
-    ASTERISKVERSION=CVS HEAD
-  endif
+  ASTERISKVERSION:=SVN-$(shell cat .svnbranch)-r$(shell cat .svnrevision)
 else
-  ASTERISKVERSIONNUM=000000
+  ifneq ($(wildcard .svn),)
+    ASTERISKVERSIONNUM=999999
+    ASTERISKVERSION=SVN-$(shell build_tools/make_svn_branch_name)
+  else
+    ASTERISKVERSIONNUM=000000
+  endif
 endif
 
 ASTCFLAGS+= $(DEBUG_THREADS)
@@ -363,7 +367,7 @@ ifeq ($(OSARCH),Darwin)
   AUDIO_LIBS=-framework CoreAudio
   ASTLINK=-Wl,-dynamic
   SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
-  OBJS+=poll.o
+  OBJS+= poll.o
   ASTCFLAGS+=-DPOLLCOMPAT
 else
 #These are used for all but Darwin
@@ -534,7 +538,7 @@ clean:
 	$(MAKE) -C stdtime clean
 
 datafiles: all
-	if test $$(id -u) = 0; then sh mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
+	if [ x`whoami` = xroot ]; then sh mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
 	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/digits
 	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/priv-callerintros
 	for x in sounds/digits/*.gsm; do \
@@ -588,17 +592,16 @@ datafiles: all
 	mkdir -p $(DESTDIR)$(AGI_DIR)
 
 update: 
-	@if [ -d CVS ]; then \
-		if [ -f patches/.applied ]; then \
-			patches=`cat patches/.applied`; \
-		fi; \
-		if [ ! -z "$$patches" ]; then \
-			for x in $$patches; do \
-				echo "Unapplying $$x..."; \
-				patch -R -p0 < patches/$$x; \
-			done; \
-			rm -f patches/.applied; \
+	@if [ -d .svn ]; then \
+		echo "Updating from Subversion..." ; \
+		svn update | tee update.out; \
+		rm -f .version; \
+		if [ `grep -c ^C update.out` -gt 0 ]; then \
+			echo ; echo "The following files have conflicts:" ; \
+			grep ^C update.out | cut -b4- ; \
 		fi ; \
+		rm -f update.out; \
+	elif [ -d CVS ]; then \
 		echo "Updating from CVS..." ; \
 		cvs -q -z3 update -Pd | tee update.out; \
 		rm -f .version; \
@@ -607,19 +610,8 @@ update:
 			grep ^C update.out | cut -d' ' -f2- ; \
 		fi ; \
 		rm -f update.out; \
-		if [ ! -z "$$patches" ]; then \
-			for x in $$patches; do \
-				if [ -f patches/$$x ]; then \
-					echo "Applying patch $$x..."; \
-					patch -p0 < patches/$$x; \
-					echo $$x >> patches/.applied; \
-				else \
-					echo "Patch $$x no longer relevant"; \
-				fi; \
-			done; \
-		fi; \
 	else \
-		echo "Not CVS";  \
+		echo "Not under version control";  \
 	fi
 
 NEWHEADERS=$(notdir $(wildcard include/asterisk/*.h))
@@ -636,6 +628,7 @@ bininstall: all
 	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/system
 	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/tmp
 	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/meetme
+	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/monitor
 	if [ -f asterisk ]; then $(INSTALL) -m 755 asterisk $(DESTDIR)$(ASTSBINDIR)/; fi
 	if [ -f cygwin/asterisk.exe ]; then $(INSTALL) -m 755 cygwin/asterisk.exe $(DESTDIR)$(ASTSBINDIR)/; fi
 	if [ -f asterisk.dll ]; then $(INSTALL) -m 755 asterisk.dll $(DESTDIR)$(ASTSBINDIR)/; fi
@@ -831,12 +824,12 @@ __rpm: include/asterisk/version.h spec
 	$(MAKE) DESTDIR=/tmp/asterisk install ; \
 	$(MAKE) DESTDIR=/tmp/asterisk samples ; \
 	mkdir -p /tmp/asterisk/etc/rc.d/init.d ; \
-	cp -f redhat/asterisk /tmp/asterisk/etc/rc.d/init.d/ ; \
+	cp -f contrib/init.d/rc.redhat.asterisk /tmp/asterisk/etc/rc.d/init.d/asterisk ; \
 	rpmbuild --rcfile /usr/lib/rpm/rpmrc:redhat/rpmrc -bb asterisk.spec
 
 progdocs:
-	(cat contrib/asterisk-ng-doxygen; echo "HAVE_DOT=$(HAVEDOT) \
-	PROJECT_NUMBER=$(ASTERISKVERSION)  -  $(ASTERISKVERSIONNUM)") | doxygen - 
+	(cat contrib/asterisk-ng-doxygen; echo "HAVE_DOT=$(HAVEDOT)"; \
+	echo "PROJECT_NUMBER=$(ASTERISKVERSION)") | doxygen - 
 
 mpg123:
 	@wget -V >/dev/null || (echo "You need wget" ; false )
@@ -906,47 +899,3 @@ cleantest:
 	if cmp -s .cleancount .lastclean ; then echo ; else \
 		$(MAKE) clean; cp -f .cleancount .lastclean;\
 	fi
-
-patchlist:
-	@echo "Experimental Patches:"
-	@for x in patches/*; do \
-		patch=`basename $$x`; \
-		if [ "$$patch" = "CVS" ]; then \
-			continue; \
-		fi; \
-		if grep -q ^$$patch$$ patches/.applied; then \
-			echo "$$patch (applied)"; \
-		else \
-			echo "$$patch (available)"; \
-		fi; \
-	done
-
-apply: 
-	@if [ -z "$(PATCH)" ]; then \
-		echo "Usage: make PATCH=<patchname> apply"; \
-	elif grep -q ^$(PATCH)$$ patches/.applied 2>/dev/null; then \
-		echo "Patch $(PATCH) is already applied"; \
-	elif [ -f "patches/$(PATCH)" ]; then \
-		echo "Applying patch $(PATCH)"; \
-		patch -p0 < patches/$(PATCH); \
-		echo "$(PATCH)" >> patches/.applied; \
-	else \
-		echo "No such patch $(PATCH) in patches directory"; \
-	fi
-
-unapply: 
-	@if [ -z "$(PATCH)" ]; then \
-		echo "Usage: make PATCH=<patchname> unapply"; \
-	elif grep -v -q ^$(PATCH)$$ patches/.applied 2>/dev/null; then \
-		echo "Patch $(PATCH) is not applied"; \
-	elif [ -f "patches/$(PATCH)" ]; then \
-		echo "Un-applying patch $(PATCH)"; \
-		patch -p0 -R < patches/$(PATCH); \
-		rm -f patches/.tmpapplied || :; \
-		mv patches/.applied patches/.tmpapplied; \
-		cat patches/.tmpapplied | grep -v ^$(PATCH)$$ > patches/.applied; \
-		rm -f patches/.tmpapplied; \
-	else \
-		echo "No such patch $(PATCH) in patches directory"; \
-	fi
-

README.fpm

@@ -1,8 +0,0 @@
-About Hold Music
-================
-Digium has licensed the music included with
-the Asterisk distribution From FreePlayMusic
-for use and distribution with Asterisk.  It
-is licensed ONLY for use as hold music within
-an Asterisk based PBX.
-

README.opsound

@@ -0,0 +1,22 @@
+About Hold Music
+================
+These files were obtained from http://opsound.org, where the authors placed them
+under the Creative Commons Attribution-Share Alike 2.5 license, a copy of which
+may be found at http://creativecommons.org.
+
+Credits
+================
+macroform-cold_day - Paul Shuler (Macroform)
+paulshuler@gmail.com - http://macroform.bandcamp.com/
+
+macroform-robot_dity - Paul Shuler (Macroform)
+paulshuler@gmail.com - http://macroform.bandcamp.com/
+
+macroform-the_simplicity - Paul Shuler (Macroform)
+paulshuler@gmail.com - http://macroform.bandcamp.com/
+
+manolo_camp-morning_coffee - Manolo Camp
+beatbastard@gmx.net - http://ccmixter.org/people/ManoloCamp
+
+reno_project-system - Reno Project
+renoproject@hotmail.com - http://www.jamendo.com/en/album/23661

UPGRADE.txt

@@ -52,8 +52,10 @@ Dialing:
 
 IAX: 
 
-* The naming convention for IAX channels has changed in a minor way such 
-  that the call number follows a "-" rather than a "/" character.
+* The naming convention for IAX channels has changed in two ways: 
+   1. The call number follows a "-" rather than a "/" character.
+   2. The name of the channel has been simplified to IAX2/peer-callno,
+   rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.
 
 SIP:
 
@@ -129,6 +131,11 @@ Applications:
   'mailbox options' menu, and 'change your password' option has been
   moved to option 5.
 
+* The application VoiceMailMain now only matches the 'default' context if
+  none is specified in the arguments.  (This was the previously 
+  documented behavior, however, we didn't follow that behavior.)  The old
+  behavior can be restored by setting searchcontexts=yes in voicemail.conf.
+
 Queues:
 
 * A queue is now considered empty not only if there are no members but if

agi/.cvsignore

@@ -1,3 +0,0 @@
-eagi-test
-eagi-sphinx-test
-.depend

app.c

@@ -316,8 +316,12 @@ int ast_dtmf_stream(struct ast_channel *chan,struct ast_channel *peer,char *digi
 				}
 			}
 		}
-		if (peer)
-			res = ast_autoservice_stop(peer);
+		if (peer) {
+			/* Stop autoservice on the peer channel, but don't overwrite any error condition 
+			   that has occurred previously while acting on the primary channel */	
+			if (ast_autoservice_stop(peer) && !res)
+				res = -1;
+		}
 	}
 	return res;
 }
@@ -430,11 +434,11 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
 			   const char *stop, const char *pause,
 			   const char *restart, int skipms) 
 {
-	long elapsed = 0, last_elapsed = 0;
 	char *breaks = NULL;
 	char *end = NULL;
 	int blen = 2;
 	int res;
+	long pause_restart_point = 0;
 
 	if (stop)
 		blen += strlen(stop);
@@ -456,9 +460,6 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
 	if (chan->_state != AST_STATE_UP)
 		res = ast_answer(chan);
 
-	if (chan)
-		ast_stopstream(chan);
-
 	if (file) {
 		if ((end = strchr(file,':'))) {
 			if (!strcasecmp(end, ":end")) {
@@ -469,25 +470,18 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
 	}
 
 	for (;;) {
-		struct timeval started = ast_tvnow();
-
-		if (chan)
-			ast_stopstream(chan);
+		ast_stopstream(chan);
 		res = ast_streamfile(chan, file, chan->language);
 		if (!res) {
-			if (end) {
+			if (pause_restart_point) {
+				ast_seekstream(chan->stream, pause_restart_point, SEEK_SET);
+				pause_restart_point = 0;
+			}
+			else if (end) {
 				ast_seekstream(chan->stream, 0, SEEK_END);
-				end=NULL;
-			}
-			res = 1;
-			if (elapsed) {
-				ast_stream_fastforward(chan->stream, elapsed);
-				last_elapsed = elapsed - 200;
-			}
-			if (res)
-				res = ast_waitstream_fr(chan, breaks, fwd, rev, skipms);
-			else
-				break;
+				end = NULL;
+			};
+			res = ast_waitstream_fr(chan, breaks, fwd, rev, skipms);
 		}
 
 		if (res < 1)
@@ -496,17 +490,16 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
 		/* We go at next loop if we got the restart char */
 		if (restart && strchr(restart, res)) {
 			ast_log(LOG_DEBUG, "we'll restart the stream here at next loop\n");
-			elapsed=0; /* To make sure the next stream will start at beginning */
+			pause_restart_point = 0;
 			continue;
 		}
 
-		if (pause != NULL && strchr(pause, res)) {
-			elapsed = ast_tvdiff_ms(ast_tvnow(), started) + last_elapsed;
-			for(;;) {
-				if (chan)
-					ast_stopstream(chan);
+		if (pause && strchr(pause, res)) {
+			pause_restart_point = ast_tellstream(chan->stream);
+			for (;;) {
+				ast_stopstream(chan);
 				res = ast_waitfordigit(chan, 1000);
-				if (res == 0)
+				if (!res)
 					continue;
 				else if (res == -1 || strchr(pause, res) || (stop && strchr(stop, res)))
 					break;
@@ -516,17 +509,16 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
 				continue;
 			}
 		}
+
 		if (res == -1)
 			break;
 
 		/* if we get one of our stop chars, return it to the calling function */
-		if (stop && strchr(stop, res)) {
-			/* res = 0; */
+		if (stop && strchr(stop, res))
 			break;
-		}
 	}
-	if (chan)
-		ast_stopstream(chan);
+
+	ast_stopstream(chan);
 
 	return res;
 }

apps/.cvsignore

@@ -1 +0,0 @@
-.depend

apps/app_chanspy.c

@@ -208,9 +208,9 @@ static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, s
 
 static void stop_spying(struct ast_channel *chan, struct ast_channel_spy *spy) 
 {
-	/* If our status has changed, then the channel we're spying on is gone....
+	/* If our status has changed to DONE, then the channel we're spying on is gone....
 	   DON'T TOUCH IT!!!  RUN AWAY!!! */
-	if (spy->status != CHANSPY_RUNNING)
+	if (spy->status == CHANSPY_DONE)
 		return;
 
 	if (!chan)
@@ -439,8 +439,8 @@ static int chanspy_exec(struct ast_channel *chan, void *data)
 
 	if (recbase) {
 		char filename[512];
-		snprintf(filename,sizeof(filename),"%s/%s.%ld.raw",ast_config_AST_MONITOR_DIR, recbase, time(NULL));
-		if ((fd = open(filename, O_CREAT | O_WRONLY, O_TRUNC)) <= 0) {
+		snprintf(filename,sizeof(filename),"%s/%s.%d.raw",ast_config_AST_MONITOR_DIR, recbase, (int)time(NULL));
+		if ((fd = open(filename, O_CREAT | O_WRONLY, O_TRUNC, 0644)) <= 0) {
 			ast_log(LOG_WARNING, "Cannot open %s for recording\n", filename);
 			fd = 0;
 		}
@@ -482,7 +482,7 @@ static int chanspy_exec(struct ast_channel *chan, void *data)
 					}
 				}
 				
-				if (igrp && (!spec || ((strlen(spec) < strlen(peer->name) &&
+				if (igrp && (!spec || ((strlen(spec) <= strlen(peer->name) &&
 							!strncasecmp(peer->name, spec, strlen(spec)))))) {
 					if (peer && (!bronly || ast_bridged_channel(peer)) &&
 					    !ast_check_hangup(peer) && !ast_test_flag(peer, AST_FLAG_SPYING)) {

apps/app_controlplayback.c

@@ -140,7 +140,7 @@ static int controlplayback_exec(struct ast_channel *chan, void *data)
 	} else {
 		if (res < 0) {
 			if (priority_jump || option_priority_jumping) {
-				if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
+				if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
 					ast_log(LOG_WARNING, "ControlPlayback tried to jump to priority n+101 as requested, but priority didn't exist\n");
 				}
 			}

apps/app_cut.c

@@ -36,7 +36,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
-#include "asterisk/version.h"
 #include "asterisk/app.h"
 
 /* Maximum length of any variable */
@@ -146,8 +145,9 @@ static int sort_internal(struct ast_channel *chan, char *data, char *buffer, siz
 		int blen = strlen(buffer);
 		if (element_count++) {
 			strncat(buffer + blen, ",", buflen - blen - 1);
+			blen++;
 		}
-		strncat(buffer + blen + 1, sortable_keys[count2].key, buflen - blen - 2);
+		strncat(buffer + blen, sortable_keys[count2].key, buflen - blen - 1);
 	}
 
 	return 0;
@@ -253,6 +253,8 @@ static int cut_internal(struct ast_channel *chan, char *data, char *buffer, size
 				}
 			}
 		}
+	} else {
+		return ERROR_NOARG;
 	}
 	return 0;
 }
@@ -386,7 +388,7 @@ static char *acf_cut_exec(struct ast_channel *chan, char *cmd, char *data, char
 
 	switch (cut_internal(chan, data, buf, len)) {
 	case ERROR_NOARG:
-		ast_log(LOG_ERROR, "Cut() requires an argument\n");
+		ast_log(LOG_ERROR, "CUT() requires an argument\n");
 		break;
 	case ERROR_NOMEM:
 		ast_log(LOG_ERROR, "Out of memory\n");

apps/app_dial.c

@@ -75,7 +75,7 @@ static char *descrip =
 "continue if no requested channels can be called, or if the timeout expires.\n\n"
 "  This application sets the following channel variables upon completion:\n"
 "    DIALEDTIME   - This is the time from dialing a channel until when it\n"
-"                   answers.\n" 
+"                   is disconnected.\n" 
 "    ANSWEREDTIME - This is the amount of time for actual call.\n"
 "    DIALSTATUS   - This is the status of the call:\n"
 "                   CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n" 
@@ -161,7 +161,7 @@ static char *descrip =
 "    S(x) - Hang up the call after 'x' seconds *after* the called party has\n"
 "           answered the call.\n"  	
 "    t    - Allow the called party to transfer the calling party by sending the\n"
-"           DTMF sequence defiend in features.conf.\n"
+"           DTMF sequence defined in features.conf.\n"
 "    T    - Allow the calling party to transfer the called party by sending the\n"
 "           DTMF sequence defined in features.conf.\n"
 "    w    - Allow the called party to enable recording of the call by sending\n"
@@ -512,10 +512,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
 						if (in->cid.cid_ani) {
 							if (o->chan->cid.cid_ani)
 								free(o->chan->cid.cid_ani);
-							o->chan->cid.cid_ani = malloc(strlen(in->cid.cid_ani) + 1);
-							if (o->chan->cid.cid_ani)
-								ast_copy_string(o->chan->cid.cid_ani, in->cid.cid_ani, sizeof(o->chan->cid.cid_ani));
-							else
+							o->chan->cid.cid_ani = strdup(in->cid.cid_ani);
+							if (!o->chan->cid.cid_ani)
 								ast_log(LOG_WARNING, "Out of memory\n");
 						}
 						if (o->chan->cid.cid_rdnis) 
@@ -650,6 +648,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
 					ast_hangup(o->chan);
 					o->chan = NULL;
 					ast_clear_flag(o, DIAL_STILLGOING);
+					HANDLE_CAUSE(in->hangupcause, in);
 				}
 			}
 			o = o->next;
@@ -737,6 +736,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
 	char numsubst[AST_MAX_EXTENSION];
 	char restofit[AST_MAX_EXTENSION];
 	char cidname[AST_MAX_EXTENSION];
+	char toast[80];
 	char *newnum;
 	char *l;
 	int privdb_val=0;
@@ -963,7 +963,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
 	/* If a channel group has been specified, get it for use when we create peer channels */
 	outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP");
 
-	ast_copy_flags(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID);
+	ast_copy_flags(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP);
 	cur = args.peers;
 	do {
 		/* Remember where to start next time */
@@ -1515,8 +1515,6 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
 		}
 		
 		if (!res) {
-			char toast[80];
-
 			memset(&config,0,sizeof(struct ast_bridge_config));
 			if (play_to_caller)
 				ast_set_flag(&(config.features_caller), AST_FEATURE_PLAY_WARNING);
@@ -1560,13 +1558,15 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
 			}
 			res = ast_bridge_call(chan,peer,&config);
 			time(&end_time);
-			snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
-			pbx_builtin_setvar_helper(chan, "DIALEDTIME", toast);
 			snprintf(toast, sizeof(toast), "%ld", (long)(end_time - answer_time));
 			pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", toast);
 			
-		} else 
+		} else {
+			time(&end_time);
 			res = -1;
+		}
+		snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
+		pbx_builtin_setvar_helper(chan, "DIALEDTIME", toast);
 		
 		if (res != AST_PBX_NO_HANGUP_PEER) {
 			if (!chan->_softhangup)

apps/app_directed_pickup.c

@@ -77,7 +77,7 @@ static int pickup_exec(struct ast_channel *chan, void *data)
 
 	/* Find a channel to pickup */
 	origin = ast_get_channel_by_exten_locked(exten, context);
-	if (origin) {
+	if (origin && origin->cdr) {
 		ast_cdr_getvar(origin->cdr, "dstchannel", &tmp, workspace,
 			       sizeof(workspace), 0);
 		if (tmp) {
@@ -89,6 +89,8 @@ static int pickup_exec(struct ast_channel *chan, void *data)
 		}
 		ast_mutex_unlock(&origin->lock);
 	} else {
+		if (origin)
+			ast_mutex_unlock(&origin->lock);
 		ast_log(LOG_DEBUG, "No originating channel found.\n");
 	}
 	

apps/app_externalivr.c

@@ -43,6 +43,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/linkedlists.h"
+#include "asterisk/app.h"
 
 static const char *tdesc = "External IVR Interface Application";
 
@@ -93,9 +94,9 @@ static void send_child_event(FILE *handle, const char event, const char *data,
 	char tmp[256];
 
 	if (!data) {
-		snprintf(tmp, sizeof(tmp), "%c,%10ld", event, time(NULL));
+		snprintf(tmp, sizeof(tmp), "%c,%10d", event, (int)time(NULL));
 	} else {
-		snprintf(tmp, sizeof(tmp), "%c,%10ld,%s", event, time(NULL), data);
+		snprintf(tmp, sizeof(tmp), "%c,%10d,%s", event, (int)time(NULL), data);
 	}
 
 	fprintf(handle, "%s\n", tmp);
@@ -250,10 +251,9 @@ static int app_exec(struct ast_channel *chan, void *data)
 	int res = -1;
 	int gen_active = 0;
 	int pid;
-	char *command;
 	char *argv[32];
 	int argc = 1;
-	char *buf;
+	char *buf, *command;
 	FILE *child_commands = NULL;
 	FILE *child_errors = NULL;
 	FILE *child_events = NULL;
@@ -270,11 +270,13 @@ static int app_exec(struct ast_channel *chan, void *data)
 	}
 
 	buf = ast_strdupa(data);
-	command = strsep(&buf, "|");
-	memset(argv, 0, sizeof(argv) / sizeof(argv[0]));
-	argv[0] = command;
-	while ((argc < 31) && (argv[argc++] = strsep(&buf, "|")));
-	argv[argc] = NULL;
+	if (!buf) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	argc = ast_app_separate_args(buf, '|', argv, sizeof(argv) / sizeof(argv[0]));
 
 	if (pipe(child_stdin)) {
 		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child input: %s\n", strerror(errno));
@@ -316,8 +318,8 @@ static int app_exec(struct ast_channel *chan, void *data)
 		dup2(child_stderr[1], STDERR_FILENO);
 		for (i = STDERR_FILENO + 1; i < 1024; i++)
 			close(i);
-		execv(command, argv);
-		fprintf(stderr, "Failed to execute '%s': %s\n", command, strerror(errno));
+		execv(argv[0], argv);
+		fprintf(stderr, "Failed to execute '%s': %s\n", argv[0], strerror(errno));
 		exit(1);
 	} else {
 		/* parent process */
@@ -343,8 +345,6 @@ static int app_exec(struct ast_channel *chan, void *data)
 			goto exit;
 		}
 
-		setvbuf(child_events, NULL, _IONBF, 0);
-
 		if (!(child_commands = fdopen(child_commands_fd, "r"))) {
 			ast_chan_log(LOG_WARNING, chan, "Could not open stream for child commands\n");
 			goto exit;
@@ -355,6 +355,10 @@ static int app_exec(struct ast_channel *chan, void *data)
 			goto exit;
 		}
 
+		setvbuf(child_events, NULL, _IONBF, 0);
+		setvbuf(child_commands, NULL, _IONBF, 0);
+		setvbuf(child_errors, NULL, _IONBF, 0);
+
 		res = 0;
 
 		while (1) {

apps/app_festival.c

@@ -175,7 +175,8 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 	if (chan->_state != AST_STATE_UP)
 		ast_answer(chan);
 	ast_stopstream(chan);
-
+	ast_indicate(chan, -1);
+	
 	owriteformat = chan->writeformat;
 	res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
 	if (res < 0) {
@@ -455,8 +456,20 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
 	/* This assumes only one waveform will come back, also LP is unlikely */
 	wave = 0;
 	do {
+               int read_data;
 		for (n=0; n < 3; )
-			n += read(fd,ack+n,3-n);
+               {
+                       read_data = read(fd,ack+n,3-n);
+                       /* this avoids falling in infinite loop
+                        * in case that festival server goes down
+                        * */
+                       if ( read_data == -1 )
+                       {
+                               ast_log(LOG_WARNING,"Unable to read from cache/festival fd");
+                               return -1;
+                       }
+                       n += read_data;
+               }
 		ack[3] = '\0';
 		if (strcmp(ack,"WV\n") == 0) {         /* receive a waveform */
 			ast_log(LOG_DEBUG,"Festival WV command\n");

apps/app_hasnewvoicemail.c

@@ -68,7 +68,7 @@ static char *hasnewvoicemail_descrip =
 "Assumes folder 'INBOX' if folder is not specified. Optionally sets <varname> to the number of messages\n" 
 "in that folder.\n"
 "  The option string may contain zero of the following character:\n"
-"	'j' -- jump to priority n+101, if there is new voicemail in tolder 'folder' or INBOX\n"
+"	'j' -- jump to priority n+101, if there is new voicemail in folder 'folder' or INBOX\n"
 "  This application sets the following channel variable upon completion:\n"
 "	HASVMSTATUS		The result of the new voicemail check returned as a text string as follows\n"
 "		<# of messages in the folder, 0 for NONE>\n";
@@ -90,7 +90,6 @@ static int hasvoicemail_internal(char *context, char *box, char *folder)
 		while ((vment = readdir(vmdir))) {
 			if (!strncmp(vment->d_name + 7, ".txt", 4)) {
 				count++;
-				break;
 			}
 		}
 		closedir(vmdir);
@@ -163,7 +162,7 @@ static int hasvoicemail_exec(struct ast_channel *chan, void *data)
 	if (vmcount > 0) {
 		/* Branch to the next extension */
 		if (priority_jump || option_priority_jumping) {
-			if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) 
+			if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) 
 				ast_log(LOG_WARNING, "VM box %s@%s has new voicemail, but extension %s, priority %d doesn't exist\n", vmbox, context, chan->exten, chan->priority + 101);
 		}
 	}

apps/app_md5.c

@@ -157,9 +157,9 @@ static int md5check_exec(struct ast_channel *chan, void *data)
 		ast_log(LOG_DEBUG, "ERROR: MD5 not verified: %s -- %s\n", args.md5hash, args.string);
 	pbx_builtin_setvar_helper(chan, "CHECKMD5STATUS", "NOMATCH");		
 	if (priority_jump || option_priority_jumping) {
-		if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
+		if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
 			if (option_debug > 2)
-				ast_log(LOG_DEBUG, "ERROR: Can't jump to exten+101 (e%s,p%d), sorry\n", chan->exten,chan->priority+101);
+				ast_log(LOG_DEBUG, "Can't jump to exten+101 (e%s,p%d), sorry\n", chan->exten,chan->priority+101);
 	}
 	LOCAL_USER_REMOVE(u);
 	return res;

apps/app_meetme.c

@@ -263,14 +263,17 @@ static char *istalking(int x)
 		return "(not talking)";
 }
 
-static int careful_write(int fd, unsigned char *data, int len)
+static int careful_write(int fd, unsigned char *data, int len, int block)
 {
 	int res;
 	int x;
 
 	while (len) {
-		x = ZT_IOMUX_WRITE | ZT_IOMUX_SIGEVENT;
-		res = ioctl(fd, ZT_IOMUX, &x);
+		if (block) {
+			x = ZT_IOMUX_WRITE | ZT_IOMUX_SIGEVENT;
+			res = ioctl(fd, ZT_IOMUX, &x);
+		} else
+			res = 0;
 		if (res >= 0)
 			res = write(fd, data, len);
 		if (res < 1) {
@@ -423,7 +426,7 @@ static void conf_play(struct ast_channel *chan, struct ast_conference *conf, int
 		len = 0;
 	}
 	if (data) 
-		careful_write(conf->fd, data, len);
+		careful_write(conf->fd, data, len, 1);
 
 	ast_mutex_unlock(&conflock);
 
@@ -1307,14 +1310,18 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, int c
 						   audio frames (in which case carefully writing would only
 						   have delayed the audio even further).
 						*/
-						write(fd, f->data, f->datalen);
+						/* As it turns out, we do want to use careful write.  We just
+						   don't want to block, but we do want to at least *try*
+						   to write out all the samples.
+						 */
+						careful_write(fd, f->data, f->datalen, 0);
 					}
 				} else if ((f->frametype == AST_FRAME_DTMF) && (confflags & CONFFLAG_EXIT_CONTEXT)) {
 					char tmp[2];
 
 					tmp[0] = f->subclass;
 					tmp[1] = '\0';
-					if (ast_goto_if_exists(chan, exitcontext, tmp, 1)) {
+					if (!ast_goto_if_exists(chan, exitcontext, tmp, 1)) {
 						ret = 0;
 						break;
 					} else if (option_debug > 1)
@@ -1623,7 +1630,8 @@ static struct ast_conference *find_conf(struct ast_channel *chan, char *confno,
 			if (dynamic_pin) {
 				if (dynamic_pin[0] == 'q') {
 					/* Query the user to enter a PIN */
-					ast_app_getdata(chan, "conf-getpin", dynamic_pin, AST_MAX_EXTENSION - 1, 0);
+					if (ast_app_getdata(chan, "conf-getpin", dynamic_pin, AST_MAX_EXTENSION - 1, 0) < 0)
+						return NULL;
 				}
 				cnf = build_conf(confno, dynamic_pin, "", make, dynamic);
 			} else {

apps/app_milliwatt.c

@@ -73,30 +73,29 @@ static void milliwatt_release(struct ast_channel *chan, void *data)
 static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int samples)
 {
 	struct ast_frame wf;
-	unsigned char waste[AST_FRIENDLY_OFFSET];
-	unsigned char buf[640];
+	unsigned char buf[AST_FRIENDLY_OFFSET + 640];
 	int i,*indexp = (int *) data;
 
-	if (len > sizeof(buf))
+	if (len + AST_FRIENDLY_OFFSET > sizeof(buf))
 	{
-		ast_log(LOG_WARNING,"Only doing %d bytes (%d bytes requested)\n",(int)sizeof(buf),len);
-		len = sizeof(buf);
+		ast_log(LOG_WARNING,"Only doing %d bytes (%d bytes requested)\n",(int)(sizeof(buf) - AST_FRIENDLY_OFFSET),len);
+		len = sizeof(buf) - AST_FRIENDLY_OFFSET;
 	}
-	waste[0] = 0; /* make compiler happy */
 	wf.frametype = AST_FRAME_VOICE;
 	wf.subclass = AST_FORMAT_ULAW;
 	wf.offset = AST_FRIENDLY_OFFSET;
 	wf.mallocd = 0;
-	wf.data = buf;
+	wf.data = buf + AST_FRIENDLY_OFFSET;
 	wf.datalen = len;
 	wf.samples = wf.datalen;
 	wf.src = "app_milliwatt";
 	wf.delivery.tv_sec = 0;
 	wf.delivery.tv_usec = 0;
+	wf.prev = wf.next = NULL;
 	/* create a buffer containing the digital milliwatt pattern */
 	for(i = 0; i < len; i++)
 	{
-		buf[i] = digital_milliwatt[(*indexp)++];
+		buf[AST_FRIENDLY_OFFSET + i] = digital_milliwatt[(*indexp)++];
 		*indexp &= 7;
 	}
 	if (ast_write(chan,&wf) < 0)

apps/app_mixmonitor.c

@@ -112,9 +112,9 @@ AST_APP_OPTIONS(mixmonitor_opts, {
 
 static void stopmon(struct ast_channel *chan, struct ast_channel_spy *spy) 
 {
-	/* If our status has changed, then the channel we're spying on is gone....
+	/* If our status has changed to DONE, then the channel we're spying on is gone....
 	   DON'T TOUCH IT!!!  RUN AWAY!!! */
-	if (spy->status != CHANSPY_RUNNING)
+	if (spy->status == CHANSPY_DONE)
 		return;
 
 	if (!chan)

apps/app_osplookup.c

@@ -217,6 +217,10 @@ static int ospnext_exec(struct ast_channel *chan, void *data)
 	temp = pbx_builtin_getvar_helper(chan, "OSPHANDLE");
 	result.handle = -1;
 	if (!ast_strlen_zero(temp) && (sscanf(temp, "%d", &result.handle) == 1) && (result.handle > -1)) {
+		temp = pbx_builtin_getvar_helper(chan, "OSPRESULTS");
+		if (ast_strlen_zero(temp) || (sscanf(temp, "%d", &result.numresults) != 1)) {
+			result.numresults = 0;
+		}
 		if ((res = ast_osp_next(&result, cause)) > 0) {
 			char tmp[80];
 			snprintf(tmp, sizeof(tmp), "%d", result.handle);

apps/app_page.c

@@ -40,6 +40,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/module.h"
 #include "asterisk/file.h"
 #include "asterisk/app.h"
+#include "asterisk/chanvars.h"
 
 
 static const char *tdesc = "Page Multiple Phones";
@@ -77,13 +78,14 @@ struct calloutdata {
 	char tech[64];
 	char resource[256];
 	char meetmeopts[64];
+	struct ast_variable *variables;
 };
 
 static void *page_thread(void *data)
 {
 	struct calloutdata *cd = data;
 	ast_pbx_outgoing_app(cd->tech, AST_FORMAT_SLINEAR, cd->resource, 30000,
-		"MeetMe", cd->meetmeopts, NULL, 0, cd->cidnum, cd->cidname, NULL, NULL);
+		"MeetMe", cd->meetmeopts, NULL, 0, cd->cidnum, cd->cidname, cd->variables, NULL);
 	free(cd);
 	return NULL;
 }
@@ -91,6 +93,9 @@ static void *page_thread(void *data)
 static void launch_page(struct ast_channel *chan, const char *meetmeopts, const char *tech, const char *resource)
 {
 	struct calloutdata *cd;
+	const char *varname;
+	struct ast_variable *lastvar = NULL;
+	struct ast_var_t *varptr;
 	pthread_t t;
 	pthread_attr_t attr;
 	cd = malloc(sizeof(struct calloutdata));
@@ -101,6 +106,29 @@ static void launch_page(struct ast_channel *chan, const char *meetmeopts, const
 		ast_copy_string(cd->tech, tech, sizeof(cd->tech));
 		ast_copy_string(cd->resource, resource, sizeof(cd->resource));
 		ast_copy_string(cd->meetmeopts, meetmeopts, sizeof(cd->meetmeopts));
+
+		AST_LIST_TRAVERSE(&chan->varshead, varptr, entries) {
+			if (!(varname = ast_var_full_name(varptr)))
+				continue;
+			if (varname[0] == '_') {
+				struct ast_variable *newvar = NULL;
+
+				if (varname[1] == '_') {
+					newvar = ast_variable_new(varname, ast_var_value(varptr));
+				} else {
+					newvar = ast_variable_new(&varname[1], ast_var_value(varptr));
+				}
+
+				if (newvar) {
+					if (lastvar)
+						lastvar->next = newvar;
+					else
+						cd->variables = newvar;
+					lastvar = newvar;
+				}
+			}
+		}
+
 		pthread_attr_init(&attr);
 		pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
 		if (ast_pthread_create(&t, &attr, page_thread, cd)) {

apps/app_privacy.c

@@ -193,7 +193,7 @@ static int privacy_exec (struct ast_channel *chan, void *data)
 		}
 		
 		/*Got a number, play sounds and send them on their way*/
-		if ((retries < maxretries) && res == 1 ) {
+		if ((retries < maxretries) && !res ) {
 			res = ast_streamfile(chan, "privacy-thankyou", chan->language);
 			if (!res)
 				res = ast_waitstream(chan, "");

apps/app_queue.c

@@ -484,11 +484,11 @@ static void *changethread(void *data)
 							"Membership: %s\r\n"
 							"Penalty: %d\r\n"
 							"CallsTaken: %d\r\n"
-							"LastCall: %ld\r\n"
+							"LastCall: %d\r\n"
 							"Status: %d\r\n"
 							"Paused: %d\r\n",
 						q->name, cur->interface, cur->dynamic ? "dynamic" : "static",
-						cur->penalty, cur->calls, cur->lastcall, cur->status, cur->paused);
+						cur->penalty, cur->calls, (int)cur->lastcall, cur->status, cur->paused);
 					}
 				}
 			}
@@ -755,28 +755,66 @@ static void rt_handle_member_record(struct ast_call_queue *q, char *interface, c
 	}
 }
 
+static void free_members(struct ast_call_queue *q, int all)
+{
+	/* Free non-dynamic members */
+	struct member *curm, *next, *prev = NULL;
+
+	for (curm = q->members; curm; curm = next) {
+		next = curm->next;
+		if (all || !curm->dynamic) {
+			if (prev)
+				prev->next = next;
+			else
+				q->members = next;
+			free(curm);
+		} else 
+			prev = curm;
+	}
+}
+
+static void destroy_queue(struct ast_call_queue *q)
+{
+	free_members(q, 1);
+	ast_mutex_destroy(&q->lock);
+	free(q);
+}
+
+static void remove_queue(struct ast_call_queue *q)
+{
+	struct ast_call_queue *cur, *prev = NULL;
+
+	ast_mutex_lock(&qlock);
+	for (cur = queues; cur; cur = cur->next) {
+		if (cur == q) {
+			if (prev)
+				prev->next = cur->next;
+			else
+				queues = cur->next;
+		} else {
+			prev = cur;
+		}
+	}
+	ast_mutex_unlock(&qlock);
+}
 
 /*!\brief Reload a single queue via realtime.
    \return Return the queue, or NULL if it doesn't exist.
-   \note Should be called with the global qlock locked.
-   When found, the queue is returned with q->lock locked. */
-static struct ast_call_queue *reload_queue_rt(const char *queuename, struct ast_variable *queue_vars, struct ast_config *member_config)
+   \note Should be called with the global qlock locked. */
+static struct ast_call_queue *find_queue_by_name_rt(const char *queuename, struct ast_variable *queue_vars, struct ast_config *member_config)
 {
 	struct ast_variable *v;
-	struct ast_call_queue *q, *prev_q;
+	struct ast_call_queue *q, *prev_q = NULL;
 	struct member *m, *prev_m, *next_m;
 	char *interface;
 	char *tmp, *tmp_name;
 	char tmpbuf[64];	/* Must be longer than the longest queue param name. */
 
 	/* Find the queue in the in-core list (we will create a new one if not found). */
-	q = queues;
-	prev_q = NULL;
-	while (q) {
+	for (q = queues; q; q = q->next) {
 		if (!strcasecmp(q->name, queuename)) {
 			break;
 		}
-		q = q->next;
 		prev_q = q;
 	}
 
@@ -788,6 +826,7 @@ static struct ast_call_queue *reload_queue_rt(const char *queuename, struct ast_
 				ast_mutex_unlock(&q->lock);
 				return NULL;
 			} else {
+				ast_mutex_unlock(&q->lock);
 				return q;
 			}
 		}
@@ -814,7 +853,7 @@ static struct ast_call_queue *reload_queue_rt(const char *queuename, struct ast_
 					prev_q->next = q->next;
 				}
 				ast_mutex_unlock(&q->lock);
-				free(q);
+				destroy_queue(q);
 			} else
 				ast_mutex_unlock(&q->lock);
 		}
@@ -850,10 +889,11 @@ static struct ast_call_queue *reload_queue_rt(const char *queuename, struct ast_
 		v = v->next;
 	}
 
-	/* Temporarily set members dead so we can detect deleted ones. */
+	/* Temporarily set non-dynamic members dead so we can detect deleted ones. */
 	m = q->members;
 	while (m) {
-		m->dead = 1;
+		if (!m->dynamic)
+			m->dead = 1;
 		m = m->next;
 	}
 
@@ -881,13 +921,60 @@ static struct ast_call_queue *reload_queue_rt(const char *queuename, struct ast_
 		m = next_m;
 	}
 
+	ast_mutex_unlock(&q->lock);
+
+	return q;
+}
+
+static struct ast_call_queue *load_realtime_queue(char *queuename)
+{
+	struct ast_variable *queue_vars = NULL;
+	struct ast_config *member_config = NULL;
+	struct ast_call_queue *q;
+
+	/* Find the queue in the in-core list first. */
+	ast_mutex_lock(&qlock);
+	for (q = queues; q; q = q->next) {
+		if (!strcasecmp(q->name, queuename)) {
+			break;
+		}
+	}
+	ast_mutex_unlock(&qlock);
+
+	if (!q) {
+		/*! \note Load from realtime before taking the global qlock, to avoid blocking all
+		   queue operations while waiting for the DB.
+
+		   This will be two separate database transactions, so we might
+		   see queue parameters as they were before another process
+		   changed the queue and member list as it was after the change.
+		   Thus we might see an empty member list when a queue is
+		   deleted. In practise, this is unlikely to cause a problem. */
+
+		queue_vars = ast_load_realtime("queues", "name", queuename, NULL);
+		if (queue_vars) {
+			member_config = ast_load_realtime_multientry("queue_members", "interface LIKE", "%", "queue_name", queuename, NULL);
+			if (!member_config) {
+				ast_log(LOG_ERROR, "no queue_members defined in your config (extconfig.conf).\n");
+				return NULL;
+			}
+		}
+
+		ast_mutex_lock(&qlock);
+
+		q = find_queue_by_name_rt(queuename, queue_vars, member_config);
+		if (member_config)
+			ast_config_destroy(member_config);
+		if (queue_vars)
+			ast_variables_destroy(queue_vars);
+
+		ast_mutex_unlock(&qlock);
+	}
 	return q;
 }
 
 static int join_queue(char *queuename, struct queue_ent *qe, enum queue_result *reason)
 {
-	struct ast_variable *queue_vars = NULL;
-	struct ast_config *member_config = NULL;
 	struct ast_call_queue *q;
 	struct queue_ent *cur, *prev = NULL;
 	int res = -1;
@@ -895,35 +982,12 @@ static int join_queue(char *queuename, struct queue_ent *qe, enum queue_result *
 	int inserted = 0;
 	enum queue_member_status stat;
 
-	/*! \note Load from realtime before taking the global qlock, to avoid blocking all
-	   queue operations while waiting for the DB.
-
-	   This will be two separate database transactions, so we might
-	   see queue parameters as they were before another process
-	   changed the queue and member list as it was after the change.
-	   Thus we might see an empty member list when a queue is
-	   deleted. In practise, this is unlikely to cause a problem. */
-	queue_vars = ast_load_realtime("queues", "name", queuename, NULL);
-	if (queue_vars) {
-		member_config = ast_load_realtime_multientry("queue_members", "interface LIKE", "%", "queue_name", queuename, NULL);
-		if (!member_config) {
-			ast_log(LOG_ERROR, "no queue_members defined in your config (extconfig.conf).\n");
-			return res;
-		}
-	}
+	q = load_realtime_queue(queuename);
+	if (!q)
+		return res;
 
 	ast_mutex_lock(&qlock);
-	q = reload_queue_rt(queuename, queue_vars, member_config);
-	/* Note: If found, reload_queue_rt() returns with q->lock locked. */
-	if(member_config)
-		ast_config_destroy(member_config);
-	if(queue_vars)
-		ast_variables_destroy(queue_vars);
-
-	if (!q) {
-		ast_mutex_unlock(&qlock);
-		return res;
-	}
+	ast_mutex_lock(&q->lock);
 
 	/* This is our one */
 	stat = get_member_status(q);
@@ -975,48 +1039,6 @@ ast_log(LOG_NOTICE, "Queue '%s' Join, Channel '%s', Position '%d'\n", q->name, q
 	return res;
 }
 
-static void free_members(struct ast_call_queue *q, int all)
-{
-	/* Free non-dynamic members */
-	struct member *curm, *next, *prev;
-
-	curm = q->members;
-	prev = NULL;
-	while(curm) {
-		next = curm->next;
-		if (all || !curm->dynamic) {
-			if (prev)
-				prev->next = next;
-			else
-				q->members = next;
-			free(curm);
-		} else 
-			prev = curm;
-		curm = next;
-	}
-}
-
-static void destroy_queue(struct ast_call_queue *q)
-{
-	struct ast_call_queue *cur, *prev = NULL;
-
-	ast_mutex_lock(&qlock);
-	for (cur = queues; cur; cur = cur->next) {
-		if (cur == q) {
-			if (prev)
-				prev->next = cur->next;
-			else
-				queues = cur->next;
-		} else {
-			prev = cur;
-		}
-	}
-	ast_mutex_unlock(&qlock);
-	free_members(q, 1);
-        ast_mutex_destroy(&q->lock);
-	free(q);
-}
-
 static int play_file(struct ast_channel *chan, char *filename)
 {
 	int res;
@@ -1224,6 +1246,7 @@ ast_log(LOG_NOTICE, "Queue '%s' Leave, Channel '%s'\n", q->name, qe->chan->name
 	ast_mutex_unlock(&q->lock);
 	if (q->dead && !q->count) {	
 		/* It's dead and nobody is in it, so kill it */
+		remove_queue(q);
 		destroy_queue(q);
 	}
 }
@@ -1261,11 +1284,11 @@ static int update_status(struct ast_call_queue *q, struct member *member, int st
 					"Membership: %s\r\n"
 					"Penalty: %d\r\n"
 					"CallsTaken: %d\r\n"
-					"LastCall: %ld\r\n"
+					"LastCall: %d\r\n"
 					"Status: %d\r\n"
 					"Paused: %d\r\n",
 				q->name, cur->interface, cur->dynamic ? "dynamic" : "static",
-				cur->penalty, cur->calls, cur->lastcall, cur->status, cur->paused);
+				cur->penalty, cur->calls, (int)cur->lastcall, cur->status, cur->paused);
 			}
 			break;
 		}
@@ -1304,7 +1327,7 @@ static int compare_weight(struct ast_call_queue *rq, struct member *member)
 		ast_mutex_lock(&q->lock);
 		if (q->count && q->members) {
 			for (mem = q->members; mem; mem = mem->next) {
-				if (mem == member) {
+				if (!strcmp(mem->interface, member->interface)) {
 					ast_log(LOG_DEBUG, "Found matching member %s in queue '%s'\n", mem->interface, q->name);
 					if (q->weight > rq->weight) {
 						ast_log(LOG_DEBUG, "Queue '%s' (weight %d, calls %d) is preferred over '%s' (weight %d, calls %d)\n", q->name, q->weight, q->count, rq->name, rq->weight, rq->count);
@@ -2404,41 +2427,42 @@ static int add_to_queue(char *queuename, char *interface, int penalty, int pause
 	struct member *new_member;
 	int res = RES_NOSUCHQUEUE;
 
+	/* \note Ensure the appropriate realtime queue is loaded.  Note that this
+	 * short-circuits if the queue is already in memory. */
+	q = load_realtime_queue(queuename);
+
 	ast_mutex_lock(&qlock);
-	for (q = queues ; q ; q = q->next) {
+
+	if (q) {
 		ast_mutex_lock(&q->lock);
-		if (!strcmp(q->name, queuename)) {
-			if (interface_exists(q, interface) == NULL) {
-				new_member = create_queue_member(interface, penalty, paused);
+		if (interface_exists(q, interface) == NULL) {
+			new_member = create_queue_member(interface, penalty, paused);
 
-				if (new_member != NULL) {
-					new_member->dynamic = 1;
-					new_member->next = q->members;
-					q->members = new_member;
-					manager_event(EVENT_FLAG_AGENT, "QueueMemberAdded",
-						"Queue: %s\r\n"
-						"Location: %s\r\n"
-						"Membership: %s\r\n"
-						"Penalty: %d\r\n"
-						"CallsTaken: %d\r\n"
-						"LastCall: %ld\r\n"
-						"Status: %d\r\n"
-						"Paused: %d\r\n",
-					q->name, new_member->interface, new_member->dynamic ? "dynamic" : "static",
-					new_member->penalty, new_member->calls, new_member->lastcall, new_member->status, new_member->paused);
+			if (new_member != NULL) {
+				new_member->dynamic = 1;
+				new_member->next = q->members;
+				q->members = new_member;
+				manager_event(EVENT_FLAG_AGENT, "QueueMemberAdded",
+					"Queue: %s\r\n"
+					"Location: %s\r\n"
+					"Membership: %s\r\n"
+					"Penalty: %d\r\n"
+					"CallsTaken: %d\r\n"
+					"LastCall: %d\r\n"
+					"Status: %d\r\n"
+					"Paused: %d\r\n",
+				q->name, new_member->interface, new_member->dynamic ? "dynamic" : "static",
+				new_member->penalty, new_member->calls, (int)new_member->lastcall, new_member->status, new_member->paused);
 					
-					if (dump)
-						dump_queue_members(q);
+				if (dump)
+					dump_queue_members(q);
 
-					res = RES_OKAY;
-				} else {
-					res = RES_OUTOFMEMORY;
-				}
+				res = RES_OKAY;
 			} else {
-				res = RES_EXISTS;
+				res = RES_OUTOFMEMORY;
 			}
-			ast_mutex_unlock(&q->lock);
-			break;
+		} else {
+			res = RES_EXISTS;
 		}
 		ast_mutex_unlock(&q->lock);
 	}
@@ -3243,7 +3267,7 @@ static void reload_queues(void)
 			else
 				queues = q->next;
 			if (!q->count) {
-				free(q);
+				destroy_queue(q);
 			} else
 				ast_log(LOG_WARNING, "XXX Leaking a little memory :( XXX\n");
 		} else {
@@ -3272,7 +3296,13 @@ static int __queues_show(int manager, int fd, int argc, char **argv, int queue_s
 	time(&now);
 	if ((!queue_show && argc != 2) || (queue_show && argc != 3))
 		return RESULT_SHOWUSAGE;
+
+	/* We only want to load realtime queues when a specific queue is asked for. */
+	if (queue_show)
+		load_realtime_queue(argv[2]);
+
 	ast_mutex_lock(&qlock);
+
 	q = queues;
 	if (!q) {	
 		ast_mutex_unlock(&qlock);
@@ -3435,13 +3465,13 @@ static int manager_queues_status( struct mansession *s, struct message *m )
 						"Membership: %s\r\n"
 						"Penalty: %d\r\n"
 						"CallsTaken: %d\r\n"
-						"LastCall: %ld\r\n"
+						"LastCall: %d\r\n"
 						"Status: %d\r\n"
 						"Paused: %d\r\n"
 						"%s"
 						"\r\n",
 							q->name, mem->interface, mem->dynamic ? "dynamic" : "static",
-							mem->penalty, mem->calls, mem->lastcall, mem->status, mem->paused, idText);
+							mem->penalty, mem->calls, (int)mem->lastcall, mem->status, mem->paused, idText);
 				}
 			}
 			/* List Queue Entries */

apps/app_record.c

@@ -277,6 +277,7 @@ static int record_exec(struct ast_channel *chan, void *data)
 			
 			if (res) {
 				ast_log(LOG_WARNING, "Problem writing frame\n");
+				ast_frfree(f);
 				break;
 			}
 			
@@ -295,16 +296,15 @@ static int record_exec(struct ast_channel *chan, void *data)
 					break;
 				}
 			}
-		}
-		if (f->frametype == AST_FRAME_VIDEO) {
+		} else if (f->frametype == AST_FRAME_VIDEO) {
 			res = ast_writestream(s, f);
 			
 			if (res) {
 				ast_log(LOG_WARNING, "Problem writing frame\n");
+				ast_frfree(f);
 				break;
 			}
-		}
-		if ((f->frametype == AST_FRAME_DTMF) &&
+		} else if ((f->frametype == AST_FRAME_DTMF) &&
 		    (f->subclass == terminator)) {
 			ast_frfree(f);
 			break;

apps/app_skel.c

@@ -3,7 +3,7 @@
  *
  * Copyright (C) <Year>, <Your Name Here>
  *
- * <Your Name Here> <<You Email Here>>
+ * <Your Name Here> <<Your Email Here>>
  *
  * See http://www.asterisk.org for more information about
  * the Asterisk project. Please do not directly contact
@@ -20,7 +20,7 @@
  *
  * \brief Skeleton application
  * 
- * This is a skeleton for development of an Asterisk application */
+ * This is a skeleton for development of an Asterisk application
  * \ingroup applications
  */
 
@@ -77,7 +77,6 @@ static int app_exec(struct ast_channel *chan, void *data)
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "%s requires an argument (dummy|[options])\n",app);
-		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 

apps/app_sms.c

@@ -694,7 +694,7 @@ static void sms_readfile (sms_t * h, char *fn)
 		}
 		while (fgets (line, sizeof (line), s))
 		{								 /* process line in file */
-			char *p;
+			unsigned char *p;
 			for (p = line; *p && *p != '\n' && *p != '\r'; p++);
 			*p = 0;					 /* strip eoln */
 			p = line;
@@ -1379,8 +1379,8 @@ static int sms_exec (struct ast_channel *chan, void *data)
 		ast_copy_string (h.cli, chan->cid.cid_num, sizeof (h.cli));
 
 	{
-		char *d = data,
-			*p,
+		unsigned char *p;
+		unsigned char *d = data,
 			answer = 0;
 		if (!*d || *d == '|') {
 			ast_log (LOG_ERROR, "Requires queue name\n");
@@ -1449,7 +1449,7 @@ static int sms_exec (struct ast_channel *chan, void *data)
 			d = p;
 			h.udl = 0;
 			while (*p && h.udl < SMSLEN)
-				h.ud[h.udl++] = utf8decode((unsigned char **)&p);
+				h.ud[h.udl++] = utf8decode(&p);
 			if (is7bit (h.dcs) && packsms7 (0, h.udhl, h.udh, h.udl, h.ud) < 0)
 				ast_log (LOG_WARNING, "Invalid 7 bit GSM data\n");
 			if (is8bit (h.dcs) && packsms8 (0, h.udhl, h.udh, h.udl, h.ud) < 0)