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/1.2.0@212958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
remove extraneous svn:executable properties
2026-03-27 16:23:30 +00:00 · 2009-08-18 20:42:51 +00:00 · 2005-11-29 18:24:39 +00:00 · 2005-11-29 18:09:49 +00:00 · 2005-11-27 17:18:34 +00:00 · 2005-11-17 04:26:07 +00:00
612 changed files with 138095 additions and 55936 deletions
--- a/.cleancount
+++ b/.cleancount
@@ -0,0 +1 @@
+8
--- a/23
+++ b/23
@@ -1,23 +1,22 @@
-Asterisk 1.0-RC-1 Known Major Bugs
-==================================
-* Some people still have issues with H.323
-* QuickNet driver still not entirely stable
-
 Asterisk Bug Tracking Information
 =================================

-To learn about and report Asterisk bugs or make feature
-requests, please visit the official Asterisk Bug Tracker
-at:
+To learn about and report Asterisk bugs, please visit 
+the official Asterisk Bug Tracker at:

 	http://bugs.digium.com

 For more information on using the bug tracker, or to 
-learn how you can contribute by acting as a bug marshal
+learn how you can contribute by acting as a bug marshall
 please see:

-	http://www.digium.com/bugtracker.html
+	http://www.digium.com/index.php?menu=bugguidelines
+
+If you would like to submit a feature request, please
+resist the temptation to post it to the bug tracker.
+Feature requests should be posted to the asterisk-dev
+mailing list, located at:
+
+	http://lists.digium.com 

 Thank you!
-
-Mark
--- a/688
+++ b/688
@@ -1,573 +1,127 @@
- NOTE: Corrections or additions to the ChangeLog may be submitted 
-       to http://bugs.digium.com.  Documentation and formatting
-       fixes are not listed here.  A complete listing of changes
-       is available through the Asterisk-CVS mailing list hosted
-       at http://lists.digium.com.
+Changes since Asterisk 1.2.0-beta2:

- -- chan_zap
-    -- Asterisk will now also look in the regular context for the fax extension while
-       executing a macro.  Previously, for this to work, the fax extension would have
-       to be included in the macro definition.
-    -- On some systems, ALERTING will be sent after PROCEEDING, so code has been
-       added to account for this case.
-    -- If no extension is specified on an overlap call, the 's' extension will be used.
- -- chan_sip
-    -- We no longer send a "to" tag on "100 Trying" messages, as it is inappropriate
-       to do so.
-    -- We now respond correctly to an invite for T.38 with a "488 Not acceptable here"
-    -- We now discard saved tags on 401/407 responses in case the provider we're talking
-       to tries to pull a dirty trick on us and change it.
-    -- rtptimeout options will now be correctly set on a peer basis rather than only global
- -- chan_mgcp
-    -- Fixed setting of accountcode
-    -- Fixed where *67 to block callerid only worked for first call
- -- chan_agent
-    -- We now will not pass audio until the agent has acked the call if the configuration
-       is set up for the agent to do so.
- -- chan_alsa
-    -- Fixed problems with the unloading of this module
- -- res_agi
-    -- A fix has been added to prevent calls from being hung up when more than one
-       call is executing an AGI script calling the GET DATA command.
-    -- AGI scripts will now continue to run even if a file was not found with the
-       GET DATA command.
-    -- When calling SAY NUMBER with a number like 09, we will now say "nine" instead
-       of "zero"
- -- app_dial
-    -- There was a problem where text frames would not be forwarded before the channel
-       has been answered. 
- -- app_disa
-    -- Fixed the timeout used when no password is set
- -- app_queue
-    -- Distinctive ring has been fixed to work for queue members
-  -- rtp
-    -- Fixed a logic error when setting the "rtpchecksums" option
- -- say.c
-    -- A problem has been fixed with saying the date in Spanish.
- -- Makefile
-    -- A line was missing for the autosupport script that caused "make rpm" to fail
- -- format_wav_gsm
-    -- Fixed a problem with wav formatting that prevented files from being played
-       in some media players
- -- pbx_spool
-    -- Fixed if the last line of text in a file for the call spool did not contain
-       a new line, it would not be processed
- -- logger
-    -- Fixed the logger so that color escape sequences wouldn't be sent to the logs
- -- format_sln
-    -- A lot of changes were made to correctly handle signed linear format on
-       big endian machines
- -- asterisk.conf
-     -- fix 'highpriority' option for asterisk.conf
+   * Cygwin build system portability
+   * Optional generation of outbound silence during channel recording

-Asterisk 1.0.7
+Changes since Asterisk 1.2.0-beta1:

- -- chan_sip
-    -- The fix for some codec availibility issues in 1.0.6 caused music on hold problems,
-       but has now been fixed.
- -- chan_skinny
-    -- A check has been added to avoid a crash.
- -- chan_iax2
-    -- A feature has been added to CVS head to have the option of sending timestamps with
-       trunk frames.  It is not supported in 1.0, but a change has been made so that it
-       will at least not choke if sent trunk timestamps.
- -- app_voicemail
-    -- Some checks have been added to avoid a crash.
- -- speex
-    -- The path /usr/include/speex has been added for a place to look for the speex header.
+   * Many, many bug fixes
+   * Documentation and sample configuration updates
+   * Vastly improved presence/subscription support in the SIP channel driver
+   * A new (experimental) mISDN channel driver
+   * A new monitoring application (MixMonitor)
+   * More portability fixes for non-Linux platforms
+   * New dialplan functions replacing old applications
+   * Significant deadlock and performance upgrades for the Manager interface
+   * An upgrade to the 'new' dialplan expression parser for all users
+   * New Zaptel echo cancellers with improved performance
+   * Support for the latest OSP toolkit from TransNexus
+   * Support user-controlled volume adjustment in MeetMe application
+   * More dialplan applications now return status variables instead of priority jumping
+   * Much more powerful ENUM support in the dialplan
+   * SIP domain support for authentication and virtual hosting
+   * Many PRI protocol updates and fixes, including more complete Q.SIG support
+   * New applications: Pickup() and Page()

-Asterisk 1.0.6
+Changes since Asterisk 1.0:

- -- chan_iax2:
-    -- Fixed a bug dealing with a division by zero that could cause a crash
- -- chan_sip:
-    -- Behavior was changed so that when a registration fails due to DNS resolution issues,
-       a retry will be attempted in 20 seconds.
-    -- Peer settings were not reset to null values when reloading the configuration file.
-       Behavior has been changed so that these values are now cleared.
-    -- 'restrictcid' now properly works on MySQL peers.
-    -- Only use the default callerid if it has been specified.
-    -- Asterisk was not sending the same From: line in SIP messages during certain times.
-       Fixed to make sure it stays the same. This makes some providers happier, to a working state.
-    -- Certain circumstances involving a blank callerid caused asterisk to segmentation fault.
-    -- There was a problem incorrectly matching codec availablity when global preferences were 
-       different from that of the user.  To fix this, processing of SDP data has been moved 
-       to after determining who the call is coming from.
-    -- Asterisk would run out of RTP ports while waiting for SUBSCRIBE's to expire even though
-       an RTP port isn't needed in this case.  This has been fixed by releasing the ports early.
- -- chan_zap:
-    -- During a certain scenario when using flash and '#' transfers you would hear the
-       other person and the music they were hearing. This has been fixed.
-    -- A fix for a compilation issue with gcc4 was added.
- -- chan_modem_bestdata:
-    -- A fix for a compilation issue with gcc4 was added.
- -- format_g729:
-    -- Treat a 10-byte read as an end of file indication instead of an error. Some G729 encoders
-       like to put 10-bytes at the end to indicate this.
- -- res_features:
-    -- During certain situations when parking a call, both endpoints would get musiconhold.
-       This has been fixed so the individual who parked the call will hear the digits and not
-       musiconhold.
- -- app_dial:
-    -- DIALEDPEERNUMBER is now being set, so if you attempted to use it in the past and failed
-       it should work now.
-    -- A callerid change caused many headaches, this has been reversed to the original 1.0 behavior.
-    -- A crash caused with the combination of the 'g' option and # transfer was fixed.
- -- app_voicemail:
-    -- If two people hit the voicemail system at the same time, and were leaving a message
-       the second message was overwriting the first. This has been fixed so that each one
-       is distinct and will not overwrite eachother.
- -- cdr_tds:
-    -- If the server you were using was going down, it had the potential to bring your asterisk
-       server down with it. Extra stuff has been added so as to bring in more error/connection
-       checking.
- -- cdr_pgsql:
-    -- This will now attempt to reconnect after a connection problem.
- -- IAXY firmware:
-    -- This has been updated to version 23.  It includes a fix for lost registrations.
- -- internals
-    -- Behavior was changed for 'show codec <number>' to make it more intuitive. (kshumard)
-    -- DNS failures and asterisk do not get along too well, this is not totally the case anymore.
-    -- Asterisk will now handle DNS failures at startup more gracefully, and won't crash and
-       burn.
-    -- Choosing to append to a wave file would render the outputted wave file corrupt. Appending
-       now works again.
-    -- If you failed to define certain keys, asterisk had the potential to crash when seeing if you had
-       used them.
-    -- Attempting to use such things as ${EXTEN:-1} gave a wrong return value. However, this was never
-       a documented feature...
+This list currently only containts changes made from the end of November until
+March 26, 2005.

-Asterisk 1.0.5
-
- -- chan_zap
-    -- fix a callerid bug introduced in 1.0.4
- -- app_queue
-    -- fix some penalty behavior
-
-Asterisk 1.0.4
-
- -- general
-    -- fix memory leak evident with extensive use of variables
-    -- update IAXy firmware to version 22
-       -- enable some special write protection
-       -- enable outbound DTMF
-    -- fix seg fault with incorrect usage of SetVar
-    -- other minor fixes including typos and doc updates
- -- chan_sip
-   -- fix codecs to not be case sensitive
-   -- Re-use auth credentials
-   -- fix MWI when using type=friend
-   -- fix global NAT option
- -- chan_agent / chan_local
-   -- fix incorrect use count
- -- chan_zap
-   -- Allow CID rings to be configured in zapata.conf
-      -- no more patching needed for UK CID
- -- app_macro 
-    -- allow Macros to exit with '*' or '#' like regular extension processing
- -- app_voicemail
-   -- don't allow '#' as a password
-   -- add option to save voicemail before going to the operator
-   -- fix global operator=yes
- -- app_read
-   -- return 0 instead of -1 if user enters nothing
- -- res_agi
-    -- don't exit AGI when file not found to stream
-    -- send script parameter when using FastAGI
- 
-Asterisk 1.0.3
-
- -- chan_zap
-    -- fix seg fault when doing *0 to flash a trunk
- -- rtp
-    -- seg fault fix
- -- chan_sip
-    -- fix to prevent seg fault when attempting a transfer
-    -- fix bug with supervised transfers
-    -- fix codec preferences
- -- chan_h323
-    -- fix compilation problem
- -- chan_iax2
-   -- avoid a deadlock related to a static config of a BUNCH of peers
- -- cdr_pgsql
-    -- fix memory leak when reading config
- -- Numerous other minor bug fixes
-
-Asterisk 1.0.2
-
- -- Major bugfix release
-
-Asterisk 1.0.1
-
- -- Added AGI over TCP support
- -- Add ability to purge callers from queue if no agents are logged in
- -- Fix inband PRI indication detection
- -- Fix for MGCP - always request digits if no RTP stream
- -- Fixed seg fault for ast_control_streamfile
- -- Make pick-up extension configurable via features.conf 
- -- Numerous other bug fixes
-
-Asterisk 1.0.0
- -- Use Q.931 standard cause codes for asterisk cause codes
- -- Bug fixes from the bug tracker
-Asterisk 1.0-RC2 
- -- Additional CDR backends
- -- Allow muted to reconnect
- -- Call parking improvements (including SIP parking support)
- -- Added licensed hold music from FreePlayMusic
- -- GR-303 and Zap improvements
- -- More bug fixes from the bug tracker
- -- Improved FreeBSD/OpenBSD/MacOS X support
-Asterisk 1.0-RC1
- -- Innumerable bug fixes and features from the bug tracker
- -- Added Open Settlement Protocol (OSP) support
- -- Added Non-facility Associated Signalling (NFAS) Support
- -- Added alarm Monitoring support
- -- Added new MeetMe options
- -- Added GR-303 Support
- -- Added trunk groups
- -- ADPCM Standardization
- -- Numerous bug fixes
- -- Add IAX2 Firmware Support
- -- Add G.726 support
- -- Add ices/icecast support
- -- Numerous bug fixes
-Asterisk 0.7.2
- -- Countless small bug fixes from bug tracker
- -- DSP Fixes
- -- Fix unloading of Zaptel
- -- Pass Caller*ID/ANI properly on call forwarding
- -- Add indication for Italy
-Asterisk 0.7.1
- -- Fixed timed include context's and GotoIfTime
- -- Fixed chan_h323 it now gets remote ip properly instead of 127.0.0.1
-Asterisk 0.7.0
- -- Removed MP3 format and codec
- -- Can now load and unload SIP,IAX,IAX2,H323 channels without core
- -- Fixed various compiler warnings and clean up source tree
- -- Preliminary AES Support
- -- Fix SIP REINVITE
- -- Outbound SIP registration behind NAT using externip
- -- More CLI documentation and clean up
- -- Pin numbers on MeeMe
- -- Dynamic MeetMe conferences are more consistent with static conferences
- -- Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP}, ${ACCONTCODE}
- -- ODBC support for logging CDRs
- -- Indications for Norway and New Zeland
- -- Major redesign of app_voicemail
- -- Syslog support
- -- Reload logfiles with CLI command 'logger reload' and rotate logs with "logger rotate'
- -- Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now appear on remote console 
- -- Properly reaping any zombie processes
- -- Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail, ZapScan, Random, ResetCDR, NoCDR
- -- Make PRI Hangup Cause available to the dialplan
- -- Verify included contexts in extensions.conf
- -- Add DESTDIR support for building RPMs and packages
- -- Do route lookups on OpenBSD
- -- Add support for building on FreeBSD and OS X
- -- Add support for PostgreSQL in Voicemail
- -- Translate SIP hangup cause to PRI hangup cause where needed
- -- Better support for MOH in IAX2
- -- Fix SIP problem where channels were not removed on BYE
- -- Display codecs by name
- -- Remove MySQL and put PGSql instead for licensing reasons
- -- Better capability matching in SIP
- -- Full IBR4 compliance for chan_zap
- -- More flexible CDR handling
- -- Distinguish between BUSY and FAILURE on outbound calls
- -- Add initial support for SCCP via chan_skinny
- -- Better support for Future Group B signaling
-Asterisk 0.5.0
- -- Retain IAX2 and SIP registrations past shutdown/crash and restart
- -- True data mode bridging when possible
- -- H.323 build improvements
- -- Agent Callback-login support
- -- RFC2833 Improvements
- -- Add thread debugging
- -- Add optional pedantic SIP checking for Pingtel
- -- Allow extension names, include context, switch to use global vars.
- -- Allow variables in extensions.conf to reference previously defined ones
- -- Merge voicemail enhancements (app_voicemail2)
- -- Add multiple queueing strategies
- -- Merge support for 'T'
- -- Allow pending agent calling (Agent/:1)
- -- Add groupings to agents.conf
- -- Add video support to IAX2
- -- Zaptel optimize playback
- -- Add video support to SIP
- -- Make RTP ports configurable
- -- Add RDNIS support to SIP and IAX2
- -- Add transfer app (implement in SIP and IAX2)
- -- Make voicemail segmentable by context (app_voicemail2)
- -- Major restructuring of voicemail (app_voicemail2)
- -- Add initial ENUM support
- -- Add malloc debugging support
- -- Add preliminary Voicetronix support
- -- Add iLBC codec
-Asterisk 0.4.0
- -- Merge and edit Nick's FXO dial support
- -- Reengineer SIP registration (outbound)
- -- Support call pickup on SIP and compatibly with ZAP
- -- Support 302 Redirect on SIP
- -- Management interface improvements
- -- Add "hint" support
- -- Improve call forwarding using new "Local" channel driver.
- -- Add "Local" channel
- -- Substantial SIP enhancements including retransmissions
- -- Enforce case sensitivity on extension/context names
- -- Add monitor support (Thanks, Mahmut)
- -- Add experimental "trunk" option to IAX2 for high density VoIP
- -- Add experimental "debug channel" command
- -- Add 'C' flag to dial command to reset call detail record (handy for calling cards)
- -- Add NAT and dynamic support to MGCP
- -- Allow selection of in-band, out-of-band, or INFO based DTMF
- -- Add contributed "*80" support to blacklist numbers (Thanks James!)
- -- Add "NAT" option to sip user, peer, friend
- -- Add experimental "IAX2" protocol
- -- Change special variable "EXTEN-n" to "EXTEN:n" to follow Bash syntax
- -- Add "Enhanced" AGI with audio pass-through (voice recognition anyone?)
- -- Choose best priority from codec from allow/disallow
- -- Reject SIP calls to self
- -- Allow SIP registration to provide an alternative contact
- -- Make HOLD on SIP make use of asterisk MOH
- -- Add supervised transfer (tested with Pingtel only)
- -- Allow maxexpirey and defaultexpirey to be runtime configurable for SIP
- -- Preliminary codec 13 support (RFC3389)
- -- Add app_authenticate for general purpose authentication
- -- Optimize RTP and smoother
- -- Create special variable "EXTEN-n" where it is extension stripped by n MSD
- -- Fix uninitialized frame pointer in channel.c
- -- Add global variables support under [globals] of extensions.conf
- -- Add macro support (show application Macro)
- -- Allow [123-5] etc in extensions
- -- Allow format of App(arg1,arg2,...) instead of just App,arg1|arg2 in dialplan
- -- Add message waiting indicator to SIP
- -- Fix double free bug in channel.c
-Asterisk 0.3.0
- -- Add fastfoward, rewind, seek, and truncate functions to streams
- -- Support registration
- -- Add G729 format
- -- Permit applications to return a digit indicating new extension
- -- Change "SHUTDOWN" to "STOP" in commands
- -- SIP "Hold" fixes and VXML URI support
- -- New chan_zap with 160 sample chunk size
- -- Add DTMF, MF, and Fax tone detector to dsp routines
- -- Allow overlap dialing (inbound) on PRI
- -- Enable tone detection with PRI
- -- Add special information tone detection
- -- Add Asterisk DB support
- -- Add pulse dialing
- -- Re-record all system prompts
- -- Change "timelen" to samples for better accuracy
- -- Move to editline, eliminating readline dependency
- -- Add peer "poke" support to SIP and IAX
- -- Add experimental call progress detection
- -- Add SIP authentication (digest)
- -- Add RDNIS
- -- Reroute faxes to "fax" extension
- -- Create ISDN/modem group concept
- -- Centralize indication
- -- Add initial MGCP support
- -- SIP debugging cleanup
- -- SIP reload
- -- SIP commands (show channels, etc)
- -- Add optional busy detection
- -- Add Visual Message Waiting Indicator (MDMF and SDMF)
- -- Add ambiguous extension matching
- -- Add *69
- -- Major SIP enhancements from SIPit
- -- Rewrite of ZAP CLASS features using subchannels
- -- Enhanced call parking
- -- Add extended outgoing spool support (pbx_spool)
-Asterisk 0.2.0
- -- Outbound origination API
- -- Call management improvements
- -- Add Do Not Disturb (*78, *79)
- -- Add agents
- -- Document variables
- -- Add transfer capability on the console
- -- Add SpeeX codec translator
- -- Add call queues
- -- Add setcallerid functionality (AGI, application)
- -- Add special variables ${CALLERID}, ${EXTEN}, ${CONTEXT}, ${PRIORITY}
- -- Don't echo cancel on pure TDM connections by default
- -- Implement Async GOTO
- -- Differentiate softhangups
- -- Add date/time
-Asterisk 0.1.12
- -- Fix for Big Endian machines
- -- MySQL CDR Engine
- -- Various SIP fixes and enhancements
- -- Add "zapateller application and arbitrary tone pairs
- -- Don't always start at "s"
- -- Separate linear mode for pseudo and real
- -- Add initial RTP and SIP support (no jitter buffer yet, unknown stability)
- -- Add 'h' extension, executed on hangup
- -- Add duration timer to message info
- -- Add web based voicemail checking ("make webvmail")
- -- Add ast_queue_frame function and eliminate frame pipes in most drivers
- -- Centralize host access (and possibly future ACL's)
- -- Add Caller*ID on PhoneJack (Thanks Nathan)
- -- Add "safe_asterisk" wrapper script to auto-restart Asterisk
- -- Indicate ringback on chan_phone
- -- Add answer confirmation (press '#' to confirm answer)
- -- Add distinctive ring support (e.g. Dial,Zap/4r2)
- -- Add ANSI/vt100 color support
- -- Make parking configurable through parking.conf
- -- Fix the empty voicemail problem
- -- Add Music On Hold
- -- Add ADSI Compiler (app_adsiprog)
- -- Extensive DISA re-work to improve tone generation
- -- Reset all idle channels every 10 minutes on a PRI
- -- Reset channels which are hungup with "channel in use"
- -- Implement VNAK support in chan_iax
- -- Fix chan_oss to support proper hangups and autoanswer
- -- Make shutdown properly hangup channels
- -- Add idling capability to chan_zap for idle-net
- -- Add "MeetMe" conferencing app (app_meetme)
- -- Add timing information to include
-Asterisk 0.1.11
- -- Add ISDN RAS capability
- -- Add stutter dialtone to Chan Zap
- -- Add "#include" capability to config files.
- -- Add call-forward variable to Chan Zap (*72, *73)
- -- Optimize IAX flow when transfer isn't possible
- -- Allow transmission of ANI over IAX
-Asterisk 0.1.10
- -- Make ast_readstring parameter be the max # of digits, not the max size with \0
- -- Make up any missing messages on the fly
- -- Add support for specific DTMF interruption to saying numbers
- -- Add new "u" and "b" options to condense busy/unavail handling
- -- Add support for RSA authentication on IAX calls
- -- Add support for ADSI compatible CPE
- -- Outgoing call queue
- -- Remote dialplan fixes for Quicknet
- -- Added AGI commands supporting TDD functions (RECEIVE CHAR & TDD MODE)
- -- Added TDD support (send/receive text in chan_zap)
- -- Fix all strncpy references
- -- Implement CSV CDR backend
- -- Implement Call Detail Records
-Asterisk 0.1.9
- -- Implement IAX quelching
- -- Allow Caller*ID to be overridden and suggested
- -- Configure defaults to use IAXTEL
- -- Allow remote dialplan polling via IAX
- -- Eliminate ast_longest_extension
- -- Implement dialplan request/reply
- -- Let peers have allow/disallow for codecs
- -- Change allow/deny to permit/deny in IAX
- -- Allow dialplan entries to match Caller*ID as well
- -- Added AGI (Asterisk Gateway Interface) scripting interface (app_agi)
- -- Added chan_zap for zapata telephony kernel interface, removed chan_tor
- -- Add convenience functions
- -- Fix race condition in channel hangup
- -- Fix memory leaks in both asterisk and iax frame allocations
- -- Add "iax show stats" command and -DTRACE_FRAMES (for frame tracing)
- -- Add DISA application (Thanks to Jim Dixon)
- -- Add IAX transfer support
- -- Add URL and HTML transmission
- -- Add application for sending images
- -- Add RedHat RPM spec file and build capability
- -- Fix GSM WAV file format bug
- -- Move ignorepat to main dialplan
- -- Add ability to specificy TOS bits in IAX
- -- Allow username:password in IAX strings
- -- Updates to PhoneJack interface
- -- Allow "servermail" in voicemail.conf to override e-mail in "from" line
- -- Add 'skip' option to app_playback
- -- Reject IAX calls on unknown extensions
- -- Fix version stuff
-Asterisk 0.1.8
- -- Keep track of version information
- -- Add -f to cause Asterisk not to fork
- -- Keep important information in voicemail .txt file
- -- Adtran Voice over Frame Relay updates
- -- Implement option setting/querying of channel drivers
- -- IAX performance improvements and protocol fixes
- -- Substantial enhancement of console channel driver
- -- Add IAX registration.  Now IAX can dynamically register
- -- Add flash-hook transfer on tormenta channels
- -- Added Three Way Calling on tormenta channels
- -- Start on concept of zombie channel
- -- Add Call Waiting CallerID
- -- Keep track of who registeres contexts, includes, and extensions
- -- Added Call Waiting(tm), *67, *70, and *82 codes
- -- Move parked calls into "parkedcalls" context by default
- -- Allow dialplan to be displayed
- -- Allow "=>" instead of just "=" to make instantiation clearer
- -- Asterisk forks if called with no arguments
- -- Add remote control by running asterisk -vvvc
- -- Adjust verboseness with "set verbose" now
- -- No longer requires libaudiofile
- -- Install beep
- -- Make PBX Config module reload extensions on SIGHUP
- -- Allow modules to be reloaded when SIGHUP is received
- -- Variables now contain line numbers
- -- Make dialer send in band signalling
- -- Add record application
- -- Added PRI signalling to Tormenta driver
- -- Allow use of BYEXTENSION in "Goto"
- -- Allow adjustment of gains on tormenta channels
- -- Added raw PCM file format support
- -- Add U-law translator
- -- Fix DTMF handling in bridge code
- -- Fix access control with IAX
-* Asterisk 0.1.7
- -- Update configuration files and add some missing sounds
- -- Added ability to include one context in another
- -- Rewrite of PBX switching
- -- Major mods to dialler application
- -- Added Caller*ID spill reception
- -- Added Dialogic VOX file format support
- -- Added ADPCM Codec
- -- Add Tormenta driver (RBS signalling)
- -- Add Caller*ID spill creation
- -- Rewrite of translation layer entirely
- -- Add ability to run PBX without additional thread
-* Asterisk 0.1.6
- -- Make app_dial handle a lack of translators smoothly
- -- Add ISDN4Linux support -- dtmf is weird...
- -- Minor bug fixes
-* Asterisk 0.1.5
- -- Fix a small mistake in IAX
- -- Fix the QuickNet driver to work with newer cards
-* Asterisk 0.1.4
- -- Update VoFR some more
- -- Fix the QuickNet driver to work with LineJack
- -- Add ability to pass images for IAX.
-* Asterisk 0.1.3
- -- Update VoFR for latest sangoma code
- -- Update QuickNet Driver
- -- Add text message handling
- -- Fix transfers to use "default" if not in current context
- -- Add call parking
- -- Improve format/content negotiation
- -- Added support for multiple languages
- -- Bug fixes, as always...
-* Asterisk 0.1.2
- -- Updated README file with a "Getting Started" section
- -- Added sample sounds and configuration files.
- -- Added LPC10 very low bandwidth (low quality) compression
- -- Enhanced translation selection mechanism.
- -- Enhanced IAX jitter buffer, improved reliability
- -- Support echo cancelation on PhoneJack
- -- Updated PhoneJack driver to std. Telephony interface
- -- Added app_echo for evaluating VoIP latency
- -- Added app_system to execute arbitrary programs
- -- Updated sample configuration files
- -- Added OSS channel driver (full duplex only)
- -- Added IAX implementation
- -- Fixed some deadlocks.
- -- A whole bunch of bug fixes
-* Asterisk 0.1.1
- -- Revised translator, fixed some general race conditions throughout *
- -- Made dialer somewhat more aware of incompatible voice channels
- -- Added Voice Modem driver and A/Open Modem Driver stub
- -- Added MP3 decoder channel
- -- Added Microsoft WAV49 support
- -- Revised License -- Pure GPL, nothing else
- -- Modified Copyright statement since code is still currently owned by author
- -- Added RAW GSM headerless data format
- -- Innumerable bug fixes
-* Asterisk 0.1.0
- -- Initial Release
+   * Add new applications:
+     -- AgentMonitorOutgoing
+     -- Curl
+     -- ExecIf
+     -- ExecIfTime
+     -- IAX2Provision
+     -- MacroExit
+     -- MacroIf
+     -- PauseQueueMember
+     -- ReadFile
+     -- SetRDNIS
+     -- SIPAddHeader
+     -- SIPGetHeader
+     -- StartMusicOnHold
+     -- StopMusicOnHold
+     -- UnpauseQueueMember
+     -- WaitForSilence
+     -- While / EndWhile
+   * app Answer
+     -- added delay option
+   * app ChanIsAvail
+     -- added 's' option
+   * app Dial
+     -- add option to specify the class for musiconhold with m option
+   * app EnumLookup
+     -- added "reload enum" for configuration
+   * app Goto
+     -- added relative priorities
+   * app GotoIf
+     -- added relative priorities
+   * app MeetMe
+     -- added 'i' option
+     -- added 'r' option
+     -- added 'T' option
+     -- added 'P' option
+     -- added 'c' option
+     -- added adminpin to meetme.conf
+     -- added reload command
+   * app PrivacyManager
+     -- add config file privacy.conf
+   * app queue
+     -- queues.conf
+        -- added persistentmembers option to queues.conf
+        -- changed music option to musiconhold
+        -- added weight option
+        -- added note about why agent groups probably shouldn't be used
+        -- added timeoutrestart option
+   * app Read
+     -- added attempts parameter
+     -- added timeout parameter
+   * app Record
+     -- added 'q' option
+   * app SendDTMF
+     -- add timeout option
+   * app SMS
+     -- document alternative syntax for queueing messages
+   * app Voicemail
+     -- add info about VM_CATEGORY
+     -- voicemail.conf
+        -- added usedirectory option
+        -- added VM_CIDNUM and VM_CIDNAME in message config
+   * chan IAX2
+     -- new jitterbuffer
+     -- added setvar option
+     -- added regex to iax2 show peers/users
+     -- allow multiple bindaddr lines in iax.conf
+     -- added reload command
+     -- added forcejitterbuffer option
+     -- added note about specifying bindport before bindaddr
+     -- added trunktimestamps option
+   * chan Agent
+     -- added agent logoff CLI command
+   * chan OSS
+     -- added Flash CLI command
+   * chan SIP
+     -- added setvar option
+     -- added compactheaders option
+     -- added usereqphone option
+     -- added registertimeout option
+     -- added externhost option
+     -- added sip notify CLI command
+     -- added sip_notify.conf
+     -- added allowguest option
+   * chan Zap
+     -- added hanguponplarityswitch option
+     -- added sendcalleridafter option
+     -- added priresetinterval option
+     -- added TON/NPI config options (the ones right above the resetinterval option)
+     -- added answeronpolarityswitch option
+     -- added "never" for resetinterval
+   * extensions
+     -- allow '*' when including files (#include "sip-*.conf")
+     -- added eswitch
+   * General
+     -- added #exec syntax for including output from a command
+     -- added show features CLI command
+     -- added configuration templates for category inheritance
--- a/341
+++ b/341
@@ -0,0 +1,341 @@
+
+		    GNU GENERAL PUBLIC LICENSE
+		       Version 2, June 1991
+
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+ 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
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+
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+possible use to the public, the best way to achieve this is to make it
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+
+  <signature of Ty Coon>, 1 April 1989
+  Ty Coon, President of Vice
+
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+Public License instead of this License.
--- a/50
+++ b/50
@@ -1,6 +1,6 @@

 === DEVELOPMENT SUPPORT ===
-I'd like to thank the following companies for helping fund development of
+We'd like to thank the following companies for helping fund development of
 Asterisk:

 Pilosoft, Inc. - for supporting ADSI development in Asterisk
@@ -26,50 +26,70 @@ Wasim - Hangup detect
 PhoneJack and Linejack card to the project.  (http://www.quicknet.net)

 === MISCELLANEOUS PATCHES ===
-Anthony Minessale - Countless big and small fixes, and relentless forward push
-       anthmct@yahoo.com              http://www.asterlink.com
+Jim Dixon - Zapata Telephony and app_rpt
+	http://www.zapatatelephony.org/app_rpt.html
+Russell Bryant - Asterisk 1.0 maintainer and misc. enhancements
+	russelb@clemson.edu
+Anthony Minessale II - Countless big and small fixes, and relentless forward push
+	anthmct@yahoo.com              http://www.asterlink.com
 James Golovich - Innumerable contributions
-       You can find him and asterisk-perl at http://asterisk.gnuinter.net
+	You can find him and asterisk-perl at http://asterisk.gnuinter.net
 Andre Bierwirth - Extension hints and status
 Oliver Daudey - ISDN4Linux fixes
 Pauline Middelink - ISDN4Linux patches and some general patches.
-       She can be found at http://www.polyware.nl/~middelink/En/
+	She can be found at http://www.polyware.nl/~middelink/En/
 Jean-Denis Girard - Various contributions from the South Pacific Islands
-       jd-girard@esoft.pf             http://www.esoft.pf
-William Jordan / Vonage - MySQL enhancmenets to Voicemail
+	jd-girard@esoft.pf             http://www.esoft.pf
+William Jordan / Vonage - MySQL enhancements to Voicemail
 	wjordan@vonage.com
 Jac Kersing - Various fixes
 Steven Critchfield - Seek and Trunc functions for playback and recording
-       critch@basesys.com
+	critch@basesys.com
 Jefferson Noxon - app_lookupcidname, app_db, and various other contributions
 Klaus-Peter Junghanns - in-band DTMF on SIP and MGCP
 Ross Finlayson - Dynamic RTP payload support
 Mahmut Fettahlioglu - Audio recording, music-on-hold changes, alaw file
-       format, and various fixes. Can be contacted at mahmut@oa.com.au
+	format, and various fixes. Can be contacted at mahmut@oa.com.au
 James Dennis - Cisco SIP compatibility patches to work with SIP service
-       providers. Can be contacted at asterisk@jdennis.net
-Tilghman Lesher - - Route lookup code; ast_localtime(), ast_say_date_with_format(); 
-                    GotoIfTime, Random, SayUnixTime, HasNewVoicemail, and Cut applications, 
-		    along with various other patches. http://asterisk.drunkcoder.com/
+	providers. Can be contacted at asterisk@jdennis.net
+Tilghman Lesher - ast_localtime(); ast_say_date_with_format(); 
+	GotoIfTime, Random, SayUnixTime, HasNewVoicemail applications;
+	CUT, SORT, EVAL, CURL, FIELDQTY, STRFTIME, QUEUEAGENT* functions;
+	and other innumerable bug fixes. http://asterisk.drunkcoder.com/
 Jayson Vantuyl - Manager protocol changes, various other bugs.
 	jvantuyl@computingedge.net
 Thorsten Lockert - OpenBSD, FreeBSD ports, making MacOS X port run on 10.3,
 	dialplan include verification, route lookup on OpenBSD, various other
 	bugs. tholo@sigmasoft.com
 Brian West - ODBC support and Bug Marshaling  
+Josh Roberson - chan_zap reload support, Advanced Voicemail Features, other misc. patches,
+	and Bug Marshalling. - josh@asteriasgi.com, http://www.asteriasgi.com
 William Waites - syslog support, SIP NAT traversal for SIP-UA. ww@styx.org
 Rich Murphey - Porting to FreeBSD, NetBSD, OpenBSD, and Darwin.
 	rich@whiteoaklabs.com  http://whiteoaklabs.com
+Simon Lockhart - Porting to Solaris (based on work of Logan ???)	
+	simon@slimey.org
+Olle E. Johansson - SIP RFC compliance, documentation and testing, testing, testing
+	oej@edvina.net, http://edvina.net
+Steve Kann - new jitter buffer for IAX2
+	stevek@stevek.com
+Constantine Filin - major contributions to the Asterisk Realtime Architecture
+Steve Murphy - privacy support
+Claude Patry - bug fixes, feature enhancements, and bug marshalling
+	cpatry@gmail.com

 === OTHER CONTRIBUTIONS ===
 John Todd - Monkey sounds and associated teletorture prompt
+Michael Jerris - bug marshaling
+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.opsound.org

 === OTHER SOURCE CODE IN ASTERISK ===

-I did not implement the codecs in asterisk.  Here is the copyright on the
+Digium did not implement the codecs in Asterisk.  Here is the copyright on the
 GSM source:

 Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
@@ -94,7 +114,7 @@ And the copyright on the ADPCM source:
 Copyright 1992 by Stichting Mathematisch Centrum, Amsterdam, The
 Netherlands.

-                        All Rights Reserved
+	                 All Rights Reserved

 Permission to use, copy, modify, and distribute this software and its
 documentation for any purpose and without fee is hereby granted,
--- a/372
+++ b/372
@@ -0,0 +1,372 @@
+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
+	* channels/chan_skinny:  Don't register channel type until ready, code formatting updates
+
+2005-11-16  Josh Roberson <josh@asteriasgi.com>
+
+	* Makefile:  Update to fix non-responsive remote console on Darwin (OSX)(issue #5757)
+
+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)
+
+	* channel.c (ast_queue_hangup): ensure that the channel lock is held before changing its fields... (issue #5770)
+
+	* res/res_musiconhold.c: don't spit out incorrect log messages (and leak memory) during reload (issue #5766)
+
+	* channels/chan_sip.c (process_sdp): don't pass video codec number into ast_getformatname(), it is not valid input for that function (issue #5764)
+
+	* pbx/pbx_ael.c (match_assignment): properly parse equal signs surrounded by whitespace (issue #5761)
+
+	* doc/README.realtime: document the limitations of using FreeTDS with Realtime (issue #5767)
+
+2005-11-15  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* Makefile: use -g3 for compiler to include macro information for debugger
+
+	* astmm.c (__ast_vasprintf): don't re-use the ap list without copying it; that's not safe on some platforms (issue #5035)
+
+	* doc/README.backtrace: add note about properly building Asterisk to be able to produce backtraces; wrap text and remove DOS line endings
+
+	* channels/chan_sip.c (add_codec_to_sdp): add 'annexb=no' to G.729A SDP (issue #5539)
+
+	* channels/chan_alsa.c (alsa_hangup): handle autohangup properly (issue #5672)
+
+	* channels/chan_misdn.c (and other files): various fixes (issue #5739)
+
+	* channels/chan_sip.c (handle_request_info): properly forward 'flash' events received via SIP INFO (issue #5751, different patch)
+
+	* apps/app_disa.c (disa_exec): don't duplicate constant strings when not needed
+
+	* apps/app_playback.c (playback_exec): use correct logic tests for options (issue #5752)
+
+	* apps/app_disa.c (disa_exec): use standard arg parsing routines (issue #5736)
+
+2005-11-15  Russell Bryant <russell@digium.com>
+
+	* manager.c: Don't crash on a SetVar action if the channel name is not set, or variable's value is not set (issue #5760)
+
+	* doc/README.variables: Add application exit status variables
+
+2005-11-14  Josh Roberson <josh@asteriasgi.com>
+	
+	* manager.c: Fix crash on variable passing from AMI originate (issue #5737)
+
+2005-11-14  Russell Bryant  <russell@digium.com>
+
+	* many files: Merge doxygen documentation updates. (issue #5605)
+
+	* apps/app_dial.c: Fix typo in RetryDial description.
+
+2005-11-12  Russell Bryant  <russell@digium.com>
+
+	* channels/chan_oss.c: Fix a typo in an error message.
+
+2005-11-11  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* Asterisk 1.2.0-rc2 released.
+	
+2005-11-11  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* channels/chan_sip.c (thread_safe_rand): ensure that threads don't get the same random number (issue #5712)
+
+	* apps/app_voicemail.c (forward_message): correct bugs in message forwarding (issue #5718)
+	(copy_message): use correct path for locking (issue #5704)
+
+	* apps/app_dial.c (wait_for_answer): correct flag copying for automon feature (issue #5720)
+
+	* channels/chan_iax2.c: correct comment
+
+	* apps/app_voicemail.c (close_mailbox): correct previous commit (issue #5663)
+	(vm_change_password): fix password change writing (issue #5721)
+
+	* channels/chan_sip.c (transmit_invite): remove useless debug message; don't try to add OSP tokens to OPTIONS pings
+
+	* apps/app_voicemail.c (close_mailbox): properly remove deleted messages at mailbox close time (issue #5663)
+
+2005-11-11  Mark Spencer <markster@digium.com>
+
+	* channels/chan_zap.c (zt_bridge): only enable/disable DTMF detection on SUB_REAL channels
+
+2005-11-10  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* channels/chan_iax2.c: ensure that system headers that provide basic types are included first (issue #5713)
+
+2005-11-11  Russell Bryant  <russell@digium.com>
+
+	* many files in apps/: Clean up application descriptions. Clarify some wording and make sure they wrap at 80 characters.
+
+2005-11-10  Mark Spencer <markster@digium.com>
+
+	* rtp.c (ast_rtp_raw_write): use unsigned int for return value from calc_txstamp() (issue #5595)
+	(calc_txstamp): never return a value that was less than zero before being turned into 'unsigned int' (issue #5595)
+
+2005-11-10  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* include/asterisk/chanspy.h: move spy-related stuff into separate header so chan_h323 can build (issue #5590)
+
+	* include/asterisk/linkedlists.h (AST_LIST_HEAD_SET_NOLOCK): properly initialize tail pointer when list head is directly set (issue #5669)
+
+	* app.c (ast_app_parse_options): ok, so we aren't all perfect... let's make the while loop actually work properly here (issue #5684)
+
+	* apps/app_disa.c (disa_exec): correct password file parsing (issue #5676)
+
+	* apps/app_meetme.c (conf_run): don't restrict admin users from joining a locked conference (issue #5680)
+
+	* channels/chan_misdn.c: include stdio.h (issue #5671)
+	* channels/chan_misdn_config.c: fix prototype warning (issue #5671)
+
+	* pbx.c: remove apps that were deprecated before 1.0 was released (issue #5673)
+
+	* apps/app_striplsd.c, apps/app_substring.c: remove apps that were deprecated before 1.0 was released (issue #5673)
+
+	* include/asterisk/lock.h (PTHREAD_MUTEX_RECURSIVE_NP): work around header problems on Cygwin (issue #5668)
+
+	* pbx/pbx_ael.c: handle switch default cases inside macros properly (issue #5354)
+
+	* configs/voicemail.conf.sample (format): add strong warning about changing format list when mailboxes contain messages (issue #5689)
+
+	* many files: ensure that system headers are included before Asterisk headers (issue #5693)
+
+	* channels/chan_iax2.c (complete_iax2_show_peer): don't return from function without releasing lock (issue #5685)
+
+	* channels/iax2-provision.c (iax_provision_reload): don't leak memory (issue #5700)
+
+	* pbx/pbx_ael.c (handle_macro): don't leak memory (issue #5701)
+	(handle_context): ditto
+
+	* res/res_features.c (load_config): properly initialize referenced variable (issue #5703)
+
+	* apps/app_queue.c (rqm_exec): correct segfault problem (issue #5705)
+	(aqm_exec): ditto
+
+	* app.c (ast_app_parse_options): don't increment 's' until after checking for NULL (related to issue #5630)
+
+	* apps/app_rpt.c: solve a memory leak (config structure was not freed) (issue #5706)
+
+2005-11-10  Russell Bryant <russell@digium.com>
+
+	* app.c (ast_app_separate_args): Don't consider the open parenthesis as part of the arguments to an option. (issue #5630)
+
+	* many files: Change all references to ast_separate_app_args to ast_app_separate_args
+
+	* many files in apps/: Clean up some application descriptions. Make sure all descriptions in changed files are wrapped at 80 characters.
+
+2005-11-09  Russell Bryant <russell@digium.com>
+
+	* pbx.c: Clean up descriptions of built-in dialplan applications. Changes include clearer wording and not referring to return values.
+
+2005-11-09  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* channels/chan_iax2.c (update_registry): don't complain about unspecifed registration expiration intervals, just use the minimum
+
+2005-11-08  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* Asterisk 1.2.0-rc1 released.
+
+	* include/asterisk/file.h: add test to ensure that stdio.h is included before this file (issue #5658)
+
+	* res/res_odbc.c (odbc_prepare_and_execute): add new API call for use to properly handle prepared statements across server disconnects (issue #5563)
+
+	* pbx.c (pbx_substitute_variables_helper_full): use already-substituted buffer for parsing variable name (issue #5664)
+
+	* channels/chan_zap.c (zt_request): return AST_CAUSE_CONGESTION when a group-channel is requested and the group exists but all channels are busy (issue #3360, related fix)
+	* channels/chan_iax2.c (create_addr): treat UNREACHABLE as AST_CAUSE_UNREGISTERED so that it will generate CHANUNAVAIL from app_dial (issue #3360)
+
+	* res/res_features.c (ast_bridge_call_thread_launch): set SCHED_RR separately from thread creation, so it won't fail when running as non-root (issue #5601, different fix)
+
+	* pbx.c (pbx_builtin_pushvar_helper): add new API function for setting variables that can exist multiple times (issue #2720)
+	* apps/Makefile (APPS): add app_stack (issue #2720)
+	* apps/app_stack.c: new applications (issue #2720)
+
+	* apps/app_meetme.c: fix two audio delay problems related to using non-Zap channels in conferences (issues #3599 and #4252)
+	* configs/meetme.conf.sample: add documentation of new 'audiobuffers' setting to control buffering on incoming audio from non-Zap channels
+
+	* channels/chan_local.c (local_call): move channel variables from incoming to outgoing instead of inheriting them (issue #5604)
+
+	* many files: add explicit include of stdio.h (issue #5650)
+
+2005-11-07  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* UPGRADE.txt (Parking): add note about new parking behavior (issue #5532)
+
+	* many files: more Cygwin compatibility, and proper getloadavg() prototype/macro (issue #5569)
+
+	* include/asterisk/lock.h (__ast_pthread_mutex_lock): correct build with DETECT_DEADLOCKS defined (issue #5570)
+
+2005-11-07  Russell Bryant  <russell@digium.com>
+
+	* apps/app_queue.c: upgrade to new arg/option API and implement priority jumping control (issue #5580)
+	* many files: Add missing include of stdio.h, and remove some duplicate and unused header includes
+
+	* include/asterisk/app.h: Increment the arg_index in the options structure to fix applicaiton options that have arguments to them
+
+2005-11-07  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* cryptostub.c: include necessary headers
+	* include/asterisk/crypto.h: don't include unnecessary headers
+
+	* manager.c (action_setvar): add support for setting global variables (issue #5571)
+
+	* Makefile: correct cross-compilation issue introduced in Cygwin patches (issue #5572)
+
+	* apps/app_voicemail.c: upgrade to new arg/option API and implement priority jumping control (issue #5649)
+
+	* asterisk.c (main): setpriority() failure is not a reason to stop the process (issue #5581)
+
+	* say.c (ast_say_date_with_format_da): say hours properly (issue #5576)
+
+	* manager.c (astman_get_variables): restore old multiple-variable behavior for "Variable" header (issue #5585)
+
+	* many files: don't check for NULL before calling ast_strlen_zero, it can do it itself (issue #5648)
+
+	* pbx.c (handle_show_hints): use proper state-to-string function for hint state (issue #5583)
+
+	* rtp.c: use unsigned format for debug packet output (issue #5595)
+
+	* asterisk.c (main): force a dnsmgr background refresh after all other modules are initialized (issue #5599)
+	* dnsmgr.c: add ability to start a background refresh on demand (issue #5599)
+
+	* apps/app_dial.c (HANDLE_CAUSE): set CDR disposition to match cause code (issue #5602)
+
+	* asterisk.c: support 'runuser' and 'rungroup' options in asterisk.conf (issue #5621)
+
+	* res/Makefile, apps/Makefile, channels/Makefile, Makefile: support WITHOUT_ZAPTEL define to forcibly avoid building Zaptel support (issue #5634)
+
+	* Makefile: various fixes (issue #5633)
+
+	* apps/app_osplookup.c: upgrade to new arg/option API and implement priority jumping control
+
+	* channels/chan_misdn.c: various fixes (issue #5639)
+	* channels/misdn/isdn_lib.c: various fixes (issue #5639)
+
+	* apps/app_playback.c: upgrade to new arg/option API and implement priority jumping control
+
+	* apps/app_privacy.c: upgrade to new arg/option API and implement priority jumping control
+
+	* apps/app_sendtext.c: upgrade to new arg/option API and implement priority jumping control
+
+	* apps/app_transfer.c: upgrade to new arg/option API and implement priority jumping control
+
+	* apps/app_txtcidname.c: upgrade to new arg/option API and implement priority jumping control
+
+	* Makefile: restore function of 'dont-optimize'
+
+	* config.c (config_text_file_load): don't generate log message when stat() fails
+
+	* many files: clean up application documentation to not refer to return values, since they cannot be used in the dialplan (work done by Neil Lewis)
+
+2005-11-06  Russell Bryant <russell@digium.com>
+
+	* many files: alphabetize options in applicaiton descriptions
+
+	* channels/chan_iax2.c: Use an enum to define iax peer/user flags as well as the pvt structure state.  Use the ast_flags macros for checking or setting the state.
+
+	* sounds.txt: Add missing words from the description of the vm-opts prompt
+
+	* apps/app_externalivr.c: Add a space that fixes building on older versions of gcc
+
+	* many files: Add doxygen updates to categorize modules into groups. Convert a lot of comments over to doxygen style. Add some text giving a basic overview of channels.
+
+	* many files: Update applications to add an exit status variable, make priority jumping optional, and use new args parsing macros
+
+	* pbx.c cdr.c res/res_features.c apps/app_dial.c include/asterisk/cdr.h: Convert some built-in applications to use new args parsing macros.  Change ast_cdr_reset to take a pointer to an ast_flags structure instead of an integer for flags.
+
+	* channels/chan_agent.c: Don't loop forever on an invalid options string
+
+	* apps/app_disa.c apps/app_forkcdr.c: Fix to use correct arguments to ast_cdr_reset
+
+2005-11-05  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* Makefile: don't rebuild asterisk/build.h unless the asterisk binary is going to be relinked for some other reason (stops spurious recompile/link every time 'make' is issued); clean up variable substitutions to use consistent syntax
+	* asterisk.c: don't include asterisk/build.h (it's unnecessary)
+	* cli.c: don't include asterisk/build.h, use extern references to buildinfo.c
+	* buildinfo.c: new file to hold version info strings
+
+2005-11-04  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* apps/app_mixmonitor.c (mixmonitor_exec): correct app name in an error message
+
+2005-11-04  Russell Bryant  <russell@digium.com>
+
+	* channels/chan_iax2.c: Create a function that stores a peer's status in a given buffer.  Use this function in "iax2 show peers" and "iax2 show peer <peername>".  Also, add the peer's status as an option to the IAXPEER dialplan function.
+
+2005-11-04  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* include/asterisk/compiler.h: don't try to use always_inline on old compilers
+
+2005-11-03  Russell Bryant  <russell@digium.com>
+
+	* res/res_agi.c: initialize buffer for result so that the contents are always valid in the response to GET FULL VARIABLE
+
+2005-11-03  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* doc/README.variables: document DYNAMIC_FEATURES
+
+	* res/res_features.c (ast_bridge_call): remove unused variables
+
+	* apps/app_dial.c (dial_exec_full): simplify options and flag usage
+
+	* include/asterisk/app.h: re-work application arg/option parsing APIs for consistent naming, add doxygen docs for option API
+	* many files: update to new APIs
+
+2005-11-02  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* apps/app_dial.c (dial_exec_full): convert to use API calls for argument/option parsing
+
+	* include/asterisk/channel.h: add doxygen docs for silence generator APIs
+
+	* channel.c (ast_channel_bridge): simplify native-bridge return logic, remove 'unsuccessful' message since it causes too many questions :-)
+
+2005-11-01  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* stdtime/localtime.c: fix build failure on uClibc systems (issue #5558)
+	* devicestate.c: same
+
+	* many files: make chan_misdn actually build (issue #5566)
+
+	* many files: more Cygwin build system support (issue #4678)
+
+	* apps/app_parkandannounce.c (parkandannounce_exec): supply parent channel to ast_request_and_dial so channel variables can be inherited (issue #5564)
+	* include/asterisk/channel.h: add parent_channel field
+	* channel.c (__ast_request_and_dial): use parent_channel field to inherit variables into new channel
+
+	* apps/app_cut.c (cut_internal): use ast_app_separate_args() instead of open code (issue #5560)
+
+	* apps/app_mixmonitor.c (launch_monitor_thread): ast_strlen_zero can handle NULL input (issue #5561)
+	(mixmonitor_exec): same
+
+	* res/res_features.c (ast_feature_request_and_dial): ensure that channel variables are inherited from the channel placing the call (issue #5499)
+
+	* utils.c (getloadavg): change to using _BSD_SOURCE as the indicator for whether this function is present or not (issue #5549)
+	
+	* include/asterisk/utils.h (ast_slinear_saturated_add): force to be inlined whenever possible
+	(ast_slinear_saturated_multiply): same
+	(ast_slinear_saturated_divide): same
+	(inaddrcmp): same
+	* include/asterisk/strings.h (ast_strlen_zero): force to be inlined whenever possible
+	* include/asterisk/compiler.h (force_inline): add macro to force inlining of functions
+	
+	* app.c (ast_play_and_record): use ast_silence_generator during recording if requested
+	* asterisk.c: add global option to enable silence-during-record (issue #5135)
+	* channel.c (silence_generator_alloc): new
+	(silence_generator_release): new
+	(silence_generator_generate): new
+	(ast_channel_start_silence_generator): new API call to start generating silence on a channel
+	(ast_channel_stop_silence_generator): parallel call to stop silence generation
+	* apps/app_record.c (record_exec): use ast_silence_generator during recording if requested
+
+2005-11-01  Kevin P. Fleming  <kpfleming@digium.com>
+
+	* Asterisk 1.2.0-beta2 released.
+
--- a/381
+++ b/381
@@ -1,341 +1,56 @@
+Asterisk is distributed under the GNU General Public License version 2
+and is also available under alternative licenses negotiated directly
+with Digium, Inc. If you obtained Asterisk under the GPL, then the GPL
+applies to all loadable Asterisk modules used on your system as well,
+except as defined below. The GPL (version 2) is included in this
+source tree in the file COPYING.

-		    GNU GENERAL PUBLIC LICENSE
-		       Version 2, June 1991
+Digium, Inc. (formerly Linux Support Services) holds copyright
+and/or sufficient licenses to all components of the Asterisk
+package, and therefore can grant, at its sole discretion, the ability
+for companies, individuals, or organizations to create proprietary or
+Open Source (even if not GPL) modules which may be dynamically linked at
+runtime with the portions of Asterisk which fall under our
+copyright/license umbrella, or are distributed under more flexible
+licenses than GPL.

- Copyright (C) 1989, 1991 Free Software Foundation, Inc.
-                       59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- Everyone is permitted to copy and distribute verbatim copies
- of this license document, but changing it is not allowed.
+If you wish to use our code in other GPL programs, don't worry --
+there is no requirement that you provide the same exception in your
+GPL'd products (although if you've written a module for Asterisk we
+would strongly encourage you to make the same exception that we do).

-			    Preamble
+Specific permission is also granted to link Asterisk with OpenSSL and
+OpenH323.

-  The licenses for most software are designed to take away your
-freedom to share and change it.  By contrast, the GNU General Public
-License is intended to guarantee your freedom to share and change free
-software--to make sure the software is free for all its users.  This
-General Public License applies to most of the Free Software
-Foundation's software and to any other program whose authors commit to
-using it.  (Some other Free Software Foundation software is covered by
-the GNU Library General Public License instead.)  You can apply it to
-your programs, too.
+In addition, Asterisk implements two management/control protocols: the
+Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface
+(AGI). It is our belief that applications using these protocols to
+manage or control an Asterisk instance do not have to be licensed
+under the GPL or a compatible license, as we believe these protocols
+do not create a 'derivative work' as referred to in the GPL. However,
+should any court or other judiciary body find that these protocols do
+fall under the terms of the GPL, then we hereby grant you a license to
+use these protocols in combination with Asterisk in external
+applications licensed under any license you wish.

-  When we speak of free software, we are referring to freedom, not
-price.  Our General Public Licenses are designed to make sure that you
-have the freedom to distribute copies of free software (and charge for
-this service if you wish), that you receive source code or can get it
-if you want it, that you can change the software or use pieces of it
-in new free programs; and that you know you can do these things.
+The 'Asterisk' name and logos are trademarks owned by Digium, Inc.,
+and use of them is subject to our trademark licensing policies. If you
+wish to use these trademarks for purposes other than simple
+redistribution of Asterisk source code obtained from Digium, you
+should contact our licensing department to determine the necessary
+steps you must take.

-  To protect your rights, we need to make restrictions that forbid
-anyone to deny you these rights or to ask you to surrender the rights.
-These restrictions translate to certain responsibilities for you if you
-distribute copies of the software, or if you modify it.
+If you have any questions regarding our licensing policy, please
+contact us:

-  For example, if you distribute copies of such a program, whether
-gratis or for a fee, you must give the recipients all the rights that
-you have.  You must make sure that they, too, receive or can get the
-source code.  And you must show them these terms so they know their
-rights.
+1.877.546.8963 (via telephone in the USA)
+1.256.428.6000 (via telephone outside the USA)
+1.256.864.0464 (via FAX inside or outside the USA)
+IAX2/misery.digium.com/6000 (via IAX2)
+licensing@digium.com (via email)

-  We protect your rights with two steps: (1) copyright the software, and
-(2) offer you this license which gives you legal permission to copy,
-distribute and/or modify the software.
-
-  Also, for each author's protection and ours, we want to make certain
-that everyone understands that there is no warranty for this free
-software.  If the software is modified by someone else and passed on, we
-want its recipients to know that what they have is not the original, so
-that any problems introduced by others will not reflect on the original
-authors' reputations.
-
-  Finally, any free program is threatened constantly by software
-patents.  We wish to avoid the danger that redistributors of a free
-program will individually obtain patent licenses, in effect making the
-program proprietary.  To prevent this, we have made it clear that any
-patent must be licensed for everyone's free use or not licensed at all.
-
-  The precise terms and conditions for copying, distribution and
-modification follow.
-
-		    GNU GENERAL PUBLIC LICENSE
-   TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
-
-  0. This License applies to any program or other work which contains
-a notice placed by the copyright holder saying it may be distributed
-under the terms of this General Public License.  The "Program", below,
-refers to any such program or work, and a "work based on the Program"
-means either the Program or any derivative work under copyright law:
-that is to say, a work containing the Program or a portion of it,
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-language.  (Hereinafter, translation is included without limitation in
-the term "modification".)  Each licensee is addressed as "you".
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-Activities other than copying, distribution and modification are not
-covered by this License; they are outside its scope.  The act of
-running the Program is not restricted, and the output from the Program
-is covered only if its contents constitute a work based on the
-Program (independent of having been made by running the Program).
-Whether that is true depends on what the Program does.
-
-  1. You may copy and distribute verbatim copies of the Program's
-source code as you receive it, in any medium, provided that you
-conspicuously and appropriately publish on each copy an appropriate
-copyright notice and disclaimer of warranty; keep intact all the
-notices that refer to this License and to the absence of any warranty;
-and give any other recipients of the Program a copy of this License
-along with the Program.
-
-You may charge a fee for the physical act of transferring a copy, and
-you may at your option offer warranty protection in exchange for a fee.
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-  2. You may modify your copy or copies of the Program or any portion
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-    interactive use in the most ordinary way, to print or display an
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-    notice that there is no warranty (or else, saying that you provide
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-
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-possible use to the public, the best way to achieve this is to make it
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-
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-to attach them to the start of each source file to most effectively
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-    Copyright (C) 19yy  <name of author>
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-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation; either version 2 of the License, or
-    (at your option) any later version.
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-
-    Gnomovision version 69, Copyright (C) 19yy name of author
-    Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
-    This is free software, and you are welcome to redistribute it
-    under certain conditions; type `show c' for details.
-
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-parts of the General Public License.  Of course, the commands you use may
-be called something other than `show w' and `show c'; they could even be
-mouse-clicks or menu items--whatever suits your program.
-
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-school, if any, to sign a "copyright disclaimer" for the program, if
-necessary.  Here is a sample; alter the names:
-
-  Yoyodyne, Inc., hereby disclaims all copyright interest in the program
-  `Gnomovision' (which makes passes at compilers) written by James Hacker.
-
-  <signature of Ty Coon>, 1 April 1989
-  Ty Coon, President of Vice
-
-This General Public License does not permit incorporating your program into
-proprietary programs.  If your program is a subroutine library, you may
-consider it more useful to permit linking proprietary applications with the
-library.  If this is what you want to do, use the GNU Library General
-Public License instead of this License.
+Digium, Inc.
+150 West Park Loop
+Suite 100
+Huntsville, AL  35806
+USA
--- a/900
+++ b/900
--- a/159
+++ b/159
@@ -1,12 +1,17 @@
 The Asterisk Open Source PBX
 by Mark Spencer <markster@digium.com>
-Copyright (C) 2001-2004 Digium
+and the Asterisk.org developer community
+
+Copyright (C) 2001-2005 Digium, Inc.
+and other copyright holders.
 ================================================================
+
 * SECURITY
  It is imperative that you read and fully understand the contents of
-  the SECURITY file before you attempt to configure an Asterisk server.
+the SECURITY file before you attempt to configure and run an Asterisk
+server.

-* WHAT IS ASTERISK
+* WHAT IS ASTERISK ?
  Asterisk is an Open Source PBX and telephony toolkit.  It is, in a
 sense, middleware between Internet and telephony channels on the bottom,
 and Internet and telephony applications at the top.  For more information
@@ -14,95 +19,118 @@ on the project itself, please visit the Asterisk home page at:

           http://www.asterisk.org

-In addition you'll find lot's of information compiled by the Asterisk
+In addition you'll find lots of information compiled by the Asterisk
 community on this Wiki:

           http://www.voip-info.org/wiki-Asterisk

-* LICENSING
-  Asterisk is distributed under GNU General Public License.  The GPL also
-must apply to all loadable modules as well, except as defined below.
+There is a book on Asterisk published by O'Reilly under the
+Creative Commons License. It is available in book stores as well
+as in a downloadable version on the http://www.asteriskdocs.org
+web site.

-  Digium, Inc. (formerly Linux Support Services) retains copyright to all 
-of the core Asterisk system, and therefore can grant, at its sole discretion, 
-the ability for companies, individuals, or organizations to create proprietary
-or Open Source (but non-GPL'd) modules which may be dynamically linked at
-runtime with the portions of Asterisk which fall under our copyright
-umbrella, or are distributed under more flexible licenses than GPL.  
-
-
-  If you wish to use our code in other GPL programs, don't worry -- there
-is no requirement that you provide the same exemption in your GPL'd
-products (although if you've written a module for Asterisk we would
-strongly encourage you to make the same exemption that we do).
-
-  Specific permission is also granted to OpenSSL and OpenH323 to link to
-Asterisk.
-
-  If you have any questions, whatsoever, regarding our licensing policy,
-please contact us.
-
-  Modules that are GPL-licensed and not available under Digium's 
-licensing scheme are added to the Asterisk-addons CVS module.
-  
-* REQUIRED COMPONENTS
+* SUPPORTED OPERATING SYSTEMS

 == Linux ==
-  Currently, the Asterisk Open Source PBX is only known to run on the
-Linux OS, although it may be portable to other UNIX-like operating systems
-(like FreeBSD) as well. 
+  The Asterisk Open Source PBX is developed and tested primarily on the
+GNU/Linux operating system, and is supported on every major GNU/Linux
+distribution.

+== Others ==
+  Asterisk has also been 'ported' and reportedly runs properly on other
+operating systems as well, including Sun Solaris, Apple's Mac OS X, and
+the BSD variants.

 * GETTING STARTED

-First, be sure you've got supported hardware (but note that you don't need ANY hardware, not even a soundcard) to install and run Asterisk. Supported are:
+  First, be sure you've got supported hardware (but note that you don't need
+ANY special hardware, not even a soundcard) to install and run Asterisk.
+
+  Supported telephony hardware includes:

 	* All Wildcard (tm) products from Digium (www.digium.com)
 	* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
-	* Full Duplex Sound Card supported by Linux
-	* Adtran Atlas 800 Plus
-	* ISDN4Linux compatible ISDN card
-	* Tormenta Dual T1 card (www.bsdtelephony.com.mx)
+	* any full duplex sound card supported by ALSA or OSS
+        * VoiceTronix OpenLine products

-Hint: CAPI compatible ISDN cards can be run using the add-on channel chan_capi.
+The are several drivers for ISDN BRI cards available from third party sources.
+Check the voip-info.org wiki for more information on chan_capi, chan_misdn and 
+zaphfc.

-So let's proceed:
+* UPGRADING FROM VERSION 1.0
+
+  If you are updating from a previous version of Asterisk, make sure you
+read the UPGRADE.txt file in the source directory. There are some files
+and configuration options that you will have to change, even though we
+made every effort possible to maintain backwards compatibility.
+
+  In order to discover new features to use, please check the configuration
+examples in the /configs directory of the source code distribution. 
+To discover the major new features of Asterisk 1.2, please visit 
+http://www.astricon.net/asterisk1-2/
+
+* NEW INSTALLATIONS
+
+  Ensure that your system contains a compatible compiler and development
+libraries.  Asterisk requires either the GNU Compiler Collection (GCC) version
+3.0 or higher, or a compiler that supports the C99 specification and some of
+the gcc language extensions.  In addition, your system needs to have the C
+library headers available, and the headers and libraries for OpenSSL,
+ncurses and zlib.
+On many distributions, these files are installed by packages with names like
+'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
+
+  So let's proceed:

 1) Run "make"
+
+  Assuming the build completes successfully:
+
 2) Run "make install"

-If this is your first time working with Asterisk, you may wish to install
+  Each time you update or checkout from CVS, you are strongly encouraged 
+to ensure all previous object files are removed to avoid internal 
+inconsistency in Asterisk. Normally, this is automatically done with 
+the presence of the file .cleancount, which increments each time a 'make clean'
+is required, and the file .lastclean, which contains the last .cleancount used. 
+
+  If this is your first time working with Asterisk, you may wish to install
 the sample PBX, with demonstration extensions, etc.  If so, run:

-	"make samples"
+3) "make samples"

-Doing so will overwrite any existing config files you have. If you are lacking a soundcard you won't be able to use the DIAL command on the console, though.
+  Doing so will overwrite any existing config files you have.

-Finally, you can launch Asterisk with:
+  Finally, you can launch Asterisk in the foreground mode (not a daemon)
+with:

-	./asterisk -vvvc
+# asterisk -vvvc

-You'll see a bunch of verbose messages fly by your screen as Asterisk
+  You'll see a bunch of verbose messages fly by your screen as Asterisk
 initializes (that's the "very very verbose" mode).  When it's ready, if
 you specified the "c" then you'll get a command line console, that looks
 like this:

 *CLI>

-You can type "help" at any time to get help with the system.  For help
+  You can type "help" at any time to get help with the system.  For help
 with a specific command, type "help <command>".  To start the PBX using
 your sound card, you can type "dial" to dial the PBX.  Then you can use
 "answer", "hangup", and "dial" to simulate the actions of a telephone.
-Remember that if you don't have a full duplex sound card (And asterisk
-will tell you somewhere in its verbose messages if you do/don't) than it
+Remember that if you don't have a full duplex sound card (and Asterisk
+will tell you somewhere in its verbose messages if you do/don't) then it
 won't work right (not yet).

-Feel free to look over the configuration files in /etc/asterisk, where
+  "man asterisk" at the Unix/Linux command prompt will give you detailed
+information on how to start and stop Asterisk, as well as all the command
+line options for starting Asterisk.
+
+  Feel free to look over the configuration files in /etc/asterisk, where
 you'll find a lot of information about what you can do with Asterisk.

 * ABOUT CONFIGURATION FILES

-All Asterisk configuration files share a common format.  Comments are
+  All Asterisk configuration files share a common format.  Comments are
 delimited by ';' (since '#' of course, being a DTMF digit, may occur in
 many places).  A configuration file is divided into sections whose names
 appear in []'s.  Each section typically contains two types of statements,
@@ -111,12 +139,12 @@ parameters'.  Internally the use of '=' and '=>' is exactly the same, so
 they're used only to help make the configuration file easier to
 understand, and do not affect how it is actually parsed.

-Entries of the form 'variable=value' set the value of some parameter in
-asterisk.  For example, in tormenta.conf, one might specify:
+  Entries of the form 'variable=value' set the value of some parameter in
+asterisk.  For example, in zapata.conf, one might specify:

 	switchtype=national

-In order to indicate to Asterisk that the switch they are connecting to is
+in order to indicate to Asterisk that the switch they are connecting to is
 of the type "national".  In general, the parameter will apply to
 instantiations which occur below its specification.  For example, if the
 configuration file read:
@@ -127,18 +155,18 @@ configuration file read:
 	switchtype = dms100
 	channel => 25-47

-Then, the "national" switchtype would be applied to channels one through
+the "national" switchtype would be applied to channels one through
 four and channels 10 through 12, whereas the "dms100" switchtype would
 apply to channels 25 through 47.
  
-The "object => parameters" instantiates an object with the given
+  The "object => parameters" instantiates an object with the given
 parameters.  For example, the line "channel => 25-47" creates objects for
-the channels 25 through 47 of the tormenta card, obtaining the settings
+the channels 25 through 47 of the card, obtaining the settings
 from the variables specified above.

 * SPECIAL NOTE ON TIME
  
-  Those using SIP phones should be aware the Asterisk is sensitive to
+  Those using SIP phones should be aware that Asterisk is sensitive to
 large jumps in time.  Manually changing the system time using date(1)
 (or other similar commands) may cause SIP registrations and other
 internal processes to fail.  If your system cannot keep accurate time
@@ -156,7 +184,7 @@ apparent.  The use of daylight savings time in a Linux system is
 purely a user interface issue and does not affect the operation of the
 Linux kernel or Asterisk.  The system clock on Linux kernels operates
 on UTC.  UTC does not use daylight savings time.
-   
+
  Also note that this issue is separate from the clocking of TDM
 channels, and is known to at least affect SIP registrations.

@@ -196,18 +224,17 @@ these changes to take effect.
 above you can try adding the command "ulimit -n 8192" to the script
 that starts Asterisk.

-
 * MORE INFORMATION

-See the doc directory for more documentation.
+  See the doc directory for more documentation on various features. Again,
+please read all the configuration samples that include documentation on
+the configuration options.

-Finally, you may wish to visit the web site and join the mailing list if
+  Finally, you may wish to visit the web site and join the mailing list if
 you're interested in getting more information.

-   http://www.asterisk.org/index.php?menu=support
+   http://www.asterisk.org/support

-Welcome to the growing worldwide community of Asterisk users!
+  Welcome to the growing worldwide community of Asterisk users!

 Mark Spencer
-
-
--- a/28
+++ b/28
@@ -4,6 +4,32 @@ PLEASE READ THE FOLLOWING IMPORTANT SECURITY RELATED INFORMATION.
 IMPROPER CONFIGURATION OF ASTERISK COULD ALLOW UNAUTHORIZED USE OF YOUR
 FACILITIES, POTENTIALLY INCURRING SUBSTANTIAL CHARGES.

+Asterisk security involves both network security (encryption, authentication)
+as well as dialplan security (authorization - who can access services in 
+your pbx). If you are setting up Asterisk in production use, please make
+sure you understand the issues involved.
+
+* NETWORK SECURITY
+
+If you install Asterisk and use the "make samples" command to install
+a demonstration configuration, Asterisk will open a few ports for accepting
+VoIP calls. Check the channel configuration files for the ports and IP addresses.
+
+If you enable the manager interface in manager.conf, please make sure that
+you access manager in a safe environment or protect it with SSH or other
+VPN solutions.
+
+For all TCP/IP connections in Asterisk, you can set ACL lists that
+will permit or deny network access to Asterisk services. Please check
+the "permit" and "deny" configuration options in manager.conf and
+the VoIP channel configurations - i.e. sip.conf and iax.conf.
+
+The IAX2 protocol supports strong RSA key authentication as well as
+AES encryption of voice and signalling. The SIP channel does not
+support encryption in this version of Asterisk.
+
+* DIALPLAN SECURITY
+
 First and foremost remember this:

 USE THE EXTENSION CONTEXTS TO ISOLATE OUTGOING OR TOLL SERVICES FROM ANY
@@ -38,4 +64,4 @@ exten => 6123,Dial(Zap/1)

 DON'T FORGET TO TAKE THE DEMO CONTEXT OUT OF YOUR DEFAULT CONTEXT.  There
 isn't really a security reason, it just will keep people from wanting to 
-play with your asterisk setup remotely.
+play with your Asterisk setup remotely.
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -0,0 +1,198 @@
+Information for Upgrading From Previous Asterisk Releases
+=========================================================
+
+Compiling:
+
+* The Asterisk 1.2 source code now uses C language features
+  supported only by 'modern' C compilers.  Generally, this means GCC
+  version 3.0 or higher, although some GCC 2.96 releases will also
+  work.  Some non-GCC compilers that support C99 and the common GCC
+  extensions (including anonymous structures and unions) will also
+  work.  All releases of GCC 2.95 do _not_ have the requisite feature
+  support; systems using that compiler will need to be upgraded to
+  a more recent compiler release.
+
+Dialplan Expressions:
+
+* The dialplan expression parser (which handles $[ ... ] constructs)
+  has gone through a major upgrade, but has one incompatible change:
+  spaces are no longer required around expression operators, including
+  string comparisons. However, you can now use quoting to keep strings
+  together for comparison. For more details, please read the
+  doc/README.variables file, and check over your dialplan for possible
+  problems.
+
+Agents:
+
+* The default for ackcall has been changed to "no" instead of "yes" 
+  because of a bug which caused the "yes" behavior to generally act like
+  "no".  You may need to adjust the value if your agents behave 
+  differently than you expect with respect to acknowledgement.
+
+* The AgentCallBackLogin application now requires a second '|' before
+  specifying an extension@context.  This is to distinguish the options
+  string from the extension, so that they do not conflict.  See
+  'show application AgentCallbackLogin' for more details.
+
+Parking:
+
+* Parking behavior has changed slightly; when a parked call times out,
+  Asterisk will attempt to deliver the call back to the extension that
+  parked it, rather than the 's' extension. If that extension is busy
+  or unavailable, the parked call will be lost.
+
+Dialing:
+
+* The Caller*ID of the outbound leg is now the extension that was 
+  called, rather than the Caller*ID of the inbound leg of the call.  The 
+  "o" flag for Dial can be used to restore the original behavior if 
+  desired.  Note that if you are looking for the originating callerid
+  from the manager event, there is a new manager event "Dial" which 
+  provides the source and destination channels and callerid.
+
+IAX: 
+
+* The naming convention for IAX channels has changed in a minor way such 
+  that the call number follows a "-" rather than a "/" character.
+
+SIP:
+
+* The global option "port" in 1.0.X that is used to set which port to
+  bind to has been changed to "bindport" to be more consistent with
+  the other channel drivers and to avoid confusion with the "port"
+  option for users/peers.
+
+* The "Registry" event now uses "Username" rather than "User" for 
+  consistency with IAX.
+
+Applications:
+
+* With the addition of dialplan functions (which operate similarly
+  to variables), the SetVar application has been renamed to Set.
+
+* The CallerPres application has been removed.  Use SetCallerPres 
+  instead.  It accepts both numeric and symbolic names.
+
+* The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
+  CheckGroup have been deprecated in favor of functions.  Here is a
+  table of their replacements:
+
+  GetGroupCount([groupname][@category]	       GROUP_COUNT([groupname][@category])	Set(GROUPCOUNT=${GROUP_COUNT()})
+  GroupMatchCount(groupmatch[@category])       GROUP_MATCH_COUNT(groupmatch[@category])	Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
+  SetGroup(groupname[@category])	       GROUP([category])=groupname		Set(GROUP()=test)
+  CheckGroup(max[@category])		       N/A					GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)
+
+  Note that CheckGroup does not have a direct replacement.  There is
+  also a new function called GROUP_LIST() which will return a space
+  separated list of all of the groups set on a channel.  The GROUP()
+  function can also return the name of the group set on a channel when
+  used in a read environment.
+
+* The applications DBGet and DBPut have been deprecated in favor of
+  functions.  Here is a table of their replacements:
+
+  DBGet(foo=family/key)        Set(foo=${DB(family/key)})
+  DBPut(family/key=${foo})     Set(DB(family/key)=${foo})
+
+* The application SetLanguage has been deprecated in favor of the
+  function LANGUAGE().
+
+  SetLanguage(fr)		Set(LANGUAGE()=fr)
+
+  The LANGUAGE function can also return the currently set language:
+
+  Set(MYLANG=${LANGUAGE()})
+
+* The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
+  have been deprecated in favor of the function TIMEOUT(timeouttype):
+
+  AbsoluteTimeout(300)		Set(TIMEOUT(absolute)=300)
+  DigitTimeout(15)		Set(TIMEOUT(digit)=15)
+  ResponseTimeout(15)		Set(TIMEOUT(response)=15)
+
+  The TIMEOUT() function can also return the currently set timeouts:
+
+  Set(DTIMEOUT=${TIMEOUT(digit)})
+
+* The applications SetCIDName, SetCIDNum, and SetRDNIS have been
+  deprecated in favor of the CALLERID(datatype) function:
+
+  SetCIDName(Joe Cool)		Set(CALLERID(name)=Joe Cool)
+  SetCIDNum(2025551212)		Set(CALLERID(number)=2025551212)
+  SetRDNIS(2024561414)		Set(CALLERID(RDNIS)=2024561414)
+
+* The application Record now uses the period to separate the filename
+  from the format, rather than the colon.
+
+* The application VoiceMail now supports a 'temporary' greeting for each
+  mailbox. This greeting can be recorded by using option 4 in the
+  'mailbox options' menu, and 'change your password' option has been
+  moved to option 5.
+
+Queues:
+
+* A queue is now considered empty not only if there are no members but if
+  none of the members are available (e.g. agents not logged on).  To
+  restore the original behavior, use "leavewhenempty=strict" or 
+  "joinwhenempty=strict" instead of "=yes" for those options.
+
+* It is now possible to use multi-digit extensions in the exit context
+  for a queue (although you should not have overlapping extensions,
+  as there is no digit timeout). This means that the EXITWITHKEY event
+  in queue_log can now contain a key field with more than a single
+  character in it.
+
+Extensions:
+
+* By default, there is a new option called "autofallthrough" in
+  extensions.conf that is set to yes.  Asterisk 1.0 (and earlier) 
+  behavior was to wait for an extension to be dialed after there were no 
+  more extensions to execute.  "autofallthrough" changes this behavior
+  so that the call will immediately be terminated with BUSY,
+  CONGESTION, or HANGUP based on Asterisk's best guess.  If you are
+  writing an extension for IVR, you must use the WaitExten application
+  if "autofallthrough" is set to yes.
+
+AGI:
+
+* AGI scripts did not always get SIGHUP at the end, previously.  That 
+  behavior has been fixed.  If you do not want your script to terminate 
+  at the end of AGI being called (e.g. on a hangup) then set SIGHUP to 
+  be ignored within your application.
+
+* CallerID is reported with agi_callerid and agi_calleridname instead
+  of a single parameter holding both.
+
+Music On Hold:
+
+* The preferred format for musiconhold.conf has changed; please see the
+  sample configuration file for the new format. The existing format
+  is still supported but will generate warnings when the module is loaded.
+
+chan_modem:
+
+* All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
+  in this release, and will be removed in the next major Asterisk release.
+  Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
+  path for aopen and bestdata modem users.
+
+MeetMe:
+
+* The conference application now allows users to increase/decrease their
+  speaking volume and listening volume (independently of each other and 
+  other users); the 'admin' and 'user' menus have changed, and new sound 
+  files are included with this release. However, if a user calling in 
+  over a Zaptel channel that does NOT have hardware DTMF detection 
+  increases their speaking volume, it is likely they will no longer be 
+  able to enter/exit the menu or make any further adjustments, as the  
+  software DTMF detector will not be able to recognize the DTMF coming 
+  from their device.
+
+GetVar Manager Action:
+
+* Previously, the behavior of the GetVar manager action reported the value
+  of a variable in the following manner:
+   > name: value
+  This has been changed to a manner similar to the SetVar action and is now
+   > Variable: name
+   > Value: value
--- a/acl.c
+++ b/acl.c
@@ -1,14 +1,25 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Various sorts of access control
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Various sorts of access control
+ * 
 */

 #include <stdio.h>
@@ -18,25 +29,50 @@
 #include <signal.h>
 #include <errno.h>
 #include <unistd.h>
-#include <asterisk/acl.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/utils.h>
-#include <asterisk/lock.h>
+#include <netinet/in.h>
 #include <arpa/inet.h>
 #include <sys/socket.h>
 #include <netdb.h>
 #include <net/if.h>
+#include <netinet/in.h>
 #include <netinet/in_systm.h>
 #include <netinet/ip.h>
 #include <sys/ioctl.h>
+
 #if defined(__OpenBSD__) || defined(__NetBSD__) || defined(__FreeBSD__)
 #include <fcntl.h>
 #include <net/route.h>
-
-AST_MUTEX_DEFINE_STATIC(routeseq_lock);
 #endif

+#if defined(SOLARIS)
+#include <sys/sockio.h>
+#endif
+
+/* netinet/ip.h may not define the following (See RFCs 791 and 1349) */
+#if !defined(IPTOS_LOWCOST)
+#define       IPTOS_LOWCOST           0x02
+#endif
+
+#if !defined(IPTOS_MINCOST)
+#define       IPTOS_MINCOST           IPTOS_LOWCOST
+#endif
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/acl.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
+#include "asterisk/lock.h"
+#include "asterisk/srv.h"
+#include "asterisk/compat.h"
+
+#if defined(__OpenBSD__) || defined(__NetBSD__) || defined(__FreeBSD__)
+AST_MUTEX_DEFINE_STATIC(routeseq_lock);
+#endif

 struct ast_ha {
 	/* Host access rule */
@@ -77,12 +113,10 @@ static void ast_copy_ha(struct ast_ha *from, struct ast_ha *to)
 static struct ast_ha *ast_duplicate_ha(struct ast_ha *original)
 {
 	struct ast_ha *new_ha = malloc(sizeof(struct ast_ha));
-
 	/* Copy from original to new object */
 	ast_copy_ha(original, new_ha); 

-	return(new_ha);
-
+	return new_ha;
 }

 /* Create duplicate HA link list */
@@ -93,7 +127,7 @@ struct ast_ha *ast_duplicate_ha_list(struct ast_ha *original)
 	struct ast_ha *ret = NULL;
 	struct ast_ha *link,*prev=NULL;

-	while(start) {
+	while (start) {
 		link = ast_duplicate_ha(start);  /* Create copy of this object */
 		if (prev)
 			prev->next = link;		/* Link previous to this object */
@@ -104,34 +138,34 @@ struct ast_ha *ast_duplicate_ha_list(struct ast_ha *original)
 		start = start->next;		/* Go to next object */
 		prev = link;			/* Save pointer to this object */
 	}
-	return (ret);    			/* Return start of list */
+	return ret;    			/* Return start of list */
 }

 struct ast_ha *ast_append_ha(char *sense, char *stuff, struct ast_ha *path)
 {
 	struct ast_ha *ha = malloc(sizeof(struct ast_ha));
-	char *nm="255.255.255.255";
-	char tmp[256] = "";
+	char *nm = "255.255.255.255";
+	char tmp[256];
 	struct ast_ha *prev = NULL;
 	struct ast_ha *ret;
-	int x,z;
+	int x, z;
 	unsigned int y;
 	ret = path;
-	while(path) {
+	while (path) {
 		prev = path;
 		path = path->next;
 	}
 	if (ha) {
-		strncpy(tmp, stuff, sizeof(tmp) - 1);
+		ast_copy_string(tmp, stuff, sizeof(tmp));
 		nm = strchr(tmp, '/');
-		if (!nm)
+		if (!nm) {
 			nm = "255.255.255.255";
-		else {
+		} else {
 			*nm = '\0';
 			nm++;
 		}
 		if (!strchr(nm, '.')) {
-			if ((sscanf(nm, "%i", &x) == 1) && (x >= 0) && (x <= 32)) {
+			if ((sscanf(nm, "%d", &x) == 1) && (x >= 0) && (x <= 32)) {
 				y = 0;
 				for (z=0;z<x;z++) {
 					y >>= 1;
@@ -140,12 +174,12 @@ struct ast_ha *ast_append_ha(char *sense, char *stuff, struct ast_ha *path)
 				ha->netmask.s_addr = htonl(y);
 			}
 		} else if (!inet_aton(nm, &ha->netmask)) {
-			ast_log(LOG_WARNING, "%s not a valid netmask\n", nm);
+			ast_log(LOG_WARNING, "%s is not a valid netmask\n", nm);
 			free(ha);
 			return path;
 		}
 		if (!inet_aton(tmp, &ha->netaddr)) {
-			ast_log(LOG_WARNING, "%s not a valid IP\n", tmp);
+			ast_log(LOG_WARNING, "%s is not a valid IP\n", tmp);
 			free(ha);
 			return path;
 		}
@@ -156,12 +190,13 @@ struct ast_ha *ast_append_ha(char *sense, char *stuff, struct ast_ha *path)
 			ha->sense = AST_SENSE_DENY;
 		}
 		ha->next = NULL;
-		if (prev)
+		if (prev) {
 			prev->next = ha;
-		else
+		} else {
 			ret = ha;
+		}
 	}
-	ast_log(LOG_DEBUG, "%s/%s appended to acl for peer\n",stuff, nm);
+	ast_log(LOG_DEBUG, "%s/%s appended to acl for peer\n", stuff, nm);
 	return ret;
 }

@@ -169,7 +204,7 @@ int ast_apply_ha(struct ast_ha *ha, struct sockaddr_in *sin)
 {
 	/* Start optimistic */
 	int res = AST_SENSE_ALLOW;
-	while(ha) {
+	while (ha) {
 		char iabuf[INET_ADDRSTRLEN];
 		char iabuf2[INET_ADDRSTRLEN];
 		/* DEBUG */
@@ -179,17 +214,29 @@ int ast_apply_ha(struct ast_ha *ha, struct sockaddr_in *sin)
 			ast_inet_ntoa(iabuf2, sizeof(iabuf2), ha->netaddr));
 		/* For each rule, if this address and the netmask = the net address
 		   apply the current rule */
-		if ((sin->sin_addr.s_addr & ha->netmask.s_addr) == (ha->netaddr.s_addr))
+		if ((sin->sin_addr.s_addr & ha->netmask.s_addr) == ha->netaddr.s_addr)
 			res = ha->sense;
 		ha = ha->next;
 	}
 	return res;
 }

-int ast_get_ip(struct sockaddr_in *sin, char *value)
+int ast_get_ip_or_srv(struct sockaddr_in *sin, const char *value, const char *service)
 {
 	struct hostent *hp;
 	struct ast_hostent ahp;
+	char srv[256];
+	char host[256];
+	int tportno = ntohs(sin->sin_port);
+	if (inet_aton(value, &sin->sin_addr))
+		return 0;
+	if (service) {
+		snprintf(srv, sizeof(srv), "%s.%s", service, value);
+		if (ast_get_srv(NULL, host, sizeof(host), &tportno, srv) > 0) {
+			sin->sin_port = htons(tportno);
+			value = host;
+		}
+	}
 	hp = ast_gethostbyname(value, &ahp);
 	if (hp) {
 		memcpy(&sin->sin_addr, hp->h_addr, sizeof(sin->sin_addr));
@@ -200,188 +247,87 @@ int ast_get_ip(struct sockaddr_in *sin, char *value)
 	return 0;
 }

+int ast_str2tos(const char *value, int *tos)
+{
+	int fval;
+	if (sscanf(value, "%i", &fval) == 1)
+		*tos = fval & 0xff;
+	else if (!strcasecmp(value, "lowdelay"))
+		*tos = IPTOS_LOWDELAY;
+	else if (!strcasecmp(value, "throughput"))
+		*tos = IPTOS_THROUGHPUT;
+	else if (!strcasecmp(value, "reliability"))
+		*tos = IPTOS_RELIABILITY;
+	else if (!strcasecmp(value, "mincost"))
+		*tos = IPTOS_MINCOST;
+	else if (!strcasecmp(value, "none"))
+		*tos = 0;
+	else
+		return -1;
+	return 0;
+}
+
+int ast_get_ip(struct sockaddr_in *sin, const char *value)
+{
+	return ast_get_ip_or_srv(sin, value, NULL);
+}
+
 /* iface is the interface (e.g. eth0); address is the return value */
-int ast_lookup_iface(char *iface, struct in_addr *address) {
+int ast_lookup_iface(char *iface, struct in_addr *address) 
+{
 	int mysock, res = 0;
 	struct my_ifreq ifreq;

 	memset(&ifreq, 0, sizeof(ifreq));
-	strncpy(ifreq.ifrn_name,iface,sizeof(ifreq.ifrn_name) - 1);
+	ast_copy_string(ifreq.ifrn_name, iface, sizeof(ifreq.ifrn_name));

-	mysock = socket(PF_INET,SOCK_DGRAM,IPPROTO_IP);
-	res = ioctl(mysock,SIOCGIFADDR,&ifreq);
+	mysock = socket(PF_INET, SOCK_DGRAM, IPPROTO_IP);
+	res = ioctl(mysock, SIOCGIFADDR, &ifreq);

 	close(mysock);
 	if (res < 0) {
 		ast_log(LOG_WARNING, "Unable to get IP of %s: %s\n", iface, strerror(errno));
-		memcpy((char *)address,(char *)&__ourip,sizeof(__ourip));
+		memcpy((char *)address, (char *)&__ourip, sizeof(__ourip));
 		return -1;
 	} else {
-		memcpy((char *)address,(char *)&ifreq.ifru_addr.sin_addr,sizeof(ifreq.ifru_addr.sin_addr));
+		memcpy((char *)address, (char *)&ifreq.ifru_addr.sin_addr, sizeof(ifreq.ifru_addr.sin_addr));
 		return 0;
 	}
 }

 int ast_ouraddrfor(struct in_addr *them, struct in_addr *us)
 {
-#if defined(__OpenBSD__) || defined(__NetBSD__) || defined(__FreeBSD__)
-	struct sockaddr_in *sin;
-	struct sockaddr *sa;
-	struct {
-		struct	rt_msghdr m_rtm;
-		char	m_space[512];
-	} m_rtmsg;
-	char iabuf[INET_ADDRSTRLEN];
-	char *cp, *p;
-	int i, l, s, seq, flags;
-	pid_t pid = getpid();
-	static int routeseq;	/* Protected by "routeseq_lock" mutex */
+	int s;
+	struct sockaddr_in sin;
+	socklen_t slen;

-	p = ast_strdupa(ast_inet_ntoa(iabuf, sizeof(iabuf), *them));
-	memset(us, 0, sizeof(struct in_addr));
-
-	memset(&m_rtmsg, 0, sizeof(m_rtmsg));
-	m_rtmsg.m_rtm.rtm_type = RTM_GET;
-	m_rtmsg.m_rtm.rtm_version = RTM_VERSION;
-	ast_mutex_lock(&routeseq_lock);
-	seq = ++routeseq;
-	ast_mutex_unlock(&routeseq_lock);
-	m_rtmsg.m_rtm.rtm_seq = seq;
-	m_rtmsg.m_rtm.rtm_addrs = RTA_DST | RTA_IFA;
-	m_rtmsg.m_rtm.rtm_msglen = sizeof(struct rt_msghdr) + sizeof(struct sockaddr_in);
-	sin = (struct sockaddr_in *)m_rtmsg.m_space;
-	sin->sin_family = AF_INET;
-	sin->sin_len = sizeof(struct sockaddr_in);
-	sin->sin_addr = *them;
-
-	if ((s = socket(PF_ROUTE, SOCK_RAW, AF_UNSPEC)) < 0) {
-		ast_log(LOG_ERROR, "Error opening routing socket\n");
+	s = socket(PF_INET, SOCK_DGRAM, 0);
+	if (s < 0) {
+		ast_log(LOG_WARNING, "Cannot create socket\n");
 		return -1;
 	}
-	flags = fcntl(s, F_GETFL);
-	fcntl(s, F_SETFL, flags | O_NONBLOCK);
-	if (write(s, (char *)&m_rtmsg, m_rtmsg.m_rtm.rtm_msglen) < 0) {
-		ast_log(LOG_ERROR, "Error writing to routing socket: %s\n", strerror(errno));
+	sin.sin_family = AF_INET;
+	sin.sin_port = 5060;
+	sin.sin_addr = *them;
+	if (connect(s, (struct sockaddr *)&sin, sizeof(sin))) {
+		ast_log(LOG_WARNING, "Cannot connect\n");
 		close(s);
 		return -1;
 	}
-	do {
-		l = read(s, (char *)&m_rtmsg, sizeof(m_rtmsg));
-	} while (l > 0 && (m_rtmsg.m_rtm.rtm_seq != seq || m_rtmsg.m_rtm.rtm_pid != pid));
-	if (l < 0) {
-		if (errno != EAGAIN)
-			ast_log(LOG_ERROR, "Error reading from routing socket\n");
+	slen = sizeof(sin);
+	if (getsockname(s, (struct sockaddr *)&sin, &slen)) {
+		ast_log(LOG_WARNING, "Cannot get socket name\n");
 		close(s);
 		return -1;
 	}
 	close(s);
-
-	if (m_rtmsg.m_rtm.rtm_version != RTM_VERSION) {
-		ast_log(LOG_ERROR, "Unsupported route socket protocol version\n");
-		return -1;
-	}
-
-	if (m_rtmsg.m_rtm.rtm_msglen != l)
-		ast_log(LOG_WARNING, "Message length mismatch, in packet %d, returned %d\n",
-				m_rtmsg.m_rtm.rtm_msglen, l);
-
-	if (m_rtmsg.m_rtm.rtm_errno) {
-		ast_log(LOG_ERROR, "RTM_GET got %s (%d)\n",
-				strerror(m_rtmsg.m_rtm.rtm_errno), m_rtmsg.m_rtm.rtm_errno);
-		return -1;
-	}
-
-	cp = (char *)m_rtmsg.m_space;
-	if (m_rtmsg.m_rtm.rtm_addrs)
-		for (i = 1; i; i <<= 1)
-			if (m_rtmsg.m_rtm.rtm_addrs & i) {
-				sa = (struct sockaddr *)cp;
-				if (i == RTA_IFA && sa->sa_family == AF_INET) {
-					sin = (struct sockaddr_in *)sa;
-					*us = sin->sin_addr;
-					ast_log(LOG_DEBUG, "Found route to %s, output from our address %s.\n", p, ast_inet_ntoa(iabuf, sizeof(iabuf), *us));
-					return 0;
-				}
-				cp += sa->sa_len > 0 ?
-					  (1 + ((sa->sa_len - 1) | (sizeof(long) - 1))) :
-					  sizeof(long);
-			}
-
-	ast_log(LOG_DEBUG, "No route found for address %s!\n", p);
-	return -1;
-#else
-	FILE *PROC;
-	unsigned int remote_ip;
-	int res = 1;
-	char line[256];
-	remote_ip = them->s_addr;
-	
-	PROC = fopen("/proc/net/route","r");
-	if (!PROC) {
-		bzero(us,sizeof(struct in_addr));
-		return -1;
-	}
-	/* First line contains headers */
-	fgets(line,sizeof(line),PROC);
-
-	while (!feof(PROC)) {
-		char iface[256];
-		unsigned int dest, gateway, mask;
-		int i,fieldnum;
-		char *fields[40];
-
-		fgets(line,sizeof(line),PROC);
-
-		fieldnum = 0;
-		for (i=0;i<sizeof(line);i++) {
-			char *offset;
-
-			fields[fieldnum++] = line + i;
-			offset = strchr(line + i,'\t');
-			if (offset == NULL) {
-				/* Exit loop */
-				break;
-			} else if (fieldnum >= 9) {
-				/* Short-circuit: can't break at 8, since the end of field 7 is figured when fieldnum=8 */
-				break;
-			} else {
-				*offset = '\0';
-				i = offset - line;
-			}
-		}
-		if (fieldnum >= 8) {
-
-			sscanf(fields[0],"%s",iface);
-			sscanf(fields[1],"%x",&dest);
-			sscanf(fields[2],"%x",&gateway);
-			sscanf(fields[7],"%x",&mask);
-#if 0
-			{ char iabuf[INET_ADDRSTRLEN]; 
-			printf("Addr: %s %08x Dest: %08x Mask: %08x\n", ast_inet_ntoa(iabuf, sizeof(iabuf), *them), remote_ip, dest, mask); }
-#endif		
-			/* Looks simple, but here is the magic */
-			if (((remote_ip & mask) ^ dest) == 0) {
-				res = ast_lookup_iface(iface,us);
-				break;
-			}
-		}
-	}
-	fclose(PROC);
-	if (res == 1) {
-		ast_log(LOG_WARNING, "Yikes!  No default route?!!\n");
-		bzero(us,sizeof(struct in_addr));
-		return -2;
-	} else if (res) {
-		/* We've already warned in subroutine */
-		return -1;
- 	}
+	*us = sin.sin_addr;
 	return 0;
-#endif
 }

 int ast_find_ourip(struct in_addr *ourip, struct sockaddr_in bindaddr)
 {
-	char ourhost[MAXHOSTNAMELEN]="";
+	char ourhost[MAXHOSTNAMELEN] = "";
 	struct ast_hostent ahp;
 	struct hostent *hp;
 	struct in_addr saddr;
@@ -392,7 +338,7 @@ int ast_find_ourip(struct in_addr *ourip, struct sockaddr_in bindaddr)
 		return 0;
 	}
 	/* try to use our hostname */
-	if (gethostname(ourhost, sizeof(ourhost)-1)) {
+	if (gethostname(ourhost, sizeof(ourhost) - 1)) {
 		ast_log(LOG_WARNING, "Unable to get hostname\n");
 	} else {
 		hp = ast_gethostbyname(ourhost, &ahp);
--- a/aescrypt.c
+++ b/aescrypt.c
@@ -30,7 +30,9 @@
 ---------------------------------------------------------------------------
 Issue Date: 26/08/2003

- This file contains the code for implementing encryption and decryption
+*/
+/*! \file
+\brief  This file contains the code for implementing encryption and decryption
 for AES (Rijndael) for block and key sizes of 16, 24 and 32 bytes. It
 can optionally be replaced by code written in assembler using NASM. For
 further details see the file aesopt.h
--- a/aeskey.c
+++ b/aeskey.c
@@ -30,7 +30,9 @@
 ---------------------------------------------------------------------------
 Issue Date: 26/08/2003

- This file contains the code for implementing the key schedule for AES
+*/
+/*! \file
+\brief  This file contains the code for implementing the key schedule for AES
 (Rijndael) for block and key sizes of 16, 24, and 32 bytes. See aesopt.h
 for further details including optimisation.
 */
--- a/aesopt.h
+++ b/aesopt.h
@@ -135,7 +135,8 @@
 #ifndef _AESOPT_H
 #define _AESOPT_H

-#include <asterisk/aes.h>
+#include "asterisk/aes.h"
+#include "asterisk/endian.h"

 /*  CONFIGURATION - USE OF DEFINES

@@ -146,25 +147,6 @@
    #if clauses.
 */

-/*  PLATFORM SPECIFIC INCLUDES */
-
-#if defined( __OpenBSD__ )
-#  include <machine/types.h>
-#  include <sys/endian.h>
-#elif defined( __FreeBSD__ ) || defined( __NetBSD__ )
-#  include <sys/types.h>
-#  include <sys/endian.h>
-#elif defined( BSD ) && ( BSD >= 199103 ) || defined(__APPLE__)
-#  include <machine/endian.h>
-#elif defined( __GNUC__ ) || defined( __GNU_LIBRARY__ )
-#  include <endian.h>
-#if !defined(__APPLE__)
-#  include <byteswap.h>
-#endif
-#elif defined( linux )
-#  include <endian.h>
-#endif
-
 /*  BYTE ORDER IN 32-BIT WORDS

    To obtain the highest speed on processors with 32-bit words, this code
--- a/agi/Makefile
+++ b/agi/Makefile
@@ -1,11 +1,11 @@
 #
 # Asterisk -- A telephony toolkit for Linux.
 # 
-# Makefile for PBX frontends (dynamically loaded)
+# Makefile for AGI-related stuff
 #
-# Copyright (C) 1999, Mark Spencer
+# Copyright (C) 1999-2005, Digium
 #
-# Mark Spencer <markster@linux-support.net>
+# Mark Spencer <markster@digium.com>
 #
 # This program is free software, distributed under the terms of
 # the GNU General Public License
@@ -15,6 +15,15 @@ AGIS=agi-test.agi eagi-test eagi-sphinx-test

 CFLAGS+=

+LIBS=
+ifeq ($(OSARCH),SunOS)
+  LIBS=-lsocket -lnsl ../strcompat.o
+endif
+
+ifeq ($(findstring BSD,${OSARCH}),BSD)
+  CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
+endif
+
 all: depend $(AGIS)

 install: all
@@ -22,10 +31,10 @@ install: all
 	for x in $(AGIS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(AGI_DIR) ; done

 eagi-test: eagi-test.o
-	$(CC) $(CFLAGS) -o eagi-test eagi-test.o
+	$(CC) $(CFLAGS) -o eagi-test eagi-test.o $(LIBS)

 eagi-sphinx-test: eagi-sphinx-test.o
-	$(CC) $(CFLAGS) -o eagi-sphinx-test eagi-sphinx-test.o
+	$(CC) $(CFLAGS) -o eagi-sphinx-test eagi-sphinx-test.o $(LIBS) 

 clean:
 	rm -f *.so *.o look .depend eagi-test eagi-sphinx-test
@@ -40,4 +49,4 @@ endif
 depend: .depend

 .depend:
-	../mkdep $(CFLAGS) `ls *.c`
+	../build_tools/mkdep $(CFLAGS) `ls *.c`
--- a/agi/eagi-sphinx-test.c
+++ b/agi/eagi-sphinx-test.c
@@ -17,6 +17,7 @@
 #include <netinet/in.h>
 #include <arpa/inet.h>
 #include <netdb.h>
+#include <asterisk/compat.h>

 #define AUDIO_FILENO (STDERR_FILENO + 1)

--- a/agi/eagi-test.c
+++ b/agi/eagi-test.c
@@ -11,6 +11,7 @@
 #include <errno.h>
 #include <string.h>
 #include <sys/select.h>
+#include <asterisk/compat.h>

 #define AUDIO_FILENO (STDERR_FILENO + 1)

--- a/alaw.c
+++ b/alaw.c
@@ -1,17 +1,32 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * u-Law to Signed linear conversion
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
 */

-#include <asterisk/alaw.h>
+/*! \file
+ *
+ * \brief u-Law to Signed linear conversion
+ * 
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/alaw.h"

 #define AMI_MASK 0x55

--- a/app.c
+++ b/app.c
--- a/apps/Makefile
+++ b/apps/Makefile
@@ -1,57 +1,87 @@
 #
 # Asterisk -- A telephony toolkit for Linux.
 # 
-# Makefile for PBX frontends (dynamically loaded)
+# Makefile for PBX applications
 #
-# Copyright (C) 1999, Mark Spencer
+# Copyright (C) 1999-2005, Digium
 #
-# Mark Spencer <markster@linux-support.net>
+# Mark Spencer <markster@digium.com>
 #
 # This program is free software, distributed under the terms of
 # the GNU General Public License
 #

-USE_MYSQL_VM_INTERFACE=0
-USE_POSTGRES_VM_INTERFACE=0
-
-#APPS=app_dial.so app_playback.so app_directory.so app_intercom.so app_mp3.so 
 APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
     app_system.so app_echo.so app_record.so app_image.so app_url.so app_disa.so \
-     app_qcall.so app_adsiprog.so app_getcpeid.so app_milliwatt.so \
+     app_adsiprog.so app_getcpeid.so app_milliwatt.so \
     app_zapateller.so app_setcallerid.so app_festival.so \
-     app_queue.so app_senddtmf.so app_parkandannounce.so app_striplsd.so \
-     app_setcidname.so app_lookupcidname.so app_substring.so app_macro.so \
+     app_queue.so app_senddtmf.so app_parkandannounce.so \
+     app_setcidname.so app_lookupcidname.so app_macro.so \
     app_authenticate.so app_softhangup.so app_lookupblacklist.so \
     app_waitforring.so app_privacy.so app_db.so app_chanisavail.so \
     app_enumlookup.so app_transfer.so app_setcidnum.so app_cdr.so \
     app_hasnewvoicemail.so app_sayunixtime.so app_cut.so app_read.so \
     app_setcdruserfield.so app_random.so app_ices.so app_eval.so \
-     app_nbscat.so app_sendtext.so app_exec.so app_sms.so \
+     app_nbscat.so app_sendtext.so app_exec.so \
     app_groupcount.so app_txtcidname.so app_controlplayback.so \
     app_talkdetect.so app_alarmreceiver.so app_userevent.so app_verbose.so \
-     app_test.so app_forkcdr.so
-
-ifneq (${OSARCH},Darwin)
-APPS+=app_intercom.so
-endif
+     app_test.so app_forkcdr.so app_math.so app_realtime.so \
+     app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \
+     app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so \
+     app_dictate.so app_externalivr.so app_directed_pickup.so \
+     app_mixmonitor.so app_stack.so

+#
+# Obsolete things...
+#
 #APPS+=app_sql_postgres.so
 #APPS+=app_sql_odbc.so
+
+#
+# Experimental things
+#
+#APPS+=app_ivrdemo.so
+#APPS+=app_skel.so
 #APPS+=app_rpt.so

-APPS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo "app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so" ; fi)
-APPS+=$(shell if [ -f /usr/local/include/zaptel.h ]; then echo "app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so" ; fi)
-APPS+=$(shell if [ -f /usr/include/osp/osp.h ]; then echo "app_osplookup.so" ; fi)
+ifndef WITHOUT_ZAPTEL
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/zaptel.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/zaptel.h),)
+  APPS+=app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so app_page.so
+endif
+endif # WITHOUT_ZAPTEL

+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/osp/osp.h $(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
+APPS+=app_osplookup.so
+endif
+
+ifeq ($(findstring BSD,${OSARCH}),BSD)
+CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
+endif
+
+CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
+ifneq ($(shell if [[ 0x`$(CROSS_COMPILE_BIN)curl-config --vernum` -ge 0x70907 ]]; then echo "OK" ; fi),)
+  ifneq (${CURLLIBS},)
+    APPS+=app_curl.so
+  endif 
+endif
+
+ifeq (${OSARCH},CYGWIN)
+CYGSOLINK=-Wl,--out-implib=lib$@.a -Wl,--export-all-symbols
+CYGSOLIB=-L.. -L. -L../res -lasterisk.dll -lres_features.so -lres_adsi.so -lres_monitor.so
+else
 CFLAGS+=-fPIC
-
-ifeq ($(USE_POSTGRES_VM_INTERFACE),1)
-CFLAGS+=-DUSEPOSTGRESVM
+APPS+=app_sms.so
 endif

-ifeq ($(USE_MYSQL_VM_INTERFACE),1)
-CFLAGS+=-DUSEMYSQLVM
-endif
+#
+# If you have UnixODBC you can use ODBC voicemail
+# storage
+#
+# Uncomment to use ODBC storage
+#CFLAGS+=-DUSE_ODBC_STORAGE
+# Uncomment for extended ODBC voicemail storage
+#CFLAGS+=-DEXTENDED_ODBC_STORAGE
+# See doc/README.odbcstorage for more information

 all: $(APPS)

@@ -59,46 +89,45 @@ clean:
 	rm -f *.so *.o look .depend

 %.so : %.o
-	$(CC) $(SOLINK) -o $@ $<
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB}

 app_rpt.so : app_rpt.o
-	$(CC) $(SOLINK) -o $@ $< -ltonezone
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -ltonezone

 install: all
 	for x in $(APPS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done
 	rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
+	rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so

-app_voicemail.so : app_voicemail.o
-ifeq ($(USE_MYSQL_VM_INTERFACE),1)
-	$(CC) $(SOLINK) -o $@ $(MLFLAGS) $< -L/usr/lib/mysql -lmysqlclient -lz
-else
-ifeq ($(USE_POSTGRES_VM_INTERFACE),1)
-	$(CC) $(SOLINK) -o $@ $(MLFLAGS) $< -lpq
-else
-	$(CC) $(SOLINK) -o $@ $(MLFLAGS) $< 
-endif
-endif
+app_curl.so: app_curl.o
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} $(CURLLIBS)

 app_sql_postgres.o: app_sql_postgres.c
 	$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres.o app_sql_postgres.c

 app_sql_postgres.so: app_sql_postgres.o
-	$(CC) $(SOLINK) -o $@ $< -L/usr/local/pgsql/lib -lpq
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -L/usr/local/pgsql/lib -lpq

 app_sql_odbc.so: app_sql_odbc.o
-	$(CC) $(SOLINK) -o $@ $< -lodbc
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -lodbc

 look:	look.c
 	$(CC) -pipe -O6 -g look.c -o look -lncurses

+ifeq (SunOS,$(shell uname))
+app_chanspy.so: app_chanspy.o
+	$(CC) $(SOLINK) -o $@ $< -lrt
+endif
+
+
 ifneq ($(wildcard .depend),)
-include .depend
+ include .depend
 endif

 depend: .depend

 .depend:
-	../mkdep $(CFLAGS) `ls *.c`
+	../build_tools/mkdep $(CFLAGS) `ls *.c`

 env:
 	env
--- a/apps/app_adsiprog.c
+++ b/apps/app_adsiprog.c
@@ -1,26 +1,28 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Program Asterisk ADSI Scripts into phone
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief Program Asterisk ADSI Scripts into phone
+ * 
+ * \ingroup applications
 */

 #include <sys/types.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/adsi.h>
-#include <asterisk/options.h>
-#include <asterisk/utils.h>
-#include <asterisk/lock.h>
 #include <netinet/in.h>
 #include <stdlib.h>
 #include <unistd.h>
@@ -30,8 +32,19 @@
 #include <stdio.h>
 #include <errno.h>

-#include "../asterisk.h"
-#include "../astconf.h"
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/adsi.h"
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
+#include "asterisk/lock.h"

 static char *tdesc = "Asterisk ADSI Programming Application";

@@ -42,9 +55,8 @@ static char *synopsis = "Load Asterisk ADSI Scripts into phone";
 /* #define DUMP_MESSAGES */

 static char *descrip =
-"  ADSIProg(script): Programs an ADSI Phone with the given script.\n"
-"If none is specified, the default is used.  Returns 0 unless CPE\n" 
-"is hungup.\n";
+"  ADSIProg(script): This application programs an ADSI Phone with the given\n"
+"script. If nothing is specified, the default script (asterisk.adsi) is used.\n";

 STANDARD_LOCAL_USER;

@@ -159,9 +171,9 @@ struct adsi_script {
 	struct adsi_flag flags[7];

 	/* Stuff from adsi script */
-	char sec[5];
+	unsigned char sec[5];
 	char desc[19];
-	char fdn[5];
+	unsigned char fdn[5];
 	int ver;
 };

@@ -400,7 +412,7 @@ static struct adsi_flag *getflagbyname(struct adsi_script *state, char *name, ch
 		ast_log(LOG_WARNING, "No more flag space at line %d of %s\n", lineno, script);
 		return NULL;
 	}
-	strncpy(state->flags[state->numflags].vname, name, sizeof(state->flags[state->numflags].vname) - 1);
+	ast_copy_string(state->flags[state->numflags].vname, name, sizeof(state->flags[state->numflags].vname));
 	state->flags[state->numflags].id = state->numflags + 1;
 	state->numflags++;
 	return &state->flags[state->numflags-1];
@@ -503,7 +515,7 @@ static struct adsi_soft_key *getkeybyname(struct adsi_script *state, char *name,
 		ast_log(LOG_WARNING, "No more key space at line %d of %s\n", lineno, script);
 		return NULL;
 	}
-	strncpy(state->keys[state->numkeys].vname, name, sizeof(state->keys[state->numkeys].vname) - 1);
+	ast_copy_string(state->keys[state->numkeys].vname, name, sizeof(state->keys[state->numkeys].vname));
 	state->keys[state->numkeys].id = state->numkeys + 2;
 	state->numkeys++;
 	return &state->keys[state->numkeys-1];
@@ -519,7 +531,7 @@ static struct adsi_subscript *getsubbyname(struct adsi_script *state, char *name
 		ast_log(LOG_WARNING, "No more subscript space at line %d of %s\n", lineno, script);
 		return NULL;
 	}
-	strncpy(state->subs[state->numsubs].vname, name, sizeof(state->subs[state->numsubs].vname) - 1);
+	ast_copy_string(state->subs[state->numsubs].vname, name, sizeof(state->subs[state->numsubs].vname));
 	state->subs[state->numsubs].id = state->numsubs;
 	state->numsubs++;
 	return &state->subs[state->numsubs-1];
@@ -538,7 +550,7 @@ static struct adsi_state *getstatebyname(struct adsi_script *state, char *name,
 		ast_log(LOG_WARNING, "No more state space at line %d of %s\n", lineno, script);
 		return NULL;
 	}
-	strncpy(state->states[state->numstates].vname, name, sizeof(state->states[state->numstates].vname) - 1);
+	ast_copy_string(state->states[state->numstates].vname, name, sizeof(state->states[state->numstates].vname));
 	state->states[state->numstates].id = state->numstates + 1;
 	state->numstates++;
 	return &state->states[state->numstates-1];
@@ -557,7 +569,7 @@ static struct adsi_display *getdisplaybyname(struct adsi_script *state, char *na
 		ast_log(LOG_WARNING, "No more display space at line %d of %s\n", lineno, script);
 		return NULL;
 	}
-	strncpy(state->displays[state->numdisplays].vname, name, sizeof(state->displays[state->numdisplays].vname) - 1);
+	ast_copy_string(state->displays[state->numdisplays].vname, name, sizeof(state->displays[state->numdisplays].vname));
 	state->displays[state->numdisplays].id = state->numdisplays + 1;
 	state->numdisplays++;
 	return &state->displays[state->numdisplays-1];
@@ -1043,7 +1055,7 @@ static int adsi_process(struct adsi_script *state, char *buf, char *script, int
 					break;
 				}
 			} else {
-				strncpy(tmp2, tmp, sizeof(tmp2) - 1);
+				ast_copy_string(tmp2, tmp, sizeof(tmp2));
 			}
 			if (strlen(tmp2) > 18) {
 				ast_log(LOG_WARNING, "Truncating full name to 18 characters at line %d of %s\n", lineno, script);
@@ -1336,7 +1348,7 @@ static struct adsi_script *compile_script(char *script)
 	int x, err;
 	struct adsi_script *scr;
 	if (script[0] == '/')
-		strncpy(fn, script, sizeof(fn) - 1);
+		ast_copy_string(fn, script, sizeof(fn));
 	else
 		snprintf(fn, sizeof(fn), "%s/%s", (char *)ast_config_AST_CONFIG_DIR, script);
 	f = fopen(fn, "r");
@@ -1425,7 +1437,7 @@ static int adsi_prog(struct ast_channel *chan, char *script)
 {
 	struct adsi_script *scr;
 	int x;
-	char buf[1024];
+	unsigned char buf[1024];
 	int bytes;
 	scr = compile_script(script);
 	if (!scr) 
@@ -1540,9 +1552,12 @@ static int adsi_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
 	struct localuser *u;
-	if (!data || ast_strlen_zero(data))
-		data = "asterisk.adsi";
+
 	LOCAL_USER_ADD(u);
+	
+	if (ast_strlen_zero(data))
+		data = "asterisk.adsi";
+	
 	if (!adsi_available(chan)) {
 		if (option_verbose > 2)
 			ast_verbose(VERBOSE_PREFIX_3 "ADSI Unavailable on CPE.  Not bothering to try.\n");
@@ -1551,14 +1566,21 @@ static int adsi_exec(struct ast_channel *chan, void *data)
 			ast_verbose(VERBOSE_PREFIX_3 "ADSI Available on CPE.  Attempting Upload.\n");
 		res = adsi_prog(chan, data);
 	}
+
 	LOCAL_USER_REMOVE(u);
+	
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);	
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_alarmreceiver.c
+++ b/apps/app_alarmreceiver.c
@@ -1,38 +1,34 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Central Station Alarm receiver for Ademco Contact ID  
- * 
- * Copyright (C) 2004 Steve Rodgers
+ * Copyright (C)  2004 - 2005 Steve Rodgers
 *
 * Steve Rodgers <hwstar@rodgers.sdcoxmail.com>
 *
- * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief Central Station Alarm receiver for Ademco Contact ID  
+ * \author Steve Rodgers <hwstar@rodgers.sdcoxmail.com>
+ * 
 * *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** 
 *
 * Use at your own risk. Please consult the GNU GPL license document included with Asterisk details. *
 *
 * *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING ***
 *
+ * \ingroup applications
 */ 
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/ulaw.h>
-#include <asterisk/options.h>
-#include <asterisk/app.h>
-#include <asterisk/dsp.h>
-#include <asterisk/config.h>
-#include <asterisk/localtime.h>
-#include <asterisk/callerid.h>
-#include <asterisk/astdb.h>
 #include <string.h>
 #include <stdlib.h>
 #include <stdio.h>
@@ -41,6 +37,26 @@
 #include <unistd.h>
 #include <sys/time.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/ulaw.h"
+#include "asterisk/options.h"
+#include "asterisk/app.h"
+#include "asterisk/dsp.h"
+#include "asterisk/config.h"
+#include "asterisk/localtime.h"
+#include "asterisk/callerid.h"
+#include "asterisk/astdb.h"
+
 #define ALMRCV_CONFIG "alarmreceiver.conf"
 #define ADEMCO_CONTACT_ID "ADEMCO_CONTACT_ID"

@@ -57,13 +73,15 @@ static char *app = "AlarmReceiver";

 static char *synopsis = "Provide support for receving alarm reports from a burglar or fire alarm panel";
 static char *descrip =
-"Alarm receiver application for Asterisk. Only 1 signalling format is supported at this time:\n"
-"Ademco Contact ID. This application should be called whenever there is an alarm panel calling in\n"
-"to dump its events. The application will handshake with the alarm panel, and receive events,\n"
-"validate them, handshake them, and store them until the panel hangs up. Once the panel hangs up,\n"
-"the application will run the command line specified by the eventcmd setting in alarmreceiver.conf\n"
-"and pipe the events to the standard input of the application. Alarmreceiver.conf also contains settings\n"
-"for DTMF timing, and for the loudness of the acknowledgement tones.\n";
+"  AlarmReceiver(): Only 1 signalling format is supported at this time: Ademco\n"
+"Contact ID. This application should be called whenever there is an alarm\n"
+"panel calling in to dump its events. The application will handshake with the\n"
+"alarm panel, and receive events, validate them, handshake them, and store them\n"
+"until the panel hangs up. Once the panel hangs up, the application will run the\n"
+"system command specified by the eventcmd setting in alarmreceiver.conf and pipe\n"
+"the events to the standard input of the application. The configuration file also\n"
+"contains settings for DTMF timing, and for the loudness of the acknowledgement\n" 
+"tones.\n";

 /* Config Variables */

@@ -103,7 +121,7 @@ static void database_increment( char *key )
 	char value[16];
 	
 	
-	if(!strlen(db_family))
+	if (ast_strlen_zero(db_family))
 		return; /* If not defined, don't do anything */
 	
 	res = ast_db_get(db_family, key, value, sizeof(value) - 1);
@@ -213,20 +231,6 @@ static int send_tone_burst(struct ast_channel *chan, float freq, int duration, i
 	return res;
 }

-/*
-* Return the difference in milliseconds between two timeval structs
-*/
-
-static int ms_diff(struct timeval *tv1, struct timeval *tv2){
-
-	int	ms;
-	
-	ms = (tv1->tv_sec - tv2->tv_sec) * 1000;
-	ms += (tv1->tv_usec - tv2->tv_usec) / 1000;
-	
-	return(ms);
-}
-
 /*
 * Receive a string of DTMF digits where the length of the digit string is known in advance. Do not give preferential
 * treatment to any digit value, and allow separate time out values to be specified for the first digit and all subsequent
@@ -244,14 +248,12 @@ static int receive_dtmf_digits(struct ast_channel *chan, char *digit_string, int
 	int i = 0;
 	int r;
 	struct ast_frame *f;
-	struct timeval now, lastdigittime;
+	struct timeval lastdigittime;
 	
-	gettimeofday(&lastdigittime,NULL);
+	lastdigittime = ast_tvnow();
 	for(;;){
-		gettimeofday(&now,NULL);
-		
 		  /* if outa time, leave */
-		if (ms_diff(&now,&lastdigittime) > 
+		if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) >
 		    ((i > 0) ? sdto : fdto)){
 			if(option_verbose >= 4)
 				ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: DTMF Digit Timeout on %s\n", chan->name);
@@ -296,7 +298,7 @@ static int receive_dtmf_digits(struct ast_channel *chan, char *digit_string, int
 		if(i >= length)
 			break;
 		
-		gettimeofday(&lastdigittime,NULL);
+		lastdigittime = ast_tvnow();
 	}
 	
 	digit_string[i] = '\0'; /* Nul terminate the end of the digit string */
@@ -318,8 +320,8 @@ static int write_metadata( FILE *logfile, char *signalling_type, struct ast_chan
 	char timestamp[80];
 	
 	/* Extract the caller ID location */
-	
-	strncpy(workstring, chan->callerid, sizeof(workstring) - 1);
+	if (chan->cid.cid_num)
+		ast_copy_string(workstring, chan->cid.cid_num, sizeof(workstring));
 	workstring[sizeof(workstring) - 1] = '\0';
 	
 	ast_callerid_parse(workstring, &cn, &cl);
@@ -394,11 +396,11 @@ static int log_events(struct ast_channel *chan,  char *signalling_type, event_no
 	FILE *logfile;
 	event_node_t *elp = event;
 	
-	if(strlen(event_spool_dir)){
+	if (!ast_strlen_zero(event_spool_dir)) {
 		
 		/* Make a template */
 		
-		strncpy(workstring, event_spool_dir, sizeof(workstring) - 1);
+		ast_copy_string(workstring, event_spool_dir, sizeof(workstring));
 		strncat(workstring, event_file, sizeof(workstring) - strlen(workstring) - 1);
 		
 		/* Make the temporary file */
@@ -518,7 +520,7 @@ static int receive_ademco_contact_id( struct ast_channel *chan, void *data, int
 				ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Incomplete string: %s, trying again...\n", event);

 			if(!got_some_digits){
-				got_some_digits = (strlen(event)) ? 1 : 0;
+				got_some_digits = (!ast_strlen_zero(event)) ? 1 : 0;
 				ack_retries++;
 			}
 			continue;	
@@ -590,7 +592,7 @@ static int receive_ademco_contact_id( struct ast_channel *chan, void *data, int
 		memset(enew, 0, sizeof(event_node_t));
 		
 		enew->next = NULL;
-		strncpy(enew->data, event, sizeof(enew->data) - 1);
+		ast_copy_string(enew->data, event, sizeof(enew->data));

 		/*
 		* Insert event onto end of list
@@ -653,17 +655,19 @@ static int alarmreceiver_exec(struct ast_channel *chan, void *data)

 	if (ast_set_write_format(chan,AST_FORMAT_ULAW)){
 		ast_log(LOG_WARNING, "AlarmReceiver: Unable to set write format to Mu-law on %s\n",chan->name);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	
 	if (ast_set_read_format(chan,AST_FORMAT_ULAW)){
 		ast_log(LOG_WARNING, "AlarmReceiver: Unable to set read format to Mu-law on %s\n",chan->name);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}

 	/* Set default values for this invokation of the application */
 	
-	strncpy(signalling_type, ADEMCO_CONTACT_ID, sizeof(signalling_type) - 1);
+	ast_copy_string(signalling_type, ADEMCO_CONTACT_ID, sizeof(signalling_type));


 	/* Answer the channel if it is not already */
@@ -715,7 +719,7 @@ static int alarmreceiver_exec(struct ast_channel *chan, void *data)
 	* Do we exec a command line at the end?
 	*/
 	
-	if((!res) && (strlen(event_app)) && (event_head)){
+	if((!res) && (!ast_strlen_zero(event_app)) && (event_head)){
 		ast_log(LOG_DEBUG,"Alarmreceiver: executing: %s\n", event_app);
 		ast_safe_system(event_app);
 	}
@@ -747,7 +751,7 @@ static int load_config(void)

 	/* Read in the config file */

-	cfg = ast_load(ALMRCV_CONFIG);
+	cfg = ast_config_load(ALMRCV_CONFIG);
                                                                                                                                  
 	if(!cfg){
 	
@@ -760,7 +764,7 @@ static int load_config(void)
 		p = ast_variable_retrieve(cfg, "general", "eventcmd");
 		
 		if(p){
-			strncpy(event_app, p, sizeof(event_app) - 1);
+			ast_copy_string(event_app, p, sizeof(event_app));
 			event_app[sizeof(event_app) - 1] = '\0';
 		}
 		
@@ -800,24 +804,24 @@ static int load_config(void)
 		p = ast_variable_retrieve(cfg, "general", "eventspooldir");
 			
 		if(p){
-			strncpy(event_spool_dir, p, sizeof(event_spool_dir) - 1);
+			ast_copy_string(event_spool_dir, p, sizeof(event_spool_dir));
 			event_spool_dir[sizeof(event_spool_dir) - 1] = '\0';
 		}
 		
 		p = ast_variable_retrieve(cfg, "general", "timestampformat");
 			
 		if(p){
-			strncpy(time_stamp_format, p, sizeof(time_stamp_format) - 1);
+			ast_copy_string(time_stamp_format, p, sizeof(time_stamp_format));
 			time_stamp_format[sizeof(time_stamp_format) - 1] = '\0';
 		}

 		p = ast_variable_retrieve(cfg, "general", "db-family");
                                                                                                                                            
 		if(p){
-			strncpy(db_family, p, sizeof(db_family) - 1);
+			ast_copy_string(db_family, p, sizeof(db_family));
 			db_family[sizeof(db_family) - 1] = '\0';
 		}
-		ast_destroy(cfg);
+		ast_config_destroy(cfg);
 	}
 	return 0;

@@ -830,8 +834,13 @@ static int load_config(void)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_authenticate.c
+++ b/apps/app_authenticate.c
@@ -1,32 +1,47 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Execute arbitrary authenticate commands
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief Execute arbitrary authenticate commands
+ * 
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/app.h>
-#include <asterisk/astdb.h>
-#include <asterisk/utils.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <errno.h>
-#include <stdlib.h>
 #include <stdio.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+#include "asterisk/astdb.h"
+#include "asterisk/utils.h"
+
 static char *tdesc = "Authentication Application";

 static char *app = "Authenticate";
@@ -34,20 +49,24 @@ static char *app = "Authenticate";
 static char *synopsis = "Authenticate a user";

 static char *descrip =
-"  Authenticate(password[|options]): Requires a user to enter a"
-"given password in order to continue execution.  If the\n"
-"password begins with the '/' character, it is interpreted as\n"
-"a file which contains a list of valid passwords (1 per line).\n"
-"an optional set of opions may be provided by concatenating any\n"
-"of the following letters:\n"
-"     a - Set account code to the password that is entered\n"
-"     d - Interpret path as database key, not literal file\n"
-"     r - Remove database key upon successful entry (valid with 'd' only)\n"
-"\n"
-"When using a database key, the value associated with the key can be\n"
-"anything.\n"
-"Returns 0 if the user enters a valid password within three\n"
-"tries, or -1 otherwise (or on hangup).\n";
+"  Authenticate(password[|options]): This application asks the caller to enter a\n"
+"given password in order to continue dialplan execution. If the password begins\n"
+"with the '/' character, it is interpreted as a file which contains a list of\n"
+"valid passwords, listed 1 password per line in the file.\n"
+"  When using a database key, the value associated with the key can be anything.\n"
+"Users have three attempts to authenticate before the channel is hung up. If the\n"
+"passsword is invalid, the 'j' option is specified, and priority n+101 exists,\n"
+"dialplan execution will continnue at this location.\n"
+"  Options:\n"
+"     a - Set the channels' account code to the password that is entered\n"
+"     d - Interpret the given path as database key, not a literal file\n"
+"     j - Support jumping to n+101 if authentication fails\n"
+"     m - Interpret the given path as a file which contains a list of account\n"
+"         codes and password hashes delimited with ':', listed one per line in\n"
+"         the file. When one of the passwords is matched, the channel will have\n"
+"         its account code set to the corresponding account code in the file.\n"
+"     r - Remove the database key upon successful entry (valid with 'd' only)\n"
+;

 STANDARD_LOCAL_USER;

@@ -56,17 +75,21 @@ LOCAL_USER_DECL;
 static int auth_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
+	int jump = 0;
 	int retries;
 	struct localuser *u;
 	char password[256]="";
 	char passwd[256];
 	char *opts;
 	char *prompt;
-	if (!data || ast_strlen_zero(data)) {
+	
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Authenticate requires an argument(password)\n");
 		return -1;
 	}
+	
 	LOCAL_USER_ADD(u);
+
 	if (chan->_state != AST_STATE_UP) {
 		res = ast_answer(chan);
 		if (res) {
@@ -74,6 +97,7 @@ static int auth_exec(struct ast_channel *chan, void *data)
 			return -1;
 		}
 	}
+	
 	strncpy(password, data, sizeof(password) - 1);
 	opts=strchr(password, '|');
 	if (opts) {
@@ -81,6 +105,8 @@ static int auth_exec(struct ast_channel *chan, void *data)
 		opts++;
 	} else
 		opts = "";
+	if (strchr(opts, 'j'))
+		jump = 1;
 	/* Start asking for password */
 	prompt = "agent-pass";
 	for (retries = 0; retries < 3; retries++) {
@@ -105,17 +131,44 @@ static int auth_exec(struct ast_channel *chan, void *data)
 				f = fopen(password, "r");
 				if (f) {
 					char buf[256] = "";
-					while(!feof(f)) {
+					char md5passwd[33] = "";
+					char *md5secret = NULL;
+
+					while (!feof(f)) {
 						fgets(buf, sizeof(buf), f);
 						if (!feof(f) && !ast_strlen_zero(buf)) {
 							buf[strlen(buf) - 1] = '\0';
-							if (!ast_strlen_zero(buf) && !strcmp(passwd, buf))
-								break;
+							if (strchr(opts, 'm')) {
+								md5secret = strchr(buf, ':');
+								if (md5secret == NULL)
+									continue;
+								*md5secret = '\0';
+								md5secret++;
+								ast_md5_hash(md5passwd, passwd);
+								if (!strcmp(md5passwd, md5secret)) {
+									if (strchr(opts, 'a'))
+										ast_cdr_setaccount(chan, buf);
+									break;
+								}
+							} else {
+								if (!strcmp(passwd, buf)) {
+									if (strchr(opts, 'a'))
+										ast_cdr_setaccount(chan, buf);
+									break;
+								}
+							}
 						}
 					}
 					fclose(f);
-					if (!ast_strlen_zero(buf) && !strcmp(passwd, buf))
-						break;
+					if (!ast_strlen_zero(buf)) {
+						if (strchr(opts, 'm')) {
+							if (md5secret && !strcmp(md5passwd, md5secret))
+								break;
+						} else {
+							if (!strcmp(passwd, buf))
+								break;
+						}
+					}
 				} else 
 					ast_log(LOG_WARNING, "Unable to open file '%s' for authentication: %s\n", password, strerror(errno));
 			}
@@ -127,17 +180,19 @@ static int auth_exec(struct ast_channel *chan, void *data)
 		prompt="auth-incorrect";
 	}
 	if ((retries < 3) && !res) {
-		if (strchr(opts, 'a')) 
+		if (strchr(opts, 'a') && !strchr(opts, 'm')) 
 			ast_cdr_setaccount(chan, passwd);
 		res = ast_streamfile(chan, "auth-thankyou", chan->language);
 		if (!res)
 			res = ast_waitstream(chan, "");
 	} else {
-		if (!res)
-			res = ast_streamfile(chan, "vm-goodbye", chan->language);
-		if (!res)
-			res = ast_waitstream(chan, "");
-		res = -1;
+		if (jump && ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
+			res = 0;
+		} else {
+			if (!ast_streamfile(chan, "vm-goodbye", chan->language))
+				res = ast_waitstream(chan, "");
+			res = -1;
+		}
 	}
 	LOCAL_USER_REMOVE(u);
 	return res;
@@ -145,8 +200,13 @@ static int auth_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+	
+	return res;
 }

 int load_module(void)
--- a/apps/app_cdr.c
+++ b/apps/app_cdr.c
@@ -1,28 +1,47 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Applictions connected with CDR engine
- * 
- * Copyright (C) 2003, Digium
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Martin Pycko <martinp@digium.com>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief Applications connected with CDR engine
+ *
+ * \ingroup applications
 */

 #include <sys/types.h>
-#include <asterisk/channel.h>
-#include <asterisk/module.h>
-#include <asterisk/pbx.h>
 #include <stdlib.h>

+#include "asterisk.h"

-static char *tdesc = "Make sure asterisk doesn't save CDR for a certain call";
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/channel.h"
+#include "asterisk/module.h"
+#include "asterisk/pbx.h"
+
+
+static char *tdesc = "Tell Asterisk to not maintain a CDR for the current call";
+
+static char *nocdr_descrip = 
+"  NoCDR(): This application will tell Asterisk not to maintain a CDR for the\n"
+"current call.\n";

-static char *nocdr_descrip = "NoCDR(): makes sure there won't be any CDR written for a certain call";
 static char *nocdr_app = "NoCDR";
-static char *nocdr_synopsis = "Make sure asterisk doesn't save CDR for a certain call";
+static char *nocdr_synopsis = "Tell Asterisk to not maintain a CDR for the current call";

 STANDARD_LOCAL_USER;

@@ -30,17 +49,29 @@ LOCAL_USER_DECL;

 static int nocdr_exec(struct ast_channel *chan, void *data)
 {
+	struct localuser *u;
+	
+	LOCAL_USER_ADD(u);
+
 	if (chan->cdr) {
 		ast_cdr_free(chan->cdr);
 		chan->cdr = NULL;
 	}
+
+	LOCAL_USER_REMOVE(u);
+
 	return 0;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(nocdr_app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(nocdr_app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_chanisavail.c
+++ b/apps/app_chanisavail.c
@@ -1,49 +1,65 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
- *
- * Check if Channel is Available
- * 
- * Copyright (C) 2003, Digium
- *
- * Mark Spencer <markster@digium.com>
- * James Golovich <james@gnuinter.net>
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License
- *
+* Asterisk -- An open source telephony toolkit.
+*
+* Copyright (C) 1999 - 2005, Digium, Inc.
+*
+* Mark Spencer <markster@digium.com>
+* James Golovich <james@gnuinter.net>
+*
+* See http://www.asterisk.org for more information about
+* the Asterisk project. Please do not directly contact
+* any of the maintainers of this project for assistance;
+* the project provides a web site, mailing lists and IRC
+* channels for your use.
+*
+* This program is free software, distributed under the terms of
+* the GNU General Public License Version 2. See the LICENSE file
+* at the top of the source tree.
+*/
+
+/*! \file
+* \brief Check if Channel is Available
+* 
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/app.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 #include <unistd.h>
 #include <errno.h>
-#include <string.h>
-#include <stdlib.h>
 #include <sys/ioctl.h>

-static char *tdesc = "Check if channel is available";
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+#include "asterisk/devicestate.h"
+#include "asterisk/options.h"
+
+static char *tdesc = "Check channel availability";

 static char *app = "ChanIsAvail";

-static char *synopsis = "Check if channel is available";
+static char *synopsis = "Check channel availability";

 static char *descrip = 
-"  ChanIsAvail(Technology/resource[&Technology2/resource2...]): \n"
-"Checks is any of the requested channels are available.  If none\n"
-"of the requested channels are available the new priority will be\n"
-"n+101 (unless such a priority does not exist or on error, in which\n"
-"case ChanIsAvail will return -1).\n"
-"If any of the requested channels are available, the next priority will be n+1,\n"
-"the channel variable ${AVAILCHAN} will be set to the name of the available channel\n"
-"and the ChanIsAvail app will return 0.\n"
-"${AVAILORIGCHAN} is the canonical channel name that was used to create the channel.\n"
-"${AVAILSTATUS} is the status code for the channel.\n";
+"  ChanIsAvail(Technology/resource[&Technology2/resource2...][|options]): \n"
+"This application will check to see if any of the specified channels are\n"
+"available. The following variables will be set by this application:\n"
+"  ${AVAILCHAN}     - the name of the available channel, if one exists\n"
+"  ${AVAILORIGCHAN} - the canonical channel name that was used to create the channel\n"
+"  ${AVAILSTATUS}   - the status code for the available channel\n"
+"  Options:\n"
+"    s - Consider the channel unavailable if the channel is in use at all\n"
+"    j - Support jumping to priority n+101 if no channel is available\n";

 STANDARD_LOCAL_USER;

@@ -51,18 +67,29 @@ LOCAL_USER_DECL;

 static int chanavail_exec(struct ast_channel *chan, void *data)
 {
-	int res=-1;
+	int res=-1, inuse=-1, option_state=0, priority_jump=0;
+	int status;
 	struct localuser *u;
-	char info[512], tmp[512], *peers, *tech, *number, *rest, *cur;
+	char *info, tmp[512], trychan[512], *peers, *tech, *number, *rest, *cur, *options, *stringp;
 	struct ast_channel *tempchan;

-	if (!data) {
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "ChanIsAvail requires an argument (Zap/1&Zap/2)\n");
 		return -1;
 	}
+
 	LOCAL_USER_ADD(u);

-	strncpy(info, (char *)data, sizeof(info)-1);
+	info = ast_strdupa(data); 
+	stringp = info;
+	strsep(&stringp, "|");
+	options = strsep(&stringp, "|");
+	if (options) {
+		if (strchr(options, 's'))
+			option_state = 1;
+		if (strchr(options, 'j'))
+			priority_jump = 1;
+	}
 	peers = info;
 	if (peers) {
 		cur = peers;
@@ -77,31 +104,47 @@ static int chanavail_exec(struct ast_channel *chan, void *data)
 			number = strchr(tech, '/');
 			if (!number) {
 				ast_log(LOG_WARNING, "ChanIsAvail argument takes format ([technology]/[device])\n");
+				LOCAL_USER_REMOVE(u);
 				return -1;
 			}
 			*number = '\0';
 			number++;
-			if ((tempchan = ast_request(tech, chan->nativeformats, number))) {
+			
+			if (option_state) {
+				/* If the pbx says in use then don't bother trying further.
+				   This is to permit testing if someone's on a call, even if the 
+	 			   channel can permit more calls (ie callwaiting, sip calls, etc).  */
+                               
+				snprintf(trychan, sizeof(trychan), "%s/%s",cur,number);
+				status = inuse = ast_device_state(trychan);
+			}
+			if ((inuse <= 1) && (tempchan = ast_request(tech, chan->nativeformats, number, &status))) {
 					pbx_builtin_setvar_helper(chan, "AVAILCHAN", tempchan->name);
 					/* Store the originally used channel too */
 					snprintf(tmp, sizeof(tmp), "%s/%s", tech, number);
 					pbx_builtin_setvar_helper(chan, "AVAILORIGCHAN", tmp);
+					snprintf(tmp, sizeof(tmp), "%d", status);
+					pbx_builtin_setvar_helper(chan, "AVAILSTATUS", tmp);
 					ast_hangup(tempchan);
 					tempchan = NULL;
 					res = 1;
 					break;
+			} else {
+				snprintf(tmp, sizeof(tmp), "%d", status);
+				pbx_builtin_setvar_helper(chan, "AVAILSTATUS", tmp);
 			}
 			cur = rest;
 		} while (cur);
 	}
-
 	if (res < 1) {
 		pbx_builtin_setvar_helper(chan, "AVAILCHAN", "");
 		pbx_builtin_setvar_helper(chan, "AVAILORIGCHAN", "");
-		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-			chan->priority+=100;
-		else
-			return -1;
+		if (priority_jump || option_priority_jumping) {
+			if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
+				LOCAL_USER_REMOVE(u);
+				return -1;
+			}
+		}
 	}

 	LOCAL_USER_REMOVE(u);
@@ -110,8 +153,13 @@ static int chanavail_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res = 0;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+	
+	return res;
 }

 int load_module(void)
--- a/apps/app_chanspy.c
+++ b/apps/app_chanspy.c
@@ -0,0 +1,587 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2005 Anthony Minessale II (anthmct@yahoo.com)
+ *
+ * Disclaimed to Digium
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief ChanSpy: Listen in on any channel.
+ * 
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <ctype.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/chanspy.h"
+#include "asterisk/features.h"
+#include "asterisk/options.h"
+#include "asterisk/app.h"
+#include "asterisk/utils.h"
+#include "asterisk/say.h"
+#include "asterisk/pbx.h"
+#include "asterisk/translate.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+
+AST_MUTEX_DEFINE_STATIC(modlock);
+
+#define AST_NAME_STRLEN 256
+#define ALL_DONE(u, ret) LOCAL_USER_REMOVE(u); return ret;
+#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
+
+static const char *synopsis = "Listen to the audio of an active channel\n";
+static const char *app = "ChanSpy";
+static const char *desc = 
+"  ChanSpy([chanprefix][|options]): This application is used to listen to the\n"
+"audio from an active Asterisk channel. This includes the audio coming in and\n"
+"out of the channel being spied on. If the 'chanprefix' parameter is specified,\n"
+"only channels beginning with this string will be spied upon.\n"
+"  While Spying, the following actions may be performed:\n"
+"    - Dialing # cycles the volume level.\n"
+"    - Dialing * will stop spying and look for another channel to spy on.\n"
+"    - Dialing a series of digits followed by # builds a channel name to append\n"
+"      to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing\n"
+"      the digits '1234#' while spying will begin spying on the channel,\n"
+"      'Agent/1234'.\n"
+"  Options:\n"
+"    b - Only spy on channels involved in a bridged call.\n"
+"    g(grp) - Match only channels where their ${SPYGROUP} variable is set to\n"
+"             'grp'.\n"
+"    q - Don't play a beep when beginning to spy on a channel.\n"
+"    r[(basename)] - Record the session to the monitor spool directory. An\n"
+"                    optional base for the filename may be specified. The\n"
+"                    default is 'chanspy'.\n"
+"    v([value]) - Adjust the initial volume in the range from -4 to 4. A\n"
+"                 negative value refers to a quieter setting.\n"
+;
+
+static const char *chanspy_spy_type = "ChanSpy";
+
+enum {
+	OPTION_QUIET	 = (1 << 0),	/* Quiet, no announcement */
+	OPTION_BRIDGED   = (1 << 1),	/* Only look at bridged calls */
+	OPTION_VOLUME    = (1 << 2),	/* Specify initial volume */
+	OPTION_GROUP     = (1 << 3),	/* Only look at channels in group */
+	OPTION_RECORD    = (1 << 4),	/* Record */
+} chanspy_opt_flags;
+
+enum {
+	OPT_ARG_VOLUME = 0,
+	OPT_ARG_GROUP,
+	OPT_ARG_RECORD,
+	OPT_ARG_ARRAY_SIZE,
+} chanspy_opt_args;
+
+AST_APP_OPTIONS(chanspy_opts, {
+	AST_APP_OPTION('q', OPTION_QUIET),
+	AST_APP_OPTION('b', OPTION_BRIDGED),
+	AST_APP_OPTION_ARG('v', OPTION_VOLUME, OPT_ARG_VOLUME),
+	AST_APP_OPTION_ARG('g', OPTION_GROUP, OPT_ARG_GROUP),
+	AST_APP_OPTION_ARG('r', OPTION_RECORD, OPT_ARG_RECORD),
+});
+
+STANDARD_LOCAL_USER;
+LOCAL_USER_DECL;
+
+struct chanspy_translation_helper {
+	/* spy data */
+	struct ast_channel_spy spy;
+	int fd;
+	int volfactor;
+};
+
+static struct ast_channel *local_channel_walk(struct ast_channel *chan) 
+{
+	struct ast_channel *ret;
+	ast_mutex_lock(&modlock);	
+	if ((ret = ast_channel_walk_locked(chan))) {
+		ast_mutex_unlock(&ret->lock);
+	}
+	ast_mutex_unlock(&modlock);			
+	return ret;
+}
+
+static struct ast_channel *local_get_channel_begin_name(char *name) 
+{
+	struct ast_channel *chan, *ret = NULL;
+	ast_mutex_lock(&modlock);
+	chan = local_channel_walk(NULL);
+	while (chan) {
+		if (!strncmp(chan->name, name, strlen(name))) {
+			ret = chan;
+			break;
+		}
+		chan = local_channel_walk(chan);
+	}
+	ast_mutex_unlock(&modlock);
+	
+	return ret;
+}
+
+static void *spy_alloc(struct ast_channel *chan, void *data)
+{
+	/* just store the data pointer in the channel structure */
+	return data;
+}
+
+static void spy_release(struct ast_channel *chan, void *data)
+{
+	/* nothing to do */
+}
+
+static int spy_generate(struct ast_channel *chan, void *data, int len, int samples) 
+{
+	struct chanspy_translation_helper *csth = data;
+	struct ast_frame *f;
+		
+	if (csth->spy.status != CHANSPY_RUNNING)
+		/* Channel is already gone more than likely */
+		return -1;
+
+	ast_mutex_lock(&csth->spy.lock);
+	f = ast_channel_spy_read_frame(&csth->spy, samples);
+	ast_mutex_unlock(&csth->spy.lock);
+		
+	if (!f)
+		return 0;
+		
+	if (ast_write(chan, f)) {
+		ast_frfree(f);
+		return -1;
+	}
+
+	if (csth->fd)
+		write(csth->fd, f->data, f->datalen);
+
+	ast_frfree(f);
+
+	return 0;
+}
+
+
+static struct ast_generator spygen = {
+	.alloc = spy_alloc,
+	.release = spy_release,
+	.generate = spy_generate, 
+};
+
+static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, struct ast_channel_spy *spy) 
+{
+	int res;
+	struct ast_channel *peer;
+
+	ast_log(LOG_NOTICE, "Attaching %s to %s\n", spychan->name, chan->name);
+
+	ast_mutex_lock(&chan->lock);
+	res = ast_channel_spy_add(chan, spy);
+	ast_mutex_unlock(&chan->lock);
+
+	if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan))) {
+		ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);	
+	}
+
+	return res;
+}
+
+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....
+	   DON'T TOUCH IT!!!  RUN AWAY!!! */
+	if (spy->status != CHANSPY_RUNNING)
+		return;
+
+	if (!chan)
+		return;
+
+	ast_mutex_lock(&chan->lock);
+	ast_channel_spy_remove(chan, spy);
+	ast_mutex_unlock(&chan->lock);
+};
+
+/* Map 'volume' levels from -4 through +4 into
+   decibel (dB) settings for channel drivers
+*/
+static signed char volfactor_map[] = {
+	-24,
+	-18,
+	-12,
+	-6,
+	0,
+	6,
+	12,
+	18,
+	24,
+};
+
+/* attempt to set the desired gain adjustment via the channel driver;
+   if successful, clear it out of the csth structure so the
+   generator will not attempt to do the adjustment itself
+*/
+static void set_volume(struct ast_channel *chan, struct chanspy_translation_helper *csth)
+{
+	signed char volume_adjust = volfactor_map[csth->volfactor + 4];
+
+	if (!ast_channel_setoption(chan, AST_OPTION_TXGAIN, &volume_adjust, sizeof(volume_adjust), 0))
+		csth->volfactor = 0;
+}
+
+static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int *volfactor, int fd) 
+{
+	struct chanspy_translation_helper csth;
+	int running, res = 0, x = 0;
+	char inp[24];
+	char *name=NULL;
+	struct ast_frame *f;
+
+	running = (chan && !ast_check_hangup(chan) && spyee && !ast_check_hangup(spyee));
+
+	if (running) {
+		memset(inp, 0, sizeof(inp));
+		name = ast_strdupa(spyee->name);
+		if (option_verbose >= 2)
+			ast_verbose(VERBOSE_PREFIX_2 "Spying on channel %s\n", name);
+
+		memset(&csth, 0, sizeof(csth));
+		ast_set_flag(&csth.spy, CHANSPY_FORMAT_AUDIO);
+		ast_set_flag(&csth.spy, CHANSPY_TRIGGER_NONE);
+		ast_set_flag(&csth.spy, CHANSPY_MIXAUDIO);
+		csth.spy.type = chanspy_spy_type;
+		csth.spy.status = CHANSPY_RUNNING;
+		csth.spy.read_queue.format = AST_FORMAT_SLINEAR;
+		csth.spy.write_queue.format = AST_FORMAT_SLINEAR;
+		ast_mutex_init(&csth.spy.lock);
+		csth.volfactor = *volfactor;
+		set_volume(chan, &csth);
+		csth.spy.read_vol_adjustment = csth.volfactor;
+		csth.spy.write_vol_adjustment = csth.volfactor;
+		csth.fd = fd;
+
+		if (start_spying(spyee, chan, &csth.spy))
+			running = 0;
+	}
+
+	if (running) {
+		running = 1;
+		ast_activate_generator(chan, &spygen, &csth);
+
+		while (csth.spy.status == CHANSPY_RUNNING &&
+		       chan && !ast_check_hangup(chan) &&
+		       spyee &&
+		       !ast_check_hangup(spyee) &&
+		       running == 1 &&
+		       (res = ast_waitfor(chan, -1) > -1)) {
+			if ((f = ast_read(chan))) {
+				res = 0;
+				if (f->frametype == AST_FRAME_DTMF) {
+					res = f->subclass;
+				}
+				ast_frfree(f);
+				if (!res) {
+					continue;
+				}
+			} else {
+				break;
+			}
+			if (x == sizeof(inp)) {
+				x = 0;
+			}
+			if (res < 0) {
+				running = -1;
+			}
+			if (res == 0) {
+				continue;
+			} else if (res == '*') {
+				running = 0; 
+			} else if (res == '#') {
+				if (!ast_strlen_zero(inp)) {
+					running = x ? atoi(inp) : -1;
+					break;
+				} else {
+					(*volfactor)++;
+					if (*volfactor > 4) {
+						*volfactor = -4;
+					}
+					if (option_verbose > 2) {
+						ast_verbose(VERBOSE_PREFIX_3 "Setting spy volume on %s to %d\n", chan->name, *volfactor);
+					}
+					csth.volfactor = *volfactor;
+					set_volume(chan, &csth);
+					csth.spy.read_vol_adjustment = csth.volfactor;
+					csth.spy.write_vol_adjustment = csth.volfactor;
+				}
+			} else if (res >= 48 && res <= 57) {
+				inp[x++] = res;
+			}
+		}
+		ast_deactivate_generator(chan);
+		stop_spying(spyee, &csth.spy);
+
+		if (option_verbose >= 2) {
+			ast_verbose(VERBOSE_PREFIX_2 "Done Spying on channel %s\n", name);
+		}
+	} else {
+		running = 0;
+	}
+
+	ast_mutex_destroy(&csth.spy.lock);
+
+	return running;
+}
+
+static int chanspy_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	struct ast_channel *peer=NULL, *prev=NULL;
+	char name[AST_NAME_STRLEN],
+		peer_name[AST_NAME_STRLEN + 5],
+		*args,
+		*ptr = NULL,
+		*options = NULL,
+		*spec = NULL,
+		*argv[5],
+		*mygroup = NULL,
+		*recbase = NULL;
+	int res = -1,
+		volfactor = 0,
+		silent = 0,
+		argc = 0,
+		bronly = 0,
+		chosen = 0,
+		count=0,
+		waitms = 100,
+		num = 0,
+		oldrf = 0,
+		oldwf = 0,
+		fd = 0;
+	struct ast_flags flags;
+	signed char zero_volume = 0;
+
+	if (!(args = ast_strdupa((char *)data))) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	oldrf = chan->readformat;
+	oldwf = chan->writeformat;
+	if (ast_set_read_format(chan, AST_FORMAT_SLINEAR) < 0) {
+		ast_log(LOG_ERROR, "Could Not Set Read Format.\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	
+	if (ast_set_write_format(chan, AST_FORMAT_SLINEAR) < 0) {
+		ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	ast_answer(chan);
+
+	ast_set_flag(chan, AST_FLAG_SPYING); /* so nobody can spy on us while we are spying */
+
+	if ((argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0])))) {
+		spec = argv[0];
+		if ( argc > 1) {
+			options = argv[1];
+		}
+		if (ast_strlen_zero(spec) || !strcmp(spec, "all")) {
+			spec = NULL;
+		}
+	}
+	
+	if (options) {
+		char *opts[OPT_ARG_ARRAY_SIZE];
+		ast_app_parse_options(chanspy_opts, &flags, opts, options);
+		if (ast_test_flag(&flags, OPTION_GROUP)) {
+			mygroup = opts[1];
+		}
+		if (ast_test_flag(&flags, OPTION_RECORD)) {
+			if (!(recbase = opts[2])) {
+				recbase = "chanspy";
+			}
+		}
+		silent = ast_test_flag(&flags, OPTION_QUIET);
+		bronly = ast_test_flag(&flags, OPTION_BRIDGED);
+		if (ast_test_flag(&flags, OPTION_VOLUME) && opts[1]) {
+			int vol;
+
+			if ((sscanf(opts[0], "%d", &vol) != 1) || (vol > 4) || (vol < -4))
+				ast_log(LOG_NOTICE, "Volume factor must be a number between -4 and 4\n");
+			else
+				volfactor = vol;
+			}
+	}
+
+	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) {
+			ast_log(LOG_WARNING, "Cannot open %s for recording\n", filename);
+			fd = 0;
+		}
+	}
+
+	for(;;) {
+		if (!silent) {
+			res = ast_streamfile(chan, "beep", chan->language);
+			if (!res)
+				res = ast_waitstream(chan, "");
+			if (res < 0) {
+				ast_clear_flag(chan, AST_FLAG_SPYING);
+				break;
+			}
+		}
+
+		count = 0;
+		res = ast_waitfordigit(chan, waitms);
+		if (res < 0) {
+			ast_clear_flag(chan, AST_FLAG_SPYING);
+			break;
+		}
+				
+		peer = local_channel_walk(NULL);
+		prev=NULL;
+		while(peer) {
+			if (peer != chan) {
+				char *group = NULL;
+				int igrp = 1;
+
+				if (peer == prev && !chosen) {
+					break;
+				}
+				chosen = 0;
+				group = pbx_builtin_getvar_helper(peer, "SPYGROUP");
+				if (mygroup) {
+					if (!group || strcmp(mygroup, group)) {
+						igrp = 0;
+					}
+				}
+				
+				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)) {
+						int x = 0;
+						strncpy(peer_name, "spy-", 5);
+						strncpy(peer_name + strlen(peer_name), peer->name, AST_NAME_STRLEN);
+						ptr = strchr(peer_name, '/');
+						*ptr = '\0';
+						ptr++;
+						for (x = 0 ; x < strlen(peer_name) ; x++) {
+							if (peer_name[x] == '/') {
+								break;
+							}
+							peer_name[x] = tolower(peer_name[x]);
+						}
+
+						if (!silent) {
+							if (ast_fileexists(peer_name, NULL, NULL) != -1) {
+								res = ast_streamfile(chan, peer_name, chan->language);
+								if (!res)
+									res = ast_waitstream(chan, "");
+								if (res)
+									break;
+							} else
+								res = ast_say_character_str(chan, peer_name, "", chan->language);
+							if ((num=atoi(ptr))) 
+								ast_say_digits(chan, atoi(ptr), "", chan->language);
+						}
+						count++;
+						prev = peer;
+						res = channel_spy(chan, peer, &volfactor, fd);
+						if (res == -1) {
+							break;
+						} else if (res > 1 && spec) {
+							snprintf(name, AST_NAME_STRLEN, "%s/%d", spec, res);
+							if ((peer = local_get_channel_begin_name(name))) {
+								chosen = 1;
+							}
+							continue;
+						}
+					}
+				}
+			}
+			if ((peer = local_channel_walk(peer)) == NULL) {
+				break;
+			}
+		}
+		waitms = count ? 100 : 5000;
+	}
+	
+
+	if (fd > 0) {
+		close(fd);
+	}
+
+	if (oldrf && ast_set_read_format(chan, oldrf) < 0) {
+		ast_log(LOG_ERROR, "Could Not Set Read Format.\n");
+	}
+	
+	if (oldwf && ast_set_write_format(chan, oldwf) < 0) {
+		ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
+	}
+
+	ast_clear_flag(chan, AST_FLAG_SPYING);
+
+	ast_channel_setoption(chan, AST_OPTION_TXGAIN, &zero_volume, sizeof(zero_volume), 0);
+
+	ALL_DONE(u, res);
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, chanspy_exec, synopsis, desc);
+}
+
+char *description(void)
+{
+	return (char *) synopsis;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_controlplayback.c
+++ b/apps/app_controlplayback.c
@@ -1,42 +1,69 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Trivial application to control playback a sound file
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief Trivial application to control playback of a sound file
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/app.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/utils.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>

-static char *tdesc = "Control Playback Application";
+#include "asterisk.h"

-static char *app = "ControlPlayback";
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

-static char *synopsis = "Play a file with fast forward and rewind";
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/app.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/utils.h"
+#include "asterisk/options.h"

-static char *descrip = 
-"ControlPlayback(filename[|skipms[|ffchar[|rewchar[|stopchar[|pausechr]]]]]):\n"
-"  Plays  back  a  given  filename (do not put extension). Options may also\n"
-"  be included following a pipe symbol.  You can use * and # to rewind and\n"
-"  fast forward the playback specified. If 'stopchar' is added the file will\n"
-"  terminate playback when 'stopchar' is pressed. Returns -1 if the channel\n"
-"  was hung up, or if the file does not exist. Returns 0 otherwise.\n\n"
-"  Example:  exten => 1234,1,ControlPlayback(file|4000|*|#|1|0)\n\n";
+static const char *tdesc = "Control Playback Application";
+
+static const char *app = "ControlPlayback";
+
+static const char *synopsis = "Play a file with fast forward and rewind";
+
+static const char *descrip = 
+"  ControlPlayback(file[|skipms[|ff[|rew[|stop[|pause[|restart|options]]]]]]]):\n"
+"This application will play back the given filename. By default, the '*' key\n"
+"can be used to rewind, and the '#' key can be used to fast-forward.\n"
+"Parameters:\n"
+"  skipms  - This is number of milliseconds to skip when rewinding or\n"
+"            fast-forwarding.\n"
+"  ff      - Fast-forward when this DTMF digit is received.\n"
+"  rew     - Rewind when this DTMF digit is received.\n"
+"  stop    - Stop playback when this DTMF digit is received.\n"
+"  pause   - Pause playback when this DTMF digit is received.\n"
+"  restart - Restart playback when this DTMF digit is received.\n"
+"Options:\n"
+"  j - Jump to priority n+101 if the requested file is not found.\n"
+"This application sets the following channel variable upon completion:\n"
+"  CPLAYBACKSTATUS -  This variable contains the status of the attempt as a text\n"
+"                     string, one of: SUCCESS | USERSTOPPED | ERROR\n";

 STANDARD_LOCAL_USER;

@@ -44,75 +71,99 @@ LOCAL_USER_DECL;

 static int is_on_phonepad(char key)
 {
-	return (key == 35 || key == 42 || (key >= 48 && key <= 57)) ? 1 : 0;
+	return key == 35 || key == 42 || (key >= 48 && key <= 57);
 }

 static int controlplayback_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
+	int res = 0, priority_jump = 0;
 	int skipms = 0;
 	struct localuser *u;
-	char tmp[256];
-	char *skip = NULL, *fwd = NULL, *rev = NULL, *stop = NULL, *pause = NULL, *file = NULL;
-
-	if (!data || ast_strlen_zero((char *)data)) {
+	char *tmp;
+	int argc;
+	char *argv[8];
+	enum arg_ids {
+		arg_file = 0,
+		arg_skip = 1,
+		arg_fwd = 2,
+		arg_rev = 3,
+		arg_stop = 4,
+		arg_pause = 5,
+		arg_restart = 6,
+		options = 7,
+	};
+	
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "ControlPlayback requires an argument (filename)\n");
 		return -1;
 	}

-	strncpy(tmp, (char *)data, sizeof(tmp)-1);
-	file = tmp;
+	LOCAL_USER_ADD(u);
+	
+	tmp = ast_strdupa(data);
+	memset(argv, 0, sizeof(argv));

-	if ((skip=strchr(tmp,'|'))) {
-		*skip++ = '\0';
-		fwd=strchr(skip,'|');
-		if (fwd) {
-			*fwd++ = '\0';
-			rev = strchr(fwd,'|');
-			if (rev) {
-				*rev++ = '\0';
-				stop = strchr(rev,'|');
-				if (stop) {
-					*stop++ = '\0';
-					pause = strchr(stop,'|');
-					if (pause) {
-						*pause++ = '\0';
-					}
-				}
-			}
-		}
+	argc = ast_app_separate_args(tmp, '|', argv, sizeof(argv) / sizeof(argv[0]));
+
+	if (argc < 1) {
+		ast_log(LOG_WARNING, "ControlPlayback requires an argument (filename)\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
 	}

-	skipms = skip ? atoi(skip) : 3000;
+	skipms = argv[arg_skip] ? atoi(argv[arg_skip]) : 3000;
 	if (!skipms)
 		skipms = 3000;

-	if (!fwd || !is_on_phonepad(*fwd))
-		fwd = "#";
-	if (!rev || !is_on_phonepad(*rev))
-		rev = "*";
-	if (stop && !is_on_phonepad(*stop))
-		stop = NULL;
-	if (pause && !is_on_phonepad(*pause))
-		pause = NULL;
+	if (!argv[arg_fwd] || !is_on_phonepad(*argv[arg_fwd]))
+		argv[arg_fwd] = "#";
+	if (!argv[arg_rev] || !is_on_phonepad(*argv[arg_rev]))
+		argv[arg_rev] = "*";
+	if (argv[arg_stop] && !is_on_phonepad(*argv[arg_stop]))
+		argv[arg_stop] = NULL;
+	if (argv[arg_pause] && !is_on_phonepad(*argv[arg_pause]))
+		argv[arg_pause] = NULL;
+	if (argv[arg_restart] && !is_on_phonepad(*argv[arg_restart]))
+		argv[arg_restart] = NULL;

-	LOCAL_USER_ADD(u);
+	if (argv[options]) {
+		if (strchr(argv[options], 'j'))
+			priority_jump = 1;
+	}

-	res = ast_control_streamfile(chan, file, fwd, rev, stop, pause, skipms);
+	res = ast_control_streamfile(chan, argv[arg_file], argv[arg_fwd], argv[arg_rev], argv[arg_stop], argv[arg_pause], argv[arg_restart], skipms);
+
+	/* If we stopped on one of our stop keys, return 0  */
+	if (argv[arg_stop] && strchr(argv[arg_stop], res)) {
+		res = 0;
+		pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "USERSTOPPED");
+	} else {
+		if (res < 0) {
+			if (priority_jump || option_priority_jumping) {
+				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");
+				}
+			}
+			res = 0;
+			pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "ERROR");
+		} else
+			pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "SUCCESS");
+	}

 	LOCAL_USER_REMOVE(u);
-	
-	/* If we stopped on one of our stop keys, return 0  */
-	if(stop && strchr(stop, res)) 
-		res = 0;
-	
+
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
@@ -122,12 +173,13 @@ int load_module(void)

 char *description(void)
 {
-	return tdesc;
+	return (char *) tdesc;
 }

 int usecount(void)
 {
 	int res;
+
 	STANDARD_USECOUNT(res);
 	return res;
 }
--- a/apps/app_curl.c
+++ b/apps/app_curl.c
@@ -0,0 +1,256 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C)  2004 - 2005, Tilghman Lesher
+ *
+ * Tilghman Lesher <curl-20050919@the-tilghman.com>
+ * and Brian Wilkins <bwilkins@cfl.rr.com> (Added POST option)
+ *
+ * app_curl.c is distributed with no restrictions on usage or
+ * redistribution.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ * \brief Curl - App to load a URL
+ * 
+ * \ingroup applications
+ */
+ 
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <curl/curl.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/cli.h"
+#include "asterisk/options.h"
+#include "asterisk/module.h"
+
+static char *tdesc = "Load external URL";
+
+static char *app = "Curl";
+
+static char *synopsis = "Load an external URL";
+
+static char *descrip = 
+"  Curl(URL[|postdata]): This application will request the specified URL.\n"
+"It is mainly used for signalling external applications of an event.\n"
+"Parameters:\n"
+"  URL      - This is the external URL to request.\n"
+"  postdata - This information will be treated as POST data.\n"
+"This application will set the following variable:\n"
+"  CURL - This variable will contain the resulting page.\n"
+"This application has been deprecated in favor of the CURL function.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+struct MemoryStruct {
+	char *memory;
+	size_t size;
+};
+
+static void *myrealloc(void *ptr, size_t size)
+{
+	/* There might be a realloc() out there that doesn't like reallocing
+	   NULL pointers, so we take care of it here */
+	if (ptr)
+		return realloc(ptr, size);
+	else
+		return malloc(size);
+}
+
+static size_t WriteMemoryCallback(void *ptr, size_t size, size_t nmemb, void *data)
+{
+	register int realsize = size * nmemb;
+	struct MemoryStruct *mem = (struct MemoryStruct *)data;
+
+	mem->memory = (char *)myrealloc(mem->memory, mem->size + realsize + 1);
+	if (mem->memory) {
+		memcpy(&(mem->memory[mem->size]), ptr, realsize);
+		mem->size += realsize;
+		mem->memory[mem->size] = 0;
+	}
+	return realsize;
+}
+
+static int curl_internal(struct MemoryStruct *chunk, char *url, char *post)
+{
+	CURL *curl;
+
+	curl_global_init(CURL_GLOBAL_ALL);
+	curl = curl_easy_init();
+
+	if (!curl) {
+		return -1;
+	}
+
+	curl_easy_setopt(curl, CURLOPT_URL, url);
+	curl_easy_setopt(curl, CURLOPT_WRITEFUNCTION, WriteMemoryCallback);
+	curl_easy_setopt(curl, CURLOPT_WRITEDATA, (void *)chunk);
+	curl_easy_setopt(curl, CURLOPT_USERAGENT, "asterisk-libcurl-agent/1.0");
+
+	if (post) {
+		curl_easy_setopt(curl, CURLOPT_POST, 1);
+		curl_easy_setopt(curl, CURLOPT_POSTFIELDS, post);
+	}
+
+	curl_easy_perform(curl);
+	curl_easy_cleanup(curl);
+	return 0;
+}
+
+static int curl_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u;
+	char *info, *post_data=NULL, *url;
+	struct MemoryStruct chunk = { NULL, 0 };
+	static int dep_warning = 0;
+	
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The application Curl is deprecated.  Please use the CURL() function instead.\n");
+		dep_warning = 1;
+	}
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Curl requires an argument (URL)\n");
+		return -1;
+	}
+	
+	LOCAL_USER_ADD(u);
+	
+	if ((info = ast_strdupa(data))) {
+		url = strsep(&info, "|");
+		post_data = info;
+	} else {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (! curl_internal(&chunk, url, post_data)) {
+		if (chunk.memory) {
+			chunk.memory[chunk.size] = '\0';
+			if (chunk.memory[chunk.size - 1] == 10)
+				chunk.memory[chunk.size - 1] = '\0';
+
+			pbx_builtin_setvar_helper(chan, "CURL", chunk.memory);
+
+			free(chunk.memory);
+		}
+	} else {
+		ast_log(LOG_ERROR, "Cannot allocate curl structure\n");
+		res = -1;
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static char *acf_curl_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
+	struct localuser *u;
+	char *info, *post_data=NULL, *url;
+	struct MemoryStruct chunk = { NULL, 0 };
+
+	*buf = '\0';
+	
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "CURL requires an argument (URL)\n");
+		return buf;
+	}
+
+	LOCAL_USER_ACF_ADD(u);
+
+	info = ast_strdupa(data);
+	if (!info) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return buf;
+	}
+	
+	url = strsep(&info, "|");
+	post_data = info;
+	
+	if (! curl_internal(&chunk, url, post_data)) {
+		if (chunk.memory) {
+			chunk.memory[chunk.size] = '\0';
+			if (chunk.memory[chunk.size - 1] == 10)
+				chunk.memory[chunk.size - 1] = '\0';
+
+			ast_copy_string(buf, chunk.memory, len);
+			free(chunk.memory);
+		}
+	} else {
+		ast_log(LOG_ERROR, "Cannot allocate curl structure\n");
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return buf;
+}
+
+struct ast_custom_function acf_curl = {
+	.name = "CURL",
+	.synopsis = "Retrieves the contents of a URL",
+	.syntax = "CURL(url[|post-data])",
+	.desc =
+	"  url       - URL to retrieve\n"
+	"  post-data - Optional data to send as a POST (GET is default action)\n",
+	.read = acf_curl_exec,
+};
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_custom_function_unregister(&acf_curl);
+	res |= ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+	
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+
+	res = ast_custom_function_register(&acf_curl);
+	res |= ast_register_application(app, curl_exec, synopsis, descrip);
+
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_cut.c
+++ b/apps/app_cut.c
@@ -1,81 +1,179 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
- *
- * Cut application
+ * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
 *
 * Tilghman Lesher <app_cut__v003@the-tilghman.com>
 *
- * $Id$
- *
 * This code is released by the author with no restrictions on usage.
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ * \brief Cut application
+ *
+ * \ingroup applications
 */

 #include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/options.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#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 */
 #define MAXRESULT	1024

-static char *tdesc = "Cuts up variables";
+static char *tdesc = "Cut out information from a string";

 static char *app_cut = "Cut";

-static char *cut_synopsis = "Splits a variable's content using the specified delimiter";
+static char *cut_synopsis = "Splits a variable's contents using the specified delimiter";

 static char *cut_descrip =
-"Usage: Cut(newvar=varname,delimiter,fieldspec)\n"
+"  Cut(newvar=varname,delimiter,fieldspec): This applicaiton will split the\n"
+"contents of a variable based on the given delimeter and store the result in\n"
+"a new variable.\n"
+"Parameters:\n"
 "  newvar    - new variable created from result string\n"
 "  varname   - variable you want cut\n"
 "  delimiter - defaults to '-'\n"
 "  fieldspec - number of the field you want (1-based offset)\n"
-"            may also be specified as a range (with -)\n"
-"            or group of ranges and fields (with &)\n" 
-"  Returns 0 or -1 on hangup or error.\n";
+"              may also be specified as a range (with -)\n"
+"              or group of ranges and fields (with &)\n"
+"This application has been deprecated in favor of the CUT function.\n";
+
+static char *app_sort = "Sort";
+static char *app_sort_synopsis = "Sorts a list of keywords and values";
+static char *app_sort_descrip =
+"  Sort(newvar=key1:val1[,key2:val2[[...],keyN:valN]]): This application will\n"
+"sort the list provided in ascending order. The result will be stored in the\n"
+"specified variable name.\n"
+"  This applicaiton has been deprecated in favor of the SORT function.\n";

 STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

-static int cut_exec(struct ast_channel *chan, void *data)
+struct sortable_keys {
+	char *key;
+	float value;
+};
+
+static int sort_subroutine(const void *arg1, const void *arg2)
 {
-	int res=0;
-	struct localuser *u;
-	char *s, *newvar=NULL, *varname=NULL, *delimiter=NULL, *field=NULL;
+	const struct sortable_keys *one=arg1, *two=arg2;
+	if (one->value < two->value) {
+		return -1;
+	} else if (one->value == two->value) {
+		return 0;
+	} else {
+		return 1;
+	}
+}
+
+#define ERROR_NOARG	(-1)
+#define ERROR_NOMEM	(-2)
+#define ERROR_USAGE	(-3)
+
+static int sort_internal(struct ast_channel *chan, char *data, char *buffer, size_t buflen)
+{
+	char *strings, *ptrkey, *ptrvalue;
+	int count=1, count2, element_count=0;
+	struct sortable_keys *sortable_keys;
+
+	memset(buffer, 0, buflen);
+
+	if (!data) {
+		return ERROR_NOARG;
+	}
+
+	strings = ast_strdupa((char *)data);
+	if (!strings) {
+		return ERROR_NOMEM;
+	}
+
+	for (ptrkey = strings; *ptrkey; ptrkey++) {
+		if (*ptrkey == '|') {
+			count++;
+		}
+	}
+
+	sortable_keys = alloca(count * sizeof(struct sortable_keys));
+	if (!sortable_keys) {
+		return ERROR_NOMEM;
+	}
+
+	memset(sortable_keys, 0, count * sizeof(struct sortable_keys));
+
+	/* Parse each into a struct */
+	count2 = 0;
+	while ((ptrkey = strsep(&strings, "|"))) {
+		ptrvalue = index(ptrkey, ':');
+		if (!ptrvalue) {
+			count--;
+			continue;
+		}
+		*ptrvalue = '\0';
+		ptrvalue++;
+		sortable_keys[count2].key = ptrkey;
+		sscanf(ptrvalue, "%f", &sortable_keys[count2].value);
+		count2++;
+	}
+
+	/* Sort the structs */
+	qsort(sortable_keys, count, sizeof(struct sortable_keys), sort_subroutine);
+
+	for (count2 = 0; count2 < count; count2++) {
+		int blen = strlen(buffer);
+		if (element_count++) {
+			strncat(buffer + blen, ",", buflen - blen - 1);
+		}
+		strncat(buffer + blen + 1, sortable_keys[count2].key, buflen - blen - 2);
+	}
+
+	return 0;
+}
+
+static int cut_internal(struct ast_channel *chan, char *data, char *buffer, size_t buflen)
+{
+	char *s, *args[3], *varname=NULL, *delimiter=NULL, *field=NULL;
 	int args_okay = 0;

-	LOCAL_USER_ADD(u);
+	memset(buffer, 0, buflen);

 	/* Check and parse arguments */
 	if (data) {
 		s = ast_strdupa((char *)data);
 		if (s) {
-			newvar = strsep(&s, "=");
-			if (newvar && (newvar[0] != '\0')) {
-				varname = strsep(&s, "|");
-				if (varname && (varname[0] != '\0')) {
-					delimiter = strsep(&s, "|");
-					if (delimiter) {
-						field = strsep(&s, "|");
-						if (field) {
-							args_okay = 1;
-						}
-					}
-				}
+			ast_app_separate_args(s, '|', args, 3);
+			varname = args[0];
+			delimiter = args[1];
+			field = args[2];
+
+			if (field) {
+				args_okay = 1;
 			}
 		} else {
-			ast_log(LOG_ERROR, "Out of memory\n");
-			res = -1;
+			return ERROR_NOMEM;
 		}
 	}

@@ -83,16 +181,12 @@ static int cut_exec(struct ast_channel *chan, void *data)
 		char d, ds[2];
 		char *tmp = alloca(strlen(varname) + 4);
 		char varvalue[MAXRESULT], *tmp2=varvalue;
-		char retstring[MAXRESULT];
-
-		memset(retstring, 0, MAXRESULT);

 		if (tmp) {
 			snprintf(tmp, strlen(varname) + 4, "${%s}", varname);
 			memset(varvalue, 0, sizeof(varvalue));
 		} else {
-			ast_log(LOG_ERROR, "Out of memory");
-			return -1;
+			return ERROR_NOMEM;
 		}

 		if (delimiter[0])
@@ -124,9 +218,7 @@ static int cut_exec(struct ast_channel *chan, void *data)
 					/* single number */
 					num2 = num1;
 				} else {
-					ast_log(LOG_ERROR, "Cut(): Illegal range '%s'\n", nextgroup);
-					ast_log(LOG_ERROR, "Usage: %s\n", cut_synopsis);
-					return -1;
+					return ERROR_USAGE;
 				}

 				/* Get to start, if any */
@@ -139,7 +231,7 @@ static int cut_exec(struct ast_channel *chan, void *data)

 				/* Most frequent problem is the expectation of reordering fields */
 				if ((num1 > 0) && (curfieldnum > num1)) {
-					ast_log(LOG_WARNING, "Cut(): we're already past the field you wanted?\n");
+					ast_log(LOG_WARNING, "We're already past the field you wanted?\n");
 				}

 				/* Re-null tmp2 if we added 1 to NULL */
@@ -149,35 +241,216 @@ static int cut_exec(struct ast_channel *chan, void *data)
 				/* Output fields until we either run out of fields or num2 is reached */
 				while ((tmp2 != NULL) && (curfieldnum <= num2)) {
 					char *tmp3 = strsep(&tmp2, ds);
-					int curlen = strlen(retstring);
+					int curlen = strlen(buffer);

-					if (strlen(retstring)) {
-						snprintf(retstring + curlen, MAXRESULT - curlen, "%c%s", d, tmp3);
+					if (curlen) {
+						snprintf(buffer + curlen, buflen - curlen, "%c%s", d, tmp3);
 					} else {
-						snprintf(retstring, MAXRESULT, "%s", tmp3);
+						snprintf(buffer, buflen, "%s", tmp3);
 					}

 					curfieldnum++;
 				}
 			}
 		}
+	}
+	return 0;
+}

-		pbx_builtin_setvar_helper(chan, newvar, retstring);
+static int sort_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *varname, *strings, result[512] = "";
+	static int dep_warning=0;
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The application Sort is deprecated.  Please use the SORT() function instead.\n");
+		dep_warning=1;
 	}

+	if (!data) {
+		ast_log(LOG_ERROR, "Sort() requires an argument\n");
+		return 0;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	strings = ast_strdupa((char *)data);
+	if (!strings) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+
+	varname = strsep(&strings, "=");
+	switch (sort_internal(chan, strings, result, sizeof(result))) {
+	case ERROR_NOARG:
+		ast_log(LOG_ERROR, "Sort() requires an argument\n");
+		res = 0;
+		break;
+	case ERROR_NOMEM:
+		ast_log(LOG_ERROR, "Out of memory\n");
+		res = -1;
+		break;
+	case 0:
+		pbx_builtin_setvar_helper(chan, varname, result);
+		res = 0;
+		break;
+	default:
+		ast_log(LOG_ERROR, "Unknown internal error\n");
+		res = -1;
+	}
 	LOCAL_USER_REMOVE(u);
 	return res;
 }

+static int cut_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *s, *newvar=NULL, result[512];
+	static int dep_warning = 0;
+
+	LOCAL_USER_ADD(u);
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The application Cut is deprecated.  Please use the CUT() function instead.\n");
+		dep_warning=1;
+	}
+
+	/* Check and parse arguments */
+	if (data) {
+		s = ast_strdupa((char *)data);
+		if (s) {
+			newvar = strsep(&s, "=");
+		} else {
+			ast_log(LOG_ERROR, "Out of memory\n");
+			LOCAL_USER_REMOVE(u);
+			return -1;
+		}
+	}
+
+	switch (cut_internal(chan, s, result, sizeof(result))) {
+	case ERROR_NOARG:
+		ast_log(LOG_ERROR, "Cut() requires an argument\n");
+		res = 0;
+		break;
+	case ERROR_NOMEM:
+		ast_log(LOG_ERROR, "Out of memory\n");
+		res = -1;
+		break;
+	case ERROR_USAGE:
+		ast_log(LOG_ERROR, "Usage: %s\n", cut_synopsis);
+		res = 0;
+		break;
+	case 0:
+		pbx_builtin_setvar_helper(chan, newvar, result);
+		res = 0;
+		break;
+	default:
+		ast_log(LOG_ERROR, "Unknown internal error\n");
+		res = -1;
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static char *acf_sort_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
+	struct localuser *u;
+
+	LOCAL_USER_ACF_ADD(u);
+
+	switch (sort_internal(chan, data, buf, len)) {
+	case ERROR_NOARG:
+		ast_log(LOG_ERROR, "SORT() requires an argument\n");
+		break;
+	case ERROR_NOMEM:
+		ast_log(LOG_ERROR, "Out of memory\n");
+		break;
+	case 0:
+		break;
+	default:
+		ast_log(LOG_ERROR, "Unknown internal error\n");
+	}
+	LOCAL_USER_REMOVE(u);
+	return buf;
+}
+
+static char *acf_cut_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
+	struct localuser *u;
+
+	LOCAL_USER_ACF_ADD(u);
+
+	switch (cut_internal(chan, data, buf, len)) {
+	case ERROR_NOARG:
+		ast_log(LOG_ERROR, "Cut() requires an argument\n");
+		break;
+	case ERROR_NOMEM:
+		ast_log(LOG_ERROR, "Out of memory\n");
+		break;
+	case ERROR_USAGE:
+		ast_log(LOG_ERROR, "Usage: %s\n", cut_synopsis);
+		break;
+	case 0:
+		break;
+	default:
+		ast_log(LOG_ERROR, "Unknown internal error\n");
+	}
+	LOCAL_USER_REMOVE(u);
+	return buf;
+}
+
+struct ast_custom_function acf_sort = {
+	.name = "SORT",
+	.synopsis = "Sorts a list of key/vals into a list of keys, based upon the vals",
+	.syntax = "SORT(key1:val1[...][,keyN:valN])",
+	.desc =
+"Takes a comma-separated list of keys and values, each separated by a colon, and returns a\n"
+"comma-separated list of the keys, sorted by their values.  Values will be evaluated as\n"
+"floating-point numbers.\n",
+	.read = acf_sort_exec,
+};
+
+struct ast_custom_function acf_cut = {
+	.name = "CUT",
+	.synopsis = "Slices and dices strings, based upon a named delimiter.",
+	.syntax = "CUT(<varname>,<char-delim>,<range-spec>)",
+	.desc =
+"  varname    - variable you want cut\n"
+"  char-delim - defaults to '-'\n"
+"  range-spec - number of the field you want (1-based offset)\n"
+"             may also be specified as a range (with -)\n"
+"             or group of ranges and fields (with &)\n",
+	.read = acf_cut_exec,
+};
+
 int unload_module(void)
 {
+	int res;
+
+	res = ast_custom_function_unregister(&acf_cut);
+	res |= ast_custom_function_unregister(&acf_sort);
+	res |= ast_unregister_application(app_sort);
+	res |= ast_unregister_application(app_cut);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app_cut);
+
+	return res;
 }

 int load_module(void)
 {
-	return ast_register_application(app_cut, cut_exec, cut_synopsis, cut_descrip);
+	int res;
+
+	res = ast_custom_function_register(&acf_cut);
+	res |= ast_custom_function_register(&acf_sort);
+	res |= ast_register_application(app_sort, sort_exec, app_sort_synopsis, app_sort_descrip);
+	res |= ast_register_application(app_cut, cut_exec, cut_synopsis, cut_descrip);
+
+	return res;
 }

 char *description(void)
--- a/apps/app_datetime.c
+++ b/apps/app_datetime.c
@@ -1,27 +1,43 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Time of day - Report the time of day
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/say.h>
+/*! \file
+ * \brief Time of day - Report the time of day
+ * 
+ * \ingroup applications
+ */
+
+#include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
-#include <stdlib.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/say.h"

 static char *tdesc = "Date and Time";

@@ -30,8 +46,7 @@ static char *app = "DateTime";
 static char *synopsis = "Say the date and time";

 static char *descrip = 
-"  DateTime():  Says the current date and time.  Returns -1 on hangup or 0\n"
-"otherwise.\n";
+"  DateTime(): This application will say the current date and time.\n";

 STANDARD_LOCAL_USER;

@@ -54,8 +69,13 @@ static int datetime_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_db.c
+++ b/apps/app_db.c
@@ -1,50 +1,74 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Database access functions
- *
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 * Copyright (C) 2003, Jefferson Noxon
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
 * Jefferson Noxon <jeff@debian.org>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
 */

-#include <sys/types.h>
-#include <asterisk/options.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/astdb.h>
-#include <asterisk/lock.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <string.h>
-#include <stdlib.h>
+/*! \file
+ *
+ * \brief Database access functions
+ *
+ * \ingroup applications
+ */

-static char *tdesc = "Database access functions for Asterisk extension logic";
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <sys/types.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/options.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/astdb.h"
+#include "asterisk/lock.h"
+#include "asterisk/options.h"
+
+static char *tdesc = "Database Access Functions";

 static char *g_descrip =
-	"  DBget(varname=family/key): Retrieves a value from the Asterisk\n"
-	"database and stores it in the given variable.  Always returns 0.  If the\n"
-	"requested key is not found, jumps to priority n+101 if available.\n";
+"  DBget(varname=family/key[|options]): This application will retrieve a value\n"
+"from the Asterisk database and store it in the given variable.\n"
+"  Options:\n"
+"    j - Jump to priority n+101 if the requested family/key isn't found.\n"
+"  This application sets the following channel variable upon completion:\n"
+"    DBGETSTATUS - This variable will contain the status of the attempt\n"
+"                  FOUND | NOTFOUND \n"
+"  This application has been deprecated in favor of the DB function.\n";

 static char *p_descrip =
-	"  DBput(family/key=value): Stores the given value in the Asterisk\n"
-	"database.  Always returns 0.\n";
+"  DBput(family/key=value): This application will store the given value in the\n"
+"specified location in the Asterisk database.\n"
+"  This application has been deprecated in favor of the DB function.\n";

 static char *d_descrip =
-	"  DBdel(family/key): Deletes a key from the Asterisk database.  Always\n"
-	"returns 0.\n";
+"  DBdel(family/key): This applicaiton will delete a key from the Asterisk\n"
+"database.\n";

 static char *dt_descrip =
-	"  DBdeltree(family[/keytree]): Deletes a family or keytree from the Asterisk\n"
-	"database.  Always returns 0.\n";
+"  DBdeltree(family[/keytree]): This application will delete a family or keytree\n"
+"from the Asterisk database\n";

 static char *g_app = "DBget";
 static char *p_app = "DBput";
@@ -60,27 +84,29 @@ STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

-static int deltree_exec (struct ast_channel *chan, void *data)
+static int deltree_exec(struct ast_channel *chan, void *data)
 {
-	int arglen;
 	char *argv, *family, *keytree;
+	struct localuser *u;

-	arglen = strlen (data);
-	argv = alloca (arglen + 1);
-	if (!argv) {	/* Why would this fail? */
-		ast_log (LOG_DEBUG, "Memory allocation failed\n");
+	LOCAL_USER_ADD(u);
+
+	argv = ast_strdupa(data);
+	if (!argv) {
+		ast_log(LOG_ERROR, "Memory allocation failed\n");
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
-	memcpy (argv, data, arglen + 1);

-	if (strchr (argv, '/')) {
-		family = strsep (&argv, "/");
-		keytree = strsep (&argv, "\0");
+	if (strchr(argv, '/')) {
+		family = strsep(&argv, "/");
+		keytree = strsep(&argv, "\0");
 			if (!family || !keytree) {
-				ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+				ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+				LOCAL_USER_REMOVE(u);
 				return 0;
 			}
-		if (!strlen (keytree))
+		if (ast_strlen_zero(keytree))
 			keytree = 0;
 	} else {
 		family = argv;
@@ -89,167 +115,209 @@ static int deltree_exec (struct ast_channel *chan, void *data)

 	if (option_verbose > 2)	{
 		if (keytree)
-			ast_verbose (VERBOSE_PREFIX_3 "DBdeltree: family=%s, keytree=%s\n", family, keytree);
+			ast_verbose(VERBOSE_PREFIX_3 "DBdeltree: family=%s, keytree=%s\n", family, keytree);
 		else
-			ast_verbose (VERBOSE_PREFIX_3 "DBdeltree: family=%s\n", family);
+			ast_verbose(VERBOSE_PREFIX_3 "DBdeltree: family=%s\n", family);
 	}

-	if (ast_db_deltree (family, keytree)) {
+	if (ast_db_deltree(family, keytree)) {
 		if (option_verbose > 2)
-			ast_verbose (VERBOSE_PREFIX_3 "DBdeltree: Error deleting key from database.\n");
+			ast_verbose(VERBOSE_PREFIX_3 "DBdeltree: Error deleting key from database.\n");
 	}

+	LOCAL_USER_REMOVE(u);
+
 	return 0;
 }

-static int del_exec (struct ast_channel *chan, void *data)
+static int del_exec(struct ast_channel *chan, void *data)
 {
-	int arglen;
 	char *argv, *family, *key;
+	struct localuser *u;

-	arglen = strlen (data);
-	argv = alloca (arglen + 1);
-	if (!argv) {	/* Why would this fail? */
-		ast_log (LOG_DEBUG, "Memory allocation failed\n");
+	LOCAL_USER_ADD(u);
+
+	argv = ast_strdupa(data);
+	if (!argv) {
+		ast_log (LOG_ERROR, "Memory allocation failed\n");
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
-	memcpy (argv, data, arglen + 1);

-	if (strchr (argv, '/')) {
-		family = strsep (&argv, "/");
-		key = strsep (&argv, "\0");
+	if (strchr(argv, '/')) {
+		family = strsep(&argv, "/");
+		key = strsep(&argv, "\0");
 		if (!family || !key) {
-			ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			LOCAL_USER_REMOVE(u);
 			return 0;
 		}
 		if (option_verbose > 2)
-			ast_verbose (VERBOSE_PREFIX_3 "DBdel: family=%s, key=%s\n", family, key);
-		if (ast_db_del (family, key)) {
+			ast_verbose(VERBOSE_PREFIX_3 "DBdel: family=%s, key=%s\n", family, key);
+		if (ast_db_del(family, key)) {
 			if (option_verbose > 2)
-				ast_verbose (VERBOSE_PREFIX_3 "DBdel: Error deleting key from database.\n");
+				ast_verbose(VERBOSE_PREFIX_3 "DBdel: Error deleting key from database.\n");
 		}
 	} else {
-		ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
+		ast_log(LOG_DEBUG, "Ignoring, no parameters\n");
 	}
+
+	LOCAL_USER_REMOVE(u);
+	
 	return 0;
 }

-static int put_exec (struct ast_channel *chan, void *data)
+static int put_exec(struct ast_channel *chan, void *data)
 {
-	int arglen;
 	char *argv, *value, *family, *key;
+	static int dep_warning = 0;
+	struct localuser *u;

-	arglen = strlen (data);
-	argv = alloca (arglen + 1);
-	if (!argv) {	/* Why would this fail? */
-		ast_log (LOG_DEBUG, "Memory allocation failed\n");
+	LOCAL_USER_ADD(u);
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "This application has been deprecated, please use the ${DB(family/key)} function instead.\n");
+		dep_warning = 1;
+	}
+	
+	argv = ast_strdupa(data);
+	if (!argv) {
+		ast_log(LOG_ERROR, "Memory allocation failed\n");
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
-	memcpy (argv, data, arglen + 1);

-	if (strchr (argv, '/') && strchr (argv, '=')) {
-		family = strsep (&argv, "/");
-		key = strsep (&argv, "=");
-		value = strsep (&argv, "\0");
+	if (strchr(argv, '/') && strchr(argv, '=')) {
+		family = strsep(&argv, "/");
+		key = strsep(&argv, "=");
+		value = strsep(&argv, "\0");
 		if (!value || !family || !key) {
-			ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			LOCAL_USER_REMOVE(u);
 			return 0;
 		}
 		if (option_verbose > 2)
-			ast_verbose (VERBOSE_PREFIX_3 "DBput: family=%s, key=%s, value=%s\n", family, key, value);
-		if (ast_db_put (family, key, value)) {
+			ast_verbose(VERBOSE_PREFIX_3 "DBput: family=%s, key=%s, value=%s\n", family, key, value);
+		if (ast_db_put(family, key, value)) {
 			if (option_verbose > 2)
-				ast_verbose (VERBOSE_PREFIX_3 "DBput: Error writing value to database.\n");
+				ast_verbose(VERBOSE_PREFIX_3 "DBput: Error writing value to database.\n");
 		}

 	} else	{
 		ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
 	}
+
+	LOCAL_USER_REMOVE(u);
+
 	return 0;
 }

-static int get_exec (struct ast_channel *chan, void *data)
+static int get_exec(struct ast_channel *chan, void *data)
 {
-	int arglen;
-	char *argv, *varname, *family, *key;
+	char *argv, *varname, *family, *key, *options = NULL;
 	char dbresult[256];
+	static int dep_warning = 0;
+	int priority_jump = 0;
+	struct localuser *u;

-	arglen = strlen (data);
-	argv = alloca (arglen + 1);
-	if (!argv) {	/* Why would this fail? */
-		ast_log (LOG_DEBUG, "Memory allocation failed\n");
+	LOCAL_USER_ADD(u);
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "This application has been deprecated, please use the ${DB(family/key)} function instead.\n");
+		dep_warning = 1;
+	}
+	
+	argv = ast_strdupa(data);
+	if (!argv) {
+		ast_log(LOG_ERROR, "Memory allocation failed\n");
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
-	memcpy (argv, data, arglen + 1);

-	if (strchr (argv, '=') && strchr (argv, '/')) {
-		varname = strsep (&argv, "=");
-		family = strsep (&argv, "/");
-		key = strsep (&argv, "\0");
+	if (strchr(argv, '=') && strchr(argv, '/')) {
+		varname = strsep(&argv, "=");
+		family = strsep(&argv, "/");
+		if (strchr((void *)&argv, '|')) {
+			key = strsep(&argv, "|");
+			options = strsep(&argv, "\0");
+		} else
+			key = strsep(&argv, "\0");
+			
 		if (!varname || !family || !key) {
-			ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			LOCAL_USER_REMOVE(u);
 			return 0;
 		}
+
+		if (options) {
+			if (strchr(options, 'j'))
+				priority_jump = 1;
+		}
+			
 		if (option_verbose > 2)
-			ast_verbose (VERBOSE_PREFIX_3 "DBget: varname=%s, family=%s, key=%s\n", varname, family, key);
-		if (!ast_db_get (family, key, dbresult, sizeof (dbresult) - 1)) {
-			pbx_builtin_setvar_helper (chan, varname, dbresult);
+			ast_verbose(VERBOSE_PREFIX_3 "DBget: varname=%s, family=%s, key=%s\n", varname, family, key);
+		if (!ast_db_get(family, key, dbresult, sizeof (dbresult) - 1)) {
+			pbx_builtin_setvar_helper(chan, varname, dbresult);
 			if (option_verbose > 2)
-				ast_verbose (VERBOSE_PREFIX_3 "DBget: set variable %s to %s\n", varname, dbresult);
+				ast_verbose(VERBOSE_PREFIX_3 "DBget: set variable %s to %s\n", varname, dbresult);
+			pbx_builtin_setvar_helper(chan, "DBGETSTATUS", "FOUND");
 		} else {
 			if (option_verbose > 2)
-			ast_verbose (VERBOSE_PREFIX_3 "DBget: Value not found in database.\n");
-			/* Send the call to n+101 priority, where n is the current priority */
-			if (ast_exists_extension (chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-				chan->priority += 100;
+			ast_verbose(VERBOSE_PREFIX_3 "DBget: Value not found in database.\n");
+			if (priority_jump || option_priority_jumping) {
+				/* Send the call to n+101 priority, where n is the current priority */
+				ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+			}
+			pbx_builtin_setvar_helper(chan, "DBGETSTATUS", "NOTFOUND");
 		}
-
 	} else {
-		ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
+		ast_log(LOG_DEBUG, "Ignoring, no parameters\n");
 	}
+
+	LOCAL_USER_REMOVE(u);
+
 	return 0;
 }

-int unload_module (void)
+int unload_module(void)
 {
 	int retval;

+	retval = ast_unregister_application(dt_app);
+	retval |= ast_unregister_application(d_app);
+	retval |= ast_unregister_application(p_app);
+	retval |= ast_unregister_application(g_app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	retval = ast_unregister_application (dt_app);
-	retval |= ast_unregister_application (d_app);
-	retval |= ast_unregister_application (p_app);
-	retval |= ast_unregister_application (g_app);

 	return retval;
 }

-int load_module (void)
+int load_module(void)
 {
 	int retval;

-	retval = ast_register_application (g_app, get_exec, g_synopsis, g_descrip);
-	if (!retval)
-		retval = ast_register_application (p_app, put_exec, p_synopsis, p_descrip);
-	if (!retval)
-		retval = ast_register_application (d_app, del_exec, d_synopsis, d_descrip);
-	if (!retval)
-		retval = ast_register_application (dt_app, deltree_exec, dt_synopsis, dt_descrip);
+	retval = ast_register_application(g_app, get_exec, g_synopsis, g_descrip);
+	retval |= ast_register_application(p_app, put_exec, p_synopsis, p_descrip);
+	retval |= ast_register_application(d_app, del_exec, d_synopsis, d_descrip);
+	retval |= ast_register_application(dt_app, deltree_exec, dt_synopsis, dt_descrip);
+	
 	return retval;
 }

-char *description (void)
+char *description(void)
 {
 	return tdesc;
 }

-int usecount (void)
+int usecount(void)
 {
 	int res;
-	STANDARD_USECOUNT (res);
+	STANDARD_USECOUNT(res);
 	return res;
 }

-char *key ()
+char *key()
 {
 	return ASTERISK_GPL_KEY;
 }
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
--- a/apps/app_dictate.c
+++ b/apps/app_dictate.c
@@ -0,0 +1,358 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2005, Anthony Minessale II
+ *
+ * Anthony Minessale II <anthmct@yahoo.com>
+ *
+ * Donated by Sangoma Technologies <http://www.samgoma.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Virtual Dictation Machine Application For Asterisk
+ *
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <sys/stat.h>	/* for mkdir */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/say.h"
+#include "asterisk/lock.h"
+#include "asterisk/app.h"
+
+static char *tdesc = "Virtual Dictation Machine";
+static char *app = "Dictate";
+static char *synopsis = "Virtual Dictation Machine";
+static char *desc = "  Dictate([<base_dir>])\n"
+"Start dictation machine using optional base dir for files.\n";
+
+
+STANDARD_LOCAL_USER;
+LOCAL_USER_DECL;
+
+typedef enum {
+	DFLAG_RECORD = (1 << 0),
+	DFLAG_PLAY = (1 << 1),
+	DFLAG_TRUNC = (1 << 2),
+	DFLAG_PAUSE = (1 << 3),
+} dflags;
+
+typedef enum {
+	DMODE_INIT,
+	DMODE_RECORD,
+	DMODE_PLAY
+} dmodes;
+
+#define ast_toggle_flag(it,flag) if(ast_test_flag(it, flag)) ast_clear_flag(it, flag); else ast_set_flag(it, flag)
+
+static int play_and_wait(struct ast_channel *chan, char *file, char *digits) 
+{
+	int res = -1;
+	if (!ast_streamfile(chan, file, chan->language)) {
+		res = ast_waitstream(chan, digits);
+	}
+	return res;
+}
+
+static int dictate_exec(struct ast_channel *chan, void *data)
+{
+	char *mydata, *argv[2], *path = NULL, filein[256];
+	char dftbase[256];
+	char *base;
+	struct ast_flags flags = {0};
+	struct ast_filestream *fs;
+	struct ast_frame *f = NULL;
+	struct localuser *u;
+	int ffactor = 320 * 80,
+		res = 0,
+		argc = 0,
+		done = 0,
+		oldr = 0,
+		lastop = 0,
+		samples = 0,
+		speed = 1,
+		digit = 0,
+		len = 0,
+		maxlen = 0,
+		mode = 0;
+		
+	LOCAL_USER_ADD(u);
+	
+	snprintf(dftbase, sizeof(dftbase), "%s/dictate", ast_config_AST_SPOOL_DIR);
+	if (!ast_strlen_zero(data) && (mydata = ast_strdupa(data))) {
+		argc = ast_app_separate_args(mydata, '|', argv, sizeof(argv) / sizeof(argv[0]));
+	}
+	
+	if (argc) {
+		base = argv[0];
+	} else {
+		base = dftbase;
+	}
+
+	oldr = chan->readformat;
+	if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR)) < 0) {
+		ast_log(LOG_WARNING, "Unable to set to linear mode.\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	ast_answer(chan);
+	ast_safe_sleep(chan, 200);
+	for(res = 0; !res;) {
+		if (ast_app_getdata(chan, "dictate/enter_filename", filein, sizeof(filein), 0) || 
+			ast_strlen_zero(filein)) {
+			res = -1;
+			break;
+		}
+		
+		mkdir(base, 0755);
+		len = strlen(base) + strlen(filein) + 2;
+		if (!path || len > maxlen) {
+			path = alloca(len);
+			memset(path, 0, len);
+			maxlen = len;
+		} else {
+			memset(path, 0, maxlen);
+		}
+
+		snprintf(path, len, "%s/%s", base, filein);
+		fs = ast_writefile(path, "raw", NULL, O_CREAT|O_APPEND, 0, 0700);
+		mode = DMODE_PLAY;
+		memset(&flags, 0, sizeof(flags));
+		ast_set_flag(&flags, DFLAG_PAUSE);
+		digit = play_and_wait(chan, "dictate/forhelp", AST_DIGIT_ANY);
+		done = 0;
+		speed = 1;
+		res = 0;
+		lastop = 0;
+		samples = 0;
+		while (!done && ((res = ast_waitfor(chan, -1)) > -1) && fs && (f = ast_read(chan))) {
+			if (digit) {
+				struct ast_frame fr = {AST_FRAME_DTMF, digit};
+				ast_queue_frame(chan, &fr);
+				digit = 0;
+			}
+			if ((f->frametype == AST_FRAME_DTMF)) {
+				int got = 1;
+				switch(mode) {
+				case DMODE_PLAY:
+					switch(f->subclass) {
+					case '1':
+						ast_set_flag(&flags, DFLAG_PAUSE);
+						mode = DMODE_RECORD;
+						break;
+					case '2':
+						speed++;
+						if (speed > 4) {
+							speed = 1;
+						}
+						res = ast_say_number(chan, speed, AST_DIGIT_ANY, chan->language, (char *) NULL);
+						break;
+					case '7':
+						samples -= ffactor;
+						if(samples < 0) {
+							samples = 0;
+						}
+						ast_seekstream(fs, samples, SEEK_SET);
+						break;
+					case '8':
+						samples += ffactor;
+						ast_seekstream(fs, samples, SEEK_SET);
+						break;
+						
+					default:
+						got = 0;
+					}
+					break;
+				case DMODE_RECORD:
+					switch(f->subclass) {
+					case '1':
+						ast_set_flag(&flags, DFLAG_PAUSE);
+						mode = DMODE_PLAY;
+						break;
+					case '8':
+						ast_toggle_flag(&flags, DFLAG_TRUNC);
+						lastop = 0;
+						break;
+					default:
+						got = 0;
+					}
+					break;
+				default:
+					got = 0;
+				}
+				if (!got) {
+					switch(f->subclass) {
+					case '#':
+						done = 1;
+						continue;
+						break;
+					case '*':
+						ast_toggle_flag(&flags, DFLAG_PAUSE);
+						if (ast_test_flag(&flags, DFLAG_PAUSE)) {
+							digit = play_and_wait(chan, "dictate/pause", AST_DIGIT_ANY);
+						} else {
+							digit = play_and_wait(chan, mode == DMODE_PLAY ? "dictate/playback" : "dictate/record", AST_DIGIT_ANY);
+						}
+						break;
+					case '0':
+						ast_set_flag(&flags, DFLAG_PAUSE);
+						digit = play_and_wait(chan, "dictate/paused", AST_DIGIT_ANY);
+						switch(mode) {
+						case DMODE_PLAY:
+							digit = play_and_wait(chan, "dictate/play_help", AST_DIGIT_ANY);
+							break;
+						case DMODE_RECORD:
+							digit = play_and_wait(chan, "dictate/record_help", AST_DIGIT_ANY);
+							break;
+						}
+						if (digit == 0) {
+							digit = play_and_wait(chan, "dictate/both_help", AST_DIGIT_ANY);
+						} else if (digit < 0) {
+							done = 1;
+							break;
+						}
+						break;
+					}
+				}
+				
+			} else if (f->frametype == AST_FRAME_VOICE) {
+				switch(mode) {
+					struct ast_frame *fr;
+					int x;
+				case DMODE_PLAY:
+					if (lastop != DMODE_PLAY) {
+						if (ast_test_flag(&flags, DFLAG_PAUSE)) {
+							digit = play_and_wait(chan, "dictate/playback_mode", AST_DIGIT_ANY);
+							if (digit == 0) {
+								digit = play_and_wait(chan, "dictate/paused", AST_DIGIT_ANY);
+							} else if (digit < 0) {
+								break;
+							}
+						}
+						if (lastop != DFLAG_PLAY) {
+							lastop = DFLAG_PLAY;
+							ast_closestream(fs);
+							fs = ast_openstream(chan, path, chan->language);
+							ast_seekstream(fs, samples, SEEK_SET);
+							chan->stream = NULL;
+						}
+						lastop = DMODE_PLAY;
+					}
+
+					if (!ast_test_flag(&flags, DFLAG_PAUSE)) {
+						for (x = 0; x < speed; x++) {
+							if ((fr = ast_readframe(fs))) {
+								ast_write(chan, fr);
+								samples += fr->samples;
+								ast_frfree(fr);
+								fr = NULL;
+							} else {
+								samples = 0;
+								ast_seekstream(fs, 0, SEEK_SET);
+							}
+						}
+					}
+					break;
+				case DMODE_RECORD:
+					if (lastop != DMODE_RECORD) {
+						int oflags = O_CREAT | O_WRONLY;
+						if (ast_test_flag(&flags, DFLAG_PAUSE)) {						
+							digit = play_and_wait(chan, "dictate/record_mode", AST_DIGIT_ANY);
+							if (digit == 0) {
+								digit = play_and_wait(chan, "dictate/paused", AST_DIGIT_ANY);
+							} else if (digit < 0) {
+								break;
+							}
+						}
+						lastop = DMODE_RECORD;
+						ast_closestream(fs);
+						if ( ast_test_flag(&flags, DFLAG_TRUNC)) {
+							oflags |= O_TRUNC;
+							digit = play_and_wait(chan, "dictate/truncating_audio", AST_DIGIT_ANY);
+						} else {
+							oflags |= O_APPEND;
+						}
+						fs = ast_writefile(path, "raw", NULL, oflags, 0, 0700);
+						if (ast_test_flag(&flags, DFLAG_TRUNC)) {
+							ast_seekstream(fs, 0, SEEK_SET);
+							ast_clear_flag(&flags, DFLAG_TRUNC);
+						} else {
+							ast_seekstream(fs, 0, SEEK_END);
+						}
+					}
+					if (!ast_test_flag(&flags, DFLAG_PAUSE)) {
+						res = ast_writestream(fs, f);
+					}
+					break;
+				}
+				
+			}
+
+			ast_frfree(f);
+		}
+	}
+	if (oldr) {
+		ast_set_read_format(chan, oldr);
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+	
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, dictate_exec, synopsis, desc);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
+
--- a/apps/app_directed_pickup.c
+++ b/apps/app_directed_pickup.c
@@ -0,0 +1,167 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2005, Joshua Colp
+ *
+ * Joshua Colp <jcolp@asterlink.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Directed Call Pickup Support
+ *
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/app.h"
+
+static const char *tdesc = "Directed Call Pickup Application";
+static const char *app = "Pickup";
+static const char *synopsis = "Directed Call Pickup";
+static const char *descrip =
+"  Pickup(extension[@context]): This application can pickup any ringing channel\n"
+"that is calling the specified extension. If no context is specified, the current\n"
+"context will be used.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int pickup_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u = NULL;
+	struct ast_channel *origin = NULL, *target = NULL;
+	char *tmp = NULL, *exten = NULL, *context = NULL;
+	char workspace[256] = "";
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Pickup requires an argument (extension) !\n");
+		return -1;	
+	}
+
+	LOCAL_USER_ADD(u);
+	
+	/* Get the extension and context if present */
+	exten = data;
+	context = strchr(data, '@');
+	if (context) {
+		*context = '\0';
+		context++;
+	}
+
+	/* Find a channel to pickup */
+	origin = ast_get_channel_by_exten_locked(exten, context);
+	if (origin) {
+		ast_cdr_getvar(origin->cdr, "dstchannel", &tmp, workspace,
+			       sizeof(workspace), 0);
+		if (tmp) {
+			/* We have a possible channel... now we need to find it! */
+			target = ast_get_channel_by_name_locked(tmp);
+		} else {
+			ast_log(LOG_DEBUG, "No target channel found.\n");
+			res = -1;
+		}
+		ast_mutex_unlock(&origin->lock);
+	} else {
+		ast_log(LOG_DEBUG, "No originating channel found.\n");
+	}
+	
+	if (res)
+		goto out;
+
+	if (target && (!target->pbx) && ((target->_state == AST_STATE_RINGING) || (target->_state == AST_STATE_RING))) {
+		ast_log(LOG_DEBUG, "Call pickup on chan '%s' by '%s'\n", target->name,
+			chan->name);
+		res = ast_answer(chan);
+		if (res) {
+			ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan->name);
+			res = -1;
+			goto out;
+		}
+		res = ast_queue_control(chan, AST_CONTROL_ANSWER);
+		if (res) {
+			ast_log(LOG_WARNING, "Unable to queue answer on '%s'\n",
+				chan->name);
+			res = -1;
+			goto out;
+		}
+		res = ast_channel_masquerade(target, chan);
+		if (res) {
+			ast_log(LOG_WARNING, "Unable to masquerade '%s' into '%s'\n", chan->name, target->name);
+			res = -1;
+			goto out;
+		}
+	} else {
+		ast_log(LOG_DEBUG, "No call pickup possible...\n");
+		res = -1;
+	}
+	/* Done */
+ out:
+	if (target) 
+		ast_mutex_unlock(&target->lock);
+	
+	LOCAL_USER_REMOVE(u);
+
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, pickup_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_directory.c
+++ b/apps/app_directory.c
@@ -1,51 +1,74 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Provide a directory of extensions
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Provide a directory of extensions
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/config.h>
-#include <asterisk/say.h>
-#include <asterisk/utils.h>
 #include <string.h>
 #include <ctype.h>
 #include <stdlib.h>
 #include <stdio.h>
-#include "../asterisk.h"
-#include "../astconf.h"
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/config.h"
+#include "asterisk/say.h"
+#include "asterisk/utils.h"

 static char *tdesc = "Extension Directory";
 static char *app = "Directory";

 static char *synopsis = "Provide directory of voicemail extensions";
 static char *descrip =
-"  Directory(vm-context[|dial-context[|options]]): Presents the user with a directory\n"
-"of extensions from which they  may  select  by name. The  list  of  names \n"
-"and  extensions  is discovered from  voicemail.conf. The  vm-context  argument\n"
-"is required, and specifies  the  context  of voicemail.conf to use.  The\n"
-"dial-context is the context to use for dialing the users, and defaults to\n"
-"the vm-context if unspecified. The 'f' option causes the directory to match\n"
-"based on the first name in voicemail.conf instead of the last name.\n"
-"Returns 0 unless the user hangs up. It  also sets up the channel on exit\n"
-"to enter the extension the user selected.\n";
+"  Directory(vm-context[|dial-context[|options]]): This application will present\n"
+"the calling channel with a directory of extensions from which they can search\n"
+"by name. The list of names and corresponding extensions is retrieved from the\n"
+"voicemail configuration file, voicemail.conf.\n"
+"  This applicaiton will immediate exit if one of the following DTMF digits are\n"
+"received and the extension to jump to exists:\n"
+"    0 - Jump to the 'o' extension, if it exists.\n"
+"    * - Jump to the 'a' extension, if it exists.\n\n"
+"  Parameters:\n"
+"    vm-context   - This is the context within voicemail.conf to use for the\n"
+"                   Directory.\n"
+"    dial-context - This is the dialplan context to use when looking for an\n"
+"                   extension that the user has selected, or when jumping to the\n"
+"                   'o' or 'a' extension.\n\n"
+"  Options:\n"
+"    f - Allow the caller to enter the first name of a user in the directory\n"
+"        instead of using the last name.\n"; 

 /* For simplicity, I'm keeping the format compatible with the voicemail config,
   but i'm open to suggestions for isolating it */

-#define DIRECTORY_CONFIG "voicemail.conf"
+#define VOICEMAIL_CONFIG "voicemail.conf"

 /* How many digits to read in */
 #define NUMDIGITS 3
@@ -176,11 +199,7 @@ static int play_mailbox_owner(struct ast_channel *chan, char *context, char *dia
 				case '1':
 					/* Name selected */
 					loop = 0;
-					if (ast_exists_extension(chan,dialcontext,ext,1,chan->callerid)) {
-						strncpy(chan->exten, ext, sizeof(chan->exten)-1);
-						chan->priority = 0;
-						strncpy(chan->context, dialcontext, sizeof(chan->context)-1);
-					} else {
+					if (ast_goto_if_exists(chan, dialcontext, ext, 1)) {
 						ast_log(LOG_WARNING,
 							"Can't find extension '%s' in context '%s'.  "
 							"Did you pass the wrong context to Directory?\n",
@@ -210,6 +229,65 @@ static int play_mailbox_owner(struct ast_channel *chan, char *context, char *dia
 	return(res);
 }

+static struct ast_config *realtime_directory(char *context)
+{
+	struct ast_config *cfg;
+	struct ast_config *rtdata;
+	struct ast_category *cat;
+	struct ast_variable *var;
+	char *mailbox;
+	char *fullname;
+	char *hidefromdir;
+	char tmp[100];
+
+	/* Load flat file config. */
+	cfg = ast_config_load(VOICEMAIL_CONFIG);
+
+	if (!cfg) {
+		/* Loading config failed. */
+		ast_log(LOG_WARNING, "Loading config failed.\n");
+		return NULL;
+	}
+
+	/* Get realtime entries, categorized by their mailbox number
+	   and present in the requested context */
+	rtdata = ast_load_realtime_multientry("voicemail", "mailbox LIKE", "%", "context", context, NULL);
+
+	/* if there are no results, just return the entries from the config file */
+	if (!rtdata)
+		return cfg;
+
+	/* Does the context exist within the config file? If not, make one */
+	cat = ast_category_get(cfg, context);
+	if (!cat) {
+		cat = ast_category_new(context);
+		if (!cat) {
+			ast_log(LOG_WARNING, "Out of memory\n");
+			ast_config_destroy(cfg);
+			return NULL;
+		}
+		ast_category_append(cfg, cat);
+	}
+
+	mailbox = ast_category_browse(rtdata, NULL);
+	while (mailbox) {
+		fullname = ast_variable_retrieve(rtdata, mailbox, "fullname");
+		hidefromdir = ast_variable_retrieve(rtdata, mailbox, "hidefromdir");
+		snprintf(tmp, sizeof(tmp), "no-password,%s,hidefromdir=%s",
+			 fullname ? fullname : "",
+			 hidefromdir ? hidefromdir : "no");
+		var = ast_variable_new(mailbox, tmp);
+		if (var)
+			ast_variable_append(cat, var);
+		else
+			ast_log(LOG_WARNING, "Out of memory adding mailbox '%s'\n", mailbox);
+		mailbox = ast_category_browse(rtdata, mailbox);
+	}
+	ast_config_destroy(rtdata);
+
+	return cfg;
+}
+
 static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *context, char *dialcontext, char digit, int last)
 {
 	/* Read in the first three digits..  "digit" is the first digit, already read */
@@ -221,12 +299,34 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
 	int lastuserchoice = 0;
 	char *start, *pos, *conv,*stringp=NULL;

-	if (!context || ast_strlen_zero(context)) {
+	if (ast_strlen_zero(context)) {
 		ast_log(LOG_WARNING,
 			"Directory must be called with an argument "
 			"(context in which to interpret extensions)\n");
 		return -1;
 	}
+	if (digit == '0') {
+		if (!ast_goto_if_exists(chan, chan->context, "o", 1) ||
+		    (!ast_strlen_zero(chan->macrocontext) &&
+		     !ast_goto_if_exists(chan, chan->macrocontext, "o", 1))) {
+			return 0;
+		} else {
+			ast_log(LOG_WARNING, "Can't find extension 'o' in current context.  "
+				"Not Exiting the Directory!\n");
+			res = 0;
+		}
+	}	
+	if (digit == '*') {
+		if (!ast_goto_if_exists(chan, chan->context, "a", 1) ||
+		    (!ast_strlen_zero(chan->macrocontext) &&
+		     !ast_goto_if_exists(chan, chan->macrocontext, "a", 1))) {
+			return 0;
+		} else {
+			ast_log(LOG_WARNING, "Can't find extension 'a' in current context.  "
+				"Not Exiting the Directory!\n");
+			res = 0;
+		}
+	}	
 	memset(ext, 0, sizeof(ext));
 	ext[0] = digit;
 	res = 0;
@@ -239,12 +339,12 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
 			while(v) {
 				/* Find a candidate extension */
 				start = strdup(v->value);
-				if (start) {
+				if (start && !strcasestr(start, "hidefromdir=yes")) {
 					stringp=start;
 					strsep(&stringp, ",");
 					pos = strsep(&stringp, ",");
 					if (pos) {
-						strncpy(name, pos, sizeof(name) - 1);
+						ast_copy_string(name, pos, sizeof(name));
 						/* Grab the last name */
 						if (last && strrchr(pos,' '))
 							pos = strrchr(pos, ' ') + 1;
@@ -276,11 +376,11 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
 						lastuserchoice = 0;
 						break;
 					case '1':
-						/* user pressed '1' and extensions exists */
+						/* user pressed '1' and extensions exists;
+						   play_mailbox_owner will already have done
+						   a goto() on the channel
+						 */
 						lastuserchoice = res;
-						strncpy(chan->context, dialcontext, sizeof(chan->context) - 1);
-						strncpy(chan->exten, v->name, sizeof(chan->exten) - 1);
-						chan->priority = 0;
 						break;
 					case '*':
 						/* user pressed '*' to skip something found */
@@ -316,17 +416,13 @@ static int directory_exec(struct ast_channel *chan, void *data)
 	int last = 1;
 	char *context, *dialcontext, *dirintro, *options;

-	if (!data) {
-		ast_log(LOG_WARNING, "directory requires an argument (context[,dialcontext])\n");
-		return -1;
-	}
-	cfg = ast_load(DIRECTORY_CONFIG);
-	if (!cfg) {
-		ast_log(LOG_WARNING, "Unable to open directory configuration %s\n", DIRECTORY_CONFIG);
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Directory requires an argument (context[,dialcontext])\n");
 		return -1;
 	}
+
 	LOCAL_USER_ADD(u);
-top:
+
 	context = ast_strdupa(data);
 	dialcontext = strchr(context, '|');
 	if (dialcontext) {
@@ -341,43 +437,60 @@ top:
 		}
 	} else	
 		dialcontext = context;
+
+	cfg = realtime_directory(context);
+	if (!cfg) {
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
 	dirintro = ast_variable_retrieve(cfg, context, "directoryintro");
-	if (!dirintro || ast_strlen_zero(dirintro))
+	if (ast_strlen_zero(dirintro))
 		dirintro = ast_variable_retrieve(cfg, "general", "directoryintro");
-	if (!dirintro || ast_strlen_zero(dirintro)) {
+	if (ast_strlen_zero(dirintro)) {
 		if (last)
 			dirintro = "dir-intro";	
 		else
 			dirintro = "dir-intro-fn";
 	}
+
 	if (chan->_state != AST_STATE_UP) 
 		res = ast_answer(chan);
-	if (!res)
-		res = ast_streamfile(chan, dirintro, chan->language);
-	if (!res)
-		res = ast_waitstream(chan, AST_DIGIT_ANY);
-	ast_stopstream(chan);
-	if (!res)
-		res = ast_waitfordigit(chan, 5000);
-	if (res > 0) {
-		res = do_directory(chan, cfg, context, dialcontext, res, last);
-		if (res > 0) {
+
+	for (;;) {
+		if (!res)
+			res = ast_streamfile(chan, dirintro, chan->language);
+		if (!res)
 			res = ast_waitstream(chan, AST_DIGIT_ANY);
-			ast_stopstream(chan);
-			if (res >= 0) {
-				goto top;
+		ast_stopstream(chan);
+		if (!res)
+			res = ast_waitfordigit(chan, 5000);
+		if (res > 0) {
+			res = do_directory(chan, cfg, context, dialcontext, res, last);
+			if (res > 0) {
+				res = ast_waitstream(chan, AST_DIGIT_ANY);
+				ast_stopstream(chan);
+				if (res >= 0) {
+					continue;
+				}
 			}
 		}
+		break;
 	}
-	ast_destroy(cfg);
+	ast_config_destroy(cfg);
 	LOCAL_USER_REMOVE(u);
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_disa.c
+++ b/apps/app_disa.c
@@ -1,33 +1,52 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * DISA -- Direct Inward System Access Application  6/20/2001
- * 
- * Copyright (C) 2001, Linux Support Services, Inc.
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Jim Dixon <jim@lambdatel.com>
 *
 * Made only slightly more sane by Mark Spencer <markster@digium.com>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief DISA -- Direct Inward System Access Application
+ *
+ * \author Jim Dixon <jim@lambdatel.com>
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/indications.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/ulaw.h>
 #include <string.h>
 #include <stdlib.h>
 #include <stdio.h>
 #include <math.h>
 #include <sys/time.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/app.h"
+#include "asterisk/indications.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/ulaw.h"
+#include "asterisk/callerid.h"

 static char *tdesc = "DISA (Direct Inward System Access) Application";

@@ -36,30 +55,27 @@ static char *app = "DISA";
 static char *synopsis = "DISA (Direct Inward System Access)";

 static char *descrip = 
-	"DISA (Direct Inward System Access) -- Allows someone from outside\n"
-	"the telephone switch (PBX) to obtain an \"internal\" system dialtone\n"
-	"and to place calls from it as if they were placing a call from within\n"
-	"the switch. A user calls a number that connects to the DISA application\n"
-	"and is given dialtone. The user enters their passcode, followed by the\n"
-	"pound sign (#). If the passcode is correct, the user is then given\n"
+	"DISA(<numeric passcode>[|<context>]) or disa(<filename>)\n"
+	"The DISA, Direct Inward System Access, application allows someone from \n"
+	"outside the telephone switch (PBX) to obtain an \"internal\" system \n"
+	"dialtone and to place calls from it as if they were placing a call from \n"
+	"within the switch.\n"
+	"DISA plays a dialtone. The user enters their numeric passcode, followed by\n"
+	"the pound sign (#). If the passcode is correct, the user is then given\n"
 	"system dialtone on which a call may be placed. Obviously, this type\n"
 	"of access has SERIOUS security implications, and GREAT care must be\n"
 	"taken NOT to compromise your security.\n\n"
 	"There is a possibility of accessing DISA without password. Simply\n"
-	"exchange your password with no-password.\n\n"
-	"  Example: exten => s,1,DISA,no-password|local\n\n"
-	"but be aware of using this for your security compromising.\n\n"
+	"exchange your password with \"no-password\".\n\n"
+	"    Example: exten => s,1,DISA(no-password|local)\n\n"
+	"Be aware that using this compromises the security of your PBX.\n\n"
 	"The arguments to this application (in extensions.conf) allow either\n"
-	"specification of a single global password (that everyone uses), or\n"
-	"individual passwords contained in a file. It also allow specification\n"
+	"specification of a single global passcode (that everyone uses), or\n"
+	"individual passcodes contained in a file. It also allow specification\n"
 	"of the context on which the user will be dialing. If no context is\n"
-	"specified, the DISA application defaults the context to \"disa\"\n"
-	"presumably that a normal system will have a special context set up\n"
-	"for DISA use with some or a lot of restrictions. The arguments are\n"
-	"one of the following:\n\n"
-	"    numeric-passcode\n"
-	"    numeric-passcode|context\n"
-	"    full-pathname-of-file-that-contains-passcodes\n\n"
+	"specified, the DISA application defaults the context to \"disa\".\n"
+	"Presumably a normal system will have a special context set up\n"
+	"for DISA use with some or a lot of restrictions. \n\n"
 	"The file that contains the passcodes (if used) allows specification\n"
 	"of either just a passcode (defaulting to the \"disa\" context, or\n"
 	"passcode|context on each line of the file. The file may contain blank\n"
@@ -67,35 +83,29 @@ static char *descrip =
 	"above arguments may have |new-callerid-string appended to them, to\n"
 	"specify a new (different) callerid to be used for this call, for\n"
 	"example: numeric-passcode|context|\"My Phone\" <(234) 123-4567> or \n"
-	"full-pathname-of-passcode-file|\"My Phone\" <(234) 123-4567>. Note that\n"
+	"full-pathname-of-passcode-file|\"My Phone\" <(234) 123-4567>.  Last\n"
+	"but not least, |mailbox[@context] may be appended, which will cause\n"
+	"a stutter-dialtone (indication \"dialrecall\") to be used, if the\n"
+	"specified mailbox contains any new messages, for example:\n"
+	"numeric-passcode|context||1234 (w/a changing callerid).  Note that\n"
 	"in the case of specifying the numeric-passcode, the context must be\n"
 	"specified if the callerid is specified also.\n\n"
-	"If login is successful, the application parses the dialed number in\n"
-	"the specified (or default) context, and returns 0 with the new extension\n"
-	"context filled-in and the priority set to 1, so that the PBX may\n"
-	"re-apply the routing tables to it and complete the call normally.";
-
+	"If login is successful, the application looks up the dialed number in\n"
+	"the specified (or default) context, and executes it if found.\n"
+	"If the user enters an invalid extension and extension \"i\" (invalid) \n"
+	"exists in the context, it will be used.\n";

 STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

-static int firstdigittimeout = 20000; /* 20 seconds first digit timeout */
-static int digittimeout = 10000; /* 10 seconds subsequent digit timeout */
-
-static int ms_diff(struct timeval *tv1, struct timeval *tv2)
-{
-int	ms;
-	
-	ms = (tv1->tv_sec - tv2->tv_sec) * 1000;
-	ms += (tv1->tv_usec - tv2->tv_usec) / 1000;
-	return(ms);
-}
-
-static void play_dialtone(struct ast_channel *chan)
+static void play_dialtone(struct ast_channel *chan, char *mailbox)
 {
 	const struct tone_zone_sound *ts = NULL;
-	ts = ast_get_indication_tone(chan->zone, "dial");
+	if(ast_app_has_voicemail(mailbox, NULL))
+		ts = ast_get_indication_tone(chan->zone, "dialrecall");
+	else
+		ts = ast_get_indication_tone(chan->zone, "dial");
 	if (ts)
 		ast_playtones_start(chan, 0, ts->data, 0);
 	else
@@ -105,49 +115,67 @@ static void play_dialtone(struct ast_channel *chan)
 static int disa_exec(struct ast_channel *chan, void *data)
 {
 	int i,j,k,x,did_ignore;
+	int firstdigittimeout = 20000;
+	int digittimeout = 10000;
 	struct localuser *u;
-	char tmp[256],arg2[256]="",exten[AST_MAX_EXTENSION],acctcode[20]="";
-	char *ourcontext,*ourcallerid;
+	char *tmp, exten[AST_MAX_EXTENSION],acctcode[20]="";
+	char pwline[256];
+	char ourcidname[256],ourcidnum[256];
 	struct ast_frame *f;
-	struct timeval lastout, now, lastdigittime;
+	struct timeval lastdigittime;
 	int res;
 	time_t rstart;
 	FILE *fp;
-	char *stringp=NULL;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(passcode);
+		AST_APP_ARG(context);
+		AST_APP_ARG(cid);
+		AST_APP_ARG(mailbox);
+	);

-	if (ast_set_write_format(chan,AST_FORMAT_ULAW))
-	{
-		ast_log(LOG_WARNING, "Unable to set write format to Mu-law on %s\n",chan->name);
-		return -1;
-	}
-	if (ast_set_read_format(chan,AST_FORMAT_ULAW))
-	{
-		ast_log(LOG_WARNING, "Unable to set read format to Mu-law on %s\n",chan->name);
-		return -1;
-	}
-	lastout.tv_sec = lastout.tv_usec = 0;
-	if (!data || !strlen((char *)data)) {
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "disa requires an argument (passcode/passcode file)\n");
 		return -1;
 	}
-	strncpy(tmp, (char *)data, sizeof(tmp)-1);
-	stringp=tmp;
-	strsep(&stringp, "|");
-	ourcontext = strsep(&stringp, "|");
-	/* if context specified, save 2nd arg and parse third */
-	if (ourcontext) {
-		strncpy(arg2,ourcontext, sizeof(arg2) - 1);
-		ourcallerid = strsep(&stringp,"|");
-	}
-	  /* if context not specified, use "disa" */
-	else {
-		arg2[0] = 0;
-		ourcallerid = NULL;
-		ourcontext = "disa";
-	}
+
 	LOCAL_USER_ADD(u);
-	if (chan->_state != AST_STATE_UP)
-	{
+	
+	if (chan->pbx) {
+		firstdigittimeout = chan->pbx->rtimeout*1000;
+		digittimeout = chan->pbx->dtimeout*1000;
+	}
+	
+	if (ast_set_write_format(chan,AST_FORMAT_ULAW)) {
+		ast_log(LOG_WARNING, "Unable to set write format to Mu-law on %s\n",chan->name);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	if (ast_set_read_format(chan,AST_FORMAT_ULAW)) {
+		ast_log(LOG_WARNING, "Unable to set read format to Mu-law on %s\n",chan->name);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	
+	ast_log(LOG_DEBUG, "Digittimeout: %d\n", digittimeout);
+	ast_log(LOG_DEBUG, "Responsetimeout: %d\n", firstdigittimeout);
+
+	tmp = ast_strdupa(data);
+	if (!tmp) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}	
+
+	AST_STANDARD_APP_ARGS(args, tmp);
+
+	if (ast_strlen_zero(args.context)) 
+		args.context = "disa";	
+	if (ast_strlen_zero(args.mailbox))
+		args.mailbox = "";
+
+	ast_log(LOG_DEBUG, "Mailbox: %s\n",args.mailbox);
+	
+	if (chan->_state != AST_STATE_UP) {
 		/* answer */
 		ast_answer(chan);
 	}
@@ -157,22 +185,19 @@ static int disa_exec(struct ast_channel *chan, void *data)
 	acctcode[0] = 0;
 	/* can we access DISA without password? */ 

-	ast_log(LOG_DEBUG, "Context: %s\n",ourcontext);
+	ast_log(LOG_DEBUG, "Context: %s\n",args.context);

-	if (!strcasecmp(tmp, "no-password"))
-	{;
+	if (!strcasecmp(args.passcode, "no-password")) {
 		k |= 1; /* We have the password */
 		ast_log(LOG_DEBUG, "DISA no-password login success\n");
 	}
-	gettimeofday(&lastdigittime,NULL);
+	lastdigittime = ast_tvnow();

-	play_dialtone(chan);
+	play_dialtone(chan, args.mailbox);

-	for(;;)
-	{
-		gettimeofday(&now,NULL);
+	for (;;) {
 		  /* if outa time, give em reorder */
-		if (ms_diff(&now,&lastdigittime) > 
+		if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) > 
 		    ((k&2) ? digittimeout : firstdigittimeout))
 		{
 			ast_log(LOG_DEBUG,"DISA %s entry timeout on chan %s\n",
@@ -215,7 +240,7 @@ static int disa_exec(struct ast_channel *chan, void *data)
 			k|=2; /* We have the first digit */ 
 			ast_playtones_stop(chan);
 		}
-		gettimeofday(&lastdigittime,NULL);
+		lastdigittime = ast_tvnow();
 		  /* got a DTMF tone */
 		if (i < AST_MAX_EXTENSION) /* if still valid number of digits */
 		{
@@ -224,43 +249,45 @@ static int disa_exec(struct ast_channel *chan, void *data)
 				if (j == '#') /* end of password */
 				{
 					  /* see if this is an integer */
-					if (sscanf(tmp,"%d",&j) < 1)
+					if (sscanf(args.passcode,"%d",&j) < 1)
 					   { /* nope, it must be a filename */
-						fp = fopen(tmp,"r");
+						fp = fopen(args.passcode,"r");
 						if (!fp)
 						   {
-							ast_log(LOG_WARNING,"DISA password file %s not found on chan %s\n",tmp,chan->name);
+							ast_log(LOG_WARNING,"DISA password file %s not found on chan %s\n",args.passcode,chan->name);
 							LOCAL_USER_REMOVE(u);
 							return -1;
 						   }
-						tmp[0] = 0;
-						while(fgets(tmp,sizeof(tmp) - 1,fp))
+						pwline[0] = 0;
+						while(fgets(pwline,sizeof(pwline) - 1,fp))
 						   {
-							char *stringp=NULL,*stringp2;
-							if (!tmp[0]) continue;
-							if (tmp[strlen(tmp) - 1] == '\n') 
-								tmp[strlen(tmp) - 1] = 0;
-							if (!tmp[0]) continue;
+							if (!pwline[0]) continue;
+							if (pwline[strlen(pwline) - 1] == '\n') 
+								pwline[strlen(pwline) - 1] = 0;
+							if (!pwline[0]) continue;
 							  /* skip comments */
-							if (tmp[0] == '#') continue;
-							if (tmp[0] == ';') continue;
-							stringp=tmp;
-							strsep(&stringp, "|");
-							stringp2=strsep(&stringp, "|");
-							if (stringp2) {
-								ourcontext=stringp2;
-								stringp2=strsep(&stringp, "|");
-								if (stringp2) ourcallerid=stringp2;
-							}
+							if (pwline[0] == '#') continue;
+							if (pwline[0] == ';') continue;
+
+							AST_STANDARD_APP_ARGS(args, pwline);
+			
+							ast_log(LOG_DEBUG, "Mailbox: %s\n",args.mailbox);
+
 							  /* password must be in valid format (numeric) */
-							if (sscanf(tmp,"%d",&j) < 1) continue;
+							if (sscanf(args.passcode,"%d",&j) < 1) continue;
 							  /* if we got it */
-							if (!strcmp(exten,tmp)) break;
+							if (!strcmp(exten,args.passcode)) {
+								if (ast_strlen_zero(args.context))
+									args.context = "disa";
+								if (ast_strlen_zero(args.mailbox))
+									args.mailbox = "";
+								break;
+							}
 						   }
 						fclose(fp);
 					   }
 					  /* compare the two */
-					if (strcmp(exten,tmp))
+					if (strcmp(exten,args.passcode))
 					{
 						ast_log(LOG_WARNING,"DISA on chan %s got bad password %s\n",chan->name,exten);
 						goto reorder;
@@ -268,12 +295,12 @@ static int disa_exec(struct ast_channel *chan, void *data)
 					}
 					 /* password good, set to dial state */
 					ast_log(LOG_DEBUG,"DISA on chan %s password is good\n",chan->name);
-					play_dialtone(chan);
+					play_dialtone(chan, args.mailbox);

 					k|=1; /* In number mode */
 					i = 0;  /* re-set buffer pointer */
 					exten[sizeof(acctcode)] = 0;
-					strncpy(acctcode,exten, sizeof(acctcode) - 1);
+					ast_copy_string(acctcode, exten, sizeof(acctcode));
 					exten[0] = 0;
 					ast_log(LOG_DEBUG,"Successful DISA log-in on chan %s\n",chan->name);
 					continue;
@@ -285,8 +312,8 @@ static int disa_exec(struct ast_channel *chan, void *data)
 			if (!(k&1)) continue; /* if getting password, continue doing it */
 			  /* if this exists */

-			if (ast_ignore_pattern(ourcontext, exten)) {
-				play_dialtone(chan);
+			if (ast_ignore_pattern(args.context, exten)) {
+				play_dialtone(chan, "");
 				did_ignore = 1;
 			} else
 				if (did_ignore) {
@@ -295,30 +322,43 @@ static int disa_exec(struct ast_channel *chan, void *data)
 				}

 			  /* if can do some more, do it */
-			if (!ast_matchmore_extension(chan,ourcontext,exten,1, chan->callerid)) {
+			if (!ast_matchmore_extension(chan,args.context,exten,1, chan->cid.cid_num)) {
 				break;
 			}
 		}
 	}

-	if ((k==3) && (ast_exists_extension(chan,ourcontext,exten,1, chan->callerid)))
-	{
-		ast_playtones_stop(chan);
-		/* We're authenticated and have a valid extension */
-		if (ourcallerid && *ourcallerid)
-		{
-			if (chan->callerid) free(chan->callerid);
-			chan->callerid = strdup(ourcallerid);
+	if (k == 3) {
+		int recheck = 0;
+		struct ast_flags flags = { AST_CDR_FLAG_POSTED };
+
+		if (!ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
+			pbx_builtin_setvar_helper(chan, "INVALID_EXTEN", exten);
+			exten[0] = 'i';
+			exten[1] = '\0';
+			recheck = 1;
+		}
+		if (!recheck || ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
+			ast_playtones_stop(chan);
+			/* We're authenticated and have a target extension */
+			if (!ast_strlen_zero(args.cid))
+			{
+				ast_callerid_split(args.cid, ourcidname, sizeof(ourcidname), ourcidnum, sizeof(ourcidnum));
+				ast_set_callerid(chan, ourcidnum, ourcidname, ourcidnum);
+			}
+
+			if (!ast_strlen_zero(acctcode))
+				ast_copy_string(chan->accountcode, acctcode, sizeof(chan->accountcode));
+
+			ast_cdr_reset(chan->cdr, &flags);
+			ast_explicit_goto(chan, args.context, exten, 1);
+			LOCAL_USER_REMOVE(u);
+			return 0;
 		}
-		strncpy(chan->exten, exten, sizeof(chan->exten) - 1);
-		strncpy(chan->context, ourcontext, sizeof(chan->context) - 1);
-		strncpy(chan->accountcode, acctcode, sizeof(chan->accountcode) - 1);
-		chan->priority = 0;
-		ast_cdr_init(chan->cdr,chan);
-		LOCAL_USER_REMOVE(u);
-		return 0;
 	}

+	/* Received invalid, but no "i" extension exists in the given context */
+
 reorder:

 	ast_indicate(chan,AST_CONTROL_CONGESTION);
@@ -340,8 +380,13 @@ reorder:

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_dumpchan.c
+++ b/apps/app_dumpchan.c
@@ -0,0 +1,192 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2004 - 2005, Anthony Minessale II.
+ *
+ * Anthony Minessale <anthmct@yahoo.com>
+ *
+ * disclaimed to Digium
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Application to dump channel variables
+ * 
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
+#include "asterisk/lock.h"
+#include "asterisk/utils.h"
+
+static char *tdesc = "Dump Info About The Calling Channel";
+static char *app = "DumpChan";
+static char *synopsis = "Dump Info About The Calling Channel";
+static char *desc = 
+"   DumpChan([<min_verbose_level>])\n"
+"Displays information on channel and listing of all channel\n"
+"variables. If min_verbose_level is specified, output is only\n"
+"displayed when the verbose level is currently set to that number\n"
+"or greater. \n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int ast_serialize_showchan(struct ast_channel *c, char *buf, size_t size)
+{
+	struct timeval now;
+	long elapsed_seconds=0;
+	int hour=0, min=0, sec=0;
+	char cgrp[256];
+	char pgrp[256];
+	
+	now = ast_tvnow();
+	memset(buf,0,size);
+	if (!c)
+		return 0;
+
+	if (c->cdr) {
+		elapsed_seconds = now.tv_sec - c->cdr->start.tv_sec;
+		hour = elapsed_seconds / 3600;
+		min = (elapsed_seconds % 3600) / 60;
+		sec = elapsed_seconds % 60;
+	}
+
+	snprintf(buf,size, 
+			 "Name=               %s\n"
+			 "Type=               %s\n"
+			 "UniqueID=           %s\n"
+			 "CallerID=           %s\n"
+			 "CallerIDName=       %s\n"
+			 "DNIDDigits=         %s\n"
+			 "State=              %s (%d)\n"
+			 "Rings=              %d\n"
+			 "NativeFormat=       %d\n"
+			 "WriteFormat=        %d\n"
+			 "ReadFormat=         %d\n"
+			 "1stFileDescriptor=  %d\n"
+			 "Framesin=           %d %s\n"
+			 "Framesout=          %d %s\n"
+			 "TimetoHangup=       %ld\n"
+			 "ElapsedTime=        %dh%dm%ds\n"
+			 "Context=            %s\n"
+			 "Extension=          %s\n"
+			 "Priority=           %d\n"
+			 "CallGroup=          %s\n"
+			 "PickupGroup=        %s\n"
+			 "Application=        %s\n"
+			 "Data=               %s\n"
+			 "Blocking_in=        %s\n",
+			 c->name,
+			 c->type,
+			 c->uniqueid,
+			 (c->cid.cid_num ? c->cid.cid_num : "(N/A)"),
+			 (c->cid.cid_name ? c->cid.cid_name : "(N/A)"),
+			 (c->cid.cid_dnid ? c->cid.cid_dnid : "(N/A)" ),
+			 ast_state2str(c->_state),
+			 c->_state,
+			 c->rings,
+			 c->nativeformats,
+			 c->writeformat,
+			 c->readformat,
+			 c->fds[0], c->fin & 0x7fffffff, (c->fin & 0x80000000) ? " (DEBUGGED)" : "",
+			 c->fout & 0x7fffffff, (c->fout & 0x80000000) ? " (DEBUGGED)" : "", (long)c->whentohangup,
+			 hour,
+			 min,
+			 sec,
+			 c->context,
+			 c->exten,
+			 c->priority,
+			 ast_print_group(cgrp, sizeof(cgrp), c->callgroup),
+			 ast_print_group(pgrp, sizeof(pgrp), c->pickupgroup),
+			 ( c->appl ? c->appl : "(N/A)" ),
+			 ( c-> data ? (!ast_strlen_zero(c->data) ? c->data : "(Empty)") : "(None)"),
+			 (ast_test_flag(c, AST_FLAG_BLOCKING) ? c->blockproc : "(Not Blocking)"));
+
+	return 0;
+}
+
+static int dumpchan_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char vars[1024];
+	char info[1024];
+	int level = 0;
+	static char *line = "================================================================================";
+	
+	LOCAL_USER_ADD(u);
+
+	if (!ast_strlen_zero(data)) {
+		level = atoi(data);
+	}
+
+	pbx_builtin_serialize_variables(chan, vars, sizeof(vars));
+	ast_serialize_showchan(chan, info, sizeof(info));
+	if (option_verbose >= level)
+		ast_verbose("\nDumping Info For Channel: %s:\n%s\nInfo:\n%s\nVariables:\n%s%s\n",chan->name, line, info, vars, line);
+
+	LOCAL_USER_REMOVE(u);
+	
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, dumpchan_exec, synopsis, desc);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
+
--- a/apps/app_echo.c
+++ b/apps/app_echo.c
@@ -1,26 +1,43 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Echo application -- play back what you hear to evaluate latency
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Echo application -- play back what you hear to evaluate latency
+ * 
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
 #include <stdlib.h>
+#include <stdio.h>
 #include <unistd.h>
 #include <string.h>
-#include <stdlib.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"

 static char *tdesc = "Simple Echo Application";

@@ -29,8 +46,9 @@ static char *app = "Echo";
 static char *synopsis = "Echo audio read back to the user";

 static char *descrip = 
-"  Echo():  Echo audio read from channel back to the channel. Returns 0\n"
-"if the user exits with the '#' key, or -1 if the user hangs up.\n";
+"  Echo():  Echo audio read from channel back to the channel. \n"
+"User can exit the application by either pressing the '#' key, \n"
+"or hanging up.\n";

 STANDARD_LOCAL_USER;

@@ -73,8 +91,13 @@ static int echo_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_enumlookup.c
+++ b/apps/app_enumlookup.c
@@ -1,49 +1,69 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Time of day - Report the time of day
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Enumlookup - lookup entry in ENUM
+ *
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/options.h>
-#include <asterisk/config.h>
-#include <asterisk/module.h>
-#include <asterisk/enum.h>
-#include <asterisk/utils.h>
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
+#include <unistd.h>
 #include <ctype.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/options.h"
+#include "asterisk/config.h"
+#include "asterisk/module.h"
+#include "asterisk/enum.h"
+#include "asterisk/utils.h"
+#include "asterisk/app.h"
+#include "asterisk/options.h"
+
 static char *tdesc = "ENUM Lookup";

 static char *app = "EnumLookup";

 static char *synopsis = "Lookup number in ENUM";

-static char *descrip = 
-"  EnumLookup(exten):  Looks up an extension via ENUM and sets\n"
+static char *descrip =
+"  EnumLookup(exten[|option]):  Looks up an extension via ENUM and sets\n"
 "the variable 'ENUM'. For VoIP URIs this variable will \n"
 "look like 'TECHNOLOGY/URI' with the appropriate technology.\n"
-"Returns -1 on hangup, or 0 on completion regardless of whether the \n"
-"lookup was successful. \n"
 "Currently, the enumservices SIP, H323, IAX, IAX2 and TEL are recognized. \n"
-"A good SIP, H323, IAX or IAX2 entry will result in normal priority handling, \n"
-"whereas a good TEL entry will increase the priority by 51 (if existing).\n"
-"If the lookup was *not* successful and there exists a priority n + 101,\n"
-"then that priority will be taken next.\n" ;
+"\nReturns status in the ENUMSTATUS channel variable:\n"
+"    ERROR	Failed to do a lookup\n"
+"    <tech>	Technology of the successful lookup: SIP, H323, IAX, IAX2 or TEL\n"
+"    BADURI	Got URI Asterisk does not understand.\n"
+"  The option string may contain zero or the following character:\n"
+"       'j' -- jump to +101 priority if the lookup isn't successful.\n"
+"	       and jump to +51 priority on a TEL entry.\n";

 #define ENUM_CONFIG "enum.conf"

@@ -54,24 +74,63 @@ STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

+/*--- enumlookup_exec: Look up number in ENUM and return result */
 static int enumlookup_exec(struct ast_channel *chan, void *data)
 {
-	int res=0;
+	int res=0,priority_jump=0;
 	char tech[80];
 	char dest[80];
 	char tmp[256];
-	char *c,*t;
+	char *c,*t = NULL;
+	static int dep_warning=0;
 	struct localuser *u;
-	if (!data || ast_strlen_zero(data)) {
+	char *parse;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(d);
+		AST_APP_ARG(o);
+	);
+
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "EnumLookup requires an argument (extension)\n");
-		res = 1;
+		return -1;
 	}
+		
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The application EnumLookup is deprecated.  Please use the ENUMLOOKUP() function instead.\n");
+		dep_warning = 1;
+	}
+
 	LOCAL_USER_ADD(u);
-	if (!res) {
-		res = ast_get_enum(chan, data, dest, sizeof(dest), tech, sizeof(tech));
-		printf("ENUM got '%d'\n", res);
+
+	parse = ast_strdupa(data);
+	if (!parse) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
 	}
-	LOCAL_USER_REMOVE(u);
+
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	tech[0] = '\0';
+	dest[0] = '\0';
+
+	if (args.o) {
+		if (strchr(args.o, 'j'))
+			priority_jump = 1;
+	}
+
+	res = ast_get_enum(chan, args.d, dest, sizeof(dest), tech, sizeof(tech), NULL, NULL);
+	
+	if (!res) {	/* Failed to do a lookup */
+		if (priority_jump || option_priority_jumping) {
+			/* Look for a "busy" place */
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		}
+		pbx_builtin_setvar_helper(chan, "ENUMSTATUS", "ERROR");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+	pbx_builtin_setvar_helper(chan, "ENUMSTATUS", tech);
 	/* Parse it out */
 	if (res > 0) {
 		if (!strcasecmp(tech, "SIP")) {
@@ -87,7 +146,7 @@ static int enumlookup_exec(struct ast_channel *chan, void *data)
 			snprintf(tmp, sizeof(tmp), "%s/%s", h323driver, c);
 /* do a s!;.*!! on the H323 URI */
 			t = strchr(c,';');
-			if (t) 
+                       if (t)
 				*t = 0;
 			pbx_builtin_setvar_helper(chan, "ENUM", tmp);
 		} else if (!strcasecmp(tech, "iax")) {
@@ -112,7 +171,7 @@ static int enumlookup_exec(struct ast_channel *chan, void *data)
 				res = 0;
 			} else {
 /* now copy over the number, skipping all non-digits and stop at ; or NULL */
-				t = tmp;	
+                               t = tmp;
 				while( *c && (*c != ';') && (t - tmp < (sizeof(tmp) - 1))) {
 					if (isdigit(*c))
 						*t++ = *c;
@@ -121,38 +180,37 @@ static int enumlookup_exec(struct ast_channel *chan, void *data)
 				*t = 0;
 				pbx_builtin_setvar_helper(chan, "ENUM", tmp);
 				ast_log(LOG_NOTICE, "tel: ENUM set to \"%s\"\n", tmp);
-				if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 51, chan->callerid))
-					chan->priority += 50;
-				else
-					res = 0;
+				if (priority_jump || option_priority_jumping) {
+					if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 51))
+						res = 0;
+				}
 			}
 		} else if (!ast_strlen_zero(tech)) {
 			ast_log(LOG_NOTICE, "Don't know how to handle technology '%s'\n", tech);
+			pbx_builtin_setvar_helper(chan, "ENUMSTATUS", "BADURI");
 			res = 0;
 		}
 	}
-	if (!res) {
-		/* Look for a "busy" place */
-		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-			chan->priority += 100;
-	} else if (res > 0)
-		res = 0;
-	return res;
+
+	LOCAL_USER_REMOVE(u);
+
+	return 0;
 }

+/*--- load_config: Load enum.conf and find out how to handle H.323 */
 static int load_config(void)
 {
 	struct ast_config *cfg;
 	char *s;

-	cfg = ast_load(ENUM_CONFIG);
+	cfg = ast_config_load(ENUM_CONFIG);
 	if (cfg) {
 		if (!(s=ast_variable_retrieve(cfg, "general", "h323driver"))) {
 			strncpy(h323driver, H323DRIVERDEFAULT, sizeof(h323driver) - 1);
 		} else {
 			strncpy(h323driver, s, sizeof(h323driver) - 1);
 		}
-		ast_destroy(cfg);
+		ast_config_destroy(cfg);
 		return 0;
 	}
 	ast_log(LOG_NOTICE, "No ENUM Config file, using defaults\n");
@@ -160,30 +218,39 @@ static int load_config(void)
 }


+/*--- unload_module: Unload this application from PBX */
 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

+/*--- load_module: Load this application into PBX */
 int load_module(void)
 {
 	int res;
+	
 	res = ast_register_application(app, enumlookup_exec, synopsis, descrip);
-	if (res)
-		return(res);
-	if ((res=load_config())) {
-		return(res);
-	}
-	return(0);
+	
+	if (!res)
+		res = load_config();
+	
+	return res;
 }

+/*--- reload: Reload configuration file */
 int reload(void)
 {
-	return(load_config());
+	return load_config();
 }


+/*--- description: Describe module */
 char *description(void)
 {
 	return tdesc;
--- a/apps/app_eval.c
+++ b/apps/app_eval.c
@@ -1,28 +1,43 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Eval application
- *
- * Copyright (c) 2004 Tilghman Lesher.  All rights reserved.
+ * Copyright (c) 2004 - 2005, Tilghman Lesher.  All rights reserved.
 *
 * Tilghman Lesher <app_eval__v001@the-tilghman.com>
 *
- * $Id$
- *
 * This code is released by the author with no restrictions on usage.
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ * \brief Eval application
+ *
+ * \author Tilghman Lesher <app_eval__v001@the-tilghman.com>
+ *
+ * \ingroup applications
 */

 #include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/options.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"

 /* Maximum length of any variable */
 #define MAXRESULT	1024
@@ -49,8 +64,14 @@ static int eval_exec(struct ast_channel *chan, void *data)
 	int res=0;
 	struct localuser *u;
 	char *s, *newvar=NULL, tmp[MAXRESULT];
+	static int dep_warning = 0;

 	LOCAL_USER_ADD(u);
+	
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "This application has been deprecated in favor of the dialplan function, EVAL\n");
+		dep_warning = 1;
+	}

 	/* Check and parse arguments */
 	if (data) {
@@ -74,8 +95,13 @@ static int eval_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app_eval);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app_eval);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_exec.c
+++ b/apps/app_exec.c
@@ -1,28 +1,44 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Exec application
- *
- * Copyright (c) 2004 Tilghman Lesher.  All rights reserved.
+ * Copyright (c) 2004 - 2005, Tilghman Lesher.  All rights reserved.
 *
 * Tilghman Lesher <app_exec__v001@the-tilghman.com>
 *
- * $Id$
- *
 * This code is released by the author with no restrictions on usage.
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ *
+ * \brief Exec application
+ *
+ * \author Tilghman Lesher <app_exec__v001@the-tilghman.com>
+ *
+ * \ingroup applications
 */

 #include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/options.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"

 /* Maximum length of any variable */
 #define MAXRESULT	1024
@@ -31,13 +47,14 @@ static char *tdesc = "Executes applications";

 static char *app_exec = "Exec";

-static char *exec_synopsis = "Exec(Appname(arguments))";
+static char *exec_synopsis = "Executes internal application";

 static char *exec_descrip =
-"Exec(appname(arguments))\n"
+"Usage: Exec(appname(arguments))\n"
 "  Allows an arbitrary application to be invoked even when not\n"
-"hardcoded into the dialplan.  Returns whatever value the\n"
-"app returns or -2 when the app cannot be found.\n";
+"hardcoded into the dialplan. To invoke external applications\n"
+"see the application System. Returns whatever value the\n"
+"app returns or a non-zero value if the app cannot be found.\n";

 STANDARD_LOCAL_USER;

@@ -71,7 +88,7 @@ static int exec_exec(struct ast_channel *chan, void *data)
 					res = pbx_exec(chan, app, args, 1);
 				} else {
 					ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
-					res = -2;
+					res = -1;
 				}
 			}
 		} else {
@@ -86,8 +103,13 @@ static int exec_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app_exec);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app_exec);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_externalivr.c
+++ b/apps/app_externalivr.c
@@ -0,0 +1,579 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Kevin P. Fleming <kpfleming@digium.com>
+ *
+ * Portions taken from the file-based music-on-hold work
+ * created by Anthony Minessale II in res_musiconhold.c
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief External IVR application interface
+ * 
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <errno.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/linkedlists.h"
+
+static const char *tdesc = "External IVR Interface Application";
+
+static const char *app = "ExternalIVR";
+
+static const char *synopsis = "Interfaces with an external IVR application";
+
+static const char *descrip = 
+"  ExternalIVR(command[|arg[|arg...]]): Forks an process to run the supplied command,\n"
+"and starts a generator on the channel. The generator's play list is\n"
+"controlled by the external application, which can add and clear entries\n"
+"via simple commands issued over its stdout. The external application\n"
+"will receive all DTMF events received on the channel, and notification\n"
+"if the channel is hung up. The application will not be forcibly terminated\n"
+"when the channel is hung up.\n"
+"See doc/README.externalivr for a protocol specification.\n";
+
+/* XXX the parser in gcc 2.95 gets confused if you don't put a space between 'name' and the comma */
+#define ast_chan_log(level, channel, format, ...) ast_log(level, "%s: " format, channel->name , ## __VA_ARGS__)
+
+struct playlist_entry {
+	AST_LIST_ENTRY(playlist_entry) list;
+	char filename[1];
+};
+
+struct localuser {
+	struct ast_channel *chan;
+	struct localuser *next;
+	AST_LIST_HEAD(playlist, playlist_entry) playlist;
+	AST_LIST_HEAD(finishlist, playlist_entry) finishlist;
+	int abort_current_sound;
+	int playing_silence;
+	int option_autoclear;
+};
+
+LOCAL_USER_DECL;
+
+struct gen_state {
+	struct localuser *u;
+	struct ast_filestream *stream;
+	struct playlist_entry *current;
+	int sample_queue;
+};
+
+static void send_child_event(FILE *handle, const char event, const char *data,
+			     const struct ast_channel *chan)
+{
+	char tmp[256];
+
+	if (!data) {
+		snprintf(tmp, sizeof(tmp), "%c,%10ld", event, time(NULL));
+	} else {
+		snprintf(tmp, sizeof(tmp), "%c,%10ld,%s", event, time(NULL), data);
+	}
+
+	fprintf(handle, "%s\n", tmp);
+	ast_chan_log(LOG_DEBUG, chan, "sent '%s'\n", tmp);
+}
+
+static void *gen_alloc(struct ast_channel *chan, void *params)
+{
+	struct localuser *u = params;
+	struct gen_state *state;
+
+	state = calloc(1, sizeof(*state));
+
+	if (!state)
+		return NULL;
+
+	state->u = u;
+
+	return state;
+}
+
+static void gen_closestream(struct gen_state *state)
+{
+	if (!state->stream)
+		return;
+
+	ast_closestream(state->stream);
+	state->u->chan->stream = NULL;
+	state->stream = NULL;
+}
+
+static void gen_release(struct ast_channel *chan, void *data)
+{
+	struct gen_state *state = data;
+
+	gen_closestream(state);
+	free(data);
+}
+
+/* caller has the playlist locked */
+static int gen_nextfile(struct gen_state *state)
+{
+	struct localuser *u = state->u;
+	char *file_to_stream;
+	
+	u->abort_current_sound = 0;
+	u->playing_silence = 0;
+	gen_closestream(state);
+
+	while (!state->stream) {
+		state->current = AST_LIST_REMOVE_HEAD(&u->playlist, list);
+		if (state->current) {
+			file_to_stream = state->current->filename;
+		} else {
+			file_to_stream = "silence-10";
+			u->playing_silence = 1;
+		}
+
+		if (!(state->stream = ast_openstream_full(u->chan, file_to_stream, u->chan->language, 1))) {
+			ast_chan_log(LOG_WARNING, u->chan, "File '%s' could not be opened: %s\n", file_to_stream, strerror(errno));
+			if (!u->playing_silence) {
+				continue;
+			} else { 
+				break;
+			}
+		}
+	}
+
+	return (!state->stream);
+}
+
+static struct ast_frame *gen_readframe(struct gen_state *state)
+{
+	struct ast_frame *f = NULL;
+	struct localuser *u = state->u;
+	
+	if (u->abort_current_sound ||
+	    (u->playing_silence && AST_LIST_FIRST(&u->playlist))) {
+		gen_closestream(state);
+		AST_LIST_LOCK(&u->playlist);
+		gen_nextfile(state);
+		AST_LIST_UNLOCK(&u->playlist);
+	}
+
+	if (!(state->stream && (f = ast_readframe(state->stream)))) {
+		if (state->current) {
+			AST_LIST_LOCK(&u->finishlist);
+			AST_LIST_INSERT_TAIL(&u->finishlist, state->current, list);
+			AST_LIST_UNLOCK(&u->finishlist);
+			state->current = NULL;
+		}
+		if (!gen_nextfile(state))
+			f = ast_readframe(state->stream);
+	}
+
+	return f;
+}
+
+static int gen_generate(struct ast_channel *chan, void *data, int len, int samples)
+{
+	struct gen_state *state = data;
+	struct ast_frame *f = NULL;
+	int res = 0;
+
+	state->sample_queue += samples;
+
+	while (state->sample_queue > 0) {
+		if (!(f = gen_readframe(state)))
+			return -1;
+
+		res = ast_write(chan, f);
+		ast_frfree(f);
+		if (res < 0) {
+			ast_chan_log(LOG_WARNING, chan, "Failed to write frame: %s\n", strerror(errno));
+			return -1;
+		}
+		state->sample_queue -= f->samples;
+	}
+
+	return res;
+}
+
+static struct ast_generator gen =
+{
+	alloc: gen_alloc,
+	release: gen_release,
+	generate: gen_generate,
+};
+
+static struct playlist_entry *make_entry(const char *filename)
+{
+	struct playlist_entry *entry;
+
+	entry = calloc(1, sizeof(*entry) + strlen(filename) + 10);
+
+	if (!entry)
+		return NULL;
+
+	strcpy(entry->filename, filename);
+
+	return entry;
+}
+
+static int app_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u = NULL;
+	struct playlist_entry *entry;
+	const char *args = data;
+	int child_stdin[2] = { 0,0 };
+	int child_stdout[2] = { 0,0 };
+	int child_stderr[2] = { 0,0 };
+	int res = -1;
+	int gen_active = 0;
+	int pid;
+	char *command;
+	char *argv[32];
+	int argc = 1;
+	char *buf;
+	FILE *child_commands = NULL;
+	FILE *child_errors = NULL;
+	FILE *child_events = NULL;
+
+	LOCAL_USER_ADD(u);
+	
+	AST_LIST_HEAD_INIT(&u->playlist);
+	AST_LIST_HEAD_INIT(&u->finishlist);
+	u->abort_current_sound = 0;
+	
+	if (ast_strlen_zero(args)) {
+		ast_log(LOG_WARNING, "ExternalIVR requires a command to execute\n");
+		goto exit;
+	}
+
+	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 (pipe(child_stdin)) {
+		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child input: %s\n", strerror(errno));
+		goto exit;
+	}
+
+	if (pipe(child_stdout)) {
+		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child output: %s\n", strerror(errno));
+		goto exit;
+	}
+
+	if (pipe(child_stderr)) {
+		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child errors: %s\n", strerror(errno));
+		goto exit;
+	}
+
+	if (chan->_state != AST_STATE_UP) {
+		ast_answer(chan);
+	}
+
+	if (ast_activate_generator(chan, &gen, u) < 0) {
+		ast_chan_log(LOG_WARNING, chan, "Failed to activate generator\n");
+		goto exit;
+	} else
+		gen_active = 1;
+
+	pid = fork();
+	if (pid < 0) {
+		ast_log(LOG_WARNING, "Failed to fork(): %s\n", strerror(errno));
+		goto exit;
+	}
+
+	if (!pid) {
+		/* child process */
+		int i;
+
+		dup2(child_stdin[0], STDIN_FILENO);
+		dup2(child_stdout[1], STDOUT_FILENO);
+		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));
+		exit(1);
+	} else {
+		/* parent process */
+		int child_events_fd = child_stdin[1];
+		int child_commands_fd = child_stdout[0];
+		int child_errors_fd = child_stderr[0];
+		struct ast_frame *f;
+		int ms;
+		int exception;
+		int ready_fd;
+		int waitfds[2] = { child_errors_fd, child_commands_fd };
+		struct ast_channel *rchan;
+
+		close(child_stdin[0]);
+		child_stdin[0] = 0;
+		close(child_stdout[1]);
+		child_stdout[1] = 0;
+		close(child_stderr[1]);
+		child_stderr[1] = 0;
+
+		if (!(child_events = fdopen(child_events_fd, "w"))) {
+			ast_chan_log(LOG_WARNING, chan, "Could not open stream for child events\n");
+			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;
+		}
+
+		if (!(child_errors = fdopen(child_errors_fd, "r"))) {
+			ast_chan_log(LOG_WARNING, chan, "Could not open stream for child errors\n");
+			goto exit;
+		}
+
+		res = 0;
+
+		while (1) {
+			if (ast_test_flag(chan, AST_FLAG_ZOMBIE)) {
+				ast_chan_log(LOG_NOTICE, chan, "Is a zombie\n");
+				res = -1;
+				break;
+			}
+
+			if (ast_check_hangup(chan)) {
+				ast_chan_log(LOG_NOTICE, chan, "Got check_hangup\n");
+				send_child_event(child_events, 'H', NULL, chan);
+				res = -1;
+				break;
+			}
+
+			ready_fd = 0;
+			ms = 100;
+			errno = 0;
+			exception = 0;
+
+			rchan = ast_waitfor_nandfds(&chan, 1, waitfds, 2, &exception, &ready_fd, &ms);
+
+			if (!AST_LIST_EMPTY(&u->finishlist)) {
+				AST_LIST_LOCK(&u->finishlist);
+				while ((entry = AST_LIST_REMOVE_HEAD(&u->finishlist, list))) {
+					send_child_event(child_events, 'F', entry->filename, chan);
+					free(entry);
+				}
+				AST_LIST_UNLOCK(&u->finishlist);
+			}
+
+			if (rchan) {
+				/* the channel has something */
+				f = ast_read(chan);
+				if (!f) {
+					ast_chan_log(LOG_NOTICE, chan, "Returned no frame\n");
+					send_child_event(child_events, 'H', NULL, chan);
+					res = -1;
+					break;
+				}
+
+				if (f->frametype == AST_FRAME_DTMF) {
+					send_child_event(child_events, f->subclass, NULL, chan);
+					if (u->option_autoclear) {
+						if (!u->abort_current_sound && !u->playing_silence)
+							send_child_event(child_events, 'T', NULL, chan);
+						AST_LIST_LOCK(&u->playlist);
+						while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+							send_child_event(child_events, 'D', entry->filename, chan);
+							free(entry);
+						}
+						if (!u->playing_silence)
+							u->abort_current_sound = 1;
+						AST_LIST_UNLOCK(&u->playlist);
+					}
+				} else if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP)) {
+					ast_chan_log(LOG_NOTICE, chan, "Got AST_CONTROL_HANGUP\n");
+					send_child_event(child_events, 'H', NULL, chan);
+					ast_frfree(f);
+					res = -1;
+					break;
+				}
+				ast_frfree(f);
+			} else if (ready_fd == child_commands_fd) {
+				char input[1024];
+
+				if (exception || feof(child_commands)) {
+					ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
+					res = -1;
+					break;
+				}
+
+				if (!fgets(input, sizeof(input), child_commands))
+					continue;
+
+				command = ast_strip(input);
+
+				ast_chan_log(LOG_DEBUG, chan, "got command '%s'\n", input);
+
+				if (strlen(input) < 4)
+					continue;
+
+				if (input[0] == 'S') {
+					if (ast_fileexists(&input[2], NULL, NULL) == -1) {
+						ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
+						send_child_event(child_events, 'Z', NULL, chan);
+						strcpy(&input[2], "exception");
+					}
+					if (!u->abort_current_sound && !u->playing_silence)
+						send_child_event(child_events, 'T', NULL, chan);
+					AST_LIST_LOCK(&u->playlist);
+					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+						send_child_event(child_events, 'D', entry->filename, chan);
+						free(entry);
+					}
+					if (!u->playing_silence)
+						u->abort_current_sound = 1;
+					entry = make_entry(&input[2]);
+					if (entry)
+						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
+					AST_LIST_UNLOCK(&u->playlist);
+				} else if (input[0] == 'A') {
+					if (ast_fileexists(&input[2], NULL, NULL) == -1) {
+						ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
+						send_child_event(child_events, 'Z', NULL, chan);
+						strcpy(&input[2], "exception");
+					}
+					entry = make_entry(&input[2]);
+					if (entry) {
+						AST_LIST_LOCK(&u->playlist);
+						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
+						AST_LIST_UNLOCK(&u->playlist);
+					}
+				} else if (input[0] == 'H') {
+					ast_chan_log(LOG_NOTICE, chan, "Hanging up: %s\n", &input[2]);
+					send_child_event(child_events, 'H', NULL, chan);
+					break;
+				} else if (input[0] == 'O') {
+					if (!strcasecmp(&input[2], "autoclear"))
+						u->option_autoclear = 1;
+					else if (!strcasecmp(&input[2], "noautoclear"))
+						u->option_autoclear = 0;
+					else
+						ast_chan_log(LOG_WARNING, chan, "Unknown option requested '%s'\n", &input[2]);
+				}
+			} else if (ready_fd == child_errors_fd) {
+				char input[1024];
+
+				if (exception || feof(child_errors)) {
+					ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
+					res = -1;
+					break;
+				}
+
+				if (fgets(input, sizeof(input), child_errors)) {
+					command = ast_strip(input);
+					ast_chan_log(LOG_NOTICE, chan, "stderr: %s\n", command);
+				}
+			} else if ((ready_fd < 0) && ms) { 
+				if (errno == 0 || errno == EINTR)
+					continue;
+
+				ast_chan_log(LOG_WARNING, chan, "Wait failed (%s)\n", strerror(errno));
+				break;
+			}
+		}
+	}
+
+ exit:
+	if (gen_active)
+		ast_deactivate_generator(chan);
+
+	if (child_events)
+		fclose(child_events);
+
+	if (child_commands)
+		fclose(child_commands);
+
+	if (child_errors)
+		fclose(child_errors);
+
+	if (child_stdin[0])
+		close(child_stdin[0]);
+
+	if (child_stdin[1])
+		close(child_stdin[1]);
+
+	if (child_stdout[0])
+		close(child_stdout[0]);
+
+	if (child_stdout[1])
+		close(child_stdout[1]);
+
+	if (child_stderr[0])
+		close(child_stderr[0]);
+
+	if (child_stderr[1])
+		close(child_stderr[1]);
+
+	while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list)))
+		free(entry);
+
+	LOCAL_USER_REMOVE(u);
+
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, app_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_festival.c
+++ b/apps/app_festival.c
@@ -1,26 +1,29 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Connect to festival
- * 
 * Copyright (C) 2002, Christos Ricudis
 *
 * Christos Ricudis <ricudis@itc.auth.gr>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Connect to festival
+ * 
+ * \ingroup applications
 */

 #include <sys/types.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/md5.h>
-#include <asterisk/config.h>
-#include <asterisk/utils.h>
-#include <asterisk/lock.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
@@ -37,6 +40,19 @@
 #include <fcntl.h>
 #include <ctype.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/md5.h"
+#include "asterisk/config.h"
+#include "asterisk/utils.h"
+#include "asterisk/lock.h"

 #define FESTIVAL_CONFIG "festival.conf"

@@ -120,7 +136,7 @@ static int send_waveform_to_fd(char *waveform, int length, int fd) {
                if (x != fd)
                        close(x);
        }
-//IAS
+/*IAS */
 #ifdef __PPC__  
 	for( x=0; x<length; x+=2)
 	{
@@ -215,7 +231,7 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 						res = -1;
 						break;
 					}
-					if (res < needed) { // last frame
+					if (res < needed) { /* last frame */
 						ast_log(LOG_DEBUG, "Last frame\n");
 						res=0;
 						break;
@@ -230,8 +246,9 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 	}
 	close(fds[0]);
 	close(fds[1]);
-//	if (pid > -1)
-//		kill(pid, SIGKILL);
+
+/*	if (pid > -1) */
+/*		kill(pid, SIGKILL); */
 	if (!res && owriteformat)
 		ast_set_write_format(chan, owriteformat);
 	return res;
@@ -276,13 +293,21 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
 	int fdesc = -1;
 	char buffer[16384];
 	int seekpos = 0;	
-	char data[256] = "";
+	char *data;	
 	char *intstr;
-	
 	struct ast_config *cfg;
-	cfg = ast_load(FESTIVAL_CONFIG);
+
+	if (ast_strlen_zero(vdata)) {
+		ast_log(LOG_WARNING, "festival requires an argument (text)\n");
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	cfg = ast_config_load(FESTIVAL_CONFIG);
 	if (!cfg) {
 		ast_log(LOG_WARNING, "No such configuration file %s\n", FESTIVAL_CONFIG);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	if (!(host = ast_variable_retrieve(cfg, "general", "host"))) {
@@ -304,19 +329,23 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
 	if (!(festivalcommand = ast_variable_retrieve(cfg, "general", "festivalcommand"))) {
 		festivalcommand = "(tts_textasterisk \"%s\" 'file)(quit)\n";
 	}
-	if (!vdata || ast_strlen_zero(vdata)) {
-		ast_log(LOG_WARNING, "festival requires an argument (text)\n");
-		ast_destroy(cfg);
+	
+	data = ast_strdupa(vdata);
+	if (!data) {
+		ast_log(LOG_ERROR, "Out of memery\n");
+		ast_config_destroy(cfg);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
-	strncpy(data, vdata, sizeof(data) - 1);
-	if ((intstr = strchr(data, '|'))) {
+
+	intstr = strchr(data, '|');
+	if (intstr) {	
 		*intstr = '\0';
 		intstr++;
 		if (!strcasecmp(intstr, "any"))
 			intstr = AST_DIGIT_ANY;
 	}
-	LOCAL_USER_ADD(u);
+	
 	ast_log(LOG_DEBUG, "Text passed to festival server : %s\n",(char *)data);
 	/* Connect to local festival server */
 	
@@ -324,7 +353,8 @@ static int festival_exec(struct ast_channel *chan, void *vdata)

    	if (fd < 0) {
 		ast_log(LOG_WARNING,"festival_client: can't get socket\n");
-		ast_destroy(cfg);
+		ast_config_destroy(cfg);
+		LOCAL_USER_REMOVE(u);
        	return -1;
 	}
        memset(&serv_addr, 0, sizeof(serv_addr));
@@ -333,7 +363,8 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
 	        serverhost = ast_gethostbyname(host, &ahp);
 	        if (serverhost == (struct hostent *)0) {
        	    	ast_log(LOG_WARNING,"festival_client: gethostbyname failed\n");
-			ast_destroy(cfg);
+			ast_config_destroy(cfg);
+			LOCAL_USER_REMOVE(u);
 	            	return -1;
        	}
 	        memmove(&serv_addr.sin_addr,serverhost->h_addr, serverhost->h_length);
@@ -343,7 +374,8 @@ static int festival_exec(struct ast_channel *chan, void *vdata)

 	if (connect(fd, (struct sockaddr *)&serv_addr, sizeof(serv_addr)) != 0) {
 		ast_log(LOG_WARNING,"festival_client: connect to server failed\n");
-		ast_destroy(cfg);
+		ast_config_destroy(cfg);
+		LOCAL_USER_REMOVE(u);
        	return -1;
    	}
    	
@@ -446,16 +478,21 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
    		}
 	} while (strcmp(ack,"OK\n") != 0);
 	close(fd);
-	ast_destroy(cfg);
-	LOCAL_USER_REMOVE(u);                                                                                
+	ast_config_destroy(cfg);
+	LOCAL_USER_REMOVE(u);
 	return res;

 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_flash.c
+++ b/apps/app_flash.c
@@ -1,34 +1,52 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * App to flash a zap trunk
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to flash a zap trunk
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/image.h>
-#include <asterisk/options.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <errno.h>
 #include <sys/ioctl.h>
 #ifdef __linux__
 #include <linux/zaptel.h>
 #else
 #include <zaptel.h>
 #endif /* __linux__ */
-#include <string.h>
-#include <errno.h>
-#include <stdlib.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/options.h"

 static char *tdesc = "Flash zap trunk application";

@@ -39,8 +57,7 @@ static char *synopsis = "Flashes a Zap Trunk";
 static char *descrip = 
 "  Flash(): Sends a flash on a zap trunk.  This is only a hack for\n"
 "people who want to perform transfers and such via AGI and is generally\n"
-"quite useless otherwise.  Returns 0 on success or -1 if this is not\n"
-"a zap trunk\n";
+"quite useless oths application will only work on Zap trunks.\n";

 STANDARD_LOCAL_USER;

@@ -92,8 +109,13 @@ static int flash_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_forkcdr.c
+++ b/apps/app_forkcdr.c
@@ -1,60 +1,96 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Fork CDR application
- * Copyright Anthony Minessale anthmct@yahoo.com
+ * Copyright (C) 1999 - 2005, Anthony Minessale anthmct@yahoo.com
 * Development of this app Sponsered/Funded  by TAAN Softworks Corp
- * 
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Fork CDR application
+ * 
+ * \ingroup applications
 */

-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/cdr.h>
-#include <asterisk/module.h>
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <pthread.h>
+#include <unistd.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/cdr.h"
+#include "asterisk/module.h"

 static char *tdesc = "Fork The CDR into 2 separate entities.";
 static char *app = "ForkCDR";
 static char *synopsis = 
 "Forks the Call Data Record";
 static char *descrip = 
-"  ForkCDR():  Causes the Call Data Record to fork an additional\n"
-	"cdr record starting from the time of the fork call\n";
+"  ForkCDR([options]):  Causes the Call Data Record to fork an additional\n"
+	"cdr record starting from the time of the fork call\n"
+"If the option 'v' is passed all cdr variables will be passed along also.\n"
+"";


 STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

+static void ast_cdr_fork(struct ast_channel *chan) 
+{
+	struct ast_cdr *cdr;
+	struct ast_cdr *newcdr;
+	struct ast_flags flags = { AST_CDR_FLAG_KEEP_VARS };

-static void ast_cdr_clone(struct ast_cdr *cdr) {
-	struct ast_cdr *newcdr = ast_cdr_alloc();
-	memcpy(newcdr,cdr,sizeof(struct ast_cdr));
-	ast_cdr_append(cdr,newcdr);
-	gettimeofday(&newcdr->start, NULL);
-	memset(&newcdr->answer, 0, sizeof(newcdr->answer));
-	ast_cdr_add_flag(cdr,AST_CDR_FLAG_CHILD|AST_CDR_FLAG_LOCKED);
-}
+	cdr = chan->cdr;

-static void ast_cdr_fork(struct ast_channel *chan) {
-	if(chan && chan->cdr) {
-		ast_cdr_clone(chan->cdr);
-	}
+	while (cdr->next)
+		cdr = cdr->next;
+	
+	if (!(newcdr = ast_cdr_dup(cdr)))
+		return;
+	
+	ast_cdr_append(cdr, newcdr);
+	ast_cdr_reset(newcdr, &flags);
+	
+	if (!ast_test_flag(cdr, AST_CDR_FLAG_KEEP_VARS))
+		ast_cdr_free_vars(cdr, 0);
+	
+	ast_set_flag(cdr, AST_CDR_FLAG_CHILD | AST_CDR_FLAG_LOCKED);
 }

 static int forkcdr_exec(struct ast_channel *chan, void *data)
 {
-	int res=0;
+	int res = 0;
 	struct localuser *u;
+
+	if (!chan->cdr) {
+		ast_log(LOG_WARNING, "Channel does not have a CDR\n");
+		return 0;
+	}
+
 	LOCAL_USER_ADD(u);

+	if (!ast_strlen_zero(data))
+		ast_set2_flag(chan->cdr, strchr(data, 'v'), AST_CDR_FLAG_KEEP_VARS);
+	
 	ast_cdr_fork(chan);

 	LOCAL_USER_REMOVE(u);
@@ -63,8 +99,13 @@ static int forkcdr_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_getcpeid.c
+++ b/apps/app_getcpeid.c
@@ -1,28 +1,45 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Execute arbitrary system commands
- * 
- * Copyright (C) 1999-2005, Digium
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Get ADSI CPE ID
+ * 
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/adsi.h>
-#include <asterisk/options.h>
 #include <stdlib.h>
+#include <stdio.h>
 #include <unistd.h>
 #include <string.h>
-#include <stdlib.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/adsi.h"
+#include "asterisk/options.h"

 static char *tdesc = "Get ADSI CPE ID";

@@ -32,8 +49,7 @@ static char *synopsis = "Get ADSI CPE ID";

 static char *descrip =
 "  GetCPEID: Obtains and displays ADSI CPE ID and other information in order\n"
-"to properly setup zapata.conf for on-hook operations.\n"
-"Returns -1 on hangup only.\n";
+"to properly setup zapata.conf for on-hook operations.\n";

 STANDARD_LOCAL_USER;

@@ -119,8 +135,13 @@ static int cpeid_exec(struct ast_channel *chan, void *idata)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_groupcount.c
+++ b/apps/app_groupcount.c
@@ -1,146 +1,131 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Group Manipulation Applications
- *
- * Copyright (c) 2004 Digium
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Group Manipulation Applications
+ *
+ * \ingroup applications
 */

 #include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/options.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/utils.h>
+#include <sys/types.h>
+#include <regex.h>

-static char *tdesc = "Group Management Routines";
+#include "asterisk.h"

-static char *app_group_count = "GetGroupCount";
-static char *app_group_set = "SetGroup";
-static char *app_group_check = "CheckGroup";
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

-static char *group_count_synopsis = "GetGroupCount([groupname][@category])";
-static char *group_set_synopsis = "SetGroup(groupname[@category])";
-static char *group_check_synopsis = "CheckGroup(max[@category])";
-
-static char *group_count_descrip =
-"GetGroupCount([group][@category])\n"
-"  Calculates the group count for the specified group, or uses\n"
-"the current channel's group if not specifed (and non-empty).\n"
-"Stores result in GROUPCOUNT.  Always returns 0.\n";
-
-static char *group_set_descrip =
-"SetGroup(group)\n"
-"  Sets the channel group to the specified value.  Equivalent to\n"
-"SetVar(GROUP=group).  Always returns 0.\n";
-
-static char *group_check_descrip =
-"CheckGroup(max)\n"
-"  Checks that the current number of total channels in the\n"
-"current channel's group does not exceed 'max'.  If the number\n"
-"does not exceed 'max', we continue to the next step. If the\n"
-"number does in fact exceed max, if priority n+101 exists, then\n"
-"execution continues at that step, otherwise -1 is returned.\n";
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/utils.h"
+#include "asterisk/cli.h"
+#include "asterisk/app.h"

 STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

-#define DEFAULT_CATEGORY "GROUP"
-
-static int group_get_count(char *group, char *category)
-{
-	struct ast_channel *chan;
-	int count = 0;
-	char *test;
-	if (group && !ast_strlen_zero(group)) {
-		chan = ast_channel_walk_locked(NULL);
-		while(chan) {
-			test = pbx_builtin_getvar_helper(chan, category);
-			if (test && !strcasecmp(test, group))
-				count++;
-			ast_mutex_unlock(&chan->lock);
-			chan = ast_channel_walk_locked(chan);
-		}
-	}
-	return count;
-}
-
 static int group_count_exec(struct ast_channel *chan, void *data)
 {
-	int res=0;
+	int res = 0;
 	int count;
 	struct localuser *u;
-	char *group=NULL;
-	char *cat = NULL;
-	char ret[80]="";
-	char tmp[256]="";
+	char group[80] = "";
+	char category[80] = "";
+	char ret[80] = "";
+	char *grp;
+	static int deprecation_warning = 0;

 	LOCAL_USER_ADD(u);

-	/* Check and parse arguments */
-	if (data && !ast_strlen_zero(data)) {
-		strncpy(tmp, data, sizeof(tmp) - 1);
-		group = tmp;
-		cat = strchr(tmp, '@');
-		if (cat) {
-			*cat = '\0';
-			cat++;
-		}
+	if (!deprecation_warning) {
+	        ast_log(LOG_WARNING, "The GetGroupCount application has been deprecated, please use the GROUP_COUNT function.\n");
+		deprecation_warning = 1;
 	}
-	if (cat)
-		snprintf(ret, sizeof(ret), "GROUP_%s", cat);
-	else
-		strncpy(ret, DEFAULT_CATEGORY, sizeof(ret) - 1);

-	if (!group || ast_strlen_zero(group)) {
-		group = pbx_builtin_getvar_helper(chan, ret);
+	ast_app_group_split_group(data, group, sizeof(group), category, sizeof(category));
+
+	if (ast_strlen_zero(group)) {
+		grp = pbx_builtin_getvar_helper(chan, category);
+		strncpy(group, grp, sizeof(group) - 1);
 	}
-	count = group_get_count(group, ret);
+
+	count = ast_app_group_get_count(group, category);
 	snprintf(ret, sizeof(ret), "%d", count);
 	pbx_builtin_setvar_helper(chan, "GROUPCOUNT", ret);
+
 	LOCAL_USER_REMOVE(u);
+
+	return res;
+}
+
+static int group_match_count_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	int count;
+	struct localuser *u;
+	char group[80] = "";
+	char category[80] = "";
+	char ret[80] = "";
+	static int deprecation_warning = 0;
+
+	LOCAL_USER_ADD(u);
+
+	if (!deprecation_warning) {
+	        ast_log(LOG_WARNING, "The GetGroupMatchCount application has been deprecated, please use the GROUP_MATCH_COUNT function.\n");
+		deprecation_warning = 1;
+	}
+
+	ast_app_group_split_group(data, group, sizeof(group), category, sizeof(category));
+
+	if (!ast_strlen_zero(group)) {
+		count = ast_app_group_match_get_count(group, category);
+		snprintf(ret, sizeof(ret), "%d", count);
+		pbx_builtin_setvar_helper(chan, "GROUPCOUNT", ret);
+	}
+
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }

 static int group_set_exec(struct ast_channel *chan, void *data)
 {
-	int res=0;
+	int res = 0;
 	struct localuser *u;
-	char ret[80] = "";
-	char tmp[256] = "";
-	char *cat=NULL, *group=NULL;
+	static int deprecation_warning = 0;

 	LOCAL_USER_ADD(u);
-
-	/* Check and parse arguments */
-	if (data && !ast_strlen_zero(data)) {
-		strncpy(tmp, data, sizeof(tmp) - 1);
-		group = tmp;
-		cat = strchr(tmp, '@');
-		if (cat) {
-			*cat = '\0';
-			cat++;
-		}
+	
+	if (!deprecation_warning) {
+	        ast_log(LOG_WARNING, "The SetGroup application has been deprecated, please use the GROUP() function.\n");
+		deprecation_warning = 1;
 	}
-	if (cat)
-		snprintf(ret, sizeof(ret), "GROUP_%s", cat);
-	else
-		strncpy(ret, DEFAULT_CATEGORY, sizeof(ret) - 1);

-	if (group && !ast_strlen_zero(group)) {
-		pbx_builtin_setvar_helper(chan, ret, group);
-	} else
+	if (ast_app_group_set_channel(chan, data))
 		ast_log(LOG_WARNING, "SetGroup requires an argument (group name)\n");

 	LOCAL_USER_REMOVE(u);
@@ -149,60 +134,191 @@ static int group_set_exec(struct ast_channel *chan, void *data)

 static int group_check_exec(struct ast_channel *chan, void *data)
 {
-	int res=0;
+	int res = 0;
 	int max, count;
 	struct localuser *u;
-	char ret[80] = "";
-	char tmp[256] = "";
-	char *cat, *group;
+	char limit[80]="";
+	char category[80]="";
+	static int deprecation_warning = 0;
+	char *parse;
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(max);
+		AST_APP_ARG(options);
+	);

 	LOCAL_USER_ADD(u);

-	if (data && !ast_strlen_zero(data)) {
-		strncpy(tmp, data, sizeof(tmp) - 1);
-		group = tmp;
-		cat = strchr(tmp, '@');
-		if (cat) {
-			*cat = '\0';
-			cat++;
-		}
-	 	if ((sscanf((char *)tmp, "%i", &max) == 1) && (max > -1)) {
-			if (cat)
-				snprintf(ret, sizeof(ret), "GROUP_%s", cat);
-			else
-				strncpy(ret, DEFAULT_CATEGORY, sizeof(ret) - 1);
-			
-			count = group_get_count(pbx_builtin_getvar_helper(chan, ret), ret);
-			if (count > max) {
-				if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-					chan->priority += 100;
-				else
+	if (!deprecation_warning) {
+	        ast_log(LOG_WARNING, "The CheckGroup application has been deprecated, please use a combination of the GotoIf application and the GROUP_COUNT() function.\n");
+		deprecation_warning = 1;
+	}
+
+	if (!(parse = ast_strdupa(data))) {
+		ast_log(LOG_WARNING, "Memory Error!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
+	if (ast_strlen_zero(args.max)) {
+		ast_log(LOG_WARNING, "CheckGroup requires an argument(max[@category][|options])\n");
+		return res;
+	}
+
+  	ast_app_group_split_group(args.max, limit, sizeof(limit), category, sizeof(category));
+
+ 	if ((sscanf(limit, "%d", &max) == 1) && (max > -1)) {
+		count = ast_app_group_get_count(pbx_builtin_getvar_helper(chan, category), category);
+		if (count > max) {
+			pbx_builtin_setvar_helper(chan, "CHECKGROUPSTATUS", "OVERMAX");
+			if (priority_jump || option_priority_jumping) {
+				if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
 					res = -1;
 			}
 		} else
-			ast_log(LOG_WARNING, "CheckGroup requires a positive integer argument (max)\n");
+			pbx_builtin_setvar_helper(chan, "CHECKGROUPSTATUS", "OK");
 	} else
-		ast_log(LOG_WARNING, "CheckGroup requires an argument(max)\n");
+		ast_log(LOG_WARNING, "CheckGroup requires a positive integer argument (max)\n");
+
 	LOCAL_USER_REMOVE(u);
 	return res;
 }

+static int group_show_channels(int fd, int argc, char *argv[])
+{
+#define FORMAT_STRING  "%-25s  %-20s  %-20s\n"
+
+	struct ast_channel *c = NULL;
+	int numchans = 0;
+	struct ast_var_t *current;
+	struct varshead *headp;
+	regex_t regexbuf;
+	int havepattern = 0;
+
+	if (argc < 3 || argc > 4)
+		return RESULT_SHOWUSAGE;
+	
+	if (argc == 4) {
+		if (regcomp(&regexbuf, argv[3], REG_EXTENDED | REG_NOSUB))
+			return RESULT_SHOWUSAGE;
+		havepattern = 1;
+	}
+
+	ast_cli(fd, FORMAT_STRING, "Channel", "Group", "Category");
+	while ( (c = ast_channel_walk_locked(c)) != NULL) {
+		headp=&c->varshead;
+		AST_LIST_TRAVERSE(headp,current,entries) {
+			if (!strncmp(ast_var_name(current), GROUP_CATEGORY_PREFIX "_", strlen(GROUP_CATEGORY_PREFIX) + 1)) {
+				if (!havepattern || !regexec(&regexbuf, ast_var_value(current), 0, NULL, 0)) {
+					ast_cli(fd, FORMAT_STRING, c->name, ast_var_value(current),
+						(ast_var_name(current) + strlen(GROUP_CATEGORY_PREFIX) + 1));
+					numchans++;
+				}
+			} else if (!strcmp(ast_var_name(current), GROUP_CATEGORY_PREFIX)) {
+				if (!havepattern || !regexec(&regexbuf, ast_var_value(current), 0, NULL, 0)) {
+					ast_cli(fd, FORMAT_STRING, c->name, ast_var_value(current), "(default)");
+					numchans++;
+				}
+			}
+		}
+		numchans++;
+		ast_mutex_unlock(&c->lock);
+	}
+
+	if (havepattern)
+		regfree(&regexbuf);
+
+	ast_cli(fd, "%d active channel%s\n", numchans, (numchans != 1) ? "s" : "");
+	return RESULT_SUCCESS;
+#undef FORMAT_STRING
+}
+
+static char *tdesc = "Group Management Routines";
+
+static char *app_group_count = "GetGroupCount";
+static char *app_group_set = "SetGroup";
+static char *app_group_check = "CheckGroup";
+static char *app_group_match_count = "GetGroupMatchCount";
+
+static char *group_count_synopsis = "Get the channel count of a group";
+static char *group_set_synopsis = "Set the channel's group";
+static char *group_check_synopsis = "Check the channel count of a group against a limit";
+static char *group_match_count_synopsis = "Get the channel count of all groups that match a pattern";
+
+static char *group_count_descrip =
+"Usage: GetGroupCount([groupname][@category])\n"
+"  Calculates the group count for the specified group, or uses\n"
+"the current channel's group if not specifed (and non-empty).\n"
+"Stores result in GROUPCOUNT. \n"
+"This application has been deprecated, please use the function\n"
+"GroupCount.\n";
+
+static char *group_set_descrip =
+"Usage: SetGroup(groupname[@category])\n"
+"  Sets the channel group to the specified value.  Equivalent to\n"
+"Set(GROUP=group).  Always returns 0.\n";
+
+static char *group_check_descrip =
+"Usage: CheckGroup(max[@category][|options])\n"
+"  Checks that the current number of total channels in the\n"
+"current channel's group does not exceed 'max'.  If the number\n"
+"does not exceed 'max', we continue to the next step. \n"
+" The option string may contain zero of the following character:\n"
+"	'j' -- jump to n+101 priority if the number does in fact exceed max,\n"
+"              and priority n+101 exists. Execuation then continues at that\n"
+"	       step, otherwise -1 is returned.\n"
+" This application sets the following channel variable upon successful completion:\n"
+"	CHECKGROUPSTATUS  The status of the check that the current channel's\n"
+"			  group does not exceed 'max'. It's value is one of\n"
+"		OK | OVERMAX \n";	
+
+static char *group_match_count_descrip =
+"Usage: GetGroupMatchCount(groupmatch[@category])\n"
+"  Calculates the group count for all groups that match the specified\n"
+"pattern. Uses standard regular expression matching (see regex(7)).\n"
+"Stores result in GROUPCOUNT.  Always returns 0.\n"
+"This application has been deprecated, please use the function\n"
+"GroupMatchCount.\n";
+
+static char show_channels_usage[] = 
+"Usage: group show channels [pattern]\n"
+"       Lists all currently active channels with channel group(s) specified.\n       Optional regular expression pattern is matched to group names for each channel.\n";
+
+static struct ast_cli_entry  cli_show_channels =
+	{ { "group", "show", "channels", NULL }, group_show_channels, "Show active channels with group(s)", show_channels_usage};
+
 int unload_module(void)
 {
 	int res;
-	STANDARD_HANGUP_LOCALUSERS;
-	res = ast_unregister_application(app_group_count);
+
+	res = ast_cli_unregister(&cli_show_channels);
+	res |= ast_unregister_application(app_group_count);
 	res |= ast_unregister_application(app_group_set);
 	res |= ast_unregister_application(app_group_check);
+	res |= ast_unregister_application(app_group_match_count);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
 	return res;
 }

 int load_module(void)
 {
 	int res;
+
 	res = ast_register_application(app_group_count, group_count_exec, group_count_synopsis, group_count_descrip);
 	res |= ast_register_application(app_group_set, group_set_exec, group_set_synopsis, group_set_descrip);
 	res |= ast_register_application(app_group_check, group_check_exec, group_check_synopsis, group_check_descrip);
+	res |= ast_register_application(app_group_match_count, group_match_count_exec, group_match_count_synopsis, group_match_count_descrip);
+	res |= ast_cli_register(&cli_show_channels);
+	
 	return res;
 }

--- a/apps/app_hasnewvoicemail.c
+++ b/apps/app_hasnewvoicemail.c
@@ -1,8 +1,7 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * HasVoicemail application
- * Changes Copyright (c) 2004 Todd Freeman <freeman@andrews.edu>
+ * Changes Copyright (c) 2004 - 2005 Todd Freeman <freeman@andrews.edu>
 * 
 * 95% based on HasNewVoicemail by:
 * 
@@ -10,144 +9,241 @@
 * 
 * Tilghman Lesher <asterisk-hasnewvoicemail-app@the-tilghman.com>
 *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions
- * are met:
- * 1. Redistributions of source code must retain the above copyright
- *    notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- *    notice, this list of conditions and the following disclaimer in the
- *    documentation and/or other materials provided with the distribution.
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
- * THIS SOFTWARE IS PROVIDED BY THE CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR
- * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO
- * EVENT SHALL THE CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
 */

-#include <sys/types.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/lock.h>
-#include <asterisk/utils.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <string.h>
-#include <stdlib.h>
-#include <dirent.h>
+/*! \file
+ *
+ * \brief HasVoicemail application
+ *
+ * \ingroup applications
+ */

-#include "../astconf.h"
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <dirent.h>
+#include <sys/types.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/utils.h"
+#include "asterisk/app.h"
+#include "asterisk/options.h"

 static char *tdesc = "Indicator for whether a voice mailbox has messages in a given folder.";
 static char *app_hasvoicemail = "HasVoicemail";
-static char *hasvoicemail_synopsis = "Conditionally branches to priority + 101";
+static char *hasvoicemail_synopsis = "Conditionally branches to priority + 101 with the right options set";
 static char *hasvoicemail_descrip =
-"HasVoicemail(vmbox[@context][:folder][|varname])\n"
-"  Branches to priority + 101, if there is voicemail in folder indicated."
+"HasVoicemail(vmbox[/folder][@context][|varname[|options]])\n"
 "  Optionally sets <varname> to the number of messages in that folder."
-"  Assumes folder of INBOX if not specified.\n";
+"  Assumes folder of INBOX if not specified.\n"
+"  The option string may contain zero or the following character:\n"
+"	'j' -- jump to priority n+101, if there is voicemail in the folder indicated.\n"
+"  This application sets the following channel variable upon completion:\n"
+"	HASVMSTATUS		The result of the voicemail check returned as a text string as follows\n"
+"		<# of messages in the folder, 0 for NONE>\n";

 static char *app_hasnewvoicemail = "HasNewVoicemail";
-static char *hasnewvoicemail_synopsis = "Conditionally branches to priority + 101";
+static char *hasnewvoicemail_synopsis = "Conditionally branches to priority + 101 with the right options set";
 static char *hasnewvoicemail_descrip =
-"HasNewVoicemail(vmbox[@context][|varname])\n"
-"  Branches to priority + 101, if there is voicemail in folder INBOX."
-"  Optionally sets <varname> to the number of messages in that folder.\n";
+"HasNewVoicemail(vmbox[/folder][@context][|varname[|options]])\n"
+"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"
+"  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";

 STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

-static int hasvoicemail_exec(struct ast_channel *chan, void *data)
+static int hasvoicemail_internal(char *context, char *box, char *folder)
 {
-	int res=0;
-	struct localuser *u;
-	char vmpath[256], *temps, *input, *varname = NULL, *vmbox, *vmfolder = "INBOX", *context = "default";
+	char vmpath[256];
 	DIR *vmdir;
 	struct dirent *vment;
-	int vmcount = 0;
+	int count=0;

+	snprintf(vmpath,sizeof(vmpath), "%s/voicemail/%s/%s/%s", (char *)ast_config_AST_SPOOL_DIR, context, box, folder);
+	if ((vmdir = opendir(vmpath))) {
+		/* No matter what the format of VM, there will always be a .txt file for each message. */
+		while ((vment = readdir(vmdir))) {
+			if (!strncmp(vment->d_name + 7, ".txt", 4)) {
+				count++;
+				break;
+			}
+		}
+		closedir(vmdir);
+	}
+	return count;
+}
+
+static int hasvoicemail_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	char *input, *varname = NULL, *vmbox, *context = "default";
+	char *vmfolder;
+	int vmcount = 0;
+	static int dep_warning = 0;
+	int priority_jump = 0;
+	char tmp[12];
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(vmbox);
+		AST_APP_ARG(varname);
+		AST_APP_ARG(options);
+	);
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The applications HasVoicemail and HasNewVoicemail have been deprecated.  Please use the VMCOUNT() function instead.\n");
+		dep_warning = 1;
+	}
+	
 	if (!data) {
-		ast_log(LOG_WARNING, "HasVoicemail requires an argument (vm-box[@context][:folder]|varname)\n");
+		ast_log(LOG_WARNING, "HasVoicemail requires an argument (vm-box[/folder][@context][|varname[|options]])\n");
 		return -1;
 	}
+
 	LOCAL_USER_ADD(u);

 	input = ast_strdupa((char *)data);
-	if (input) {
-		temps = input;
-		if ((temps = strsep(&input, "|"))) {
-			if (input && !ast_strlen_zero(input))
-				varname = input;
-			input = temps;
-		}
-		if ((temps = strsep(&input, ":"))) {
-			if (input && !ast_strlen_zero(input))
-				vmfolder = input;
-			input = temps;
-		}
-		if ((vmbox = strsep(&input, "@")))
-			if (input && !ast_strlen_zero(input))
-				context = input;
-		if (!vmbox)
-			vmbox = input;
-
-		snprintf(vmpath,sizeof(vmpath), "%s/voicemail/%s/%s/%s", (char *)ast_config_AST_SPOOL_DIR, context, vmbox, vmfolder);
-		if (!(vmdir = opendir(vmpath))) {
-			ast_log(LOG_NOTICE, "Voice mailbox %s at %s does not exist\n", vmbox, vmpath);
-		} else {
-
-			/* No matter what the format of VM, there will always be a .txt file for each message. */
-			while ((vment = readdir(vmdir)))
-				if (!strncmp(vment->d_name + 7,".txt",4))
-					vmcount++;
-			closedir(vmdir);
-		}
-		/* Set the count in the channel variable */
-		if (varname) {
-			char tmp[12];
-			snprintf(tmp, sizeof(tmp), "%d", vmcount);
-			pbx_builtin_setvar_helper(chan, varname, tmp);
-		}
-
-		if (vmcount > 0) {
-			/* Branch to the next extension */
-			if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid)) {
-				chan->priority += 100;
-			} else
-				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);
-		}
-	} else {
+	if (! input) {
 		ast_log(LOG_ERROR, "Out of memory error\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
 	}

+	AST_STANDARD_APP_ARGS(args, input);
+
+	if ((vmbox = strsep(&args.vmbox, "@")))
+		if (!ast_strlen_zero(args.vmbox))
+			context = args.vmbox;
+	if (!vmbox)
+		vmbox = args.vmbox;
+
+	vmfolder = strchr(vmbox, '/');
+	if (vmfolder) {
+		*vmfolder = '\0';
+		vmfolder++;
+	} else {
+		vmfolder = "INBOX";
+	}
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
+	vmcount = hasvoicemail_internal(context, vmbox, vmfolder);
+	/* Set the count in the channel variable */
+	if (varname) {
+		snprintf(tmp, sizeof(tmp), "%d", vmcount);
+		pbx_builtin_setvar_helper(chan, varname, tmp);
+	}
+
+	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)) 
+				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);
+		}
+	}
+
+	snprintf(tmp, sizeof(tmp), "%d", vmcount);
+	pbx_builtin_setvar_helper(chan, "HASVMSTATUS", tmp);
+	
 	LOCAL_USER_REMOVE(u);
-	return res;
+
+	return 0;
 }

+static char *acf_vmcount_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
+	struct localuser *u;
+	char *args, *context, *box, *folder;
+
+	LOCAL_USER_ACF_ADD(u);
+
+	buf[0] = '\0';
+
+	args = ast_strdupa(data);
+	if (!args) {
+		ast_log(LOG_ERROR, "Out of memory");
+		LOCAL_USER_REMOVE(u);
+		return buf;
+	}
+
+	box = strsep(&args, "|");
+	if (strchr(box, '@')) {
+		context = box;
+		box = strsep(&context, "@");
+	} else {
+		context = "default";
+	}
+
+	if (args) {
+		folder = args;
+	} else {
+		folder = "INBOX";
+	}
+
+	snprintf(buf, len, "%d", hasvoicemail_internal(context, box, folder));
+
+	LOCAL_USER_REMOVE(u);
+	
+	return buf;
+}
+
+struct ast_custom_function acf_vmcount = {
+	.name = "VMCOUNT",
+	.synopsis = "Counts the voicemail in a specified mailbox",
+	.syntax = "VMCOUNT(vmbox[@context][|folder])",
+	.desc =
+	"  context - defaults to \"default\"\n"
+	"  folder  - defaults to \"INBOX\"\n",
+	.read = acf_vmcount_exec,
+};
+
 int unload_module(void)
 {
 	int res;
-	STANDARD_HANGUP_LOCALUSERS;
-	res = ast_unregister_application(app_hasvoicemail);
+	
+	res = ast_custom_function_unregister(&acf_vmcount);
+	res |= ast_unregister_application(app_hasvoicemail);
 	res |= ast_unregister_application(app_hasnewvoicemail);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
 	return res;
 }

 int load_module(void)
 {
 	int res;
-	res = ast_register_application(app_hasvoicemail, hasvoicemail_exec, hasvoicemail_synopsis, hasvoicemail_descrip);
+
+	res = ast_custom_function_register(&acf_vmcount);
+	res |= ast_register_application(app_hasvoicemail, hasvoicemail_exec, hasvoicemail_synopsis, hasvoicemail_descrip);
 	res |= ast_register_application(app_hasnewvoicemail, hasvoicemail_exec, hasnewvoicemail_synopsis, hasnewvoicemail_descrip);
+
 	return res;
 }

--- a/apps/app_ices.c
+++ b/apps/app_ices.c
@@ -1,24 +1,28 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Stream to an icecast server via ICES (see contrib/asterisk-ices.xml)
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Stream to an icecast server via ICES (see contrib/asterisk-ices.xml)
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/frame.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
 #include <string.h>
 #include <stdio.h>
 #include <signal.h>
@@ -27,7 +31,19 @@
 #include <fcntl.h>
 #include <sys/time.h>
 #include <errno.h>
-#include "../astconf.h"
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/frame.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"

 #define ICES "/usr/bin/ices"
 #define LOCAL_ICES "/usr/local/bin/ices"
@@ -41,8 +57,7 @@ static char *synopsis = "Encode and stream using 'ices'";
 static char *descrip = 
 "  ICES(config.xml) Streams to an icecast server using ices\n"
 "(available separately).  A configuration file must be supplied\n"
-"for ices (see examples/asterisk-ices.conf).  Returns  -1  on\n"
-"hangup or 0 otherwise.\n";
+"for ices (see examples/asterisk-ices.conf). \n";

 STANDARD_LOCAL_USER;

@@ -85,20 +100,24 @@ static int ices_exec(struct ast_channel *chan, void *data)
 	struct ast_frame *f;
 	char filename[256]="";
 	char *c;
-	last.tv_usec = 0;
-	last.tv_sec = 0;
-	if (!data || !strlen(data)) {
+
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "ICES requires an argument (configfile.xml)\n");
 		return -1;
 	}
+
+	LOCAL_USER_ADD(u);
+	
+	last = ast_tv(0, 0);
+	
 	if (pipe(fds)) {
 		ast_log(LOG_WARNING, "Unable to create pipe\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	flags = fcntl(fds[1], F_GETFL);
 	fcntl(fds[1], F_SETFL, flags | O_NONBLOCK);
 	
-	LOCAL_USER_ADD(u);
 	ast_stopstream(chan);

 	if (chan->_state != AST_STATE_UP)
@@ -108,6 +127,7 @@ static int ices_exec(struct ast_channel *chan, void *data)
 		close(fds[0]);
 		close(fds[1]);
 		ast_log(LOG_WARNING, "Answer failed!\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}

@@ -117,6 +137,7 @@ static int ices_exec(struct ast_channel *chan, void *data)
 		close(fds[0]);
 		close(fds[1]);
 		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	if (((char *)data)[0] == '/')
@@ -159,18 +180,26 @@ static int ices_exec(struct ast_channel *chan, void *data)
 		}
 	}
 	close(fds[1]);
-	LOCAL_USER_REMOVE(u);
+	
 	if (pid > -1)
 		kill(pid, SIGKILL);
 	if (!res && oreadformat)
 		ast_set_read_format(chan, oreadformat);
+
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_image.c
+++ b/apps/app_image.c
@@ -1,26 +1,46 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * App to transmit an image
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to transmit an image
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/image.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/app.h"
+#include "asterisk/options.h"

 static char *tdesc = "Image Transmission Application";

@@ -29,12 +49,15 @@ static char *app = "SendImage";
 static char *synopsis = "Send an image file";

 static char *descrip = 
-"  SendImage(filename): Sends an image on a channel. If the channel\n"
-"does not support  image transport, and there exists  a  step  with\n"
-"priority n + 101, then  execution  will  continue  at  that  step.\n"
-"Otherwise,  execution  will continue at  the  next priority level.\n"
-"SendImage only  returns  0 if  the  image was sent correctly or if\n"
-"the channel does not support image transport, and -1 otherwise.\n";
+"  SendImage(filename): Sends an image on a channel. \n"
+"If the channel supports image transport but the image send\n"
+"fails, the channel will be hung up. Otherwise, the dialplan\n"
+"continues execution.\n"
+"The option string may contain the following character:\n"
+"	'j' -- jump to priority n+101 if the channel doesn't support image transport\n"
+"This application sets the following channel variable upon completion:\n"
+"	SENDIMAGESTATUS		The status is the result of the attempt as a text string, one of\n"
+"		OK | NOSUPPORT \n";			

 STANDARD_LOCAL_USER;

@@ -44,26 +67,61 @@ static int sendimage_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
 	struct localuser *u;
-	if (!data || !strlen((char *)data)) {
-		ast_log(LOG_WARNING, "SendImage requires an argument (filename)\n");
+	char *parse;
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(filename);
+		AST_APP_ARG(options);
+	);
+	
+	LOCAL_USER_ADD(u);
+
+	if (!(parse = ast_strdupa(data))) {
+		ast_log(LOG_WARNING, "Memory Error!\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
-	LOCAL_USER_ADD(u);
+
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	if (ast_strlen_zero(args.filename)) {
+		ast_log(LOG_WARNING, "SendImage requires an argument (filename[|options])\n");
+		return -1;
+	}
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
 	if (!ast_supports_images(chan)) {
 		/* Does not support transport */
-		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-			chan->priority += 100;
+		if (priority_jump || option_priority_jumping)
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		pbx_builtin_setvar_helper(chan, "SENDIMAGESTATUS", "NOSUPPORT");
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
-	res = ast_send_image(chan, data);
+
+	res = ast_send_image(chan, args.filename);
+	
+	if (!res)
+		pbx_builtin_setvar_helper(chan, "SENDIMAGESTATUS", "OK");
+	
 	LOCAL_USER_REMOVE(u);
+	
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res; 
 }

 int load_module(void)
--- a/apps/app_intercom.c
+++ b/apps/app_intercom.c
@@ -1,24 +1,29 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Use /dev/dsp as an intercom.
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Use /dev/dsp as an intercom.
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/frame.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
+#include <stdio.h>
 #include <unistd.h>
 #include <errno.h>
 #include <sys/ioctl.h>
@@ -35,6 +40,19 @@
 #include <soundcard.h>
 #endif

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/frame.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+
 #ifdef __OpenBSD__
 #define DEV_DSP "/dev/audio"
 #else
@@ -51,8 +69,7 @@ static char *app = "Intercom";
 static char *synopsis = "(Obsolete) Send to Intercom";
 static char *descrip = 
 "  Intercom(): Sends the user to the intercom (i.e. /dev/dsp).  This program\n"
-"is generally considered  obselete by the chan_oss module.  Returns 0 if the\n"
-"user exits with a DTMF tone, or -1 if they hangup.\n";
+"is generally considered  obselete by the chan_oss module.  User can terminate\n"with a DTMF tone, or by hangup.\n";

 STANDARD_LOCAL_USER;

@@ -141,6 +158,7 @@ static int intercom_exec(struct ast_channel *chan, void *data)
 	res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
 	if (res < 0) {
 		ast_log(LOG_WARNING, "Unable to set format to signed linear on channel %s\n", chan->name);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	/* Read packets from the channel */
@@ -168,18 +186,27 @@ static int intercom_exec(struct ast_channel *chan, void *data)
 				res = -1;
 		}
 	}
-	LOCAL_USER_REMOVE(u);
+	
 	if (!res)
 		ast_set_read_format(chan, oreadformat);
+
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }

 int unload_module(void)
 {
-	STANDARD_HANGUP_LOCALUSERS;
+	int res;
+
 	if (sound > -1)
 		close(sound);
-	return ast_unregister_application(app);
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_ivrdemo.c
+++ b/apps/app_ivrdemo.c
@@ -0,0 +1,144 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief IVR Demo application
+ * 
+ * \ingroup applications
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/app.h"
+
+static char *tdesc = "IVR Demo Application";
+static char *app = "IVRDemo";
+static char *synopsis = 
+"  This is a skeleton application that shows you the basic structure to create your\n"
+"own asterisk applications and demonstrates the IVR demo.\n";
+
+static int ivr_demo_func(struct ast_channel *chan, void *data)
+{
+	ast_verbose("IVR Demo, data is %s!\n", (char *)data);
+	return 0;
+}
+
+AST_IVR_DECLARE_MENU(ivr_submenu, "IVR Demo Sub Menu", 0, 
+{
+	{ "s", AST_ACTION_BACKGROUND, "demo-abouttotry" },
+	{ "s", AST_ACTION_WAITOPTION },
+	{ "1", AST_ACTION_PLAYBACK, "digits/1" },
+	{ "1", AST_ACTION_PLAYBACK, "digits/1" },
+	{ "1", AST_ACTION_RESTART },
+	{ "2", AST_ACTION_PLAYLIST, "digits/2;digits/3" },
+	{ "3", AST_ACTION_CALLBACK, ivr_demo_func },
+	{ "4", AST_ACTION_TRANSFER, "demo|s|1" },
+	{ "*", AST_ACTION_REPEAT },
+	{ "#", AST_ACTION_UPONE  },
+	{ NULL }
+});
+
+AST_IVR_DECLARE_MENU(ivr_demo, "IVR Demo Main Menu", 0, 
+{
+	{ "s", AST_ACTION_BACKGROUND, "demo-congrats" },
+	{ "g", AST_ACTION_BACKGROUND, "demo-instruct" },
+	{ "g", AST_ACTION_WAITOPTION },
+	{ "1", AST_ACTION_PLAYBACK, "digits/1" },
+	{ "1", AST_ACTION_RESTART },
+	{ "2", AST_ACTION_MENU, &ivr_submenu },
+	{ "2", AST_ACTION_RESTART },
+	{ "i", AST_ACTION_PLAYBACK, "invalid" },
+	{ "i", AST_ACTION_REPEAT, (void *)(unsigned long)2 },
+	{ "#", AST_ACTION_EXIT },
+	{ NULL },
+});
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int skel_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "skel requires an argument (filename)\n");
+		return -1;
+	}
+	
+	LOCAL_USER_ADD(u);
+
+	/* Do our thing here */
+
+	if (chan->_state != AST_STATE_UP)
+		res = ast_answer(chan);
+	if (!res)
+		res = ast_ivr_menu_run(chan, &ivr_demo, data);
+	
+	LOCAL_USER_REMOVE(u);
+
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+	
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+	
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, skel_exec, tdesc, synopsis);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_lookupblacklist.c
+++ b/apps/app_lookupblacklist.c
@@ -1,29 +1,49 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * App to lookup the callerid number, and see if it is blacklisted
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to lookup the callerid number, and see if it is blacklisted
+ *
+ * \ingroup applications
+ * 
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/options.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/image.h>
-#include <asterisk/callerid.h>
-#include <asterisk/astdb.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/callerid.h"
+#include "asterisk/astdb.h"
+#include "asterisk/options.h"

 static char *tdesc = "Look up Caller*ID name/number from blacklist database";

@@ -32,14 +52,14 @@ static char *app = "LookupBlacklist";
 static char *synopsis = "Look up Caller*ID name/number from blacklist database";

 static char *descrip =
-  "  LookupBlacklist: Looks up the Caller*ID number on the active\n"
-  "channel in the Asterisk database (family 'blacklist').  If the\n"
-  "number is found, and if there exists a priority n + 101,\n"
-  "where 'n' is the priority of the current instance, then  the\n"
-  "channel  will  be  setup  to continue at that priority level.\n"
-  "Otherwise, it returns 0.  Does nothing if no Caller*ID was received on the\n"
-  "channel.\n"
-  "Example: database put blacklist <name/number> 1\n";
+  "  LookupBlacklist(options): Looks up the Caller*ID number on the active\n"
+  "channel in the Asterisk database (family 'blacklist').  \n"
+  "The option string may contain the following character:\n"
+  "	'j' -- jump to n+101 priority if the number/name is found in the blacklist\n"
+  "This application sets the following channel variable upon completion:\n"
+  "	LOOKUPBLSTATUS		The status of the Blacklist lookup as a text string, one of\n"
+  "		FOUND | NOTFOUND\n"
+  "Example: exten => 1234,1,LookupBlacklist()\n";

 STANDARD_LOCAL_USER;

@@ -48,47 +68,54 @@ LOCAL_USER_DECL;
 static int
 lookupblacklist_exec (struct ast_channel *chan, void *data)
 {
-	char old_cid[144] = "", *num, *name;
 	char blacklist[1];
-	char shrunknum[64] = "";
 	struct localuser *u;
 	int bl = 0;
+	int priority_jump = 0;

-	LOCAL_USER_ADD (u);
-	if (chan->callerid)
-	{
-		strncpy (old_cid, chan->callerid, sizeof (old_cid) - 1);
-		ast_callerid_parse (old_cid, &name, &num);
-		if (num)
-			strncpy (shrunknum, num, sizeof (shrunknum) - 1);
-		else
-			num = shrunknum;
-		
-		ast_shrink_phone_number (shrunknum);
-		if (!ast_db_get ("blacklist", shrunknum, blacklist, sizeof (blacklist)))
-		{
+	LOCAL_USER_ADD(u);
+
+	if (!ast_strlen_zero(data)) {
+		if (strchr(data, 'j'))
+			priority_jump = 1;
+	}
+
+	if (chan->cid.cid_num) {
+		if (!ast_db_get("blacklist", chan->cid.cid_num, blacklist, sizeof (blacklist))) {
 			if (option_verbose > 2)
-				ast_log(LOG_NOTICE, "Blacklisted number %s found\n",shrunknum);
-			bl = 1;
-		}
-		else if (!ast_db_get ("blacklist", name, blacklist, sizeof (blacklist)))
-		{
-			if (option_verbose > 2)
-				ast_log (LOG_NOTICE,"Blacklisted name \"%s\" found\n",name);
+				ast_log(LOG_NOTICE, "Blacklisted number %s found\n",chan->cid.cid_num);
 			bl = 1;
 		}
 	}
-	
-	if (bl && ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-		chan->priority+=100;
-	LOCAL_USER_REMOVE (u);
+	if (chan->cid.cid_name) {
+		if (!ast_db_get("blacklist", chan->cid.cid_name, blacklist, sizeof (blacklist))) {
+			if (option_verbose > 2)
+				ast_log (LOG_NOTICE,"Blacklisted name \"%s\" found\n",chan->cid.cid_name);
+			bl = 1;
+		}
+	}
+
+	if (bl) {
+		if (priority_jump || option_priority_jumping) 
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		pbx_builtin_setvar_helper(chan, "LOOKUPBLSTATUS", "FOUND");
+	} else
+		pbx_builtin_setvar_helper(chan, "LOOKUPBLSTATUS", "NOTFOUND");	
+
+	LOCAL_USER_REMOVE(u);
+
 	return 0;
 }

 int unload_module (void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application (app);
+
+	return res;	
 }

 int load_module (void)
--- a/apps/app_lookupcidname.c
+++ b/apps/app_lookupcidname.c
@@ -1,29 +1,47 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * App to set callerid name from database, based on directory number
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to set callerid name from database, based on directory number
+ * 
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/options.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/image.h>
-#include <asterisk/callerid.h>
-#include <asterisk/astdb.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/callerid.h"
+#include "asterisk/astdb.h"

 static char *tdesc = "Look up CallerID Name from local database";

@@ -37,7 +55,7 @@ static char *descrip =
  "Caller*ID name.  Does nothing if no Caller*ID was received on the\n"
  "channel.  This is useful if you do not subscribe to Caller*ID\n"
  "name delivery, or if you want to change the names on some incoming\n"
-  "calls.  Always returns 0.\n";
+  "calls.\n";

 STANDARD_LOCAL_USER;

@@ -46,32 +64,18 @@ LOCAL_USER_DECL;
 static int
 lookupcidname_exec (struct ast_channel *chan, void *data)
 {
-  char old_cid[144] = "", *num, *name;
-  char new_cid[144];
  char dbname[64];
-  char shrunknum[64] = "";
  struct localuser *u;

  LOCAL_USER_ADD (u);
-  if (chan->callerid)
-    {
-      strncpy (old_cid, chan->callerid, sizeof (old_cid) - 1);
-      ast_callerid_parse (old_cid, &name, &num);	/* this destroys the original string */
-      if (num)			/* It's possible to get an empty number */
-	strncpy (shrunknum, num, sizeof (shrunknum) - 1);
-      else
-	num = shrunknum;
-      ast_shrink_phone_number (shrunknum);
-      if (!ast_db_get ("cidname", shrunknum, dbname, sizeof (dbname)))
-	{
-	  snprintf (new_cid, sizeof (new_cid), "\"%s\" <%s>", dbname, num);
-	  ast_set_callerid (chan, new_cid, 0);
-	  if (option_verbose > 2)
-	    ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID to %s\n",
-			 new_cid);
+  if (chan->cid.cid_num) {
+	if (!ast_db_get ("cidname", chan->cid.cid_num, dbname, sizeof (dbname))) {
+		ast_set_callerid (chan, NULL, dbname, NULL);
+		  if (option_verbose > 2)
+		    ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID name to %s\n",
+				 dbname);
 	}
-
-    }
+  }
  LOCAL_USER_REMOVE (u);
  return 0;
 }
@@ -79,8 +83,13 @@ lookupcidname_exec (struct ast_channel *chan, void *data)
 int
 unload_module (void)
 {
-  STANDARD_HANGUP_LOCALUSERS;
-  return ast_unregister_application (app);
+	int res;
+
+	res = ast_unregister_application (app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }

 int
--- a/apps/app_macro.c
+++ b/apps/app_macro.c
@@ -1,50 +1,87 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Macro Implementation
- * 
- * Copyright (C) 2003, Digium
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Dial plan macro Implementation
+ * 
+ * \ingroup applications
 */

-#include <sys/types.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/options.h>
-#include <asterisk/utils.h>
-#include <asterisk/lock.h>
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
+#include <unistd.h>
+#include <sys/types.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/options.h"
+#include "asterisk/config.h"
+#include "asterisk/utils.h"
+#include "asterisk/lock.h"

 #define MAX_ARGS 80

+/* special result value used to force macro exit */
+#define MACRO_EXIT_RESULT 1024
+
 static char *tdesc = "Extension Macros";

 static char *descrip =
 "  Macro(macroname|arg1|arg2...): Executes a macro using the context\n"
 "'macro-<macroname>', jumping to the 's' extension of that context and\n"
-"executing each step, then returning when the steps end.  The calling\n"
-"extension, context, and priority are stored in ${MACRO_EXTEN}, \n"
+"executing each step, then returning when the steps end. \n"
+"The calling extension, context, and priority are stored in ${MACRO_EXTEN}, \n"
 "${MACRO_CONTEXT} and ${MACRO_PRIORITY} respectively.  Arguments become\n"
-"${ARG1}, ${ARG2}, etc in the macro context.  Macro returns -1 if\n"
-"any step in the macro returns -1, and 0 otherwise.  If you Goto out\n"
-"of the Macro context, the Macro will terminate and control will be return\n"
-"at the location of the Goto.  Otherwise if ${MACRO_OFFSET} is set at\n"
-"termination, Macro will attempt to continue at priority\n"
-"MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.\n";
+"${ARG1}, ${ARG2}, etc in the macro context.\n"
+"If you Goto out of the Macro context, the Macro will terminate and control\n"
+"will be returned at the location of the Goto.\n"
+"If ${MACRO_OFFSET} is set at termination, Macro will attempt to continue\n"
+"at priority MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.\n";
+
+static char *if_descrip =
+"  MacroIf(<expr>?macroname_a[|arg1][:macroname_b[|arg1]])\n"
+"Executes macro defined in <macroname_a> if <expr> is true\n"
+"(otherwise <macroname_b> if provided)\n"
+"Arguments and return values as in application macro()\n";
+
+static char *exit_descrip =
+"  MacroExit():\n"
+"Causes the currently running macro to exit as if it had\n"
+"ended normally by running out of priorities to execute.\n"
+"If used outside a macro, will likely cause unexpected\n"
+"behavior.\n";

 static char *app = "Macro";
+static char *if_app = "MacroIf";
+static char *exit_app = "MacroExit";

 static char *synopsis = "Macro Implementation";
+static char *if_synopsis = "Conditional Macro Implementation";
+static char *exit_synopsis = "Exit From Macro";

 STANDARD_LOCAL_USER;

@@ -52,187 +89,284 @@ LOCAL_USER_DECL;

 static int macro_exec(struct ast_channel *chan, void *data)
 {
-  char tmp[256] = "";
-  char *cur, *rest;
-  char *macro;
-  char fullmacro[80];
-  char varname[80];
-  char *oldargs[MAX_ARGS + 1] = { NULL, };
-  int argc, x;
-  int res=0;
-  char oldexten[256]="";
-  int oldpriority;
-  char pc[80];
-  char oldcontext[256] = "";
-  char *offsets;
-  int offset;
-  int setmacrocontext=0;
+	char *tmp;
+	char *cur, *rest;
+	char *macro;
+	char fullmacro[80];
+	char varname[80];
+	char *oldargs[MAX_ARGS + 1] = { NULL, };
+	int argc, x;
+	int res=0;
+	char oldexten[256]="";
+	int oldpriority;
+	char pc[80], depthc[12];
+	char oldcontext[AST_MAX_CONTEXT] = "";
+	char *offsets;
+	int offset, depth;
+	int setmacrocontext=0;
+	int autoloopflag;
  
-  char *save_macro_exten;
-  char *save_macro_context;
-  char *save_macro_priority;
-  char *save_macro_offset;
-  
-  if (!data || ast_strlen_zero(data)) {
-    ast_log(LOG_WARNING, "Invalid Macro incantation\n");
-    return 0;
-  }
-  strncpy(tmp, data, sizeof(tmp) - 1);
-  rest = tmp;
-  macro = strsep(&rest, "|");
-  if (!macro || ast_strlen_zero(macro)) {
-  	ast_log(LOG_WARNING, "Invalid macro name specified\n");
-	return 0;
-  }
-  snprintf(fullmacro, sizeof(fullmacro), "macro-%s", macro);
-  if (!ast_exists_extension(chan, fullmacro, "s", 1, chan->callerid)) {
-  	if (!ast_context_find(fullmacro)) 
-		ast_log(LOG_WARNING, "No such context '%s' for macro '%s'\n", fullmacro, macro);
-	else
-	  	ast_log(LOG_WARNING, "Context '%s' for macro '%s' lacks 's' extension, priority 1\n", fullmacro, macro);
-	return 0;
-  }
-  /* Save old info */
-  oldpriority = chan->priority;
-  strncpy(oldexten, chan->exten, sizeof(oldexten) - 1);
-  strncpy(oldcontext, chan->context, sizeof(oldcontext) - 1);
-  if (ast_strlen_zero(chan->macrocontext)) {
-	strncpy(chan->macrocontext, chan->context, sizeof(chan->macrocontext) - 1);
-	strncpy(chan->macroexten, chan->exten, sizeof(chan->macroexten) - 1);
-	chan->macropriority = chan->priority;
-	setmacrocontext=1;
-  }
-  argc = 1;
-  /* Save old macro variables */
-  save_macro_exten = pbx_builtin_getvar_helper(chan, "MACRO_EXTEN");
-  if (save_macro_exten) save_macro_exten = strdup(save_macro_exten);
-  pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", oldexten);
-
-  save_macro_context = pbx_builtin_getvar_helper(chan, "MACRO_CONTEXT");
-  if (save_macro_context) save_macro_context = strdup(save_macro_context);
-  pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", oldcontext);
-
-  save_macro_priority = pbx_builtin_getvar_helper(chan, "MACRO_PRIORITY");
-  if (save_macro_priority) save_macro_priority = strdup(save_macro_priority);
-  snprintf(pc, sizeof(pc), "%d", oldpriority);
-  pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", pc);
-  
-  save_macro_offset = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET");
-  if (save_macro_offset) save_macro_offset = strdup(save_macro_offset);
-  pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", NULL);
-
-  /* Setup environment for new run */
-  chan->exten[0] = 's';
-  chan->exten[1] = '\0';
-  strncpy(chan->context, fullmacro, sizeof(chan->context) - 1);
-  chan->priority = 1;
-
-  while((cur = strsep(&rest, "|")) && (argc < MAX_ARGS)) {
-  	/* Save copy of old arguments if we're overwriting some, otherwise
-	   let them pass through to the other macro */
-  	snprintf(varname, sizeof(varname), "ARG%d", argc);
-	oldargs[argc] = pbx_builtin_getvar_helper(chan, varname);
-	if (oldargs[argc])
-		oldargs[argc] = strdup(oldargs[argc]);
-	pbx_builtin_setvar_helper(chan, varname, cur);
-	argc++;
-  }
-  while(ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->callerid)) {
-	if ((res = ast_spawn_extension(chan, chan->context, chan->exten, chan->priority, chan->callerid))) {
-		/* Something bad happened, or a hangup has been requested. */
-		if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F')) ||
-		    (res == '*') || (res == '#')) {
-			/* Just return result as to the previous application as if it had been dialed */
-			ast_log(LOG_DEBUG, "Oooh, got something to jump out with ('%c')!\n", res);
-			break;
-		}
-		switch(res) {
-		case AST_PBX_KEEPALIVE:
-			if (option_debug)
-				ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited KEEPALIVE in macro %s on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
-			else if (option_verbose > 1)
-				ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited KEEPALIVE in macro '%s' on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
-			goto out;
-			break;
-		default:
-			if (option_debug)
-				ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
-			else if (option_verbose > 1)
-				ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
-			goto out;
-		}
+	char *save_macro_exten;
+	char *save_macro_context;
+	char *save_macro_priority;
+	char *save_macro_offset;
+	struct localuser *u;
+ 
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Macro() requires arguments. See \"show application macro\" for help.\n");
+		return -1;
 	}
-	if (strcasecmp(chan->context, fullmacro)) {
-		if (option_verbose > 1)
-			ast_verbose(VERBOSE_PREFIX_2 "Channel '%s' jumping out of macro '%s'\n", chan->name, macro);
-		break;
-	}
-	/* don't stop executing extensions when we're in "h" */
-	if (chan->_softhangup && strcasecmp(oldexten,"h")) {
-		ast_log(LOG_DEBUG, "Extension %s, priority %d returned normally even though call was hung up\n",
-			chan->exten, chan->priority);
-		goto out;
-	}
-	chan->priority++;
-  }
-out:
-  for (x=1;x<argc;x++) {
-  	/* Restore old arguments and delete ours */
-	snprintf(varname, sizeof(varname), "ARG%d", x);
-  	if (oldargs[x]) {
-		pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
-		free(oldargs[x]);
+
+	LOCAL_USER_ADD(u);
+
+	/* Count how many levels deep the rabbit hole goes */
+	tmp = pbx_builtin_getvar_helper(chan, "MACRO_DEPTH");
+	if (tmp) {
+		sscanf(tmp, "%d", &depth);
 	} else {
-		pbx_builtin_setvar_helper(chan, varname, NULL);
+		depth = 0;
 	}
-  }

-  /* Restore macro variables */
-  pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
-  if (save_macro_exten) free(save_macro_exten);
-  pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", save_macro_context);
-  if (save_macro_context) free(save_macro_context);
-  pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
-  if (save_macro_priority) free(save_macro_priority);
-  if (setmacrocontext) {
-	chan->macrocontext[0] = '\0';
-	chan->macroexten[0] = '\0';
-	chan->macropriority = 0;
-  }
+	if (depth >= 7) {
+		ast_log(LOG_ERROR, "Macro():  possible infinite loop detected.  Returning early.\n");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+	snprintf(depthc, sizeof(depthc), "%d", depth + 1);
+	pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);

-  if (!strcasecmp(chan->context, fullmacro)) {
-  	/* If we're leaving the macro normally, restore original information */
-	chan->priority = oldpriority;
-	strncpy(chan->context, oldcontext, sizeof(chan->context) - 1);
-	if (!(chan->_softhangup & AST_SOFTHANGUP_ASYNCGOTO)) {
-		/* Copy the extension, so long as we're not in softhangup, where we could be given an asyncgoto */
-		strncpy(chan->exten, oldexten, sizeof(chan->exten) - 1);
-		if ((offsets = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"))) {
-			/* Handle macro offset if it's set by checking the availability of step n + offset + 1, otherwise continue
-			   normally if there is any problem */
-			if (sscanf(offsets, "%d", &offset) == 1) {
-				if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + offset + 1, chan->callerid)) {
-					chan->priority += offset;
+	tmp = ast_strdupa(data);
+	rest = tmp;
+	macro = strsep(&rest, "|");
+	if (ast_strlen_zero(macro)) {
+		ast_log(LOG_WARNING, "Invalid macro name specified\n");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+	snprintf(fullmacro, sizeof(fullmacro), "macro-%s", macro);
+	if (!ast_exists_extension(chan, fullmacro, "s", 1, chan->cid.cid_num)) {
+  		if (!ast_context_find(fullmacro)) 
+			ast_log(LOG_WARNING, "No such context '%s' for macro '%s'\n", fullmacro, macro);
+		else
+	  		ast_log(LOG_WARNING, "Context '%s' for macro '%s' lacks 's' extension, priority 1\n", fullmacro, macro);
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+	
+	/* Save old info */
+	oldpriority = chan->priority;
+	ast_copy_string(oldexten, chan->exten, sizeof(oldexten));
+	ast_copy_string(oldcontext, chan->context, sizeof(oldcontext));
+	if (ast_strlen_zero(chan->macrocontext)) {
+		ast_copy_string(chan->macrocontext, chan->context, sizeof(chan->macrocontext));
+		ast_copy_string(chan->macroexten, chan->exten, sizeof(chan->macroexten));
+		chan->macropriority = chan->priority;
+		setmacrocontext=1;
+	}
+	argc = 1;
+	/* Save old macro variables */
+	save_macro_exten = pbx_builtin_getvar_helper(chan, "MACRO_EXTEN");
+	if (save_macro_exten) 
+		save_macro_exten = strdup(save_macro_exten);
+	pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", oldexten);
+
+	save_macro_context = pbx_builtin_getvar_helper(chan, "MACRO_CONTEXT");
+	if (save_macro_context)
+		save_macro_context = strdup(save_macro_context);
+	pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", oldcontext);
+
+	save_macro_priority = pbx_builtin_getvar_helper(chan, "MACRO_PRIORITY");
+	if (save_macro_priority) 
+		save_macro_priority = strdup(save_macro_priority);
+	snprintf(pc, sizeof(pc), "%d", oldpriority);
+	pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", pc);
+  
+	save_macro_offset = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET");
+	if (save_macro_offset) 
+		save_macro_offset = strdup(save_macro_offset);
+	pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", NULL);
+
+	/* Setup environment for new run */
+	chan->exten[0] = 's';
+	chan->exten[1] = '\0';
+	ast_copy_string(chan->context, fullmacro, sizeof(chan->context));
+	chan->priority = 1;
+
+	while((cur = strsep(&rest, "|")) && (argc < MAX_ARGS)) {
+  		/* Save copy of old arguments if we're overwriting some, otherwise
+	   	let them pass through to the other macro */
+  		snprintf(varname, sizeof(varname), "ARG%d", argc);
+		oldargs[argc] = pbx_builtin_getvar_helper(chan, varname);
+		if (oldargs[argc])
+			oldargs[argc] = strdup(oldargs[argc]);
+		pbx_builtin_setvar_helper(chan, varname, cur);
+		argc++;
+	}
+	autoloopflag = ast_test_flag(chan, AST_FLAG_IN_AUTOLOOP);
+	ast_set_flag(chan, AST_FLAG_IN_AUTOLOOP);
+	while(ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
+		/* Reset the macro depth, if it was changed in the last iteration */
+		pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
+		if ((res = ast_spawn_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num))) {
+			/* Something bad happened, or a hangup has been requested. */
+			if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F')) ||
+		    	(res == '*') || (res == '#')) {
+				/* Just return result as to the previous application as if it had been dialed */
+				ast_log(LOG_DEBUG, "Oooh, got something to jump out with ('%c')!\n", res);
+				break;
+			}
+			switch(res) {
+	        	case MACRO_EXIT_RESULT:
+                        	res = 0;
+				goto out;
+			case AST_PBX_KEEPALIVE:
+				if (option_debug)
+					ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited KEEPALIVE in macro %s on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
+				else if (option_verbose > 1)
+					ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited KEEPALIVE in macro '%s' on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
+				goto out;
+				break;
+			default:
+				if (option_debug)
+					ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
+				else if (option_verbose > 1)
+					ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
+				goto out;
+			}
+		}
+		if (strcasecmp(chan->context, fullmacro)) {
+			if (option_verbose > 1)
+				ast_verbose(VERBOSE_PREFIX_2 "Channel '%s' jumping out of macro '%s'\n", chan->name, macro);
+			break;
+		}
+		/* don't stop executing extensions when we're in "h" */
+		if (chan->_softhangup && strcasecmp(oldexten,"h")) {
+			ast_log(LOG_DEBUG, "Extension %s, priority %d returned normally even though call was hung up\n",
+				chan->exten, chan->priority);
+			goto out;
+		}
+		chan->priority++;
+  	}
+	out:
+	/* Reset the depth back to what it was when the routine was entered (like if we called Macro recursively) */
+	snprintf(depthc, sizeof(depthc), "%d", depth);
+	pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
+
+	ast_set2_flag(chan, autoloopflag, AST_FLAG_IN_AUTOLOOP);
+  	for (x=1; x<argc; x++) {
+  		/* Restore old arguments and delete ours */
+		snprintf(varname, sizeof(varname), "ARG%d", x);
+  		if (oldargs[x]) {
+			pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
+			free(oldargs[x]);
+		} else {
+			pbx_builtin_setvar_helper(chan, varname, NULL);
+		}
+  	}
+
+	/* Restore macro variables */
+	pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
+	if (save_macro_exten)
+		free(save_macro_exten);
+	pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", save_macro_context);
+	if (save_macro_context)
+		free(save_macro_context);
+	pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
+	if (save_macro_priority)
+		free(save_macro_priority);
+	if (setmacrocontext) {
+		chan->macrocontext[0] = '\0';
+		chan->macroexten[0] = '\0';
+		chan->macropriority = 0;
+	}
+
+	if (!strcasecmp(chan->context, fullmacro)) {
+  		/* If we're leaving the macro normally, restore original information */
+		chan->priority = oldpriority;
+		ast_copy_string(chan->context, oldcontext, sizeof(chan->context));
+		if (!(chan->_softhangup & AST_SOFTHANGUP_ASYNCGOTO)) {
+			/* Copy the extension, so long as we're not in softhangup, where we could be given an asyncgoto */
+			ast_copy_string(chan->exten, oldexten, sizeof(chan->exten));
+			if ((offsets = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"))) {
+				/* Handle macro offset if it's set by checking the availability of step n + offset + 1, otherwise continue
+			   	normally if there is any problem */
+				if (sscanf(offsets, "%d", &offset) == 1) {
+					if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + offset + 1, chan->cid.cid_num)) {
+						chan->priority += offset;
+					}
 				}
 			}
 		}
 	}
-  }

-  pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
-  if (save_macro_offset) free(save_macro_offset);
-  return res;
+	pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
+	if (save_macro_offset)
+		free(save_macro_offset);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static int macroif_exec(struct ast_channel *chan, void *data) 
+{
+	char *expr = NULL, *label_a = NULL, *label_b = NULL;
+	int res = 0;
+	struct localuser *u;
+
+	LOCAL_USER_ADD(u);
+
+	expr = ast_strdupa(data);
+	if (!expr) {
+		ast_log(LOG_ERROR, "Out of Memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if ((label_a = strchr(expr, '?'))) {
+		*label_a = '\0';
+		label_a++;
+		if ((label_b = strchr(label_a, ':'))) {
+			*label_b = '\0';
+			label_b++;
+		}
+		if (ast_true(expr))
+			macro_exec(chan, label_a);
+		else if (label_b) 
+			macro_exec(chan, label_b);
+	} else
+		ast_log(LOG_WARNING, "Invalid Syntax.\n");
+
+	LOCAL_USER_REMOVE(u);
+
+	return res;
+}
+			
+static int macro_exit_exec(struct ast_channel *chan, void *data)
+{
+	return MACRO_EXIT_RESULT;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(if_app);
+	res |= ast_unregister_application(exit_app);
+	res |= ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
 {
-	return ast_register_application(app, macro_exec, synopsis, descrip);
+	int res;
+
+	res = ast_register_application(exit_app, macro_exit_exec, exit_synopsis, exit_descrip);
+	res |= ast_register_application(if_app, macroif_exec, if_synopsis, if_descrip);
+	res |= ast_register_application(app, macro_exec, synopsis, descrip);
+
+	return res;
 }

 char *description(void)
--- a/apps/app_math.c
+++ b/apps/app_math.c
@@ -0,0 +1,297 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2004 - 2005, Andy Powell 
+ *
+ * Updated by Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief A simple math application
+ * 
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <sys/time.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <time.h>
+#include <dirent.h>
+#include <ctype.h>
+#include <sys/file.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/options.h"
+#include "asterisk/config.h"
+#include "asterisk/say.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+#include "asterisk/manager.h"
+#include "asterisk/localtime.h"
+#include "asterisk/cli.h"
+#include "asterisk/utils.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+
+static char *tdesc = "Basic Math Functions";
+
+static char *app_math = "Math";
+
+static char *math_synopsis = "Performs Mathematical Functions";
+
+static char *math_descrip =
+"Math(returnvar,<number1><op><number 2>\n\n"
+"Perform floating point calculation on number 1 to number 2 and \n"
+"store the result in returnvar.  Valid ops are: \n"
+"    +,-,/,*,%,<,>,>=,<=,==\n"
+"and behave as their C equivalents.\n";
+
+#define ADDFUNCTION 0
+#define DIVIDEFUNCTION 1
+#define MULTIPLYFUNCTION 2
+#define SUBTRACTFUNCTION 3
+#define MODULUSFUNCTION 4
+
+#define GTFUNCTION 5
+#define LTFUNCTION 6
+#define GTEFUNCTION 7
+#define LTEFUNCTION 8
+#define EQFUNCTION 9
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int math_exec(struct ast_channel *chan, void *data) 
+{
+	float fnum1;
+	float fnum2;
+	float ftmp = 0;
+	char *op;
+	int iaction=-1;
+	static int deprecation_warning = 0;
+
+	/* dunno, big calulations :D */
+	char user_result[30];
+
+	char *s;
+	char *mvar, *mvalue1, *mvalue2=NULL;
+		
+	struct localuser *u;
+
+	if (!deprecation_warning) {
+		ast_log(LOG_WARNING, "Math() is deprecated, please use Set(var=${MATH(...)} instead.\n");
+		deprecation_warning = 1;
+	}
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "No parameters passed. !\n");
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	s = ast_strdupa(data);
+	if (!s) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	mvar = strsep(&s, "|");
+	mvalue1 = strsep(&s, "|");
+	
+	if ((op = strchr(mvalue1, '+'))) {
+		iaction = ADDFUNCTION;
+		*op = '\0';
+	} else if ((op = strchr(mvalue1, '-'))) {
+		iaction = SUBTRACTFUNCTION;
+		*op = '\0';
+	} else if ((op = strchr(mvalue1, '*'))) {
+		iaction = MULTIPLYFUNCTION;
+		*op = '\0';
+	} else if ((op = strchr(mvalue1, '/'))) {
+		iaction = DIVIDEFUNCTION;
+		*op = '\0';
+	} else if ((op = strchr(mvalue1, '>'))) {
+		iaction = GTFUNCTION;
+		*op = '\0';
+		if (*(op+1) == '=') {
+			op++;
+			*op = '\0';
+			iaction = GTEFUNCTION;
+		}
+	} else if ((op = strchr(mvalue1, '<'))) {
+		iaction = LTFUNCTION;
+		*op = '\0';
+		if (*(op+1) == '=') {
+			op++;
+			*op = '\0';
+			iaction = LTEFUNCTION;
+		}
+	} else if ((op = strchr(mvalue1, '='))) {
+		iaction = GTFUNCTION;
+		*op = '\0';
+		if (*(op+1) == '=') {
+			op++;
+			*op = '\0';
+			iaction = EQFUNCTION;
+		} else
+			op = NULL;
+	} 
+	
+	if (op) 
+		mvalue2 = op + 1;
+		
+	if (!mvar || !mvalue1 || !mvalue2) {
+		ast_log(LOG_WARNING, "Supply all the parameters - just this once, please\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (!strcmp(mvar,"")) {
+		ast_log(LOG_WARNING, "No return variable set.\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (sscanf(mvalue1, "%f", &fnum1) != 1) {
+		ast_log(LOG_WARNING, "'%s' is not a valid number\n", mvalue1);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (sscanf(mvalue2, "%f", &fnum2) != 1) {
+		ast_log(LOG_WARNING, "'%s' is not a valid number\n", mvalue2);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	switch (iaction) {
+	case ADDFUNCTION :
+		ftmp = fnum1 + fnum2;
+		break;
+	case DIVIDEFUNCTION :
+		if (fnum2 <=0)
+			ftmp = 0; /* can't do a divide by 0 */
+		else
+			ftmp = (fnum1 / fnum2);
+		break;
+	case MULTIPLYFUNCTION :
+		ftmp = (fnum1 * fnum2);
+		break;
+	case SUBTRACTFUNCTION :
+		ftmp = (fnum1 - fnum2);
+		break;
+	case MODULUSFUNCTION : {
+		int inum1 = fnum1;
+		int inum2 = fnum2;
+			
+		ftmp = (inum1 % inum2);
+		
+		break;
+		}
+	case GTFUNCTION :
+		if (fnum1 > fnum2)
+			strcpy(user_result, "TRUE");
+		else
+			strcpy(user_result, "FALSE");
+		break;
+	case LTFUNCTION :
+		if (fnum1 < fnum2)
+			strcpy(user_result, "TRUE");
+		else
+			strcpy(user_result, "FALSE");
+		break;
+	case GTEFUNCTION :
+		if (fnum1 >= fnum2)
+			strcpy(user_result, "TRUE");
+		else
+			strcpy(user_result, "FALSE");
+		break;
+	case LTEFUNCTION :
+		if (fnum1 <= fnum2)
+			strcpy(user_result, "TRUE");
+		else
+			strcpy(user_result, "FALSE");
+		break;					
+	case EQFUNCTION :
+		if (fnum1 == fnum2)
+			strcpy(user_result, "TRUE");
+		else
+			strcpy(user_result, "FALSE");
+		break;
+	default :
+		ast_log(LOG_WARNING, "Something happened that neither of us should be proud of %d\n", iaction);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (iaction < GTFUNCTION || iaction > EQFUNCTION) 
+		snprintf(user_result,sizeof(user_result),"%f",ftmp);
+		
+	pbx_builtin_setvar_helper(chan, mvar, user_result);	
+	
+	LOCAL_USER_REMOVE(u);
+	return 0;
+}
+
+int unload_module(void)
+{
+	int res;
+	
+	res = ast_unregister_application(app_math);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app_math, math_exec, math_synopsis, math_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
+
+/* Fading everything to black and blue... */
--- a/apps/app_md5.c
+++ b/apps/app_md5.c
@@ -0,0 +1,205 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2005, Olle E. Johansson, Edvina.net
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+
+/*! \file
+ *
+ * \brief MD5 checksum application
+ * 
+ * \todo Remove this deprecated application in 1.3dev
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/utils.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/app.h"
+
+static char *tdesc_md5 = "MD5 checksum applications";
+static char *app_md5 = "MD5";
+static char *desc_md5 = "Calculate MD5 checksum";
+static char *synopsis_md5 = 
+"  MD5(<var>=<string>): Calculates a MD5 checksum on <string>.\n"
+"Returns hash value in a channel variable. \n";
+
+static char *app_md5check = "MD5Check";
+static char *desc_md5check = "Check MD5 checksum";
+static char *synopsis_md5check = 
+"  MD5Check(<md5hash>|<string>[|options]): Calculates a MD5 checksum on <string>\n"
+"and compares it with the hash. Returns 0 if <md5hash> is correct for <string>.\n"
+"The option string may contain zero or more of the following characters:\n"
+"	'j' -- jump to priority n+101 if the hash and string do not match \n"
+"This application sets the following channel variable upon completion:\n"
+"	CHECKMD5STATUS	The status of the MD5 check, one of the following\n"
+"		MATCH | NOMATCH\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+/*--- md5_exec: Calculate MD5 checksum (hash) on given string and
+	return it in channel variable ---*/
+static int md5_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *varname= NULL; /* Variable to set */
+	char *string = NULL; /* String to calculate on */
+	char retvar[50]; /* Return value */
+	static int dep_warning = 0;
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "This application has been deprecated, please use the MD5 function instead.\n");
+		dep_warning = 1;
+	}	
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Syntax: md5(<varname>=<string>) - missing argument!\n");
+		return -1;
+	}
+	
+	LOCAL_USER_ADD(u);
+
+	memset(retvar,0, sizeof(retvar));
+	string = ast_strdupa(data);
+	varname = strsep(&string,"=");
+	if (ast_strlen_zero(varname)) {
+		ast_log(LOG_WARNING, "Syntax: md5(<varname>=<string>) - missing argument!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	ast_md5_hash(retvar, string);
+	pbx_builtin_setvar_helper(chan, varname, retvar);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+/*--- md5check_exec: Calculate MD5 checksum and compare it with
+	existing checksum. ---*/
+static int md5check_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *string = NULL; /* String to calculate on */
+	char newhash[50]; /* Return value */
+	static int dep_warning = 0;
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(md5hash);
+		AST_APP_ARG(string);
+		AST_APP_ARG(options);
+	);
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "This application has been deprecated, please use the CHECK_MD5 function instead.\n");
+		dep_warning = 1;
+	}
+	
+	LOCAL_USER_ADD(u);
+
+	if (!(string = ast_strdupa(data))) {
+		ast_log(LOG_WARNING, "Memory Error!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args, string);
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
+	if (ast_strlen_zero(args.md5hash) || ast_strlen_zero(args.string)) {
+		ast_log(LOG_WARNING, "Syntax: MD5Check(<md5hash>|<string>[|options]) - missing argument!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	memset(newhash,0, sizeof(newhash));
+
+	ast_md5_hash(newhash, args.string);
+	if (!strcmp(newhash, args.md5hash)) {	/* Verification ok */
+		if (option_debug > 2)
+			ast_log(LOG_DEBUG, "MD5 verified ok: %s -- %s\n", args.md5hash, args.string);
+		pbx_builtin_setvar_helper(chan, "CHECKMD5STATUS", "MATCH");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+	if (option_debug > 2)
+		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 (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);
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app_md5);
+	res |= ast_unregister_application(app_md5check);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+
+	res = ast_register_application(app_md5check, md5check_exec, desc_md5check, synopsis_md5check);
+	res |= ast_register_application(app_md5, md5_exec, desc_md5, synopsis_md5);
+	
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc_md5;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_meetme.c
+++ b/apps/app_meetme.c
--- a/apps/app_milliwatt.c
+++ b/apps/app_milliwatt.c
@@ -1,28 +1,45 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Digital Milliwatt Test
- * 
- * Copyright (C) 2002, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Digital Milliwatt Test
+ * 
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
+#include <unistd.h>
 #include <errno.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+
 static char *tdesc = "Digital Milliwatt (mu-law) Test Application";

 static char *app = "Milliwatt";
@@ -111,6 +128,7 @@ static int milliwatt_exec(struct ast_channel *chan, void *data)
 	if (ast_activate_generator(chan,&milliwattgen,"milliwatt") < 0)
 	{
 		ast_log(LOG_WARNING,"Failed to activate generator on '%s'\n",chan->name);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	while(!ast_safe_sleep(chan, 10000));
@@ -121,8 +139,13 @@ static int milliwatt_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_mixmonitor.c
+++ b/apps/app_mixmonitor.c
@@ -0,0 +1,466 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2005, Anthony Minessale II
+ * Copyright (C) 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ * Kevin P. Fleming <kpfleming@digium.com>
+ *
+ * Based on app_muxmon.c provided by
+ * Anthony Minessale II <anthmct@yahoo.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief MixMonitor() - Record a call and mix the audio during the recording
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/chanspy.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/cli.h"
+#include "asterisk/options.h"
+#include "asterisk/app.h"
+#include "asterisk/linkedlists.h"
+
+#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
+
+static const char *tdesc = "Mixed Audio Monitoring Application";
+static const char *app = "MixMonitor";
+static const char *synopsis = "Record a call and mix the audio during the recording";
+static const char *desc = ""
+"  MixMonitor(<file>.<ext>[|<options>[|<command>]])\n\n"
+"Records the audio on the current channel to the specified file.\n"
+"If the filename is an absolute path, uses that path, otherwise\n"
+"creates the file in the configured monitoring directory from\n"
+"asterisk.conf.\n\n"
+"Valid options:\n"
+" a      - Append to the file instead of overwriting it.\n"
+" b      - Only save audio to the file while the channel is bridged.\n"
+"          Note: does not include conferences.\n"
+" v(<x>) - Adjust the heard volume by a factor of <x> (range -4 to 4)\n"	
+" V(<x>) - Adjust the spoken volume by a factor of <x> (range -4 to 4)\n"	
+" W(<x>) - Adjust the both heard and spoken volumes by a factor of <x>\n"
+"         (range -4 to 4)\n\n"	
+"<command> will be executed when the recording is over\n"
+"Any strings matching ^{X} will be unescaped to ${X} and \n"
+"all variables will be evaluated at that time.\n"
+"The variable MIXMONITOR_FILENAME will contain the filename used to record.\n"
+"";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static const char *mixmonitor_spy_type = "MixMonitor";
+
+struct mixmonitor {
+	struct ast_channel *chan;
+	char *filename;
+	char *post_process;
+	unsigned int flags;
+	int readvol;
+	int writevol;
+};
+
+enum {
+	MUXFLAG_APPEND = (1 << 1),
+	MUXFLAG_BRIDGED = (1 << 2),
+	MUXFLAG_VOLUME = (1 << 3),
+	MUXFLAG_READVOLUME = (1 << 4),
+	MUXFLAG_WRITEVOLUME = (1 << 5),
+} mixmonitor_flags;
+
+enum {
+	OPT_ARG_READVOLUME = 0,
+	OPT_ARG_WRITEVOLUME,
+	OPT_ARG_VOLUME,
+	OPT_ARG_ARRAY_SIZE,
+} mixmonitor_args;
+
+AST_APP_OPTIONS(mixmonitor_opts, {
+	AST_APP_OPTION('a', MUXFLAG_APPEND),
+	AST_APP_OPTION('b', MUXFLAG_BRIDGED),
+	AST_APP_OPTION_ARG('v', MUXFLAG_READVOLUME, OPT_ARG_READVOLUME),
+	AST_APP_OPTION_ARG('V', MUXFLAG_WRITEVOLUME, OPT_ARG_WRITEVOLUME),
+	AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
+});
+
+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....
+	   DON'T TOUCH IT!!!  RUN AWAY!!! */
+	if (spy->status != CHANSPY_RUNNING)
+		return;
+
+	if (!chan)
+		return;
+
+	ast_mutex_lock(&chan->lock);
+	ast_channel_spy_remove(chan, spy);
+	ast_mutex_unlock(&chan->lock);
+}
+
+static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy) 
+{
+	struct ast_channel *peer;
+	int res;
+
+	if (!chan)
+		return -1;
+
+	ast_mutex_lock(&chan->lock);
+	res = ast_channel_spy_add(chan, spy);
+	ast_mutex_unlock(&chan->lock);
+		
+	if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan)))
+		ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);	
+
+	return res;
+}
+
+#define SAMPLES_PER_FRAME 160
+
+static void *mixmonitor_thread(void *obj) 
+{
+	struct mixmonitor *mixmonitor = obj;
+	struct ast_channel_spy spy;
+	struct ast_filestream *fs = NULL;
+	char *ext, *name;
+	unsigned int oflags;
+	struct ast_frame *f;
+	char post_process[1024] = "";
+	
+	STANDARD_INCREMENT_USECOUNT;
+
+	name = ast_strdupa(mixmonitor->chan->name);
+
+	oflags = O_CREAT|O_WRONLY;
+	oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
+		
+	if ((ext = strrchr(mixmonitor->filename, '.'))) {
+		*(ext++) = '\0';
+	} else {
+		ext = "raw";
+	}
+
+	fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644);
+	if (!fs) {
+		ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
+		goto out;
+	}
+
+	if (ast_test_flag(mixmonitor, MUXFLAG_APPEND))
+		ast_seekstream(fs, 0, SEEK_END);
+	
+	memset(&spy, 0, sizeof(spy));
+	ast_set_flag(&spy, CHANSPY_FORMAT_AUDIO);
+	ast_set_flag(&spy, CHANSPY_MIXAUDIO);
+	spy.type = mixmonitor_spy_type;
+	spy.status = CHANSPY_RUNNING;
+	spy.read_queue.format = AST_FORMAT_SLINEAR;
+	spy.write_queue.format = AST_FORMAT_SLINEAR;
+	if (mixmonitor->readvol) {
+		ast_set_flag(&spy, CHANSPY_READ_VOLADJUST);
+		spy.read_vol_adjustment = mixmonitor->readvol;
+	}
+	if (mixmonitor->writevol) {
+		ast_set_flag(&spy, CHANSPY_WRITE_VOLADJUST);
+		spy.write_vol_adjustment = mixmonitor->writevol;
+	}
+	ast_mutex_init(&spy.lock);
+
+	if (startmon(mixmonitor->chan, &spy)) {
+		ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
+			spy.type, mixmonitor->chan->name);
+		goto out2;
+	}
+
+	if (option_verbose > 1)
+		ast_verbose(VERBOSE_PREFIX_2 "Begin MixMonitor Recording %s\n", name);
+	
+	while (1) {
+		struct ast_frame *next;
+		int write;
+
+		ast_mutex_lock(&spy.lock);
+
+		ast_channel_spy_trigger_wait(&spy);
+		
+		if (ast_check_hangup(mixmonitor->chan) || spy.status != CHANSPY_RUNNING) {
+			ast_mutex_unlock(&spy.lock);
+			break;
+		}
+		
+		while (1) {
+			if (!(f = ast_channel_spy_read_frame(&spy, SAMPLES_PER_FRAME)))
+				break;
+
+			write = (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) ||
+				 ast_bridged_channel(mixmonitor->chan));
+
+			/* it is possible for ast_channel_spy_read_frame() to return a chain
+			   of frames if a queue flush was necessary, so process them
+			*/
+			for (; f; f = next) {
+				next = f->next;
+				if (write)
+					ast_writestream(fs, f);
+				ast_frfree(f);
+			}
+		}
+
+		ast_mutex_unlock(&spy.lock);
+	}
+	
+	if (mixmonitor->post_process) {
+		char *p;
+
+		for (p = mixmonitor->post_process; *p ; p++) {
+			if (*p == '^' && *(p+1) == '{') {
+				*p = '$';
+			}
+		}
+		pbx_substitute_variables_helper(mixmonitor->chan, mixmonitor->post_process, post_process, sizeof(post_process) - 1);
+	}
+
+	stopmon(mixmonitor->chan, &spy);
+
+	if (option_verbose > 1)
+		ast_verbose(VERBOSE_PREFIX_2 "End MixMonitor Recording %s\n", name);
+
+	if (!ast_strlen_zero(post_process)) {
+		if (option_verbose > 2)
+			ast_verbose(VERBOSE_PREFIX_2 "Executing [%s]\n", post_process);
+		ast_safe_system(post_process);
+	}
+
+out2:
+	ast_mutex_destroy(&spy.lock);
+
+	if (fs)
+		ast_closestream(fs);
+
+out:
+	free(mixmonitor);
+
+	STANDARD_DECREMENT_USECOUNT;
+
+	return NULL;
+}
+
+static void launch_monitor_thread(struct ast_channel *chan, const char *filename, unsigned int flags,
+				  int readvol, int writevol, const char *post_process) 
+{
+	pthread_attr_t attr;
+	pthread_t thread;
+	struct mixmonitor *mixmonitor;
+	int len;
+
+	len = sizeof(*mixmonitor) + strlen(filename) + 1;
+	if (!ast_strlen_zero(post_process))
+		len += strlen(post_process) + 1;
+
+	if (!(mixmonitor = calloc(1, len))) {
+		ast_log(LOG_ERROR, "Memory Error!\n");
+		return;
+	}
+
+	mixmonitor->chan = chan;
+	mixmonitor->filename = (char *) mixmonitor + sizeof(*mixmonitor);
+	strcpy(mixmonitor->filename, filename);
+	if (!ast_strlen_zero(post_process)) {
+		mixmonitor->post_process = mixmonitor->filename + strlen(filename) + 1;
+		strcpy(mixmonitor->post_process, post_process);
+	}
+	mixmonitor->readvol = readvol;
+	mixmonitor->writevol = writevol;
+	mixmonitor->flags = flags;
+
+	pthread_attr_init(&attr);
+	pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
+	ast_pthread_create(&thread, &attr, mixmonitor_thread, mixmonitor);
+	pthread_attr_destroy(&attr);
+}
+
+static int mixmonitor_exec(struct ast_channel *chan, void *data)
+{
+	int x, readvol = 0, writevol = 0;
+	struct localuser *u;
+	struct ast_flags flags = {0};
+	char *parse;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(filename);
+		AST_APP_ARG(options);
+		AST_APP_ARG(post_process);
+	);
+	
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	if (!(parse = ast_strdupa(data))) {
+		ast_log(LOG_WARNING, "Memory Error!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args, parse);
+	
+	if (ast_strlen_zero(args.filename)) {
+		ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (args.options) {
+		char *opts[OPT_ARG_ARRAY_SIZE] = { NULL, };
+
+		ast_app_parse_options(mixmonitor_opts, &flags, opts, args.options);
+
+		if (ast_test_flag(&flags, MUXFLAG_READVOLUME)) {
+			if (ast_strlen_zero(opts[OPT_ARG_READVOLUME])) {
+				ast_log(LOG_WARNING, "No volume level was provided for the heard volume ('v') option.\n");
+			} else if ((sscanf(opts[OPT_ARG_READVOLUME], "%d", &x) != 1) || (x < -4) || (x > 4)) {
+				ast_log(LOG_NOTICE, "Heard volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_READVOLUME]);
+			} else {
+				readvol = get_volfactor(x);
+			}
+		}
+		
+		if (ast_test_flag(&flags, MUXFLAG_WRITEVOLUME)) {
+			if (ast_strlen_zero(opts[OPT_ARG_WRITEVOLUME])) {
+				ast_log(LOG_WARNING, "No volume level was provided for the spoken volume ('V') option.\n");
+			} else if ((sscanf(opts[OPT_ARG_WRITEVOLUME], "%d", &x) != 1) || (x < -4) || (x > 4)) {
+				ast_log(LOG_NOTICE, "Spoken volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_WRITEVOLUME]);
+			} else {
+				writevol = get_volfactor(x);
+			}
+		}
+		
+		if (ast_test_flag(&flags, MUXFLAG_VOLUME)) {
+			if (ast_strlen_zero(opts[OPT_ARG_VOLUME])) {
+				ast_log(LOG_WARNING, "No volume level was provided for the combined volume ('W') option.\n");
+			} else if ((sscanf(opts[OPT_ARG_VOLUME], "%d", &x) != 1) || (x < -4) || (x > 4)) {
+				ast_log(LOG_NOTICE, "Combined volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_VOLUME]);
+			} else {
+				readvol = writevol = get_volfactor(x);
+			}
+		}
+	}
+
+	/* if not provided an absolute path, use the system-configured monitoring directory */
+	if (args.filename[0] != '/') {
+		char *build;
+
+		build = alloca(strlen(ast_config_AST_MONITOR_DIR) + strlen(args.filename) + 3);
+		sprintf(build, "%s/%s", ast_config_AST_MONITOR_DIR, args.filename);
+		args.filename = build;
+	}
+
+	pbx_builtin_setvar_helper(chan, "MIXMONITOR_FILENAME", args.filename);
+	launch_monitor_thread(chan, args.filename, flags.flags, readvol, writevol, args.post_process);
+
+	LOCAL_USER_REMOVE(u);
+
+	return 0;
+}
+
+static int mixmonitor_cli(int fd, int argc, char **argv) 
+{
+	struct ast_channel *chan;
+
+	if (argc < 3)
+		return RESULT_SHOWUSAGE;
+
+	if (!(chan = ast_get_channel_by_name_prefix_locked(argv[2], strlen(argv[2])))) {
+		ast_cli(fd, "No channel matching '%s' found.\n", argv[2]);
+		return RESULT_SUCCESS;
+	}
+
+	if (!strcasecmp(argv[1], "start"))
+		mixmonitor_exec(chan, argv[3]);
+	else if (!strcasecmp(argv[1], "stop"))
+		ast_channel_spy_stop_by_type(chan, mixmonitor_spy_type);
+
+	ast_mutex_unlock(&chan->lock);
+
+	return RESULT_SUCCESS;
+}
+
+
+static struct ast_cli_entry cli_mixmonitor = {
+	{ "mixmonitor", NULL, NULL },
+	mixmonitor_cli, 
+	"Execute a MixMonitor command",
+	"mixmonitor <start|stop> <chan_name> [<args>]\n"
+};
+
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_cli_unregister(&cli_mixmonitor);
+	res |= ast_unregister_application(app);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+
+	res = ast_cli_register(&cli_mixmonitor);
+	res |= ast_register_application(app, mixmonitor_exec, synopsis, desc);
+
+	return res;
+}
+
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_mp3.c
+++ b/apps/app_mp3.c
@@ -1,24 +1,28 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Silly application to play an MP3 file -- uses mpg123
- * 
- * Copyright (C) 1999-2004, Digium, Inc.
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Silly application to play an MP3 file -- uses mpg123
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/frame.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
 #include <string.h>
 #include <stdio.h>
 #include <signal.h>
@@ -27,6 +31,19 @@
 #include <fcntl.h>
 #include <sys/time.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/frame.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+
 #define LOCAL_MPG_123 "/usr/local/bin/mpg123"
 #define MPG_123 "/usr/bin/mpg123"

@@ -37,9 +54,9 @@ static char *app = "MP3Player";
 static char *synopsis = "Play an MP3 file or stream";

 static char *descrip = 
-"  MP3Player(location) Executes mpg123 to play the given location\n"
-"which typically would be a  filename  or  a URL. Returns  -1  on\n"
-"hangup or 0 otherwise. User can exit by pressing any key\n.";
+"  MP3Player(location) Executes mpg123 to play the given location,\n"
+"which typically would be a filename or a URL. User can exit by pressing\n"
+"any key on the dialpad, or by hanging up."; 

 STANDARD_LOCAL_USER;

@@ -104,60 +121,51 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 	int pid = -1;
 	int owriteformat;
 	int timeout = 2000;
-	struct timeval now, next;
+	struct timeval next;
 	struct ast_frame *f;
 	struct myframe {
 		struct ast_frame f;
 		char offset[AST_FRIENDLY_OFFSET];
 		short frdata[160];
 	} myf;
-	if (!data) {
+	
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "MP3 Playback requires an argument (filename)\n");
 		return -1;
 	}
+
+	LOCAL_USER_ADD(u);
+
 	if (pipe(fds)) {
 		ast_log(LOG_WARNING, "Unable to create pipe\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
-	LOCAL_USER_ADD(u);
+	
 	ast_stopstream(chan);

 	owriteformat = chan->writeformat;
 	res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
 	if (res < 0) {
 		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	
-	gettimeofday(&now, NULL);
 	res = mp3play((char *)data, fds[1]);
 	if (!strncasecmp((char *)data, "http://", 7)) {
 		timeout = 10000;
 	}
 	/* Wait 1000 ms first */
-	next = now;
+	next = ast_tvnow();
 	next.tv_sec += 1;
 	if (res >= 0) {
 		pid = res;
 		/* Order is important -- there's almost always going to be mp3...  we want to prioritize the
 		   user */
 		for (;;) {
-			gettimeofday(&now, NULL);
-			ms = (next.tv_sec - now.tv_sec) * 1000;
-			ms += (next.tv_usec - now.tv_usec) / 1000;
-#if 0
-			printf("ms: %d\n", ms);
-#endif			
+			ms = ast_tvdiff_ms(next, ast_tvnow());
 			if (ms <= 0) {
-#if 0
-				{
-					static struct timeval last;
-					struct timeval tv;
-					gettimeofday(&tv, NULL);
-					printf("Since last: %ld\n", (tv.tv_sec - last.tv_sec) * 1000 + (tv.tv_usec - last.tv_usec) / 1000);
-					last = tv;
-				}
-#endif
 				res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata), timeout);
 				if (res > 0) {
 					myf.f.frametype = AST_FRAME_VOICE;
@@ -179,14 +187,7 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 					res = 0;
 					break;
 				}
-				next.tv_usec += res / 2 * 125;
-				if (next.tv_usec >= 1000000) {
-					next.tv_usec -= 1000000;
-					next.tv_sec++;
-				}
-#if 0
-				printf("Next: %d\n", ms);
-#endif				
+				next = ast_tvadd(next, ast_samp2tv(myf.f.samples, 8000));
 			} else {
 				ms = ast_waitfor(chan, ms);
 				if (ms < 0) {
@@ -214,18 +215,26 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 	}
 	close(fds[0]);
 	close(fds[1]);
-	LOCAL_USER_REMOVE(u);
+	
 	if (pid > -1)
 		kill(pid, SIGKILL);
 	if (!res && owriteformat)
 		ast_set_write_format(chan, owriteformat);
+
+	LOCAL_USER_REMOVE(u);
+	
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+	
+	return res;
 }

 int load_module(void)
--- a/apps/app_nbscat.c
+++ b/apps/app_nbscat.c
@@ -1,24 +1,28 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Silly application to play an NBScat file -- uses nbscat8k
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Silly application to play an NBScat file -- uses nbscat8k
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/frame.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
 #include <string.h>
 #include <stdio.h>
 #include <signal.h>
@@ -28,9 +32,26 @@
 #include <sys/time.h>
 #include <sys/socket.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/frame.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+
 #define LOCAL_NBSCAT "/usr/local/bin/nbscat8k"
 #define NBSCAT "/usr/bin/nbscat8k"

+#ifndef AF_LOCAL
+#define AF_LOCAL AF_UNIX
+#endif
+
 static char *tdesc = "Silly NBS Stream Application";

 static char *app = "NBScat";
@@ -39,8 +60,7 @@ static char *synopsis = "Play an NBS local stream";

 static char *descrip = 
 "  NBScat: Executes nbscat to listen to the local NBS stream.\n"
-"Returns  -1  on\n hangup or 0 otherwise. User can exit by \n"
-"pressing any key\n.";
+"User can exit by pressing any key\n.";

 STANDARD_LOCAL_USER;

@@ -90,52 +110,43 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 	int ms = -1;
 	int pid = -1;
 	int owriteformat;
-	struct timeval now, next;
+	struct timeval next;
 	struct ast_frame *f;
 	struct myframe {
 		struct ast_frame f;
 		char offset[AST_FRIENDLY_OFFSET];
 		short frdata[160];
 	} myf;
+	
+	LOCAL_USER_ADD(u);
+
 	if (socketpair(AF_LOCAL, SOCK_STREAM, 0, fds)) {
 		ast_log(LOG_WARNING, "Unable to create socketpair\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
-	LOCAL_USER_ADD(u);
+	
 	ast_stopstream(chan);

 	owriteformat = chan->writeformat;
 	res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
 	if (res < 0) {
 		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	
 	res = NBScatplay(fds[1]);
 	/* Wait 1000 ms first */
-	next = now;
+	next = ast_tvnow();
 	next.tv_sec += 1;
 	if (res >= 0) {
 		pid = res;
 		/* Order is important -- there's almost always going to be mp3...  we want to prioritize the
 		   user */
 		for (;;) {
-			gettimeofday(&now, NULL);
-			ms = (next.tv_sec - now.tv_sec) * 1000;
-			ms += (next.tv_usec - now.tv_usec) / 1000;
-#if 0
-			printf("ms: %d\n", ms);
-#endif			
+			ms = ast_tvdiff_ms(next, ast_tvnow());
 			if (ms <= 0) {
-#if 0
-				{
-					static struct timeval last;
-					struct timeval tv;
-					gettimeofday(&tv, NULL);
-					printf("Since last: %ld\n", (tv.tv_sec - last.tv_sec) * 1000 + (tv.tv_usec - last.tv_usec) / 1000);
-					last = tv;
-				}
-#endif
 				res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata));
 				if (res > 0) {
 					myf.f.frametype = AST_FRAME_VOICE;
@@ -157,14 +168,7 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 					res = 0;
 					break;
 				}
-				next.tv_usec += res / 2 * 125;
-				if (next.tv_usec >= 1000000) {
-					next.tv_usec -= 1000000;
-					next.tv_sec++;
-				}
-#if 0
-				printf("Next: %d\n", ms);
-#endif				
+				next = ast_tvadd(next, ast_samp2tv(myf.f.samples, 8000));
 			} else {
 				ms = ast_waitfor(chan, ms);
 				if (ms < 0) {
@@ -192,18 +196,26 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 	}
 	close(fds[0]);
 	close(fds[1]);
-	LOCAL_USER_REMOVE(u);
+	
 	if (pid > -1)
 		kill(pid, SIGKILL);
 	if (!res && owriteformat)
 		ast_set_write_format(chan, owriteformat);
+
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_osplookup.c
+++ b/apps/app_osplookup.c
@@ -1,33 +1,52 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Time of day - Report the time of day
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/options.h>
-#include <asterisk/config.h>
-#include <asterisk/module.h>
-#include <asterisk/utils.h>
-#include <asterisk/causes.h>
-#include <asterisk/astosp.h>
+/*! \file
+ *
+ * \brief Open Settlement Protocol Lookup
+ * 
+ * \ingroup applications
+ */
+
+#include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
-#include <stdlib.h>
 #include <ctype.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/options.h"
+#include "asterisk/config.h"
+#include "asterisk/module.h"
+#include "asterisk/utils.h"
+#include "asterisk/causes.h"
+#include "asterisk/astosp.h"
+#include "asterisk/app.h"
+#include "asterisk/options.h"
+
 static char *tdesc = "OSP Lookup";

 static char *app = "OSPLookup";
@@ -47,23 +66,33 @@ static char *descrip =
 " ${OSPHANDLE}: The OSP Handle for anything remaining\n"
 " ${OSPRESULTS}: The number of OSP results total remaining\n"
 "\n"
-"If the lookup was *not* successful and there exists a priority n + 101,\n"
-"then that priority will be taken next.\n" ;
+"The option string may contain the following character:\n"
+"	'j' -- jump to n+101 priority if the lookup was NOT successful\n"
+"This application sets the following channel variable upon completion:\n"
+"	OSPLOOKUPSTATUS	The status of the OSP Lookup attempt as a text string, one of\n"
+"		SUCCESS | FAILED \n";
+

 static char *descrip2 = 
-"  OSPNext:  Looks up the next OSP Destination for ${OSPHANDLE}\n"
+"  OSPNext(cause[|options]):  Looks up the next OSP Destination for ${OSPHANDLE}\n"
 "See OSPLookup for more information\n"
 "\n"
-"If the lookup was *not* successful and there exists a priority n + 101,\n"
-"then that priority will be taken next.\n" ;
+"The option string may contain the following character:\n"
+"	'j' -- jump to n+101 priority if the lookup was NOT successful\n"
+"This application sets the following channel variable upon completion:\n"
+"	OSPNEXTSTATUS	The status of the OSP Next attempt as a text string, one of\n"
+"		SUCCESS | FAILED \n";

 static char *descrip3 = 
-"  OSPFinish(status):  Records call state for ${OSPHANDLE}, according to\n"
-"status, which should be one of BUSY, CONGESTION, ANSWER, NOANSWER, or NOCHANAVAIL\n"
-"or coincidentally, just what the Dial application stores in its ${DIALSTATUS}\n"
+"  OSPFinish(status[|options]):  Records call state for ${OSPHANDLE}, according to\n"
+"status, which should be one of BUSY, CONGESTION, ANSWER, NOANSWER, or CHANUNAVAIL\n"
+"or coincidentally, just what the Dial application stores in its ${DIALSTATUS}.\n"
 "\n"
-"If the finishing was *not* successful and there exists a priority n + 101,\n"
-"then that priority will be taken next.\n" ;
+"The option string may contain the following character:\n"
+"	'j' -- jump to n+101 priority if the finish attempt was NOT successful\n"
+"This application sets the following channel variable upon completion:\n"
+"	OSPFINISHSTATUS	The status of the OSP Finish attempt as a text string, one of\n"
+"		SUCCESS | FAILED \n";

 STANDARD_LOCAL_USER;

@@ -92,43 +121,58 @@ static int osplookup_exec(struct ast_channel *chan, void *data)
 	int res=0;
 	struct localuser *u;
 	char *temp;
-	char *provider, *opts=NULL;
 	struct ast_osp_result result;
-	if (!data || ast_strlen_zero(data) || !(temp = ast_strdupa(data))) {
-		ast_log(LOG_WARNING, "OSPLookup requires an argument (extension)\n");
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(extension);
+		AST_APP_ARG(provider);
+		AST_APP_ARG(options);
+	);
+	
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "OSPLookup requires an argument OSPLookup(exten[|provider[|options]])\n");
 		return -1;
 	}
-	provider = strchr(temp, '|');
-	if (provider) {
-		*provider = '\0';
-		provider++;
-		opts = strchr(provider, '|');
-		if (opts) {
-			*opts = '\0';
-			opts++;
-		}
-	}
+
 	LOCAL_USER_ADD(u);
-	ast_log(LOG_DEBUG, "Whoo hoo, looking up OSP on '%s' via '%s'\n", temp, provider ? provider : "<default>");
-	if ((res = ast_osp_lookup(chan, provider, temp, chan->callerid, &result)) > 0) {
+
+	temp = ast_strdupa(data);
+	if (!temp) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args, temp);
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
+	ast_log(LOG_DEBUG, "Whoo hoo, looking up OSP on '%s' via '%s'\n", args.extension, args.provider ? args.provider : "<default>");
+	if ((res = ast_osp_lookup(chan, args.provider, args.extension, chan->cid.cid_num, &result)) > 0) {
 		char tmp[80];
 		snprintf(tmp, sizeof(tmp), "%d", result.handle);
-		pbx_builtin_setvar_helper(chan, "OSPHANDLE", tmp);
-		pbx_builtin_setvar_helper(chan, "OSPTECH", result.tech);
-		pbx_builtin_setvar_helper(chan, "OSPDEST", result.dest);
-		pbx_builtin_setvar_helper(chan, "OSPTOKEN", result.token);
+		pbx_builtin_setvar_helper(chan, "_OSPHANDLE", tmp);
+		pbx_builtin_setvar_helper(chan, "_OSPTECH", result.tech);
+		pbx_builtin_setvar_helper(chan, "_OSPDEST", result.dest);
+		pbx_builtin_setvar_helper(chan, "_OSPTOKEN", result.token);
 		snprintf(tmp, sizeof(tmp), "%d", result.numresults);
-		pbx_builtin_setvar_helper(chan, "OSPRESULTS", tmp);
+		pbx_builtin_setvar_helper(chan, "_OSPRESULTS", tmp);
+		pbx_builtin_setvar_helper(chan, "OSPLOOKUPSTATUS", "SUCCESS");
+
 	} else {
-		if (!res)
-			ast_log(LOG_NOTICE, "OSP Lookup failed for '%s' (provider '%s')\n", temp, provider ? provider : "<default>");
-		else
-			ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Lookup for '%s' (provider '%s')!\n", chan->name, temp, provider ? provider : "<default>" );
+		if (!res) {
+			ast_log(LOG_NOTICE, "OSP Lookup failed for '%s' (provider '%s')\n", args.extension, args.provider ? args.provider : "<default>");
+			pbx_builtin_setvar_helper(chan, "OSPLOOKUPSTATUS", "FAILED");
+		} else
+			ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Lookup for '%s' (provider '%s')!\n", chan->name, args.extension, args.provider ? args.provider : "<default>" );
 	}
 	if (!res) {
 		/* Look for a "busy" place */
-		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-			chan->priority += 100;
+		if (priority_jump || option_priority_jumping)
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
 	} else if (res > 0)
 		res = 0;
 	LOCAL_USER_REMOVE(u);
@@ -142,23 +186,47 @@ static int ospnext_exec(struct ast_channel *chan, void *data)
 	char *temp;
 	int cause;
 	struct ast_osp_result result;
-	if (!data || ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "OSPNext should have an argument (cause)\n");
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(cause);
+		AST_APP_ARG(options);
+	);
+	
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "OSPNext should have an argument (cause[|options])\n");
+		return -1;
 	}
+
 	LOCAL_USER_ADD(u);
-	cause = str2cause((char *)data);
+
+	temp = ast_strdupa(data);
+	if (!temp) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args, temp);
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
+	cause = str2cause(args.cause);
 	temp = pbx_builtin_getvar_helper(chan, "OSPHANDLE");
 	result.handle = -1;
-	if (temp && strlen(temp) && (sscanf(temp, "%i", &result.handle) == 1) && (result.handle > -1)) {
+	if (!ast_strlen_zero(temp) && (sscanf(temp, "%d", &result.handle) == 1) && (result.handle > -1)) {
 		if ((res = ast_osp_next(&result, cause)) > 0) {
 			char tmp[80];
 			snprintf(tmp, sizeof(tmp), "%d", result.handle);
-			pbx_builtin_setvar_helper(chan, "OSPHANDLE", tmp);
-			pbx_builtin_setvar_helper(chan, "OSPTECH", result.tech);
-			pbx_builtin_setvar_helper(chan, "OSPDEST", result.dest);
-			pbx_builtin_setvar_helper(chan, "OSPTOKEN", result.token);
+			pbx_builtin_setvar_helper(chan, "_OSPHANDLE", tmp);
+			pbx_builtin_setvar_helper(chan, "_OSPTECH", result.tech);
+			pbx_builtin_setvar_helper(chan, "_OSPDEST", result.dest);
+			pbx_builtin_setvar_helper(chan, "_OSPTOKEN", result.token);
 			snprintf(tmp, sizeof(tmp), "%d", result.numresults);
-			pbx_builtin_setvar_helper(chan, "OSPRESULTS", tmp);
+			pbx_builtin_setvar_helper(chan, "_OSPRESULTS", tmp);
+			pbx_builtin_setvar_helper(chan, "OSPNEXTSTATUS", "SUCCESS");
 		}
 	} else {
 		if (!res) {
@@ -166,13 +234,14 @@ static int ospnext_exec(struct ast_channel *chan, void *data)
 				ast_log(LOG_NOTICE, "OSP Lookup Next failed for handle '%d'\n", result.handle);
 			else
 				ast_log(LOG_DEBUG, "No OSP handle specified\n");
+			pbx_builtin_setvar_helper(chan, "OSPNEXTSTATUS", "FAILED");	
 		} else
 			ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Next!\n", chan->name);
 	}
 	if (!res) {
 		/* Look for a "busy" place */
-		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-			chan->priority += 100;
+		if (priority_jump || option_priority_jumping)
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
 	} else if (res > 0)
 		res = 0;
 	LOCAL_USER_REMOVE(u);
@@ -187,9 +256,33 @@ static int ospfinished_exec(struct ast_channel *chan, void *data)
 	int cause;
 	time_t start=0, duration=0;
 	struct ast_osp_result result;
-	if (!data || ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "OSPFinish should have an argument (cause)\n");
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(status);
+		AST_APP_ARG(options);
+	);
+	
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "OSPFinish should have an argument (status[|options])\n");
+		return -1;
 	}
+
+	LOCAL_USER_ADD(u);
+
+	temp = ast_strdupa(data);
+	if (!temp) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args, temp);
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
 	if (chan->cdr) {
 		start = chan->cdr->answer.tv_sec;
 		if (start)
@@ -198,13 +291,14 @@ static int ospfinished_exec(struct ast_channel *chan, void *data)
 			duration = 0;
 	} else
 		ast_log(LOG_WARNING, "OSPFinish called on channel '%s' with no CDR!\n", chan->name);
-	LOCAL_USER_ADD(u);
-	cause = str2cause((char *)data);
+	
+	cause = str2cause(args.status);
 	temp = pbx_builtin_getvar_helper(chan, "OSPHANDLE");
 	result.handle = -1;
-	if (temp && strlen(temp) && (sscanf(temp, "%i", &result.handle) == 1) && (result.handle > -1)) {
+	if (!ast_strlen_zero(temp) && (sscanf(temp, "%d", &result.handle) == 1) && (result.handle > -1)) {
 		if (!ast_osp_terminate(result.handle, cause, start, duration)) {
-			pbx_builtin_setvar_helper(chan, "OSPHANDLE", "");
+			pbx_builtin_setvar_helper(chan, "_OSPHANDLE", "");
+			pbx_builtin_setvar_helper(chan, "OSPFINISHSTATUS", "SUCCESS");
 			res = 1;
 		}
 	} else {
@@ -213,13 +307,14 @@ static int ospfinished_exec(struct ast_channel *chan, void *data)
 				ast_log(LOG_NOTICE, "OSP Finish failed for handle '%d'\n", result.handle);
 			else
 				ast_log(LOG_DEBUG, "No OSP handle specified\n");
+			pbx_builtin_setvar_helper(chan, "OSPFINISHSTATUS", "FAILED");
 		} else
 			ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Terminate!\n", chan->name);
 	}
 	if (!res) {
 		/* Look for a "busy" place */
-		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-			chan->priority += 100;
+		if (priority_jump || option_priority_jumping)
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
 	} else if (res > 0)
 		res = 0;
 	LOCAL_USER_REMOVE(u);
@@ -230,26 +325,25 @@ static int ospfinished_exec(struct ast_channel *chan, void *data)
 int unload_module(void)
 {
 	int res;
-	STANDARD_HANGUP_LOCALUSERS;
+	
 	res = ast_unregister_application(app3);
 	res |= ast_unregister_application(app2);
 	res |= ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
 	return res;
 }

 int load_module(void)
 {
 	int res;
+	
 	res = ast_register_application(app, osplookup_exec, synopsis, descrip);
-	if (res)
-		return(res);
-	res = ast_register_application(app2, ospnext_exec, synopsis2, descrip2);
-	if (res)
-		return(res);
-	res = ast_register_application(app3, ospfinished_exec, synopsis3, descrip3);
-	if (res)
-		return(res);
-	return(0);
+	res |= ast_register_application(app2, ospnext_exec, synopsis2, descrip2);
+	res |= ast_register_application(app3, ospfinished_exec, synopsis3, descrip3);
+	
+	return res;
 }

 int reload(void)
--- a/apps/app_page.c
+++ b/apps/app_page.c
@@ -0,0 +1,206 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (c) 2004 - 2005 Digium, Inc.  All rights reserved.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * This code is released under the GNU General Public License
+ * version 2.0.  See LICENSE for more information.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ *
+ * \brief page() - Paging application
+ *
+ * \ingroup applications
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/options.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/file.h"
+#include "asterisk/app.h"
+
+
+static const char *tdesc = "Page Multiple Phones";
+
+static const char *app_page= "Page";
+
+static const char *page_synopsis = "Pages phones";
+
+static const char *page_descrip =
+"Page(Technology/Resource&Technology2/Resource2[|options])\n"
+"  Places outbound calls to the given technology / resource and dumps\n"
+"them into a conference bridge as muted participants.  The original\n"
+"caller is dumped into the conference as a speaker and the room is\n"
+"destroyed when the original caller leaves.  Valid options are:\n"
+"        d - full duplex audio\n"
+"	 q - quiet, do not play beep to caller\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+enum {
+	PAGE_DUPLEX = (1 << 0),
+	PAGE_QUIET = (1 << 1),
+} page_opt_flags;
+
+AST_APP_OPTIONS(page_opts, {
+	AST_APP_OPTION('d', PAGE_DUPLEX),
+	AST_APP_OPTION('q', PAGE_QUIET),
+});
+
+struct calloutdata {
+	char cidnum[64];
+	char cidname[64];
+	char tech[64];
+	char resource[256];
+	char meetmeopts[64];
+};
+
+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);
+	free(cd);
+	return NULL;
+}
+
+static void launch_page(struct ast_channel *chan, const char *meetmeopts, const char *tech, const char *resource)
+{
+	struct calloutdata *cd;
+	pthread_t t;
+	pthread_attr_t attr;
+	cd = malloc(sizeof(struct calloutdata));
+	if (cd) {
+		memset(cd, 0, sizeof(struct calloutdata));
+		ast_copy_string(cd->cidnum, chan->cid.cid_num ? chan->cid.cid_num : "", sizeof(cd->cidnum));
+		ast_copy_string(cd->cidname, chan->cid.cid_name ? chan->cid.cid_name : "", sizeof(cd->cidname));
+		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));
+		pthread_attr_init(&attr);
+		pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
+		if (ast_pthread_create(&t, &attr, page_thread, cd)) {
+			ast_log(LOG_WARNING, "Unable to create paging thread: %s\n", strerror(errno));
+			free(cd);
+		}
+	}
+}
+
+static int page_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	char *options;
+	char *tech, *resource;
+	char meetmeopts[80];
+	struct ast_flags flags = { 0 };
+	unsigned int confid = rand();
+	struct ast_app *app;
+	char *tmp;
+	int res=0;
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	if (!(app = pbx_findapp("MeetMe"))) {
+		ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	};
+
+	options = ast_strdupa(data);
+	if (!options) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	tmp = strsep(&options, "|");
+	if (options)
+		ast_app_parse_options(page_opts, &flags, NULL, options);
+
+	snprintf(meetmeopts, sizeof(meetmeopts), "%ud|%sqxdw", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m");
+	while ((tech = strsep(&tmp, "&"))) {
+		if ((resource = strchr(tech, '/'))) {
+			*resource++ = '\0';
+			launch_page(chan, meetmeopts, tech, resource);
+		} else {
+			ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
+		}
+	}
+	if (!ast_test_flag(&flags, PAGE_QUIET)) {
+		res = ast_streamfile(chan, "beep", chan->language);
+		if (!res)
+			res = ast_waitstream(chan, "");
+	}
+	if (!res) {
+		snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%sqxd", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t");
+		pbx_exec(chan, app, meetmeopts, 1);
+	}
+
+	LOCAL_USER_REMOVE(u);
+
+	return -1;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res =  ast_unregister_application(app_page);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
+}
+
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_parkandannounce.c
+++ b/apps/app_parkandannounce.c
@@ -1,36 +1,51 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
 *
- * ParkAndAnnounce application for Asterisk
 * Author: Ben Miller <bgmiller@dccinc.com>
 *    With TONS of help from Mark!
- * 
- * Asterisk is Copyrighted as follows
- * Copyright (C) 1999, Mark Spencer
 *
- * Mark Spencer <markster@linux-support.net>
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
 */

-#include <sys/types.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/channel_pvt.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/features.h>
-#include <asterisk/options.h>
-#include <asterisk/logger.h>
-#include <asterisk/say.h>
-#include <asterisk/lock.h>
+/*! \file
+ *
+ * \brief ParkAndAnnounce application for Asterisk
+ * 
+ * \ingroup applications
+ */

 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
+#include <unistd.h>
+#include <sys/types.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/features.h"
+#include "asterisk/options.h"
+#include "asterisk/logger.h"
+#include "asterisk/say.h"
+#include "asterisk/lock.h"

 static char *tdesc = "Call Parking and Announce Application";

@@ -64,18 +79,23 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
 	char *s,*orig_s;

 	struct ast_channel *dchan;
+	struct outgoing_helper oh;
 	int outstate;

 	struct localuser *u;
-	if(!data || (data && !strlen(data))) {
+
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "ParkAndAnnounce requires arguments: (announce:template|timeout|dial|[return_context])\n");
 		return -1;
 	}
  
+	LOCAL_USER_ADD(u);
+
 	l=strlen(data)+2;
 	orig_s=malloc(l);
 	if(!orig_s) {
 		ast_log(LOG_WARNING, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	s=orig_s;
@@ -85,6 +105,7 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
 	if(! template) {
 		ast_log(LOG_WARNING, "PARK: An announce template must be defined\n");
 		free(orig_s);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
  
@@ -96,6 +117,7 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
 	if(!dial) {
 		ast_log(LOG_WARNING, "PARK: A dial resource must be specified i.e: Console/dsp or Zap/g1/5551212\n");
 		free(orig_s);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	} else {
 		dialtech=strsep(&dial, "/");
@@ -128,6 +150,7 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
 	if(atoi(priority) < 0) {
 		ast_log(LOG_WARNING, "Priority '%s' must be a number > 0\n", priority);
 		free(orig_s);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	/* At this point we have a priority and maybe an extension and a context */
@@ -140,16 +163,13 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
 		chan->priority++;
 	}

-
 	if(option_verbose > 2) {
-		ast_verbose( VERBOSE_PREFIX_3 "Return Context: (%s,%s,%d) ID: %s\n", chan->context,chan->exten, chan->priority, chan->callerid);
-		if(!ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->callerid)) {
+		ast_verbose( VERBOSE_PREFIX_3 "Return Context: (%s,%s,%d) ID: %s\n", chan->context,chan->exten, chan->priority, chan->cid.cid_num);
+		if(!ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
 			ast_verbose( VERBOSE_PREFIX_3 "Warning: Return Context Invalid, call will return to default|s\n");
 		}
 	}
  
-	LOCAL_USER_ADD(u);
-
 	/* we are using masq_park here to protect * from touching the channel once we park it.  If the channel comes out of timeout
 	before we are done announcing and the channel is messed with, Kablooeee.  So we use Masq to prevent this.  */

@@ -161,7 +181,9 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)

 	/* Now place the call to the extention */

-	dchan = ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, dialstr,30000, &outstate, chan->callerid);
+	memset(&oh, 0, sizeof(oh));
+	oh.parent_channel = chan;
+	dchan = __ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, dialstr,30000, &outstate, chan->cid.cid_num, chan->cid.cid_name, &oh);

 	if(dchan) {
 		if(dchan->_state == AST_STATE_UP) {
@@ -215,9 +237,10 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)

 	ast_stopstream(dchan);  
 	ast_hangup(dchan);
-
-	LOCAL_USER_REMOVE(u);
 	free(orig_s);
+	
+	LOCAL_USER_REMOVE(u);
+	
 	return res;
 }

@@ -225,8 +248,13 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
--- a/apps/app_playback.c
+++ b/apps/app_playback.c
@@ -1,43 +1,68 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Trivial application to playback a sound file
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Trivial application to playback a sound file
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/utils.h>
 #include <string.h>
 #include <stdlib.h>
+#include <stdio.h>

-static char *tdesc = "Trivial Playback Application";
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/utils.h"
+#include "asterisk/options.h"
+#include "asterisk/app.h"
+
+static char *tdesc = "Sound File Playback Application";

 static char *app = "Playback";

 static char *synopsis = "Play a file";

 static char *descrip = 
-"  Playback(filename[|option]):  Plays  back  a  given  filename (do not put\n"
-"extension). Options may also be  included following a pipe symbol. The 'skip'\n"
-"option causes the playback of the message to  be  skipped  if  the  channel\n"
-"is not in the 'up' state (i.e. it hasn't been  answered  yet. If 'skip' is \n"
+"  Playback(filename[&filename2...][|option]):  Plays back given filenames (do not put\n"
+"extension). Options may also be included following a pipe symbol. The 'skip'\n"
+"option causes the playback of the message to be skipped if the channel\n"
+"is not in the 'up' state (i.e. it hasn't been  answered  yet). If 'skip' is \n"
 "specified, the application will return immediately should the channel not be\n"
-"off hook.  Otherwise, unless 'noanswer' is specified, the channel channel will\n"
+"off hook.  Otherwise, unless 'noanswer' is specified, the channel will\n"
 "be answered before the sound is played. Not all channels support playing\n"
-"messages while still hook. Returns -1 if the channel was hung up, or if the\n"
-"file does not exist. Returns 0 otherwise.\n";
+"messages while still on hook. If 'j' is specified, the application\n"
+"will jump to priority n+101 if present when a file specified to be played\n"
+"does not exist.\n"
+"This application sets the following channel variable upon completion:\n"
+" PLAYBACKSTATUS    The status of the playback attempt as a text string, one of\n"
+"               SUCCESS | FAILED\n"
+;

 STANDARD_LOCAL_USER;

@@ -45,26 +70,43 @@ LOCAL_USER_DECL;

 static int playback_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
+	int res = 0, mres = 0;
 	struct localuser *u;
-	char tmp[256];
-	char *options;
+	char *tmp = NULL;
 	int option_skip=0;
 	int option_noanswer = 0;
-	char *stringp;
-	if (!data || ast_strlen_zero((char *)data)) {
+	char *front = NULL, *back = NULL;
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(filenames);
+		AST_APP_ARG(options);
+	);
+	
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Playback requires an argument (filename)\n");
 		return -1;
 	}
-	strncpy(tmp, (char *)data, sizeof(tmp)-1);
-	stringp=tmp;
-	strsep(&stringp, "|");
-	options = strsep(&stringp, "|");
-	if (options && !strcasecmp(options, "skip"))
-		option_skip = 1;
-	if (options && !strcasecmp(options, "noanswer"))
-		option_noanswer = 1;
+
 	LOCAL_USER_ADD(u);
+
+	tmp = ast_strdupa(data);
+	if (!tmp) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;	
+	}
+
+	AST_STANDARD_APP_ARGS(args, tmp);
+
+	if (args.options) {
+		if (strcasestr(args.options, "skip"))
+			option_skip = 1;
+		if (strcasestr(args.options, "noanswer"))
+			option_noanswer = 1;
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+	
 	if (chan->_state != AST_STATE_UP) {
 		if (option_skip) {
 			/* At the user's option, skip if the line is not up */
@@ -76,14 +118,29 @@ static int playback_exec(struct ast_channel *chan, void *data)
 	}
 	if (!res) {
 		ast_stopstream(chan);
-		res = ast_streamfile(chan, tmp, chan->language);
-		if (!res) 
-			res = ast_waitstream(chan, "");
-		else {
-			ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
-			res = 0;
+		front = tmp;
+		while (!res && front) {
+			if ((back = strchr(front, '&'))) {
+				*back = '\0';
+				back++;
+			}
+			res = ast_streamfile(chan, front, chan->language);
+			if (!res) { 
+				res = ast_waitstream(chan, "");	
+				ast_stopstream(chan);
+			} else {
+				ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
+				if (priority_jump || option_priority_jumping)
+					ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+				res = 0;
+				mres = 1;
+			}
+			front = back;
 		}
-		ast_stopstream(chan);
+		if (mres)
+			pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "FAILED");
+		else
+			pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "SUCCESS");
 	}
 	LOCAL_USER_REMOVE(u);
 	return res;
@@ -91,8 +148,13 @@ static int playback_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_privacy.c
+++ b/apps/app_privacy.c
@@ -1,30 +1,49 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Block all calls without Caller*ID, require phone # to be entered
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Block all calls without Caller*ID, require phone # to be entered
+ * 
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/options.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/image.h>
-#include <asterisk/callerid.h>
-#include <asterisk/app.h>
-#include <asterisk/config.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/utils.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/callerid.h"
+#include "asterisk/app.h"
+#include "asterisk/config.h"

 #define PRIV_CONFIG "privacy.conf"

@@ -35,40 +54,56 @@ static char *app = "PrivacyManager";
 static char *synopsis = "Require phone number to be entered, if no CallerID sent";

 static char *descrip =
-  "  PrivacyManager: If no Caller*ID is sent, PrivacyManager answers the\n"
-  "channel and asks the caller to enter their 10 digit phone number.\n"
-  "The caller is given 3 attempts.  If after 3 attempts, they do not enter\n"
-  "their 10 digit phone number, and if there exists a priority n + 101,\n"
-  "where 'n' is the priority of the current instance, then  the\n"
-  "channel  will  be  setup  to continue at that priority level.\n"
-  "Otherwise, it returns 0.  Does nothing if Caller*ID was received on the\n"
-  "channel.\n";
+  "  PrivacyManager([maxretries[|minlength[|options]]]): If no Caller*ID \n"
+  "is sent, PrivacyManager answers the channel and asks the caller to\n"
+  "enter their phone number. The caller is given 3 attempts to do so.\n"
+  "The application does nothing if Caller*ID was received on the channel.\n"
+  "  Configuration file privacy.conf contains two variables:\n"
+  "   maxretries  default 3  -maximum number of attempts the caller is allowed \n"
+  "               to input a callerid.\n"
+  "   minlength   default 10 -minimum allowable digits in the input callerid number.\n"
+  "If you don't want to use the config file and have an i/o operation with\n"
+  "every call, you can also specify maxretries and minlength as application\n"
+  "parameters. Doing so supercedes any values set in privacy.conf.\n"
+  "The option string may contain the following character: \n"
+  "  'j' -- jump to n+101 priority after <maxretries> failed attempts to collect\n"
+  "         the minlength number of digits.\n"
+  "The application sets the following channel variable upon completion: \n"
+  "PRIVACYMGRSTATUS  The status of the privacy manager's attempt to collect \n"
+  "                  a phone number from the user. A text string that is either:\n" 
+  "          SUCCESS | FAILED \n"
+;

 STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

-static int
-privacy_exec (struct ast_channel *chan, void *data)
+
+
+static int privacy_exec (struct ast_channel *chan, void *data)
 {
 	int res=0;
 	int retries;
 	int maxretries = 3;
-	int x;
+	int minlength = 10;
+	int x = 0;
 	char *s;
-	char phone[10];
-	char new_cid[144];
+	char phone[30];
 	struct localuser *u;
-	struct ast_config *cfg;
+	struct ast_config *cfg = NULL;
+	char *parse = NULL;
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(maxretries);
+		AST_APP_ARG(minlength);
+		AST_APP_ARG(options);
+	);

 	LOCAL_USER_ADD (u);
-	if (chan->callerid)
-	{
+	if (!ast_strlen_zero(chan->cid.cid_num)) {
 		if (option_verbose > 2)
 			ast_verbose (VERBOSE_PREFIX_3 "CallerID Present: Skipping\n");
-	}
-	else
-	{
+	} else {
 		/*Answer the channel if it is not already*/
 		if (chan->_state != AST_STATE_UP) {
 			res = ast_answer(chan);
@@ -77,9 +112,55 @@ privacy_exec (struct ast_channel *chan, void *data)
 				return -1;
 			}
 		}
-		/*Read in the config file*/
-		cfg = ast_load(PRIV_CONFIG);
+
+		if (!ast_strlen_zero((char *)data))
+		{
+			parse = ast_strdupa(data);
+			if (!parse) {
+				ast_log(LOG_ERROR, "Out of memory!\n");
+				LOCAL_USER_REMOVE(u);
+				return -1;
+			}
+			
+			AST_STANDARD_APP_ARGS(args, parse);
+
+			if (args.maxretries) {
+				if (sscanf(args.maxretries, "%d", &x) == 1)
+					maxretries = x;
+				else
+					ast_log(LOG_WARNING, "Invalid max retries argument\n");
+			}
+			if (args.minlength) {
+				if (sscanf(args.minlength, "%d", &x) == 1)
+					minlength = x;
+				else
+					ast_log(LOG_WARNING, "Invalid min length argument\n");
+			}
+			if (args.options)
+				if (strchr(args.options, 'j'))
+					priority_jump = 1;
+
+		}		
+
+		if (!x)
+		{
+			/*Read in the config file*/
+			cfg = ast_config_load(PRIV_CONFIG);
 		
+			if (cfg && (s = ast_variable_retrieve(cfg, "general", "maxretries"))) {
+				if (sscanf(s, "%d", &x) == 1) 
+					maxretries = x;
+				else
+					ast_log(LOG_WARNING, "Invalid max retries argument\n");
+        		}
+
+			if (cfg && (s = ast_variable_retrieve(cfg, "general", "minlength"))) {
+				if (sscanf(s, "%d", &x) == 1) 
+					minlength = x;
+				else
+					ast_log(LOG_WARNING, "Invalid min length argument\n");
+			}
+		}	
 		
 		/*Play unidentified call*/
 		res = ast_safe_sleep(chan, 1000);
@@ -88,23 +169,21 @@ privacy_exec (struct ast_channel *chan, void *data)
 		if (!res)
 			res = ast_waitstream(chan, "");

-        if (cfg && (s = ast_variable_retrieve(cfg, "general", "maxretries"))) {
-                if (sscanf(s, "%d", &x) == 1) {
-                        maxretries = x;
-                } else {
-                        ast_log(LOG_WARNING, "Invalid max retries argument\n");
-                }
-        }
-			
 		/*Ask for 10 digit number, give 3 attempts*/
 		for (retries = 0; retries < maxretries; retries++) {
 			if (!res)
-				res = ast_app_getdata(chan, "privacy-prompt", phone, sizeof(phone), 0);
+				res = ast_streamfile(chan, "privacy-prompt", chan->language);
+			if (!res)
+				res = ast_waitstream(chan, "");
+
+			if (!res ) 
+				res = ast_readstring(chan, phone, sizeof(phone) - 1, /* digit timeout ms */ 3200, /* first digit timeout */ 5000, "#");
+
 			if (res < 0)
 				break;

-			/*Make sure we get 10 digits*/
-			if (strlen(phone) == 10) 
+			/*Make sure we get at least digits*/
+			if (strlen(phone) >= minlength ) 
 				break;
 			else {
 				res = ast_streamfile(chan, "privacy-incorrect", chan->language);
@@ -114,21 +193,21 @@ privacy_exec (struct ast_channel *chan, void *data)
 		}
 		
 		/*Got a number, play sounds and send them on their way*/
-		if ((retries < maxretries) && !res) {
+		if ((retries < maxretries) && res == 1 ) {
 			res = ast_streamfile(chan, "privacy-thankyou", chan->language);
 			if (!res)
 				res = ast_waitstream(chan, "");
-			snprintf (new_cid, sizeof (new_cid), "\"%s\" <%s>", "Privacy Manager", phone);
-			ast_set_callerid (chan, new_cid, 0);
+			ast_set_callerid (chan, phone, "Privacy Manager", NULL);
 			if (option_verbose > 2)
-				ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID to %s\n",new_cid);
+				ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID to %s\n",phone);
+			pbx_builtin_setvar_helper(chan, "PRIVACYMGRSTATUS", "SUCCESS");
 		} else {
-			/*Send the call to n+101 priority, where n is the current priority*/
-			if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-				chan->priority+=100;
+			if (priority_jump || option_priority_jumping)	
+				ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+			pbx_builtin_setvar_helper(chan, "PRIVACYMGRSTATUS", "FAILED");
 		}
 		if (cfg) 
-			ast_destroy(cfg);
+			ast_config_destroy(cfg);
 	}

  LOCAL_USER_REMOVE (u);
@@ -138,8 +217,13 @@ privacy_exec (struct ast_channel *chan, void *data)
 int
 unload_module (void)
 {
-  STANDARD_HANGUP_LOCALUSERS;
-  return ast_unregister_application (app);
+	int res;
+
+	res = ast_unregister_application (app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }

 int
--- a/apps/app_qcall.c
+++ b/apps/app_qcall.c
@@ -1,394 +0,0 @@
-/** @file app_qcall.c 
- *
- * Asterisk -- A telephony toolkit for Linux.
- *
- * Call back a party and connect them to a running pbx thread
- * 
- * Copyright (C) 1999, Mark Spencer
- *
- * Mark Spencer <markster@linux-support.net>
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License
- *
- * Call a user from a file contained within a queue (/var/spool/asterisk/qcall)
- * 
- * The queue is a directory containing files with the call request information
- * as a single line of text as follows:
- * 
- * Dialstring Caller-ID Extension Maxsecs [Identifier] [Required-response]
- *
- *  Dialstring -- A Dial String (The number to be called) in the
- *  format Technology/Number, such IAX/mysys/1234 or Zap/g1/1234
- * 
- *  Caller-ID -- A Standard nomalized representation of the Caller-ID of
- *  the number being dialed (generally 10 digits in the US). Leave as
- *  "asreceived" to use the default Caller*ID
- *
- *  Extension -- The Extension (optionally Extension@context) that the
- *  user should be "transferred" to after acceptance of the call.
- *
- *  Maxsecs -- The Maximum time of the call in seconds. Specify 0 for infinite.
- *
- *  Identifier -- The "Identifier" of the request. This is used to determine
- *  the names of the audio prompt files played. The first prompt, the one that
- *  asks for the input, is just the exact string specified as the identifier.
- *  The second prompt, the one that is played after the correct input is given,
- *  (generally a "thank you" recording), is the specified string with "-ok" 
- *  added to the end. So, if you specify "foo" as the identifier, your first
- *  prompt file that will be played will be "foo" and the second one will be
- *  "foo-ok".  If omitted no prompt is given
- *
- *  Required-Response (Optional) -- Specify a digit string to be used as the
- *  acceptance "code" if you desire it to be something other then "1". This
- *  can be used to implement some sort of PIN or security system. It may be
- *  more then a single character.
- *
- * NOTE: It is important to remember that the process that creates these
- * files needs keep and maintain a write lock (using flock with the LOCK_EX
- * option) when writing these files.
- *
- */
- 
-#include <asterisk/lock.h>
-#include <asterisk/utils.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/options.h>
-#include <stdio.h>
-#include <unistd.h>
-#include <string.h>
-#include <stdlib.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <errno.h>
-#include <dirent.h>
-#include <ctype.h>
-#include <sys/stat.h>
-#include <sys/time.h>
-#include <sys/file.h>
-#include "../astconf.h"
-
-static char qdir[255];
-static  char *tdesc = "Call from Queue";
-static  pthread_t qcall_thread;
-static int debug = 0;
-STANDARD_LOCAL_USER;
-LOCAL_USER_DECL;
-
-#define	OLDESTOK	14400		/* not any more then this number of secs old */
-#define	INITIALONE	1		/* initial wait before the first one in secs */
-#define	NEXTONE		600		/* wait before trying it again in secs */
-#define	MAXWAITFORANSWER 45000		/* max call time before answer */
-/* define either one or both of these two if your application requires it */
-#if	0
-#define	ACCTCODE	"SOMETHING"	/* Account code */
-#define	AMAFLAGS AST_CDR_BILLING	/* AMA flags */
-#endif
-/* define this if you want to have a particular CLID display on the user's
-   phone when they receive the call */
-#if	0
-#define	OURCLID	"2564286275"		/* The callerid to be displayed when calling */
-#endif
-
-static void *qcall_do(void *arg);
-
-static void *qcall(void *ignore)
-{
-pthread_t dialer_thread;
-DIR *dirp;
-FILE *fp;
-struct dirent *dp;
-char fname[80];
-struct stat mystat;
-time_t	t;
-void *arg;
-pthread_attr_t attr;
-
-	time(&t);
-	if (debug) printf("@@@@ qcall starting at %s",ctime(&t));
-	for(;;)
-	   {
-		time(&t);
-		dirp = opendir(qdir);
-		if (!dirp)
-		   {
-			perror("app_qcall:Cannot open queue directory");
-			break;
-		   }
-		while((dp = readdir(dirp)) != NULL)
-		   {
-			if (dp->d_name[0] == '.') continue;
-			snprintf(fname, sizeof(fname), "%s/%s", qdir, dp->d_name);
-			if (stat(fname,&mystat) == -1)
-			   {
-				perror("app_qcall:stat");
-				fprintf(stderr,"%s\n",fname);
-				continue;
-			   }
-			  /* if not a regular file, skip it */
-			if ((mystat.st_mode & S_IFMT) != S_IFREG) continue;
-			  /* if not yet .... */
-			if (mystat.st_atime == mystat.st_mtime)
-			   {  /* first time */
-				if ((mystat.st_atime + INITIALONE) > t) 
-					continue;
-			   }
-			else
-			   { /* already looked at once */
-				if ((mystat.st_atime + NEXTONE) > t) continue;
-			   }
-			  /* if too old */
-			if (mystat.st_mtime < (t - OLDESTOK))
-			   {
-				/* kill it, its too old */
-				unlink(fname);
-				continue;
-			   }				
-			 /* "touch" file's access time */
-			fp = fopen(fname,"r");
-			if (fp) fclose(fp);
-			/* make a copy of the filename string, so that we
-				may go on and use the buffer */
-			arg = (void *) strdup(fname);
-		        pthread_attr_init(&attr);
- 		        pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
-			if (ast_pthread_create(&dialer_thread,&attr,qcall_do,arg) == -1)
-			   {
-				perror("qcall: Cannot create thread");
-				continue;
-			   }
-		   }
-		closedir(dirp);
-		sleep(1);
-	   }
-	pthread_exit(NULL);
-}
-	
-/* single thread with one file (request) to dial */
-static void *qcall_do(void *arg)
-{
-	char fname[300] = "";
-	char dialstr[300];
-	char extstr[300];
-	char ident[300] = "";
-	char reqinp[300] = "";
-	char buf[300];
-	char clid[300],*tele,*context;
-	FILE *fp;
-	int ms = MAXWAITFORANSWER,maxsecs;
-	struct ast_channel *channel;
-	time_t	t;
-
-	  /* get the filename from the arg */
-	strncpy(fname,(char *)arg, sizeof(fname) - 1);
-	free(arg);
-	time(&t);
-	fp = fopen(fname,"r");
-	if (!fp) /* if cannot open request file */
-	   {
-		perror("qcall_do:fopen");
-		fprintf(stderr,"%s\n",fname);
-		unlink(fname);
-		pthread_exit(NULL);
-	   }
-	/* lock the file */
-	if (flock(fileno(fp),LOCK_EX) == -1)
-	   {
-		perror("qcall_do:flock");
-		fprintf(stderr,"%s\n",fname);
-		pthread_exit(NULL);
-	   }
-	/* default required input for acknowledgement */
-	reqinp[0] = '1';
-	reqinp[1] = '\0';
-	/* default no ident */
-	ident[0] = '\0';  /* default no ident */
-	if (fscanf(fp,"%s %s %s %d %s %s",dialstr,clid,
-		extstr,&maxsecs,ident,reqinp) < 4)
-	   {
-		fprintf(stderr,"qcall_do:file line invalid in file %s:\n",fname);
-		pthread_exit(NULL);
-	   }
-	flock(fileno(fp),LOCK_UN);
-	fclose(fp);
-	tele = strchr(dialstr,'/');
-	if (!tele)
-	   {
-		fprintf(stderr,"qcall_do:Dial number must be in format tech/number\n");
-		unlink(fname);
-		pthread_exit(NULL);
-	   }
-	*tele++ = 0;
-	channel = ast_request(dialstr,AST_FORMAT_SLINEAR,tele);
-	if (channel)
-	   {
-		ast_set_read_format(channel,AST_FORMAT_SLINEAR);
-		ast_set_write_format(channel,AST_FORMAT_SLINEAR);
-#ifdef	OURCLID
-		if (channel->callerid)
-			free(channel->callerid);
-		channel->callerid = strdup(OURCLID);
-		if (channel->ani)
-			free(channel->ani);
-		channel->ani = strdup(OURCLID);
-#endif		
-		channel->whentohangup = 0;
-		channel->appl = "AppQcall";
-		channel->data = "(Outgoing Line)";
-		if (option_verbose > 2)
-			ast_verbose(VERBOSE_PREFIX_3 "Qcall initiating call to %s/%s on %s (%s)\n",
-				dialstr,tele,channel->name,fname);
-		ast_call(channel,tele,MAXWAITFORANSWER);
-	   }
-	else
-	   {
-		fprintf(stderr,"qcall_do:Sorry unable to obtain channel\n");
-		pthread_exit(NULL);
-	   }
-	if (strcasecmp(clid, "asreceived")) {
-		if (channel->callerid) free(channel->callerid);
-		channel->callerid = NULL;
-		if (channel->ani) free(channel->ani);
-		channel->ani = NULL;
-	}
-	if (channel->_state == AST_STATE_UP)
-	if (debug) printf("@@@@ Autodial:Line is Up\n");
-	if (option_verbose > 2)
-	ast_verbose(VERBOSE_PREFIX_3 "Qcall waiting for answer on %s\n",
-		channel->name);
-	while(ms > 0){
-		struct ast_frame *f;
-		ms = ast_waitfor(channel,ms);
-		f = ast_read(channel);
-		if (!f)
-		   {
-			if (debug) printf("@@@@ qcall_do:Hung Up\n");
-			unlink(fname);
-			break;
-		   }
-		if (f->frametype == AST_FRAME_CONTROL)
-		   {
-			if (f->subclass == AST_CONTROL_HANGUP)
-			   {
-				if (debug) printf("@@@@ qcall_do:Hung Up\n");
-				unlink(fname);
-				ast_frfree(f);
-				break;
-			   }
-			if (f->subclass == AST_CONTROL_ANSWER)
-			   {
-				if (debug) printf("@@@@ qcall_do:Phone Answered\n");
-				if (channel->_state == AST_STATE_UP)
-				   {
-					unlink(fname);
-					if (option_verbose > 2)
-						ast_verbose(VERBOSE_PREFIX_3 "Qcall got answer on %s\n",
-							channel->name);
-					usleep(1500000);
-					if (strlen(ident)) {
-						ast_streamfile(channel,ident,0);
-						if (ast_readstring(channel,buf,strlen(reqinp),10000,5000,"#"))
-						{
-							ast_stopstream(channel);
-							if (debug) printf("@@@@ qcall_do: timeout or hangup in dtmf read\n");
-							ast_frfree(f);
-							break;
-						}
-						ast_stopstream(channel);
-						if (strcmp(buf,reqinp)) /* if not match */
-						{
-							if (debug) printf("@@@@ qcall_do: response (%s) does not match required (%s)\n",buf,reqinp);
-							ast_frfree(f);
-							break;
-						}
-						ast_frfree(f);
-					}
-					/* okay, now we go for it */
-					context = strchr(extstr,'@');
-					if (!context) context = "default";
-					else *context++ = 0;
-					if (option_verbose > 2)
-						ast_verbose(VERBOSE_PREFIX_3 "Qcall got accept, now putting through to %s@%s on %s\n",
-							extstr,context,channel->name);
-					if (strlen(ident)) {
-						strncat(ident,"-ok", sizeof(ident) - strlen(ident) - 1);
-						/* if file existent, play it */
-						if (!ast_streamfile(channel,ident,0))
-						{
-							ast_waitstream(channel,"");
-							ast_stopstream(channel);
-						}
-					}
-					if (strcasecmp(clid, "asreceived")) {
-						channel->callerid = strdup(clid);
-						channel->ani = strdup(clid);
-					}
-					channel->language[0] = 0;
-					channel->dnid = strdup(extstr);
-#ifdef	AMAFLAGS
-					channel->amaflags = AMAFLAGS;
-#endif
-#ifdef	ACCTCODE
-					strncpy(channel->accountcode, ACCTCODE, sizeof(chan->accountcode) - 1);
-#else
-					channel->accountcode[0] = 0;
-#endif
-					if (maxsecs)  /* if finite length call */
-					   {
-						time(&channel->whentohangup);
-						channel->whentohangup += maxsecs;
-					   }
-					strncpy(channel->exten, extstr, sizeof(channel->exten) - 1);
-					strncpy(channel->context, context, sizeof(channel->context) - 1);
-					channel->priority = 1;
-					if(debug) printf("Caller ID is %s\n", channel->callerid);
-					ast_pbx_run(channel);
-					pthread_exit(NULL);
-				}
-			}
-			else if(f->subclass==AST_CONTROL_RINGING)
-				if (debug) printf("@@@@ qcall_do:Phone Ringing end\n");
-		}
-		ast_frfree(f);
-	}
-	ast_hangup(channel);
-	if (debug) printf("@@@@ qcall_do:Hung up channel\n");
-	pthread_exit(NULL);
-	return NULL;
-}
-
-int unload_module(void)
-{
-	STANDARD_HANGUP_LOCALUSERS;
-	return 0;
-}
-
-int load_module(void)
-{
-	snprintf(qdir, sizeof(qdir), "%s/%s", ast_config_AST_SPOOL_DIR, "qcall");
-	mkdir(qdir,0760);
-	ast_pthread_create(&qcall_thread,NULL,qcall,NULL);
-	return 0;
-}
-
-char *description(void)
-{
-	return tdesc;
-}
-
-int usecount(void)
-{
-	int res;
-	STANDARD_USECOUNT(res);
-	return res;
-}
-
-char *key()
-{
-	return ASTERISK_GPL_KEY;
-}
--- a/apps/app_queue.c
+++ b/apps/app_queue.c
--- a/apps/app_random.c
+++ b/apps/app_random.c
@@ -1,27 +1,43 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Random application
- * 
- * Copyright (c) 2003-2004 Tilghman Lesher.  All rights reserved.
+ * Copyright (c) 2003 - 2005 Tilghman Lesher.  All rights reserved.
 *
- * Tilghman Lesher <asterisk__app_random__20040111@the-tilghman.com>
+ * Tilghman Lesher <asterisk__app_random__200508@the-tilghman.com>
 *
 * This code is released by the author with no restrictions on usage or distribution.
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ *
+ * \brief Random application
+ *
+ * \author Tilghman Lesher <asterisk__app_random__200508@the-tilghman.com>
+ * \ingroup applications
 */

-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/options.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
 #include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"

 static char *tdesc = "Random goto";

@@ -37,75 +53,59 @@ STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

+static char random_state[256];
+
 static int random_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
 	struct localuser *u;

 	char *s;
-	char *exten, *pri, *context;
 	char *prob;
-	int probint, priorityint;
-
-	if (!data) {
+	int probint;
+	
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Random requires an argument ([probability]:[[context|]extension|]priority)\n");
 		return -1;
 	}
+	
 	LOCAL_USER_ADD(u);
-	s = ast_strdupa((void *) data);
+
+	s = ast_strdupa(data);
+	if (!s) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}

 	prob = strsep(&s,":");
 	if ((!prob) || (sscanf(prob, "%d", &probint) != 1))
 		probint = 0;

 	if ((random() % 100) + probint > 100) {
-		context = strsep(&s, "|");
-		exten = strsep(&s, "|");
-		if (!exten) {
-			/* Only a priority */
-			pri = context;
-			exten = NULL;
-			context = NULL;
-		} else {
-			pri = strsep(&s, "|");
-			if (!pri) {
-				pri = exten;
-				exten = context;
-				context = NULL;
-			}
-		}
-		if (!pri) {
-			ast_log(LOG_WARNING, "No label specified\n");
-			LOCAL_USER_REMOVE(u);
-			return -1;
-		} else if (sscanf(pri, "%d", &priorityint) != 1) {
-			ast_log(LOG_WARNING, "Priority '%s' must be a number > 0\n", pri);
-			LOCAL_USER_REMOVE(u);
-			return -1;
-		}
-		/* At this point we have a priority and */
-		/* maybe an extension and a context     */
-		chan->priority = priorityint - 1;
-		if (exten && strcasecmp(exten, "BYEXTENSION"))
-			strncpy(chan->exten, exten, sizeof(chan->exten)-1);
-		if (context)
-			strncpy(chan->context, context, sizeof(chan->context)-1);
+		res = ast_parseable_goto(chan, s);
 		if (option_verbose > 2)
 			ast_verbose( VERBOSE_PREFIX_3 "Random branches to (%s,%s,%d)\n",
 				chan->context,chan->exten, chan->priority+1);
-		LOCAL_USER_REMOVE(u);
 	}
+	LOCAL_USER_REMOVE(u);
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+	
+	res = ast_unregister_application(app_random);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app_random);
+
+	return res;	
 }

 int load_module(void)
 {
+	initstate((getppid() * 65535 + getpid()) % RAND_MAX, random_state, 256);
 	return ast_register_application(app_random, random_exec, random_synopsis, random_descrip);
 }

--- a/apps/app_read.c
+++ b/apps/app_read.c
@@ -1,28 +1,46 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Trivial application to read a variable
- * 
- * Copyright (C) 2003, Digium
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Trivial application to read a variable
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/app.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/options.h>
-#include <asterisk/utils.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/app.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/options.h"
+#include "asterisk/utils.h"

 static char *tdesc = "Read Variable Application";

@@ -31,64 +49,117 @@ static char *app = "Read";
 static char *synopsis = "Read a variable";

 static char *descrip = 
-"  Read(variable[|filename][|maxdigits][|option])\n\n"
-"Reads a #-terminated string of digits from the user in to the given variable,\n"
-"optionally playing a given filename first.\n"
+"  Read(variable[|filename][|maxdigits][|option][|attempts][|timeout])\n\n"
+"Reads a #-terminated string of digits a certain number of times from the\n"
+"user in to the given variable.\n"
+"  filename   -- file to play before reading digits.\n"
 "  maxdigits  -- maximum acceptable number of digits. Stops reading after\n"
 "                maxdigits have been entered (without requiring the user to\n"
 "                press the '#' key).\n"
 "                Defaults to 0 - no limit - wait for the user press the '#' key.\n"
 "                Any value below 0 means the same. Max accepted value is 255.\n"
 "  option     -- may be 'skip' to return immediately if the line is not up,\n"
-"                or 'noanswer' to read digits even if the line is not up.\n\n"
-"Returns -1 on hangup or error and 0 otherwise.\n";
+"                or 'noanswer' to read digits even if the line is not up.\n"
+"  attempts   -- if greater than 1, that many attempts will be made in the \n"
+"                event no data is entered.\n"
+"  timeout    -- if greater than 0, that value will override the default timeout.\n\n"
+"Read should disconnect if the function fails or errors out.\n";

 STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

+#define ast_next_data(instr,ptr,delim) if((ptr=strchr(instr,delim))) { *(ptr) = '\0' ; ptr++;}
+
 static int read_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
 	struct localuser *u;
 	char tmp[256];
-	char argdata[256] = "";
-	char *varname;
-	char *filename;
-	char *stringp;
-	char *maxdigitstr;
-	char *options;
+	char *timeout = NULL;
+	char *varname = NULL;
+	char *filename = NULL;
+	char *loops;
+	char *maxdigitstr=NULL;
+	char *options=NULL;
 	int option_skip = 0;
 	int option_noanswer = 0;
 	int maxdigits=255;
-	if (!data || ast_strlen_zero((char *)data)) {
+	int tries = 1;
+	int to = 0;
+	int x = 0;
+	char *argcopy = NULL;
+	char *args[8];
+
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Read requires an argument (variable)\n");
 		return -1;
 	}
-	strncpy(argdata, (char *)data, sizeof(argdata)-1);
-	stringp=argdata;
-	varname = strsep(&stringp, "|");
-	filename = strsep(&stringp, "|");
-	maxdigitstr = strsep(&stringp,"|");
-	options = strsep(&stringp, "|");
-	if (options && !strcasecmp(options, "skip"))
-		option_skip = 1;
-	if (options && !strcasecmp(options, "noanswer"))
-		option_noanswer = 1;
-	if (!(filename) || ast_strlen_zero(filename)) 
-		filename = NULL;
-	if (maxdigitstr) {
-	    maxdigits = atoi(maxdigitstr);
-	    if ((maxdigits<1) || (maxdigits>255)) {
-    		maxdigits = 255;
-	    } else
-		ast_verbose(VERBOSE_PREFIX_3 "Accepting a maximum of %i digits.\n", maxdigits);
-	}	
-	if (!(varname) || ast_strlen_zero(varname)) {
-		ast_log(LOG_WARNING, "Read requires an variable name\n");
+
+	LOCAL_USER_ADD(u);
+	
+	argcopy = ast_strdupa(data);
+	if (!argcopy) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
-	LOCAL_USER_ADD(u);
+
+	if (ast_app_separate_args(argcopy, '|', args, sizeof(args) / sizeof(args[0])) < 1) {
+		ast_log(LOG_WARNING, "Cannot Parse Arguments.\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	varname = args[x++];
+	filename = args[x++];
+	maxdigitstr = args[x++];
+	options = args[x++];
+	loops = args[x++];
+	timeout = args[x++];
+	
+	if (options) { 
+		if (!strcasecmp(options, "skip"))
+			option_skip = 1;
+		else if (!strcasecmp(options, "noanswer"))
+			option_noanswer = 1;
+		else {
+			if (strchr(options, 's'))
+				option_skip = 1;
+			if (strchr(options, 'n'))
+				option_noanswer = 1;
+		}
+	}
+
+	if(loops) {
+		tries = atoi(loops);
+		if(tries <= 0)
+			tries = 1;
+	}
+
+	if(timeout) {
+		to = atoi(timeout);
+		if (to <= 0)
+			to = 0;
+		else
+			to *= 1000;
+	}
+
+	if (ast_strlen_zero(filename)) 
+		filename = NULL;
+	if (maxdigitstr) {
+		maxdigits = atoi(maxdigitstr);
+		if ((maxdigits<1) || (maxdigits>255)) {
+    			maxdigits = 255;
+		} else if (option_verbose > 2)
+			ast_verbose(VERBOSE_PREFIX_3 "Accepting a maximum of %d digits.\n", maxdigits);
+	}
+	if (ast_strlen_zero(varname)) {
+		ast_log(LOG_WARNING, "Invalid! Usage: Read(variable[|filename][|maxdigits][|option][|attempts][|timeout])\n\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	
 	if (chan->_state != AST_STATE_UP) {
 		if (option_skip) {
 			/* At the user's option, skip if the line is not up */
@@ -101,14 +172,29 @@ static int read_exec(struct ast_channel *chan, void *data)
 		}
 	}
 	if (!res) {
-		ast_stopstream(chan);
-		res = ast_app_getdata(chan, filename, tmp, maxdigits, 0);
-		if (res > -1) {
-			pbx_builtin_setvar_helper(chan, varname, tmp);
-			ast_verbose(VERBOSE_PREFIX_3 "User entered '%s'\n", tmp);
-			res = 0;
-		} else {
-			ast_verbose(VERBOSE_PREFIX_3 "User disconnected\n");
+		while(tries && !res) {
+			ast_stopstream(chan);
+			res = ast_app_getdata(chan, filename, tmp, maxdigits, to);
+			if (res > -1) {
+				pbx_builtin_setvar_helper(chan, varname, tmp);
+				if (!ast_strlen_zero(tmp)) {
+					if (option_verbose > 2)
+						ast_verbose(VERBOSE_PREFIX_3 "User entered '%s'\n", tmp);
+					tries = 0;
+				} else {
+					tries--;
+					if (option_verbose > 2) {
+						if (tries)
+							ast_verbose(VERBOSE_PREFIX_3 "User entered nothing, %d chance%s left\n", tries, (tries != 1) ? "s" : "");
+						else
+							ast_verbose(VERBOSE_PREFIX_3 "User entered nothing.\n");
+					}
+				}
+				res = 0;
+			} else {
+				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "User disconnected\n");
+			}
 		}
 	}
 	LOCAL_USER_REMOVE(u);
@@ -117,8 +203,13 @@ static int read_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_readfile.c
+++ b/apps/app_readfile.c
@@ -0,0 +1,143 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Matt O'Gorman <mogorman@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief ReadFile application -- Reads in a File for you.
+ *
+ * \ingroup applications
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/app.h"
+#include "asterisk/module.h"
+
+static char *tdesc = "Stores output of file into a variable";
+
+static char *app_readfile = "ReadFile";
+
+static char *readfile_synopsis = "ReadFile(varname=file,length)";
+
+static char *readfile_descrip =
+"ReadFile(varname=file,length)\n"
+"  Varname - Result stored here.\n"
+"  File - The name of the file to read.\n"
+"  Length - Maximum number of characters to capture.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+
+static int readfile_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *s, *varname=NULL, *file=NULL, *length=NULL, *returnvar=NULL;
+	int len=0;
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "ReadFile require an argument!\n");
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	s = ast_strdupa(data);
+	if (!s) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	varname = strsep(&s, "=");
+	file = strsep(&s, "|");
+	length = s;
+
+	if (!varname || !file) {
+		ast_log(LOG_ERROR, "No file or variable specified!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (length) {
+		if ((sscanf(length, "%d", &len) != 1) || (len < 0)) {
+			ast_log(LOG_WARNING, "%s is not a positive number, defaulting length to max\n", length);
+			len = 0;
+		}
+	}
+
+	returnvar = ast_read_textfile(file);
+	if(len > 0){
+		if(len < strlen(returnvar))
+			returnvar[len]='\0';
+		else
+			ast_log(LOG_WARNING,"%s is longer than %d, and %d \n", file, len, (int)strlen(returnvar));
+	}
+	pbx_builtin_setvar_helper(chan, varname, returnvar);
+	free(returnvar);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app_readfile);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
+}
+
+int load_module(void)
+{
+	return ast_register_application(app_readfile, readfile_exec, readfile_synopsis, readfile_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_realtime.c
+++ b/apps/app_realtime.c
@@ -0,0 +1,262 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Anthony Minessale <anthmct@yahoo.com>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief RealTime App
+ * 
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/options.h"
+#include "asterisk/pbx.h"
+#include "asterisk/config.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/cli.h"
+
+#define next_one(var) var = var->next
+#define crop_data(str) { *(str) = '\0' ; (str)++; }
+
+static char *tdesc = "Realtime Data Lookup/Rewrite";
+static char *app = "RealTime";
+static char *uapp = "RealTimeUpdate";
+static char *synopsis = "Realtime Data Lookup";
+static char *usynopsis = "Realtime Data Rewrite";
+static char *USAGE = "RealTime(<family>|<colmatch>|<value>[|<prefix>])";
+static char *UUSAGE = "RealTimeUpdate(<family>|<colmatch>|<value>|<newcol>|<newval>)";
+static char *desc = "Use the RealTime config handler system to read data into channel variables.\n"
+"RealTime(<family>|<colmatch>|<value>[|<prefix>])\n\n"
+"All unique column names will be set as channel variables with optional prefix to the name.\n"
+"e.g. prefix of 'var_' would make the column 'name' become the variable ${var_name}\n\n";
+static char *udesc = "Use the RealTime config handler system to update a value\n"
+"RealTimeUpdate(<family>|<colmatch>|<value>|<newcol>|<newval>)\n\n"
+"The column <newcol> in 'family' matching column <colmatch>=<value> will be updated to <newval>\n";
+
+STANDARD_LOCAL_USER;
+LOCAL_USER_DECL;
+
+static int cli_load_realtime(int fd, int argc, char **argv) 
+{
+	char *header_format = "%30s  %-30s\n";
+	struct ast_variable *var=NULL;
+
+	if(argc<5) {
+		ast_cli(fd, "You must supply a family name, a column to match on, and a value to match to.\n");
+		return RESULT_FAILURE;
+	}
+
+	var = ast_load_realtime(argv[2], argv[3], argv[4], NULL);
+
+	if(var) {
+		ast_cli(fd, header_format, "Column Name", "Column Value");
+		ast_cli(fd, header_format, "--------------------", "--------------------");
+		while(var) {
+			ast_cli(fd, header_format, var->name, var->value);
+			var = var->next;
+		}
+	} else {
+		ast_cli(fd, "No rows found matching search criteria.\n");
+	}
+	return RESULT_SUCCESS;
+}
+
+static int cli_update_realtime(int fd, int argc, char **argv) {
+	int res = 0;
+
+	if(argc<7) {
+		ast_cli(fd, "You must supply a family name, a column to update on, a new value, column to match, and value to to match.\n");
+		ast_cli(fd, "Ex: realtime update sipfriends name bobsphone port 4343\n will execute SQL as UPDATE sipfriends SET port = 4343 WHERE name = bobsphone\n");
+		return RESULT_FAILURE;
+	}
+
+	res = ast_update_realtime(argv[2], argv[3], argv[4], argv[5], argv[6], NULL);
+
+	if(res < 0) {
+		ast_cli(fd, "Failed to update. Check the debug log for possible SQL related entries.\n");
+		return RESULT_SUCCESS;
+	}
+
+       ast_cli(fd, "Updated %d RealTime record%s.\n", res, (res != 1) ? "s" : "");
+
+	return RESULT_SUCCESS;
+}
+
+static char cli_load_realtime_usage[] =
+"Usage: realtime load <family> <colmatch> <value>\n"
+"       Prints out a list of variables using the RealTime driver.\n";
+
+static struct ast_cli_entry cli_load_realtime_cmd = {
+        { "realtime", "load", NULL, NULL }, cli_load_realtime,
+        "Used to print out RealTime variables.", cli_load_realtime_usage, NULL };
+
+static char cli_update_realtime_usage[] =
+"Usage: realtime update <family> <colmatch> <value>\n"
+"       Update a single variable using the RealTime driver.\n";
+
+static struct ast_cli_entry cli_update_realtime_cmd = {
+        { "realtime", "update", NULL, NULL }, cli_update_realtime,
+        "Used to update RealTime variables.", cli_update_realtime_usage, NULL };
+
+static int realtime_update_exec(struct ast_channel *chan, void *data) 
+{
+	char *family=NULL, *colmatch=NULL, *value=NULL, *newcol=NULL, *newval=NULL;
+	struct localuser *u;
+	int res = 0;
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_ERROR,"Invalid input: usage %s\n",UUSAGE);
+		return -1;
+	}
+	
+	LOCAL_USER_ADD(u);
+
+	if ((family = ast_strdupa(data))) {
+		if ((colmatch = strchr(family,'|'))) {
+			crop_data(colmatch);
+			if ((value = strchr(colmatch,'|'))) {
+				crop_data(value);
+				if ((newcol = strchr(value,'|'))) {
+					crop_data(newcol);
+					if ((newval = strchr(newcol,'|'))) 
+						crop_data(newval);
+				}
+			}
+		}
+	}
+	if (! (family && value && colmatch && newcol && newval) ) {
+		ast_log(LOG_ERROR,"Invalid input: usage %s\n",UUSAGE);
+		res = -1;
+	} else {
+		ast_update_realtime(family,colmatch,value,newcol,newval,NULL);
+	}
+
+	LOCAL_USER_REMOVE(u);
+	
+	return res;
+}
+
+
+static int realtime_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	struct ast_variable *var, *itt;
+	char *family=NULL, *colmatch=NULL, *value=NULL, *prefix=NULL, *vname=NULL;
+	size_t len;
+		
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_ERROR,"Invalid input: usage %s\n",USAGE);
+		return -1;
+	}
+	
+	LOCAL_USER_ADD(u);
+
+	if ((family = ast_strdupa(data))) {
+		if ((colmatch = strchr(family,'|'))) {
+			crop_data(colmatch);
+			if ((value = strchr(colmatch,'|'))) {
+				crop_data(value);
+				if ((prefix = strchr(value,'|')))
+					crop_data(prefix);
+			}
+		}
+	}
+	if (! (family && value && colmatch) ) {
+		ast_log(LOG_ERROR,"Invalid input: usage %s\n",USAGE);
+		res = -1;
+	} else {
+		if (option_verbose > 3)
+			ast_verbose(VERBOSE_PREFIX_4"Realtime Lookup: family:'%s' colmatch:'%s' value:'%s'\n",family,colmatch,value);
+		if ((var = ast_load_realtime(family, colmatch, value, NULL))) {
+			for (itt = var; itt; itt = itt->next) {
+				if(prefix) {
+					len = strlen(prefix) + strlen(itt->name) + 2;
+					vname = alloca(len);
+					snprintf(vname,len,"%s%s",prefix,itt->name);
+					
+				} else 
+					vname = itt->name;
+
+				pbx_builtin_setvar_helper(chan, vname, itt->value);
+			}
+			ast_variables_destroy(var);
+		} else if (option_verbose > 3)
+			ast_verbose(VERBOSE_PREFIX_4"No Realtime Matches Found.\n");
+	}
+	
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_cli_unregister(&cli_load_realtime_cmd);
+	res |= ast_cli_unregister(&cli_update_realtime_cmd);
+	res |= ast_unregister_application(uapp);
+	res |= ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+
+	res = ast_cli_register(&cli_load_realtime_cmd);
+	res |= ast_cli_register(&cli_update_realtime_cmd);
+	res |= ast_register_application(uapp, realtime_update_exec, usynopsis, udesc);
+	res |= ast_register_application(app, realtime_exec, synopsis, desc);
+
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
+
--- a/apps/app_record.c
+++ b/apps/app_record.c
@@ -1,26 +1,46 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Trivial application to record a sound file
- * 
- * Copyright (C) 2001, Linux Support Services, Inc.
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Matthew Fredrickson <creslin@linux-support.net>
+ * Matthew Fredrickson <creslin@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Trivial application to record a sound file
+ *
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/dsp.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/dsp.h"
+#include "asterisk/utils.h"
+#include "asterisk/options.h"

 static char *tdesc = "Trivial Record Application";

@@ -29,20 +49,26 @@ static char *app = "Record";
 static char *synopsis = "Record to a file";

 static char *descrip = 
-"  Record(filename:format|silence[|maxduration][|option])\n\n"
+"  Record(filename.format|silence[|maxduration][|options])\n\n"
 "Records from the channel into a given filename. If the file exists it will\n"
 "be overwritten.\n"
 "- 'format' is the format of the file type to be recorded (wav, gsm, etc).\n"
 "- 'silence' is the number of seconds of silence to allow before returning.\n"
 "- 'maxduration' is the maximum recording duration in seconds. If missing\n"
 "or 0 there is no maximum.\n"
-"- 'option' may be 'skip' to return immediately if the line is not up,\n"
-"or 'noanswer' to attempt to record even if the line is not up.\n\n"
+"- 'options' may contain any of the following letters:\n"
+"     'a' : append to existing recording rather than replacing\n"
+"     'n' : do not answer, but record anyway if line not yet answered\n"
+"     'q' : quiet (do not play a beep tone)\n"
+"     's' : skip recording if the line is not yet answered\n"
+"     't' : use alternate '*' terminator key instead of default '#'\n"
+"\n"
 "If filename contains '%d', these characters will be replaced with a number\n"
 "incremented by one each time the file is recorded. \n\n"
-"Formats: g723, g729, gsm, h263, ulaw, alaw, vox, wav, WAV\n\n"
+"Use 'show file formats' to see the available formats on your system\n\n"
 "User can press '#' to terminate the recording and continue to the next priority.\n\n"
-"Returns -1 when the user hangs up.\n";
+"If the user should hangup during a recording, all data will be lost and the\n"
+"application will teminate. \n";

 STANDARD_LOCAL_USER;

@@ -53,12 +79,10 @@ static int record_exec(struct ast_channel *chan, void *data)
 	int res = 0;
 	int count = 0;
 	int percentflag = 0;
-	char fil[256];
-	char tmp[256];
-	char ext[10];
-	char *vdata;
+	char *filename, *ext = NULL, *silstr, *maxstr, *options;
+	char *vdata, *p;
 	int i = 0;
-	int j = 0;
+	char tmp[256];

 	struct ast_filestream *s = '\0';
 	struct localuser *u;
@@ -69,120 +93,107 @@ static int record_exec(struct ast_channel *chan, void *data)
 	int dspsilence = 0;
 	int silence = 0;		/* amount of silence to allow */
 	int gotsilence = 0;		/* did we timeout for silence? */
-	char silencestr[5];
-	char durationstr[8];
-	int maxduration = 0;		/* max duration of recording */
+	int maxduration = 0;		/* max duration of recording in milliseconds */
 	int gottimeout = 0;		/* did we timeout for maxduration exceeded? */
-	time_t timeout = 0;
-	char option[16];
 	int option_skip = 0;
 	int option_noanswer = 0;
+	int option_append = 0;
+	int terminator = '#';
+	int option_quiet = 0;
 	int rfmt = 0;
 	int flags;
-	char *end=NULL;
-	char *p=NULL;
-
-
-
+	int waitres;
+	struct ast_silence_generator *silgen = NULL;
+	
 	/* The next few lines of code parse out the filename and header from the input string */
-	if (!data) { /* no data implies no filename or anything is present */
+	if (ast_strlen_zero(data)) { /* no data implies no filename or anything is present */
 		ast_log(LOG_WARNING, "Record requires an argument (filename)\n");
 		return -1;
 	}
+
+	LOCAL_USER_ADD(u);
+
+	/* Yay for strsep being easy */
 	vdata = ast_strdupa(data);
-	
-	p = vdata;
-	while(p && (p=strchr(p,':'))) {
-		end=p;
-		if(!strcasecmp(end,":end")) {
-			*end='\0';
-			end++;
-			break;
-		}
-		p++;
-		end=NULL;
-	}
-	
-
-	for (; vdata[i] && (vdata[i] != ':') && (vdata[i] != '|'); i++ ) {
-		if ((vdata[i] == '%') && (vdata[i+1] == 'd')) {
-			percentflag = 1;                      /* the wildcard is used */
-		}
-		if (j < sizeof(fil) - 1)
-			fil[j++] = vdata[i];
-	}
-	fil[j] = '\0';
-
-	if (vdata[i] != ':') {
-		ast_log(LOG_WARNING, "No extension found\n");
+	if (!vdata) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
-	i++;

-	j = 0;
-	for (; vdata[i] && (vdata[i] != '|'); i++)
-		if (j < sizeof(ext) - 1)
-			ext[j++] = vdata[i];
-	ext[j] = '\0';
-
-	if (vdata[i] == '|')
-		i++;
-
-	j = 0;
-	for (; vdata[i] && (vdata[i] != '|'); i++)
-		if (j < sizeof(silencestr) - 1)
-			silencestr[j++] = vdata[i];
-	silencestr[j] = '\0';
-
-	if (j > 0) {
-		silence = atoi(silencestr);
-		if (silence > 0)
-			silence *= 1000;
+	p = vdata;
+	filename = strsep(&p, "|");
+	silstr = strsep(&p, "|");
+	maxstr = strsep(&p, "|");	
+	options = strsep(&p, "|");
+	
+	if (filename) {
+		if (strstr(filename, "%d"))
+			percentflag = 1;
+		ext = strrchr(filename, '.'); /* to support filename with a . in the filename, not format */
+		if (!ext)
+			ext = strchr(filename, ':');
+		if (ext) {
+			*ext = '\0';
+			ext++;
+		}
 	}
-
-	if (vdata[i] == '|')
-		i++;
-
-	j = 0;
-	for (; vdata[i] && (vdata[i] != '|'); i++)
-		if (j < sizeof(durationstr) - 1)
-			durationstr[j++] = vdata[i];
-	durationstr[j] = '\0';
-
-	if (j > 0)
-		maxduration = atoi(durationstr);
-
-	if (vdata[i] == '|')
-		i++;
-
-	j = 0;
-	for (; vdata[i] && (vdata[i] != '|'); i++)
-		if (j < sizeof(option) - 1)
-			option[j++] = vdata[i];
-	option[j] = '\0';
-
-	if (!strcasecmp(option, "skip"))
-		option_skip = 1;
-	if (!strcasecmp(option, "noanswer"))
-		option_noanswer = 1;
-
+	if (!ext) {
+		ast_log(LOG_WARNING, "No extension specified to filename!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	if (silstr) {
+		if ((sscanf(silstr, "%d", &i) == 1) && (i > -1)) {
+			silence = i * 1000;
+		} else if (!ast_strlen_zero(silstr)) {
+			ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", silstr);
+		}
+	}
+	
+	if (maxstr) {
+		if ((sscanf(maxstr, "%d", &i) == 1) && (i > -1))
+			/* Convert duration to milliseconds */
+			maxduration = i * 1000;
+		else if (!ast_strlen_zero(maxstr))
+			ast_log(LOG_WARNING, "'%s' is not a valid maximum duration\n", maxstr);
+	}
+	if (options) {
+		/* Retain backwards compatibility with old style options */
+		if (!strcasecmp(options, "skip"))
+			option_skip = 1;
+		else if (!strcasecmp(options, "noanswer"))
+			option_noanswer = 1;
+		else {
+			if (strchr(options, 's'))
+				option_skip = 1;
+			if (strchr(options, 'n'))
+				option_noanswer = 1;
+			if (strchr(options, 'a'))
+				option_append = 1;
+			if (strchr(options, 't'))
+				terminator = '*';
+			if (strchr(options, 'q'))
+				option_quiet = 1;
+		}
+	}
+	
 	/* done parsing */
-
 	
 	/* these are to allow the use of the %d in the config file for a wild card of sort to
 	  create a new file with the inputed name scheme */
 	if (percentflag) {
 		do {
-			snprintf(tmp, sizeof(tmp), fil, count);
+			snprintf(tmp, sizeof(tmp), filename, count);
 			count++;
 		} while ( ast_fileexists(tmp, ext, chan->language) != -1 );
 		pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
 	} else
-		strncpy(tmp, fil, sizeof(tmp)-1);
+		strncpy(tmp, filename, sizeof(tmp)-1);
 	/* end of routine mentioned */
-
-	LOCAL_USER_ADD(u);
-
+	
+	
+	
 	if (chan->_state != AST_STATE_UP) {
 		if (option_skip) {
 			/* At the user's option, skip if the line is not up */
@@ -194,7 +205,12 @@ static int record_exec(struct ast_channel *chan, void *data)
 		}
 	}
 	
-	if (!res) {
+	if (res) {
+		ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
+		goto out;
+	}
+	
+	if (!option_quiet) {
 		/* Some code to play a nice little beep to signify the start of the record operation */
 		res = ast_streamfile(chan, "beep", chan->language);
 		if (!res) {
@@ -203,118 +219,139 @@ static int record_exec(struct ast_channel *chan, void *data)
 			ast_log(LOG_WARNING, "ast_streamfile failed on %s\n", chan->name);
 		}
 		ast_stopstream(chan);
-
-		/* The end of beep code.  Now the recording starts */
-
-
-		if (silence > 0) {
-	        	rfmt = chan->readformat;
-        		res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
-        		if (res < 0) {
-                		ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
-        		        return -1;
-        		}
-			sildet = ast_dsp_new();
-        		if (!sildet) {
-                		ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
-                		return -1;
-		       	}
-			ast_dsp_set_threshold(sildet, 256);
+	}
+		
+	/* The end of beep code.  Now the recording starts */
+		
+	if (silence > 0) {
+		rfmt = chan->readformat;
+		res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
+		if (res < 0) {
+			ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
+			LOCAL_USER_REMOVE(u);
+			return -1;
 		}
+		sildet = ast_dsp_new();
+		if (!sildet) {
+			ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
+			LOCAL_USER_REMOVE(u);
+			return -1;
+		}
+		ast_dsp_set_threshold(sildet, 256);
+	} 
+		
+		
+	flags = option_append ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
+	s = ast_writefile( tmp, ext, NULL, flags , 0, 0644);
+		
+	if (!s) {
+		ast_log(LOG_WARNING, "Could not create file %s\n", filename);
+		goto out;
+	}

-
-		flags = end ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
-		s = ast_writefile( tmp, ext, NULL, flags , 0, 0644);
-
-
-		if (s) {
-			if (maxduration > 0)
-				timeout = time(NULL) + (time_t)maxduration;
-
-			while (ast_waitfor(chan, -1) > -1) {
-				if (maxduration > 0 && time(NULL) > timeout) {
-					gottimeout = 1;
-					break;
+	if (option_transmit_silence_during_record)
+		silgen = ast_channel_start_silence_generator(chan);
+	
+	/* Request a video update */
+	ast_indicate(chan, AST_CONTROL_VIDUPDATE);
+	
+	if (maxduration <= 0)
+		maxduration = -1;
+	
+	while ((waitres = ast_waitfor(chan, maxduration)) > -1) {
+		if (maxduration > 0) {
+			if (waitres == 0) {
+				gottimeout = 1;
+				break;
+			}
+			maxduration = waitres;
+		}
+		
+		f = ast_read(chan);
+		if (!f) {
+			res = -1;
+			break;
+		}
+		if (f->frametype == AST_FRAME_VOICE) {
+			res = ast_writestream(s, f);
+			
+			if (res) {
+				ast_log(LOG_WARNING, "Problem writing frame\n");
+				break;
+			}
+			
+			if (silence > 0) {
+				dspsilence = 0;
+				ast_dsp_silence(sildet, f, &dspsilence);
+				if (dspsilence) {
+					totalsilence = dspsilence;
+				} else {
+					totalsilence = 0;
 				}
-
-				f = ast_read(chan);
-				if (!f) {
-					res = -1;
-					break;
-				}
-				if (f->frametype == AST_FRAME_VOICE) {
-					res = ast_writestream(s, f);
-					
-					if (res) {
-						ast_log(LOG_WARNING, "Problem writing frame\n");
-						break;
-					}
-
-					if (silence > 0) {
-						dspsilence = 0;
-						ast_dsp_silence(sildet, f, &dspsilence);
-						if (dspsilence) {
-							totalsilence = dspsilence;
-						} else {
-							totalsilence = 0;
-						}
-	                                	if (totalsilence > silence) {
-        	                        	        /* Ended happily with silence */
-							ast_frfree(f);
-							gotsilence = 1;
-                	                	        break;
-	                     	        	}
-					}
-				}
-				if (f->frametype == AST_FRAME_VIDEO) {
-					res = ast_writestream(s, f);
-					
-					if (res) {
-						ast_log(LOG_WARNING, "Problem writing frame\n");
-						break;
-					}
-				}
-				if ((f->frametype == AST_FRAME_DTMF) &&
-					(f->subclass == '#')) {
+				if (totalsilence > silence) {
+					/* Ended happily with silence */
 					ast_frfree(f);
+					gotsilence = 1;
 					break;
 				}
-				ast_frfree(f);
 			}
-			if (!f) {
-					ast_log(LOG_DEBUG, "Got hangup\n");
-					res = -1;
+		}
+		if (f->frametype == AST_FRAME_VIDEO) {
+			res = ast_writestream(s, f);
+			
+			if (res) {
+				ast_log(LOG_WARNING, "Problem writing frame\n");
+				break;
 			}
+		}
+		if ((f->frametype == AST_FRAME_DTMF) &&
+		    (f->subclass == terminator)) {
+			ast_frfree(f);
+			break;
+		}
+		ast_frfree(f);
+	}
+	if (!f) {
+		ast_log(LOG_DEBUG, "Got hangup\n");
+		res = -1;
+	}
+			
+	if (gotsilence) {
+		ast_stream_rewind(s, silence-1000);
+		ast_truncstream(s);
+	} else if (!gottimeout) {
+		/* Strip off the last 1/4 second of it */
+		ast_stream_rewind(s, 250);
+		ast_truncstream(s);
+	}
+	ast_closestream(s);

-			if (gotsilence) {
-				ast_stream_rewind(s, silence-1000);
-				ast_truncstream(s);
-			} else if (!gottimeout) {
-				/* Strip off the last 1/4 second of it */
-				ast_stream_rewind(s, 250);
-				ast_truncstream(s);
-			}
-			ast_closestream(s);
-		} else			
-			ast_log(LOG_WARNING, "Could not create file %s\n", fil);
-	} else
-		ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
-
-	LOCAL_USER_REMOVE(u);
+	if (silgen)
+		ast_channel_stop_silence_generator(chan, silgen);
+	
+ out:
 	if ((silence > 0) && rfmt) {
-	        res = ast_set_read_format(chan, rfmt);
-        	if (res)
-        	        ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name);
+		res = ast_set_read_format(chan, rfmt);
+		if (res)
+			ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name);
 		if (sildet)
 			ast_dsp_free(sildet);
 	}
+
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_rpt.c
+++ b/apps/app_rpt.c
--- a/apps/app_sayunixtime.c
+++ b/apps/app_sayunixtime.c
@@ -1,27 +1,43 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * SayUnixTime application
- * 
 * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
 *
 * Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com>
 *
 * This code is released by the author with no restrictions on usage.
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ *
+ * \brief SayUnixTime application
+ * 
+ * \ingroup applications
 */

 #include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/options.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/say.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/say.h"


 static char *tdesc = "Say time";
@@ -38,8 +54,7 @@ static char *sayunixtime_descrip =
 "  timezone: timezone, see /usr/share/zoneinfo for a list.\n"
 "              defaults to machine default.\n"
 "  format:   a format the time is to be said in.  See voicemail.conf.\n"
-"              defaults to \"ABdY 'digits/at' IMp\"\n"
-"  Returns 0 or -1 on hangup.\n";
+"              defaults to \"ABdY 'digits/at' IMp\"\n";
 static char *datetime_descrip =
 "DateTime([unixtime][|[timezone][|format]])\n"
 "  unixtime: time, in seconds since Jan 1, 1970.  May be negative.\n"
@@ -47,8 +62,7 @@ static char *datetime_descrip =
 "  timezone: timezone, see /usr/share/zoneinfo for a list.\n"
 "              defaults to machine default.\n"
 "  format:   a format the time is to be said in.  See voicemail.conf.\n"
-"              defaults to \"ABdY 'digits/at' IMp\"\n"
-"  Returns 0 or -1 on hangup.\n";
+"              defaults to \"ABdY 'digits/at' IMp\"\n";

 STANDARD_LOCAL_USER;

@@ -58,16 +72,23 @@ static int sayunixtime_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
 	struct localuser *u;
-	char *s,*zone=NULL,*timec;
+	char *s,*zone=NULL,*timec,*format;
 	time_t unixtime;
-	char *format = "ABdY 'digits/at' IMp";
 	struct timeval tv;
-
+	
 	LOCAL_USER_ADD(u);

-	gettimeofday(&tv,NULL);
+	tv = ast_tvnow();
 	unixtime = (time_t)tv.tv_sec;

+	if( !strcasecmp(chan->language, "da" ) ) {
+		format = "A dBY HMS";
+	} else if ( !strcasecmp(chan->language, "de" ) ) {
+		format = "A dBY HMS";
+	} else {
+		format = "ABdY 'digits/at' IMp";
+	} 
+
 	if (data) {
 		s = data;
 		s = ast_strdupa(s);
@@ -105,22 +126,23 @@ static int sayunixtime_exec(struct ast_channel *chan, void *data)
 int unload_module(void)
 {
 	int res;
-	STANDARD_HANGUP_LOCALUSERS;
+	
 	res = ast_unregister_application(app_sayunixtime);
-	if (! res)
-		return ast_unregister_application(app_datetime);
-	else
-		return res;
+	res |= ast_unregister_application(app_datetime);
+
+	STANDARD_HANGUP_LOCALUSERS;
+	
+	return res;
 }

 int load_module(void)
 {
 	int res;
+	
 	res = ast_register_application(app_sayunixtime, sayunixtime_exec, sayunixtime_synopsis, sayunixtime_descrip);
-	if (! res)
-		return ast_register_application(app_datetime, sayunixtime_exec, sayunixtime_synopsis, datetime_descrip);
-	else
-		return res;
+	res |= ast_register_application(app_datetime, sayunixtime_exec, sayunixtime_synopsis, datetime_descrip);
+	
+	return res;
 }

 char *description(void)
--- a/apps/app_senddtmf.c
+++ b/apps/app_senddtmf.c
@@ -1,28 +1,46 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * App to send DTMF digits
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to send DTMF digits
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/options.h>
-#include <asterisk/utils.h>
-#include <asterisk/app.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
+#include "asterisk/app.h"

 static char *tdesc = "Send DTMF digits Application";

@@ -31,9 +49,10 @@ static char *app = "SendDTMF";
 static char *synopsis = "Sends arbitrary DTMF digits";

 static char *descrip = 
-"  SendDTMF(digits): Sends DTMF digits on a channel. \n"
-"  Accepted digits: 0-9, *#abcd\n"
-" Returns 0 on success or -1 on a hangup.\n";
+" SendDTMF(digits[|timeout_ms]): Sends DTMF digits on a channel. \n"
+" Accepted digits: 0-9, *#abcd\n"
+" The application will either pass the assigned digits or terminate if it\n"
+" encounters an error.\n";

 STANDARD_LOCAL_USER;

@@ -43,22 +62,48 @@ static int senddtmf_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
 	struct localuser *u;
-	char *digits = data;
+	char *digits = NULL, *to = NULL;
+	int timeout = 250;

-	if (!digits || ast_strlen_zero(digits)) {
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "SendDTMF requires an argument (digits or *#aAbBcCdD)\n");
+		return 0;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	digits = ast_strdupa(data);
+	if (!digits) {
+		ast_log(LOG_ERROR, "Out of Memory!\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
-	LOCAL_USER_ADD(u);
-	res = ast_dtmf_stream(chan,NULL,digits,250);
+
+	if ((to = strchr(digits,'|'))) {
+		*to = '\0';
+		to++;
+		timeout = atoi(to);
+	}
+		
+	if(timeout <= 0)
+		timeout = 250;
+
+	res = ast_dtmf_stream(chan,NULL,digits,timeout);
+		
 	LOCAL_USER_REMOVE(u);
+
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_sendtext.c
+++ b/apps/app_sendtext.c
@@ -1,41 +1,67 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * App to transmit a text message
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to transmit a text message
+ * 
+ * Requires support of sending text messages from channel driver
+ *
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/channel_pvt.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/image.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>

-static char *tdesc = "Send Text Applications";
+#include "asterisk.h"

-static char *app = "SendText";
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

-static char *synopsis = "Send a Text Message";
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/options.h"
+#include "asterisk/app.h"

-static char *descrip = 
-"  SendText(text): Sends text to client.  If the client\n"
-"does not support text transport, and  there  exists  a  step  with\n"
-"priority  n + 101,  then  execution  will  continue  at that step.\n"
-"Otherwise, execution will continue at  the  next  priority  level.\n"
-"SendText only returns 0  if  the  text  was  sent  correctly  or  if\n"
-"the channel  does  not  support text transport,  and -1 otherwise.\n";
+static const char *tdesc = "Send Text Applications";
+
+static const char *app = "SendText";
+
+static const char *synopsis = "Send a Text Message";
+
+static const char *descrip = 
+"  SendText(text[|options]): Sends text to current channel (callee).\n"
+"Result of transmission will be stored in the SENDTEXTSTATUS\n"
+"channel variable:\n"
+"      SUCCESS      Transmission succeeded\n"
+"      FAILURE      Transmission failed\n"
+"      UNSUPPORTED  Text transmission not supported by channel\n"
+"\n"
+"At this moment, text is supposed to be 7 bit ASCII in most channels.\n"
+"The option string many contain the following character:\n"
+"'j' -- jump to n+101 priority if the channel doesn't support\n"
+"       text transport\n";

 STANDARD_LOCAL_USER;

@@ -45,30 +71,64 @@ static int sendtext_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
 	struct localuser *u;
-	if (!data || !strlen((char *)data)) {
-		ast_log(LOG_WARNING, "SendText requires an argument (text)\n");
+	char *status = "UNSUPPORTED";
+	char *parse = NULL;
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(text);
+		AST_APP_ARG(options);
+	);
+		
+	LOCAL_USER_ADD(u);	
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "SendText requires an argument (text[|options])\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
+	} else {
+		parse = ast_strdupa(data);
+		if (!parse) {
+			ast_log(LOG_ERROR, "Out of memory!\n");
+			LOCAL_USER_REMOVE(u);
+			return -1;
+		}
 	}
-	LOCAL_USER_ADD(u);
+	
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
 	ast_mutex_lock(&chan->lock);
-	if (!chan->pvt->send_text) {
+	if (!chan->tech->send_text) {
 		ast_mutex_unlock(&chan->lock);
 		/* Does not support transport */
-		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-			chan->priority += 100;
+		if (priority_jump || option_priority_jumping)
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
 		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
+	status = "FAILURE";
 	ast_mutex_unlock(&chan->lock);
-	res = ast_sendtext(chan, (char *)data);
+	res = ast_sendtext(chan, args.text);
+	if (!res)
+		status = "SUCCESS";
+	pbx_builtin_setvar_helper(chan, "SENDTEXTSTATUS", status);
 	LOCAL_USER_REMOVE(u);
-	return res;
+	return 0;
 }

 int unload_module(void)
 {
+	int res;
+	
+	res = ast_unregister_application(app);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
@@ -78,13 +138,15 @@ int load_module(void)

 char *description(void)
 {
-	return tdesc;
+	return (char *) tdesc;
 }

 int usecount(void)
 {
 	int res;
+
 	STANDARD_USECOUNT(res);
+
 	return res;
 }

--- a/apps/app_setcallerid.c
+++ b/apps/app_setcallerid.c
@@ -1,27 +1,45 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * App to set callerid
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to set callerid
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/image.h>
-#include <asterisk/callerid.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/callerid.h"

 static char *app2 = "SetCallerPres";

@@ -31,25 +49,9 @@ STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

-static struct {
-	int val;
-	char *name;
-} preses[] = {
-	{  AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED, "allowed_not_screened" },
-	{  AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN, "allowed_passed_screen" },
-	{  AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN, "allowed_failed_screen" },
-	{  AST_PRES_ALLOWED_NETWORK_NUMBER, "allowed" },
-	{  AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED	, "prohib_not_screened" },
-	{  AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN, "prohib_passed_screen" },
-	{  AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN, "prohib_failed_screen" },
-	{  AST_PRES_PROHIB_NETWORK_NUMBER, "prohib" },
-	{  AST_PRES_NUMBER_NOT_AVAILABLE, "unavailable" },
-};
-
 static char *descrip2 = 
-"  SetCallerPres(presentation): Set Caller*ID presentation on\n"
-"a call to a new value.  Sets ANI as well if a flag is used.\n"
-"Always returns 0.  Valid presentations are:\n"
+"  SetCallerPres(presentation): Set Caller*ID presentation on a call.\n"
+"  Valid presentations are:\n"
 "\n"
 "      allowed_not_screened    : Presentation Allowed, Not Screened\n"
 "      allowed_passed_screen   : Presentation Allowed, Passed Screen\n" 
@@ -65,33 +67,23 @@ static char *descrip2 =

 static int setcallerid_pres_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
-	char tmp[256] = "";
 	struct localuser *u;
-	int x;
-	char *opts;
 	int pres = -1;
-	if (data)
-		strncpy(tmp, (char *)data, sizeof(tmp) - 1);
-	opts = strchr(tmp, '|');
-	if (opts) {
-		*opts = '\0';
-		opts++;
-	}
-	for (x=0;x<sizeof(preses) / sizeof(preses[0]);x++) {
-		if (!strcasecmp(preses[x].name, tmp)) {
-			pres = preses[x].val;
-			break;
-		}
-	}
+
+	LOCAL_USER_ADD(u);
+	
+	pres = ast_parse_caller_presentation(data);
+
 	if (pres < 0) {
-		ast_log(LOG_WARNING, "'%s' is not a valid presentation (see 'show application SetCallerPres')\n", tmp);
+		ast_log(LOG_WARNING, "'%s' is not a valid presentation (see 'show application SetCallerPres')\n",
+			(char *) data);
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
-	LOCAL_USER_ADD(u);
-	chan->callingpres = pres;
+	
+	chan->cid.cid_pres = pres;
 	LOCAL_USER_REMOVE(u);
-	return res;
+	return 0;
 }


@@ -104,17 +96,32 @@ static char *synopsis = "Set CallerID";

 static char *descrip = 
 "  SetCallerID(clid[|a]): Set Caller*ID on a call to a new\n"
-"value.  Sets ANI as well if a flag is used.  Always returns 0\n";
+"value.  Sets ANI as well if a flag is used. \n";

 static int setcallerid_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
-	char tmp[256] = "";
+	char *tmp = NULL;
+	char name[256];
+	char num[256];
 	struct localuser *u;
 	char *opt;
 	int anitoo = 0;
-	if (data)
-		strncpy(tmp, (char *)data, sizeof(tmp) - 1);
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "SetCallerID requires an argument!\n");
+		return 0;
+	}
+	
+	LOCAL_USER_ADD(u);
+	
+	tmp = ast_strdupa(data);
+	if (!tmp) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	
 	opt = strchr(tmp, '|');
 	if (opt) {
 		*opt = '\0';
@@ -122,23 +129,35 @@ static int setcallerid_exec(struct ast_channel *chan, void *data)
 		if (*opt == 'a')
 			anitoo = 1;
 	}
-	LOCAL_USER_ADD(u);
-	ast_set_callerid(chan, strlen(tmp) ? tmp : NULL, anitoo);
+	
+	ast_callerid_split(tmp, name, sizeof(name), num, sizeof(num));
+	ast_set_callerid(chan, num, name, anitoo ? num : NULL);
+
 	LOCAL_USER_REMOVE(u);
+	
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app2);
+	res |= ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	ast_unregister_application(app2);
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
 {
-	ast_register_application(app2, setcallerid_pres_exec, synopsis2, descrip2);
-	return ast_register_application(app, setcallerid_exec, synopsis, descrip);
+	int res;
+	
+	res = ast_register_application(app2, setcallerid_pres_exec, synopsis2, descrip2);
+	res |= ast_register_application(app, setcallerid_exec, synopsis, descrip);
+
+	return res;
 }

 char *description(void)
--- a/apps/app_setcdruserfield.c
+++ b/apps/app_setcdruserfield.c
@@ -1,28 +1,45 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Applictions connected with CDR engine
- * 
- * Copyright (C) 2003, Digium
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Justin Huff <jjhuff@mspin.net>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Applictions connected with CDR engine
+ * 
+ * \ingroup applications
 */

 #include <sys/types.h>
-#include <asterisk/channel.h>
-#include <asterisk/cdr.h>
-#include <asterisk/module.h>
-#include <asterisk/pbx.h>
-#include <asterisk/logger.h>
-#include <asterisk/config.h>
-#include <asterisk/manager.h>
-#include <asterisk/utils.h>
 #include <stdlib.h>
 #include <string.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/channel.h"
+#include "asterisk/cdr.h"
+#include "asterisk/module.h"
+#include "asterisk/pbx.h"
+#include "asterisk/logger.h"
+#include "asterisk/config.h"
+#include "asterisk/manager.h"
+#include "asterisk/utils.h"
+

 static char *tdesc = "CDR user field apps";

@@ -35,8 +52,7 @@ static char *setcdruserfield_descrip =
               "       can use for data not stored anywhere else in the record.\n"
               "       CDR records can be used for billing or storing other arbitrary data\n"
               "       (I.E. telephone survey responses)\n"
-               "       Also see AppendCDRUserField().\n"
-               "       Always returns 0\n";
+               "       Also see AppendCDRUserField().\n";

 		
 static char *setcdruserfield_app = "SetCDRUserField";
@@ -51,8 +67,7 @@ static char *appendcdruserfield_descrip =
               "       can use for data not stored anywhere else in the record.\n"
               "       CDR records can be used for billing or storing other arbitrary data\n"
               "       (I.E. telephone survey responses)\n"
-               "       Also see SetCDRUserField().\n"
-               "       Always returns 0\n";
+               "       Also see SetCDRUserField().\n";
 		
 static char *appendcdruserfield_app = "AppendCDRUserField";
 static char *appendcdruserfield_synopsis = "Append to the CDR user field";
@@ -76,13 +91,7 @@ static int action_setcdruserfield(struct mansession *s, struct message *m)
 		astman_send_error(s, m, "No UserField specified");
 		return 0;
 	}
-	c = ast_channel_walk_locked(NULL);
-	while (c) {
-		if (!strcasecmp(c->name, channel))
-			break;
-		ast_mutex_unlock(&c->lock);
-		c = ast_channel_walk_locked(c);
-	}
+	c = ast_get_channel_by_name_locked(channel);
 	if (!c) {
 		astman_send_error(s, m, "No such channel");
 		return 0;
@@ -101,9 +110,9 @@ static int setcdruserfield_exec(struct ast_channel *chan, void *data)
 	struct localuser *u;
 	int res = 0;
 	
-	LOCAL_USER_ADD(u)
-	if (chan->cdr && data) 
-	{
+	LOCAL_USER_ADD(u);
+
+	if (chan->cdr && data) {
 		ast_cdr_setuserfield(chan, (char*)data);
 	}

@@ -117,9 +126,9 @@ static int appendcdruserfield_exec(struct ast_channel *chan, void *data)
 	struct localuser *u;
 	int res = 0;
 	
-	LOCAL_USER_ADD(u)
-	if (chan->cdr && data) 
-	{
+	LOCAL_USER_ADD(u);
+
+	if (chan->cdr && data) {
 		ast_cdr_appenduserfield(chan, (char*)data);
 	}

@@ -131,19 +140,24 @@ static int appendcdruserfield_exec(struct ast_channel *chan, void *data)
 int unload_module(void)
 {
 	int res;
-	STANDARD_HANGUP_LOCALUSERS;
+	
 	res = ast_unregister_application(setcdruserfield_app);
 	res |= ast_unregister_application(appendcdruserfield_app);
-	ast_manager_unregister("SetCDRUserField");
+	res |= ast_manager_unregister("SetCDRUserField");
+
+	STANDARD_HANGUP_LOCALUSERS;
+
 	return res;
 }

 int load_module(void)
 {
 	int res;
+
 	res = ast_register_application(setcdruserfield_app, setcdruserfield_exec, setcdruserfield_synopsis, setcdruserfield_descrip);
 	res |= ast_register_application(appendcdruserfield_app, appendcdruserfield_exec, appendcdruserfield_synopsis, appendcdruserfield_descrip);
-	ast_manager_register("SetCDRUserField", EVENT_FLAG_CALL, action_setcdruserfield, "Set the CDR UserField");
+	res |= ast_manager_register("SetCDRUserField", EVENT_FLAG_CALL, action_setcdruserfield, "Set the CDR UserField");
+	
 	return res;
 }

--- a/apps/app_setcidname.c
+++ b/apps/app_setcidname.c
@@ -1,28 +1,46 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * App to set callerid
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to set callerid
+ *
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/image.h>
-#include <asterisk/callerid.h>
-#include <asterisk/utils.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/callerid.h"
+#include "asterisk/utils.h"

 static char *tdesc = "Set CallerID Name";

@@ -34,7 +52,9 @@ static char *descrip =
 "  SetCIDName(cname[|a]): Set Caller*ID Name on a call to a new\n"
 "value, while preserving the original Caller*ID number.  This is\n"
 "useful for providing additional information to the called\n"
-"party. Sets ANI as well if a flag is used.  Always returns 0\n";
+"party. \n"
+"SetCIDName has been deprecated in favor of the function\n"
+"CALLERID(name)\n";

 STANDARD_LOCAL_USER;

@@ -42,46 +62,51 @@ LOCAL_USER_DECL;

 static int setcallerid_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
-	char tmp[256] = "";
-	char oldcid[256] = "", *l, *n;
-	char newcid[256] = "";
+	char *tmp = NULL;	
 	struct localuser *u;
 	char *opt;
-	int anitoo = 0;
-	if (data)
-		strncpy(tmp, (char *)data, sizeof(tmp) - 1);
+	static int deprecation_warning = 0;
+
+	if (!deprecation_warning) {
+		ast_log(LOG_WARNING, "SetCIDName is deprecated, please use Set(CALLERID(name)=value) instead.\n");
+		deprecation_warning = 1;
+	}
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_ERROR, "SetCIDName requires an argument!\n");
+		return 0;	
+	}
+
+	LOCAL_USER_ADD(u);
+
+	tmp = ast_strdupa(data);
+	if (!tmp) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	
 	opt = strchr(tmp, '|');
 	if (opt) {
 		*opt = '\0';
-		opt++;
-		if (*opt == 'a')
-			anitoo = 1;
 	}
-	LOCAL_USER_ADD(u);
-	if (chan->callerid) {
-		strncpy(oldcid, chan->callerid, sizeof(oldcid) - 1);
-		ast_callerid_parse(oldcid, &n, &l);
-		n = tmp;
-		if (!ast_strlen_zero(n)) {
-			if (l && !ast_strlen_zero(l))
-				snprintf(newcid, sizeof(newcid), "\"%s\" <%s>", n, l);
-			else
-				strncpy(newcid, tmp, sizeof(newcid) - 1);
-		} else if (l && !ast_strlen_zero(l)) {
-			strncpy(newcid, l, sizeof(newcid) - 1);
-		}
-	} else
-		strncpy(newcid, tmp, sizeof(newcid) - 1);
-	ast_set_callerid(chan, !ast_strlen_zero(newcid) ? newcid : NULL, anitoo);
+	
+	ast_set_callerid(chan, NULL, tmp, NULL);
+
 	LOCAL_USER_REMOVE(u);
-	return res;
+	
+	return 0;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_setcidnum.c
+++ b/apps/app_setcidnum.c
@@ -1,29 +1,47 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * App to set callerid
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
 * Oliver Daudey <traveler@xs4all.nl>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to set callerid number
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/image.h>
-#include <asterisk/callerid.h>
-#include <asterisk/utils.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/callerid.h"
+#include "asterisk/utils.h"

 static char *tdesc = "Set CallerID Number";

@@ -35,7 +53,9 @@ static char *descrip =
 "  SetCIDNum(cnum[|a]): Set Caller*ID Number on a call to a new\n"
 "value, while preserving the original Caller*ID name.  This is\n"
 "useful for providing additional information to the called\n"
-"party. Sets ANI as well if a flag is used.  Always returns 0\n";
+"party. Sets ANI as well if a flag is used.\n"
+"SetCIDNum has been deprecated in favor of the function\n"
+"CALLERID(number)\n";

 STANDARD_LOCAL_USER;

@@ -44,14 +64,24 @@ LOCAL_USER_DECL;
 static int setcallerid_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
-	char tmp[256] = "";
-	char oldcid[256] = "", *l, *n;
-	char newcid[256] = "";
 	struct localuser *u;
 	char *opt;
 	int anitoo = 0;
+	char *tmp = NULL;
+	static int deprecation_warning = 0;
+
+	LOCAL_USER_ADD(u);
+	
+	if (!deprecation_warning) {
+		ast_log(LOG_WARNING, "SetCIDNum is deprecated, please use Set(CALLERID(number)=value) instead.\n");
+		deprecation_warning = 1;
+	}
+
 	if (data)
-		strncpy(tmp, (char *)data, sizeof(tmp) - 1);
+		tmp = ast_strdupa(data);
+	else
+		tmp = "";
+	
 	opt = strchr(tmp, '|');
 	if (opt) {
 		*opt = '\0';
@@ -59,30 +89,23 @@ static int setcallerid_exec(struct ast_channel *chan, void *data)
 		if (*opt == 'a')
 			anitoo = 1;
 	}
-	LOCAL_USER_ADD(u);
-	if (chan->callerid) {
-		strncpy(oldcid, chan->callerid, sizeof(oldcid) - 1);
-		ast_callerid_parse(oldcid, &n, &l);
-		l = tmp;
-		if (!ast_strlen_zero(l)) {
-			if (n && !ast_strlen_zero(n))
-				snprintf(newcid, sizeof(newcid), "\"%s\" <%s>", n, l);
-			else
-				strncpy(newcid, tmp, sizeof(newcid) - 1);
-		} else if (n && !ast_strlen_zero(n)) {
-			strncpy(newcid, n, sizeof(newcid) - 1);
-		}
-	} else
-		strncpy(newcid, tmp, sizeof(newcid) - 1);
-	ast_set_callerid(chan, !ast_strlen_zero(newcid) ? newcid : NULL, anitoo);
+	
+	ast_set_callerid(chan, tmp, NULL, anitoo ? tmp : NULL);
+
 	LOCAL_USER_REMOVE(u);
+	
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+	
+	res = ast_unregister_application(app);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_setrdnis.c
+++ b/apps/app_setrdnis.c
@@ -0,0 +1,132 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ * Oliver Daudey <traveler@xs4all.nl>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to set rdnis
+ *
+ * \ingroup applications
+ */
+ 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/callerid.h"
+#include "asterisk/utils.h"
+
+static char *tdesc = "Set RDNIS Number";
+
+static char *app = "SetRDNIS";
+
+static char *synopsis = "Set RDNIS Number";
+
+static char *descrip = 
+"  SetRDNIS(cnum): Set RDNIS Number on a call to a new\n"
+"value.\n"
+"SetRDNIS has been deprecated in favor of the function\n"
+"CALLERID(rdnis)\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int setrdnis_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	char *opt, *n, *l;
+	char *tmp = NULL;
+	static int deprecation_warning = 0;
+
+	LOCAL_USER_ADD(u);
+	
+	if (!deprecation_warning) {
+		ast_log(LOG_WARNING, "SetRDNIS is deprecated, please use Set(CALLERID(rdnis)=value) instead.\n");
+		deprecation_warning = 1;
+	}
+
+	if (data)
+		tmp = ast_strdupa(data);
+	else
+		tmp = "";	
+
+	opt = strchr(tmp, '|');
+	if (opt)
+		*opt = '\0';
+	
+	n = l = NULL;
+	ast_callerid_parse(tmp, &n, &l);
+	if (l) {
+		ast_shrink_phone_number(l);
+		ast_mutex_lock(&chan->lock);
+		if (chan->cid.cid_rdnis)
+			free(chan->cid.cid_rdnis);
+		chan->cid.cid_rdnis = (l[0]) ? strdup(l) : NULL;
+		ast_mutex_unlock(&chan->lock);
+	}
+
+	LOCAL_USER_REMOVE(u);
+	
+	return 0;
+}
+
+int unload_module(void)
+{
+	int res;
+	
+	res = ast_unregister_application(app);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, setrdnis_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_settransfercapability.c
+++ b/apps/app_settransfercapability.c
@@ -0,0 +1,145 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2005, Frank Sautter, levigo holding gmbh, www.levigo.de
+ *
+ * Frank Sautter - asterisk+at+sautter+dot+com 
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to set the ISDN Transfer Capability
+ * 
+ * \ingroup applications
+ */
+ 
+#include <string.h>
+#include <stdlib.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/options.h"
+#include "asterisk/transcap.h"
+
+
+static char *app = "SetTransferCapability";
+
+static char *synopsis = "Set ISDN Transfer Capability";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static struct {	int val; char *name; } transcaps[] = {
+	{ AST_TRANS_CAP_SPEECH,				"SPEECH" },
+	{ AST_TRANS_CAP_DIGITAL,			"DIGITAL" },
+	{ AST_TRANS_CAP_RESTRICTED_DIGITAL,	"RESTRICTED_DIGITAL" },
+	{ AST_TRANS_CAP_3_1K_AUDIO,			"3K1AUDIO" },
+	{ AST_TRANS_CAP_DIGITAL_W_TONES,	"DIGITAL_W_TONES" },
+	{ AST_TRANS_CAP_VIDEO,				"VIDEO" },
+};
+
+static char *descrip = 
+"  SetTransferCapability(transfercapability): Set the ISDN Transfer \n"
+"Capability of a call to a new value.\n"
+"Valid Transfer Capabilities are:\n"
+"\n"
+"  SPEECH             : 0x00 - Speech (default, voice calls)\n"
+"  DIGITAL            : 0x08 - Unrestricted digital information (data calls)\n"
+"  RESTRICTED_DIGITAL : 0x09 - Restricted digital information\n"
+"  3K1AUDIO           : 0x10 - 3.1kHz Audio (fax calls)\n"
+"  DIGITAL_W_TONES    : 0x11 - Unrestricted digital information with tones/announcements\n"
+"  VIDEO              : 0x18 - Video:\n"
+"\n"
+;
+
+static int settransfercapability_exec(struct ast_channel *chan, void *data)
+{
+	char *tmp = NULL;
+	struct localuser *u;
+	int x;
+	char *opts;
+	int transfercapability = -1;
+	
+	LOCAL_USER_ADD(u);
+	
+	if (data)
+		tmp = ast_strdupa(data);
+	else
+		tmp = "";
+
+	opts = strchr(tmp, '|');
+	if (opts)
+		*opts = '\0';
+	
+	for (x = 0; x < (sizeof(transcaps) / sizeof(transcaps[0])); x++) {
+		if (!strcasecmp(transcaps[x].name, tmp)) {
+			transfercapability = transcaps[x].val;
+			break;
+		}
+	}
+	if (transfercapability < 0) {
+		ast_log(LOG_WARNING, "'%s' is not a valid transfer capability (see 'show application SetTransferCapability')\n", tmp);
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+		
+	chan->transfercapability = (unsigned short)transfercapability;
+	
+	if (option_verbose > 2)
+		ast_verbose(VERBOSE_PREFIX_3 "Setting transfer capability to: 0x%.2x - %s.\n", transfercapability, tmp);			
+	
+	LOCAL_USER_REMOVE(u);
+
+	return 0;
+}
+
+
+int unload_module(void)
+{
+	int res;
+	
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, settransfercapability_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return synopsis;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_skel.c
+++ b/apps/app_skel.c
@@ -1,59 +1,135 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Skeleton application
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) <Year>, <Your Name Here>
 *
- * Mark Spencer <markster@linux-support.net>
+ * <Your Name Here> <<You Email Here>>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
 */

-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/lock.h>
+/*! \file
+ *
+ * \brief Skeleton application
+ * 
+ * This is a skeleton for development of an Asterisk application */
+ * \ingroup applications
+ */
+
+#include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/app.h"
+
 static char *tdesc = "Trivial skeleton Application";
-static char *app = "skel";
+static char *app = "Skel";
 static char *synopsis = 
-"  This is a skeleton application that shows you the basic structure to create your\n"
-"own asterisk applications.\n";
+"Skeleton application.";
+static char *descrip = "This application is a template to build other applications from.\n"
+ " It shows you the basic structure to create your own Asterisk applications.\n";
+
+#define OPTION_A	(1 << 0)	/* Option A */
+#define OPTION_B	(1 << 1)	/* Option B(n) */
+#define OPTION_C	(1 << 2)	/* Option C(str) */
+#define OPTION_NULL	(1 << 3)	/* Dummy Termination */
+
+AST_DECLARE_OPTIONS(app_opts,{
+	['a'] = { OPTION_A },
+	['b'] = { OPTION_B, 1 },
+	['c'] = { OPTION_C, 2 }
+});

 STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

-static int skel_exec(struct ast_channel *chan, void *data)
+static int app_exec(struct ast_channel *chan, void *data)
 {
-	int res=0;
+	int res = 0;
+	struct ast_flags flags;
 	struct localuser *u;
-	if (!data) {
-		ast_log(LOG_WARNING, "skel requires an argument (filename)\n");
+	char *options=NULL;
+	char *dummy = NULL;
+	char *args;
+	int argc = 0;
+	char *opts[2];
+	char *argv[2];
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "%s requires an argument (dummy|[options])\n",app);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
+
 	LOCAL_USER_ADD(u);
+	
 	/* Do our thing here */
+
+	/* We need to make a copy of the input string if we are going to modify it! */
+	args = ast_strdupa(data);	
+	if (!args) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	
+	if ((argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0])))) {
+		dummy = argv[0];
+		options = argv[1];
+		ast_parseoptions(app_opts, &flags, opts, options);
+	}
+
+	if (!ast_strlen_zero(dummy)) 
+		ast_log(LOG_NOTICE, "Dummy value is : %s\n", dummy);
+
+	if (ast_test_flag(&flags, OPTION_A))
+		ast_log(LOG_NOTICE, "Option A is set\n");
+
+	if (ast_test_flag(&flags, OPTION_B))
+		ast_log(LOG_NOTICE,"Option B is set with : %s\n", opts[0] ? opts[0] : "<unspecified>");
+
+	if (ast_test_flag(&flags, OPTION_C))
+		ast_log(LOG_NOTICE,"Option C is set with : %s\n", opts[1] ? opts[1] : "<unspecified>");
+
 	LOCAL_USER_REMOVE(u);
+	
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
 {
-	return ast_register_application(app, skel_exec, synopsis, tdesc);
+	return ast_register_application(app, app_exec, synopsis, descrip);
 }

 char *description(void)
--- a/apps/app_sms.c
+++ b/apps/app_sms.c
--- a/apps/app_softhangup.c
+++ b/apps/app_softhangup.c
@@ -1,34 +1,54 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * SoftHangup application
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief SoftHangup application
+ * 
+ * \ingroup applications
 */

-#include <sys/types.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/lock.h>
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
+#include <unistd.h>
+#include <sys/types.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"

 static char *synopsis = "Soft Hangup Application";

 static char *tdesc = "Hangs up the requested channel";

-static char *desc = "  SoftHangup(Technology/resource)\n"
-"Hangs up the requested channel.  Always returns 0\n";
+static char *desc = "  SoftHangup(Technology/resource|options)\n"
+"Hangs up the requested channel.  If there are no channels to hangup,\n"
+"the application will report it.\n"
+"- 'options' may contain the following letter:\n"
+"     'a' : hang up all channels on a specified device instead of a single resource\n";

 static char *app = "SoftHangup";

@@ -40,21 +60,46 @@ static int softhangup_exec(struct ast_channel *chan, void *data)
 {
 	struct localuser *u;
 	struct ast_channel *c=NULL;
-	if (!data) {
+	char *options, *cut, *cdata, *match;
+	char name[AST_CHANNEL_NAME] = "";
+	int all = 0;
+	
+	if (ast_strlen_zero(data)) {
                ast_log(LOG_WARNING, "SoftHangup requires an argument (Technology/resource)\n");
 		return 0;
 	}
+	
 	LOCAL_USER_ADD(u);
+
+	cdata = ast_strdupa(data);
+	match = strsep(&cdata, "|");
+	options = strsep(&cdata, "|");
+	all = options && strchr(options,'a');
 	c = ast_channel_walk_locked(NULL);
 	while (c) {
-		if (!strcasecmp(c->name, data)) {
-			ast_softhangup(c, AST_SOFTHANGUP_EXPLICIT);
-			ast_mutex_unlock(&c->lock);
-			break;
-		}
+		strncpy(name, c->name, sizeof(name)-1);
 		ast_mutex_unlock(&c->lock);
+		/* XXX watch out, i think it is wrong to access c-> after unlocking! */
+		if (all) {
+			/* CAPI is set up like CAPI[foo/bar]/clcnt */ 
+			if (!strcmp(c->type,"CAPI")) 
+				cut = strrchr(name,'/');
+			/* Basically everything else is Foo/Bar-Z */
+			else
+				cut = strchr(name,'-');
+			/* Get rid of what we've cut */
+			if (cut)
+				*cut = 0;
+		}
+		if (!strcasecmp(name, match)) {
+			ast_log(LOG_WARNING, "Soft hanging %s up.\n",c->name);
+			ast_softhangup(c, AST_SOFTHANGUP_EXPLICIT);
+			if(!all)
+				break;
+		}
 		c = ast_channel_walk_locked(c);
 	}
+	
 	LOCAL_USER_REMOVE(u);

 	return 0;
@@ -62,8 +107,13 @@ static int softhangup_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_sql_postgres.c
+++ b/apps/app_sql_postgres.c
@@ -1,36 +1,50 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Connect to PostgreSQL
- * 
 * Copyright (C) 2002, Christos Ricudis
 *
 * Christos Ricudis <ricudis@itc.auth.gr>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Connect to PostgreSQL
+ * 
+ * \ingroup applications
 */

-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/linkedlists.h>
-#include <asterisk/chanvars.h>
-#include <asterisk/lock.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
-#include <stdlib.h>
 #include <sys/types.h>
 #include <stdio.h>
 #include <unistd.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/linkedlists.h"
+#include "asterisk/chanvars.h"
+#include "asterisk/lock.h"
+
 #include "libpq-fe.h"

-#define EXTRA_LOG 0
-
-
 static char *tdesc = "Simple PostgreSQL Interface";

 static char *app = "PGSQL";
@@ -189,7 +203,7 @@ static int del_identifier(int identifier,int identifier_type) {
 		AST_LIST_TRAVERSE(headp,i,entries) {
 			if ((i->identifier==identifier) && 
 			    (i->identifier_type==identifier_type)) {
-				AST_LIST_REMOVE(headp,i,ast_PGSQL_id,entries);
+				AST_LIST_REMOVE(headp,i,entries);
 				free(i);
 				found=1;
 				break;
@@ -224,7 +238,7 @@ static int aPGSQL_connect(struct ast_channel *chan, void *data) {
 	s1=malloc(l);
 	strncpy(s1, data, l -1);
 	stringp=s1;
-	strsep(&stringp," "); // eat the first token, we already know it :P 
+	strsep(&stringp," "); /* eat the first token, we already know it :P  */
 	var=strsep(&stringp," ");
 	optionstring=strsep(&stringp,"\n");
 		
@@ -265,9 +279,9 @@ static int aPGSQL_query(struct ast_channel *chan, void *data) {
 	s2=malloc(l);
 	strncpy(s1, data, l - 1);
 	stringp=s1;
-	strsep(&stringp," "); // eat the first token, we already know it :P 
+	strsep(&stringp," "); /* eat the first token, we already know it :P  */
 	s3=strsep(&stringp," ");
-	while (1) {	// ugly trick to make branches with break;
+	while (1) {	/* ugly trick to make branches with break; */
 		var=s3;
 		s4=strsep(&stringp," ");
 		id=atoi(s4);
@@ -328,10 +342,10 @@ static int aPGSQL_fetch(struct ast_channel *chan, void *data) {
 	s2=malloc(l);
 	strncpy(s1, data, l - 1);
 	stringp=s1;
-	strsep(&stringp," "); // eat the first token, we already know it :P 
+	strsep(&stringp," "); /* eat the first token, we already know it :P  */
 	fetchid_var=strsep(&stringp," ");
-	while (1) {	// ugly trick to make branches with break;
-	  var=fetchid_var; // fetchid
+	while (1) {	/* ugly trick to make branches with break; */
+	  var=fetchid_var; /* fetchid */
 		fnd=0;
 		
 		AST_LIST_TRAVERSE(headp,variables,entries) {
@@ -348,19 +362,19 @@ static int aPGSQL_fetch(struct ast_channel *chan, void *data) {
 		}

 		s4=strsep(&stringp," ");
-		id=atoi(s4); // resultid
+		id=atoi(s4); /* resultid */
 		if ((PGSQLres=find_identifier(id,AST_PGSQL_ID_RESID))==NULL) {
 			ast_log(LOG_WARNING,"Invalid result identifier %d passed in aPGSQL_fetch\n",id);
 			res=-1;
 			break;
 		}
-		id=atoi(s7); //fetchid
+		id=atoi(s7); /*fetchid */
 		if ((lalares=find_identifier(id,AST_PGSQL_ID_FETCHID))==NULL) {
-	    i=0;       // fetching the very first row
+	    i=0;       /* fetching the very first row */
 		} else {
 			i=*lalares;
 			free(lalares);
-	    del_identifier(id,AST_PGSQL_ID_FETCHID); // will re-add it a bit later
+	    del_identifier(id,AST_PGSQL_ID_FETCHID); /* will re-add it a bit later */
 		}

 	  if (i<PQntuples(PGSQLres)) {
@@ -381,11 +395,11 @@ static int aPGSQL_fetch(struct ast_channel *chan, void *data) {
 			pbx_builtin_setvar_helper(chan,s5,s6);
 		}
 			lalares=malloc(sizeof(int));
-	    *lalares = ++i; // advance to the next row
+	    *lalares = ++i; /* advance to the next row */
 	    id1 = add_identifier(AST_PGSQL_ID_FETCHID,lalares);
 		} else {
 	    ast_log(LOG_WARNING,"ast_PGSQL_fetch : EOF\n");
-	    id1 = 0; // no more rows
+	    id1 = 0; /* no more rows */
 		}
 		snprintf(s, sizeof(s), "%d", id1);
 	  ast_log(LOG_WARNING,"Setting var '%s' to value '%s'\n",fetchid_var,s);
@@ -411,7 +425,7 @@ static int aPGSQL_reset(struct ast_channel *chan, void *data) {
 	s1=malloc(l);
 	strncpy(s1, data, l - 1);
 	stringp=s1;
-	strsep(&stringp," "); // eat the first token, we already know it :P 
+	strsep(&stringp," "); /* eat the first token, we already know it :P  */
 	s3=strsep(&stringp," ");
 	id=atoi(s3);
 	if ((karoto=find_identifier(id,AST_PGSQL_ID_CONNID))==NULL) {
@@ -437,7 +451,7 @@ static int aPGSQL_clear(struct ast_channel *chan, void *data) {
 	s1=malloc(l);
 	strncpy(s1, data, l - 1);
 	stringp=s1;
-	strsep(&stringp," "); // eat the first token, we already know it :P 
+	strsep(&stringp," "); /* eat the first token, we already know it :P  */
 	s3=strsep(&stringp," ");
 	id=atoi(s3);
 	if ((karoto=find_identifier(id,AST_PGSQL_ID_RESID))==NULL) {
@@ -467,7 +481,7 @@ static int aPGSQL_disconnect(struct ast_channel *chan, void *data) {
 	s1=malloc(l);
 	strncpy(s1, data, l - 1);
 	stringp=s1;
-	strsep(&stringp," "); // eat the first token, we already know it :P 
+	strsep(&stringp," "); /* eat the first token, we already know it :P  */
 	s3=strsep(&stringp," ");
 	id=atoi(s3);
 	if ((karoto=find_identifier(id,AST_PGSQL_ID_CONNID))==NULL) {
@@ -485,23 +499,19 @@ static int aPGSQL_debug(struct ast_channel *chan, void *data) {
 	ast_log(LOG_WARNING,"Debug : %s\n",(char *)data);
 	return(0);
 }
-		
-	

 static int PGSQL_exec(struct ast_channel *chan, void *data)
 {
 	struct localuser *u;
 	int result;

-#if EXTRA_LOG
-	printf("PRSQL_exec: data=%s\n",(char*)data);
-#endif
-
-	if (!data) {
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "APP_PGSQL requires an argument (see manual)\n");
 		return -1;
 	}
+	
 	LOCAL_USER_ADD(u);
+
 	result=0;
 	
 	if (strncasecmp("connect",data,strlen("connect"))==0) {
@@ -524,14 +534,19 @@ static int PGSQL_exec(struct ast_channel *chan, void *data)
 	}
 		
 	LOCAL_USER_REMOVE(u);                                                                                
+	
 	return result;
-
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+	
+	return res;
 }

 int load_module(void)
--- a/apps/app_stack.c
+++ b/apps/app_stack.c
@@ -0,0 +1,197 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (c) 2004-2005 Tilghman Lesher <app_stack_v002@the-tilghman.com>.
+ *
+ * This code is released by the author with no restrictions on usage.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Stack applications Gosub, Return, etc.
+ * 
+ * \ingroup applications
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+#include "asterisk/options.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/chanvars.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/config.h"
+
+#define STACKVAR	"~GOSUB~STACK~"
+
+static const char *tdesc = "Stack Routines";
+
+static const char *app_gosub = "Gosub";
+static const char *app_gosubif = "GosubIf";
+static const char *app_return = "Return";
+static const char *app_pop = "StackPop";
+
+static const char *gosub_synopsis = "Jump to label, saving return address";
+static const char *gosubif_synopsis = "Jump to label, saving return address";
+static const char *return_synopsis = "Return from gosub routine";
+static const char *pop_synopsis = "Remove one address from gosub stack";
+
+static const char *gosub_descrip =
+"Gosub([[context|]exten|]priority)\n"
+"  Jumps to the label specified, saving the return address.\n"
+"  Returns 0 if the label exists or -1 otherwise.\n";
+static const char *gosubif_descrip =
+"Gosub(condition?labeliftrue[:labeliffalse])\n"
+"  If the condition is true, then jump to labeliftrue.  If false, jumps to\n"
+"labeliffalse, if specified.  In either case, a jump saves the return point\n"
+"in the dialplan, to be returned to with a Return.\n"
+"  Returns 0 if the label exists or -1 otherwise.\n";
+static const char *return_descrip =
+"Return()\n"
+"  Jumps to the last label in the stack, removing it.\n"
+"  Returns 0 if there's a label in the stack or -1 otherwise.\n";
+static const char *pop_descrip =
+"StackPop()\n"
+"  Removes last label in the stack, discarding it.\n"
+"  Always returns 0, even if the stack is empty.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int pop_exec(struct ast_channel *chan, void *data)
+{
+	pbx_builtin_setvar_helper(chan, STACKVAR, NULL);
+
+	return 0;
+}
+
+static int return_exec(struct ast_channel *chan, void *data)
+{
+	char *label = pbx_builtin_getvar_helper(chan, STACKVAR);
+
+	if (ast_strlen_zero(label)) {
+		ast_log(LOG_ERROR, "Return without Gosub: stack is empty\n");
+		return -1;
+	} else if (ast_parseable_goto(chan, label)) {
+		ast_log(LOG_WARNING, "No next statement after Gosub?\n");
+		return -1;
+	}
+
+	pbx_builtin_setvar_helper(chan, STACKVAR, NULL);
+	return 0;
+}
+
+static int gosub_exec(struct ast_channel *chan, void *data)
+{
+	char newlabel[AST_MAX_EXTENSION * 2 + 3 + 11];
+	struct localuser *u;
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_ERROR, "%s requires an argument: %s([[context|]exten|]priority)\n", app_gosub, app_gosub);
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+	snprintf(newlabel, sizeof(newlabel), "%s|%s|%d", chan->context, chan->exten, chan->priority + 1);
+
+	if (ast_parseable_goto(chan, data)) {
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	pbx_builtin_pushvar_helper(chan, STACKVAR, newlabel);
+	LOCAL_USER_REMOVE(u);
+
+	return 0;
+}
+
+static int gosubif_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	char *condition="", *label1, *label2, *args;
+	int res=0;
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "GosubIf requires an argument\n");
+		return 0;
+	}
+
+	args = ast_strdupa((char *)data);
+	if (!args) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	condition = strsep(&args, "?");
+	label1 = strsep(&args, ":");
+	label2 = args;
+
+	if (ast_true(condition)) {
+		if (label1) {
+			res = gosub_exec(chan, label1);
+		}
+	} else if (label2) {
+		res = gosub_exec(chan, label2);
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	ast_unregister_application(app_return);
+	ast_unregister_application(app_pop);
+	ast_unregister_application(app_gosubif);
+	ast_unregister_application(app_gosub);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return 0;
+}
+
+int load_module(void)
+{
+	ast_register_application(app_pop, pop_exec, pop_synopsis, pop_descrip);
+	ast_register_application(app_return, return_exec, return_synopsis, return_descrip);
+	ast_register_application(app_gosubif, gosubif_exec, gosubif_synopsis, gosubif_descrip);
+	ast_register_application(app_gosub, gosub_exec, gosub_synopsis, gosub_descrip);
+
+	return 0;
+}
+
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
--- a/apps/app_striplsd.c
+++ b/apps/app_striplsd.c
@@ -1,97 +0,0 @@
-/*
- * Asterisk -- A telephony toolkit for Linux.
- *
- * Skeleton application
- * 
- * Copyright (C) 1999, Mark Spencer
- *
- * Mark Spencer <markster@linux-support.net>
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License
- */
-
-#include <sys/types.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/lock.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <string.h>
-#include <stdlib.h>
-
-static char *tdesc = "Strip trailing digits";
-
-static char *descrip =
-"  StripLSD(count): Strips the trailing  'count'  digits  from  the  channel's\n"
-"associated extension. For example, the  number  5551212 when stripped with a\n"
-"count of 4 would be changed to 555.  This app always returns 0, and the PBX\n"
-"will continue processing at the next priority for the *new* extension.\n"
-"  So, for  example, if  priority 3 of 5551212  is  StripLSD 4, the next step\n"
-"executed will be priority 4 of 555.  If you switch into an  extension which\n"
-"has no first step, the PBX will treat it as though the user dialed an\n"
-"invalid extension.\n";
-
-static char *app = "StripLSD";
-
-static char *synopsis = "Strip Least Significant Digits";
-
-STANDARD_LOCAL_USER;
-
-LOCAL_USER_DECL;
-
-static int striplsd_exec(struct ast_channel *chan, void *data)
-{
-	char newexten[AST_MAX_EXTENSION] = "";
-	int maxbytes = 0;
-	int stripcount = 0;
-	int extlen = strlen(chan->exten);
-
-	maxbytes = sizeof(newexten) - 1;
-	if (data) {
-		stripcount = atoi(data);
-	}
-	if (!stripcount) {
-		ast_log(LOG_DEBUG, "Ignoring, since number of digits to strip is 0\n");
-		return 0;
-	}
-	if (extlen > stripcount) {
-		if (extlen - stripcount <= maxbytes) {
-			maxbytes = extlen - stripcount;
-		}
-		strncpy(newexten, chan->exten, maxbytes);
-	}
-	strncpy(chan->exten, newexten, sizeof(chan->exten)-1);
-	return 0;
-}
-
-int unload_module(void)
-{
-	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
-}
-
-int load_module(void)
-{
-	return ast_register_application(app, striplsd_exec, synopsis, descrip);
-}
-
-char *description(void)
-{
-	return tdesc;
-}
-
-int usecount(void)
-{
-	int res;
-	STANDARD_USECOUNT(res);
-	return res;
-}
-
-char *key()
-{
-	return ASTERISK_GPL_KEY;
-}
--- a/apps/app_substring.c
+++ b/apps/app_substring.c
@@ -1,134 +0,0 @@
-/*
- * Asterisk -- A telephony toolkit for Linux.
- *
- * Skeleton application
- * 
- * Copyright (C) 1999, Mark Spencer
- *
- * Mark Spencer <markster@linux-support.net>
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License
- */
-
-#include <sys/types.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/pbx.h>
-#include <asterisk/lock.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <string.h>
-#include <stdlib.h>
-
-static char *tdesc = "(Deprecated) Save substring digits in a given variable";
-
-static char *descrip =
-"  (Deprecated, use ${variable:a:b} instead)\n"
-"\n"
-"  SubString(variable=string_of_digits|count1|count2): Assigns the substring\n"
-"of string_of_digits to a given variable. Parameter count1 may be positive\n"
-"or negative. If it's positive then we skip the first count1 digits from the\n"
-"left. If it's negative, we move count1 digits counting from the end of\n"
-"the string to the left. Parameter count2 implies how many digits we are\n"
-"taking from the point that count1 placed us. If count2 is negative, then\n"
-"that many digits are omitted from the end.\n"
-"For example:\n"
-"exten => _NXXXXXX,1,SubString,test=2564286161|0|3\n"
-"assigns the area code (3 first digits) to variable test.\n"
-"exten => _NXXXXXX,1,SubString,test=2564286161|-7|7\n"
-"assigns the last 7 digits to variable test.\n"
-"exten => _NXXXXXX,1,SubString,test=2564286161|0|-4\n" 
-"assigns all but the last 4 digits to variable test.\n" 
-"If there are no parameters it'll return with -1.\n"
-"If there wrong parameters it go on and return with 0\n";
-
-static char *app = "SubString";
-
-static char *synopsis = "(Deprecated) Save substring digits in a given variable";
-
-STANDARD_LOCAL_USER;
-
-LOCAL_USER_DECL;
-
-static int substring_exec(struct ast_channel *chan, void *data)
-{
-  char newexten[AST_MAX_EXTENSION] = "";
-  char *count1, *count2;
-  char *first, *second, *stringp;
-
-  stringp=alloca(strlen(data)+1);
-  ast_log(LOG_WARNING, "The use of Substring application is deprecated. Please use ${variable:a:b} instead\n");
-  strncpy(stringp,data,strlen(data));
-  if (strchr(stringp,'|')&&strchr(stringp,'=')) {
-    int icount1,icount2;
-    first=strsep(&stringp,"=");
-    second=strsep(&stringp,"|");
-    count1=strsep(&stringp,"|");
-    count2=strsep(&stringp,"\0");
-    if (!first || !second || !count1 || !count2) {
-      ast_log(LOG_DEBUG, "Ignoring, since there is no argument: variable or string or count1 or count2\n");
-      return 0;
-    }
-    icount1=atoi(count1);
-    icount2=atoi(count2);
-	if (icount2<0) {
-	  icount2 = icount2 + strlen(second);
-    }
-    if (abs(icount1)>strlen(second)) {
-      ast_log(LOG_WARNING, "Limiting count1 parameter because it exceeds the length of the string\n");
-      if (icount1>=0)
-        icount1=strlen(second);
-      else
-        icount1=0;
-    }
-    if ((icount1<0 && icount2>-icount1) || (icount1>=0 && icount1+icount2>strlen(second))) {
-      ast_log(LOG_WARNING, "Limiting count2 parameter because it exceeds the length of the string\n");
-      if (icount1>=0)
-      	icount2=strlen(second)-icount1;
-      else
-      	icount2=strlen(second)+icount1;
-    }
-    if (first&&second) {
-      if (icount1>=0)
-        strncpy(newexten,second+icount1,icount2);
-      else
-        strncpy(newexten,second+strlen(second)+icount1,icount2);
-      pbx_builtin_setvar_helper(chan,first,newexten);
-    }
-  } else {
-    ast_log(LOG_DEBUG, "Ignoring, no parameters\n");
-  }
-  return 0;
-}
-
-int unload_module(void)
-{
-	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
-}
-
-int load_module(void)
-{
-	return ast_register_application(app, substring_exec, synopsis, descrip);
-}
-
-char *description(void)
-{
-	return tdesc;
-}
-
-int usecount(void)
-{
-	int res;
-	STANDARD_USECOUNT(res);
-	return res;
-}
-
-char *key()
-{
-	return ASTERISK_GPL_KEY;
-}
--- a/apps/app_system.c
+++ b/apps/app_system.c
@@ -1,29 +1,47 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Execute arbitrary system commands
- * 
- * Copyright (C) 1999-2004, Digium, Inc.
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Execute arbitrary system commands
+ * 
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/app.h>
 #include <stdlib.h>
+#include <stdio.h>
 #include <unistd.h>
 #include <string.h>
-#include <stdlib.h>
 #include <errno.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+#include "asterisk/options.h"
+
 static char *tdesc = "Generic System() application";

 static char *app = "System";
@@ -34,19 +52,34 @@ static char *synopsis = "Execute a system command";

 static char *synopsis2 = "Try executing a system command";

+static char *chanvar = "SYSTEMSTATUS";
+
 static char *descrip =
-"  System(command): Executes a command  by  using  system(). Returns -1 on\n"
-"failure to execute the specified command. If  the command itself executes\n"
-"but is in error, and if there exists a priority n + 101, where 'n' is the\n"
-"priority of the current instance, then  the  channel  will  be  setup  to\n"
-"continue at that priority level.  Otherwise, System returns 0.\n";
+"  System(command): Executes a command  by  using  system(). If the command\n"
+"fails, the console should report a fallthrough. \n"
+"Result of execution is returned in the SYSTEMSTATUS channel variable:\n"
+"   FAILURE	Could not execute the specified command\n"
+"   SUCCESS	Specified command successfully executed\n"
+"\n"
+"Old behaviour:\n"
+"If the command itself executes but is in error, and if there exists\n"
+"a priority n + 101, where 'n' is the priority of the current instance,\n"
+"then  the  channel  will  be  setup to continue at that priority level.\n"
+"Note that this jump functionality has been deprecated and will only occur\n"
+"if the global priority jumping option is enabled in extensions.conf.\n";

 static char *descrip2 =
-"  TrySystem(command): Executes a command  by  using  system(). Returns 0\n"
-"on any situation. If  the command itself executes but is in error, and if\n"
+"  TrySystem(command): Executes a command  by  using  system().\n"
+"on any situation.\n"
+"Result of execution is returned in the SYSTEMSTATUS channel variable:\n"
+"   FAILURE	Could not execute the specified command\n"
+"   SUCCESS	Specified command successfully executed\n"
+"   APPERROR	Specified command successfully executed, but returned error code\n"
+"\n"
+"Old behaviour:\nIf  the command itself executes but is in error, and if\n"
 "there exists a priority n + 101, where 'n' is the priority of the current\n"
 "instance, then  the  channel  will  be  setup  to continue at that\n"
-"priority level.  Otherwise, System returns 0.\n";
+"priority level.  Otherwise, System will terminate.\n";

 STANDARD_LOCAL_USER;

@@ -56,27 +89,40 @@ static int system_exec_helper(struct ast_channel *chan, void *data, int failmode
 {
 	int res=0;
 	struct localuser *u;
-	if (!data) {
+	
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "System requires an argument(command)\n");
+		pbx_builtin_setvar_helper(chan, chanvar, "FAILURE");
 		return failmode;
 	}
+
 	LOCAL_USER_ADD(u);
+
 	/* Do our thing here */
 	res = ast_safe_system((char *)data);
 	if ((res < 0) && (errno != ECHILD)) {
 		ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
+		pbx_builtin_setvar_helper(chan, chanvar, "FAILURE");
 		res = failmode;
 	} else if (res == 127) {
 		ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
+		pbx_builtin_setvar_helper(chan, chanvar, "FAILURE");
 		res = failmode;
 	} else {
-		if (res < 0)
+		if (res < 0) 
 			res = 0;
-		if (res && ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid)) 
-			chan->priority+=100;
+		if (option_priority_jumping && res)
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+
+		if (res != 0)
+			pbx_builtin_setvar_helper(chan, chanvar, "APPERROR");
+		else
+			pbx_builtin_setvar_helper(chan, chanvar, "SUCCESS");
 		res = 0;
-	}
+	} 
+
 	LOCAL_USER_REMOVE(u);
+
 	return res;
 }

@@ -92,15 +138,24 @@ static int trysystem_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+	res |= ast_unregister_application(app2);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	ast_unregister_application(app2);
-	return ast_unregister_application(app);
+
+	return res;
 }

 int load_module(void)
 {
-	ast_register_application(app2, trysystem_exec, synopsis2, descrip2);
-	return ast_register_application(app, system_exec, synopsis, descrip);
+	int res;
+
+	res = ast_register_application(app2, trysystem_exec, synopsis2, descrip2);
+	res |= ast_register_application(app, system_exec, synopsis, descrip);
+
+	return res;
 }

 char *description(void)
--- a/apps/app_talkdetect.c
+++ b/apps/app_talkdetect.c
@@ -1,27 +1,45 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Playback a file with audio detect
- * 
- * Copyright (C) 2004, Digium, Inc.
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Playback a file with audio detect
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/utils.h>
-#include <asterisk/dsp.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/utils.h"
+#include "asterisk/dsp.h"

 static char *tdesc = "Playback with Talk Detection";

@@ -38,8 +56,7 @@ static char *descrip =
 "yet less than 'max' ms is followed by silence for at least 'sil' ms then\n"
 "the audio playback is aborted and processing jumps to the 'talk' extension\n"
 "if available.  If unspecified, sil, min, and max default to 1000, 100, and\n"
-"infinity respectively.  Returns -1 on hangup, and 0 on successful playback\n"
-"completion with no exit conditions.\n";
+"infinity respectively.\n";

 STANDARD_LOCAL_USER;

@@ -49,23 +66,33 @@ static int background_detect_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
 	struct localuser *u;
-	char tmp[256];
+	char *tmp;
 	char *options;
 	char *stringp;
 	struct ast_frame *fr;
 	int notsilent=0;
-	struct timeval start = { 0, 0}, end = {0, 0};
+	struct timeval start = { 0, 0};
 	int sil = 1000;
 	int min = 100;
 	int max = -1;
 	int x;
 	int origrformat=0;
 	struct ast_dsp *dsp;
-	if (!data || ast_strlen_zero((char *)data)) {
+	
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "BackgroundDetect requires an argument (filename)\n");
 		return -1;
 	}
-	strncpy(tmp, (char *)data, sizeof(tmp)-1);
+
+	LOCAL_USER_ADD(u);
+
+	tmp = ast_strdupa(data);
+	if (!tmp) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}	
+
 	stringp=tmp;
 	strsep(&stringp, "|");
 	options = strsep(&stringp, "|");
@@ -85,7 +112,6 @@ static int background_detect_exec(struct ast_channel *chan, void *data)
 	}
 	ast_log(LOG_DEBUG, "Preparing detect of '%s', sil=%d,min=%d,max=%d\n", 
 						tmp, sil, min, max);
-	LOCAL_USER_ADD(u);
 	if (chan->_state != AST_STATE_UP) {
 		/* Otherwise answer unless we're supposed to send this while on-hook */
 		res = ast_answer(chan);
@@ -124,7 +150,7 @@ static int background_detect_exec(struct ast_channel *chan, void *data)
 						char t[2];
 						t[0] = fr->subclass;
 						t[1] = '\0';
-						if (ast_canmatch_extension(chan, chan->context, t, 1, chan->callerid)) {
+						if (ast_canmatch_extension(chan, chan->context, t, 1, chan->cid.cid_num)) {
 							/* They entered a valid  extension, or might be anyhow */
 							res = fr->subclass;
 							ast_frfree(fr);
@@ -138,18 +164,19 @@ static int background_detect_exec(struct ast_channel *chan, void *data)
 							/* We've been quiet a little while */
 							if (notsilent) {
 								/* We had heard some talking */
-								gettimeofday(&end, NULL);
-								ms = (end.tv_sec - start.tv_sec) * 1000;
-								ms += (end.tv_usec - start.tv_usec) / 1000;
+								ms = ast_tvdiff_ms(ast_tvnow(), start);
 								ms -= sil;
 								if (ms < 0)
 									ms = 0;
 								if ((ms > min) && ((max < 0) || (ms < max))) {
+									char ms_str[10];
 									ast_log(LOG_DEBUG, "Found qualified token of %d ms\n", ms);
-									if (ast_exists_extension(chan, chan->context, "talk", 1, chan->callerid)) {
-										strncpy(chan->exten, "talk", sizeof(chan->exten) -1 );
-										chan->priority = 0;
-									}
+
+									/* Save detected talk time (in milliseconds) */ 
+									sprintf(ms_str, "%d", ms );	
+									pbx_builtin_setvar_helper(chan, "TALK_DETECTED", ms_str);
+									
+									ast_goto_if_exists(chan, chan->context, "talk", 1);
 									res = 0;
 									ast_frfree(fr);
 									break;
@@ -160,7 +187,7 @@ static int background_detect_exec(struct ast_channel *chan, void *data)
 						} else {
 							if (!notsilent) {
 								/* Heard some audio, mark the begining of the token */
-								gettimeofday(&start, NULL);
+								start = ast_tvnow();
 								ast_log(LOG_DEBUG, "Start of voice token!\n");
 								notsilent = 1;
 							}
@@ -191,8 +218,13 @@ static int background_detect_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_test.c
+++ b/apps/app_test.c
@@ -1,32 +1,53 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Applications to test connection and produce report in text file
- * 
- * Copyright (C) 2004, Digium, Inc.
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 * Russell Bryant <russelb@clemson.edu>
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Applications to test connection and produce report in text file
+ * 
+ * \ingroup applications
 */

-#include <sys/types.h>
-#include <asterisk/channel.h>
-#include <asterisk/options.h>
-#include <asterisk/module.h>
-#include <asterisk/logger.h>
-#include <asterisk/lock.h>
-#include <asterisk/app.h>
-#include <asterisk/pbx.h>
-#include <asterisk/utils.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 #include <unistd.h>
 #include <fcntl.h>
-#include "../astconf.h"
+#include <sys/types.h>
+#include <sys/stat.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/channel.h"
+#include "asterisk/options.h"
+#include "asterisk/module.h"
+#include "asterisk/logger.h"
+#include "asterisk/lock.h"
+#include "asterisk/app.h"
+#include "asterisk/pbx.h"
+#include "asterisk/utils.h"
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;

 static char *tdesc = "Interface Test Application";

@@ -51,7 +72,7 @@ static int measurenoise(struct ast_channel *chan, int ms, char *who)
 	int samples=0;
 	int x;
 	short *foo;
-	struct timeval start, tv;
+	struct timeval start;
 	struct ast_frame *f;
 	int rformat;
 	rformat = chan->readformat;
@@ -59,11 +80,9 @@ static int measurenoise(struct ast_channel *chan, int ms, char *who)
 		ast_log(LOG_NOTICE, "Unable to set to linear mode!\n");
 		return -1;
 	}
-	gettimeofday(&start, NULL);
+	start = ast_tvnow();
 	for(;;) {
-		gettimeofday(&tv, NULL);
-		mssofar = (tv.tv_sec - start.tv_sec) * 1000;
-		mssofar += (tv.tv_usec - start.tv_usec) / 1000;
+		mssofar = ast_tvdiff_ms(ast_tvnow(), start);
 		if (mssofar > ms)
 			break;
 		res = ast_waitfor(chan, ms - mssofar);
@@ -110,10 +129,6 @@ static int sendnoise(struct ast_channel *chan, int ms)
 	return res;	
 }

-STANDARD_LOCAL_USER;
-
-LOCAL_USER_DECL;
-
 static int testclient_exec(struct ast_channel *chan, void *data)
 {
 	struct localuser *u;
@@ -122,14 +137,15 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 	char fn[80];
 	char serverver[80];
 	FILE *f;
-	LOCAL_USER_ADD(u);
 	
 	/* Check for test id */
-	if (!testid || ast_strlen_zero(testid)) {
+	if (ast_strlen_zero(testid)) {
 		ast_log(LOG_WARNING, "TestClient requires an argument - the test id\n");
 		return -1;
 	}
 	
+	LOCAL_USER_ADD(u);
+
 	if (chan->_state != AST_STATE_UP)
 		res = ast_answer(chan);
 	
@@ -138,15 +154,18 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 	/* Transmit client version */
 	if (!res)
 		res = ast_dtmf_stream(chan, NULL, "8378*1#", 0);
-	ast_log(LOG_DEBUG, "Transmit client version\n");
+	if (option_debug)
+		ast_log(LOG_DEBUG, "Transmit client version\n");
 	
 	/* Read server version */
-	ast_log(LOG_DEBUG, "Read server version\n");
+	if (option_debug)
+		ast_log(LOG_DEBUG, "Read server version\n");
 	if (!res) 
 		res = ast_app_getdata(chan, NULL, serverver, sizeof(serverver) - 1, 0);
 	if (res > 0)
 		res = 0;
-	ast_log(LOG_DEBUG, "server version: %s\n", serverver);
+	if (option_debug)
+		ast_log(LOG_DEBUG, "server version: %s\n", serverver);
 		
 	if (res > 0)
 		res = 0;
@@ -158,7 +177,8 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 		res = ast_dtmf_stream(chan, NULL, testid, 0);		
 	if (!res) 
 		res = ast_dtmf_stream(chan, NULL, "#", 0);		
-	ast_log(LOG_DEBUG, "send test identifier: %s\n", testid);
+	if (option_debug)
+		ast_log(LOG_DEBUG, "send test identifier: %s\n", testid);

 	if ((res >=0) && (!ast_strlen_zero(testid))) {
 		/* Make the directory to hold the test results in case it's not there */
@@ -174,7 +194,8 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 			
 			if (!res) {
 				/* Step 1: Wait for "1" */
-				ast_log(LOG_DEBUG, "TestClient: 2.  Wait DTMF 1\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 2.  Wait DTMF 1\n");
 				res = ast_waitfordigit(chan, 3000);
 				fprintf(f, "WAIT DTMF 1:   %s\n", (res != '1') ? "FAIL" : "PASS");
 				if (res == '1')
@@ -186,7 +207,8 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 				res = ast_safe_sleep(chan, 1000);
 			if (!res) {
 				/* Step 2: Send "2" */
-				ast_log(LOG_DEBUG, "TestClient: 2.  Send DTMF 2\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 2.  Send DTMF 2\n");
 				res = ast_dtmf_stream(chan, NULL, "2", 0);
 				fprintf(f, "SEND DTMF 2:   %s\n", (res < 0) ? "FAIL" : "PASS");
 				if (res > 0)
@@ -194,7 +216,8 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 3: Wait one second */
-				ast_log(LOG_DEBUG, "TestClient: 3.  Wait one second\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 3.  Wait one second\n");
 				res = ast_safe_sleep(chan, 1000);
 				fprintf(f, "WAIT 1 SEC:    %s\n", (res < 0) ? "FAIL" : "PASS");
 				if (res > 0)
@@ -202,7 +225,8 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 			}			
 			if (!res) {
 				/* Step 4: Measure noise */
-				ast_log(LOG_DEBUG, "TestClient: 4.  Measure noise\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 4.  Measure noise\n");
 				res = measurenoise(chan, 5000, "TestClient");
 				fprintf(f, "MEASURENOISE:  %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
 				if (res > 0)
@@ -210,7 +234,8 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 5: Wait for "4" */
-				ast_log(LOG_DEBUG, "TestClient: 5.  Wait DTMF 4\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 5.  Wait DTMF 4\n");
 				res = ast_waitfordigit(chan, 3000);
 				fprintf(f, "WAIT DTMF 4:   %s\n", (res != '4') ? "FAIL" : "PASS");
 				if (res == '4')
@@ -220,13 +245,15 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 6: Transmit tone noise */
-				ast_log(LOG_DEBUG, "TestClient: 6.  Transmit tone\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 6.  Transmit tone\n");
 				res = sendnoise(chan, 6000);
 				fprintf(f, "SENDTONE:      %s\n", (res < 0) ? "FAIL" : "PASS");
 			}
 			if (!res || (res == '5')) {
 				/* Step 7: Wait for "5" */
-				ast_log(LOG_DEBUG, "TestClient: 7.  Wait DTMF 5\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 7.  Wait DTMF 5\n");
 				if (!res)
 					res = ast_waitfordigit(chan, 3000);
 				fprintf(f, "WAIT DTMF 5:   %s\n", (res != '5') ? "FAIL" : "PASS");
@@ -237,7 +264,8 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 8: Wait one second */
-				ast_log(LOG_DEBUG, "TestClient: 8.  Wait one second\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 8.  Wait one second\n");
 				res = ast_safe_sleep(chan, 1000);
 				fprintf(f, "WAIT 1 SEC:    %s\n", (res < 0) ? "FAIL" : "PASS");
 				if (res > 0)
@@ -245,7 +273,8 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 9: Measure noise */
-				ast_log(LOG_DEBUG, "TestClient: 6.  Measure tone\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 6.  Measure tone\n");
 				res = measurenoise(chan, 4000, "TestClient");
 				fprintf(f, "MEASURETONE:   %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
 				if (res > 0)
@@ -253,7 +282,8 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 10: Send "7" */
-				ast_log(LOG_DEBUG, "TestClient: 7.  Send DTMF 7\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 7.  Send DTMF 7\n");
 				res = ast_dtmf_stream(chan, NULL, "7", 0);
 				fprintf(f, "SEND DTMF 7:   %s\n", (res < 0) ? "FAIL" : "PASS");
 				if (res > 0)
@@ -261,7 +291,8 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 11: Wait for "8" */
-				ast_log(LOG_DEBUG, "TestClient: 11.  Wait DTMF 8\n");
+				if (option_debug)
+					ast_log(LOG_DEBUG, "TestClient: 11.  Wait DTMF 8\n");
 				res = ast_waitfordigit(chan, 3000);
 				fprintf(f, "WAIT DTMF 8:   %s\n", (res != '8') ? "FAIL" : "PASS");
 				if (res == '8')
@@ -269,12 +300,13 @@ static int testclient_exec(struct ast_channel *chan, void *data)
 				else
 					res = -1;
 			}
-			if (!res) {
+			if (option_debug && !res ) {
 				/* Step 12: Hangup! */
 				ast_log(LOG_DEBUG, "TestClient: 12.  Hangup\n");
 			}

-			ast_log(LOG_DEBUG, "-- TEST COMPLETE--\n");
+			if (option_debug)
+				ast_log(LOG_DEBUG, "-- TEST COMPLETE--\n");
 			fprintf(f, "-- END TEST--\n");
 			fclose(f);
 			res = -1;
@@ -299,13 +331,16 @@ static int testserver_exec(struct ast_channel *chan, void *data)
 	if (chan->_state != AST_STATE_UP)
 		res = ast_answer(chan);
 	/* Read version */
-	ast_log(LOG_DEBUG, "Read client version\n");
+	if (option_debug)
+		ast_log(LOG_DEBUG, "Read client version\n");
 	if (!res) 
 		res = ast_app_getdata(chan, NULL, testid, sizeof(testid) - 1, 0);
 	if (res > 0)
 		res = 0;
-	ast_log(LOG_DEBUG, "client version: %s\n", testid);
-	ast_log(LOG_DEBUG, "Transmit server version\n");
+	if (option_debug) {
+		ast_log(LOG_DEBUG, "client version: %s\n", testid);
+		ast_log(LOG_DEBUG, "Transmit server version\n");
+	}
 	res = ast_safe_sleep(chan, 1000);
 	if (!res)
 		res = ast_dtmf_stream(chan, NULL, "8378*1#", 0);
@@ -314,7 +349,8 @@ static int testserver_exec(struct ast_channel *chan, void *data)

 	if (!res) 
 		res = ast_app_getdata(chan, NULL, testid, sizeof(testid) - 1, 0);		
-	ast_log(LOG_DEBUG, "read test identifier: %s\n", testid);
+	if (option_debug) 
+		ast_log(LOG_DEBUG, "read test identifier: %s\n", testid);
 	/* Check for sneakyness */
 	if (strchr(testid, '/'))
 		res = -1;
@@ -333,7 +369,8 @@ static int testserver_exec(struct ast_channel *chan, void *data)
 			res = ast_safe_sleep(chan, 1000);
 			if (!res) {
 				/* Step 1: Send "1" */
-				ast_log(LOG_DEBUG, "TestServer: 1.  Send DTMF 1\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 1.  Send DTMF 1\n");
 				res = ast_dtmf_stream(chan, NULL, "1", 0);
 				fprintf(f, "SEND DTMF 1:   %s\n", (res < 0) ? "FAIL" : "PASS");
 				if (res > 0)
@@ -341,7 +378,8 @@ static int testserver_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 2: Wait for "2" */
-				ast_log(LOG_DEBUG, "TestServer: 2.  Wait DTMF 2\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 2.  Wait DTMF 2\n");
 				res = ast_waitfordigit(chan, 3000);
 				fprintf(f, "WAIT DTMF 2:   %s\n", (res != '2') ? "FAIL" : "PASS");
 				if (res == '2')
@@ -351,7 +389,8 @@ static int testserver_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 3: Measure noise */
-				ast_log(LOG_DEBUG, "TestServer: 3.  Measure noise\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 3.  Measure noise\n");
 				res = measurenoise(chan, 6000, "TestServer");
 				fprintf(f, "MEASURENOISE:  %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
 				if (res > 0)
@@ -359,7 +398,8 @@ static int testserver_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 4: Send "4" */
-				ast_log(LOG_DEBUG, "TestServer: 4.  Send DTMF 4\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 4.  Send DTMF 4\n");
 				res = ast_dtmf_stream(chan, NULL, "4", 0);
 				fprintf(f, "SEND DTMF 4:   %s\n", (res < 0) ? "FAIL" : "PASS");
 				if (res > 0)
@@ -368,7 +408,8 @@ static int testserver_exec(struct ast_channel *chan, void *data)
 		
 			if (!res) {
 				/* Step 5: Wait one second */
-				ast_log(LOG_DEBUG, "TestServer: 5.  Wait one second\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 5.  Wait one second\n");
 				res = ast_safe_sleep(chan, 1000);
 				fprintf(f, "WAIT 1 SEC:    %s\n", (res < 0) ? "FAIL" : "PASS");
 				if (res > 0)
@@ -377,7 +418,8 @@ static int testserver_exec(struct ast_channel *chan, void *data)
 		
 			if (!res) {
 				/* Step 6: Measure noise */
-				ast_log(LOG_DEBUG, "TestServer: 6.  Measure tone\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 6.  Measure tone\n");
 				res = measurenoise(chan, 4000, "TestServer");
 				fprintf(f, "MEASURETONE:   %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
 				if (res > 0)
@@ -386,7 +428,8 @@ static int testserver_exec(struct ast_channel *chan, void *data)

 			if (!res) {
 				/* Step 7: Send "5" */
-				ast_log(LOG_DEBUG, "TestServer: 7.  Send DTMF 5\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 7.  Send DTMF 5\n");
 				res = ast_dtmf_stream(chan, NULL, "5", 0);
 				fprintf(f, "SEND DTMF 5:   %s\n", (res < 0) ? "FAIL" : "PASS");
 				if (res > 0)
@@ -395,14 +438,16 @@ static int testserver_exec(struct ast_channel *chan, void *data)

 			if (!res) {
 				/* Step 8: Transmit tone noise */
-				ast_log(LOG_DEBUG, "TestServer: 8.  Transmit tone\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 8.  Transmit tone\n");
 				res = sendnoise(chan, 6000);
 				fprintf(f, "SENDTONE:      %s\n", (res < 0) ? "FAIL" : "PASS");
 			}
 		
 			if (!res || (res == '7')) {
 				/* Step 9: Wait for "7" */
-				ast_log(LOG_DEBUG, "TestServer: 9.  Wait DTMF 7\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 9.  Wait DTMF 7\n");
 				if (!res)
 					res = ast_waitfordigit(chan, 3000);
 				fprintf(f, "WAIT DTMF 7:   %s\n", (res != '7') ? "FAIL" : "PASS");
@@ -415,7 +460,8 @@ static int testserver_exec(struct ast_channel *chan, void *data)
 				res = ast_safe_sleep(chan, 1000);
 			if (!res) {
 				/* Step 10: Send "8" */
-				ast_log(LOG_DEBUG, "TestServer: 10.  Send DTMF 8\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 10.  Send DTMF 8\n");
 				res = ast_dtmf_stream(chan, NULL, "8", 0);
 				fprintf(f, "SEND DTMF 8:   %s\n", (res < 0) ? "FAIL" : "PASS");
 				if (res > 0)
@@ -423,12 +469,13 @@ static int testserver_exec(struct ast_channel *chan, void *data)
 			}
 			if (!res) {
 				/* Step 11: Wait for hangup to arrive! */
-				ast_log(LOG_DEBUG, "TestServer: 11.  Waiting for hangup\n");
+				if (option_debug) 
+					ast_log(LOG_DEBUG, "TestServer: 11.  Waiting for hangup\n");
 				res = ast_safe_sleep(chan, 10000);
 				fprintf(f, "WAIT HANGUP:   %s\n", (res < 0) ? "PASS" : "FAIL");
 			}

-			ast_log(LOG_DEBUG, "-- TEST COMPLETE--\n");
+			ast_log(LOG_NOTICE, "-- TEST COMPLETE--\n");
 			fprintf(f, "-- END TEST--\n");
 			fclose(f);
 			res = -1;
@@ -444,15 +491,24 @@ static int testserver_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(testc_app);
+	res |= ast_unregister_application(tests_app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	ast_unregister_application(testc_app);
-	return ast_unregister_application(tests_app);
+
+	return res;	
 }

 int load_module(void)
 {
-	ast_register_application(testc_app, testclient_exec, testc_synopsis, testc_descrip);
-	return ast_register_application(tests_app, testserver_exec, tests_synopsis, tests_descrip);
+	int res;
+
+	res = ast_register_application(testc_app, testclient_exec, testc_synopsis, testc_descrip);
+	res |= ast_register_application(tests_app, testserver_exec, tests_synopsis, tests_descrip);
+
+	return res;
 }

 char *description(void)
--- a/apps/app_transfer.c
+++ b/apps/app_transfer.c
@@ -1,71 +1,160 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Transfer a caller
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Transfer a caller
+ * 
+ * Requires transfer support from channel driver
+ *
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
+#include <unistd.h>

-static char *tdesc = "Transfer";
+#include "asterisk.h"

-static char *app = "Transfer";
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

-static char *synopsis = "Transfer caller to remote extension";
-
-static char *descrip = 
-"  Transfer(exten):  Requests the remote caller be transferred to\n"
-"a given extension. Returns -1 on hangup, or 0 on completion\n"
-"regardless of whether the transfer was successful.  If the transfer\n"
-"was *not* supported or successful and there exists a priority n + 101,\n"
-"then that priority will be taken next.\n" ;
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/options.h"
+#include "asterisk/app.h"

 STANDARD_LOCAL_USER;

 LOCAL_USER_DECL;

+static const char *tdesc = "Transfer";
+
+static const char *app = "Transfer";
+
+static const char *synopsis = "Transfer caller to remote extension";
+
+static const char *descrip = 
+"  Transfer([Tech/]dest[|options]):  Requests the remote caller be transferred\n"
+"to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only\n"
+"an incoming call with the same channel technology will be transfered.\n"
+"Note that for SIP, if you transfer before call is setup, a 302 redirect\n"
+"SIP message will be returned to the caller.\n"
+"\nThe result of the application will be reported in the TRANSFERSTATUS\n"
+"channel variable:\n"
+"       SUCCESS      Transfer succeeded\n"
+"       FAILURE      Transfer failed\n"
+"       UNSUPPORTED  Transfer unsupported by channel driver\n"
+"The option string many contain the following character:\n"
+"'j' -- jump to n+101 priority if the channel transfer attempt\n"
+"       fails\n";
+
 static int transfer_exec(struct ast_channel *chan, void *data)
 {
-	int res=0;
+	int res;
+	int len;
 	struct localuser *u;
-	if (!data || !strlen(data)) {
-		ast_log(LOG_WARNING, "Transfer requires an argument (destination)\n");
-		res = 1;
-	}
+	char *slash;
+	char *tech = NULL;
+	char *dest = NULL;
+	char *status;
+	char *parse;
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(dest);
+		AST_APP_ARG(options);
+	);
+
 	LOCAL_USER_ADD(u);
-	if (!res) {
-		res = ast_transfer(chan, data);
+
+	if (ast_strlen_zero((char *)data)) {
+		ast_log(LOG_WARNING, "Transfer requires an argument ([Tech/]destination[|options])\n");
+		LOCAL_USER_REMOVE(u);
+		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
+		return 0;
+	} else {
+		parse = ast_strdupa(data);
+		if (!parse) {
+			ast_log(LOG_ERROR, "Out of memory!\n");
+			LOCAL_USER_REMOVE(u);
+			return -1;
+		}
 	}
-	if (!res) {
-		/* Look for a "busy" place */
-		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-			chan->priority += 100;
+
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
 	}
-	if (res > 0)
+
+	dest = args.dest;
+
+	if ((slash = strchr(dest, '/')) && (len = (slash - dest))) {
+		tech = dest;
+		dest = slash + 1;
+		/* Allow execution only if the Tech/destination agrees with the type of the channel */
+		if (strncasecmp(chan->type, tech, len)) {
+			pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
+			LOCAL_USER_REMOVE(u);
+			return 0;
+		}
+	}
+
+	/* Check if the channel supports transfer before we try it */
+	if (!chan->tech->transfer) {
+		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "UNSUPPORTED");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+
+	res = ast_transfer(chan, dest);
+
+	if (res < 0) {
+		status = "FAILURE";
+		if (priority_jump || option_priority_jumping)
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
 		res = 0;
+	} else {
+		status = "SUCCESS";
+		res = 0;
+	}
+
+	pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", status);
+
 	LOCAL_USER_REMOVE(u);
+
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
@@ -75,13 +164,15 @@ int load_module(void)

 char *description(void)
 {
-	return tdesc;
+	return (char *) tdesc;
 }

 int usecount(void)
 {
 	int res;
+
 	STANDARD_USECOUNT(res);
+
 	return res;
 }

--- a/apps/app_txtcidname.c
+++ b/apps/app_txtcidname.c
@@ -1,32 +1,51 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Caller*id name lookup - Look up the caller's name via DNS
- * 
- * Copyright (C) 1999-2004, Digium
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Caller*id name lookup - Look up the caller's name via DNS
+ * 
+ * \ingroup applications
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/options.h>
-#include <asterisk/config.h>
-#include <asterisk/module.h>
-#include <asterisk/enum.h>
-#include <asterisk/utils.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <stdlib.h>
 #include <ctype.h>

+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/options.h"
+#include "asterisk/module.h"
+#include "asterisk/enum.h"
+#include "asterisk/utils.h"
+#include "asterisk/app.h"
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
 static char *tdesc = "TXTCIDName";

 static char *app = "TXTCIDName";
@@ -34,18 +53,14 @@ static char *app = "TXTCIDName";
 static char *synopsis = "Lookup caller name from TXT record";

 static char *descrip = 
-"  TXTLookup(CallerID):  Looks up a Caller Name via DNS and sets\n"
+"  TXTCIDName(<CallerIDNumber>[|options]):  Looks up a Caller Name via DNS and sets\n"
 "the variable 'TXTCIDNAME'. TXTCIDName will either be blank\n"
-"or return the value found in the TXT record in DNS.\n" ;
-
-#define ENUM_CONFIG "enum.conf"
-
-static char h323driver[80] = "";
-#define H323DRIVERDEFAULT "H323"
-
-STANDARD_LOCAL_USER;
-
-LOCAL_USER_DECL;
+"or return the value found in the TXT record in DNS.\n" 
+"The option string may contain the following character:\n"
+"'j' -- jump to n+101 priority if the lookup fails\n"
+"This application sets the following channel variable upon completion:\n"
+"  TXTCIDNAMESTATUS The status of the lookup as a text string, one of\n"
+"      SUCCESS | FAILED\n";

 static int txtcidname_exec(struct ast_channel *chan, void *data)
 {
@@ -53,79 +68,84 @@ static int txtcidname_exec(struct ast_channel *chan, void *data)
 	char tech[80];
 	char txt[256] = "";
 	char dest[80];
-
 	struct localuser *u;
-	if (!data || !strlen(data)) {
-		ast_log(LOG_WARNING, "TXTCIDName requires an argument (extension)\n");
-		res = 1;
-	}
+	static int dep_warning = 0;
+	char *parse = NULL;
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(cidnum);
+		AST_APP_ARG(options);
+	);
+
 	LOCAL_USER_ADD(u);
+	
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The TXTCIDName application has been deprecated in favor of the TXTCIDNAME dialplan function.\n");
+		dep_warning = 1;
+	}
+	
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "TXTCIDName requires an argument (extension[|options])\n");
+		LOCAL_USER_REMOVE(u);
+		return(0);
+	}
+
+	parse = ast_strdupa(data);
+	if (!parse) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args,parse);
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+	
 	if (!res) {
 		res = ast_get_txt(chan, data, dest, sizeof(dest), tech, sizeof(tech), txt, sizeof(txt));
 	}
-	LOCAL_USER_REMOVE(u);
+	
 	/* Parse it out */
 	if (res > 0) {
 		if (!ast_strlen_zero(txt)) {
 			pbx_builtin_setvar_helper(chan, "TXTCIDNAME", txt);
-#if 0
-			ast_log(LOG_DEBUG, "TXTCIDNAME got '%s'\n", txt);
-#endif
+			pbx_builtin_setvar_helper(chan, "TXTCIDNAMESTATUS", "SUCCESS");
+			if (option_debug > 1)
+				ast_log(LOG_DEBUG, "TXTCIDNAME got '%s'\n", txt);
 		}
 	}
 	if (!res) {
 		/* Look for a "busy" place */
-		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-			chan->priority += 100;
+		if (priority_jump || option_priority_jumping)
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		pbx_builtin_setvar_helper(chan, "TXTCIDNAMESTATUS", "FAILED");
 	} else if (res > 0)
 		res = 0;
+
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }

-static int load_config(void)
-{
-	struct ast_config *cfg;
-	char *s;
-
-	cfg = ast_load(ENUM_CONFIG);
-	if (cfg) {
-		if (!(s=ast_variable_retrieve(cfg, "general", "h323driver"))) {
-			strncpy(h323driver, H323DRIVERDEFAULT, sizeof(h323driver) - 1);
-		} else {
-			strncpy(h323driver, s, sizeof(h323driver) - 1);
-		}
-		ast_destroy(cfg);
-		return 0;
-	}
-	ast_log(LOG_NOTICE, "No ENUM Config file, using defaults\n");
-	return 0;
-}
-
-
 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
 {
-	int res;
-	res = ast_register_application(app, txtcidname_exec, synopsis, descrip);
-	if (res)
-		return(res);
-	if ((res=load_config())) {
-		return(res);
-	}
-	return(0);
+	return ast_register_application(app, txtcidname_exec, synopsis, descrip);
 }

-int reload(void)
-{
-	return(load_config());
-}
-
-
 char *description(void)
 {
 	return tdesc;
@@ -142,4 +162,3 @@ char *key()
 {
 	return ASTERISK_GPL_KEY;
 }
-
--- a/apps/app_url.c
+++ b/apps/app_url.c
@@ -1,26 +1,44 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * App to transmit a URL
- * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999 - 2005, Digium, Inc.
 *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief App to transmit a URL
+ * 
+ * \ingroup applications
 */
 
-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/translate.h>
-#include <asterisk/image.h>
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/options.h"

 static char *tdesc = "Send URL Applications";

@@ -29,15 +47,25 @@ static char *app = "SendURL";
 static char *synopsis = "Send a URL";

 static char *descrip = 
-"  SendURL(URL[|option]): Requests client go to URL.  If the client\n"
-"does not support html transport, and  there  exists  a  step  with\n"
-"priority  n + 101,  then  execution  will  continue  at that step.\n"
-"Otherwise, execution will continue at  the  next  priority  level.\n"
-"SendURL only returns 0  if  the  URL  was  sent  correctly  or  if\n"
-"the channel  does  not  support HTML transport,  and -1 otherwise.\n"
-"If the option 'wait' is  specified,  execution  will  wait  for an\n"
-"acknowledgement that  the  URL  has  been loaded before continuing\n"
-"and will return -1 if the peer is unable to load the URL\n";
+"  SendURL(URL[|option]): Requests client go to URL (IAX2) or sends the \n"
+"URL to the client (other channels).\n"
+"Result is returned in the SENDURLSTATUS channel variable:\n"
+"    SUCCESS       URL successfully sent to client\n"
+"    FAILURE       Failed to send URL\n"
+"    NOLOAD        Clien failed to load URL (wait enabled)\n"
+"    UNSUPPORTED   Channel does not support URL transport\n"
+"\n"
+"If the option 'wait' is specified, execution will wait for an\n"
+"acknowledgement that the URL has been loaded before continuing\n"
+"and will return -1 if the peer is unable to load the URL\n"
+"\n"
+"Old behaviour (deprecated): \n"
+" If the client does not support Asterisk \"html\" transport, \n"
+" and there exists a step with priority n + 101, then execution will\n"
+" continue at that step.\n"
+" Otherwise, execution will continue at the next priority level.\n"
+" SendURL only returns 0 if the URL was sent correctly  or if\n"
+" the channel does not support HTML transport, and -1 otherwise.\n";

 STANDARD_LOCAL_USER;

@@ -47,73 +75,102 @@ static int sendurl_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
 	struct localuser *u;
-	char tmp[256];
+	char *tmp;
 	char *options;
-	int option_wait=0;
+	int local_option_wait=0;
+	int local_option_jump = 0;
 	struct ast_frame *f;
 	char *stringp=NULL;
-	if (!data || !strlen((char *)data)) {
+	char *status = "FAILURE";
+	
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "SendURL requires an argument (URL)\n");
+		pbx_builtin_setvar_helper(chan, "SENDURLSTATUS", status);
 		return -1;
 	}
-	strncpy(tmp, (char *)data, sizeof(tmp)-1);
+
+	LOCAL_USER_ADD(u);
+
+	tmp = ast_strdupa(data);
+	if (!tmp) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
 	stringp=tmp;
 	strsep(&stringp, "|");
 	options = strsep(&stringp, "|");
 	if (options && !strcasecmp(options, "wait"))
-		option_wait = 1;
-	LOCAL_USER_ADD(u);
+		local_option_wait = 1;
+	if (options && !strcasecmp(options, "j"))
+		local_option_jump = 1;
+	
 	if (!ast_channel_supports_html(chan)) {
 		/* Does not support transport */
-		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-			chan->priority += 100;
+		if (local_option_jump || option_priority_jumping)
+			 ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		pbx_builtin_setvar_helper(chan, "SENDURLSTATUS", "UNSUPPORTED");
 		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
 	res = ast_channel_sendurl(chan, tmp);
-	if (res > -1) {
-		if (option_wait) {
-			for(;;) {
-				/* Wait for an event */
-				res = ast_waitfor(chan, -1);
-				if (res < 0) 
-					break;
-				f = ast_read(chan);
-				if (!f) {
-					res = -1;
-					break;
-				}
-				if (f->frametype == AST_FRAME_HTML) {
-					switch(f->subclass) {
-					case AST_HTML_LDCOMPLETE:
-						res = 0;
-						ast_frfree(f);
-						goto out;
-						break;
-					case AST_HTML_NOSUPPORT:
-						/* Does not support transport */
-						if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
-							chan->priority += 100;
-						res = 0;
-						goto out;
-						break;
-					default:
-						ast_log(LOG_WARNING, "Don't know what to do with HTML subclass %d\n", f->subclass);
-					};
-				}
-				ast_frfree(f);
-			}
-		}
+	if (res == -1) {
+		pbx_builtin_setvar_helper(chan, "SENDURLSTATUS", "FAILURE");
+		LOCAL_USER_REMOVE(u);
+		return res;
 	}
+	status = "SUCCESS";
+	if (local_option_wait) {
+		for(;;) {
+			/* Wait for an event */
+			res = ast_waitfor(chan, -1);
+			if (res < 0) 
+				break;
+			f = ast_read(chan);
+			if (!f) {
+				res = -1;
+				status = "FAILURE";
+				break;
+			}
+			if (f->frametype == AST_FRAME_HTML) {
+				switch(f->subclass) {
+				case AST_HTML_LDCOMPLETE:
+					res = 0;
+					ast_frfree(f);
+					status = "NOLOAD";
+					goto out;
+					break;
+				case AST_HTML_NOSUPPORT:
+					/* Does not support transport */
+					status ="UNSUPPORTED";
+					if (local_option_jump || option_priority_jumping)
+			 			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+					res = 0;
+					goto out;
+					break;
+				default:
+					ast_log(LOG_WARNING, "Don't know what to do with HTML subclass %d\n", f->subclass);
+				};
+			}
+			ast_frfree(f);
+		}
+	} 
 out:	
+	pbx_builtin_setvar_helper(chan, "SENDURLSTATUS", status);
 	LOCAL_USER_REMOVE(u);
 	return res;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+	
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_userevent.c
+++ b/apps/app_userevent.c
@@ -1,24 +1,43 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * UserEvent application -- send manager event
- * 
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
 *
 * This program is free software, distributed under the terms of
- * the GNU General Public License
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
 */

-#include <asterisk/lock.h>
-#include <asterisk/file.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
-#include <asterisk/manager.h>
+/*! \file
+ *
+ * \brief UserEvent application -- send manager event
+ * 
+ * \ingroup applications
+ */
+
+
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
+#include <unistd.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/manager.h"

 static char *tdesc = "Custom User Event Application";

@@ -44,32 +63,39 @@ LOCAL_USER_DECL;
 static int userevent_exec(struct ast_channel *chan, void *data)
 {
 	struct localuser *u;
-	char info[512];
-    char eventname[512];
+	char *info;
+	char eventname[512];
 	char *eventbody;

-	if (!data || !strlen(data)) {
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "UserEvent requires an argument (eventname|optional event body)\n");
 		return -1;
 	}

-	strncpy(info, (char *)data, strlen((char *)data) + AST_MAX_EXTENSION-1);
+	LOCAL_USER_ADD(u);
+
+	info = ast_strdupa(data);
+	if (!info) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
 	snprintf(eventname, sizeof(eventname), "UserEvent%s", info);
 	eventbody = strchr(eventname, '|');
 	if (eventbody) {
 		*eventbody = '\0';
 		eventbody++;
 	}
-	LOCAL_USER_ADD(u);
-
+	
 	if(eventbody) {
            ast_log(LOG_DEBUG, "Sending user event: %s, %s\n", eventname, eventbody);
-            manager_event(EVENT_FLAG_CALL, eventname, 
+            manager_event(EVENT_FLAG_USER, eventname, 
 			"Channel: %s\r\nUniqueid: %s\r\n%s\r\n",
 			chan->name, chan->uniqueid, eventbody);
 	} else {
            ast_log(LOG_DEBUG, "Sending user event: %s\n", eventname);
-            manager_event(EVENT_FLAG_CALL, eventname, 
+            manager_event(EVENT_FLAG_USER, eventname, 
 			"Channel: %s\r\nUniqueid: %s\r\n", chan->name, chan->uniqueid);
 	}

@@ -79,8 +105,13 @@ static int userevent_exec(struct ast_channel *chan, void *data)

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app);
+
+	return res;	
 }

 int load_module(void)
--- a/apps/app_verbose.c
+++ b/apps/app_verbose.c
@@ -1,25 +1,41 @@
 /*
- * Asterisk -- A telephony toolkit for Linux.
+ * Asterisk -- An open source telephony toolkit.
 *
- * Verbose application
- * 
- * Copyright (c) 2004 Tilghman Lesher.  All rights reserved.
+ * Copyright (c) 2004 - 2005 Tilghman Lesher.  All rights reserved.
 *
 * Tilghman Lesher <app_verbose_v001@the-tilghman.com>
 *
 * This code is released by the author with no restrictions on usage.
 *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ *
+ * \brief Verbose logging application
+ *
+ * \ingroup applications
 */

 #include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
-#include <asterisk/options.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <asterisk/pbx.h>
-#include <asterisk/module.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/options.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"


 static char *tdesc = "Send verbose output";
@@ -30,8 +46,7 @@ static char *verbose_synopsis = "Send arbitrary text to verbose output";

 static char *verbose_descrip =
 "Verbose([<level>|]<message>)\n"
-"  level must be an integer value.  If not specified, defaults to 0."
-"  Always returns 0.\n";
+"  level must be an integer value.  If not specified, defaults to 0.\n";

 STANDARD_LOCAL_USER;

@@ -41,6 +56,9 @@ static int verbose_exec(struct ast_channel *chan, void *data)
 {
 	char *vtext;
 	int vsize;
+	struct localuser *u;
+
+	LOCAL_USER_ADD(u);

 	if (data) {
 		vtext = ast_strdupa((char *)data);
@@ -78,13 +96,20 @@ static int verbose_exec(struct ast_channel *chan, void *data)
 		}
 	}

+	LOCAL_USER_REMOVE(u);
+
 	return 0;
 }

 int unload_module(void)
 {
+	int res;
+
+	res = ast_unregister_application(app_verbose);
+
 	STANDARD_HANGUP_LOCALUSERS;
-	return ast_unregister_application(app_verbose);
+
+	return res;	
 }

 int load_module(void)
--- a/Show More
+++ b/Show More