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/1.2.2-netsec@212958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

.lastclean

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

.version

@@ -0,0 +1 @@
+1.2.2-netsec

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 +1,708 @@
+2006-01-18 Kevin P. Fleming <kpfleming@digium.com>
+
+	* Asterisk 1.2.2-netsec Released with Ranch Networks network security
+	device support
+
+2006-01-18 00:47 +0000 [r8162]  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
+
+2006-01-17 20:10 +0000 [r8140]  Matt O'Gorman <mogorman@digium.com>
+
+	* 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 [r8122]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* asterisk.c: update CLI copyright notice
+
+2006-01-16 23:51 +0000 [r8112]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* 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 [r7986]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_agent.c: move variable to correct scope (issue
+	  #6197)
+
+2006-01-11 07:18 +0000 [r7976]  Russell Bryant <russell@digium.com>
+
+	* apps/app_voicemail.c: fix temp greetings with ODBC storage (issue
+	  #6078)
+
+2006-01-11 05:46 +0000 [r7972]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_sip.c: fix mem leak on module unload (issue #6190)
+
+2006-01-11 05:26 +0000 [r7970]  Russell Bryant <russell@digium.com>
+
+	* 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)
+
+2006-01-11 04:53 +0000 [r7965]  Russell Bryant <russell@digium.com>
+
+	* 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 [r7960]  Russell Bryant <russell@digium.com>
+
+	* pbx.c: fix locking error - lock instead of unlock
+
+2006-01-11 03:12 +0000 [r7957]  Russell Bryant <russell@digium.com>
+
+	* 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 [r7908]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* pbx/pbx_spool.c: Bug 6157 - Memory leak
+
+2006-01-09 18:37 +0000 [r7904]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* 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 [r7871]  Russell Bryant <russell@digium.com>
+
+	* apps/app_voicemail.c: fix seg fault when using greek syntax in
+	  VoicemMailMain (issue #6142)
+
+2006-01-09 04:52 +0000 [r7870]  Russell Bryant <russell@digium.com>
+
+	* 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 [r7809]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_agent.c: send device state updates for auto-logoff
+	  of agents as well
+
+2006-01-05 00:18 +0000 [r7807]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* formats/format_pcm.c, formats/format_pcm_alaw.c: doh... fseek()
+	  has no useful return value
+
+2006-01-04 23:51 +0000 [r7805]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* formats/format_pcm.c, formats/format_pcm_alaw.c: use proper
+	  fwrite() parameters and return value
+
+2006-01-04 23:45 +0000 [r7803]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* formats/format_pcm.c, formats/format_pcm_alaw.c: return properly
+	  after extending file
+
+2006-01-04 23:27 +0000 [r7800]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that
+	  ulaw/alaw sound files are filled with silence when extended (not
+	  zeroes)
+
+2006-01-04 23:02 +0000 [r7799]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channel.c: make monitoring more tolerant of peers that deliver
+	  frames in bursts
+
+2006-01-04 21:46 +0000 [r7795]  Olle Johansson <oej@edvina.net>
+
+	* res/res_features.c: Issue #5980: Removing extra CR+LF in manager
+	  events - needs port to trunk
+
+2006-01-04 21:43 +0000 [r7792]  Olle Johansson <oej@edvina.net>
+
+	* 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 [r7740]  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
+
+2006-01-03 17:00 +0000 [r7738]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_sip.c: backport rport scanning fix from trunk (bug
+	  #6071)
+
+2006-01-03 16:34 +0000 [r7736]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* 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 [r7665]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_sip.c: fix memory leak in build_rpid (issue #6070)
+
+2005-12-27 21:07 +0000 [r7663]  Russell Bryant <russell@digium.com>
+
+	* 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 [r7599]  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
+
+2005-12-22 16:17 +0000 [r7595]  Russell Bryant <russell@digium.com>
+
+	* 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 [r7557]  Russell Bryant <russell@digium.com>
+
+	* res/res_agi.c: check array bounds when parsing arguments to AGI
+	  (issue #5868)
+
+2005-12-20 18:05 +0000 [r7552]  Russell Bryant <russell@digium.com>
+
+	* channels/chan_iax2.c: backport fix for reloading peer context
+	  (issue #6007)
+
+2005-12-20 17:34 +0000 [r7550]  Russell Bryant <russell@digium.com>
+
+	* 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 [r7523]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* file.c: Bug 5988 - record append option not working
+
+2005-12-19 05:38 +0000 [r7521]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* cdr.c: Bug 6026 - segfault for the sequence NoCDR(),
+	  SetAMAFlags()
+
+2005-12-17 18:55 +0000 [r7519]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* doc/README.ael: Document that curley braces must be on the same
+	  line as the keyword.
+
+2005-12-17 17:19 +0000 [r7517]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_chanspy.c: Bug 6009 - off by one error
+
+2005-12-17 03:59 +0000 [r7515]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_sip.c: Max-Forwards headers must only be present on
+	  requests, not responses
+
+2005-12-17 03:44 +0000 [r7513]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* 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)
+
+2005-12-17 02:20 +0000 [r7510]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* 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 [r7470]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* doc/README.variables: clarify substring documentation
+
+2005-12-13 16:06 +0000 [r7468]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* utils.c: correct broken math in tvfix() for timestamp values over
+	  one million
+
+2005-12-13 05:53 +0000 [r7453]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* apps/app_dial.c: restore ability of caller to hangup calls that
+	  are still ringing (issue #5839)
+
+2005-12-13 05:14 +0000 [r7451]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_sip.c, pbx.c: ensure that hangups while incoming
+	  calls are in early state are handled properly (issue #5919)
+
+2005-12-13 04:43 +0000 [r7449]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channels/chan_agent.c: only report AGENT_IDLE for callback mode
+	  agents when they are actually idle (issue #5902)
+
+2005-12-13 04:25 +0000 [r7448]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* 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)
+
+2005-12-12 17:30 +0000 [r7435]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* 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 [r7429]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* include/asterisk/linkedlists.h: Bug 5965 - major bug in
+	  AST_LIST_REMOVE
+
+2005-12-11 01:15 +0000 [r7427]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* 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 [r7406]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Bug 5960
+
+2005-12-08 23:32 +0000 [r7404]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+	* configs/res_odbc.conf.sample: Documenting two keywords that were
+	  previously missing
+
+2005-12-08 01:05 +0000 [r7386]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* pbx.c: initialize the buffer before using it...
+
+2005-12-07 20:46 +0000 [r7382]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* 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
+
+2005-12-06 00:19 +0000 [r7349]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* contrib/firmware/iax/iaxy.bin: replace with unmodified file
+
+2005-12-06 00:08 +0000 [r7346-7347]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* contrib/firmware/iax/iaxy.bin: _really_ don't mess with line
+	  endings
+
+	* contrib/firmware/iax/iaxy.bin: don't try to expand keywords in
+	  this file
+
+2005-12-06 00:02 +0000 [r7345]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* contrib/firmware/iax/iaxy.bin: set this file to binary mode, so
+	  it won't be mangled on export/checkout
+
+2005-12-05 06:47 +0000 [r7340]  Russell Bryant <russell@digium.com>
+
+	* Makefile: remove ASTERISKVERSIONNUM from the version string given
+	  to doxygen
+
+2005-12-05 00:09 +0000 [r7337]  Russell Bryant <russell@digium.com>
+
+	* apps/app_queue.c: don't delete dynamic queue members when
+	  reloading the static members from a realtime database (issue
+	  #5922)
+
+2005-12-05 00:01 +0000 [r7335]  Russell Bryant <russell@digium.com>
+
+	* 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 [r7274]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* apps/app_page.c, pbx.c: inherit channel variables into channels
+	  created by Page() application (issue #5888)
+
+2005-12-02 00:42 +0000 [r7273]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* apps/app_voicemail.c, configs/voicemail.conf.sample, UPGRADE.txt:
+	  allow previous context-searching behavior to be used if desired
+	  (issue #5899)
+
+2005-12-02 00:39 +0000 [r7272]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* 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)
+
+2005-12-02 00:28 +0000 [r7270-7271]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* 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)
+
+2005-12-01 23:49 +0000 [r7269]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* apps/app_festival.c: don't block waiting for the Festival server
+	  forever when it goes away (issue #5882)
+
+2005-12-01 23:34 +0000 [r7268]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channel.c: ensure channel's scheduling context is freed (issue
+	  #5788)
+
+2005-12-01 23:18 +0000 [r7266]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* 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 [r7265]  Olle Johansson <oej@edvina.net>
+
+	* doc/README.misdn: Changing bug report address to the Asterisk
+	  issue tracker
+
+2005-12-01 22:21 +0000 [r7263]  Olle Johansson <oej@edvina.net>
+
+	* doc/README.jitterbuffer, doc/README.realtime: Removing references
+	  to 1.1dev, replacing with 1.2, in documentation files.
+
+2005-12-01 21:38 +0000 [r7261]  Olle Johansson <oej@edvina.net>
+
+	* 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 [r7251]  Russell Bryant <russell@digium.com>
+
+	* apps/app_externalivr.c: use ast_app_separate_args to split
+	  arguments (issue #5686)
+
+2005-12-01 01:19 +0000 [r7249]  Russell Bryant <russell@digium.com>
+
+	* 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)
+
+2005-12-01 00:44 +0000 [r7247]  Russell Bryant <russell@digium.com>
+
+	* 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)
+
+2005-11-30 21:31 +0000 [r7244]  Russell Bryant <russell@digium.com>
+
+	* redhat/asterisk (removed), Makefile: remove outdated redhat init
+	  script and provide the updated one in 'make rpm' (issue #5786)
+
+2005-11-30 21:24 +0000 [r7242]  Russell Bryant <russell@digium.com>
+
+	* 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)
+
+2005-11-30 17:39 +0000 [r7234]  Russell Bryant <russell@digium.com>
+
+	* apps/app_dial.c: fix DIALEDTIME when call has not been answered
+	  (issue #5862)
+
+2005-11-30 14:27 +0000 [r7233]  Russell Bryant <russell@digium.com>
+
+	* rtp.c: do not allow an rtp message with zero type (issue #5749)
+
+2005-11-30 07:22 +0000 [r7231]  Russell Bryant <russell@digium.com>
+
+	* pbx.c: fix hint case sensitivity (issue #5856)
+
+2005-11-30 05:26 +0000 [r7228]  Russell Bryant <russell@digium.com>
+
+	* 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 [r7226]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* doc/cdr.txt, doc/CODING-GUIDELINES, include/asterisk.h,
+	  doc/README.mp3: remove remaining CVS references
+
+2005-11-30 03:20 +0000 [r7224]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* channel.c: port memory leak fix from rev 7223 in trunk
+
+2005-11-28 04:01 +0000 [r7208]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* Makefile, build_tools/make_svn_branch_name (added): port version
+	  string computation from trunk
+
+2005-11-27 16:04 +0000 [r7199]  Kevin P. Fleming <kpfleming@digium.com>
+
+	* / (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)

Makefile

@@ -105,6 +105,9 @@ BUSYDETECT+= #-DBUSYDETECT_TONEONLY
 # Don't use together with -DBUSYDETECT_TONEONLY
 BUSYDETECT+= #-DBUSYDETECT_COMPARE_TONE_AND_SILENCE
 
+# Comment this if you want to disable MIDCOM
+MIDCOM = -DMIDCOM
+
 ifneq ($(OSARCH),SunOS)
   ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk
   ASTVARLIBDIR=$(INSTALL_PREFIX)/var/lib/asterisk
@@ -331,6 +334,7 @@ ASTCFLAGS+= $(DEBUG_THREADS)
 ASTCFLAGS+= $(TRACE_FRAMES)
 ASTCFLAGS+= $(MALLOC_DEBUG)
 ASTCFLAGS+= $(BUSYDETECT)
+ASTCFLAGS+= $(MIDCOM)
 ASTCFLAGS+= $(OPTIONS)
 ASTCFLAGS+= -fomit-frame-pointer 
 SUBDIRS=res channels pbx apps codecs formats agi cdr funcs utils stdtime

README

@@ -164,6 +164,14 @@ parameters.  For example, the line "channel => 25-47" creates objects for
 the channels 25 through 47 of the card, obtaining the settings
 from the variables specified above.
 
+* NETWORK SECURITY DEVICES
+
+This release of Asterisk contains support for network security devices
+manufactured by Ranch Networks, Inc., using their MIDCOM interface
+library. You will need the companion libmidcom-0.1.0.tar.gz file to
+build the library. Contact Ranch Networks' support department for assistance
+in building and configuring MIDCOM support.
+
 * SPECIAL NOTE ON TIME
   
   Those using SIP phones should be aware that Asterisk is sensitive to

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

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;
 }

apps/app_dial.c

@@ -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"

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) {

apps/app_meetme.c

@@ -1630,7 +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_voicemail.c

@@ -445,7 +445,7 @@ static void apply_option(struct ast_vm_user *vmu, const char *var, const char *v
 		ast_copy_string(vmu->language, value, sizeof(vmu->language));
 	} else if (!strcasecmp(var, "tz")) {
 		ast_copy_string(vmu->zonetag, value, sizeof(vmu->zonetag));
-	} else if (!strcasecmp(var, "delete")) {
+	} else if (!strcasecmp(var, "delete") || !strcasecmp(var, "deletevoicemail")) {
 		ast_set2_flag(vmu, ast_true(value), VM_DELETE);	
 	} else if (!strcasecmp(var, "saycid")){
 		ast_set2_flag(vmu, ast_true(value), VM_SAYCID);	
@@ -2613,8 +2613,8 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, struct leave_vm_
 				}
 			}
 			if (ast_fileexists(fn, NULL, NULL)) {
-				notify_new_message(chan, vmu, msgnum, duration, fmt, chan->cid.cid_num, chan->cid.cid_name);
 				STORE(dir, vmu->mailbox, vmu->context, msgnum);
+				notify_new_message(chan, vmu, msgnum, duration, fmt, chan->cid.cid_num, chan->cid.cid_name);
 				DISPOSE(dir, msgnum);
 			}
 			pbx_builtin_setvar_helper(chan, "VMSTATUS", "SUCCESS");
@@ -4764,13 +4764,14 @@ static int vm_tempgreeting(struct ast_channel *chan, struct ast_vm_user *vmu, st
 	while((cmd >= 0) && (cmd != 't')) {
 		if (cmd)
 			retries = 0;
+		RETRIEVE(prefile, -1);
 		if (ast_fileexists(prefile, NULL, NULL) > 0) {
 			switch (cmd) {
 			case '1':
 				cmd = play_record_review(chan,"vm-rec-temp",prefile, maxgreet, fmtc, 0, vmu, &duration, NULL, record_gain);
 				break;
 			case '2':
-				ast_filedelete(prefile, NULL);
+				DELETE(prefile, -1, prefile);
 				ast_play_and_wait(chan,"vm-tempremoved");
 				cmd = 't';	
 				break;
@@ -4794,6 +4795,7 @@ static int vm_tempgreeting(struct ast_channel *chan, struct ast_vm_user *vmu, st
 			play_record_review(chan,"vm-rec-temp",prefile, maxgreet, fmtc, 0, vmu, &duration, NULL, record_gain);
 			cmd = 't';	
 		}
+		DISPOSE(prefile, -1);
 	}
 	if (cmd == 't')
 		cmd = 0;
@@ -5645,10 +5647,10 @@ static int vm_box_exists(struct ast_channel *chan, void *data)
 static int vmauthenticate(struct ast_channel *chan, void *data)
 {
 	struct localuser *u;
-	char *s = data, *user=NULL, *context=NULL, mailbox[AST_MAX_EXTENSION];
+	char *s = data, *user=NULL, *context=NULL, mailbox[AST_MAX_EXTENSION] = "";
 	struct ast_vm_user vmus;
 	char *options = NULL;
-	int silent = 0;
+	int silent = 0, skipuser = 0;
 	int res = -1;
 
 	LOCAL_USER_ADD(u);
@@ -5665,6 +5667,9 @@ static int vmauthenticate(struct ast_channel *chan, void *data)
 			s = user;
 			user = strsep(&s, "@");
 			context = strsep(&s, "");
+			if (!ast_strlen_zero(user))
+				skipuser++;
+			ast_copy_string(mailbox, user, sizeof(mailbox));
 		}
 	}
 
@@ -5672,9 +5677,10 @@ static int vmauthenticate(struct ast_channel *chan, void *data)
 		silent = (strchr(options, 's')) != NULL;
 	}
 
-	if (!vm_authenticate(chan, mailbox, sizeof(mailbox), &vmus, context, NULL, 0, 3, silent)) {
+	if (!vm_authenticate(chan, mailbox, sizeof(mailbox), &vmus, context, NULL, skipuser, 3, silent)) {
 		pbx_builtin_setvar_helper(chan, "AUTH_MAILBOX", mailbox);
 		pbx_builtin_setvar_helper(chan, "AUTH_CONTEXT", vmus.context);
+		ast_play_and_wait(chan, "auth-thankyou");
 		res = 0;
 	}
 

asterisk.c

@@ -127,7 +127,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 /*! \brief Welcome message when starting a CLI interface */
 #define WELCOME_MESSAGE \
-	ast_verbose("Asterisk " ASTERISK_VERSION ", Copyright (C) 1999 - 2005 Digium, Inc. and others.\n"); \
+	ast_verbose("Asterisk " ASTERISK_VERSION ", Copyright (C) 1999 - 2006 Digium, Inc. and others.\n"); \
 	ast_verbose("Created by Mark Spencer <markster@digium.com>\n"); \
 	ast_verbose("Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.\n"); \
 	ast_verbose("This is free software, with components licensed under the GNU General Public\n"); \
@@ -2138,6 +2138,10 @@ int main(int argc, char *argv[])
 		}
 	}
 
+	if (option_console && !option_verbose) 
+		ast_verbose("[ Reading Master Configuration ]");
+	ast_readconfig();
+
 	if (option_dumpcore) {
 		struct rlimit l;
 		memset(&l, 0, sizeof(l));
@@ -2148,10 +2152,6 @@ int main(int argc, char *argv[])
 		}
 	}
 
-	if (option_console && !option_verbose) 
-		ast_verbose("[ Reading Master Configuration ]");
-	ast_readconfig();
-
 	if ((!rungroup) && !ast_strlen_zero(ast_config_AST_RUN_GROUP))
 		rungroup = ast_config_AST_RUN_GROUP;
 	if ((!runuser) && !ast_strlen_zero(ast_config_AST_RUN_USER))

channel.c

@@ -576,7 +576,7 @@ struct ast_channel *ast_channel_alloc(int needqueue)
 	tmp->fds[AST_MAX_FDS-1] = tmp->alertpipe[0];
 	/* And timing pipe */
 	tmp->fds[AST_MAX_FDS-2] = tmp->timingfd;
-	strcpy(tmp->name, "**Unkown**");
+	strcpy(tmp->name, "**Unknown**");
 	/* Initial state */
 	tmp->_state = AST_STATE_DOWN;
 	tmp->streamid = -1;

channels/Makefile

@@ -97,6 +97,11 @@ ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/libpri.so.1)$(wildcard $(CROSS
   ZAPPRI=-lpri
 endif
 
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/asterisk/modules/res_netsec.so)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/lib/asterisk/modules/res_netsec.so),)
+  CFLAGS+=-DSIP_MIDCOM
+#  NETSEC=-lnetsec -lssl -lcrypto -lstdc++
+endif
+
 ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/libmfcr2.so.1)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/lib/libmfcr2.so.1),)
   CFLAGS+=-DZAPATA_R2
   ZAPR2=-lmfcr2
@@ -207,7 +212,7 @@ chan_zap.so: chan_zap.o
 	$(CC) $(SOLINK) -o $@ $<  $(ZAPPRI) $(ZAPR2) -ltonezone
 
 chan_sip.so: chan_sip.o
-	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} chan_sip.o ${CYGSOLIB}
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} chan_sip.o ${CYGSOLIB} ${NETSEC}
 
 chan_agent.so: chan_agent.o
 	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} chan_agent.o ${CYGSOLIB} ${CYG_CHAN_AGENT}

channels/chan_agent.c

@@ -102,7 +102,7 @@ static const char descrip2[] =
 static const char descrip3[] =
 "  AgentMonitorOutgoing([options]):\n"
 "Tries to figure out the id of the agent who is placing outgoing call based on\n"
-"comparision of the callerid of the current interface and the global variable \n"
+"comparison of the callerid of the current interface and the global variable \n"
 "placed by the AgentCallbackLogin application. That's why it should be used only\n"
 "with the AgentCallbackLogin app. Uses the monitoring functions in chan_agent \n"
 "instead of Monitor application. That have to be configured in the agents.conf file.\n"
@@ -144,7 +144,7 @@ static char moh[80] = "default";
 
 /** Persistent Agents astdb family */
 static const char pa_family[] = "/Agents";
-/** The maximum lengh of each persistent member agent database entry */
+/** The maximum length of each persistent member agent database entry */
 #define PA_MAX_LEN 2048
 /** queues.conf [general] option */
 static int persistent_agents = 0;
@@ -1051,7 +1051,7 @@ static int read_agent_config(void)
 	urlprefix[0] = '\0';
 	savecallsin[0] = '\0';
 
-	/* Read in [general] section for persistance */
+	/* Read in [general] section for persistence */
 	if ((general_val = ast_variable_retrieve(cfg, "general", "persistentagents")))
 		persistent_agents = ast_true(general_val);
 
@@ -1695,6 +1695,7 @@ static int __login_exec(struct ast_channel *chan, void *data, int callbackmode)
 	char agent_goodbye[AST_MAX_FILENAME_LEN];
 	int update_cdr = updatecdr;
 	char *filename = "agent-loginok";
+	char tmpchan[AST_MAX_BUF] = "";
 
 	LOCAL_USER_ADD(u);
 
@@ -1790,7 +1791,7 @@ static int __login_exec(struct ast_channel *chan, void *data, int callbackmode)
 				gettimeofday(&p->lastdisc, NULL);
 				p->lastdisc.tv_sec++;
 
-				/* Set Channel Specific Agent Overides */
+				/* Set Channel Specific Agent Overrides */
 				if (pbx_builtin_getvar_helper(chan, "AGENTACKCALL") && strlen(pbx_builtin_getvar_helper(chan, "AGENTACKCALL"))) {
 					if (!strcasecmp(pbx_builtin_getvar_helper(chan, "AGENTACKCALL"), "always"))
 						p->ackcall = 2;
@@ -1818,14 +1819,13 @@ static int __login_exec(struct ast_channel *chan, void *data, int callbackmode)
 					if (option_verbose > 2)
 						ast_verbose(VERBOSE_PREFIX_3 "Saw variable AGENTWRAPUPTIME=%s, setting wrapuptime to: %d for Agent '%s'.\n",tmpoptions,p->wrapuptime,p->agent);
 				}
-				/* End Channel Specific Agent Overides */
+				/* End Channel Specific Agent Overrides */
 				if (!p->chan) {
 					char last_loginchan[80] = "";
 					long logintime;
 					snprintf(agent, sizeof(agent), "Agent/%s", p->agent);
 
 					if (callbackmode) {
-						char tmpchan[AST_MAX_BUF] = "";
 						int pos = 0;
 						/* Retrieve login chan */
 						for (;;) {
@@ -2360,7 +2360,7 @@ static void reload_agents(void)
 	}
 	ast_mutex_unlock(&agentlock);
 	if (db_tree) {
-		ast_log(LOG_NOTICE, "Agents sucessfully reloaded from database.\n");
+		ast_log(LOG_NOTICE, "Agents successfully reloaded from database.\n");
 		ast_db_freetree(db_tree);
 	}
 }
@@ -2417,7 +2417,7 @@ static int agent_devicestate(void *data)
 
 /**
  * Initialize the Agents module.
- * This funcion is being called by Asterisk when loading the module. Among other thing it registers applications, cli commands and reads the cofiguration file.
+ * This function is being called by Asterisk when loading the module. Among other thing it registers applications, cli commands and reads the cofiguration file.
  *
  * @returns int Always 0.
  */

channels/chan_sip.c

@@ -86,6 +86,10 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/astosp.h"
 #endif
 
+#ifdef SIP_MIDCOM
+#include "asterisk/res_netsec.h"
+#endif
+
 #ifndef DEFAULT_USERAGENT
 #define DEFAULT_USERAGENT "Asterisk PBX"
 #endif
@@ -679,6 +683,10 @@ static struct sip_pvt {
 	
 	struct ast_dsp *vad;			/*!< Voice Activation Detection dsp */
 	
+#ifdef SIP_MIDCOM
+  void *r;
+#endif
+	
 	struct sip_peer *peerpoke;		/*!< If this calls is to poke a peer, which one */
 	struct sip_registry *registry;		/*!< If this is a REGISTER call, to which registry */
 	struct ast_rtp *rtp;			/*!< RTP Session */
@@ -922,6 +930,25 @@ static const struct cfsubscription_types *find_subscription_type(enum subscripti
 static int transmit_state_notify(struct sip_pvt *p, int state, int full, int substate);
 static char *gettag(struct sip_request *req, char *header, char *tagbuf, int tagbufsize);
 
+#ifdef SIP_MIDCOM
+static void sip_rtp_get_peer_audio_helper(void *p, struct sockaddr_in *them);
+static void sip_rtp_get_peer_video_helper(void *p, struct sockaddr_in *them);
+static void sip_rtp_get_us_audio_helper(void *p, struct sockaddr_in *sin);
+static void sip_rtp_get_us_video_helper(void *p, struct sockaddr_in *vsin);
+static void sip_map_hook_struct(void *p, void *r);
+static void *sip_get_hook_struct(void *p);
+static int sip_get_flag_novideo(void *p);
+static int sip_cmp_sa_addr(void *p, struct sockaddr_in *addr);
+static void sip_get_recv_addr(void *p, struct in_addr *addr);
+static char *sip_get_username(void *p);
+static struct ast_channel *sip_channel_helper(void *p);
+static struct ast_channel *sip_bridged_channel_helper(void *p);
+static int sip_get_capability_helper(void *p);
+static void sip_softhangup_helper(void *p);
+
+extern struct ast_sip_hook_cb *m_cb;
+#endif
+
 /*! \brief Definition of this channel for PBX channel registration */
 static const struct ast_channel_tech sip_tech = {
 	.type = channeltype,
@@ -2095,6 +2122,11 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner)
 	if (sip_debug_test_pvt(p))
 		ast_verbose("Destroying call '%s'\n", p->callid);
 
+#ifdef SIP_MIDCOM
+	if (m_cb)
+	  m_cb->__sip_destroy_hook(p);
+#endif
+
 	if (dumphistory)
 		sip_dump_history(p);
 
@@ -2419,6 +2451,12 @@ static int sip_hangup(struct ast_channel *ast)
 	if (ast->_state != AST_STATE_UP)
 		needcancel = 1;
 
+#ifdef SIP_MIDCOM
+        /* For callee to shutdown, send "BYE" instead of "CANCEL"
+           -- this needs to be verified */
+        if (m_cb && ast_test_flag(p, SIP_OUTGOING)) needcancel = 0;
+#endif
+
 	/* Disconnect */
 	p = ast->tech_pvt;
 	if (p->vad) {
@@ -4339,8 +4377,22 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
 		ast_rtp_get_us(p->vrtp, &vsin);
 
 	if (p->redirip.sin_addr.s_addr) {
+#ifdef SIP_MIDCOM
+	  if (m_cb && p->r) {
+	    struct sockaddr_in redirip_hook;
+	    char iabuf2[INET_ADDRSTRLEN];
+	    m_cb->ast_get_redirip_audio_hook(p->r, &redirip_hook);
+	    ast_log(LOG_DEBUG, "Replacing %s:%d by %s:%d in SDP before sending to %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->redirip.sin_addr), ntohs(p->redirip.sin_port), ast_inet_ntoa(iabuf2, sizeof(iabuf2), redirip_hook.sin_addr), ntohs(redirip_hook.sin_port), p->username);
+	    dest.sin_port = redirip_hook.sin_port;
+	    dest.sin_addr = redirip_hook.sin_addr;
+	  } else {
 		dest.sin_port = p->redirip.sin_port;
 		dest.sin_addr = p->redirip.sin_addr;
+	  }
+#else
+		dest.sin_port = p->redirip.sin_port;
+		dest.sin_addr = p->redirip.sin_addr;
+#endif
 		if (p->redircodecs)
 			capability = p->redircodecs;
 	} else {
@@ -4351,8 +4403,22 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
 	/* Determine video destination */
 	if (p->vrtp) {
 		if (p->vredirip.sin_addr.s_addr) {
+#ifdef SIP_MIDCOM
+		  if (m_cb && p->r) {
+		    struct sockaddr_in vredirip_hook;
+		    char iabuf2[INET_ADDRSTRLEN];
+		    m_cb->ast_get_vredirip_video_hook(p->r, &vredirip_hook);
+		    ast_log(LOG_DEBUG, "Replacing %s:%d by %s:%d in video SDP before sending to %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->vredirip.sin_addr), ntohs(p->vredirip.sin_port), ast_inet_ntoa(iabuf2, sizeof(iabuf2), vredirip_hook.sin_addr), ntohs(vredirip_hook.sin_port), p->username);
+		    vdest.sin_port = vredirip_hook.sin_port;
+		    vdest.sin_addr = vredirip_hook.sin_addr;
+		  } else {
 			vdest.sin_port = p->vredirip.sin_port;
 			vdest.sin_addr = p->vredirip.sin_addr;
+		  }
+#else
+			vdest.sin_port = p->vredirip.sin_port;
+			vdest.sin_addr = p->vredirip.sin_addr;
+#endif
 		} else {
 			vdest.sin_addr = p->ourip;
 			vdest.sin_port = vsin.sin_port;
@@ -4509,6 +4575,14 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
 	} else {
 		ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
 	}
+#ifdef SIP_MIDCOM
+	if (m_cb) {
+	  if (!m_cb->transmit_response_with_sdp_hook(p)) { 
+	    ast_log(LOG_NOTICE, "Failed transmit_response_with_sdp_hook()\n");
+	    return -1;
+	  }
+	}
+#endif
 	return send_response(p, &resp, retrans, seqno);
 }
 
@@ -4570,6 +4644,19 @@ static int determine_firstline_parts( struct sip_request *req )
 static int transmit_reinvite_with_sdp(struct sip_pvt *p)
 {
 	struct sip_request req;
+
+#ifdef SIP_MIDCOM
+	if (m_cb) {
+	  if (!m_cb->transmit_reinvite_with_sdp_hook(p)) { 
+	    ast_log(LOG_NOTICE, "Failed transmit_reinvite_with_sdp_hook()\n");
+	    if (p->owner)
+	      ast_queue_hangup(p->owner);
+	    else
+	      ast_set_flag(p, SIP_NEEDDESTROY);
+	  }
+	}
+#endif
+
 	if (ast_test_flag(p, SIP_REINVITE_UPDATE))
 		reqprep(&req, p, SIP_UPDATE, 0, 1);
 	else 
@@ -9467,6 +9554,16 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
 		p->authtries = 0;
 		if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
 			process_sdp(p, req);
+#ifdef SIP_MIDCOM
+			if (m_cb) {
+			  if (!m_cb->handle_response_invite_hook(p)) {
+			    if (p->owner)
+			      ast_queue_hangup(p->owner);
+			    else
+			      ast_set_flag(p, SIP_NEEDDESTROY);
+			  }
+			}
+#endif
 		}
 
 		/* Parse contact header for continued conversation */
@@ -10308,6 +10405,19 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 				ast_set_flag(p, SIP_NEEDDESTROY);	
 				return -1;
 			}
+#ifdef SIP_MIDCOM
+			if (m_cb) {
+			  if (!m_cb->handle_request_invite_hook((void *)p)) {
+			    ast_log(LOG_NOTICE, "Failed to NAT for (%s)\n", get_header(req, "From"));
+			    if (ignore)
+			      transmit_response(p, "403 Forbidden", req);
+			    else
+			      transmit_response_reliable(p, "403 Forbidden", req, 1);
+			    ast_set_flag(p, SIP_NEEDDESTROY);
+			    return 0;
+			  }
+			}
+#endif
 		} else {
 			p->jointcapability = p->capability;
 			ast_log(LOG_DEBUG, "Hm....  No sdp for the moment\n");
@@ -12655,8 +12765,13 @@ static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan)
 	if (!p)
 		return NULL;
 	ast_mutex_lock(&p->lock);
-	if (p->rtp && ast_test_flag(p, SIP_CAN_REINVITE))
+	if (p->rtp && ast_test_flag(p, SIP_CAN_REINVITE)) {
 		rtp =  p->rtp;
+#ifdef SIP_MIDCOM
+		if (m_cb)
+		  m_cb->ast_rtp_nat_us_audio_hook(rtp, p->r); /* change the ip port in rtp */
+#endif
+	}
 	ast_mutex_unlock(&p->lock);
 	return rtp;
 }
@@ -12671,8 +12786,13 @@ static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
 		return NULL;
 
 	ast_mutex_lock(&p->lock);
-	if (p->vrtp && ast_test_flag(p, SIP_CAN_REINVITE))
+	if (p->vrtp && ast_test_flag(p, SIP_CAN_REINVITE)) {
 		rtp = p->vrtp;
+#ifdef SIP_MIDCOM
+		if (m_cb)
+		  m_cb->ast_rtp_nat_us_video_hook(rtp, p->r); /* change the ip port in rtp */
+#endif
+	}
 	ast_mutex_unlock(&p->lock);
 	return rtp;
 }
@@ -12686,12 +12806,22 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
 	if (!p) 
 		return -1;
 	ast_mutex_lock(&p->lock);
-	if (rtp)
+		if (rtp) {
 		ast_rtp_get_peer(rtp, &p->redirip);
+#ifdef SIP_MIDCOM
+		if (m_cb)
+		  m_cb->ast_rtp_get_their_nat_audio_hook(rtp, p->r);
+#endif
+		}
 	else
 		memset(&p->redirip, 0, sizeof(p->redirip));
-	if (vrtp)
+		if (vrtp) {
 		ast_rtp_get_peer(vrtp, &p->vredirip);
+#ifdef SIP_MIDCOM
+		if (m_cb)
+		  m_cb->ast_rtp_get_their_nat_video_hook(vrtp, p->r);
+#endif
+		}
 	else
 		memset(&p->vredirip, 0, sizeof(p->vredirip));
 	p->redircodecs = codecs;
@@ -12959,6 +13089,26 @@ static struct ast_rtp_protocol sip_rtp = {
 	get_codec: sip_get_codec,
 };
 
+#ifdef SIP_MIDCOM
+/*! \brief  sip_helper: Interface structure with callbacks used to connect to midcom module --*/
+static struct ast_sip_helper_cb sip_helper = {
+	ast_rtp_get_peer_audio_helper: sip_rtp_get_peer_audio_helper,
+	ast_rtp_get_peer_video_helper: sip_rtp_get_peer_video_helper,
+	ast_rtp_get_us_audio_helper: sip_rtp_get_us_audio_helper,
+	ast_rtp_get_us_video_helper: sip_rtp_get_us_video_helper,
+	ast_map_hook_struct: sip_map_hook_struct,
+	ast_get_hook_struct: sip_get_hook_struct,
+	ast_get_flag_novideo: sip_get_flag_novideo,
+	ast_cmp_sa_addr: sip_cmp_sa_addr,
+	ast_get_recv_addr: sip_get_recv_addr,
+	ast_get_username: sip_get_username,
+	ast_channel_helper: sip_channel_helper,
+	ast_bridged_channel_helper: sip_bridged_channel_helper,
+	ast_get_capability_helper: sip_get_capability_helper,
+	ast_softhangup_helper: sip_softhangup_helper,
+};
+#endif
+
 /*! \brief  sip_poke_all_peers: Send a poke to all known peers */
 static void sip_poke_all_peers(void)
 {
@@ -13094,6 +13244,12 @@ int load_module()
 	/* Tell the RTP subdriver that we're here */
 	ast_rtp_proto_register(&sip_rtp);
 
+#ifdef SIP_MIDCOM
+	/* Register the sip helper functions */
+	if (m_cb)
+	  m_cb->ast_sip_helper_register(&sip_helper);
+#endif
+
 	/* Register dialplan applications */
 	ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode);
 
@@ -13142,6 +13298,12 @@ int unload_module()
 
 	ast_rtp_proto_unregister(&sip_rtp);
 
+#ifdef SIP_MIDCOM
+	/* Unregister the sip helper functions */
+	if (m_cb)
+	  m_cb->ast_sip_helper_unregister();
+#endif
+
 	ast_manager_unregister("SIPpeers");
 	ast_manager_unregister("SIPshowpeer");
 
@@ -13153,7 +13315,6 @@ int unload_module()
 				ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD);
 			p = p->next;
 		}
-		iflist = NULL;
 		ast_mutex_unlock(&iflock);
 	} else {
 		ast_log(LOG_WARNING, "Unable to lock the interface list\n");
@@ -13226,4 +13387,78 @@ char *description()
 	return (char *) desc;
 }
 
+#ifdef SIP_MIDCOM
+static void sip_rtp_get_peer_audio_helper(void *p, struct sockaddr_in *them)
+{
+  ast_rtp_get_peer(((struct sip_pvt*)p)->rtp, them);
+}
+
+static void sip_rtp_get_peer_video_helper(void *p, struct sockaddr_in *them)
+{
+  ast_rtp_get_peer(((struct sip_pvt*)p)->vrtp, them);
+}
+
+static void sip_rtp_get_us_audio_helper(void *p, struct sockaddr_in *sin)
+{
+  ast_rtp_get_us(((struct sip_pvt*)p)->rtp, sin);
+  sin->sin_addr = ((struct sip_pvt*)p)->ourip;
+}
+
+static void sip_rtp_get_us_video_helper(void *p, struct sockaddr_in *vsin)
+{
+  ast_rtp_get_us(((struct sip_pvt*)p)->vrtp, vsin);
+  vsin->sin_addr = ((struct sip_pvt*)p)->ourip;
+}
+
+static void sip_map_hook_struct(void *p, void *r)
+{
+  ((struct sip_pvt*)p)->r = r;
+}
+
+static void *sip_get_hook_struct(void *p)
+{
+  return ((struct sip_pvt*)p)->r;
+}
+
+static int sip_get_flag_novideo(void *p)
+{
+  return ast_test_flag((struct sip_pvt*)p, SIP_NOVIDEO);
+}
+
+static int sip_cmp_sa_addr(void *p, struct sockaddr_in *addr)
+{
+  return (((struct sip_pvt*)p)->sa.sin_addr.s_addr == addr->sin_addr.s_addr);
+}
+
+static void sip_get_recv_addr(void *p, struct in_addr *addr)
+{
+  memcpy(addr, &((struct sip_pvt *)p)->recv.sin_addr, sizeof(struct in_addr));
+}
+
+static char *sip_get_username(void *p)
+{
+  return ((struct sip_pvt*)p)->username;
+}
+
+static struct ast_channel *sip_channel_helper(void *p)
+{
+  return ((struct sip_pvt*)p)->owner;
+}
+
+static struct ast_channel *sip_bridged_channel_helper(void *p)
+{
+  return ast_bridged_channel(((struct sip_pvt*)p)->owner);
+}
+
+static int sip_get_capability_helper(void *p)
+{
+  return ((struct sip_pvt*)p)->jointcapability;
+}
+
+static void sip_softhangup_helper(void *p)
+{
+  if (p && ((struct sip_pvt *)p)->owner)
+    ast_softhangup(((struct sip_pvt *)p)->owner, AST_SOFTHANGUP_APPUNLOAD);
+}
+#endif
 

configs/voicemail.conf.sample

@@ -172,6 +172,7 @@ sendvoicemail=yes	; Context to Send voicemail from [option 5 from the advanced m
 			;     This does NOT affect option 3,3 from the advanced options menu
 ; delete=yes		; After notification, the voicemail is deleted from the server. [per-mailbox only]
 			;     This is intended for use with users who wish to receive their voicemail ONLY by email.
+			;     Note:  deletevoicemail is provided as an equivalent option for Realtime configuration.
 ; nextaftercmd=yes	; Skips to the next message after hitting 7 or 9 to delete/save current message.
 			;     [global option only at this time] 
 ; forcename=yes		; Forces a new user to record their name.  A new user is

doc/README.cdr

@@ -11,6 +11,8 @@ separated text file.
      for an updated list of supported databases, from MySQL to MsSQL
      and text files.
    * cdr_tds supports FreeTDS databases (Among them MS SQL)
+	NOTE: Please read README.tds for information on possible
+	problems with the FreeTDS driver
    * cdr_sqlite supports SQlite
    * cdr_pgsql supports PostgreSQL
 
@@ -39,28 +41,28 @@ channel configuration files, like sip.conf, iax.conf etc.
 Fields of the CDR in Asterisk
 -----------------------------
 
-   1. accountcode: What account number to use, (string, 20 characters)
-   2. src: Caller*ID number (string, 80 characters)
-   3. dst: Destination extension (string, 80 characters)
-   4. dcontext: Destination context (string, 80 characters)
-   5. clid: Caller*ID with text (80 characters)
-   6. channel: Channel used (80 characters)
-   7. dstchannel: Destination channel if appropriate (80 characters)
-   8. lastapp: Last application if appropriate (80 characters)
-   9. lastdata: Last application data (arguments) (80 characters)
-  10. start: Start of call (date/time)
-  11. answer: Answer of call (date/time)
-  12. end: End of call (date/time)
-  13. duration: Total time in system, in seconds (integer), from dial to hangup
-  14. billsec: Total time call is up, in seconds (integer), from answer to hangup
-  15. disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY
-  16. amaflags: What flags to use: DOCUMENTATION, BILL, IGNORE etc, 
-      specified on a per channel basis like accountcode.
-  17. user field: A user-defined field, maximum 255 characters 
+   1. accountcode:	What account number to use, (string, 20 characters)
+   2. src:		Caller*ID number (string, 80 characters)
+   3. dst:		Destination extension (string, 80 characters)
+   4. dcontext:		Destination context (string, 80 characters)
+   5. clid:		Caller*ID with text (80 characters)
+   6. channel:		Channel used (80 characters)
+   7. dstchannel:	Destination channel if appropriate (80 characters)
+   8. lastapp:		Last application if appropriate (80 characters)
+   9. lastdata:		Last application data (arguments) (80 characters)
+  10. start:		Start of call (date/time)
+  11. answer:		Answer of call (date/time)
+  12. end:		End of call (date/time)
+  13. duration:		Total time in system, in seconds (integer), from dial to hangup
+  14. billsec:		Total time call is up, in seconds (integer), from answer to hangup
+  15. disposition:	What happened to the call: ANSWERED, NO ANSWER, BUSY
+  16. amaflags:		What flags to use: DOCUMENTATION, BILL, IGNORE etc, 
+      			specified on a per channel basis like accountcode.
+  17. user field:	A user-defined field, maximum 255 characters 
 
 In some cases, uniqueid is appended:
 
-    * uniqueid: Unique Channel Identifier (32 characters) 
+    * uniqueid:		Unique Channel Identifier (32 characters) 
       This needs to be enabled in the source code at compile time
 
 

funcs/func_strings.c

@@ -48,8 +48,12 @@ static char *function_fieldqty(struct ast_channel *chan, char *cmd, char *data,
 	if (delim) {
 		varname = strsep(&delim, "|");
 		pbx_retrieve_variable(chan, varname, &varval, workspace, sizeof(workspace), NULL);
-		while (strsep(&varval, delim))
-			fieldcount++;
+		if (delim) {
+			while (strsep(&varval, delim))
+				fieldcount++;
+		} else if (!ast_strlen_zero(varval)) {
+			fieldcount = 1;
+		}
 		snprintf(buf, len, "%d", fieldcount);
 	} else {
 		ast_log(LOG_ERROR, "Out of memory\n");

include/asterisk/rtp.h

@@ -100,6 +100,11 @@ void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
 
 void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
 
+#ifdef MIDCOM
+void ast_rtp_nat_us(struct ast_rtp *rtp, struct sockaddr_in *our_nat);
+void ast_rtp_get_their_nat(struct ast_rtp *rtp, struct sockaddr_in *their_nat);
+#endif
+
 void ast_rtp_destroy(struct ast_rtp *rtp);
 
 void ast_rtp_reset(struct ast_rtp *rtp);

loader.c

@@ -457,8 +457,8 @@ static int ast_resource_exists(char *resource)
 static const char *loadorder[] =
 {
 	"res_",
-	"chan_",
 	"pbx_",
+	"chan_",
 	NULL,
 };
 

logger.c

@@ -461,8 +461,8 @@ int reload_logger(int rotate)
 
 	filesize_reload_needed = 0;
 
-	queue_log_init();
 	init_logger_chain();
+	queue_log_init();
 
 	if (logfiles.event_log) {
 		if (eventlog) {
@@ -589,12 +589,12 @@ int init_logger(void)
 	ast_cli_register(&rotate_logger_cli);
 	ast_cli_register(&logger_show_channels_cli);
 
-	/* initialize queue logger */
-	queue_log_init();
-
 	/* create log channels */
 	init_logger_chain();
 
+	/* initialize queue logger */
+	queue_log_init();
+
 	/* create the eventlog */
 	if (logfiles.event_log) {
 		mkdir((char *)ast_config_AST_LOG_DIR, 0755);

pbx.c

@@ -2029,7 +2029,7 @@ int ast_extension_state_del(int id, ast_state_cb_type callback)
 	    		cblist = cblist->next;
 		}
 
-		ast_mutex_lock(&hintlock);
+		ast_mutex_unlock(&hintlock);
 		return -1;
 	}
 

res/res_features.c

@@ -1280,10 +1280,12 @@ int ast_bridge_call(struct ast_channel *chan,struct ast_channel *peer,struct ast
 			if (!(monitor_app = pbx_findapp("Monitor")))
 				monitor_ok=0;
 		}
-		if ((monitor_exec = pbx_builtin_getvar_helper(chan, "AUTO_MONITOR"))) 
-			pbx_exec(chan, monitor_app, monitor_exec, 1);
-		else if ((monitor_exec = pbx_builtin_getvar_helper(peer, "AUTO_MONITOR")))
-			pbx_exec(peer, monitor_app, monitor_exec, 1);
+		if (monitor_app) {
+			if ((monitor_exec = pbx_builtin_getvar_helper(chan, "AUTO_MONITOR"))) 
+				pbx_exec(chan, monitor_app, monitor_exec, 1);
+			else if ((monitor_exec = pbx_builtin_getvar_helper(peer, "AUTO_MONITOR")))
+				pbx_exec(peer, monitor_app, monitor_exec, 1);
+		}
 	}
 	
 	set_config_flags(chan, peer, config);

rtp.c

@@ -125,6 +125,10 @@ struct ast_rtp {
 	int rtp_lookup_code_cache_code;
 	int rtp_lookup_code_cache_result;
 	int rtp_offered_from_local;
+
+#ifdef MIDCOM
+  struct sockaddr_in them_midcom_nat;
+#endif
 	struct ast_rtcp *rtcp;
 };
 
@@ -1041,6 +1045,20 @@ void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
 	memcpy(us, &rtp->us, sizeof(rtp->us));
 }
 
+#ifdef MIDCOM /* RANCH */
+void ast_rtp_nat_us(struct ast_rtp *rtp, struct sockaddr_in *our_nat)
+{
+  memcpy(&rtp->them_midcom_nat, our_nat, sizeof(rtp->them_midcom_nat));
+}
+
+void ast_rtp_get_their_nat(struct ast_rtp *rtp, struct sockaddr_in *their_nat)
+{
+  their_nat->sin_family = AF_INET;
+  their_nat->sin_port = rtp->them_midcom_nat.sin_port;
+  their_nat->sin_addr = rtp->them_midcom_nat.sin_addr;
+}
+#endif
+
 void ast_rtp_stop(struct ast_rtp *rtp)
 {
 	memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
@@ -1515,7 +1533,6 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
 	struct sockaddr_in t0, t1;
 	struct sockaddr_in vt0, vt1;
 	char iabuf[INET_ADDRSTRLEN];
-	
 	void *pvt0, *pvt1;
 	int codec0,codec1, oldcodec0, oldcodec1;
 	
@@ -1670,8 +1687,12 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
 			if (option_debug) {
 				ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n", 
 					c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t0.sin_addr), ntohs(t0.sin_port), codec0);
+			ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n", 
+				c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vt0.sin_addr), ntohs(vt0.sin_port), codec0);
 				ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", 
 					c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
+			ast_log(LOG_DEBUG, "Oooh, '%s' wasv %s:%d/(format %d)\n", 
+				c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vac0.sin_addr), ntohs(vac0.sin_port), oldcodec0);
 			}
 			if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
 				ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);

translate.c

@@ -331,6 +331,7 @@ static int show_translation(int fd, int argc, char *argv[])
 	if (argc > 4) 
 		return RESULT_SHOWUSAGE;
 
+	ast_mutex_lock(&list_lock);
 	if (argv[2] && !strcasecmp(argv[2],"recalc")) {
 		z = argv[3] ? atoi(argv[3]) : 1;
 
@@ -349,7 +350,6 @@ static int show_translation(int fd, int argc, char *argv[])
 
 	ast_cli(fd, "         Translation times between formats (in milliseconds)\n");
 	ast_cli(fd, "          Source Format (Rows) Destination Format(Columns)\n\n");
-	ast_mutex_lock(&list_lock);
 	for (x = -1; x < SHOW_TRANS; x++) {
 		/* next 2 lines run faster than using strcpy() */
 		line[0] = ' ';