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/1.2.9-netsec@212958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

.cvsignore

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

.lastclean

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

.version

@@ -0,0 +1 @@
+1.2.9-netsec

BUGS

@@ -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

CHANGES

@@ -1,655 +1,127 @@
- NOTE: Corrections or additions to the ChangeLog may be submitted to
-       http://bugs.digium.com.  Documentation and formatting fixes are not
-       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:
 
-Asterisk 1.2.0
+   * Cygwin build system portability
+   * Optional generation of outbound silence during channel recording
 
- -- Some of the major feature upgrades ...
+Changes since Asterisk 1.2.0-beta1:
 
- -- DUNDi (Distributed Universal Number Discovery -- http://www.dundi.com)
- -- AEL (Asterisk Extension Logic)
- -- Realtime Database Configuration Engine
- -- Native Music on Hold
- -- Native IAX Encryption
- -- New Jitter Buffer
- -- Q.SIG Switchtype for PRI
- -- FastAGI (AGI over TCP)
- -- Dialplan Functions
- -- ODBC Storage of Voicemail
+   * 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 1.0.10
+Changes since Asterisk 1.0:
 
- -- chan_local
-    -- In releases 1.0.8 and 1.0.9, the Local channels that are created would
-       not be masqueraded into the new channel type.  This has now been fixed.
- -- chan_sip
-    -- The 'insecure' options have been changed to support matching peersby IP
-       only, not requiring authentication on incoming invites, or both. Before,
-       to not require authentication on incoming invites also required matching
-       peers based on IP only.
- -- chan_zap
-    -- Before, call waiting could occur during the initial ringing on the line.
-       This has now been fixed.
- -- app_disa
-    -- We will now not set the accountcode if one is not supplied. 
- -- app_meetme
-    -- If the first caller into a conference hangs up while being prompted for
-       the conference pin number, the conference will no longer be held open.
- -- app_userevent
-    -- Events created with this application were indicated as a "call" event
-       instead of a "user" event.  This made the "user" event permissions
-       not work correctly.
- -- app_voicemail
-    -- When using the externpass option for voicemail, the password will be
-       immediately updated in memory as well, instead of having to wait for
-       the next time the configuration is reloaded. 
- -- app_zapras
-    -- We now ensure buffer policy is restored after RAS is done with a channel.
-       This could cause audio problems on the channel after zapras is done
-       with it. 
- -- res_agi
-    -- We now unmask the SIGHUP signal before executing an AGI script.  This
-       fixes problems where some AGI scripts would continue running long after
-       the call is over.
- -- extensions
-    -- A potential crash has been fixed when calling LEN() to get the length of
-       a string that was 80 characters or larger.
- -- logger
-    -- The Asterisk logger will automatically detect when a log file needs to
-       be rotated.  However, this feature could put Asterisk in a nasty loop
-       that would result in a crash.
- -- general
-    -- Added man pages for astgenkey, autosupport, and safe_asterisk
+This list currently only containts changes made from the end of November until
+March 26, 2005.
 
-Asterisk 1.0.9
-
- -- fix bug in callerid matching in the dialplan that was introduced in 1.0.8
-
-Asterisk 1.0.8
-
- -- 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
-
-Asterisk 1.0.7
-
- -- 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.
-
-Asterisk 1.0.6
-
- -- 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.
-    -- 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...
-
-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

CREDITS

@@ -13,6 +13,8 @@ Telesthetic - for supporting SIP development
 
 Christos Ricudis - for substantial code contributions
 
+nic.at - ENUM support in Asterisk
+
 Paul Bagyenda, Digital Solutions - for initial Voicetronix driver development
 
 === WISHLIST CONTRIBUTERS ===
@@ -85,7 +87,7 @@ Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
 	available under a Creative Commons License at http://www.asteriskdocs.org
 
 === HOLD MUSIC ===
-Music provided by www.freeplaymusic.com
+Music provided by www.opsound.org
 
 === OTHER SOURCE CODE IN ASTERISK ===
 

Makefile

@@ -36,20 +36,24 @@ endif
 # Remember the MAKELEVEL at the top
 MAKETOPLEVEL?=$(MAKELEVEL)
 
+PWD=$(shell pwd)
+
+ifneq ($(findstring dont-optimize,$(MAKECMDGOALS)),dont-optimize)
 ######### More GSM codec optimization
 ######### Uncomment to enable MMXTM optimizations for x86 architecture CPU's
 ######### which support MMX instructions.  This should be newer pentiums,
 ######### ppro's, etc, as well as the AMD K6 and K7.  
 #K6OPT  = -DK6OPT
 
+#Tell gcc to optimize the code
+OPTIMIZE+=-O6
+endif
+
 #Overwite config files on "make samples"
 OVERWRITE=y
 
-#Tell gcc to optimize the code
-OPTIMIZE+=-O6
-
-#Include debug symbols in the executables (-g) and profiling info (-pg)
-DEBUG=-g #-pg
+#Include debug and macro symbols in the executables (-g) and profiling info (-pg)
+DEBUG=-g3 #-pg
 
 #Set NOCRYPTO to yes if you do not want to have crypto support or 
 #dependencies
@@ -87,7 +91,9 @@ INSTALL_PREFIX?=
 # Files are copied here temporarily during the install process
 # For example, make DESTDIR=/tmp/asterisk woud put things in
 # /tmp/asterisk/etc/asterisk
-DESTDIR=
+# XXX watch out, put no spaces or comments after the value
+DESTDIR?=
+#DESTDIR?=/tmp/asterisk
 
 # Original busydetect routine
 BUSYDETECT = #-DBUSYDETECT
@@ -101,7 +107,10 @@ BUSYDETECT+= #-DBUSYDETECT_TONEONLY
 # Don't use together with -DBUSYDETECT_TONEONLY
 BUSYDETECT+= #-DBUSYDETECT_COMPARE_TONE_AND_SILENCE
 
-ifneq (${OSARCH},SunOS)
+# Comment this if you want to disable MIDCOM
+MIDCOM = -DMIDCOM
+
+ifneq ($(OSARCH),SunOS)
   ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk
   ASTVARLIBDIR=$(INSTALL_PREFIX)/var/lib/asterisk
   ASTETCDIR=$(INSTALL_PREFIX)/etc/asterisk
@@ -164,7 +173,7 @@ ifneq ($(wildcard ~/.asterisk.makeopts),)
   include ~/.asterisk.makeopts
 endif
 
-ifeq (${OSARCH},Linux)
+ifeq ($(OSARCH),Linux)
   ifeq ($(CROSS_COMPILE),)
     PROC?=$(shell uname -m)
   else
@@ -175,7 +184,6 @@ ifeq (${OSARCH},Linux)
     # You must have GCC 3.4 to use k8, otherwise use athlon
     PROC=k8
     #PROC=athlon
-    OPTIONS+=-m64
   endif
 
   ifeq ($(PROC),sparc64)
@@ -196,28 +204,30 @@ ifeq (${OSARCH},Linux)
       OPTIONS+=-fsigned-char -mcpu=ep9312
     else
       ifeq ($(SUB_PROC),xscale)
-        OPTIONS+=-fsigned-char -msoft-float -mcpu=xscale
+        OPTIONS+=-fsigned-char -mcpu=xscale
       else
-        OPTIONS+=-fsigned-char -msoft-float 
+        OPTIONS+=-fsigned-char 
       endif
     endif
   endif
   MPG123TARG=linux
 endif
 
-PWD=$(shell pwd)
 GREP=grep
+ID=id
 
-ifeq (${OSARCH},SunOS)
+ifeq ($(OSARCH),SunOS)
   GREP=/usr/xpg4/bin/grep
   M4=/usr/local/bin/m4
+  ID=/usr/xpg4/bin/id
 endif
 
 INCLUDE+=-Iinclude -I../include
 ASTCFLAGS+=-pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE #-DMAKE_VALGRIND_HAPPY
 ASTCFLAGS+=$(OPTIMIZE)
+ASTOBJ=-o asterisk
 
-ifeq ($(findstring BSD,${OSARCH}),BSD)
+ifeq ($(findstring BSD,$(OSARCH)),BSD)
   PROC=$(shell uname -m)
   ASTCFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
 endif
@@ -238,62 +248,97 @@ else
   endif
 endif
 
-ifeq (${OSARCH},FreeBSD)
+ifeq ($(OSARCH),FreeBSD)
   BSDVERSION=$(shell make -V OSVERSION -f $(CROSS_COMPILE_TARGET)/usr/share/mk/bsd.port.subdir.mk)
-  ASTCFLAGS+=$(shell if test ${BSDVERSION} -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
-  LIBS+=$(shell if test  ${BSDVERSION} -lt 502102 ; then echo "-lc_r"; else echo "-pthread"; fi)
+  ASTCFLAGS+=$(shell if test $(BSDVERSION) -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
+  LIBS+=$(shell if test  $(BSDVERSION) -lt 502102 ; then echo "-lc_r"; else echo "-pthread"; fi)
   ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/spandsp),)
     ASTCFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include/spandsp
   endif
   MPG123TARG=freebsd
+
+  # XXX FreeBSD paths
+  PREFIX?=/usr/local
+  ASTLIBDIR=$(INSTALL_PREFIX)$(PREFIX)/lib/asterisk
+  ASTVARLIBDIR=$(INSTALL_PREFIX)$(PREFIX)/share/asterisk
+  ASTETCDIR=$(INSTALL_PREFIX)$(PREFIX)/etc/asterisk
+  ASTSPOOLDIR=$(INSTALL_PREFIX)/var/spool/asterisk
+  ASTLOGDIR=$(INSTALL_PREFIX)/var/log/asterisk
+  ASTHEADERDIR=$(INSTALL_PREFIX)$(PREFIX)/include/asterisk
+  ASTCONFPATH=$(ASTETCDIR)/asterisk.conf
+  ASTBINDIR=$(INSTALL_PREFIX)$(PREFIX)/bin
+  ASTSBINDIR=$(INSTALL_PREFIX)$(PREFIX)/sbin
+  ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run
+  ASTMANDIR=$(INSTALL_PREFIX)$(PREFIX)/man
+  # XXX end FreeBSD paths
+
 endif # FreeBSD
 
-ifeq (${OSARCH},NetBSD)
+ifeq ($(OSARCH),NetBSD)
   ASTCFLAGS+=-pthread
   INCLUDE+=-I$(CROSS_COMPILE_TARGET)/usr/pkg/include
   MPG123TARG=netbsd
 endif
 
-ifeq (${OSARCH},OpenBSD)
+ifeq ($(OSARCH),OpenBSD)
   ASTCFLAGS+=-pthread
 endif
 
-ifeq (${OSARCH},SunOS)
+ifeq ($(OSARCH),SunOS)
   ASTCFLAGS+=-Wcast-align -DSOLARIS
   INCLUDE+=-Iinclude/solaris-compat -I$(CROSS_COMPILE_TARGET)/usr/local/ssl/include
 endif
 
+ifeq ($(findstring CYGWIN,$(OSARCH)),CYGWIN)
+  CYGLOADER=cygwin_a
+  OSARCH=CYGWIN
+  ASTOBJ=-shared -o asterisk.dll -Wl,--out-implib=libasterisk.dll.a -Wl,--export-all-symbols
+  ASTLINK=
+  LIBS+=-lpthread -lncurses -lm -lresolv
+  ASTSBINDIR=$(MODULES_DIR)
+  PROC=$(shell uname -m)
+endif
+
+ifndef WITHOUT_ZAPTEL
+
 ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/zaptel.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/zaptel.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/pkg/include/zaptel.h),)
   ASTCFLAGS+=-DZAPTEL_OPTIMIZATIONS
 endif
 
+endif # WITHOUT_ZAPTEL
+
 LIBEDIT=editline/libedit.a
 
 ifneq ($(wildcard .version),)
-  ASTERISKVERSION=$(shell cat .version)
-  ASTERISKVERSIONNUM=$(shell awk -F. '{printf "%02d%02d%02d", $$1, $$2, $$3}' .version)
-  RPMVERSION=$(shell sed 's/[-\/:]/_/g' .version)
+  ASTERISKVERSION:=$(shell cat .version)
+  ASTERISKVERSIONNUM:=$(shell awk -F. '{printf "%02d%02d%02d", $$1, $$2, $$3}' .version)
+  RPMVERSION:=$(shell sed 's/[-\/:]/_/g' .version)
 else
   RPMVERSION=unknown
 endif
 
-ifneq ($(wildcard CVS),)
+# CVS mirrors of SVN have .svnrevision files showing
+# which SVN revision they are based on, and .svnbranch
+# showing the branch they are made from
+ifneq ($(wildcard .svnrevision),)
   ASTERISKVERSIONNUM=999999
-  ifneq ($(wildcard CVS/Tag),)
-    ASTERISKVERSION=$(shell echo "CVS-`sed 's/^T//g' CVS/Tag`-`date +"%D-%T"`")
-  else
-    ASTERISKVERSION=CVS HEAD
-  endif
+  ASTERISKVERSION:=SVN-$(shell cat .svnbranch)-r$(shell cat .svnrevision)
 else
-  ASTERISKVERSIONNUM=000000
+  ifneq ($(wildcard .svn),)
+    ASTERISKVERSIONNUM=999999
+    ASTERISKVERSION=SVN-$(shell build_tools/make_svn_branch_name)
+  endif
 endif
 
 ASTCFLAGS+= $(DEBUG_THREADS)
 ASTCFLAGS+= $(TRACE_FRAMES)
 ASTCFLAGS+= $(MALLOC_DEBUG)
 ASTCFLAGS+= $(BUSYDETECT)
+ASTCFLAGS+= $(MIDCOM)
 ASTCFLAGS+= $(OPTIONS)
+ifneq ($(findstring dont-optimize,$(MAKECMDGOALS)),dont-optimize)
 ASTCFLAGS+= -fomit-frame-pointer 
+endif
 SUBDIRS=res channels pbx apps codecs formats agi cdr funcs utils stdtime
 
 OBJS=io.o sched.o logger.o frame.o loader.o config.o channel.o \
@@ -316,37 +361,39 @@ ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/dlfcn.h),)
   ASTCFLAGS+=-DDLFCNCOMPAT
 endif
 
-ifeq (${OSARCH},Linux)
+ifeq ($(OSARCH),Linux)
   LIBS+=-ldl -lpthread -lncurses -lm -lresolv  #-lnjamd
 else
   LIBS+=-lncurses -lm
 endif
 
-ifeq (${OSARCH},Darwin)
+ifeq ($(OSARCH),Darwin)
   LIBS+=-lresolv
   ASTCFLAGS+=-D__Darwin__
   AUDIO_LIBS=-framework CoreAudio
   ASTLINK=-Wl,-dynamic
   SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
+  OBJS+= poll.o
+  ASTCFLAGS+=-DPOLLCOMPAT
 else
 #These are used for all but Darwin
   ASTLINK=-Wl,-E 
   SOLINK=-shared -Xlinker -x
 endif
 
-ifeq (${OSARCH},FreeBSD)
+ifeq ($(OSARCH),FreeBSD)
   LIBS+=-lcrypto
 endif
 
-ifeq (${OSARCH},NetBSD)
+ifeq ($(OSARCH),NetBSD)
   LIBS+=-lpthread -lcrypto -lm -L$(CROSS_COMPILE_TARGET)/usr/pkg/lib -lncurses
 endif
 
-ifeq (${OSARCH},OpenBSD)
+ifeq ($(OSARCH),OpenBSD)
   LIBS+=-lcrypto -lpthread -lm -lncurses
 endif
 
-ifeq (${OSARCH},SunOS)
+ifeq ($(OSARCH),SunOS)
   LIBS+=-lpthread -ldl -lnsl -lsocket -lresolv -L$(CROSS_COMPILE_TARGET)/usr/local/ssl/lib
   OBJS+=strcompat.o
   ASTLINK=
@@ -357,6 +404,13 @@ ifeq ($(MAKETOPLEVEL),$(MAKELEVEL))
   CFLAGS+=$(ASTCFLAGS)
 endif
 
+# This is used when generating the doxygen documentation
+ifneq ($(wildcard /usr/local/bin/dot)$(wildcard /usr/bin/dot),)
+  HAVEDOT=yes
+else
+  HAVEDOT=no
+endif
+
 LIBS+=-lssl
 
 INSTALL=install
@@ -383,10 +437,10 @@ endif
 noclean: depend asterisk subdirs
 
 editline/config.h:
-	cd editline && unset CFLAGS LIBS && ./configure ; \
+	cd editline && unset CFLAGS LIBS && CFLAGS="$(OPTIMIZE)" ./configure ; \
 
 editline/libedit.a: FORCE
-	cd editline && unset CFLAGS LIBS && test -f config.h || ./configure
+	cd editline && unset CFLAGS LIBS && test -f config.h || CFLAGS="$(OPTIMIZE)" ./configure
 	$(MAKE) -C editline libedit.a
 
 db1-ast/libdb1.a: FORCE
@@ -444,18 +498,6 @@ defaults.h: FORCE
 	fi
 	rm -f $@.tmp
 
-include/asterisk/build.h:
-	build_tools/make_build_h > $@.tmp
-	if cmp -s $@.tmp $@ ; then echo ; else \
-		mv $@.tmp $@ ; \
-	fi
-	rm -f $@.tmp
-
-# only force 'build.h' to be made for a non-'install' run
-ifeq ($(findstring install,$(MAKECMDGOALS)),)
-include/asterisk/build.h: FORCE
-endif
-
 include/asterisk/version.h: FORCE
 	build_tools/make_version_h > $@.tmp
 	if cmp -s $@.tmp $@ ; then echo; else \
@@ -471,8 +513,17 @@ stdtime/libtime.a: FORCE
 		exit 1; \
 	fi
 
-asterisk: editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a $(OBJS)
-	$(CC) $(DEBUG) -o asterisk $(ASTLINK) $(OBJS) $(LIBEDIT) db1-ast/libdb1.a stdtime/libtime.a $(LIBS)
+cygwin_a:
+	$(MAKE) -C cygwin all
+
+asterisk: $(CYGLOADER) editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a $(OBJS)
+	build_tools/make_build_h > include/asterisk/build.h.tmp
+	if cmp -s include/asterisk/build.h.tmp include/asterisk/build.h ; then echo ; else \
+		mv include/asterisk/build.h.tmp include/asterisk/build.h ; \
+	fi
+	rm -f include/asterisk/build.h.tmp
+	$(CC) -c -o buildinfo.o $(CFLAGS) buildinfo.c
+	$(CC) $(DEBUG) $(ASTOBJ) $(ASTLINK) $(OBJS) buildinfo.o $(LIBEDIT) db1-ast/libdb1.a stdtime/libtime.a $(LIBS)
 
 muted: muted.o
 	$(CC) $(AUDIO_LIBS) -o muted muted.o
@@ -493,7 +544,7 @@ clean:
 	$(MAKE) -C stdtime clean
 
 datafiles: all
-	if test $$(id -u) = 0; then sh mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
+	if [ x`$(ID) -un` = xroot ]; then sh mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
 	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/digits
 	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/priv-callerintros
 	for x in sounds/digits/*.gsm; do \
@@ -547,17 +598,16 @@ datafiles: all
 	mkdir -p $(DESTDIR)$(AGI_DIR)
 
 update: 
-	@if [ -d CVS ]; then \
-		if [ -f patches/.applied ]; then \
-			patches=`cat patches/.applied`; \
-		fi; \
-		if [ ! -z "$$patches" ]; then \
-			for x in $$patches; do \
-				echo "Unapplying $$x..."; \
-				patch -R -p0 < patches/$$x; \
-			done; \
-			rm -f patches/.applied; \
+	@if [ -d .svn ]; then \
+		echo "Updating from Subversion..." ; \
+		svn update | tee update.out; \
+		rm -f .version; \
+		if [ `grep -c ^C update.out` -gt 0 ]; then \
+			echo ; echo "The following files have conflicts:" ; \
+			grep ^C update.out | cut -b4- ; \
 		fi ; \
+		rm -f update.out; \
+	elif [ -d CVS ]; then \
 		echo "Updating from CVS..." ; \
 		cvs -q -z3 update -Pd | tee update.out; \
 		rm -f .version; \
@@ -566,19 +616,8 @@ update:
 			grep ^C update.out | cut -d' ' -f2- ; \
 		fi ; \
 		rm -f update.out; \
-		if [ ! -z "$$patches" ]; then \
-			for x in $$patches; do \
-				if [ -f patches/$$x ]; then \
-					echo "Applying patch $$x..."; \
-					patch -p0 < patches/$$x; \
-					echo $$x >> patches/.applied; \
-				else \
-					echo "Patch $$x no longer relevant"; \
-				fi; \
-			done; \
-		fi; \
 	else \
-		echo "Not CVS";  \
+		echo "Not under version control";  \
 	fi
 
 NEWHEADERS=$(notdir $(wildcard include/asterisk/*.h))
@@ -595,7 +634,10 @@ bininstall: all
 	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/system
 	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/tmp
 	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/meetme
-	$(INSTALL) -m 755 asterisk $(DESTDIR)$(ASTSBINDIR)/
+	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/monitor
+	if [ -f asterisk ]; then $(INSTALL) -m 755 asterisk $(DESTDIR)$(ASTSBINDIR)/; fi
+	if [ -f cygwin/asterisk.exe ]; then $(INSTALL) -m 755 cygwin/asterisk.exe $(DESTDIR)$(ASTSBINDIR)/; fi
+	if [ -f asterisk.dll ]; then $(INSTALL) -m 755 asterisk.dll $(DESTDIR)$(ASTSBINDIR)/; fi
 	ln -sf asterisk $(DESTDIR)$(ASTSBINDIR)/rasterisk
 	$(INSTALL) -m 755 contrib/scripts/astgenkey $(DESTDIR)$(ASTSBINDIR)/
 	$(INSTALL) -m 755 contrib/scripts/autosupport $(DESTDIR)$(ASTSBINDIR)/	
@@ -603,13 +645,11 @@ bininstall: all
 		cat contrib/scripts/safe_asterisk | sed 's|__ASTERISK_SBIN_DIR__|$(ASTSBINDIR)|;' > $(DESTDIR)$(ASTSBINDIR)/safe_asterisk ;\
 		chmod 755 $(DESTDIR)$(ASTSBINDIR)/safe_asterisk;\
 	fi
-	for x in $(SUBDIRS); do $(MAKE) -C $$x install || exit 1 ; done
 	$(INSTALL) -d $(DESTDIR)$(ASTHEADERDIR)
 	$(INSTALL) -m 644 include/asterisk/*.h $(DESTDIR)$(ASTHEADERDIR)
 	if [ -n "$(OLDHEADERS)" ]; then \
 		rm -f $(addprefix $(DESTDIR)$(ASTHEADERDIR)/,$(OLDHEADERS)) ;\
 	fi
-	rm -f $(DESTDIR)$(ASTVARLIBDIR)/sounds/voicemail
 	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds
 	mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-csv
 	mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-custom
@@ -628,30 +668,10 @@ bininstall: all
 	else \
 		echo "You need to do cvs update -d not just cvs update" ; \
 	fi 
-	( cd $(DESTDIR)$(ASTVARLIBDIR)/sounds  ; ln -s $(ASTSPOOLDIR)/voicemail . )
 	if [ -f mpg123-0.59r/mpg123 ]; then $(MAKE) -C mpg123-0.59r install; fi
-	@echo " +---- Asterisk Installation Complete -------+"  
-	@echo " +                                           +"
-	@echo " +    YOU MUST READ THE SECURITY DOCUMENT    +"
-	@echo " +                                           +"
-	@echo " + Asterisk has successfully been installed. +"  
-	@echo " + If you would like to install the sample   +"  
-	@echo " + configuration files (overwriting any      +"
-	@echo " + existing config files), run:              +"  
-	@echo " +                                           +"
-	@echo " +               $(MAKE) samples                +"
-	@echo " +                                           +"
-	@echo " +-----------------  or ---------------------+"
-	@echo " +                                           +"
-	@echo " + You can go ahead and install the asterisk +"
-	@echo " + program documentation now or later run:   +"
-	@echo " +                                           +"
-	@echo " +              $(MAKE) progdocs                +"
-	@echo " +                                           +"
-	@echo " + **Note** This requires that you have      +"
-	@echo " + doxygen installed on your local system    +"
-	@echo " +-------------------------------------------+"
-	@$(MAKE) -s oldmodcheck
+
+install-subdirs:
+	for x in $(SUBDIRS); do $(MAKE) -C $$x install || exit 1 ; done
 
 NEWMODS=$(notdir $(wildcard */*.so))
 OLDMODS=$(filter-out $(NEWMODS),$(notdir $(wildcard $(DESTDIR)$(MODULES_DIR)/*.so)))
@@ -674,10 +694,32 @@ oldmodcheck:
 		echo " WARNING WARNING WARNING" ;\
 	fi
 
-install: all datafiles bininstall
+install: all datafiles bininstall install-subdirs
 	@if [ -x /usr/sbin/asterisk-post-install ]; then \
 		/usr/sbin/asterisk-post-install $(DESTDIR) . ; \
 	fi
+	@echo " +---- Asterisk Installation Complete -------+"  
+	@echo " +                                           +"
+	@echo " +    YOU MUST READ THE SECURITY DOCUMENT    +"
+	@echo " +                                           +"
+	@echo " + Asterisk has successfully been installed. +"  
+	@echo " + If you would like to install the sample   +"  
+	@echo " + configuration files (overwriting any      +"
+	@echo " + existing config files), run:              +"  
+	@echo " +                                           +"
+	@echo " +               $(MAKE) samples                +"
+	@echo " +                                           +"
+	@echo " +-----------------  or ---------------------+"
+	@echo " +                                           +"
+	@echo " + You can go ahead and install the asterisk +"
+	@echo " + program documentation now or later run:   +"
+	@echo " +                                           +"
+	@echo " +              $(MAKE) progdocs                +"
+	@echo " +                                           +"
+	@echo " + **Note** This requires that you have      +"
+	@echo " + doxygen installed on your local system    +"
+	@echo " +-------------------------------------------+"
+	@$(MAKE) -s oldmodcheck
 
 upgrade: all bininstall
 
@@ -706,22 +748,24 @@ samples: adsi
 		fi ; \
 		$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` ;\
 	done
-	if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTETCDIR)/asterisk.conf ]; then \
-		echo "[directories]" > $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo "astetcdir => $(ASTETCDIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo "astmoddir => $(MODULES_DIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo "astvarlibdir => $(ASTVARLIBDIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo "astagidir => $(AGI_DIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo "astspooldir => $(ASTSPOOLDIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo "astrundir => $(ASTVARRUNDIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo "astlogdir => $(ASTLOGDIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo "" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo "; Changing the following lines may compromise your security." >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo ";[files]" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo ";astctlpermissions = 0660" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo ";astctlowner = root" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo ";astctlgroup = apache" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
-		echo ";astctl = asterisk.ctl" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf ; \
+	if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTCONFPATH) ]; then \
+		( \
+		echo "[directories]" ; \
+		echo "astetcdir => $(ASTETCDIR)" ; \
+		echo "astmoddir => $(MODULES_DIR)" ; \
+		echo "astvarlibdir => $(ASTVARLIBDIR)" ; \
+		echo "astagidir => $(AGI_DIR)" ; \
+		echo "astspooldir => $(ASTSPOOLDIR)" ; \
+		echo "astrundir => $(ASTVARRUNDIR)" ; \
+		echo "astlogdir => $(ASTLOGDIR)" ; \
+		echo "" ; \
+		echo "; Changing the following lines may compromise your security." ; \
+		echo ";[files]" ; \
+		echo ";astctlpermissions = 0660" ; \
+		echo ";astctlowner = root" ; \
+		echo ";astctlgroup = apache" ; \
+		echo ";astctl = asterisk.ctl" ; \
+		) > $(DESTDIR)$(ASTCONFPATH) ; \
 	else \
 		echo "Skipping asterisk.conf creation"; \
 	fi
@@ -786,11 +830,12 @@ __rpm: include/asterisk/version.h spec
 	$(MAKE) DESTDIR=/tmp/asterisk install ; \
 	$(MAKE) DESTDIR=/tmp/asterisk samples ; \
 	mkdir -p /tmp/asterisk/etc/rc.d/init.d ; \
-	cp -f redhat/asterisk /tmp/asterisk/etc/rc.d/init.d/ ; \
+	cp -f contrib/init.d/rc.redhat.asterisk /tmp/asterisk/etc/rc.d/init.d/asterisk ; \
 	rpmbuild --rcfile /usr/lib/rpm/rpmrc:redhat/rpmrc -bb asterisk.spec
 
 progdocs:
-	doxygen contrib/asterisk-ng-doxygen
+	(cat contrib/asterisk-ng-doxygen; echo "HAVE_DOT=$(HAVEDOT)"; \
+	echo "PROJECT_NUMBER=$(ASTERISKVERSION)") | doxygen - 
 
 mpg123:
 	@wget -V >/dev/null || (echo "You need wget" ; false )
@@ -806,23 +851,22 @@ config:
 		$(INSTALL) -m 755 init.asterisk /etc/init.d/asterisk; \
 	fi 
 
-dont-optimize:
-	$(MAKE) OPTIMIZE= K6OPT= install
+dont-optimize: install
 
 valgrind: dont-optimize
 
-depend: include/asterisk/build.h include/asterisk/version.h .depend defaults.h 
+depend: include/asterisk/version.h .depend defaults.h 
 	for x in $(SUBDIRS); do $(MAKE) -C $$x depend || exit 1 ; done
 
 .depend: include/asterisk/version.h
-	build_tools/mkdep ${CFLAGS} $(wildcard *.c)
+	build_tools/mkdep $(CFLAGS) $(wildcard *.c)
 
 .tags-depend:
 	@echo -n ".tags-depend: " > $@
 	@find . -maxdepth 1 -name \*.c -printf "\t%p \\\\\n" >> $@
 	@find . -maxdepth 1 -name \*.h -printf "\t%p \\\\\n" >> $@
-	@find ${SUBDIRS} -name \*.c -printf "\t%p \\\\\n" >> $@
-	@find ${SUBDIRS} -name \*.h -printf "\t%p \\\\\n" >> $@
+	@find $(SUBDIRS) -name \*.c -printf "\t%p \\\\\n" >> $@
+	@find $(SUBDIRS) -name \*.h -printf "\t%p \\\\\n" >> $@
 	@find include -name \*.h -printf "\t%p \\\\\n" >> $@
 	@echo >> $@
 
@@ -830,8 +874,8 @@ depend: include/asterisk/build.h include/asterisk/version.h .depend defaults.h
 	@rm -f $@
 	@find . -maxdepth 1 -name \*.c -print >> $@
 	@find . -maxdepth 1 -name \*.h -print >> $@
-	@find ${SUBDIRS} -name \*.c -print >> $@
-	@find ${SUBDIRS} -name \*.h -print >> $@
+	@find $(SUBDIRS) -name \*.c -print >> $@
+	@find $(SUBDIRS) -name \*.h -print >> $@
 	@find include -name \*.h -print >> $@
 
 tags: .tags-depend .tags-sources
@@ -861,47 +905,3 @@ cleantest:
 	if cmp -s .cleancount .lastclean ; then echo ; else \
 		$(MAKE) clean; cp -f .cleancount .lastclean;\
 	fi
-
-patchlist:
-	@echo "Experimental Patches:"
-	@for x in patches/*; do \
-		patch=`basename $$x`; \
-		if [ "$$patch" = "CVS" ]; then \
-			continue; \
-		fi; \
-		if grep -q ^$$patch$$ patches/.applied; then \
-			echo "$$patch (applied)"; \
-		else \
-			echo "$$patch (available)"; \
-		fi; \
-	done
-
-apply: 
-	@if [ -z "$(PATCH)" ]; then \
-		echo "Usage: make PATCH=<patchname> apply"; \
-	elif grep -q ^$(PATCH)$$ patches/.applied 2>/dev/null; then \
-		echo "Patch $(PATCH) is already applied"; \
-	elif [ -f "patches/$(PATCH)" ]; then \
-		echo "Applying patch $(PATCH)"; \
-		patch -p0 < patches/$(PATCH); \
-		echo "$(PATCH)" >> patches/.applied; \
-	else \
-		echo "No such patch $(PATCH) in patches directory"; \
-	fi
-
-unapply: 
-	@if [ -z "$(PATCH)" ]; then \
-		echo "Usage: make PATCH=<patchname> unapply"; \
-	elif grep -v -q ^$(PATCH)$$ patches/.applied 2>/dev/null; then \
-		echo "Patch $(PATCH) is not applied"; \
-	elif [ -f "patches/$(PATCH)" ]; then \
-		echo "Un-applying patch $(PATCH)"; \
-		patch -p0 -R < patches/$(PATCH); \
-		rm -f patches/.tmpapplied || :; \
-		mv patches/.applied patches/.tmpapplied; \
-		cat patches/.tmpapplied | grep -v ^$(PATCH)$$ > patches/.applied; \
-		rm -f patches/.tmpapplied; \
-	else \
-		echo "No such patch $(PATCH) in patches directory"; \
-	fi
-

README

@@ -67,7 +67,7 @@ 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/
+http://edvina.net/asterisk1-2/
 
 * NEW INSTALLATIONS
 
@@ -164,6 +164,14 @@ parameters.  For example, the line "channel => 25-47" creates objects for
 the channels 25 through 47 of the card, obtaining the settings
 from the variables specified above.
 
+* NETWORK SECURITY DEVICES
+
+This release of Asterisk contains support for network security devices
+manufactured by Ranch Networks, Inc., using their MIDCOM interface
+library. You will need the companion libmidcom-0.1.0.tar.gz file to
+build the library. Contact Ranch Networks' support department for assistance
+in building and configuring MIDCOM support.
+
 * SPECIAL NOTE ON TIME
   
   Those using SIP phones should be aware that Asterisk is sensitive to

README.fpm

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

README.opsound

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

UPGRADE.txt

@@ -1,5 +1,5 @@
-Information for Upgrading From Asterisk 1.0
-===========================================
+Information for Upgrading From Previous Asterisk Releases
+=========================================================
 
 Compiling:
 
@@ -34,6 +34,13 @@ Agents:
   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 
@@ -45,8 +52,10 @@ Dialing:
 
 IAX: 
 
-* The naming convention for IAX channels has changed in a minor way such 
-  that the call number follows a "-" rather than a "/" character.
+* The naming convention for IAX channels has changed in two ways: 
+   1. The call number follows a "-" rather than a "/" character.
+   2. The name of the channel has been simplified to IAX2/peer-callno,
+   rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.
 
 SIP:
 
@@ -122,6 +131,11 @@ Applications:
   'mailbox options' menu, and 'change your password' option has been
   moved to option 5.
 
+* The application VoiceMailMain now only matches the 'default' context if
+  none is specified in the arguments.  (This was the previously 
+  documented behavior, however, we didn't follow that behavior.)  The old
+  behavior can be restored by setting searchcontexts=yes in voicemail.conf.
+
 Queues:
 
 * A queue is now considered empty not only if there are no members but if
@@ -153,6 +167,9 @@ AGI:
   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

acl.c

@@ -68,6 +68,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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);

agi/.cvsignore

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

agi/Makefile

@@ -13,7 +13,7 @@
 
 AGIS=agi-test.agi eagi-test eagi-sphinx-test
 
-CFLAGS+=
+CFLAGS+=-DNO_AST_MM
 
 LIBS=
 ifeq ($(OSARCH),SunOS)

app.c

@@ -52,11 +52,12 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #define MAX_OTHER_FORMATS 10
 
 
-/* 
+/* !
 This function presents a dialtone and reads an extension into 'collect' 
 which must be a pointer to a **pre-initilized** array of char having a 
 size of 'size' suitable for writing to.  It will collect no more than the smaller 
 of 'maxlen' or 'size' minus the original strlen() of collect digits.
+\return 0 if extension does not exist, 1 if extension exists
 */
 int ast_app_dtget(struct ast_channel *chan, const char *context, char *collect, size_t size, int maxlen, int timeout) 
 {
@@ -103,7 +104,7 @@ int ast_app_dtget(struct ast_channel *chan, const char *context, char *collect,
 
 
 
-/* set timeout to 0 for "standard" timeouts. Set timeout to -1 for 
+/*! \param timeout set timeout to 0 for "standard" timeouts. Set timeout to -1 for 
    "ludicrous time" (essentially never times out) */
 int ast_app_getdata(struct ast_channel *c, char *prompt, char *s, int maxlen, int timeout)
 {
@@ -315,8 +316,12 @@ int ast_dtmf_stream(struct ast_channel *chan,struct ast_channel *peer,char *digi
 				}
 			}
 		}
-		if (peer)
-			res = ast_autoservice_stop(peer);
+		if (peer) {
+			/* Stop autoservice on the peer channel, but don't overwrite any error condition 
+			   that has occurred previously while acting on the primary channel */	
+			if (ast_autoservice_stop(peer) && !res)
+				res = -1;
+		}
 	}
 	return res;
 }
@@ -429,11 +434,11 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
 			   const char *stop, const char *pause,
 			   const char *restart, int skipms) 
 {
-	long elapsed = 0, last_elapsed = 0;
 	char *breaks = NULL;
 	char *end = NULL;
 	int blen = 2;
 	int res;
+	long pause_restart_point = 0;
 
 	if (stop)
 		blen += strlen(stop);
@@ -455,9 +460,6 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
 	if (chan->_state != AST_STATE_UP)
 		res = ast_answer(chan);
 
-	if (chan)
-		ast_stopstream(chan);
-
 	if (file) {
 		if ((end = strchr(file,':'))) {
 			if (!strcasecmp(end, ":end")) {
@@ -468,25 +470,18 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
 	}
 
 	for (;;) {
-		struct timeval started = ast_tvnow();
-
-		if (chan)
-			ast_stopstream(chan);
+		ast_stopstream(chan);
 		res = ast_streamfile(chan, file, chan->language);
 		if (!res) {
-			if (end) {
+			if (pause_restart_point) {
+				ast_seekstream(chan->stream, pause_restart_point, SEEK_SET);
+				pause_restart_point = 0;
+			}
+			else if (end) {
 				ast_seekstream(chan->stream, 0, SEEK_END);
-				end=NULL;
-			}
-			res = 1;
-			if (elapsed) {
-				ast_stream_fastforward(chan->stream, elapsed);
-				last_elapsed = elapsed - 200;
-			}
-			if (res)
-				res = ast_waitstream_fr(chan, breaks, fwd, rev, skipms);
-			else
-				break;
+				end = NULL;
+			};
+			res = ast_waitstream_fr(chan, breaks, fwd, rev, skipms);
 		}
 
 		if (res < 1)
@@ -495,17 +490,16 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
 		/* We go at next loop if we got the restart char */
 		if (restart && strchr(restart, res)) {
 			ast_log(LOG_DEBUG, "we'll restart the stream here at next loop\n");
-			elapsed=0; /* To make sure the next stream will start at beginning */
+			pause_restart_point = 0;
 			continue;
 		}
 
-		if (pause != NULL && strchr(pause, res)) {
-			elapsed = ast_tvdiff_ms(ast_tvnow(), started) + last_elapsed;
-			for(;;) {
-				if (chan)
-					ast_stopstream(chan);
+		if (pause && strchr(pause, res)) {
+			pause_restart_point = ast_tellstream(chan->stream);
+			for (;;) {
+				ast_stopstream(chan);
 				res = ast_waitfordigit(chan, 1000);
-				if (res == 0)
+				if (!res)
 					continue;
 				else if (res == -1 || strchr(pause, res) || (stop && strchr(stop, res)))
 					break;
@@ -515,17 +509,16 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
 				continue;
 			}
 		}
+
 		if (res == -1)
 			break;
 
 		/* if we get one of our stop chars, return it to the calling function */
-		if (stop && strchr(stop, res)) {
-			/* res = 0; */
+		if (stop && strchr(stop, res))
 			break;
-		}
 	}
-	if (chan)
-		ast_stopstream(chan);
+
+	ast_stopstream(chan);
 
 	return res;
 }
@@ -544,7 +537,7 @@ int ast_play_and_wait(struct ast_channel *chan, const char *fn)
 static int global_silence_threshold = 128;
 static int global_maxsilence = 0;
 
-int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int silencethreshold, int maxsilence, const char *path)
+int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int silencethreshold, int maxsilence, const char *path, const char *acceptdtmf, const char *canceldtmf)
 {
 	int d;
 	char *fmts;
@@ -558,8 +551,8 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
 	struct ast_dsp *sildet=NULL;   	/* silence detector dsp */
 	int totalsilence = 0;
 	int dspsilence = 0;
-	int gotsilence = 0;		/* did we timeout for silence? */
 	int rfmt=0;
+	struct ast_silence_generator *silgen = NULL;
 
 	if (silencethreshold < 0)
 		silencethreshold = global_silence_threshold;
@@ -595,7 +588,7 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
 
 	while((fmt = strsep(&stringp, "|"))) {
 		if (fmtcnt > MAX_OTHER_FORMATS - 1) {
-			ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app_voicemail.c\n");
+			ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app.c\n");
 			break;
 		}
 		sfmt[fmtcnt++] = ast_strdupa(fmt);
@@ -615,8 +608,6 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
 	if (path)
 		ast_unlock_path(path);
 
-
-	
 	if (maxsilence > 0) {
 		sildet = ast_dsp_new(); /* Create the silence detector */
 		if (!sildet) {
@@ -632,9 +623,13 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
 			return -1;
 		}
 	}
+
 	/* Request a video update */
 	ast_indicate(chan, AST_CONTROL_VIDUPDATE);
 
+	if (option_transmit_silence_during_record)
+		silgen = ast_channel_start_silence_generator(chan);
+
 	if (x == fmtcnt) {
 	/* Loop forever, writing the packets we read to the writer(s), until
 	   we read a # or get a hangup */
@@ -678,7 +673,7 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
 						if (option_verbose > 2)
 							ast_verbose( VERBOSE_PREFIX_3 "Recording automatically stopped after a silence of %d seconds\n", totalsilence/1000);
 						ast_frfree(f);
-						gotsilence = 1;
+						res = 'S';
 						outmsg=2;
 						break;
 					}
@@ -693,19 +688,18 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
 				/* Write only once */
 				ast_writestream(others[0], f);
 			} else if (f->frametype == AST_FRAME_DTMF) {
-				if (f->subclass == '#') {
+				if (strchr(acceptdtmf, f->subclass)) {
 					if (option_verbose > 2)
-						ast_verbose( VERBOSE_PREFIX_3 "User ended message by pressing %c\n", f->subclass);
-					res = '#';
+						ast_verbose(VERBOSE_PREFIX_3 "User ended message by pressing %c\n", f->subclass);
+					res = f->subclass;
 					outmsg = 2;
 					ast_frfree(f);
 					break;
 				}
-				if (f->subclass == '0') {
-				/* Check for a '0' during message recording also, in case caller wants operator */
+				if (strchr(canceldtmf, f->subclass)) {
 					if (option_verbose > 2)
-						ast_verbose(VERBOSE_PREFIX_3 "User cancelled by pressing %c\n", f->subclass);
-					res = '0';
+						ast_verbose(VERBOSE_PREFIX_3 "User cancelled message by pressing %c\n", f->subclass);
+					res = f->subclass;
 					outmsg = 0;
 					ast_frfree(f);
 					break;
@@ -735,6 +729,9 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
 		ast_log(LOG_WARNING, "Error creating writestream '%s', format '%s'\n", recordfile, sfmt[x]);
 	}
 
+	if (silgen)
+		ast_channel_stop_silence_generator(chan, silgen);
+
 	*duration = end - start;
 
 	for (x=0;x<fmtcnt;x++) {
@@ -764,6 +761,14 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
 	return res;
 }
 
+static char default_acceptdtmf[] = "#";
+static char default_canceldtmf[] = "0";
+
+int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int silencethreshold, int maxsilence, const char *path)
+{
+	return ast_play_and_record_full(chan, playfile, recordfile, maxtime, fmt, duration, silencethreshold, maxsilence, path, default_acceptdtmf, default_canceldtmf);
+}
+
 int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int beep, int silencethreshold, int maxsilence)
 {
 	int d = 0;
@@ -779,7 +784,6 @@ int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordf
 	struct ast_dsp *sildet;   	/* silence detector dsp */
 	int totalsilence = 0;
 	int dspsilence = 0;
-	int gotsilence = 0;		/* did we timeout for silence? */
 	int rfmt=0;	
 	char prependfile[80];
 	
@@ -820,7 +824,7 @@ int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordf
 	
 	while((fmt = strsep(&stringp, "|"))) {
 		if (fmtcnt > MAX_OTHER_FORMATS - 1) {
-			ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app_voicemail.c\n");
+			ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app.c\n");
 			break;
 		}
 		sfmt[fmtcnt++] = ast_strdupa(fmt);
@@ -848,6 +852,7 @@ int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordf
 		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");
+			ast_dsp_free(sildet);
 			return -1;
 		}
 	}
@@ -897,7 +902,7 @@ int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordf
 					if (option_verbose > 2) 
 						ast_verbose( VERBOSE_PREFIX_3 "Recording automatically stopped after a silence of %d seconds\n", totalsilence/1000);
 					ast_frfree(f);
-					gotsilence = 1;
+					res = 'S';
 					outmsg=2;
 					break;
 					}
@@ -952,6 +957,7 @@ int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordf
 	} else {
 		ast_log(LOG_WARNING, "Error creating writestream '%s', format '%s'\n", prependfile, sfmt[x]); 
 	}
+	ast_dsp_free(sildet);
 	*duration = end - start;
 #if 0
 	if (outmsg > 1) {
@@ -1099,7 +1105,7 @@ int ast_app_group_match_get_count(char *groupmatch, char *category)
 	return count;
 }
 
-int ast_separate_app_args(char *buf, char delim, char **array, int arraylen)
+unsigned int ast_app_separate_args(char *buf, char delim, char **array, int arraylen)
 {
 	int argc;
 	char *scan;
@@ -1517,41 +1523,41 @@ char *ast_read_textfile(const char *filename)
 	return output;
 }
 
-int ast_parseoptions(const struct ast_option *options, struct ast_flags *flags, char **args, char *optstr)
+int ast_app_parse_options(const struct ast_app_option *options, struct ast_flags *flags, char **args, char *optstr)
 {
 	char *s;
 	int curarg;
-	int argloc;
+	unsigned int argloc;
 	char *arg;
 	int res = 0;
 
-	flags->flags = 0;
+	ast_clear_flag(flags, AST_FLAGS_ALL);
 
 	if (!optstr)
 		return 0;
 
 	s = optstr;
 	while (*s) {
-		curarg = *s & 0x7f;
-		flags->flags |= options[curarg].flag;
+		curarg = *s++ & 0x7f;
+		ast_set_flag(flags, options[curarg].flag);
 		argloc = options[curarg].arg_index;
-		s++;
 		if (*s == '(') {
 			/* Has argument */
-			s++;
-			arg = s;
-			while (*s && (*s != ')')) s++;
+			arg = ++s;
+			while (*s && (*s != ')'))
+				s++;
 			if (*s) {
 				if (argloc)
 					args[argloc - 1] = arg;
-				*s = '\0';
-				s++;
+				*s++ = '\0';
 			} else {
 				ast_log(LOG_WARNING, "Missing closing parenthesis for argument '%c' in string '%s'\n", curarg, arg);
 				res = -1;
 			}
-		} else if (argloc)
+		} else if (argloc) {
 			args[argloc - 1] = NULL;
+		}
 	}
+
 	return res;
 }

apps/.cvsignore

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

apps/Makefile

@@ -15,21 +15,21 @@ 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_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 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_mixmonitor.so app_stack.so
 
 #
 # Obsolete things...
@@ -44,9 +44,11 @@ APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
 #APPS+=app_skel.so
 #APPS+=app_rpt.so
 
+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 $(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
 APPS+=app_osplookup.so
@@ -63,7 +65,13 @@ ifneq ($(shell if [[ 0x`$(CROSS_COMPILE_BIN)curl-config --vernum` -ge 0x70907 ]]
   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
+APPS+=app_sms.so
+endif
 
 #
 # If you have UnixODBC you can use ODBC voicemail
@@ -78,13 +86,13 @@ CFLAGS+=-fPIC
 all: $(APPS)
 
 clean:
-	rm -f *.so *.o look .depend
+	rm -f *.so *.o .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
@@ -92,19 +100,16 @@ install: all
 	rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
 
 app_curl.so: app_curl.o
-	$(CC) $(SOLINK) -o $@ $< $(CURLLIBS)
+	$(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
+	$(CC) -pipe -I$(CROSS_COMPILE_TARGET)/usr/local/pgsql/include -I$(CROSS_COMPILE_TARGET)/usr/include/postgresql $(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
-
-look:	look.c
-	$(CC) -pipe -O6 -g look.c -o look -lncurses
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -lodbc
 
 ifeq (SunOS,$(shell uname))
 app_chanspy.so: app_chanspy.o

apps/app_adsiprog.c

@@ -19,6 +19,7 @@
 /*! \file
  * \brief Program Asterisk ADSI Scripts into phone
  * 
+ * \ingroup applications
  */
 
 #include <sys/types.h>
@@ -54,9 +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;
 

apps/app_alarmreceiver.c

@@ -18,6 +18,7 @@
 
 /*! \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 *** 
  *
@@ -25,6 +26,7 @@
  *
  * *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING ***
  *
+ * \ingroup applications
  */ 
  
 #include <string.h>
@@ -71,13 +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 */
 
@@ -554,11 +558,10 @@ static int receive_ademco_contact_id( struct ast_channel *chan, void *data, int
 
 		if(checksum){
 			database_increment("checksum-errors");
-			if(option_verbose >= 2){
+			if(option_verbose >= 2)
 				ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Nonzero checksum\n");
 			ast_log(LOG_DEBUG, "AlarmReceiver: Nonzero checksum\n");
 			continue;
-			}
 		}
 
 		/* Check the message type for correctness */

apps/app_authenticate.c

@@ -19,6 +19,7 @@
 /*! \file
  * \brief Execute arbitrary authenticate commands
  * 
+ * \ingroup applications
  */
 
 #include <stdlib.h>
@@ -48,27 +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"
-"     m - Interpret path as a file which contains a list of\n"
-"         account codes and password hashes delimited with ':'\n"
-"         one per line. When password matched, corresponding\n"
-"         account code will be set\n"
-"     j - Support jumping to n+101\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 on hangup.  If the priority n+101 exists and invalid\n"
-"authentication was entered, and the 'j' flag was specified, processing\n"
-"will jump to n+101 and 0 will be returned.\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;
 

apps/app_cdr.c

@@ -19,6 +19,7 @@
 /*! \file
  * \brief Applications connected with CDR engine
  *
+ * \ingroup applications
  */
 
 #include <sys/types.h>
@@ -33,11 +34,14 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 
 
-static char *tdesc = "Make sure asterisk doesn't save CDR for a certain call";
+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;
 

apps/app_chanisavail.c

@@ -1,32 +1,33 @@
 /*
- * 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.
- */
+* 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
- * 
+* \brief Check if Channel is Available
+* 
+ * \ingroup applications
  */
 
 #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>
 
 #include "asterisk.h"
@@ -41,26 +42,24 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/module.h"
 #include "asterisk/app.h"
 #include "asterisk/devicestate.h"
+#include "asterisk/options.h"
 
-static char *tdesc = "Check if channel is available";
+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...][|option]): \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"
-"If the option 's' is specified (state), will consider channel unavailable\n"
-"when the channel is in use at all, even if it can take another call.\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;
 
@@ -68,7 +67,7 @@ LOCAL_USER_DECL;
 
 static int chanavail_exec(struct ast_channel *chan, void *data)
 {
-	int res=-1, inuse=-1, option_state=0;
+	int res=-1, inuse=-1, option_state=0, priority_jump=0;
 	int status;
 	struct localuser *u;
 	char *info, tmp[512], trychan[512], *peers, *tech, *number, *rest, *cur, *options, *stringp;
@@ -85,8 +84,12 @@ static int chanavail_exec(struct ast_channel *chan, void *data)
 	stringp = info;
 	strsep(&stringp, "|");
 	options = strsep(&stringp, "|");
-	if (options && *options == 's')
-		option_state = 1;
+	if (options) {
+		if (strchr(options, 's'))
+			option_state = 1;
+		if (strchr(options, 'j'))
+			priority_jump = 1;
+	}
 	peers = info;
 	if (peers) {
 		cur = peers;
@@ -136,9 +139,11 @@ static int chanavail_exec(struct ast_channel *chan, void *data)
 	if (res < 1) {
 		pbx_builtin_setvar_helper(chan, "AVAILCHAN", "");
 		pbx_builtin_setvar_helper(chan, "AVAILORIGCHAN", "");
-		if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
-			LOCAL_USER_REMOVE(u);
-			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;
+			}
 		}
 	}
 

apps/app_chanspy.c

@@ -19,11 +19,13 @@
 /*! \file
  * \brief ChanSpy: Listen in on any channel.
  * 
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
 #include <ctype.h>
 
 #include "asterisk.h"
@@ -33,6 +35,7 @@ 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"
@@ -49,38 +52,55 @@ AST_MUTEX_DEFINE_STATIC(modlock);
 #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 = "Tap into any type of asterisk channel and listen to audio";
+static const char *synopsis = "Listen to the audio of an active channel\n";
 static const char *app = "ChanSpy";
-static const char *desc = "   Chanspy([<scanspec>][|<options>])\n\n"
-"Valid Options:\n"
-" - q: quiet, don't announce channels beep, etc.\n"
-" - b: bridged, only spy on channels involved in a bridged call.\n"
-" - v([-4..4]): adjust the initial volume. (negative is quieter)\n"
-" - g(grp): enforce group.  Match only calls where their ${SPYGROUP} is 'grp'.\n"
-" - r[(basename)]: Record session to monitor spool dir (with optional basename, default is 'chanspy')\n\n"
-"If <scanspec> is specified, only channel names *beginning* with that string will be scanned.\n"
-"('all' or an empty string are also both valid <scanspec>)\n\n"
-"While Spying:\n\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 to <scanspec>\n"
-"(e.g. run Chanspy(Agent) and dial 1234# while spying to jump to channel Agent/1234)\n\n"
-"";
+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";
 
-#define OPTION_QUIET	 (1 << 0)	/* Quiet, no announcement */
-#define OPTION_BRIDGED   (1 << 1)	/* Only look at bridged calls */
-#define OPTION_VOLUME    (1 << 2)	/* Specify initial volume */
-#define OPTION_GROUP     (1 << 3)   /* Only look at channels in group */
-#define OPTION_RECORD    (1 << 4)   /* Record */
+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;
 
-AST_DECLARE_OPTIONS(chanspy_opts,{
-	['q'] = { OPTION_QUIET },
-	['b'] = { OPTION_BRIDGED },
-	['v'] = { OPTION_VOLUME, 1 },
-	['g'] = { OPTION_GROUP, 2 },
-	['r'] = { OPTION_RECORD, 3 },
+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;
@@ -188,9 +208,9 @@ static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, s
 
 static void stop_spying(struct ast_channel *chan, struct ast_channel_spy *spy) 
 {
-	/* If our status has changed, then the channel we're spying on is gone....
+	/* If our status has changed to DONE, then the channel we're spying on is gone....
 	   DON'T TOUCH IT!!!  RUN AWAY!!! */
-	if (spy->status != CHANSPY_RUNNING)
+	if (spy->status == CHANSPY_DONE)
 		return;
 
 	if (!chan)
@@ -384,7 +404,7 @@ static int chanspy_exec(struct ast_channel *chan, void *data)
 
 	ast_set_flag(chan, AST_FLAG_SPYING); /* so nobody can spy on us while we are spying */
 
-	if ((argc = ast_separate_app_args(args, '|', argv, sizeof(argv) / sizeof(argv[0])))) {
+	if ((argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0])))) {
 		spec = argv[0];
 		if ( argc > 1) {
 			options = argv[1];
@@ -395,8 +415,8 @@ static int chanspy_exec(struct ast_channel *chan, void *data)
 	}
 	
 	if (options) {
-		char *opts[3];
-		ast_parseoptions(chanspy_opts, &flags, opts, 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];
 		}
@@ -419,8 +439,8 @@ static int chanspy_exec(struct ast_channel *chan, void *data)
 
 	if (recbase) {
 		char filename[512];
-		snprintf(filename,sizeof(filename),"%s/%s.%ld.raw",ast_config_AST_MONITOR_DIR, recbase, time(NULL));
-		if ((fd = open(filename, O_CREAT | O_WRONLY, O_TRUNC)) <= 0) {
+		snprintf(filename,sizeof(filename),"%s/%s.%d.raw",ast_config_AST_MONITOR_DIR, recbase, (int)time(NULL));
+		if ((fd = open(filename, O_CREAT | O_WRONLY, O_TRUNC, 0644)) <= 0) {
 			ast_log(LOG_WARNING, "Cannot open %s for recording\n", filename);
 			fd = 0;
 		}
@@ -462,7 +482,7 @@ static int chanspy_exec(struct ast_channel *chan, void *data)
 					}
 				}
 				
-				if (igrp && (!spec || ((strlen(spec) < strlen(peer->name) &&
+				if (igrp && (!spec || ((strlen(spec) <= strlen(peer->name) &&
 							!strncasecmp(peer->name, spec, strlen(spec)))))) {
 					if (peer && (!bronly || ast_bridged_channel(peer)) &&
 					    !ast_check_hangup(peer) && !ast_test_flag(peer, AST_FLAG_SPYING)) {

apps/app_controlplayback.c

@@ -19,10 +19,12 @@
 /*! \file
  * \brief Trivial application to control playback of a sound file
  * 
+ * \ingroup applications
  */
  
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -37,6 +39,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/module.h"
 #include "asterisk/translate.h"
 #include "asterisk/utils.h"
+#include "asterisk/options.h"
 
 static const char *tdesc = "Control Playback Application";
 
@@ -45,14 +48,22 @@ static const char *app = "ControlPlayback";
 static const char *synopsis = "Play a file with fast forward and rewind";
 
 static const char *descrip = 
-"ControlPlayback(filename[|skipms[|ffchar[|rewchar[|stopchar[|pausechar[|restartchar]]]]]]):\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. If 'restartchar' is added, the file\n"
-"  will restart when 'restartchar' is pressed. Returns -1 if the channel\n"
-"  was hung up. if the file does not exist jumps to n+101 if it present.\n\n"
-"  Example:  exten => 1234,1,ControlPlayback(file|4000|*|#|1|0|5)\n\n";
+"  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;
 
@@ -65,12 +76,12 @@ static int is_on_phonepad(char key)
 
 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;
 	int argc;
-	char *argv[7];
+	char *argv[8];
 	enum arg_ids {
 		arg_file = 0,
 		arg_skip = 1,
@@ -79,6 +90,7 @@ static int controlplayback_exec(struct ast_channel *chan, void *data)
 		arg_stop = 4,
 		arg_pause = 5,
 		arg_restart = 6,
+		options = 7,
 	};
 	
 	if (ast_strlen_zero(data)) {
@@ -91,7 +103,7 @@ static int controlplayback_exec(struct ast_channel *chan, void *data)
 	tmp = ast_strdupa(data);
 	memset(argv, 0, sizeof(argv));
 
-	argc = ast_separate_app_args(tmp, '|', argv, sizeof(argv) / sizeof(argv[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");
@@ -114,15 +126,28 @@ static int controlplayback_exec(struct ast_channel *chan, void *data)
 	if (argv[arg_restart] && !is_on_phonepad(*argv[arg_restart]))
 		argv[arg_restart] = NULL;
 
+	if (argv[options]) {
+		if (strchr(argv[options], 'j'))
+			priority_jump = 1;
+	}
+
 	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)) 
+	if (argv[arg_stop] && strchr(argv[arg_stop], res)) {
 		res = 0;
-
-	if (res < 0) {
-		if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
+		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);

apps/app_curl.c

@@ -20,8 +20,10 @@
 /*! \file
  * \brief Curl - App to load a URL
  * 
+ * \ingroup applications
  */
  
+#include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 #include <curl/curl.h>
@@ -46,10 +48,14 @@ static char *app = "Curl";
 static char *synopsis = "Load an external URL";
 
 static char *descrip = 
-"  Curl(URL[|postdata]): Requests the URL.  Mainly used for signalling\n"
-"external applications of an event.  Returns 0 or -1 on fatal error.\n"
-"Argument specified treated as POST data.  Also sets CURL variable with the\n"
-"resulting page.\n";
+"  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;
 

apps/app_cut.c

@@ -18,6 +18,7 @@
 /*! \file
  * \brief Cut application
  *
+ * \ingroup applications
  */
 
 #include <stdio.h>
@@ -35,34 +36,37 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
-#include "asterisk/version.h"
+#include "asterisk/app.h"
 
 /* Maximum length of any variable */
 #define MAXRESULT	1024
 
-static char *tdesc = "String manipulation";
+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>]])\n"
-"Sorts the list provided by using the value as a float to order the list of\n"
-"keywords in ascending order.  Sets the variable provided to the list of\n"
-"sorted keywords.  Always returns 0.\n";
+"  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;
 
@@ -141,8 +145,9 @@ static int sort_internal(struct ast_channel *chan, char *data, char *buffer, siz
 		int blen = strlen(buffer);
 		if (element_count++) {
 			strncat(buffer + blen, ",", buflen - blen - 1);
+			blen++;
 		}
-		strncat(buffer + blen + 1, sortable_keys[count2].key, buflen - blen - 2);
+		strncat(buffer + blen, sortable_keys[count2].key, buflen - blen - 1);
 	}
 
 	return 0;
@@ -150,7 +155,7 @@ static int sort_internal(struct ast_channel *chan, char *data, char *buffer, siz
 
 static int cut_internal(struct ast_channel *chan, char *data, char *buffer, size_t buflen)
 {
-	char *s, *varname=NULL, *delimiter=NULL, *field=NULL;
+	char *s, *args[3], *varname=NULL, *delimiter=NULL, *field=NULL;
 	int args_okay = 0;
 
 	memset(buffer, 0, buflen);
@@ -159,15 +164,13 @@ static int cut_internal(struct ast_channel *chan, char *data, char *buffer, size
 	if (data) {
 		s = ast_strdupa((char *)data);
 		if (s) {
-			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 {
 			return ERROR_NOMEM;
@@ -250,6 +253,8 @@ static int cut_internal(struct ast_channel *chan, char *data, char *buffer, size
 				}
 			}
 		}
+	} else {
+		return ERROR_NOARG;
 	}
 	return 0;
 }
@@ -383,7 +388,7 @@ static char *acf_cut_exec(struct ast_channel *chan, char *cmd, char *data, char
 
 	switch (cut_internal(chan, data, buf, len)) {
 	case ERROR_NOARG:
-		ast_log(LOG_ERROR, "Cut() requires an argument\n");
+		ast_log(LOG_ERROR, "CUT() requires an argument\n");
 		break;
 	case ERROR_NOMEM:
 		ast_log(LOG_ERROR, "Out of memory\n");

apps/app_datetime.c

@@ -19,8 +19,10 @@
 /*! \file
  * \brief Time of day - Report the time of day
  * 
+ * \ingroup applications
  */
 
+#include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
@@ -44,8 +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;
 

apps/app_db.c

@@ -22,11 +22,13 @@
  *
  * \brief Database access functions
  *
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
 #include <sys/types.h>
 
 #include "asterisk.h"
@@ -41,25 +43,32 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/module.h"
 #include "asterisk/astdb.h"
 #include "asterisk/lock.h"
+#include "asterisk/options.h"
 
-static char *tdesc = "Database access functions for Asterisk extension logic";
+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";
@@ -205,9 +214,10 @@ static int put_exec(struct ast_channel *chan, void *data)
 
 static int get_exec(struct ast_channel *chan, void *data)
 {
-	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;
 
 	LOCAL_USER_ADD(u);
@@ -227,23 +237,38 @@ static int get_exec(struct ast_channel *chan, void *data)
 	if (strchr(argv, '=') && strchr(argv, '/')) {
 		varname = strsep(&argv, "=");
 		family = strsep(&argv, "/");
-		key = strsep(&argv, "\0");
+		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");
 			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);
 			if (option_verbose > 2)
 				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 */
-			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+			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");

apps/app_dictate.c

@@ -22,11 +22,13 @@
  *
  * \brief Virtual Dictation Machine Application For Asterisk
  *
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
 #include <sys/stat.h>	/* for mkdir */
 
 #include "asterisk.h"
@@ -102,7 +104,7 @@ static int dictate_exec(struct ast_channel *chan, void *data)
 	
 	snprintf(dftbase, sizeof(dftbase), "%s/dictate", ast_config_AST_SPOOL_DIR);
 	if (!ast_strlen_zero(data) && (mydata = ast_strdupa(data))) {
-		argc = ast_separate_app_args(mydata, '|', argv, sizeof(argv) / sizeof(argv[0]));
+		argc = ast_app_separate_args(mydata, '|', argv, sizeof(argv) / sizeof(argv[0]));
 	}
 	
 	if (argc) {

apps/app_directed_pickup.c

@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2005, Joshua Colp
  *
- * Joshua Colp <jcolp@asterlink.com>
+ * Joshua Colp <jcolp@digium.com>
  *
  * See http://www.asterisk.org for more information about
  * the Asterisk project. Please do not directly contact
@@ -20,11 +20,13 @@
  *
  * \brief Directed Call Pickup Support
  *
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
 
 #include "asterisk.h"
 
@@ -40,10 +42,11 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 static const char *tdesc = "Directed Call Pickup Application";
 static const char *app = "Pickup";
-static const char *synopsis = "Directed Call Pickup application.";
+static const char *synopsis = "Directed Call Pickup";
 static const char *descrip =
-" Pickup(extension@context):\n"
-"Steals any calls to a specified extension that are in a ringing state and bridges them to the current channel. Context is an optional argument.\n";
+"  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;
 
@@ -74,7 +77,7 @@ static int pickup_exec(struct ast_channel *chan, void *data)
 
 	/* Find a channel to pickup */
 	origin = ast_get_channel_by_exten_locked(exten, context);
-	if (origin) {
+	if (origin && origin->cdr) {
 		ast_cdr_getvar(origin->cdr, "dstchannel", &tmp, workspace,
 			       sizeof(workspace), 0);
 		if (tmp) {
@@ -86,6 +89,8 @@ static int pickup_exec(struct ast_channel *chan, void *data)
 		}
 		ast_mutex_unlock(&origin->lock);
 	} else {
+		if (origin)
+			ast_mutex_unlock(&origin->lock);
 		ast_log(LOG_DEBUG, "No originating channel found.\n");
 	}
 	

apps/app_directory.c

@@ -20,6 +20,7 @@
  *
  * \brief Provide a directory of extensions
  * 
+ * \ingroup applications
  */
  
 #include <string.h>
@@ -46,18 +47,23 @@ 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.  If the user enters '0' and there\n"
-"exists an extension 'o' in the current context, the directory will exit with 0\n"
-"and call control will resume at that extension.  Entering '*' will exit similarly,\n"
-"but to the 'a' extension, much like app_voicemail's behavior.\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 */

apps/app_disa.c

@@ -3,7 +3,6 @@
  *
  * Copyright (C) 1999 - 2005, Digium, Inc.
  *
- * Jim Dixon <jim@lambdatel.com>
  *
  * Made only slightly more sane by Mark Spencer <markster@digium.com>
  *
@@ -21,7 +20,10 @@
 /*! \file
  *
  * \brief DISA -- Direct Inward System Access Application
+ *
+ * \author Jim Dixon <jim@lambdatel.com>
  * 
+ * \ingroup applications
  */
  
 #include <string.h>
@@ -93,7 +95,6 @@ static char *descrip =
 	"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;
@@ -117,14 +118,20 @@ static int disa_exec(struct ast_channel *chan, void *data)
 	int firstdigittimeout = 20000;
 	int digittimeout = 10000;
 	struct localuser *u;
-	char *tmp, arg2[256]="",exten[AST_MAX_EXTENSION],acctcode[20]="";
-	char *ourcontext,*ourcallerid,ourcidname[256],ourcidnum[256],*mailbox;
+	char *tmp, exten[AST_MAX_EXTENSION],acctcode[20]="";
+	char pwline[256];
+	char ourcidname[256],ourcidnum[256];
 	struct ast_frame *f;
 	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_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "disa requires an argument (passcode/passcode file)\n");
@@ -159,24 +166,14 @@ static int disa_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}	
 
-	stringp=tmp;
-	strsep(&stringp, "|");
-	ourcontext = strsep(&stringp, "|");
-	/* if context specified, save 2nd arg and parse third */
-	if (ourcontext) {
-		ast_copy_string(arg2, ourcontext, sizeof(arg2));
-		ourcallerid = strsep(&stringp,"|");
-	}
-	  /* if context not specified, use "disa" */
-	else {
-		arg2[0] = 0;
-		ourcallerid = NULL;
-		ourcontext = "disa";
-	}
-	mailbox = strsep(&stringp, "|");
-	if (!mailbox)
-		mailbox = "";
-	ast_log(LOG_DEBUG, "Mailbox: %s\n",mailbox);
+	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 */
@@ -188,15 +185,15 @@ 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");
 	}
 	lastdigittime = ast_tvnow();
 
-	play_dialtone(chan, mailbox);
+	play_dialtone(chan, args.mailbox);
 
 	for (;;) {
 		  /* if outa time, give em reorder */
@@ -252,48 +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;
-							}
-							mailbox = strsep(&stringp, "|");
-							if (!mailbox)
-								mailbox = "";
-							ast_log(LOG_DEBUG, "Mailbox: %s\n",mailbox);
+							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;
@@ -301,7 +295,7 @@ 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, mailbox);
+					play_dialtone(chan, args.mailbox);
 
 					k|=1; /* In number mode */
 					i = 0;  /* re-set buffer pointer */
@@ -318,7 +312,7 @@ 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)) {
+			if (ast_ignore_pattern(args.context, exten)) {
 				play_dialtone(chan, "");
 				did_ignore = 1;
 			} else
@@ -328,7 +322,7 @@ 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->cid.cid_num)) {
+			if (!ast_matchmore_extension(chan,args.context,exten,1, chan->cid.cid_num)) {
 				break;
 			}
 		}
@@ -336,27 +330,28 @@ static int disa_exec(struct ast_channel *chan, void *data)
 
 	if (k == 3) {
 		int recheck = 0;
+		struct ast_flags flags = { AST_CDR_FLAG_POSTED };
 
-		if (!ast_exists_extension(chan, ourcontext, exten, 1, chan->cid.cid_num)) {
+		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, ourcontext, exten, 1, chan->cid.cid_num)) {
+		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 (ourcallerid && *ourcallerid)
+			if (!ast_strlen_zero(args.cid))
 			{
-				ast_callerid_split(ourcallerid, ourcidname, sizeof(ourcidname), ourcidnum, sizeof(ourcidnum));
+				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, AST_CDR_FLAG_POSTED);
-			ast_explicit_goto(chan, ourcontext, exten, 1);
+			ast_cdr_reset(chan->cdr, &flags);
+			ast_explicit_goto(chan, args.context, exten, 1);
 			LOCAL_USER_REMOVE(u);
 			return 0;
 		}

apps/app_dumpchan.c

@@ -22,11 +22,13 @@
  *
  * \brief Application to dump channel variables
  * 
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
 
 #include "asterisk.h"
 
@@ -50,7 +52,7 @@ static char *desc =
 "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. Always returns 0.\n\n";
+"or greater. \n";
 
 STANDARD_LOCAL_USER;
 

apps/app_echo.c

@@ -20,9 +20,11 @@
  *
  * \brief Echo application -- play back what you hear to evaluate latency
  * 
+ * \ingroup applications
  */
 
 #include <stdlib.h>
+#include <stdio.h>
 #include <unistd.h>
 #include <string.h>
 
@@ -44,8 +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;
 

apps/app_enumlookup.c

@@ -20,12 +20,13 @@
  *
  * \brief Enumlookup - lookup entry in ENUM
  *
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
+#include <unistd.h>
 #include <ctype.h>
 
 #include "asterisk.h"
@@ -42,6 +43,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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";
 
@@ -50,20 +53,17 @@ 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"
+"  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\n"
 "Currently, the enumservices SIP, H323, IAX, IAX2 and TEL are recognized. \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"
-"\nOld, depreciated, behaviour:\n"
-"\nA SIP, H323, IAX or IAX2 entry will result in normal priority handling, \n"
-"whereas a 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" ;
+"  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"
 
@@ -77,13 +77,18 @@ 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;
+	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");
@@ -97,13 +102,30 @@ static int enumlookup_exec(struct ast_channel *chan, void *data)
 
 	LOCAL_USER_ADD(u);
 
-	tech[0] = '\0';
+	parse = ast_strdupa(data);
+	if (!parse) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
 
-	res = ast_get_enum(chan, data, dest, sizeof(dest), tech, sizeof(tech), NULL, NULL);
+	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 */
-		/* Look for a "busy" place */
-		ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		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;
@@ -158,8 +180,10 @@ 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_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 51))
-					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);

apps/app_eval.c

@@ -19,6 +19,8 @@
  * \brief Eval application
  *
  * \author Tilghman Lesher <app_eval__v001@the-tilghman.com>
+ *
+ * \ingroup applications
  */
 
 #include <stdio.h>

apps/app_exec.c

@@ -20,6 +20,8 @@
  * \brief Exec application
  *
  * \author Tilghman Lesher <app_exec__v001@the-tilghman.com>
+ *
+ * \ingroup applications
  */
 
 #include <stdio.h>

apps/app_externalivr.c

@@ -23,12 +23,13 @@
  *
  * \brief External IVR application interface
  * 
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
+#include <unistd.h>
 #include <errno.h>
 
 #include "asterisk.h"
@@ -42,6 +43,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/linkedlists.h"
+#include "asterisk/app.h"
+#include "asterisk/options.h"
 
 static const char *tdesc = "External IVR Interface Application";
 
@@ -59,7 +62,8 @@ static const char *descrip =
 "when the channel is hung up.\n"
 "See doc/README.externalivr for a protocol specification.\n";
 
-#define ast_chan_log(level, channel, format, ...) ast_log(level, "%s: " format, channel->name, ## __VA_ARGS__)
+/* 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;
@@ -91,9 +95,9 @@ static void send_child_event(FILE *handle, const char event, const char *data,
 	char tmp[256];
 
 	if (!data) {
-		snprintf(tmp, sizeof(tmp), "%c,%10ld", event, time(NULL));
+		snprintf(tmp, sizeof(tmp), "%c,%10d", event, (int)time(NULL));
 	} else {
-		snprintf(tmp, sizeof(tmp), "%c,%10ld,%s", event, time(NULL), data);
+		snprintf(tmp, sizeof(tmp), "%c,%10d,%s", event, (int)time(NULL), data);
 	}
 
 	fprintf(handle, "%s\n", tmp);
@@ -248,10 +252,9 @@ static int app_exec(struct ast_channel *chan, void *data)
 	int res = -1;
 	int gen_active = 0;
 	int pid;
-	char *command;
 	char *argv[32];
 	int argc = 1;
-	char *buf;
+	char *buf, *command;
 	FILE *child_commands = NULL;
 	FILE *child_errors = NULL;
 	FILE *child_events = NULL;
@@ -268,11 +271,13 @@ static int app_exec(struct ast_channel *chan, void *data)
 	}
 
 	buf = ast_strdupa(data);
-	command = strsep(&buf, "|");
-	memset(argv, 0, sizeof(argv) / sizeof(argv[0]));
-	argv[0] = command;
-	while ((argc < 31) && (argv[argc++] = strsep(&buf, "|")));
-	argv[argc] = NULL;
+	if (!buf) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	argc = ast_app_separate_args(buf, '|', argv, sizeof(argv) / sizeof(argv[0]));
 
 	if (pipe(child_stdin)) {
 		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child input: %s\n", strerror(errno));
@@ -309,13 +314,16 @@ static int app_exec(struct ast_channel *chan, void *data)
 		/* child process */
 		int i;
 
+		if (option_highpriority)
+			ast_set_priority(0);
+
 		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));
+		execv(argv[0], argv);
+		fprintf(stderr, "Failed to execute '%s': %s\n", argv[0], strerror(errno));
 		exit(1);
 	} else {
 		/* parent process */
@@ -341,8 +349,6 @@ static int app_exec(struct ast_channel *chan, void *data)
 			goto exit;
 		}
 
-		setvbuf(child_events, NULL, _IONBF, 0);
-
 		if (!(child_commands = fdopen(child_commands_fd, "r"))) {
 			ast_chan_log(LOG_WARNING, chan, "Could not open stream for child commands\n");
 			goto exit;
@@ -353,6 +359,10 @@ static int app_exec(struct ast_channel *chan, void *data)
 			goto exit;
 		}
 
+		setvbuf(child_events, NULL, _IONBF, 0);
+		setvbuf(child_commands, NULL, _IONBF, 0);
+		setvbuf(child_errors, NULL, _IONBF, 0);
+
 		res = 0;
 
 		while (1) {

apps/app_festival.c

@@ -20,6 +20,7 @@
  *
  * \brief Connect to festival
  * 
+ * \ingroup applications
  */
 
 #include <sys/types.h>
@@ -52,6 +53,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/config.h"
 #include "asterisk/utils.h"
 #include "asterisk/lock.h"
+#include "asterisk/options.h"
 
 #define FESTIVAL_CONFIG "festival.conf"
 
@@ -135,6 +137,9 @@ static int send_waveform_to_fd(char *waveform, int length, int fd) {
                 if (x != fd)
                         close(x);
         }
+	if (option_highpriority)
+		ast_set_priority(0);
+
 /*IAS */
 #ifdef __PPC__  
 	for( x=0; x<length; x+=2)
@@ -174,7 +179,8 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 	if (chan->_state != AST_STATE_UP)
 		ast_answer(chan);
 	ast_stopstream(chan);
-
+	ast_indicate(chan, -1);
+	
 	owriteformat = chan->writeformat;
 	res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
 	if (res < 0) {
@@ -454,8 +460,20 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
 	/* This assumes only one waveform will come back, also LP is unlikely */
 	wave = 0;
 	do {
+               int read_data;
 		for (n=0; n < 3; )
-			n += read(fd,ack+n,3-n);
+               {
+                       read_data = read(fd,ack+n,3-n);
+                       /* this avoids falling in infinite loop
+                        * in case that festival server goes down
+                        * */
+                       if ( read_data == -1 )
+                       {
+                               ast_log(LOG_WARNING,"Unable to read from cache/festival fd");
+                               return -1;
+                       }
+                       n += read_data;
+               }
 		ack[3] = '\0';
 		if (strcmp(ack,"WV\n") == 0) {         /* receive a waveform */
 			ast_log(LOG_DEBUG,"Festival WV command\n");

apps/app_flash.c

@@ -20,11 +20,13 @@
  *
  * \brief App to flash a zap trunk
  * 
+ * \ingroup applications
  */
  
+#include <stdlib.h>
+#include <stdio.h>
 #include <string.h>
 #include <errno.h>
-#include <stdlib.h>
 #include <sys/ioctl.h>
 #ifdef __linux__
 #include <linux/zaptel.h>
@@ -55,8 +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;
 

apps/app_forkcdr.c

@@ -19,12 +19,13 @@
  *
  * \brief Fork CDR application
  * 
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <pthread.h>
+#include <unistd.h>
 
 #include "asterisk.h"
 
@@ -56,26 +57,39 @@ static void ast_cdr_fork(struct ast_channel *chan)
 {
 	struct ast_cdr *cdr;
 	struct ast_cdr *newcdr;
-	if (!chan || !(cdr = chan->cdr))
-		return;
+	struct ast_flags flags = { AST_CDR_FLAG_KEEP_VARS };
+
+	cdr = chan->cdr;
+
 	while (cdr->next)
 		cdr = cdr->next;
+	
 	if (!(newcdr = ast_cdr_dup(cdr)))
 		return;
+	
 	ast_cdr_append(cdr, newcdr);
-	ast_cdr_reset(newcdr, AST_CDR_FLAG_KEEP_VARS);
+	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((char *)data, 'v'), AST_CDR_FLAG_KEEP_VARS);
+		ast_set2_flag(chan->cdr, strchr(data, 'v'), AST_CDR_FLAG_KEEP_VARS);
 	
 	ast_cdr_fork(chan);
 

apps/app_getcpeid.c

@@ -18,11 +18,13 @@
 
 /*! \file
  *
- * \brief Execute arbitrary system commands
+ * \brief Get ADSI CPE ID
  * 
+ * \ingroup applications
  */
 
 #include <stdlib.h>
+#include <stdio.h>
 #include <unistd.h>
 #include <string.h>
 
@@ -47,8 +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;
 

apps/app_groupcount.c

@@ -20,6 +20,7 @@
  *
  * \brief Group Manipulation Applications
  *
+ * \ingroup applications
  */
 
 #include <stdio.h>
@@ -139,27 +140,50 @@ static int group_check_exec(struct ast_channel *chan, void *data)
 	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 (!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 (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "CheckGroup requires an argument(max[@category])\n");
+	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;
 	}
 
-	LOCAL_USER_ADD(u);
-
-  	ast_app_group_split_group(data, limit, sizeof(limit), category, sizeof(category));
+  	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) {
-			if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
-				res = -1;
-		}
+			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
+			pbx_builtin_setvar_helper(chan, "CHECKGROUPSTATUS", "OK");
 	} else
 		ast_log(LOG_WARNING, "CheckGroup requires a positive integer argument (max)\n");
 
@@ -232,9 +256,9 @@ 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.  Always returns 0.\n"
-"This application has been deprecated, please use the function\n"
-"GroupCount.\n";
+"Stores result in GROUPCOUNT. \n"
+"Note: This application has been deprecated, please use the function\n"
+"GROUP_COUNT.\n";
 
 static char *group_set_descrip =
 "Usage: SetGroup(groupname[@category])\n"
@@ -242,20 +266,26 @@ static char *group_set_descrip =
 "Set(GROUP=group).  Always returns 0.\n";
 
 static char *group_check_descrip =
-"Usage: CheckGroup(max[@category])\n"
+"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. 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";
+"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";
+"Note: This application has been deprecated, please use the function\n"
+"GROUP_MATCH_COUNT.\n";
 
 static char show_channels_usage[] = 
 "Usage: group show channels [pattern]\n"

apps/app_hasnewvoicemail.c

@@ -24,13 +24,15 @@
  *
  * \brief HasVoicemail application
  *
+ * \ingroup applications
  */
 
-#include <sys/types.h>
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
 #include <dirent.h>
+#include <sys/types.h>
 
 #include "asterisk.h"
 
@@ -43,56 +45,146 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/module.h"
 #include "asterisk/lock.h"
 #include "asterisk/utils.h"
+#include "asterisk/app.h"
+#include "asterisk/options.h"
+#ifdef USE_ODBC_STORAGE
+#include "asterisk/res_odbc.h"
+
+static char odbc_database[80];
+static char odbc_table[80];
+#endif
 
 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[/folder][@context][|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[/folder][@context][|varname])\n"
-"  Branches to priority + 101, if there is voicemail in folder 'folder' or INBOX.\n"
-"if folder is not specified. Optionally sets <varname> to the number of messages\n" 
-"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 folder 'folder' or INBOX\n"
+"  This application sets the following channel variable upon completion:\n"
+"	HASVMSTATUS		The result of the new voicemail check returned as a text string as follows\n"
+"		<# of messages in the folder, 0 for NONE>\n";
 
 STANDARD_LOCAL_USER;
 
 LOCAL_USER_DECL;
 
-static int hasvoicemail_internal(char *context, char *box, char *folder)
+#ifdef USE_ODBC_STORAGE
+static int hasvoicemail_internal(const char *context, const char *mailbox, const char *folder)
 {
-	char vmpath[256];
-	DIR *vmdir;
-	struct dirent *vment;
-	int count=0;
+	int nummsgs = 0;
+	int res;
+	SQLHSTMT stmt;
+	char sql[256];
+	char rowdata[20];
 
-	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;
-			}
+	if (!folder)
+		folder = "INBOX";
+	/* If no mailbox, return immediately */
+	if (ast_strlen_zero(mailbox))
+		return 0;
+	if (ast_strlen_zero(context))
+		context = "default";
+
+	odbc_obj *obj;
+	obj = fetch_odbc_obj(odbc_database, 0);
+	if (obj) {
+		res = SQLAllocHandle(SQL_HANDLE_STMT, obj->con, &stmt);
+		if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
+			ast_log(LOG_WARNING, "SQL Alloc Handle failed!\n");
+			goto yuck;
 		}
-		closedir(vmdir);
+		snprintf(sql, sizeof(sql), "SELECT COUNT(*) FROM %s WHERE dir = '%s/voicemail/%s/%s/%s'", odbc_table, ast_config_AST_SPOOL_DIR, context, mailbox, folder);
+		res = SQLPrepare(stmt, sql, SQL_NTS);
+		if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {  
+			ast_log(LOG_WARNING, "SQL Prepare failed![%s]\n", sql);
+			SQLFreeHandle(SQL_HANDLE_STMT, stmt);
+			goto yuck;
+		}
+		res = odbc_smart_execute(obj, stmt);
+		if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
+			ast_log(LOG_WARNING, "SQL Execute error!\n[%s]\n\n", sql);
+			SQLFreeHandle(SQL_HANDLE_STMT, stmt);
+			goto yuck;
+		}
+		res = SQLFetch(stmt);
+		if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
+			ast_log(LOG_WARNING, "SQL Fetch error!\n[%s]\n\n", sql);
+			SQLFreeHandle(SQL_HANDLE_STMT, stmt);
+			goto yuck;
+		}
+		res = SQLGetData(stmt, 1, SQL_CHAR, rowdata, sizeof(rowdata), NULL);
+		if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
+			ast_log(LOG_WARNING, "SQL Get Data error!\n[%s]\n\n", sql);
+			SQLFreeHandle(SQL_HANDLE_STMT, stmt);
+			goto yuck;
+		}
+		nummsgs = atoi(rowdata);
+		SQLFreeHandle(SQL_HANDLE_STMT, stmt);
+	} else
+		ast_log(LOG_WARNING, "Failed to obtain database object for '%s'!\n", odbc_database);
+
+yuck:
+	return nummsgs;
+}
+
+#else
+
+static int hasvoicemail_internal(const char *context, const char *mailbox, const char *folder)
+{
+	DIR *dir;
+	struct dirent *de;
+	char fn[256];
+	int count = 0;
+
+	if (ast_strlen_zero(folder))
+		folder = "INBOX";
+	if (ast_strlen_zero(context))
+		context = "default";
+	/* If no mailbox, return immediately */
+	if (ast_strlen_zero(mailbox))
+		return 0;
+	snprintf(fn, sizeof(fn), "%s/voicemail/%s/%s/%s", ast_config_AST_SPOOL_DIR, context, mailbox, folder);
+	dir = opendir(fn);
+	if (!dir)
+		return 0;
+	while ((de = readdir(dir))) {
+		if (!strncasecmp(de->d_name, "msg", 3) && !strcasecmp(de->d_name + 8, "txt"))
+			count++;
 	}
+	closedir(dir);
 	return count;
 }
+#endif
 
 static int hasvoicemail_exec(struct ast_channel *chan, void *data)
 {
 	struct localuser *u;
-	char *temps, *input, *varname = NULL, *vmbox, *context = "default";
+	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");
@@ -100,7 +192,7 @@ static int hasvoicemail_exec(struct ast_channel *chan, void *data)
 	}
 	
 	if (!data) {
-		ast_log(LOG_WARNING, "HasVoicemail requires an argument (vm-box[/folder][@context]|varname)\n");
+		ast_log(LOG_WARNING, "HasVoicemail requires an argument (vm-box[/folder][@context][|varname[|options]])\n");
 		return -1;
 	}
 
@@ -113,18 +205,13 @@ static int hasvoicemail_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 
-	temps = input;
-	if ((temps = strsep(&input, "|"))) {
-		if (!ast_strlen_zero(input))
-			varname = input;
-		input = temps;
-	}
+	AST_STANDARD_APP_ARGS(args, input);
 
-	if ((vmbox = strsep(&input, "@")))
-		if (!ast_strlen_zero(input))
-			context = input;
+	if ((vmbox = strsep(&args.vmbox, "@")))
+		if (!ast_strlen_zero(args.vmbox))
+			context = args.vmbox;
 	if (!vmbox)
-		vmbox = input;
+		vmbox = args.vmbox;
 
 	vmfolder = strchr(vmbox, '/');
 	if (vmfolder) {
@@ -134,21 +221,31 @@ static int hasvoicemail_exec(struct ast_channel *chan, void *data)
 		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) {
-		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_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);
+		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 0;
 }
 
@@ -199,6 +296,31 @@ struct ast_custom_function acf_vmcount = {
 	.read = acf_vmcount_exec,
 };
 
+static int load_config(void)
+{
+#ifdef USE_ODBC_STORAGE
+	struct ast_config *cfg;
+	char *tmp;
+	cfg = ast_config_load("voicemail.conf");
+	if (cfg) {
+		if (! (tmp = ast_variable_retrieve(cfg, "general", "odbcstorage")))
+			tmp = "asterisk";
+		ast_copy_string(odbc_database, tmp, sizeof(odbc_database));
+
+		if (! (tmp = ast_variable_retrieve(cfg, "general", "odbctable")))
+			tmp = "voicemessages";
+		ast_copy_string(odbc_table, tmp, sizeof(odbc_table));
+		ast_config_destroy(cfg);
+	}
+#endif
+	return 0;
+}
+
+int reload(void)
+{
+	return load_config();
+}
+
 int unload_module(void)
 {
 	int res;
@@ -215,7 +337,7 @@ int unload_module(void)
 int load_module(void)
 {
 	int res;
-
+	load_config();
 	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);

apps/app_ices.c

@@ -20,6 +20,7 @@
  *
  * \brief Stream to an icecast server via ICES (see contrib/asterisk-ices.xml)
  * 
+ * \ingroup applications
  */
  
 #include <string.h>
@@ -43,6 +44,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/translate.h"
+#include "asterisk/options.h"
 
 #define ICES "/usr/bin/ices"
 #define LOCAL_ICES "/usr/local/bin/ices"
@@ -56,8 +58,7 @@ 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;
 
@@ -72,6 +73,8 @@ static int icesencode(char *filename, int fd)
 		ast_log(LOG_WARNING, "Fork failed\n");
 	if (res)
 		return res;
+	if (option_highpriority)
+		ast_set_priority(0);
 	dup2(fd, STDIN_FILENO);
 	for (x=STDERR_FILENO + 1;x<256;x++) {
 		if ((x != STDIN_FILENO) && (x != STDOUT_FILENO))

apps/app_image.c

@@ -20,10 +20,12 @@
  *
  * \brief App to transmit an image
  * 
+ * \ingroup applications
  */
  
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -37,6 +39,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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";
 
@@ -45,12 +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;
 
@@ -60,22 +67,46 @@ static int sendimage_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
 	struct localuser *u;
+	char *parse;
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(filename);
+		AST_APP_ARG(options);
+	);
 	
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "SendImage requires an argument (filename)\n");
+	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 */
-		ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		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);
 	

apps/app_intercom.c

@@ -20,8 +20,10 @@
  *
  * \brief Use /dev/dsp as an intercom.
  * 
+ * \ingroup applications
  */
  
+#include <stdio.h>
 #include <unistd.h>
 #include <errno.h>
 #include <sys/ioctl.h>
@@ -67,8 +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;
 

apps/app_ivrdemo.c

@@ -20,8 +20,10 @@
  *
  * \brief IVR Demo application
  * 
+ * \ingroup applications
  */
 
+#include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>

apps/app_lookupblacklist.c

@@ -19,11 +19,14 @@
 /*! \file
  *
  * \brief App to lookup the callerid number, and see if it is blacklisted
+ *
+ * \ingroup applications
  * 
  */
 
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -40,6 +43,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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";
 
@@ -48,14 +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;
 
@@ -67,30 +71,39 @@ lookupblacklist_exec (struct ast_channel *chan, void *data)
 	char blacklist[1];
 	struct localuser *u;
 	int bl = 0;
+	int priority_jump = 0;
 
-	LOCAL_USER_ADD (u);
-	if (chan->cid.cid_num)
-	{
-		if (!ast_db_get ("blacklist", chan->cid.cid_num, 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",chan->cid.cid_num);
 			bl = 1;
 		}
 	}
 	if (chan->cid.cid_name) {
-		if (!ast_db_get ("blacklist", chan->cid.cid_name, blacklist, sizeof (blacklist))) 
-		{
+		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)
-		ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
 
-	LOCAL_USER_REMOVE (u);
+	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;
 }
 

apps/app_lookupcidname.c

@@ -20,10 +20,12 @@
  *
  * \brief App to set callerid name from database, based on directory number
  * 
+ * \ingroup applications
  */
 
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -53,7 +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;
 

apps/app_macro.c

@@ -20,12 +20,14 @@
  *
  * \brief Dial plan macro Implementation
  * 
+ * \ingroup applications
  */
 
-#include <sys/types.h>
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
+#include <sys/types.h>
 
 #include "asterisk.h"
 
@@ -57,7 +59,6 @@ static char *descrip =
 "${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"
-"Macro returns -1 if any step in the macro returns -1, and 0 otherwise.\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";
 
@@ -103,7 +104,7 @@ static int macro_exec(struct ast_channel *chan, void *data)
 	char *offsets;
 	int offset, depth;
 	int setmacrocontext=0;
-	int autoloopflag;
+	int autoloopflag, dead = 0;
   
 	char *save_macro_exten;
 	char *save_macro_context;
@@ -215,8 +216,8 @@ static int macro_exec(struct ast_channel *chan, void *data)
 				break;
 			}
 			switch(res) {
-	        	case MACRO_EXIT_RESULT:
-                        	res = 0;
+			case MACRO_EXIT_RESULT:
+				res = 0;
 				goto out;
 			case AST_PBX_KEEPALIVE:
 				if (option_debug)
@@ -230,6 +231,7 @@ static int macro_exec(struct ast_channel *chan, void *data)
 					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);
+				dead = 1;
 				goto out;
 			}
 		}
@@ -249,37 +251,44 @@ static int macro_exec(struct ast_channel *chan, void *data)
 	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);
+	if (!dead) {
+		pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
 
-	ast_set2_flag(chan, autoloopflag, AST_FLAG_IN_AUTOLOOP);
-  	for (x=1; x<argc; x++) {
+		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]);
+			if (!dead)
+				pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
 			free(oldargs[x]);
-		} else {
+		} else if (!dead) {
 			pbx_builtin_setvar_helper(chan, varname, NULL);
 		}
   	}
 
 	/* Restore macro variables */
-	pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
+	if (!dead) {
+		pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
+		pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", save_macro_context);
+		pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
+	}
 	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) {
+
+	if (!dead && setmacrocontext) {
 		chan->macrocontext[0] = '\0';
 		chan->macroexten[0] = '\0';
 		chan->macropriority = 0;
 	}
 
-	if (!strcasecmp(chan->context, fullmacro)) {
+	if (!dead && !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));
@@ -298,7 +307,8 @@ static int macro_exec(struct ast_channel *chan, void *data)
 		}
 	}
 
-	pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
+	if (!dead)
+		pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
 	if (save_macro_offset)
 		free(save_macro_offset);
 	LOCAL_USER_REMOVE(u);
@@ -327,7 +337,7 @@ static int macroif_exec(struct ast_channel *chan, void *data)
 			*label_b = '\0';
 			label_b++;
 		}
-		if (ast_true(expr))
+		if (pbx_checkcondition(expr))
 			macro_exec(chan, label_a);
 		else if (label_b) 
 			macro_exec(chan, label_b);

apps/app_math.c

@@ -20,6 +20,7 @@
  *
  * \brief A simple math application
  * 
+ * \ingroup applications
  */
 
 #include <stdlib.h>
@@ -68,7 +69,7 @@ static char *math_descrip =
 "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.  Always returns 0.\n";
+"and behave as their C equivalents.\n";
 
 #define ADDFUNCTION 0
 #define DIVIDEFUNCTION 1

apps/app_md5.c

@@ -20,11 +20,13 @@
  * \brief MD5 checksum application
  * 
  * \todo Remove this deprecated application in 1.3dev
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
 
 #include "asterisk.h"
 
@@ -38,20 +40,25 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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. Always return 0\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>): Calculates a MD5 checksum on <string>\n"
+"  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"
-"Jumps to priority+101 if incorrect.\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;
 
@@ -100,44 +107,60 @@ static int md5check_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
 	struct localuser *u;
-	char *hash= NULL; /* Hash to compare with */
 	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;
 	}
 	
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "Syntax: MD5Check(<md5hash>,<string>) - missing argument!\n");
-		return -1;
-	}
-	
 	LOCAL_USER_ADD(u);
-	
-	memset(newhash,0, sizeof(newhash));
 
-	string = ast_strdupa(data);
-	hash = strsep(&string,"|");
-	if (ast_strlen_zero(hash)) {
-		ast_log(LOG_WARNING, "Syntax: MD5Check(<md5hash>,<string>) - missing argument!\n");
+	if (!(string = ast_strdupa(data))) {
+		ast_log(LOG_WARNING, "Memory Error!\n");
 		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
-	ast_md5_hash(newhash, string);
-	if (!strcmp(newhash, hash)) {	/* Verification ok */
+
+	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", hash, string);
+			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", hash, string);
-	if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
-		if (option_debug > 2)
-			ast_log(LOG_DEBUG, "ERROR: Can't jump to exten+101 (e%s,p%d), sorry\n", chan->exten,chan->priority+101);
+		ast_log(LOG_DEBUG, "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, "Can't jump to exten+101 (e%s,p%d), sorry\n", chan->exten,chan->priority+101);
+	}
 	LOCAL_USER_REMOVE(u);
 	return res;
 }

apps/app_milliwatt.c

@@ -20,12 +20,13 @@
  *
  * \brief Digital Milliwatt Test
  * 
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
+#include <unistd.h>
 #include <errno.h>
 
 #include "asterisk.h"
@@ -72,30 +73,29 @@ static void milliwatt_release(struct ast_channel *chan, void *data)
 static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int samples)
 {
 	struct ast_frame wf;
-	unsigned char waste[AST_FRIENDLY_OFFSET];
-	unsigned char buf[640];
+	unsigned char buf[AST_FRIENDLY_OFFSET + 640];
 	int i,*indexp = (int *) data;
 
-	if (len > sizeof(buf))
+	if (len + AST_FRIENDLY_OFFSET > sizeof(buf))
 	{
-		ast_log(LOG_WARNING,"Only doing %d bytes (%d bytes requested)\n",(int)sizeof(buf),len);
-		len = sizeof(buf);
+		ast_log(LOG_WARNING,"Only doing %d bytes (%d bytes requested)\n",(int)(sizeof(buf) - AST_FRIENDLY_OFFSET),len);
+		len = sizeof(buf) - AST_FRIENDLY_OFFSET;
 	}
-	waste[0] = 0; /* make compiler happy */
 	wf.frametype = AST_FRAME_VOICE;
 	wf.subclass = AST_FORMAT_ULAW;
 	wf.offset = AST_FRIENDLY_OFFSET;
 	wf.mallocd = 0;
-	wf.data = buf;
+	wf.data = buf + AST_FRIENDLY_OFFSET;
 	wf.datalen = len;
 	wf.samples = wf.datalen;
 	wf.src = "app_milliwatt";
 	wf.delivery.tv_sec = 0;
 	wf.delivery.tv_usec = 0;
+	wf.prev = wf.next = NULL;
 	/* create a buffer containing the digital milliwatt pattern */
 	for(i = 0; i < len; i++)
 	{
-		buf[i] = digital_milliwatt[(*indexp)++];
+		buf[AST_FRIENDLY_OFFSET + i] = digital_milliwatt[(*indexp)++];
 		*indexp &= 7;
 	}
 	if (ast_write(chan,&wf) < 0)

apps/app_mixmonitor.c

@@ -23,11 +23,13 @@
 
 /*! \file
  * \brief MixMonitor() - Record a call and mix the audio during the recording
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
 
 #include "asterisk.h"
 
@@ -36,6 +38,7 @@ 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"
@@ -85,26 +88,33 @@ struct mixmonitor {
 };
 
 enum {
-    MUXFLAG_APPEND = (1 << 1),
-    MUXFLAG_BRIDGED = (1 << 2),
-    MUXFLAG_VOLUME = (1 << 3),
-    MUXFLAG_READVOLUME = (1 << 4),
-    MUXFLAG_WRITEVOLUME = (1 << 5),
+	MUXFLAG_APPEND = (1 << 1),
+	MUXFLAG_BRIDGED = (1 << 2),
+	MUXFLAG_VOLUME = (1 << 3),
+	MUXFLAG_READVOLUME = (1 << 4),
+	MUXFLAG_WRITEVOLUME = (1 << 5),
 } mixmonitor_flags;
 
-AST_DECLARE_OPTIONS(mixmonitor_opts,{
-	['a'] = { MUXFLAG_APPEND },
-	['b'] = { MUXFLAG_BRIDGED },
-	['v'] = { MUXFLAG_READVOLUME, 1 },
-	['V'] = { MUXFLAG_WRITEVOLUME, 2 },
-	['W'] = { MUXFLAG_VOLUME, 3 },
+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....
+	/* If our status has changed to DONE, then the channel we're spying on is gone....
 	   DON'T TOUCH IT!!!  RUN AWAY!!! */
-	if (spy->status != CHANSPY_RUNNING)
+	if (spy->status == CHANSPY_DONE)
 		return;
 
 	if (!chan)
@@ -152,7 +162,7 @@ static void *mixmonitor_thread(void *obj)
 	oflags = O_CREAT|O_WRONLY;
 	oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
 		
-	if ((ext = strchr(mixmonitor->filename, '.'))) {
+	if ((ext = strrchr(mixmonitor->filename, '.'))) {
 		*(ext++) = '\0';
 	} else {
 		ext = "raw";
@@ -193,6 +203,17 @@ static void *mixmonitor_thread(void *obj)
 	if (option_verbose > 1)
 		ast_verbose(VERBOSE_PREFIX_2 "Begin MixMonitor Recording %s\n", name);
 	
+	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);
+	}
+
 	while (1) {
 		struct ast_frame *next;
 		int write;
@@ -227,17 +248,6 @@ static void *mixmonitor_thread(void *obj)
 		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)
@@ -272,7 +282,7 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename
 	int len;
 
 	len = sizeof(*mixmonitor) + strlen(filename) + 1;
-	if (post_process && !ast_strlen_zero(post_process))
+	if (!ast_strlen_zero(post_process))
 		len += strlen(post_process) + 1;
 
 	if (!(mixmonitor = calloc(1, len))) {
@@ -283,7 +293,7 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename
 	mixmonitor->chan = chan;
 	mixmonitor->filename = (char *) mixmonitor + sizeof(*mixmonitor);
 	strcpy(mixmonitor->filename, filename);
-	if (post_process && !ast_strlen_zero(post_process)) {
+	if (!ast_strlen_zero(post_process)) {
 		mixmonitor->post_process = mixmonitor->filename + strlen(filename) + 1;
 		strcpy(mixmonitor->post_process, post_process);
 	}
@@ -325,41 +335,41 @@ static int mixmonitor_exec(struct ast_channel *chan, void *data)
 	AST_STANDARD_APP_ARGS(args, parse);
 	
 	if (ast_strlen_zero(args.filename)) {
-		ast_log(LOG_WARNING, "Muxmon requires an argument (filename)\n");
+		ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
 		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 
 	if (args.options) {
-		char *opts[3] = { NULL, };
+		char *opts[OPT_ARG_ARRAY_SIZE] = { NULL, };
 
-		ast_parseoptions(mixmonitor_opts, &flags, opts, args.options);
+		ast_app_parse_options(mixmonitor_opts, &flags, opts, args.options);
 
 		if (ast_test_flag(&flags, MUXFLAG_READVOLUME)) {
-			if (!opts[0] || ast_strlen_zero(opts[0])) {
+			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[0], "%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[0]);
+			} 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 (!opts[1] || ast_strlen_zero(opts[1])) {
+			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[1], "%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[1]);
+			} 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 (!opts[2] || ast_strlen_zero(opts[2])) {
+			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[2], "%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[2]);
+			} 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);
 			}
@@ -410,7 +420,7 @@ static struct ast_cli_entry cli_mixmonitor = {
 	{ "mixmonitor", NULL, NULL },
 	mixmonitor_cli, 
 	"Execute a MixMonitor command",
-	"mixmonitor <start|stop> <chan_name> [<args>]"
+	"mixmonitor <start|stop> <chan_name> [<args>]\n"
 };
 
 

apps/app_mp3.c

@@ -20,6 +20,7 @@
  *
  * \brief Silly application to play an MP3 file -- uses mpg123
  * 
+ * \ingroup applications
  */
  
 #include <string.h>
@@ -42,6 +43,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/translate.h"
+#include "asterisk/options.h"
 
 #define LOCAL_MPG_123 "/usr/local/bin/mpg123"
 #define MPG_123 "/usr/bin/mpg123"
@@ -53,9 +55,9 @@ 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 o a URL. User can exit by pressing any key\n."
-"Returns  -1  on hangup or 0 otherwise."; 
+"  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;
 
@@ -70,6 +72,8 @@ static int mp3play(char *filename, int fd)
 		ast_log(LOG_WARNING, "Fork failed\n");
 	if (res)
 		return res;
+	if (option_highpriority)
+		ast_set_priority(0);
 	dup2(fd, STDOUT_FILENO);
 	for (x=0;x<256;x++) {
 		if (x != STDOUT_FILENO)

apps/app_nbscat.c

@@ -20,6 +20,7 @@
  *
  * \brief Silly application to play an NBScat file -- uses nbscat8k
  * 
+ * \ingroup applications
  */
  
 #include <string.h>
@@ -43,6 +44,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/translate.h"
+#include "asterisk/options.h"
 
 #define LOCAL_NBSCAT "/usr/local/bin/nbscat8k"
 #define NBSCAT "/usr/bin/nbscat8k"
@@ -59,8 +61,7 @@ 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;
 
@@ -75,6 +76,9 @@ static int NBScatplay(int fd)
 		ast_log(LOG_WARNING, "Fork failed\n");
 	if (res)
 		return res;
+	if (option_highpriority)
+		ast_set_priority(0);
+
 	dup2(fd, STDOUT_FILENO);
 	for (x=0;x<256;x++) {
 		if (x != STDOUT_FILENO)

apps/app_osplookup.c

@@ -20,8 +20,10 @@
  *
  * \brief Open Settlement Protocol Lookup
  * 
+ * \ingroup applications
  */
 
+#include <stdio.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
@@ -42,6 +44,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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";
 
@@ -62,23 +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;
 
@@ -107,11 +121,16 @@ 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;
+	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 (extension)\n");
+		ast_log(LOG_WARNING, "OSPLookup requires an argument OSPLookup(exten[|provider[|options]])\n");
 		return -1;
 	}
 
@@ -124,19 +143,15 @@ static int osplookup_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 
-	provider = strchr(temp, '|');
-	if (provider) {
-		*provider = '\0';
-		provider++;
-		opts = strchr(provider, '|');
-		if (opts) {
-			*opts = '\0';
-			opts++;
-		}
+	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", temp, provider ? provider : "<default>");
-	if ((res = ast_osp_lookup(chan, provider, temp, chan->cid.cid_num, &result)) > 0) {
+
+	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);
@@ -145,16 +160,19 @@ static int osplookup_exec(struct ast_channel *chan, void *data)
 		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, "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 */
-		ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		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);
@@ -168,28 +186,53 @@ static int ospnext_exec(struct ast_channel *chan, void *data)
 	char *temp;
 	int cause;
 	struct ast_osp_result result;
-
+	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)\n");
+		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 (!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);
-			snprintf(tmp, sizeof(tmp), "%d", result.numresults);
-			pbx_builtin_setvar_helper(chan, "_OSPRESULTS", tmp);
-		}
+	if (ast_strlen_zero(temp) || (sscanf(temp, "%d", &result.handle) != 1)) {
+		result.handle = -1;
+	}
+	temp = pbx_builtin_getvar_helper(chan, "OSPRESULTS");
+	if (ast_strlen_zero(temp) || (sscanf(temp, "%d", &result.numresults) != 1)) {
+		result.numresults = 0;
+	}
+	if ((res = ast_osp_next(&result, cause)) > 0) {
+		char tmp[80];
+		snprintf(tmp, sizeof(tmp), "%d", result.handle);
+		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, "OSPNEXTSTATUS", "SUCCESS");
 	} else {
 		if (!res) {
 			if (result.handle < 0)
@@ -198,10 +241,13 @@ static int ospnext_exec(struct ast_channel *chan, void *data)
 				ast_log(LOG_DEBUG, "No OSP handle specified\n");
 		} else
 			ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Next!\n", chan->name);
+
+		pbx_builtin_setvar_helper(chan, "OSPNEXTSTATUS", "FAILED");	
 	}
 	if (!res) {
 		/* Look for a "busy" place */
-		ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		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);
@@ -216,13 +262,32 @@ static int ospfinished_exec(struct ast_channel *chan, void *data)
 	int cause;
 	time_t start=0, duration=0;
 	struct ast_osp_result result;
-
+	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 (cause)\n");
+		ast_log(LOG_WARNING, "OSPFinish should have an argument (status[|options])\n");
 		return -1;
 	}
 
-	LOCAL_USER_ADD(u);	
+	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;
@@ -233,12 +298,13 @@ static int ospfinished_exec(struct ast_channel *chan, void *data)
 	} else
 		ast_log(LOG_WARNING, "OSPFinish called on channel '%s' with no CDR!\n", chan->name);
 	
-	cause = str2cause((char *)data);
+	cause = str2cause(args.status);
 	temp = pbx_builtin_getvar_helper(chan, "OSPHANDLE");
 	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, "OSPFINISHSTATUS", "SUCCESS");
 			res = 1;
 		}
 	} else {
@@ -247,12 +313,14 @@ 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 */
-		ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		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);

apps/app_page.c

@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (c) 2004 - 2005 Digium, Inc.  All rights reserved.
+ * Copyright (c) 2004 - 2006 Digium, Inc.  All rights reserved.
  *
  * Mark Spencer <markster@digium.com>
  *
@@ -20,6 +20,7 @@
  *
  * \brief page() - Paging application
  *
+ * \ingroup applications
  */
 
 #include <stdio.h>
@@ -39,6 +40,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/module.h"
 #include "asterisk/file.h"
 #include "asterisk/app.h"
+#include "asterisk/chanvars.h"
 
 
 static const char *tdesc = "Page Multiple Phones";
@@ -53,20 +55,21 @@ static const char *page_descrip =
 "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"
-"Always returns -1.\n";
+"        d - full duplex audio\n"
+"	 q - quiet, do not play beep to caller\n";
 
 STANDARD_LOCAL_USER;
 
 LOCAL_USER_DECL;
 
-#define PAGE_DUPLEX (1 << 0)
-#define PAGE_QUIET  (1 << 1)
+enum {
+	PAGE_DUPLEX = (1 << 0),
+	PAGE_QUIET = (1 << 1),
+} page_opt_flags;
 
-AST_DECLARE_OPTIONS(page_opts,{
-	['d'] = { PAGE_DUPLEX },
-	['q'] = { PAGE_QUIET },
+AST_APP_OPTIONS(page_opts, {
+	AST_APP_OPTION('d', PAGE_DUPLEX),
+	AST_APP_OPTION('q', PAGE_QUIET),
 });
 
 struct calloutdata {
@@ -75,13 +78,14 @@ struct calloutdata {
 	char tech[64];
 	char resource[256];
 	char meetmeopts[64];
+	struct ast_variable *variables;
 };
 
 static void *page_thread(void *data)
 {
 	struct calloutdata *cd = data;
 	ast_pbx_outgoing_app(cd->tech, AST_FORMAT_SLINEAR, cd->resource, 30000,
-		"MeetMe", cd->meetmeopts, NULL, 0, cd->cidnum, cd->cidname, NULL, NULL);
+		"MeetMe", cd->meetmeopts, NULL, 0, cd->cidnum, cd->cidname, cd->variables, NULL, NULL);
 	free(cd);
 	return NULL;
 }
@@ -89,6 +93,9 @@ static void *page_thread(void *data)
 static void launch_page(struct ast_channel *chan, const char *meetmeopts, const char *tech, const char *resource)
 {
 	struct calloutdata *cd;
+	const char *varname;
+	struct ast_variable *lastvar = NULL;
+	struct ast_var_t *varptr;
 	pthread_t t;
 	pthread_attr_t attr;
 	cd = malloc(sizeof(struct calloutdata));
@@ -99,6 +106,29 @@ static void launch_page(struct ast_channel *chan, const char *meetmeopts, const
 		ast_copy_string(cd->tech, tech, sizeof(cd->tech));
 		ast_copy_string(cd->resource, resource, sizeof(cd->resource));
 		ast_copy_string(cd->meetmeopts, meetmeopts, sizeof(cd->meetmeopts));
+
+		AST_LIST_TRAVERSE(&chan->varshead, varptr, entries) {
+			if (!(varname = ast_var_full_name(varptr)))
+				continue;
+			if (varname[0] == '_') {
+				struct ast_variable *newvar = NULL;
+
+				if (varname[1] == '_') {
+					newvar = ast_variable_new(varname, ast_var_value(varptr));
+				} else {
+					newvar = ast_variable_new(&varname[1], ast_var_value(varptr));
+				}
+
+				if (newvar) {
+					if (lastvar)
+						lastvar->next = newvar;
+					else
+						cd->variables = newvar;
+					lastvar = newvar;
+				}
+			}
+		}
+
 		pthread_attr_init(&attr);
 		pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
 		if (ast_pthread_create(&t, &attr, page_thread, cd)) {
@@ -119,6 +149,7 @@ static int page_exec(struct ast_channel *chan, void *data)
 	struct ast_app *app;
 	char *tmp;
 	int res=0;
+	char originator[AST_CHANNEL_NAME];
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
@@ -140,12 +171,21 @@ static int page_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 
+	ast_copy_string(originator, chan->name, sizeof(originator));
+	if ((tmp = strchr(originator, '-')))
+		*tmp = '\0';
+
 	tmp = strsep(&options, "|");
 	if (options)
-		ast_parseoptions(page_opts, &flags, NULL, 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, "&"))) {
+		/* don't call the originating device */
+		if (!strcasecmp(tech, originator))
+			continue;
+
 		if ((resource = strchr(tech, '/'))) {
 			*resource++ = '\0';
 			launch_page(chan, meetmeopts, tech, resource);
@@ -153,6 +193,7 @@ static int page_exec(struct ast_channel *chan, void *data)
 			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)

apps/app_parkandannounce.c

@@ -23,12 +23,14 @@
  *
  * \brief ParkAndAnnounce application for Asterisk
  * 
+ * \ingroup applications
  */
 
-#include <sys/types.h>
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
+#include <sys/types.h>
 
 #include "asterisk.h"
 
@@ -77,6 +79,7 @@ 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;
@@ -178,7 +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->cid.cid_num, chan->cid.cid_name);
+	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) {

apps/app_playback.c

@@ -20,10 +20,12 @@
  *
  * \brief Trivial application to playback a sound file
  * 
+ * \ingroup applications
  */
  
 #include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
 
 #include "asterisk.h"
 
@@ -37,6 +39,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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";
 
@@ -46,14 +50,19 @@ static char *synopsis = "Play a file";
 
 static char *descrip = 
 "  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"
+"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.  If the\n"
-"file does not exist, will jump to priority n+101 if present.\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;
 
@@ -61,14 +70,17 @@ 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 = NULL;
-	char *options = NULL;
 	int option_skip=0;
 	int option_noanswer = 0;
-	char *stringp = NULL;
 	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");
@@ -84,13 +96,16 @@ static int playback_exec(struct ast_channel *chan, void *data)
 		return -1;	
 	}
 
-	stringp = tmp;
-	strsep(&stringp, "|");
-	options = strsep(&stringp, "|");
-	if (options && !strcasecmp(options, "skip"))
-		option_skip = 1;
-	if (options && !strcasecmp(options, "noanswer"))
-		option_noanswer = 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) {
@@ -115,11 +130,17 @@ static int playback_exec(struct ast_channel *chan, void *data)
 				ast_stopstream(chan);
 			} else {
 				ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
-				ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+				if (priority_jump || option_priority_jumping)
+					ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
 				res = 0;
+				mres = 1;
 			}
 			front = back;
 		}
+		if (mres)
+			pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "FAILED");
+		else
+			pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "SUCCESS");
 	}
 	LOCAL_USER_REMOVE(u);
 	return res;

apps/app_privacy.c

@@ -20,10 +20,12 @@
  *
  * \brief Block all calls without Caller*ID, require phone # to be entered
  * 
+ * \ingroup applications
  */
 
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -52,17 +54,24 @@ 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 phone number.\n"
-  "The caller is given 3 attempts.  If after 3 attempts, they do not enter\n"
-  "at least a 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 to input a callerid.\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;
@@ -71,18 +80,24 @@ 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 minlength = 10;
-	int x;
+	int x = 0;
 	char *s;
 	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 (!ast_strlen_zero(chan->cid.cid_num)) {
@@ -97,9 +112,55 @@ privacy_exec (struct ast_channel *chan, void *data)
 				return -1;
 			}
 		}
-		/*Read in the config file*/
-		cfg = ast_config_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);
@@ -108,21 +169,6 @@ 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");
-                }
-        }
-        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");
-                }
-        }
-			
 		/*Ask for 10 digit number, give 3 attempts*/
 		for (retries = 0; retries < maxretries; retries++) {
 			if (!res)
@@ -147,16 +193,26 @@ privacy_exec (struct ast_channel *chan, void *data)
 		}
 		
 		/*Got a number, play sounds and send them on their way*/
-		if ((retries < maxretries) && res == 1 ) {
+		if ((retries < maxretries) && res >= 0 ) {
 			res = ast_streamfile(chan, "privacy-thankyou", chan->language);
 			if (!res)
 				res = ast_waitstream(chan, "");
-			ast_set_callerid (chan, phone, "Privacy Manager", NULL);
-			if (option_verbose > 2)
-				ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID to %s\n",phone);
+
+			ast_set_callerid (chan, phone, "Privacy Manager", NULL); 
+
+			/* Clear the unavailable presence bit so if it came in on PRI
+			 * the caller id will now be passed out to other channels
+			 */
+			chan->cid.cid_pres &= (AST_PRES_UNAVAILABLE ^ 0xFF);
+
+			if (option_verbose > 2) {
+				ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID to %s, callerpres to %d\n",phone,chan->cid.cid_pres);
+			}
+			pbx_builtin_setvar_helper(chan, "PRIVACYMGRSTATUS", "SUCCESS");
 		} else {
-			/* 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);
+			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_config_destroy(cfg);

apps/app_random.c

@@ -20,6 +20,7 @@
  * \brief Random application
  *
  * \author Tilghman Lesher <asterisk__app_random__200508@the-tilghman.com>
+ * \ingroup applications
  */
 
 #include <stdio.h>
@@ -81,7 +82,7 @@ static int random_exec(struct ast_channel *chan, void *data)
 	if ((!prob) || (sscanf(prob, "%d", &probint) != 1))
 		probint = 0;
 
-	if ((random() % 100) + probint > 100) {
+	if ((random() % 100) + probint >= 100) {
 		res = ast_parseable_goto(chan, s);
 		if (option_verbose > 2)
 			ast_verbose( VERBOSE_PREFIX_3 "Random branches to (%s,%s,%d)\n",

apps/app_read.c

@@ -20,10 +20,12 @@
  *
  * \brief Trivial application to read a variable
  * 
+ * \ingroup applications
  */
  
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -61,7 +63,7 @@ static char *descrip =
 "  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"
-"Returns -1 on hangup or error and 0 otherwise.\n";
+"Read should disconnect if the function fails or errors out.\n";
 
 STANDARD_LOCAL_USER;
 
@@ -103,7 +105,7 @@ static int read_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 
-	if (ast_separate_app_args(argcopy, '|', args, sizeof(args) / sizeof(args[0])) < 1) {
+	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;

apps/app_readfile.c

@@ -20,6 +20,7 @@
  *
  * \brief ReadFile application -- Reads in a File for you.
  *
+ * \ingroup applications
  */
 
 #include <stdio.h>
@@ -94,15 +95,16 @@ static int readfile_exec(struct ast_channel *chan, void *data)
 		}
 	}
 
-	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));
+	if ((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);
 	}
-	pbx_builtin_setvar_helper(chan, varname, returnvar);
-	free(returnvar);
 	LOCAL_USER_REMOVE(u);
 	return res;
 }

apps/app_realtime.c

@@ -21,11 +21,13 @@
  *
  * \brief RealTime App
  * 
+ * \ingroup applications
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
 
 #include "asterisk.h"
 

apps/app_record.c

@@ -20,10 +20,12 @@
  *
  * \brief Trivial application to record a sound file
  *
+ * \ingroup applications
  */
  
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -38,6 +40,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/translate.h"
 #include "asterisk/dsp.h"
 #include "asterisk/utils.h"
+#include "asterisk/options.h"
 
 static char *tdesc = "Trivial Record Application";
 
@@ -54,17 +57,18 @@ static char *descrip =
 "- 'maxduration' is the maximum recording duration in seconds. If missing\n"
 "or 0 there is no maximum.\n"
 "- 'options' may contain any of the following letters:\n"
-"     's' : skip recording if the line is not yet answered\n"
-"     'n' : do not answer, but record anyway if line not yet answered\n"
 "     'a' : append to existing recording rather than replacing\n"
-"     't' : use alternate '*' terminator key instead of default '#'\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"
 "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;
 
@@ -98,6 +102,8 @@ static int record_exec(struct ast_channel *chan, void *data)
 	int option_quiet = 0;
 	int rfmt = 0;
 	int flags;
+	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 (ast_strlen_zero(data)) { /* no data implies no filename or anything is present */
@@ -199,123 +205,131 @@ 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) {
-				res = ast_waitstream(chan, "");
-			} else {
-				ast_log(LOG_WARNING, "ast_streamfile failed on %s\n", chan->name);
+	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) {
+			res = ast_waitstream(chan, "");
+		} else {
+			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");
+			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;
+	}
+
+	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;
 			}
-			ast_stopstream(chan);
+			maxduration = waitres;
 		}
 		
-		/* 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) {
-			int waitres;
-
-			/* Request a video update */
-			ast_indicate(chan, AST_CONTROL_VIDUPDATE);
-
-			if (maxduration <= 0)
-				maxduration = -1;
+		f = ast_read(chan);
+		if (!f) {
+			res = -1;
+			break;
+		}
+		if (f->frametype == AST_FRAME_VOICE) {
+			res = ast_writestream(s, f);
 			
-			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;
-						}
-						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 == terminator)) {
-					ast_frfree(f);
-					break;
-				}
+			if (res) {
+				ast_log(LOG_WARNING, "Problem writing frame\n");
 				ast_frfree(f);
-			}
-			if (!f) {
-				ast_log(LOG_DEBUG, "Got hangup\n");
-				res = -1;
+				break;
 			}
 			
-			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);
+			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;
+				}
 			}
-			ast_closestream(s);
-		} else			
-			ast_log(LOG_WARNING, "Could not create file %s\n", filename);
-	} else
-		ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
+		} else if (f->frametype == AST_FRAME_VIDEO) {
+			res = ast_writestream(s, f);
+			
+			if (res) {
+				ast_log(LOG_WARNING, "Problem writing frame\n");
+				ast_frfree(f);
+				break;
+			}
+		} else 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 (silgen)
+		ast_channel_stop_silence_generator(chan, silgen);
 	
+ out:
 	if ((silence > 0) && rfmt) {
 		res = ast_set_read_format(chan, rfmt);
 		if (res)

apps/app_rpt.c

@@ -20,7 +20,7 @@
 /*
  *
  * Radio Repeater / Remote Base program 
- *  version 0.36 10/26/05
+ *  version 0.37 11/3/05
  * 
  * See http://www.zapatatelephony.org/app_rpt.html
  *
@@ -197,7 +197,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/say.h"
 #include "asterisk/localtime.h"
 
-static  char *tdesc = "Radio Repeater / Remote Base  version 0.36  10/26/2005";
+static  char *tdesc = "Radio Repeater / Remote Base  version 0.37  11/03/2005";
 
 static char *app = "Rpt";
 
@@ -4614,7 +4614,7 @@ char cmd[MAXDTMF+1] = "";
 			ast_set_write_format(myrpt->txchannel,AST_FORMAT_SLINEAR);
 			myrpt->txchannel->whentohangup = 0;
 			myrpt->txchannel->appl = "Apprpt";
-			myrpt->txchannel->data = "(Repeater Rx)";
+			myrpt->txchannel->data = "(Repeater Tx)";
 			if (option_verbose > 2)
 				ast_verbose(VERBOSE_PREFIX_3 "rpt (Tx) initiating call to %s/%s on %s\n",
 					tmpstr,tele,myrpt->txchannel->name);
@@ -5758,6 +5758,7 @@ pthread_attr_t attr;
 		if (!rpt_vars[i].rxchanname)
 		{
 			ast_log(LOG_WARNING,"Did not specify rxchanname for node %s\n",rpt_vars[i].name);
+			ast_config_destroy(cfg);
 			pthread_exit(NULL);
 		}
 		/* if is a remote, dont start one for it */
@@ -5775,6 +5776,7 @@ pthread_attr_t attr;
 		if (!rpt_vars[i].ident)
 		{
 			ast_log(LOG_WARNING,"Did not specify ident for node %s\n",rpt_vars[i].name);
+			ast_config_destroy(cfg);
 			pthread_exit(NULL);
 		}
 	        pthread_attr_init(&attr);
@@ -5821,6 +5823,7 @@ pthread_attr_t attr;
 		}
 		usleep(2000000);
 	}
+	ast_config_destroy(cfg);
 	pthread_exit(NULL);
 }
 
@@ -6022,7 +6025,7 @@ static int rpt_exec(struct ast_channel *chan, void *data)
 			return -1;
 		}
 
-		if (*b1 <= '1')
+		if (*b1 < '1')
 		{
 			ast_log(LOG_WARNING, "Node %s Invalid for connection here!!\n",b1);
 			return -1;

apps/app_sayunixtime.c

@@ -19,6 +19,7 @@
  *
  * \brief SayUnixTime application
  * 
+ * \ingroup applications
  */
 
 #include <stdio.h>
@@ -53,8 +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"
@@ -62,8 +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;
 

apps/app_senddtmf.c

@@ -20,10 +20,12 @@
  *
  * \brief App to send DTMF digits
  * 
+ * \ingroup applications
  */
  
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -47,9 +49,10 @@ static char *app = "SendDTMF";
 static char *synopsis = "Sends arbitrary DTMF digits";
 
 static char *descrip = 
-"  SendDTMF(digits[|timeout_ms]): 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, w (.5s pause)\n"
+" The application will either pass the assigned digits or terminate if it\n"
+" encounters an error.\n";
 
 STANDARD_LOCAL_USER;
 

apps/app_sendtext.c

@@ -20,10 +20,14 @@
  *
  * \brief App to transmit a text message
  * 
+ * Requires support of sending text messages from channel driver
+ *
+ * \ingroup applications
  */
  
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -38,6 +42,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/translate.h"
 #include "asterisk/image.h"
 #include "asterisk/options.h"
+#include "asterisk/app.h"
 
 static const char *tdesc = "Send Text Applications";
 
@@ -46,8 +51,7 @@ static const char *app = "SendText";
 static const char *synopsis = "Send a Text Message";
 
 static const char *descrip = 
-"  SendText(text): Sends text to current channel (callee).\n"
-"Otherwise, execution will continue at the next priority level.\n"
+"  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"
@@ -55,11 +59,9 @@ static const char *descrip =
 "      UNSUPPORTED  Text transmission not supported by channel\n"
 "\n"
 "At this moment, text is supposed to be 7 bit ASCII in most channels.\n"
-"Old deprecated behavior: \n"
-" SendText only returns 0 if the text was sent correctly or if\n"
-" the channel does not support text transport.\n"
-" If the client does not support text transport, and there exists a\n"
-" step with priority n + 101, then execution will continue at that step.\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;
 
@@ -70,26 +72,47 @@ static int sendtext_exec(struct ast_channel *chan, void *data)
 	int res = 0;
 	struct localuser *u;
 	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)\n");
+		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->tech->send_text) {
 		ast_mutex_unlock(&chan->lock);
 		/* Does not support transport */
-		if (option_priority_jumping)
+		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);

apps/app_setcallerid.c

@@ -20,10 +20,12 @@
  *
  * \brief App to set callerid
  * 
+ * \ingroup applications
  */
  
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -49,7 +51,7 @@ LOCAL_USER_DECL;
 
 static char *descrip2 = 
 "  SetCallerPres(presentation): Set Caller*ID presentation on a call.\n"
-"  Always returns 0.  Valid presentations are:\n"
+"  Valid presentations are:\n"
 "\n"
 "      allowed_not_screened    : Presentation Allowed, Not Screened\n"
 "      allowed_passed_screen   : Presentation Allowed, Passed Screen\n" 
@@ -94,7 +96,7 @@ 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)
 {

apps/app_setcdruserfield.c

@@ -20,6 +20,7 @@
  *
  * \brief Applictions connected with CDR engine
  * 
+ * \ingroup applications
  */
 
 #include <sys/types.h>
@@ -51,8 +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";
@@ -67,8 +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";

apps/app_setcidname.c

@@ -20,10 +20,12 @@
  *
  * \brief App to set callerid
  *
+ * \ingroup applications
  */
  
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -50,7 +52,7 @@ 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. Always returns 0\n"
+"party. \n"
 "SetCIDName has been deprecated in favor of the function\n"
 "CALLERID(name)\n";
 

apps/app_setcidnum.c

@@ -21,10 +21,12 @@
  *
  * \brief App to set callerid number
  * 
+ * \ingroup applications
  */
  
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -51,7 +53,7 @@ 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";
 

apps/app_setrdnis.c

@@ -21,10 +21,12 @@
  *
  * \brief App to set rdnis
  *
+ * \ingroup applications
  */
  
-#include <string.h>
 #include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
@@ -49,7 +51,7 @@ static char *synopsis = "Set RDNIS Number";
 
 static char *descrip = 
 "  SetRDNIS(cnum): Set RDNIS Number on a call to a new\n"
-"value.  Always returns 0\n"
+"value.\n"
 "SetRDNIS has been deprecated in favor of the function\n"
 "CALLERID(rdnis)\n";
 

apps/app_settransfercapability.c

@@ -20,6 +20,7 @@
  *
  * \brief App to set the ISDN Transfer Capability
  * 
+ * \ingroup applications
  */
  
 #include <string.h>
@@ -57,7 +58,7 @@ static struct {	int val; char *name; } transcaps[] = {
 static char *descrip = 
 "  SetTransferCapability(transfercapability): Set the ISDN Transfer \n"
 "Capability of a call to a new value.\n"
-"Always returns 0.  Valid Transfer Capabilities are:\n"
+"Valid Transfer Capabilities are:\n"
 "\n"
 "  SPEECH             : 0x00 - Speech (default, voice calls)\n"
 "  DIGITAL            : 0x08 - Unrestricted digital information (data calls)\n"

apps/app_skel.c

@@ -3,7 +3,7 @@
  *
  * Copyright (C) <Year>, <Your Name Here>
  *
- * <Your Name Here> <<You Email Here>>
+ * <Your Name Here> <<Your Email Here>>
  *
  * See http://www.asterisk.org for more information about
  * the Asterisk project. Please do not directly contact
@@ -20,8 +20,11 @@
  *
  * \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>
@@ -50,7 +53,7 @@ static char *descrip = "This application is a template to build other applicatio
 #define OPTION_C	(1 << 2)	/* Option C(str) */
 #define OPTION_NULL	(1 << 3)	/* Dummy Termination */
 
-AST_DECLARE_OPTIONS(app_opts,{
+AST_APP_OPTIONS(app_opts,{
 	['a'] = { OPTION_A },
 	['b'] = { OPTION_B, 1 },
 	['c'] = { OPTION_C, 2 }
@@ -74,7 +77,6 @@ static int app_exec(struct ast_channel *chan, void *data)
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "%s requires an argument (dummy|[options])\n",app);
-		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 
@@ -90,10 +92,10 @@ static int app_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 	
-	if ((argc = ast_separate_app_args(args, '|', argv, sizeof(argv) / sizeof(argv[0])))) {
+	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);
+		ast_app_parse_options(app_opts, &flags, opts, options);
 	}
 
 	if (!ast_strlen_zero(dummy)) 

apps/app_sms.c

@@ -17,17 +17,19 @@
 /*! \file
  *
  * \brief SMS application - ETSI ES 201 912 protocol 1 implimentation
+ * \ingroup applications
  * 
  */
 
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
 #include <errno.h>
-#include <sys/types.h>
-#include <sys/stat.h>
 #include <dirent.h>
 #include <ctype.h>
+#include <sys/types.h>
+#include <sys/stat.h>
 
 #include "asterisk.h"
 
@@ -66,8 +68,7 @@ static char *synopsis = "Communicates with SMS service centres and SMS capable a
 
 static char *descrip =
 	"  SMS(name|[a][s]):  SMS handles exchange of SMS data with a call to/from SMS capabale\n"
-	"phone or SMS PSTN service centre. Can send and/or receive SMS messages.\n"
-	"Returns 0 if call handled correctly, or -1 if there were any problems.\n"
+	"phone or SMS PSTN service center. Can send and/or receive SMS messages.\n"
 	"Works to ETSI ES 201 912 compatible with BT SMS PSTN service in UK\n"
 	"Typical usage is to use to handle called from the SMS service centre CLI,\n"
 	"or to set up a call using 'outgoing' or manager interface to connect\n"
@@ -693,7 +694,7 @@ static void sms_readfile (sms_t * h, char *fn)
 		}
 		while (fgets (line, sizeof (line), s))
 		{								 /* process line in file */
-			char *p;
+			unsigned char *p;
 			for (p = line; *p && *p != '\n' && *p != '\r'; p++);
 			*p = 0;					 /* strip eoln */
 			p = line;
@@ -1177,25 +1178,24 @@ static void sms_messagetx(sms_t * h)
 static int sms_generate (struct ast_channel *chan, void *data, int len, int samples)
 {
 	struct ast_frame f = { 0 };
-	unsigned char waste[AST_FRIENDLY_OFFSET];
+#define MAXSAMPLES 800
 #ifdef OUTALAW
-	unsigned char buf[800];
+	unsigned char *buf;
 #else
-	signed short buf[800];
+	short *buf;
 #endif
+#define SAMPLE2LEN sizeof(*buf)
 	sms_t *h = data;
 	int i;
 
-	if (len > sizeof (buf)) {
-		ast_log (LOG_WARNING, "Only doing %d bytes (%d bytes requested)\n", (int)(sizeof (buf) / sizeof (signed short)), len);
-		len = sizeof (buf);
-#ifdef OUTALAW
-		samples = len;
-#else
-		samples = len / 2;
-#endif
+	if (samples > MAXSAMPLES) {
+		ast_log (LOG_WARNING, "Only doing %d samples (%d requested)\n",
+			 MAXSAMPLES, samples);
+		samples = MAXSAMPLES;
 	}
-	waste[0] = 0;					 /* make compiler happy */
+	len = samples * SAMPLE2LEN + AST_FRIENDLY_OFFSET;
+	buf = alloca(len);
+
 	f.frametype = AST_FRAME_VOICE;
 #ifdef OUTALAW
 	f.subclass = AST_FORMAT_ALAW;
@@ -1205,8 +1205,7 @@ static int sms_generate (struct ast_channel *chan, void *data, int len, int samp
 	f.datalen = samples * 2;
 #endif
 	f.offset = AST_FRIENDLY_OFFSET;
-	f.mallocd = 0;
-	f.data = buf;
+	f.data = buf + AST_FRIENDLY_OFFSET;
 	f.samples = samples;
 	f.src = "app_sms";
 	/* create a buffer containing the digital sms pattern */
@@ -1378,8 +1377,8 @@ static int sms_exec (struct ast_channel *chan, void *data)
 		ast_copy_string (h.cli, chan->cid.cid_num, sizeof (h.cli));
 
 	{
-		char *d = data,
-			*p,
+		unsigned char *p;
+		unsigned char *d = data,
 			answer = 0;
 		if (!*d || *d == '|') {
 			ast_log (LOG_ERROR, "Requires queue name\n");
@@ -1448,7 +1447,7 @@ static int sms_exec (struct ast_channel *chan, void *data)
 			d = p;
 			h.udl = 0;
 			while (*p && h.udl < SMSLEN)
-				h.ud[h.udl++] = utf8decode((unsigned char **)&p);
+				h.ud[h.udl++] = utf8decode(&p);
 			if (is7bit (h.dcs) && packsms7 (0, h.udhl, h.udh, h.udl, h.ud) < 0)
 				ast_log (LOG_WARNING, "Invalid 7 bit GSM data\n");
 			if (is8bit (h.dcs) && packsms8 (0, h.udhl, h.udh, h.udl, h.ud) < 0)

apps/app_softhangup.c

@@ -20,12 +20,14 @@
  *
  * \brief SoftHangup application
  * 
+ * \ingroup applications
  */
 
-#include <sys/types.h>
 #include <stdlib.h>
-#include <unistd.h>
+#include <stdio.h>
 #include <string.h>
+#include <unistd.h>
+#include <sys/types.h>
 
 #include "asterisk.h"
 
@@ -43,7 +45,8 @@ static char *synopsis = "Soft Hangup Application";
 static char *tdesc = "Hangs up the requested channel";
 
 static char *desc = "  SoftHangup(Technology/resource|options)\n"
-"Hangs up the requested channel.  Always returns 0\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";