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/1.2.25-netsec@212958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

.cleancount

@@ -1 +1 @@
-36
+9

.lastclean

@@ -0,0 +1 @@
+9

.version

@@ -0,0 +1 @@
+1.2.25-netsec

BUGS

@@ -4,10 +4,10 @@ Asterisk Bug Tracking Information
 To learn about and report Asterisk bugs, please visit 
 the official Asterisk Bug Tracker at:
 
-	https://issues.asterisk.org
+	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.asterisk.org/developers/bug-guidelines

CHANGES

@@ -1,628 +1,127 @@
-======================================================================
-===
-=== This file documents the new and/or enhanced functionality added in
-=== the Asterisk versions listed below. This file does NOT include
-=== changes in behavior that would not be backwards compatible with
-=== previous versions; for that information see the UPGRADE.txt file
-=== and the other UPGRADE files for older releases.
-===
-======================================================================
+Changes since Asterisk 1.2.0-beta2:
 
-SIP changes
------------
- * Added a new option "prematuremedia" that defaults to "no". If you turn this
-   option on, chan_sip will not automatically initiate early media if it receives
-   audio from the incoming channel before there's been a progress indication.
+   * Cygwin build system portability
+   * Optional generation of outbound silence during channel recording
 
------------------------------------------------------------------------------------
---- Functionality changes from Asterisk 1.6.0.10 to Asterisk 1.6.0.11 -------------
------------------------------------------------------------------------------------
+Changes since Asterisk 1.2.0-beta1:
 
-SIP Changes
------------
- * Added a new 'ignoresdpversion' option to sip.conf.  When this is enabled
-   (either globally or for a specific peer), chan_sip will treat any SDP data
-   it receives as new data and update the media stream accordingly.  By
-   default, Asterisk will only modify the media stream if the SDP session
-   version received is different from the current SDP session version.  This
-   option is required to interoperate with devices that have non-standard SDP
-   session version implementations (observed with Microsoft OCS).  This option
-   is disabled by default. In addition, this behavior is automatic when the SDP received
-   is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
-   since the call will fail if Asterisk does not process the incoming SDP, Asterisk
-   will accept the SDP even if the SDP version number is not properly incremented,
-   but will generate a warning in the log indicating that the SIP peer that sent
-   the SDP should have the 'ignoresdpversion' option set.
+   * 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()
 
-------------------------------------------------------------------------------
---- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0  -------------
-------------------------------------------------------------------------------
+Changes since Asterisk 1.0:
 
-AMI - The manager (TCP/TLS/HTTP)
---------------------------------
-  * Manager has undergone a lot of changes, all of them documented
-    in doc/manager_1_1.txt
-  * Manager version has changed to 1.1
-  * Added a new action 'CoreShowChannels' to list currently defined channels
-     and some information about them. 
-  * Added a new action 'SIPshowregistry' to list SIP registrations.
-  * Added TLS support for the manager interface and HTTP server
-  * Added the URI redirect option for the built-in HTTP server
-  * The output of CallerID in Manager events is now more consistent.
-     CallerIDNum is used for number and CallerIDName for name.
-  * Enable https support for builtin web server.
-     See configs/http.conf.sample for details.
-  * Added a new action, GetConfigJSON, which can return the contents of an
-     Asterisk configuration file in JSON format.  This is intended to help
-     improve the performance of AJAX applications using the manager interface
-     over HTTP.
-  * SIP and IAX manager events now use "ChannelType" in all cases where we 
-     indicate channel driver. Previously, we used a mixture of "Channel"
-     and "ChannelDriver" headers.
-  * Added a "Bridge" action which allows you to bridge any two channels that
-     are currently active on the system.
-  * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
-     the voicemail users setup.
-  * Added 'DBDel' and 'DBDelTree' manager commands.
-  * cdr_manager now reports events via the "cdr" level, separating it from
-     the very verbose "call" level.
-  * Manager users are now stored in memory. If you change the manager account
-    list (delete or add accounts) you need to reload manager.
-  * Added Masquerade manager event for when a masquerade happens between
-     two channels.
-  * Added "manager reload" command for the CLI
-  * Lots of commands that only provided information are now allowed under the
-     Reporting privilege, instead of only under Call or System.
-  * The IAX* commands now require either System or Reporting privilege, to
-     mirror the privileges of the SIP* commands.
-  * Added ability to retrieve list of categories in a config file.
-  * Added ability to retrieve the content of a particular category.
-  * Added ability to empty a context.
-  * Created new action to create a new file.
-  * Updated delete action to allow deletion by line number with respect to category.
-  * Added new action insert to add new variable to category at specified line.
-  * Updated action newcat to allow new category to be inserted in file above another
-    existing category.
-  * Added new event "JitterBufStats" in the IAX2 channel
-  * Originate now requires the Originate privilege and, if you want to call out
-    to a subshell, it requires the System privilege, as well.  This was done to
-    enhance manager security.
+This list currently only containts changes made from the end of November until
+March 26, 2005.
 
-Dialplan functions
-------------------
-  * Added the DEVICE_STATE() dialplan function which allows retrieving any device
-     state in the dialplan, as well as creating custom device states that are
-     controllable from the dialplan.
-  * Extend CALLERID() function with "pres" and "ton" parameters to
-     fetch string representation of calling number presentation indicator
-     and numeric representation of type of calling number value.
-  * MailboxExists converted to dialplan function
-  * A new option to Dial() for telling IP phones not to count the call
-     as "missed" when dial times out and cancels.
-  * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
-     mutex.  No deadlocks are possible, as LOCK() only allows a single lock to be
-     held for any given channel.  Also, locks are automatically freed when a
-     channel is hung up.
-  * Added HINT() dialplan function that allows retrieving hint information.
-     Hints are mappings between extensions and devices for the sake of 
-     determining the state of an extension.  This function can retrieve the list
-     of devices or the name associated with a hint.
-  * Added EXTENSION_STATE() dialplan function which allows retrieving the state
-    of any extension.
-  * Added SYSINFO() dialplan function which allows retrieval of system information
-  * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
-     the existence of a dialplan target.
-  * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
-     upper and lower case, respectively.
-  * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
-     ID for the call (not the Asterisk call ID or unique ID), provided that the
-     channel driver supports this. For SIP, you get the SIP call-ID for the
-     bridged channel which you can store in the CDR with a custom field.
-  * Added the function AUDIOHOOK_INHERIT. This actually is already in Asterisk
-    1.4, but since it was added late in the release cycle, I felt it was a good
-	idea to list it here as well. See the CLI output for "core show function
-	AUDIOHOOK_INHERIT" for more details
-
-CLI Changes
------------
-  * New CLI command "core show hint" (usage: core show hint <exten>)
-  * New CLI command "core show settings"
-  * Added 'core show channels count' CLI command.
-  * Added the ability to set the core debug and verbose values on a per-file basis.
-  * Added 'queue pause member' and 'queue unpause member' CLI commands
-  * Ability to set process limits ("ulimit") without restarting Asterisk
-  * Enhanced "agi debug" to print the channel name as a prefix to the debug
-     output to make debugging on busy systems much easier.
-  * New CLI commands "dialplan set extenpatternmatching true/false"
-  * New CLI command: "core set chanvar" to set a channel variable from the CLI.
-  * Added an easy way to execute Asterisk CLI commands at startup.  Any commands
-    listed in the startup_commands section of cli.conf will get executed.
-  * Added a CLI command, "devstate change", which allows you to set custom device
-     states from the func_devstate module that provides the DEVICE_STATE() function
-     and handling of the "Custom:" devices.
-
-SIP changes
------------
- * Added a new 'faxdetect=yes|no' configuration option to sip.conf.  When this
-    option is enabled, Asterisk will watch for a CNG tone in the incoming audio
-    for a received call.  If it is detected, the channel will jump to the 
-    'fax' extension in the dialplan.
-  * Improved NAT and STUN support.
-     chan_sip now can use port numbers in bindaddr, externip and externhost
-     options, as well as contact a STUN server to detect its external address
-     for the SIP socket. See sip.conf.sample, 'NAT' section.
-  * The default SIP useragent= identifier now includes the Asterisk version
-  * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
-     If set, and the incoming request carries authentication info,
-     the username to match in the users list is taken from the Digest header
-     rather than from the From: field. This feature is considered experimental.
-  * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
-     since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
-  * The "localmask" setting was removed in version 1.2 and the reminder about it
-     being removed is now also removed.
-  * A new option "busylevel" for setting a level of calls where asterisk reports
-     a device as busy, to separate it from call-limit. This value is also added
-     to the SIP_PEER dialplan function.
-  * A new realtime family called "sipregs" is now supported to store SIP registration
-     data. If this family is defined, "sippeers" will be used for configuration and
-     "sipregs" for registrations. If it's not defined, "sippeers" will be used for
-     registration data, as before.
-  * The SIPPEER function have new options for port address, call and pickup groups
-  * Added support for T.140 realtime text in SIP/RTP
-  * The "checkmwi" option has been removed from sip.conf, as it is no longer
-     required due to the restructuring of how MWI is handled.  See the descriptions 
-     in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf 
-     for more information.
-  * Added rtpdest option to CHANNEL() dialplan function.
-  * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
-  * SIP now adds a header to the CANCEL if the call was answered by another phone
-     in the same dial command, or if the new c option in dial() is used.
-  * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
-     states it is not needed. For phones, however, that do require it the "registertrying" option
-     has been added so it can be enabled. 
-  * A new option called "callcounter" (global/peer/user level) enables call counters needed
-     for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
-     used to enable this functionality).
-  * New settings for timer T1 and timer B on a global level or per device. This makes it 
-     possible to force timeout faster on non-responsive SIP servers. These settings are
-     considered advanced, so don't use them unless you have a problem.
-  * Added a dial string option to be able to set the To: header in an INVITE to any
-     SIP uri.
-  * Added a new global and per-peer option, qualifyfreq, which allows you to configure
-     the qualify frequency.
-  * Added SIP Session Timers support (RFC 4028).  This prevents stuck SIP sessions that
-     were not properly torn down due to network or endpoint failures during an established
-     SIP session.
-  * Added experimental TCP and TLS support for SIP.  See doc/siptls.txt and 
-     configs/sip.conf.sample for more information on how it is used.
-  * Added t38pt_usertpsource option. See sip.conf.sample for details.
-
-IAX2 changes
-------------
-  * Added the trunkmaxsize configuration option to chan_iax2.
-  * Added the srvlookup option to iax.conf
-  * Added support for OSP.  The token is set and retrieved through the CHANNEL()
-     dialplan function.
-
-XMPP Google Talk/Jingle changes
--------------------------------
-  * Added the bindaddr option to gtalk.conf.
-
-Skinny changes
--------------
-  * Added skinny show device, skinny show line, and skinny show settings CLI commands.
-  * Proper codec support in chan_skinny.
-  * Added settings for IP and Ethernet QoS requests
-
-MGCP changes
-------------
-  * Added separate settings for media QoS in mgcp.conf
-
-Console Channel Driver changes
-------------------------------
-  * Added experimental support for video send & receive to chan_oss.
-    This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
-    a video source.
-
-Phone channel changes (chan_phone)
-----------------------------------
-  * Added G729 passthrough support to chan_phone for Sigma Designs boards.
-
-H.323 channel Changes
----------------------
-  * H323 remote hold notification support added (by NOTIFY message
-     and/or H.450 supplementary service)
-
-Local channel changes
----------------------
-  * The device state functionality in the Local channel driver has been updated
-     to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
-     to just UNKNOWN if the extension exists.
-  * Added jitterbuffer support for chan_local.  This allows you to use the
-     generic jitterbuffer on incoming calls going to Asterisk applications.
-     For example, this would allow you to use a jitterbuffer for an incoming
-     SIP call to Voicemail by putting a Local channel in the middle.  This
-     feature is enabled by using the 'j' option in the Dial string to the Local
-     channel in conjunction with the existing 'n' option for local channels.
-
-DAHDI channel driver (chan_dahdi) Changes
-----------------------------------------
-  * SS7 support (via libss7 library)
-  * In India, some carriers transmit CID via dtmf. Some code has been added
-     that will handle some situations. The cidstart=polarity_IN choice has been added for
-     those carriers that transmit CID via dtmf after a polarity change.
-  * CID matching information is now shown when doing 'dialplan show'.
-  * Added dahdi show version CLI command.
-  * Added setvar support to chan_dahdi.conf channel entries.
-  * Added two new options: mwimonitor and mwimonitornotify.  These options allow
-     you to enable MWI monitoring on FXO lines.  When the MWI state changes,
-     the script specified in the mwimonitornotify option is executed.  An internal
-     event indicating the new state of the mailbox is also generated, so that
-     the normal MWI facilities in Asterisk work as usual.
-  * Added signalling type 'auto', which attempts to use the same signalling type
-     for a channel as configured in DAHDI. This is primarily designed for analog
-     ports, but will also work for digital ports that are configured for FXS or FXO
-     signalling types. This mode is also the default now, so if your chan_dahdi.conf
-     does not specify signalling for a channel (which is unlikely as the sample
-     configuration file has always recommended specifying it for every channel) then
-     the 'auto' mode will be used for that channel if possible.
-  * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
-     state for a channel; also ensured that the DNDState Manager event is
-     emitted no matter how the DND state is set or cleared.
-
-New Channel Drivers
--------------------
-  * Added a new channel driver, chan_unistim.  See doc/unistim.txt and
-     configs/unistim.conf.sample for details.  This new channel driver allows
-     you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
-  * Added a new channel driver, chan_console, which uses portaudio as a cross
-     platform audio interface.  It was written as a channel driver that would
-     work with Mac CoreAudio, but portaudio supports a number of other audio
-     interfaces, as well. Note that this channel driver requires v19 or higher
-     of portaudio; older versions have a different API.
- 
-DUNDi changes
--------------
-  * Added the ability to specify arguments to the Dial application when using
-     the DUNDi switch in the dialplan.
-  * Added the ability to set weights for responses dynamically.  This can be
-     done using a global variable or a dialplan function.  Using the SHELL()
-     function would allow you to have an external script set the weight for
-     each response.
-  * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT.  These
-     functions will allow you to initiate a DUNDi query from the dialplan,
-     find out how many results there are, and access each one.
-
-ENUM changes
-------------
-  * Added two new dialplan functions, ENUMQUERY and ENUMRESULT.  These
-     functions will allow you to initiate an ENUM lookup from the dialplan,
-     and Asterisk will cache the results.  ENUMRESULT can be used to access
-     the results without doing multiple DNS queries.
-
-Voicemail Changes
------------------
-  * Added the ability to customize which sound files are used for some of the
-     prompts within the Voicemail application by changing them in voicemail.conf
-  * Added the ability for the "voicemail show users" CLI command to show users
-     configured by the dynamic realtime configuration method.
-  * MWI (Message Waiting Indication) handling has been significantly
-     restructured internally to Asterisk.  It is now totally event based
-     instead of polling based.  The voicemail application will notify other
-     modules that have subscribed to MWI events when something in the mailbox
-     changes.
-    This also means that if any other entity outside of Asterisk is changing
-     the contents of mailboxes, then the voicemail application still needs to
-     poll for changes.  Examples of situations that would require this option
-     are web interfaces to voicemail or an email client in the case of using
-     IMAP storage.  So, two new options have been added to voicemail.conf
-     to account for this: "pollmailboxes" and "pollfreq".  See the sample
-     configuration file for details.
-  * Added "tw" language support
-  * Added support for storage of greetings using an IMAP server
-  * Added ability to customize forward, reverse, stop, and pause keys for message playback
-  * SMDI is now enabled in voicemail using the smdienable option.
-  * A "lockmode" option has been added to asterisk.conf to configure the file
-     locking method used for voicemail, and potentially other things in the
-     future.  The default is the old behavior, lockfile.  However, there is a
-     new method, "flock", that uses a different method for situations where the
-     lockfile will not work, such as on SMB/CIFS mounts.
-  * Added the ability to backup deleted messages, to ease recovery in the case
-     that a user accidentally deletes a message, and discovers that they need it.
-  * Reworked the SMDI interface in Asterisk.  The new way to access SMDI information
-     is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG().  The file
-     smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
-     voicemail boxes.  The SMDI interface can also poll for MWI changes when some
-     outside entity is modifying the state of the mailbox (such as IMAP storage or
-     a web interface of some kind).
-
-Queue changes
--------------
-  * Added the general option 'shared_lastcall' so that member's wrapuptime may be
-     used across multiple queues.
-  * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and 
-     setqueueentryvar options for each queue, see queues.conf.sample for details.
-  * Added keepstats option to queues.conf which will keep queue
-     statistics during a reload.
-  * setinterfacevar option in queues.conf also now sets a variable
-     called MEMBERNAME which contains the member's name.
-  * Added 'Strategy' field to manager event QueueParams which represents
-     the queue strategy in use. 
-  * Added option to run macro when a queue member is connected to a caller, 
-     see queues.conf.sample for details.
-  * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
-     does not count paused queue members as unavailable.
-  * Added min-announce-frequency option to queues.conf which allows you to control the
-     minimum amount of time between queue announcements for use when the caller's queue
-     position changes frequently.
-  * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
-     queue log.
-  * Added ability for non-realtime queues to have realtime members
-  * Added the "linear" strategy to queues.
-  * Added the "wrandom" strategy to queues.
-  * Added new channel variable QUEUE_MIN_PENALTY
-  * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
-     rules in queuerules.conf. See configs/queuerules.conf.sample for details
-  * Added a new parameter for member definition, called state_interface. This may be
-    used so that a member may be called via one interface but have a different interface's
-    device state reported.
-
-MeetMe Changes
---------------
-  * The 'o' option to provide an optimization has been removed and its functionality 
-     has been enabled by default.
-  * When a conference is created, the UNIQUEID of the channel that caused it to be
-     created is stored.  Then, every channel that joins the conference will have the
-     MEETMEUNIQUEID channel variable set with this ID.  This can be used to relate
-     callers that come and go from long standing conferences.
-  * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
-     except it does operations on a channel by name, instead of number in a conference.
-     This is a very useful feature in combination with the 'X' option to ChanSpy.
-  * Added 'C' option to Meetme which causes a caller to continue in the dialplan
-     when kicked out.
-  * Added new RealTime functionality to provide support for scheduled conferencing.
-     This includes optional messages to the caller if they attempt to join before
-     the schedule start time, or to allow the caller to join the conference early.
-     Also included is optional support for limiting the number of callers per
-     RealTime conference.
-  * Added the S() and L() options to the MeetMe application.  These are pretty
-     much identical to the S() and L() options to Dial().  They let you set
-     timeouts for the conference, as well as have warning sounds played to
-     let the caller know how much time is left, and when it is running out.
-  * Added the ability to do "meetme concise" with the "meetme" CLI command.
-     This extends the concise capabilities of this CLI command to include
-     listing all conferences, instead of an addition to the other sub commands
-     for the "meetme" command.
-  * Added the ability to specify the music on hold class used to play into the
-     conference when there is only one member and the M option is used.
-
-Other Dialplan Application Changes
-----------------------------------
-  * Argument support for Gosub application
-  * From the to-do lists: straighten out the app timeout args:
-     Wait() app now really does 0.3 seconds- was truncating arg to an int.
-     WaitExten() same as Wait().
-     Congestion() - Now takes floating pt. argument.
-     Busy() - now takes floating pt. argument.
-     Read() - timeout now can be floating pt.
-     WaitForRing() now takes floating pt timeout arg.
-     SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
-  * Added 's' option to Page application.
-  * Added 'E', 'V', and 'P' commands to ExternalIVR.
-  * Added 'o' and 'X' options to Chanspy.
-  * Added a new dialplan application, Bridge, which allows you to bridge the
-     calling channel to any other active channel on the system.
-  * Added the ability to specify a music on hold class to play instead of ringing
-     for the SLATrunk application.
-  * The Read application no longer exits the dialplan on error.  Instead, it sets
-     READSTATUS to ERROR, which you can catch and handle separately.
-  * Added 'm' option to Directory, which lists out names, 8 at a time, instead
-     of asking for verification of each name, one at a time.
-  * Privacy() no longer uses privacy.conf, as all options are specifyable as
-     direct options to the app.
-  * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
-     for more details
-  * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
-  * The ChannelRedirect application no longer exits the dialplan if the given channel
-     does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
-     or NOCHANNEL if the given channel was not found.
-  * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
-    answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
-    from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
-    original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
-    the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
-    to obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
-
-Music On Hold Changes
----------------------
-  * A new option, "digit", has been added for music on hold classes in 
-     musiconhold.conf.  If this is set for a music on hold class, a caller
-     listening to music on hold can press this digit to switch to listening
-     to this music on hold class.
-  * Support for realtime music on hold has been added.
-  * In conjunction with the realtime music on hold, a general section has
-     been added to musiconhold.conf, its sole variable is cachertclasses. If this
-     is set, then music on hold classes found in realtime will be cached in memory.
-
-AEL Changes
------------
-  * AEL upgraded to use the Gosub with Arguments instead
-     of Macro application, to hopefully reduce the problems
-     seen with the artificially low stack ceiling that 
-     Macro bumps into. Macros can only call other Macros
-     to a depth of 7. Tests run using gosub, show depths
-     limited only by virtual memory. A small test demonstrated
-     recursive call depths of 100,000 without problems.
-     -- in addition to this, all apps that allowed a macro
-     to be called, as in Dial, queues, etc, are now allowing
-     a gosub call in similar fashion.
-  * AEL now generates LOCAL(argname) declarations when it
-     Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
-     etc. That makes the arguments local in scope. The user
-     can define their own local variables in macros, now,
-     by saying "local myvar=someval;"  or using Set() in this
-     fashion:  Set(LOCAL(myvar)=someval);  ("local" is now
-     an AEL keyword).
-  * utils/conf2ael introduced. Will convert an extensions.conf
-     file into extensions.ael. Very crude and unfinished, but 
-     will be improved as time goes by. Should be useful for a
-     first pass at conversion.
-  * aelparse will now read extensions.conf to see if a referenced
-     macro or context is there before issueing a warning.
-
-Call Features (res_features) Changes
-------------------------------------
-  * Added the parkedcalltransfers option to features.conf
-  * Added parkedcallparking option to control one touch parking w/ parking
-    pickup
-  * Added parkedcallhangup option to control disconnect feature w/ parking
-    pickup
-  * Added parkedcallrecording option to control one-touch record w/ parking
-    pickup
-  * Added BRIDGE_FEATURES variable to set available features for a channel
-  * The built-in method for doing attended transfers has been updated to
-     include some new options that allow you to have the transferee sent
-     back to the person that did the transfer if the transfer is not successful.
-     See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
-     in features.conf.sample.
-  * Added support for configuring named groups of custom call features in
-     features.conf.  This means that features can be written a single time, and
-     then mapped into groups of features for different key mappings or easier
-     access control.
-  * Updated the ParkedCall application to allow you to not specify a parking
-     extension.  If you don't specify a parking space to pick up, it will grab
-     the first one available.
-  * Added cli command 'features reload' to reload call features from features.conf
-  * Moved into core asterisk binary.
-
-Language Support Changes
-------------------------
-  * Brazilian Portuguese (pt-BR) in VM, and say.c was added
-  * Added support for the Hungarian language for saying numbers, dates, and times.
-
-AGI Changes
------------
-  * Added SPEECH commands for speech recognition. A complete listing can be found
-     using agi show.
-  * If app_stack is loaded, GOSUB is a native AGI command that may be used to
-    invoke subroutines in the dialplan.  Note that calling EXEC with Gosub
-    does not behave as expected; the native command needs to be used, instead.
-
-Logger changes
---------------
-  * Added rotatestrategy option to logger.conf, along with two new options:
-     "timestamp" which will use the time to name the logger files instead of
-     sequence number; and "rotate", which rotates the names of the logfiles,
-     similar to the way syslog rotates files.
-  * Added exec_after_rotate option to logger.conf, which allows a system
-     command to be run after rotation.  This is primarily useful with
-     rotatestrategry=rotate, to allow a limit on the number of logfiles kept
-     and to ensure that the oldest log file gets deleted.
-  * Added realtime support for the queue log
-
-Call Detail Records 
--------------------
-  * The cdr_manager module has a [mappings] feature, like cdr_custom,
-    to add fields to the manager event from the CDR variables.
-  * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
-     backend database CDR table.  Specifically, additional, non-standard
-     columns are supported, merely by setting the corresponding CDR variable in
-     your dialplan.  In addition, you may alias any column to another name (for
-     example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
-     simply "alias src => ANI" in the configuration file).  Records may be
-     posted to more than one backend, simply by specifying multiple categories
-     in the configuration file.  And finally, you may filter which CDRs get
-     posted to each backend, by specifying a filter (which the record must
-     match) for the particular category.  Filters are additive (meaning all
-     rules must match to post that CDR).
-  * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
-     module.  Specifically, you may add additional columns into the table and
-     they will be set, if you set the corresponding CDR variable name.  Also,
-     if you omit columns in your database table, they will be silently skipped
-     (but a record will still be inserted, based on what columns remain).  Note
-     that the other two features from cdr_adaptive_odbc (alias and filter) are
-     not currently supported.
-  * The ResetCDR application now has an 'e' option that re-enables a CDR if it
-     has been disabled using the NoCDR application.
-
-Miscellaneous New Modules
--------------------------
-  * Added a new CDR module, cdr_sqlite3_custom.
-  * Added a new realtime configuration module, res_config_sqlite
-  * Added a new codec translation module, codec_resample, which re-samples
-     signed linear audio between 8 kHz and 16 kHz to help support wideband
-     codecs.
-  * Added a new module, res_phoneprov, which allows auto-provisioning of phones
-     based on configuration templates that use Asterisk dialplan function and
-     variable substitution.  It should be possible to create phone profiles and
-     templates that work for the majority of phones provisioned over http. It
-     is currently only intended to provision a single user account per phone.
-     An example profile and set of templates for Polycom phones is provided.
-     NOTE: Polycom firmware is not included, but should be placed in
-     AST_DATA_DIR/phoneprov/configs to match up with the included templates.
-  * Added a new module, app_jack, which provides interfaces to JACK, the Jack
-     Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
-     provided; there is a JACK() application, and a JACK_HOOK() function.  Both
-     interfaces create an input and output JACK port.  The application makes
-     these ports the endpoint of the call.  The audio coming from the channel
-     goes out the output port and whatever comes back in on the input port is
-     what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
-     audiohook on the channel.  This lets you run the audio coming from a
-     channel through JACK, and whatever comes back in is what gets forwarded
-     on as the channel's audio.  This is very useful for building custom
-     vocoders or doing recording or analysis of the channel's audio in another
-     application.
-  * Added a new module, res_config_curl, which permits using a HTTP POST url
-     to retrieve, create, update, and delete realtime information from a remote
-     web server.  Note that this module requires func_curl.so to be loaded for
-     backend functionality.
-  * Added a new module, res_config_ldap, which permits the use of an LDAP
-     server for realtime data access.
-  * Added support for writing and running your dialplan in lua using the pbx_lua
-     module.  See configs/extensions.lua.sample for examples of how to do this.
-
-Miscellaneous 
--------------
- * res_jabber: autoprune has been disabled by default, to avoid misconfiguration 
-   that would end up being interpreted as a bug once Asterisk started removing 
-   the contacts from a user list.
-  * Ability to use libcap to set high ToS bits when non-root
-     on Linux. If configure is unable to find libcap then you
-     can use --with-cap to specify the path.
-  * Added maxfiles option to options section of asterisk.conf which allows you to specify
-     what Asterisk should set as the maximum number of open files when it loads.
-  * Added the jittertargetextra configuration option.
-  * Added support for setting the CoS for VLAN traffic (802.1p).  See the sample
-     configuration files for the IP channel drivers.  The new option is "cos".
-     This information is also documented in doc/qos.tex, or the IP Quality of Service
-     section of asterisk.pdf.
-  * When originating a call using AMI or pbx_spool that fails the reason for failure
-     will now be available in the failed extension using the REASON dialplan variable.
-  * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
-     It allows you to configure a prefix for auto-monitor recordings.
-  * A new extension pattern matching algorithm, based on a trie, is introduced
-     here, that could noticeably speed up mid-sized to large dialplans.
-     It is NOT used by default, as duplicating the behaviour of the old pattern
-     matcher is still under development. A config file option, in extensions.conf,
-     in the [general] section, called "extenpatternmatchingnew", is by default
-     set to false; setting that to true will force the use of the new algorithm.
-     Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
-     be used to switch the algorithms at run time.
-  * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
-     specifying which socket to use to connect to the running Asterisk daemon
-     (-s)
-  * Added logging to 'make update' command.  See update.log
-  * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
-     do not come from the remote party.
-  * Added the 'n' option to the SpeechBackground application to tell it to not
-     answer the channel if it has not already been answered.
-  * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
-     turned on, via the CHANNEL(trace) dialplan function.  Could be useful for
-     dialplan debugging.
-  * iLBC source code no longer included (see UPGRADE.txt for details)
-  * A new option for the configure script, --enable-internal-poll, has been added
-    for use with systems which may have a buggy implementation of the poll system
-	call. If you notice odd behavior such as the CLI being unresponsive on remote
-	consoles, you may want to try using this option. This option is enabled by default
-	on Darwin systems since it is known that the Darwin poll() implementation has
-	odd issues.
+   * 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

@@ -17,9 +17,6 @@ nic.at - ENUM support in Asterisk
 
 Paul Bagyenda, Digital Solutions - for initial Voicetronix driver development
 
-John Todd, TalkPlus, Inc.  and JR Richardson, Ntegrated Solutions. - for funding
-    the development of SIP Session Timers support.
-
 === WISHLIST CONTRIBUTERS ===
 Jeremy McNamara - SpeeX support
 Nick Seraphin - RDNIS support
@@ -30,183 +27,77 @@ Wasim - Hangup detect
 * Thanks to QuickNet Technologies for their donation of an Internet
 PhoneJack and Linejack card to the project.  (http://www.quicknet.net)
 
-* Thanks to VoipSupply for their donation of Sipura ATAs to the project for
-T.38 testing. (http://www.voipsupply.com)
-
-* Thanks to Grandstream for their donation of ATAs to the project for
-T.38 testing. (http://www.grandstream.com)
-
 === MISCELLANEOUS PATCHES ===
 Jim Dixon - Zapata Telephony and app_rpt
 	http://www.zapatatelephony.org/app_rpt.html
-
-Russell Bryant - Asterisk release manager and countless enhancements and bug
-	fixes.
-	russell(AT)digium.com
-
-Anthony Minessale II - Countless big and small fixes, and relentless forward
-	push. ChanSpy, ForkCDR, ControlPlayback, While/EndWhile, DumpChan, Dictate,
-	MacroIf, ExecIf, ExecIfTime, RetryDial, MixMonitor applications; many
-	realtime concepts and implementation pieces, including res_config_odbc;
+Russell Bryant - Asterisk 1.0 maintainer and misc. enhancements
+	russelb@clemson.edu
+Anthony Minessale II - Countless big and small fixes, and relentless forward push
+	ChanSpy, ForkCDR, ControlPlayback, While/EndWhile, DumpChan, Dictate,
+	MacroIf, ExecIf, ExecIfTime, RetryDial, MixMonitor applications; res_odbc;
+	many realtime concepts and implementation pieces, including res_config_odbc;
 	format_slin; cdr_custom; several features in Dial including L(), G() and
-	enhancements to M() and D(); several CDR enhancements including CDR
-	variables; attended transfer; one touch record; native MOH; manager
-	eventmask; command line '-t' flag to allow recording/voicemail on nfs
-	shares; #exec command and multiline comments in config files; setvar in iax
-	and sip configs.
-	anthmct(AT)yahoo.com              http://www.asterlink.com
-
-James Golovich - Innumerable contributions, including SIP TCP and TLS support.
+	enhancements to M() and D(); several CDR enhancements including CDR variables;
+	attended transfer; one touch record; native MOH; manager eventmask; command
+	line '-t' flag to allow recording/voicemail on nfs shares; #exec command and
+	multiline comments in config files; setvar in iax and sip configs.
+	anthmct@yahoo.com              http://www.asterlink.com
+James Golovich - Innumerable contributions
 	You can find him and asterisk-perl at http://asterisk.gnuinter.net
-
 Andre Bierwirth - Extension hints and status
-
+Oliver Daudey - ISDN4Linux fixes
+Pauline Middelink - ISDN4Linux patches and some general patches.
+	She can be found at http://www.polyware.nl/~middelink/En/
 Jean-Denis Girard - Various contributions from the South Pacific Islands
-	jd-girard(AT)esoft.pf             http://www.esoft.pf
-
+	jd-girard@esoft.pf             http://www.esoft.pf
 William Jordan / Vonage - MySQL enhancements to Voicemail
-	wjordan(AT)vonage.com
-
+	wjordan@vonage.com
 Jac Kersing - Various fixes
-
 Steven Critchfield - Seek and Trunc functions for playback and recording
-	critch(AT)basesys.com
-
+	critch@basesys.com
 Jefferson Noxon - app_lookupcidname, app_db, and various other contributions
-
 Klaus-Peter Junghanns - in-band DTMF on SIP and MGCP
-
 Ross Finlayson - Dynamic RTP payload support
-
 Mahmut Fettahlioglu - Audio recording, music-on-hold changes, alaw file
-	format, and various fixes. Can be contacted at mahmut(AT)oa.com.au
-
+	format, and various fixes. Can be contacted at mahmut@oa.com.au
 James Dennis - Cisco SIP compatibility patches to work with SIP service
-	providers. Can be contacted at asterisk(AT)jdennis.net
-
+	providers. Can be contacted at asterisk@jdennis.net
 Tilghman Lesher - ast_localtime(); ast_say_date_with_format(); 
-	GotoIfTime, SayUnixTime, HasNewVoicemail applications;
-	CUT, SORT, EVAL, CURL, FIELDQTY, STRFTIME, some QUEUE* functions;
-	func_odbc, cdr_adaptive_odbc, and other innumerable bug fixes.
-	tilghman(AT)digium.com            http://asterisk.drunkcoder.com/
-
+	GotoIfTime, Random, SayUnixTime, HasNewVoicemail applications;
+	CUT, SORT, EVAL, CURL, FIELDQTY, STRFTIME, QUEUEAGENT* functions;
+	and other innumerable bug fixes. http://asterisk.drunkcoder.com/
 Jayson Vantuyl - Manager protocol changes, various other bugs.
-	jvantuyl(AT)computingedge.net
-
+	jvantuyl@computingedge.net
 Thorsten Lockert - OpenBSD, FreeBSD ports, making MacOS X port run on 10.3,
-	dialplan include verification, route lookup on OpenBSD, SNMP agent
-	support (res_snmp), various other bugs. tholo(AT)sigmasoft.com
-
-Josh Roberson - chan_zap reload support, Advanced Voicemail Features, & other
-	misc. patches. - josh(AT)asteriasgi.com, http://www.asteriasgi.com
-
-William Waites - syslog support, SIP NAT traversal for SIP-UA. ww(AT)styx.org
-
+	dialplan include verification, route lookup on OpenBSD, various other
+	bugs. tholo@sigmasoft.com
+Brian West - ODBC support and Bug Marshaling  
+Josh Roberson - chan_zap reload support, Advanced Voicemail Features, other misc. patches,
+	and Bug Marshalling. - josh@asteriasgi.com, http://www.asteriasgi.com
+William Waites - syslog support, SIP NAT traversal for SIP-UA. ww@styx.org
 Rich Murphey - Porting to FreeBSD, NetBSD, OpenBSD, and Darwin.
-	rich(AT)whiteoaklabs.com  http://whiteoaklabs.com
-
+	rich@whiteoaklabs.com  http://whiteoaklabs.com
 Simon Lockhart - Porting to Solaris (based on work of Logan ???)	
-	simon(AT)slimey.org
-
-Olle E. Johansson - SIP RFC compliance, documentation and testing, testing,
-	SIP outbound proxy support, Manager 1.1 update, SIP transfer support,
-	SIP presence support, SIP call state updates (dialog-info), 
-	MiniVM - the small voicemail system, many documentation
-	updates/corrections, and many bug fixes.
-	oej(AT)edvina.net, http://edvina.net
-
+	simon@slimey.org
+Olle E. Johansson - SIP RFC compliance, documentation and testing, testing, testing
+	oej@edvina.net, http://edvina.net
 Steve Kann - new jitter buffer for IAX2
-	stevek(AT)stevek.com
-
+	stevek@stevek.com
 Constantine Filin - major contributions to the Asterisk Realtime Architecture
-
-Steve Murphy - privacy support, $[ ] parser upgrade, AEL2 parser upgrade.
-	murf(AT)digium.com
-
+Steve Murphy - privacy support
 Claude Patry - bug fixes, feature enhancements, and bug marshalling
-	cpatry(AT)gmail.com
-
-Miroslav Nachev, miro(AT)space-comm.com COSMOS Software Enterprises, Ltd.
-	- for Variable for No Answer Timeout for Attended Transfer
-
-Slav Klenov & Vanheuverzwijn Joachim - development of the generic jitterbuffer
-	Securax Ltd. info(AT)securax.be
-
-Roy Sigurd Karlsbakk - providing funding for generic jitterbuffer development
-	roy(AT)karlsbakk.net, Briiz Telecom AS
-
-Voop AS, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - funding for rewrite
-	of SIP transfers
-
-Philippe Sultan - RADIUS CDR module, many fixes to res_jabber and gtalk/jingle
-	channel drivers.
-	INRIA, http://www.inria.fr/
-
-John Martin, Aupix - Improved video support in the SIP channel
-	T.140 text support in RTP/SIP
-
-Steve Underwood - Provided T.38 pass through support.
-
-George Konstantoulakis - Support for Greek in voicemail added by InAccess
-	Networks (work funded by HOL, www.hol.gr) gkon(AT)inaccessnetworks.com
-
-Daniel Nylander - Support for Swedish and Norwegian languages in voicemail.
-	http://www.danielnylander.se/
-
-Stojan Sljivic - An option for maximum number of messsages per mailbox in
-	voicemail.  Also an issue with voicemail synchronization has been fixed.
-	GDS Partners www.gdspartners.com .  stojan.sljivic(AT)gdspartners.com
-
-Bartosz Supczinski - Support for Polish added by DIR (www.dir.pl)
-	Bartosz.Supczinski(AT)dir.pl
-
-James Rothenberger - Support for IMAP storage integration added by
-	OneBizTone LLC Work funded by University of Pennsylvania jar(AT)onebiztone.com
-
-Paul Cadach - Bringing chan_h323 up to date, bug fixes, and more!
-
-Voop AS - Financial support for a lot of work with the SIP driver and the IAX
-	trunk MTU patch
-
-Cedric Hans - Development of chan_unistim
-  cedric.hans(AT)mlkj.net
-
-Sergio Fadda - console_video: video support for chan_oss and chan_alsa
-
-Marta Carbone - console_video and the astobj2 framework
-
-Luigi Rizzo - astobj2, console_video, windows build, chan_oss cleanup,
-	and a bunch of infrastructure work (loader, new_cli, ...)
-
-Brett Bryant - digit option for musiconhold selection, ENUMQUERY and ENUMRESULT functions,
-	feature group configuration for features.conf, per-file CLI debug and verbose settings,
-	TCP and TLS support for SIP, and various bug fixes.
-	brettbryant(AT)gmail.com
-
-Sergey Tamkovich - Realtime support for MusicOnHold, store and destroy realtime methods and
-	implementations for odbc, sqlite, and pgsql realtime drivers, attended transfer updates,
-	multiple speeds for ControlPlayback, and multiple bug fixes
-	- See http://voip-info.org/users/view/sergee
-	serg(AT)voipsolutions.ru
+	cpatry@gmail.com
 
 === OTHER CONTRIBUTIONS ===
 John Todd - Monkey sounds and associated teletorture prompt
 Michael Jerris - bug marshaling
 Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
 	available under a Creative Commons License at http://www.asteriskdocs.org
-Brian M. Clapper - poll.c emulation
-	This product includes software developed by Brian M. Clapper <bmc(AT)clapper.org>
 
 === HOLD MUSIC ===
 Music provided by www.opsound.org
 
 === OTHER SOURCE CODE IN ASTERISK ===
-Asterisk uses libedit, the lightweight readline replacement from NetBSD.
-The cdr_radius module uses libradiusclient-ng, which is also from NetBSD.
-They are BSD-licensed and require the following statement:
-
-      This product includes software developed by the NetBSD
-      Foundation, Inc. and its contributors.
 
 Digium did not implement the codecs in Asterisk.  Here is the copyright on the
 GSM source:

HARDWARE

@@ -0,0 +1,70 @@
+A PBX is only really useful if you can get calls into it.  Of course, you
+can use Asterisk with VoIP calls (SIP, H.323, IAX), but you can also talk
+to the real PSTN through various cards.
+
+Supported Hardware is divided into two general groups:  Zaptel devices and 
+non-zaptel devices.  The Zaptel compatible hardware supports pseudo-TDM 
+conferencing and all call features through chan_zap, whereas non-zaptel 
+compatible hardware may have different features.
+
+Zaptel compatible hardware
+==========================
+
+-- Digium (Primary author of Asterisk)
+        http://www.digium.com, http://store.yahoo.com/asteriskpbx
+
+   * Wildcard X100P - Single FXO interface connects to Loopstart phone 
+     line
+
+   * Wildcard T400P (obsolete) - Quad T1 interface connects to four T1/PRI 
+     interfaces.  Supports RBS and PRI voice and PPP, FR, and HDLC data.
+
+   * Wildcard E400P (obsolete)- Quad E1 interface connects to four E1/PRI 
+     (or PRA) interfaces.  Supports PRA/PRI, EuroISDN voice and data.
+
+   * Wildcard T100P - Single T1 interface connects to a single T1/PRI
+     interface.  Supports RBS and PRI voice and PPP, FR, and HDLC data.
+
+   * Wildcard E100P - Single E1 interface connects to a single E1/PRI (or PRA)
+     interface.  Supports PRA/PRI, EuroISDN voice and PPP, FR, HDLC data.
+
+   * Wildcard S100U - Single FXS interface connects to a standard analog
+     telephone.
+
+   * Wildcard TDM400P - Quad Modular FXS interface connects to standard
+     analog telephones.
+
+   * Wildcard TE410P - Quad T1/E1 switchable interface.  Supports PRI and 
+     RBS signalling, as well as PPP, FR, and HDLC data modes.
+
+Non-zaptel compatible hardware
+==============================
+
+-- QuickNet, Inc. 
+       http://www.quicknet.net
+
+   * Internet PhoneJack - Single FXS interface.  Supports Linux telephony
+     interface.  DSP compression built-in.
+
+   * Internet LineJack - Single FXS or FXO interface.  Supports Linux 
+     telephony interface.
+
+
+Miscellaneous other interfaces
+==============================
+
+-- ISDN4Linux
+       http://www.isdn4linux.de/
+
+   * Any ISDN terminal adapter supported by isdn4linux should provide
+     connectivity.
+
+-- ALSA
+       http://www.alsa-project.org
+
+   * Any ALSA compatible full-duplex sound card
+
+-- OSS
+       http://www.opensound.com
+
+   * Any OSS compatible full-duplex sound card

LICENSE

@@ -7,8 +7,15 @@ source tree in the file COPYING.
 
 This package also includes various components that are not part of
 Asterisk itself; these components are in the 'contrib' directory
-and its subdirectories. These components are also distributed under the
-GPL version 2 as well.
+and its subdirectories. Most of these components are also
+distributed under the GPL version 2 as well, except for the following:
+
+contrib/firmware/iax/iaxy.bin:
+	This file is Copyright (C) Digium, Inc. and is licensed for
+	use with Digium IAXy hardware devices only. It can be
+	distributed freely as long as the distribution is in the
+	original form present in this package (not reformatted or
+	modified).
 
 Digium, Inc. (formerly Linux Support Services) holds copyright
 and/or sufficient licenses to all components of the Asterisk
@@ -24,8 +31,8 @@ there is no requirement that you provide the same exception in your
 GPL'd products (although if you've written a module for Asterisk we
 would strongly encourage you to make the same exception that we do).
 
-Specific permission is also granted to link Asterisk with OpenSSL, OpenH323
-and/or the UW IMAP Toolkit and distribute the resulting binary files.
+Specific permission is also granted to link Asterisk with OpenSSL and
+OpenH323 and distribute the resulting binary files.
 
 In addition, Asterisk implements two management/control protocols: the
 Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface
@@ -43,20 +50,19 @@ and use of them is subject to our trademark licensing policies. If you
 wish to use these trademarks for purposes other than simple
 redistribution of Asterisk source code obtained from Digium, you
 should contact our licensing department to determine the necessary
-steps you must take. For more information on this policy, please read:
-
-http://www.digium.com/en/company/profile/trademarkpolicy.php
+steps you must take.
 
 If you have any questions regarding our licensing policy, please
 contact us:
 
-+1.877.344.4861 (via telephone in the USA)
++1.877.546.8963 (via telephone in the USA)
 +1.256.428.6000 (via telephone outside the USA)
 +1.256.864.0464 (via FAX inside or outside the USA)
-IAX2/pbx.digium.com (via IAX2)
+IAX2/misery.digium.com/6000 (via IAX2)
 licensing@digium.com (via email)
 
 Digium, Inc.
-445 Jan Davis Drive
+150 West Park Loop
+Suite 100
 Huntsville, AL  35806
 USA

Makefile.moddir_rules

@@ -1,160 +0,0 @@
-#
-# Asterisk -- A telephony toolkit for Linux.
-# 
-# Makefile rules for subdirectories containing modules
-#
-# Copyright (C) 2006, Digium, Inc.
-#
-# Kevin P. Fleming <kpfleming@digium.com>
-#
-# This program is free software, distributed under the terms of
-# the GNU General Public License
-#
-
-# Makefile rules for building modules.
-
-# In most cases, we set target-specific variables for certain targets
-# (remember that they apply recursively to prerequisites).
-# Also note that we can only set one variable per rule, so we have to
-# repeat the left hand side to set multiple variables.
-
-ifeq ($(findstring LOADABLE_MODULES,$(MENUSELECT_CFLAGS)),)
-  _ASTCFLAGS+=${GC_CFLAGS}
-endif
-
-ifneq ($(findstring STATIC_BUILD,$(MENUSELECT_CFLAGS)),)
-  STATIC_BUILD=-static
-endif
-
-include $(ASTTOPDIR)/Makefile.rules
-
-# If MODULE_PREFIX is defined, use it to run the standard functions to set
-# C_MODS, CC_MODS, LOADABLE_MODS and EMBEDDED_MODS.
-# Each word of MODULE_PREFIX is a prefix for filenames that we consider
-# valid C or CC modules (eg. app, func ...). Note that the underscore
-# is added here, and does not need to be in MODULE_PREFIX
-#
-# Use MODULE_EXCLUDE to specify additional modules to exclude.
-
-ifneq ($(MODULE_PREFIX),)
-    ALL_C_MODS:=
-    ALL_C_MODS+=$(foreach p,$(MODULE_PREFIX),$(patsubst %.c,%,$(wildcard $(p)_*.c)))
-    ALL_CC_MODS:=
-    ALL_CC_MODS+=$(foreach p,$(MODULE_PREFIX),$(patsubst %.cc,%,$(wildcard $(p)_*.cc)))
- 
-    C_MODS:=$(filter-out $(MENUSELECT_$(MENUSELECT_CATEGORY)),$(ALL_C_MODS))
-    CC_MODS:=$(filter-out $(MENUSELECT_$(MENUSELECT_CATEGORY)),$(ALL_CC_MODS))
-
-    # and store in the list of embedded or loadable modules
-    ifneq ($(findstring $(MENUSELECT_CATEGORY),$(MENUSELECT_EMBED)),)
-	EMBEDDED_MODS:=$(C_MODS) $(CC_MODS)
-    else
-	LOADABLE_MODS:=$(C_MODS) $(CC_MODS)
-    endif
-endif
-
-# Both C++ and C++ sources need their module name in AST_MODULE
-# We also pass whatever _INCLUDE list is generated by menuselect
-# (they are stored in file 'makeopts')
-
-$(addsuffix .oo,$(CC_MODS)) $(addsuffix .o,$(C_MODS)):	\
-	_ASTCFLAGS+= -DAST_MODULE=\"$*\" $(MENUSELECT_OPTS_$*:%=-D%) $(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_INCLUDE))
-
-ifeq ($(findstring $(OSARCH), mingw32 cygwin ),)
-   # don't define -fPIC on mingw32 and cygwin, it is the default
-   $(LOADABLE_MODS:%=%.so): _ASTCFLAGS+=-fPIC
-endif
-
-# For loadable modules, pass _LIB and _LDFLAGS from menuselect.
-$(LOADABLE_MODS:%=%.so): LIBS+=$(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_LIB))
-$(LOADABLE_MODS:%=%.so): _ASTLDFLAGS+=$(foreach dep,$(MENUSELECT_DEPENDS_$*),$(value $(dep)_LDFLAGS))
-
-$(EMBEDDED_MODS:%=%.o): _ASTCFLAGS+=-DEMBEDDED_MODULE=$*
-
-$(addsuffix .so,$(filter $(LOADABLE_MODS),$(C_MODS))): %.so: %.o
-$(addsuffix .so,$(filter $(LOADABLE_MODS),$(CC_MODS))): %.so: %.oo
-
-modules.link: $(addsuffix .eo,$(filter $(EMBEDDED_MODS),$(C_MODS)))
-
-.PHONY: clean uninstall _all moduleinfo makeopts
-
-ifneq ($(LOADABLE_MODS),)
-_all: $(LOADABLE_MODS:%=%.so)
-ifneq ($(findstring $(OSARCH), mingw32 cygwin ),)
-  # linker options and extra libraries for cygwin
-  SOLINK=-Wl,--out-implib=lib$@.a -shared
-  LIBS+=-L$(ASTTOPDIR)/main -lasterisk -L$(ASTTOPDIR)/res $($@_LIBS)
-  # additional libraries in res/
-endif
-endif
-
-ifneq ($(EMBEDDED_MODS),)
-_all: modules.link
-__embed_ldscript:
-	@echo "../$(SUBDIR)/modules.link"
-__embed_ldflags:
-	@echo "$(foreach mod,$(filter $(EMBEDDED_MODS),$(C_MODS)),$(foreach dep,$(MENUSELECT_DEPENDS_$(mod)),$(dep)_LDFLAGS))"
-	@echo "$(foreach mod,$(filter $(EMBEDDED_MODS),$(CC_MODS)),$(foreach dep,$(MENUSELECT_DEPENDS_$(mod)),$(dep)_LDFLAGS))"
-__embed_libs:
-	@echo "$(foreach mod,$(filter $(EMBEDDED_MODS),$(C_MODS)),$(foreach dep,$(MENUSELECT_DEPENDS_$(mod)),$(dep)_LIB))"
-	@echo "$(foreach mod,$(filter $(EMBEDDED_MODS),$(CC_MODS)),$(foreach dep,$(MENUSELECT_DEPENDS_$(mod)),$(dep)_LIB))"
-else
-__embed_ldscript:
-__embed_ldflags:
-__embed_libs:
-endif
-
-modules.link:
-	@rm -f $@
-	@for file in $(patsubst %,$(SUBDIR)/%,$(filter %.eo,$^)); do echo "INPUT (../$${file})" >> $@; done
-	@for file in $(patsubst %,$(SUBDIR)/%,$(filter-out %.eo,$^)); do echo "INPUT (../$${file})" >> $@; done
-
-clean::
-	rm -f *.so *.o *.oo *.eo *.i *.ii
-	rm -f .*.d
-	rm -f *.s *.i
-	rm -f modules.link
-
-install:: all
-	@echo "Installing modules from `basename $(CURDIR)`..."
-	@for x in $(LOADABLE_MODS:%=%.so); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done
-
-uninstall::
-
-dist-clean::
-	rm -f .*.moduleinfo .moduleinfo
-	rm -f .*.makeopts .makeopts
-
-.%.moduleinfo: %.c
-	@echo "<member name=\"$*\" displayname=\"$(shell $(GREP) -e AST_MODULE_INFO $< | head -n 1 | cut -d '"' -f 2)\" remove_on_change=\"$(SUBDIR)/$*.o $(SUBDIR)/$*.so\">" > $@
-	$(AWK) -f $(ASTTOPDIR)/build_tools/get_moduleinfo $< >> $@
-	echo "</member>" >> $@
-
-.%.moduleinfo: %.cc
-	@echo "<member name=\"$*\" displayname=\"$(shell $(GREP) -e AST_MODULE_INFO $< | head -n 1 | cut -d '"' -f 2)\" remove_on_change=\"$(SUBDIR)/$*.oo $(SUBDIR)/$*.so\">" > $@
-	$(AWK) -f $(ASTTOPDIR)/build_tools/get_moduleinfo $< >> $@
-	echo "</member>" >> $@
-
-.moduleinfo:: $(addsuffix .moduleinfo,$(addprefix .,$(ALL_C_MODS) $(ALL_CC_MODS)))
-	@echo "<category name=\"MENUSELECT_$(MENUSELECT_CATEGORY)\" displayname=\"$(MENUSELECT_DESCRIPTION)\" remove_on_change=\"$(SUBDIR)/modules.link\">" > $@
-	@cat $^ >> $@
-	@echo "</category>" >> $@
-
-moduleinfo: .moduleinfo
-	@cat $<
-
-.%.makeopts: %.c
-	@$(AWK) -f $(ASTTOPDIR)/build_tools/get_makeopts $< > $@
-
-.%.makeopts: %.cc
-	@$(AWK) -f $(ASTTOPDIR)/build_tools/get_makeopts $< > $@
-
-.makeopts:: $(addsuffix .makeopts,$(addprefix .,$(ALL_C_MODS) $(ALL_CC_MODS)))
-	@cat $^ > $@
-
-makeopts: .makeopts
-	@cat $<
-
-ifneq ($(wildcard .*.d),)
-   include .*.d
-endif

Makefile.rules

@@ -1,135 +0,0 @@
-#
-# Asterisk -- A telephony toolkit for Linux.
-# 
-# Makefile rules
-#
-# Copyright (C) 2006-2008, Digium, Inc.
-#
-# Kevin P. Fleming <kpfleming@digium.com>
-#
-# This program is free software, distributed under the terms of
-# the GNU General Public License
-#
-
-# Each command is preceded by a short comment on what to do.
-# Prefixing one or the other with @\# or @ or nothing makes the desired
-# behaviour. ECHO_PREFIX prefixes the comment, CMD_PREFIX prefixes the command.
-
--include $(ASTTOPDIR)/makeopts
-
-.PHONY: dist-clean
-
-# If 'make' decides to create intermediate files to satisfy a build requirement
-# (like producing a .i from a .c), we want to keep them, so tell make to keep
-# all intermediate files
-.SECONDARY:
-
-# extra cflags to build dependencies. Recursively expanded.
-MAKE_DEPS=-MD -MT $@ -MF .$(subst /,_,$@).d -MP
-
-ifeq ($(NOISY_BUILD),)
-    ECHO_PREFIX=@
-    CMD_PREFIX=@
-else
-    ECHO_PREFIX=@\# 
-    CMD_PREFIX=
-endif
-
-OPTIMIZE?=-O6
-ifneq ($(findstring darwin,$(OSARCH)),)
-  ifeq ($(shell /usr/bin/sw_vers -productVersion | cut -c1-4),10.6)
-    # Snow Leopard has an issue with this optimization flag on large files (like chan_sip)
-    OPTIMIZE+=-fno-inline-functions
-  endif
-endif
-
-ifeq ($(findstring DONT_OPTIMIZE,$(MENUSELECT_CFLAGS)),)
-    _ASTCFLAGS+=$(OPTIMIZE)
-else
-    _ASTCFLAGS+=-O0
-endif
-
-# shortcuts for common combinations of flags; these must be recursively expanded so that
-# per-target settings will be applied
-CC_CFLAGS=$(PTHREAD_CFLAGS) $(_ASTCFLAGS) $(ASTCFLAGS)
-CXX_CFLAGS=$(PTHREAD_CFLAGS) $(filter-out -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(AST_DECLARATION_AFTER_STATEMENT),$(_ASTCFLAGS) $(ASTCFLAGS))
-CC_LDFLAGS_SO=$(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $(SOLINK) $(ASTLDFLAGS)
-CXX_LDFLAGS_SO=$(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $(SOLINK) $(ASTLDFLAGS)
-CC_LIBS=$(PTHREAD_LIBS) $(LIBS)
-CXX_LIBS=$(PTHREAD_LIBS) $(LIBS)
-
-# determine whether to double-compile so that the optimizer can report code path problems
-# this is only done when developer mode and DONT_OPTIMIZE are both enabled
-# in that case, we run the preprocessor to produce a .i or .ii file from the source
-# code, then compile once with optimizer enabled (and the output to /dev/null),
-# and if that doesn't fail then compile again with optimizer disabled
-ifeq ($(findstring DONT_OPTIMIZE,$(MENUSELECT_CFLAGS))$(AST_DEVMODE),DONT_OPTIMIZEyes)
-COMPILE_DOUBLE=yes
-endif
-
-%.o: %.s
-	$(ECHO_PREFIX) echo "   [AS] $< -> $@"
-ifeq ($(COMPILE_DOUBLE),yes)
-	$(CMD_PREFIX) $(CC) -o /dev/null -c $< $(OPTIMIZE) $(CC_CFLAGS)
-endif
-	$(CMD_PREFIX) $(CC) -o $@ -c $< $(CC_CFLAGS)
-
-%.o: %.i
-	$(ECHO_PREFIX) echo "   [CCi] $< -> $@"
-ifeq ($(COMPILE_DOUBLE),yes)
-	$(CMD_PREFIX) $(CC) -o /dev/null -c $< $(OPTIMIZE) $(CC_CFLAGS)
-endif
-	$(CMD_PREFIX) $(CC) -o $@ -c $< $(CC_CFLAGS)
-
-ifneq ($(COMPILE_DOUBLE),yes)
-%.o: %.c
-	$(ECHO_PREFIX) echo "   [CC] $< -> $@"
-	$(CMD_PREFIX) $(CC) -o $@ -c $< $(MAKE_DEPS) $(CC_CFLAGS)
-endif
-
-%.i: %.c
-	$(ECHO_PREFIX) echo "   [CPP] $< -> $@"
-	$(CMD_PREFIX) $(CC) -o $@ -E $< $(MAKE_DEPS) $(CC_CFLAGS)
-
-%.oo: %.ii
-	$(ECHO_PREFIX) echo "   [CXXi] $< -> $@"
-ifeq ($(COMPILE_DOUBLE),yes)
-	$(CMD_PREFIX) $(CXX) -o /dev/null -c $< $(OPTIMIZE) $(CXX_CFLAGS)
-endif
-	$(CMD_PREFIX) $(CXX) -o $@ -c $< $(CXX_CFLAGS)
-
-ifneq ($(COMPILE_DOUBLE),yes)
-%.oo: %.cc
-	$(ECHO_PREFIX) echo "   [CXX] $< -> $@"
-	$(CMD_PREFIX) $(CXX) -o $@ -c $< $(MAKE_DEPS) $(CXX_CFLAGS)
-endif
-
-%.ii: %.cc
-	$(ECHO_PREFIX) echo "   [CPP] $< -> $@"
-	$(CMD_PREFIX) $(CXX) -o $@ -E $< $(MAKE_DEPS) $(CXX_CFLAGS)
-
-%.so: %.o
-	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
-	$(CMD_PREFIX) $(CC) $(STATIC_BUILD) -o $@ $(CC_LDFLAGS_SO) $^ $(CC_LIBS)
-
-%.so: %.oo
-	$(ECHO_PREFIX) echo "   [LDXX] $^ -> $@"
-	$(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(CXX_LDFLAGS_SO) $^ $(CXX_LIBS)
-
-%.eo: %.o
-	$(ECHO_PREFIX) echo "   [EMBED] $< -> $@"
-	$(CMD_PREFIX) $(ASTTOPDIR)/build_tools/make_linker_eo_script $* > .$@.ld
-	$(CMD_PREFIX) $(LD) -r -T .$@.ld -o $@ $<
-	$(CMD_PREFIX) rm -f .$@.ld
-
-%.eo: %.oo
-	$(ECHO_PREFIX) echo "   [EMBED] $< -> $@"
-	$(CMD_PREFIX) $(ASTTOPDIR)/build_tools/make_linker_eo_script $* > .$@.ld
-	$(CMD_PREFIX) $(LD) -r -T .$@.ld -o $@ $<
-	$(CMD_PREFIX) rm -f .$@.ld
-
-%: %.o
-	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
-	$(CMD_PREFIX) $(CXX) $(STATIC_BUILD) -o $@ $(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $^ $(CXX_LIBS) $(ASTLDFLAGS)
-
-dist-clean:: clean

README

@@ -1,81 +1,63 @@
-===============================================================================
-===                     The Asterisk(R) Open Source PBX
-===
-===                   by Mark Spencer <markster@digium.com>
-===                  and the Asterisk.org developer community
-===
-===                    Copyright (C) 2001-2008 Digium, Inc.
-===                       and other copyright holders.
-===============================================================================
+The Asterisk Open Source PBX
+by Mark Spencer <markster@digium.com>
+and the Asterisk.org developer community
 
--------------------------------------------------------------------------------
---- SECURITY ------------------------------------------------------------------
+Copyright (C) 2001-2005 Digium, Inc.
+and other copyright holders.
+================================================================
 
+* SECURITY
   It is imperative that you read and fully understand the contents of
-the security information document before you attempt to configure and run
-an Asterisk server.
-
-  If you downloaded Asterisk as a tarball, see the security section in the PDF
-version of the documentation in doc/tex/asterisk.pdf.  Alternatively, pull up
-the HTML version of the documentation in doc/tex/asterisk/index.html.  The
-source for the security document is available in doc/tex/security.tex.
--------------------------------------------------------------------------------
-
--------------------------------------------------------------------------------
---- WHAT IS ASTERISK ? --------------------------------------------------------
+the SECURITY file before you attempt to configure and run an Asterisk
+server.
 
+* WHAT IS ASTERISK ?
   Asterisk is an Open Source PBX and telephony toolkit.  It is, in a
 sense, middleware between Internet and telephony channels on the bottom,
-and Internet and telephony applications at the top.  However, Asterisk supports
-more telephony interfaces than just Internet telephony.  Asterisk also has a
-vast amount of support for traditional PSTN telephony, as well.  For more
-information on the project itself, please visit the Asterisk home page at:
+and Internet and telephony applications at the top.  For more information
+on the project itself, please visit the Asterisk home page at:
 
            http://www.asterisk.org
 
-  In addition you'll find lots of information compiled by the Asterisk
+In addition you'll find lots of information compiled by the Asterisk
 community on this Wiki:
 
            http://www.voip-info.org/wiki-Asterisk
 
-  There is a book on Asterisk published by O'Reilly under the Creative Commons
-License. It is available in book stores as well as in a downloadable version on
-the http://www.asteriskdocs.org web site.
--------------------------------------------------------------------------------
+There is a book on Asterisk published by O'Reilly under the
+Creative Commons License. It is available in book stores as well
+as in a downloadable version on the http://www.asteriskdocs.org
+web site.
 
--------------------------------------------------------------------------------
---- SUPPORTED OPERATING SYSTEMS -----------------------------------------------
+* SUPPORTED OPERATING SYSTEMS
 
---- Linux
+== Linux ==
   The Asterisk Open Source PBX is developed and tested primarily on the
 GNU/Linux operating system, and is supported on every major GNU/Linux
 distribution.
 
---- Others
+== Others ==
   Asterisk has also been 'ported' and reportedly runs properly on other
-operating systems as well, including Sun Solaris, Apple's Mac OS X, Cygwin,
-and the BSD variants.
--------------------------------------------------------------------------------
+operating systems as well, including Sun Solaris, Apple's Mac OS X, and
+the BSD variants.
 
--------------------------------------------------------------------------------
---- GETTING STARTED -----------------------------------------------------------
+* GETTING STARTED
 
   First, be sure you've got supported hardware (but note that you don't need
-ANY special hardware, not even a sound card) to install and run Asterisk.
+ANY special hardware, not even a soundcard) to install and run Asterisk.
 
   Supported telephony hardware includes:
 
-	* All Analog and Digital Interface cards from Digium (www.digium.com)
+	* All Wildcard (tm) products from Digium (www.digium.com)
 	* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
-	* any full duplex sound card supported by ALSA, OSS, or PortAudio
-	* any ISDN card supported by mISDN on Linux (BRI)
-	* The Xorcom AstriBank channel bank
-	* VoiceTronix OpenLine products
+	* any full duplex sound card supported by ALSA or OSS
+        * VoiceTronix OpenLine products
 
--------------------------------------------------------------------------------
+The are several drivers for ISDN BRI cards available from third party sources.
+Check the voip-info.org wiki for more information on chan_capi, chan_misdn and 
+zaphfc.
 
--------------------------------------------------------------------------------
---- UPGRADING FROM AN EARLIER VERSION -----------------------------------------
+* UPGRADING FROM VERSION 1.0
 
   If you are updating from a previous version of Asterisk, make sure you
 read the UPGRADE.txt file in the source directory. There are some files
@@ -83,59 +65,44 @@ and configuration options that you will have to change, even though we
 made every effort possible to maintain backwards compatibility.
 
   In order to discover new features to use, please check the configuration
-examples in the /configs directory of the source code distribution.  For a
-list of new features in this version of Asterisk, see the CHANGES file.
--------------------------------------------------------------------------------
+examples in the /configs directory of the source code distribution. 
+To discover the major new features of Asterisk 1.2, please visit 
+http://edvina.net/asterisk1-2/
 
--------------------------------------------------------------------------------
---- NEW INSTALLATIONS ---------------------------------------------------------
+* NEW INSTALLATIONS
 
   Ensure that your system contains a compatible compiler and development
 libraries.  Asterisk requires either the GNU Compiler Collection (GCC) version
 3.0 or higher, or a compiler that supports the C99 specification and some of
 the gcc language extensions.  In addition, your system needs to have the C
-library headers available, and the headers and libraries for ncurses.
+library headers available, and the headers and libraries for OpenSSL,
+ncurses and zlib.
+On many distributions, these files are installed by packages with names like
+'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
 
-  There are many modules that have additional dependencies.  To see what
-libraries are being looked for, see ./configure --help, or run
-"make menuselect" to view the dependencies for specific modules.
+  So let's proceed:
 
-  On many distributions, these dependencies are installed by packages with names
-like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' 
-or similar.
-
-  So, let's proceed:
-
-1) Read this README file.
-
-  There are more documents than this one in the doc/ directory.  You may also
-want to check the configuration files that contain examples and reference
-guides. They are all in the configs/ directory.
-
-2) Run "./configure"
-
-  Execute the configure script to guess values for system-dependent
-variables used during compilation.
-
-3) Run "make menuselect" [optional]
-
-  This is needed if you want to select the modules that will be compiled and to
-check dependencies for various optional modules.
-
-4) Run "make"
+1) Run "make"
 
   Assuming the build completes successfully:
 
-5) Run "make install"
+2) Run "make install"
+
+  Each time you update or checkout from CVS, you are strongly encouraged 
+to ensure all previous object files are removed to avoid internal 
+inconsistency in Asterisk. Normally, this is automatically done with 
+the presence of the file .cleancount, which increments each time a 'make clean'
+is required, and the file .lastclean, which contains the last .cleancount used. 
 
   If this is your first time working with Asterisk, you may wish to install
 the sample PBX, with demonstration extensions, etc.  If so, run:
 
-6) "make samples"
+3) "make samples"
 
-  Doing so will overwrite any existing configuration files you have installed.
+  Doing so will overwrite any existing config files you have.
 
-  Finally, you can launch Asterisk in the foreground mode (not a daemon) with:
+  Finally, you can launch Asterisk in the foreground mode (not a daemon)
+with:
 
 # asterisk -vvvc
 
@@ -148,22 +115,20 @@ like this:
 
   You can type "help" at any time to get help with the system.  For help
 with a specific command, type "help <command>".  To start the PBX using
-your sound card, you can type "console dial" to dial the PBX.  Then you can use
-"console answer", "console hangup", and "console dial" to simulate the actions
-of a telephone.  Remember that if you don't have a full duplex sound card
-(and Asterisk will tell you somewhere in its verbose messages if you do/don't)
-then it won't work right (not yet).
+your sound card, you can type "dial" to dial the PBX.  Then you can use
+"answer", "hangup", and "dial" to simulate the actions of a telephone.
+Remember that if you don't have a full duplex sound card (and Asterisk
+will tell you somewhere in its verbose messages if you do/don't) then it
+won't work right (not yet).
 
   "man asterisk" at the Unix/Linux command prompt will give you detailed
 information on how to start and stop Asterisk, as well as all the command
 line options for starting Asterisk.
 
-  Feel free to look over the configuration files in /etc/asterisk, where you
-will find a lot of information about what you can do with Asterisk.
--------------------------------------------------------------------------------
+  Feel free to look over the configuration files in /etc/asterisk, where
+you'll find a lot of information about what you can do with Asterisk.
 
--------------------------------------------------------------------------------
---- ABOUT CONFIGURATION FILES -------------------------------------------------
+* ABOUT CONFIGURATION FILES
 
   All Asterisk configuration files share a common format.  Comments are
 delimited by ';' (since '#' of course, being a DTMF digit, may occur in
@@ -175,11 +140,11 @@ they're used only to help make the configuration file easier to
 understand, and do not affect how it is actually parsed.
 
   Entries of the form 'variable=value' set the value of some parameter in
-asterisk.  For example, in dahdi.conf, one might specify:
+asterisk.  For example, in zapata.conf, one might specify:
 
 	switchtype=national
 
-  In order to indicate to Asterisk that the switch they are connecting to is
+in order to indicate to Asterisk that the switch they are connecting to is
 of the type "national".  In general, the parameter will apply to
 instantiations which occur below its specification.  For example, if the
 configuration file read:
@@ -190,7 +155,7 @@ configuration file read:
 	switchtype = dms100
 	channel => 25-47
 
-  The "national" switchtype would be applied to channels one through
+the "national" switchtype would be applied to channels one through
 four and channels 10 through 12, whereas the "dms100" switchtype would
 apply to channels 25 through 47.
   
@@ -198,10 +163,16 @@ apply to channels 25 through 47.
 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.
--------------------------------------------------------------------------------
 
--------------------------------------------------------------------------------
---- SPECIAL NOTE ON TIME ------------------------------------------------------
+* 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
 large jumps in time.  Manually changing the system time using date(1)
@@ -224,10 +195,8 @@ on UTC.  UTC does not use daylight savings time.
 
   Also note that this issue is separate from the clocking of TDM
 channels, and is known to at least affect SIP registrations.
--------------------------------------------------------------------------------
 
--------------------------------------------------------------------------------
---- FILE DESCRIPTORS ----------------------------------------------------------
+* FILE DESCRIPTORS
 
   Depending on the size of your system and your configuration,
 Asterisk can consume a large number of file descriptors.  In UNIX,
@@ -240,13 +209,11 @@ everything from configuration information to voicemail storage.
   Most systems limit the number of file descriptors that Asterisk can
 have open at one time.  This can limit the number of simultaneous
 calls that your system can handle.  For example, if the limit is set
-at 1024 (a common default value) Asterisk can handle approximately 150
+at 1024 (a common default value) Asterisk can handle approxiately 150
 SIP calls simultaneously.  To change the number of file descriptors
 follow the instructions for your system below:
--------------------------------------------------------------------------------
 
--------------------------------------------------------------------------------
---- PAM-based Linux System ----------------------------------------------------
+== PAM-based Linux System ==
 
   If your system uses PAM (Pluggable Authentication Modules) edit
 /etc/security/limits.conf.  Add these lines to the bottom of the file:
@@ -264,29 +231,18 @@ these changes to take effect.
   If there are no instructions specifically adapted to your system
 above you can try adding the command "ulimit -n 8192" to the script
 that starts Asterisk.
--------------------------------------------------------------------------------
 
--------------------------------------------------------------------------------
---- MORE INFORMATION ----------------------------------------------------------
+* MORE INFORMATION
 
   See the doc directory for more documentation on various features. Again,
 please read all the configuration samples that include documentation on
 the configuration options.
 
-  If this release of Asterisk was downloaded from a tarball, then some
-additional documentation should have been included.
-     * doc/tex/asterisk.pdf --- PDF version of the documentation
-	 * doc/tex/asterisk/index.html --- HTML version of the documentation
-
   Finally, you may wish to visit the web site and join the mailing list if
 you're interested in getting more information.
 
    http://www.asterisk.org/support
 
   Welcome to the growing worldwide community of Asterisk users!
--------------------------------------------------------------------------------
 
---- Mark Spencer, and the Asterisk.org development community
-
--------------------------------------------------------------------------------
-Asterisk is a trademark of Digium, Inc.
+Mark Spencer

README-SERIOUSLY.bestpractices.txt

@@ -1,295 +0,0 @@
-==================
-| Best Practices |
-==================
-
-The purpose of this document is to define best practices when working with
-Asterisk in order to minimize possible security breaches and to provide tried
-examples in field deployments. This is a living document and is subject to 
-change over time as best practices are defined.
-
---------
-Sections
---------
-
-* Filtering Data: 
-        How to protect yourself from redial attacks
-
-* Proper Device Naming: 
-        Why to not use numbered extensions for devices
-
-* Secure Passwords: 
-        Secure passwords limit your risk to brute force attacks
-
-* Reducing Pattern Match Typos: 
-        Using the 'same' prefix, or using Goto()
-
-----------------
-Additional Links
-----------------
-
-Additional links that contain useful information about best practices or
-security are listed below.
-
-* Seven Steps to Better SIP Security:
-        http://blogs.digium.com/2009/03/28/sip-security/
-
-* Asterisk VoIP Security (webinar):
-        http://www.asterisk.org/security/webinar/
-
-
-==============
-Filtering Data
-==============
-
-In the Asterisk dialplan, several channel variables contain data potentially 
-supplied by outside sources. This could lead to a potential security concern 
-where those outside sources may send cleverly crafted strings of data which 
-could be utilized, e.g. to place calls to unexpected locations.
-
-An example of this can be found in the use of pattern matching and the ${EXTEN}
-channel variable. Note that ${EXTEN} is not the only system created channel
-variable, so it is important to be aware of where the data you're using is
-coming from.
-
-For example, this common dialplan takes 2 or more characters of data, starting 
-with a number 0-9, and then accepts any additional information supplied by the
-request.
-
-[NOTE: We use SIP in this example, but is not limited to SIP only; protocols
-       such as Jabber/XMPP or IAX2 are also susceptible to the same sort of
-       injection problem.]
-       
-
-[incoming]
-exten => _X.,1,Verbose(2,Incoming call to extension ${EXTEN})
-exten => _X.,n,Dial(SIP/${EXTEN})
-exten => _X.,n,Hangup()
-
-This dialplan may be utilized to accept calls to extensions, which then dial a
-numbered device name configured in one of the channel configuration files (such
-as sip.conf, iax.conf, etc...) (see the section Proper Device Naming for more
-information on why this approach is flawed).
-
-The example we've given above looks harmless enough until you take into
-consideration that several channel technologies accept characters that could
-be utilized in a clever attack. For example, instead of just sending a request
-to dial extension 500 (which in our example above would create the string
-SIP/500 and is then used by the Dial() application to place a call), someone
-could potentially send a string like "500&SIP/itsp/14165551212".
-
-The string "500&SIP/itsp/14165551212" would then be contained within the 
-${EXTEN} channel variable, which is then utilized by the Dial() application in
-our example, thereby giving you the dialplan line of:
-
-exten => _X.,n,Dial(SIP/500&SIP/itsp/14165551212)
-
-Our example above has now provided someone with a method to place calls out of
-your ITSP in a place where you didn't expect to allow it. There are a couple of
-ways in which you can mitigate this impact: stricter pattern matching, or using
-the FILTER() dialplan function.
-
-Strict Pattern Matching
------------------------
-
-The simple way to mitigate this problem is with a strict pattern match that does
-not utilize the period (.) or bang (!) characters to match on one-or-more 
-characters or zero-or-more characters (respectively). To fine tune our example
-to only accept three digit extensions, we could change our pattern match to
-be:
-
-exten => _XXX,n,Dial(SIP/${EXTEN})
-
-In this way, we have minimized our impact because we're not allowing anything
-other than the numbers zero through nine. But in some cases we really do need to
-handle variable pattern matches, such as when dialing international numbers
-or when we want to handle something like a SIP URI. In this case, we'll need to
-utilize the FILTER() dialplan function.
-
-Using FILTER()
---------------
-
-The FILTER() dialplan function is used to filter strings by only allowing
-characters that you have specified. This is a perfect candidate for controlling
-which characters you want to pass to the Dial() application, or any other
-application which will contain dynamic information passed to Asterisk from an
-external source. Lets take a look at how we can use FILTER() to control what
-data we allow.
-
-Using our previous example to accept any string length of 2 or more characters, 
-starting with a number of zero through nine, we can use FILTER() to limit what 
-we will accept to just numbers. Our example would then change to something like:
-
-[incoming]
-exten => _X.,1,Verbose(2,Incoming call to extension ${EXTEN})
-exten => _X.,n,Dial(SIP/${FILTER(0-9,${EXTEN})})
-exten => _X.,n,Hangup()
-
-Note how we've wrapped the ${EXTEN} channel variable with the FILTER() function
-which will then only pass back characters that fit into the numerical range that
-we've defined.
-
-Alternatively, if we didn't want to utilize the FILTER() function within the
-Dial() application directly, we could save the value to a channel variable,
-which has a side effect of being usable in other locations of your dialplan if
-necessary, and to handle error checking in a separate location.
-
-[incoming]
-exten => _X.,1,Verbose(2,Incoming call to extension ${EXTEN})
-exten => _X.,n,Set(SAFE_EXTEN=${FILTER(0-9,${EXTEN})})
-exten => _X.,n,Dial(SIP/${SAFE_EXTEN})
-exten => _X.,n,Hangup()
-
-Now we can use the ${SAFE_EXTEN} channel variable anywhere throughout the rest
-of our dialplan, knowing we've already filtered it. We could also perform an
-error check to verify that what we've received in ${EXTEN} also matches the data
-passed back by FILTER(), and to fail the call if things do not match.
-
-[incoming]
-exten => _X.,1,Verbose(2,Incoming call to extension ${EXTEN})
-exten => _X.,n,Set(SAFE_EXTEN=${FILTER(0-9,${EXTEN})})
-exten => _X.,n,GotoIf($[${EXTEN} != ${SAFE_EXTEN}]?error,1)
-exten => _X.,n,Dial(SIP/${SAFE_EXTEN})
-exten => _X.,n,Hangup()
-
-exten => error,1,Verbose(2,Values of EXTEN and SAFE_EXTEN did not match.)
-exten => error,n,Verbose(2,EXTEN: "${EXTEN}" -- SAFE_EXTEN: "${SAFE_EXTEN}")
-exten => error,n,Playback(silence/1&invalid)
-exten => error,n,Hangup()
-
-Another example would be using FILTER() to control the characters we accept when
-we're expecting to get a SIP URI for dialing.
-
-[incoming]
-exten => _[0-9a-zA-Z].,1,Verbose(2,Incoming call to extension ${EXTEN})
-exten => _[0-9a-zA-Z].,n,Dial(SIP/${FILTER(.@0-9a-zA-Z,${EXTEN})
-exten => _[0-9a-zA-Z].,n,Hangup()
-
-Of course the FILTER() function doesn't check the formatting of the incoming
-request. There is also the REGEX() dialplan function which can be used to
-determine if the string passed to it matches the regular expression you've
-created, and to take proper action on whether it matches or not. The creation of
-regular expressions is left as an exercise for the reader.
-
-More information about the FILTER() and REGEX() dialplan functions can be found
-by typing "core show function FILTER" and "core show function REGEX" from your
-Asterisk console.
-
-
-====================
-Proper Device Naming
-====================
-
-In Asterisk, the concept of an extension number being tied to a specific device
-does not exist. Asterisk is aware of devices it can call or receive calls from,
-and how you define in your dialplan how to reach those devices is up to you.
-
-Because it has become common practice to think of a specific device as having an
-extension number associated with it, it only becomes natural to think about
-naming your devices the same as the extension number you're providing it. But
-by doing this, you're limiting the powerful concept of separating user from
-extensions, and extensions from devices.
-
-It can also be a security hazard to name your devices with a number, as this can
-open you up to brute force attacks. Many of the current exploits deal with
-device configurations which utilize a number, and even worse, a password that
-matches the devices name. For example, take a look at this poorly created device
-in sip.conf:
-
-[1000]
-type=friend
-context=international_dialing
-secret=1000
-
-As implied by the context, we've permitted a device named 1000 with a password
-of 1000 to place calls internationally. If your PBX system is accessible via
-the internet, then your system will be vulnerable to expensive international
-calls. Even if your system is not accessible via the internet, people within
-your organization could get access to dialing rules you'd prefer to reserve only
-for certain people.
-
-A more secure example for the device would be to use something like the MAC
-address of the device, along with a strong password (see the section Secure
-Passwords). The following example would be more secure:
-
-[0004f2040001]
-type=friend
-context=international_dialing
-secret=aE3%B8*$jk^G
-
-Then in your dialplan, you would reference the device via the MAC address of the
-device (or if using the softphone, a MAC address of a network interface on the
-computer).
-
-Also note that you should NOT use this password, as it will likely be one of the
-first ones added to the dictionary for brute force attacks.
-
-
-================
-Secure Passwords
-================
-
-Secure passwords are necessary in many (if not all) environments, and Asterisk 
-is certainly no exception, especially when it comes to expensive long distance
-calls that could potentially cost your company hundreds or thousands of dollars
-on an expensive monthly phone bill, with little to no recourse to fight the
-charges.
-
-Whenever you are positioned to add a password to your system, whether that is
-for a device configuration, a database connection, or any other secure 
-connection, be sure to use a secure password. A good example of a secure
-password would be something like:
-
-aE3%B8*$jk^G
-
-Our password also contains 12 characters with a mixture of upper and
-lower case characters, numbers, and symbols. Because these passwords are likely 
-to only be entered once, or loaded via a configuration file, there is
-no need to create simple passwords, even in testing. Some of the holes found in
-production systems used for exploitations involve finding the one test extension
-that contains a weak password that was forgotten prior to putting a system into
-production.
-
-Using a web search you can find several online password generators such as
-http://www.strongpasswordgenerator.com or there are several scripts that can be
-used to generate a strong password.
-
-
-============================
-Reducing Pattern Match Typos
-============================
-
-As of Asterisk 1.6.2, a new method for reducing the number of complex pattern
-matches you need to enter, which can reduce typos in your dialplan, has been
-implemented. Traditionally, a dialplan with a complex pattern match would look
-something like:
-
-exten => _[3-5]XXX,1,Verbose(Incoming call to ${EXTEN})
-exten => _[3-5]XXX,n,Set(DEVICE=${DB(device/mac_address/${EXTEN})})
-exten => _[3-5]XXX,n,Set(TECHNOLOGY=${DB(device/technology/${EXTEN})})
-exten => _[3-5]XXX,n,GotoIf($[${ISNULL(${TECHNOLOGY})} | ${ISNULL(${DEVICE})}]?error,1)
-exten => _[3-5]XXX,n,Dial(${TECHNOLOGY}/${DEVICE},${GLOBAL(TIMEOUT)})
-exten => _[3-5]XXX,n,Set(vmFlag=${IF($[${DIALSTATUS} = BUSY]?b:u)})
-exten => _[3-5]XXX,n,Voicemail(${EXTEN}@${GLOBAL(VOICEMAIL_CONTEXT)},${vmFlag})
-exten => _[3-5]XXX,n,Hangup()
-
-exten => error,1,Verbose(2,Unable to lookup technology or device for extension)
-exten => error,n,Playback(silence/1&num-not-in-db)
-exten => error,n,Hangup()
-
-Of course there exists the possibility for a typo when retyping the pattern
-match _[3-5]XXX which will match on extensions 3000 through 5999. We can
-minimize this error by utilizing the same => prefix on all lines beyond the
-first one. Our same dialplan with using same => would look like the following:
-
-exten => _[3-5]XXX,1,Verbose(Incoming call to ${EXTEN})
-same => n,Set(DEVICE=${DB(device/mac_address/${EXTEN})})
-same => n,Set(TECHNOLOGY=${DB(device/technology/${EXTEN})})
-same => n,GotoIf($[${ISNULL(${TECHNOLOGY})} | ${ISNULL(${DEVICE})}]?error,1)
-same => n,Dial(${TECHNOLOGY}/${DEVICE},${GLOBAL(TIMEOUT)})
-same => n,Set(vmFlag=${IF($[${DIALSTATUS} = BUSY]?b:u)})
-same => n,Voicemail(${EXTEN}@${GLOBAL(VOICEMAIL_CONTEXT)},${vmFlag})
-same => n,Hangup()
-
-exten => error,1,Verbose(2,Unable to lookup technology or device for extension)
-same => n,Playback(silence/1&num-not-in-db)
-same => n,Hangup()

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

SECURITY

@@ -1,15 +1,15 @@
-\subsection{Introduction}
+==== Security Notes with Asterisk ====
 
-PLEASE READ THE FOLLOWING IMPORTANT SECURITY RELATED INFORMATION.
+PLEASE READ THE FOLLOWING IMPORTANT SECURITY RELATED INFORMATION.  
 IMPROPER CONFIGURATION OF ASTERISK COULD ALLOW UNAUTHORIZED USE OF YOUR
 FACILITIES, POTENTIALLY INCURRING SUBSTANTIAL CHARGES.
 
 Asterisk security involves both network security (encryption, authentication)
-as well as dialplan security (authorization - who can access services in
+as well as dialplan security (authorization - who can access services in 
 your pbx). If you are setting up Asterisk in production use, please make
 sure you understand the issues involved.
 
-\subsection{Network Security}
+* NETWORK SECURITY
 
 If you install Asterisk and use the "make samples" command to install
 a demonstration configuration, Asterisk will open a few ports for accepting
@@ -28,7 +28,7 @@ The IAX2 protocol supports strong RSA key authentication as well as
 AES encryption of voice and signalling. The SIP channel does not
 support encryption in this version of Asterisk.
 
-\subsection{Dialplan Security}
+* DIALPLAN SECURITY
 
 First and foremost remember this:
 
@@ -37,7 +37,7 @@ INCOMING CONNECTIONS.
 
 You should consider that if any channel, incoming line, etc can enter an
 extension context that it has the capability of accessing any extension
-within that context.
+within that context. 
 
 Therefore, you should NOT allow access to outgoing or toll services in
 contexts that are accessible (especially without a password) from incoming
@@ -46,35 +46,28 @@ stations within you network.  In particular, never ever put outgoing toll
 services in the "default" context.  To make things easier, you can include
 the "default" context within other private contexts by using:
 
-\begin{astlisting}
-\begin{verbatim}
 	include => default
-\end{verbatim}
-\end{astlisting}
 
 in the appropriate section.  A well designed PBX might look like this:
 
-\begin{astlisting}
-\begin{verbatim}
 [longdistance]
-exten => _91NXXNXXXXXX,1,Dial(DAHDI/g2/${EXTEN:1})
+exten => _91NXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1})
 include => local
 
 [local]
-exten => _9NXXNXXX,1,Dial(DAHDI/g2/${EXTEN:1})
+exten => _9NXXNXXX,1,Dial(Zap/g2/${EXTEN:1})
 include => default
 
 [default]
-exten => 6123,Dial(DAHDI/1)
-\end{verbatim}
-\end{astlisting}
+exten => 6123,Dial(Zap/1)
+
 
 DON'T FORGET TO TAKE THE DEMO CONTEXT OUT OF YOUR DEFAULT CONTEXT.  There
-isn't really a security reason, it just will keep people from wanting to
+isn't really a security reason, it just will keep people from wanting to 
 play with your Asterisk setup remotely.
 
-\subsection{Log Security}
+* LOG SECURITY
 
 Please note that the Asterisk log files, as well as information printed to the
-Asterisk CLI, may contain sensitive information such as passwords and call
+Asterisk CLI, may contain sensitive information such as passwords and call 
 history.  Keep this in mind when providing access to these resources.

UPGRADE-1.2.txt

@@ -1,218 +0,0 @@
-=========================================================
-===
-=== Information for upgrading from Asterisk 1.0 to 1.2
-===
-=== This file documents all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also includes advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=========================================================
-
-Compiling:
-
-* The Asterisk 1.2 source code now uses C language features
-  supported only by 'modern' C compilers.  Generally, this means GCC
-  version 3.0 or higher, although some GCC 2.96 releases will also
-  work.  Some non-GCC compilers that support C99 and the common GCC
-  extensions (including anonymous structures and unions) will also
-  work.  All releases of GCC 2.95 do _not_ have the requisite feature
-  support; systems using that compiler will need to be upgraded to
-  a more recent compiler release.
-
-Dialplan Expressions:
-
-* The dialplan expression parser (which handles $[ ... ] constructs)
-  has gone through a major upgrade, but has one incompatible change:
-  spaces are no longer required around expression operators, including
-  string comparisons. However, you can now use quoting to keep strings
-  together for comparison. For more details, please read the
-  doc/README.variables file, and check over your dialplan for possible
-  problems.
-
-Agents:
-
-* The default for ackcall has been changed to "no" instead of "yes" 
-  because of a bug which caused the "yes" behavior to generally act like
-  "no".  You may need to adjust the value if your agents behave 
-  differently than you expect with respect to acknowledgement.
-
-* The AgentCallBackLogin application now requires a second '|' before
-  specifying an extension@context.  This is to distinguish the options
-  string from the extension, so that they do not conflict.  See
-  'show application AgentCallbackLogin' for more details.
-
-Parking:
-
-* Parking behavior has changed slightly; when a parked call times out,
-  Asterisk will attempt to deliver the call back to the extension that
-  parked it, rather than the 's' extension. If that extension is busy
-  or unavailable, the parked call will be lost.
-
-Dialing:
-
-* The Caller*ID of the outbound leg is now the extension that was 
-  called, rather than the Caller*ID of the inbound leg of the call.  The 
-  "o" flag for Dial can be used to restore the original behavior if 
-  desired.  Note that if you are looking for the originating callerid
-  from the manager event, there is a new manager event "Dial" which 
-  provides the source and destination channels and callerid.
-
-IAX: 
-
-* The naming convention for IAX channels has changed in 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:
-
-* The global option "port" in 1.0.X that is used to set which port to
-  bind to has been changed to "bindport" to be more consistent with
-  the other channel drivers and to avoid confusion with the "port"
-  option for users/peers.
-
-* The "Registry" event now uses "Username" rather than "User" for 
-  consistency with IAX.
-
-Applications:
-
-* With the addition of dialplan functions (which operate similarly
-  to variables), the SetVar application has been renamed to Set.
-
-* The CallerPres application has been removed.  Use SetCallerPres 
-  instead.  It accepts both numeric and symbolic names.
-
-* The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
-  CheckGroup have been deprecated in favor of functions.  Here is a
-  table of their replacements:
-
-  GetGroupCount([groupname][@category]	       GROUP_COUNT([groupname][@category])	Set(GROUPCOUNT=${GROUP_COUNT()})
-  GroupMatchCount(groupmatch[@category])       GROUP_MATCH_COUNT(groupmatch[@category])	Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
-  SetGroup(groupname[@category])	       GROUP([category])=groupname		Set(GROUP()=test)
-  CheckGroup(max[@category])		       N/A					GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)
-
-  Note that CheckGroup does not have a direct replacement.  There is
-  also a new function called GROUP_LIST() which will return a space
-  separated list of all of the groups set on a channel.  The GROUP()
-  function can also return the name of the group set on a channel when
-  used in a read environment.
-
-* The applications DBGet and DBPut have been deprecated in favor of
-  functions.  Here is a table of their replacements:
-
-  DBGet(foo=family/key)        Set(foo=${DB(family/key)})
-  DBPut(family/key=${foo})     Set(DB(family/key)=${foo})
-
-* The application SetLanguage has been deprecated in favor of the
-  function LANGUAGE().
-
-  SetLanguage(fr)		Set(LANGUAGE()=fr)
-
-  The LANGUAGE function can also return the currently set language:
-
-  Set(MYLANG=${LANGUAGE()})
-
-* The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
-  have been deprecated in favor of the function TIMEOUT(timeouttype):
-
-  AbsoluteTimeout(300)		Set(TIMEOUT(absolute)=300)
-  DigitTimeout(15)		Set(TIMEOUT(digit)=15)
-  ResponseTimeout(15)		Set(TIMEOUT(response)=15)
-
-  The TIMEOUT() function can also return the currently set timeouts:
-
-  Set(DTIMEOUT=${TIMEOUT(digit)})
-
-* The applications SetCIDName, SetCIDNum, and SetRDNIS have been
-  deprecated in favor of the CALLERID(datatype) function:
-
-  SetCIDName(Joe Cool)		Set(CALLERID(name)=Joe Cool)
-  SetCIDNum(2025551212)		Set(CALLERID(number)=2025551212)
-  SetRDNIS(2024561414)		Set(CALLERID(RDNIS)=2024561414)
-
-* The application Record now uses the period to separate the filename
-  from the format, rather than the colon.
-
-* The application VoiceMail now supports a 'temporary' greeting for each
-  mailbox. This greeting can be recorded by using option 4 in the
-  'mailbox options' menu, and 'change your password' option has been
-  moved to option 5.
-
-* 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
-  none of the members are available (e.g. agents not logged on).  To
-  restore the original behavior, use "leavewhenempty=strict" or 
-  "joinwhenempty=strict" instead of "=yes" for those options.
-
-* It is now possible to use multi-digit extensions in the exit context
-  for a queue (although you should not have overlapping extensions,
-  as there is no digit timeout). This means that the EXITWITHKEY event
-  in queue_log can now contain a key field with more than a single
-  character in it.
-
-Extensions:
-
-* By default, there is a new option called "autofallthrough" in
-  extensions.conf that is set to yes.  Asterisk 1.0 (and earlier) 
-  behavior was to wait for an extension to be dialed after there were no 
-  more extensions to execute.  "autofallthrough" changes this behavior
-  so that the call will immediately be terminated with BUSY,
-  CONGESTION, or HANGUP based on Asterisk's best guess.  If you are
-  writing an extension for IVR, you must use the WaitExten application
-  if "autofallthrough" is set to yes.
-
-AGI:
-
-* AGI scripts did not always get SIGHUP at the end, previously.  That 
-  behavior has been fixed.  If you do not want your script to terminate 
-  at the end of AGI being called (e.g. on a hangup) then set SIGHUP to 
-  be ignored within your application.
-
-* CallerID is reported with agi_callerid and agi_calleridname instead
-  of a single parameter holding both.
-
-Music On Hold:
-
-* The preferred format for musiconhold.conf has changed; please see the
-  sample configuration file for the new format. The existing format
-  is still supported but will generate warnings when the module is loaded.
-
-chan_modem:
-
-* All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
-  in this release, and will be removed in the next major Asterisk release.
-  Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
-  path for aopen and bestdata modem users.
-
-MeetMe:
-
-* The conference application now allows users to increase/decrease their
-  speaking volume and listening volume (independently of each other and 
-  other users); the 'admin' and 'user' menus have changed, and new sound 
-  files are included with this release. However, if a user calling in 
-  over a Zaptel channel that does NOT have hardware DTMF detection 
-  increases their speaking volume, it is likely they will no longer be 
-  able to enter/exit the menu or make any further adjustments, as the  
-  software DTMF detector will not be able to recognize the DTMF coming 
-  from their device.
-
-GetVar Manager Action:
-
-* Previously, the behavior of the GetVar manager action reported the value
-  of a variable in the following manner:
-   > name: value
-  This has been changed to a manner similar to the SetVar action and is now
-   > Variable: name
-   > Value: value

UPGRADE-1.4.txt

@@ -1,498 +0,0 @@
-=========================================================
-===
-=== Information for upgrading from Asterisk 1.2 to 1.4
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also includes advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-===
-=========================================================
-
-Build Process (configure script):
-
-Asterisk now uses an autoconf-generated configuration script to learn how it
-should build itself for your system. As it is a standard script, running:
-
-$ ./configure --help
-
-will show you all the options available. This script can be used to tell the
-build process what libraries you have on your system (if it cannot find them
-automatically), which libraries you wish to have ignored even though they may
-be present, etc.
-
-You must run the configure script before Asterisk will build, although it will
-attempt to automatically run it for you with no options specified; for most
-users, that will result in a similar build to what they would have had before
-the configure script was added to the build process (except for having to run
-'make' again after the configure script is run). Note that the configure script
-does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
-when your system configuration changes or you wish to build Asterisk with 
-different options.
-
-Build Process (module selection):
-
-The Asterisk source tree now includes a basic module selection and build option
-selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
-In this tool, you can disable building of modules that you don't care about,
-turn on/off global options for the build and see which modules will not 
-(and cannot) be built because your system does not have the required external
-dependencies installed.
-
-The resulting file from menuselect is called 'menuselect.makeopts'. Note that
-the resulting menuselect.makeopts file generally contains which modules *not*
-to build. The modules listed in this file indicate which modules have unmet
-dependencies, a present conflict, or have been disabled by the user in the
-menuselect interface. Compiler Flags can also be set in the menuselect
-interface.  In this case, the resulting file contains which CFLAGS are in use,
-not which ones are not in use.
-
-If you would like to save your choices and have them applied against all
-builds, the file can be copied to '~/.asterisk.makeopts' or 
-'/etc/asterisk.makeopts'.
-
-Build Process (Makefile targets):
-
-The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
-is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
-in the menuselect tool.
-
-It is now possible to run most make targets against a single subdirectory; from
-the top level directory, for example, 'make channels' will run 'make all' in the
-'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.
-
-Sound (prompt) and Music On Hold files:
-
-Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
-use with Asterisk have been replaced with new versions produced from high quality
-master recordings, and are available in three languages (English, French and
-Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
-In addition, the music on hold files provided by opsound.org Music are now available
-in the same five formats, but no longer available in MP3 format.
-
-The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
-(as were supplied with previous releases) and the opsound.org MOH files in WAV format.
-All of the other variations can be installed by running 'make menuselect' and
-selecting the packages you wish to install; when you run 'make install', those
-packages will be downloaded and installed along with the standard files included
-in the tarball.
-
-If for some reason you expect to not have Internet access at the time you will be
-running 'make install', you can make your package selections using menuselect and
-then run 'make sounds' to download (only) the sound packages; this will leave the
-sound packages in the 'sounds' subdirectory to be used later during installation.
-
-WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
-instead of the alternate-language files being stored in subdirectories underneath
-the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
-etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
-language itself, then places all the sound files for that language under that
-directory and its subdirectories. This is the layout that will be created if you
-select non-English languages to be installed via menuselect, HOWEVER Asterisk does
-not default to this layout and will not find the files in the places it expects them
-to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
-/etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
-installed.
-
-PBX Core:
-
-* The (very old and undocumented) ability to use BYEXTENSION for dialing
-  instead of ${EXTEN} has been removed.
-  
-* Builtin (res_features) transfer functionality attempts to use the context
-  defined in TRANSFER_CONTEXT variable of the transferer channel first. If
-  not set, it uses the transferee variable. If not set in any channel, it will 
-  attempt to use the last non macro context. If not possible, it will default
-  to the current context.
-
-* The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
-  if your dialplan relies on the ability to 'run off the end' of an extension
-  and wait for a new extension without using WaitExten() to accomplish that,
-  you will need set autofallthrough to 'no' in your extensions.conf file.
- 
-Command Line Interface:
-
-* 'show channels concise', designed to be used by applications that will parse
-  its output, previously used ':' characters to separate fields. However, some
-  of those fields can easily contain that character, making the output not
-  parseable. The delimiter has been changed to '!'.
-
-Applications:
-
-* In previous Asterisk releases, many applications would jump to priority n+101
-  to indicate some kind of status or error condition.  This functionality was
-  marked deprecated in Asterisk 1.2.  An option to disable it was provided with
-  the default value set to 'on'.  The default value for the global priority
-  jumping option is now 'off'.
-
-* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
-  AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
-  and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
-  been removed in this version.  You should use the equivalent dialplan
-  function in places where you have previously used one of these applications.
-
-* The application SetGlobalVar has been deprecated.  You should replace uses
-  of this application with the following combination of Set and GLOBAL():
-  Set(GLOBAL(name)=value).  You may also access global variables exclusively by
-  using the GLOBAL() dialplan function, instead of relying on variable
-  interpolation falling back to globals when no channel variable is set.
-
-* The application SetVar has been renamed to Set.  The syntax SetVar was marked
-  deprecated in version 1.2 and is no longer recognized in this version.  The
-  use of Set with multiple argument pairs has also been deprecated.  Please
-  separate each name/value pair into its own dialplan line.
-
-* app_read has been updated to use the newer options codes, using "skip" or
-  "noanswer" will not work.  Use s or n.  Also there is a new feature i, for
-  using indication tones, so typing in skip would give you unexpected results.
-
-* OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
-
-* The CONNECT event in the queue_log from app_queue now has a second field 
-  in addition to the holdtime field. It contains the unique ID of the 
-  queue member channel that is taking the call. This is useful when trying 
-  to link recording filenames back to a particular call from the queue.  
-
-* The old/current behavior of app_queue has a serial type behavior
-  in that the queue will make all waiting callers wait in the queue
-  even if there is more than one available member ready to take
-  calls until the head caller is connected with the member they
-  were trying to get to. The next waiting caller in line then
-  becomes the head caller, and they are then connected with the
-  next available member and all available members and waiting callers
-  waits while this happens. This cycle continues until there are
-  no more available members or waiting callers, whichever comes first.
-  The new behavior, enabled by setting autofill=yes in queues.conf
-  either at the [general] level to default for all queues or 
-  to set on a per-queue level, makes sure that when the waiting 
-  callers are connecting with available members in a parallel fashion 
-  until there are no more available members or no more waiting callers,
-  whichever comes first. This is probably more along the lines of how
-  one would expect a queue should work and in most cases, you will want 
-  to enable this new behavior. If you do not specify or comment out this 
-  option, it will default to "no" to keep backward compatability with the old 
-  behavior.
-
-* Queues depend on the channel driver reporting the proper state
-  for each member of the queue. To get proper signalling on
-  queue members that use the SIP channel driver, you need to
-  enable a call limit (could be set to a high value so it
-  is not put into action) and also make sure that both inbound
-  and outbound calls are accounted for.
-
-  Example:
-
-       [general]
-       limitonpeer = yes
-
-       [peername]
-       type=friend
-       call-limit=10
-
-
-* The app_queue application now has the ability to use MixMonitor to 
-  record conversations queue members are having with queue callers. Please
-  see configs/queues.conf.sample for more information on this option.
-
-* The app_queue application strategy called 'roundrobin' has been deprecated
-  for this release. Users are encouraged to use 'rrmemory' instead, since it
-  provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
-  'rrmemory' will be renamed 'roundrobin'.
-
-* The app_queue application option called 'monitor-join' has been deprecated
-  for this release. Users are encouraged to use 'monitor-type=mixmonitor' instead,
-  since it provides the same functionality but is not dependent on soxmix or some
-  other external program in order to mix the audio.
-
-* app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
-  the 'm' option now provides the functionality of "initially muted". 
-  In practice, most existing dialplans using the 'm' flag should not notice
-  any difference, unless the keypad menu is enabled, allowing the user 
-  to unmute themsleves.
-
-* ast_play_and_record would attempt to cancel the recording if a DTMF
-  '0' was received.  This behavior was not documented in most of the
-  applications that used ast_play_and_record and the return codes from
-  ast_play_and_record weren't checked for properly.
-  ast_play_and_record has been changed so that '0' no longer cancels a
-  recording.  If you want to allow DTMF digits to cancel an
-  in-progress recording use ast_play_and_record_full which allows you
-  to specify which DTMF digits can be used to accept a recording and
-  which digits can be used to cancel a recording.
-
-* ast_app_messagecount has been renamed to ast_app_inboxcount.  There is now a
-  new ast_app_messagecount function which takes a single context/mailbox/folder
-  mailbox specification and returns the message count for that folder only.
-  This addresses the deficiency of not being able to count the number of
-  messages in folders other than INBOX and Old.
-
-* The exit behavior of the AGI applications has changed. Previously, when
-  a connection to an AGI server failed, the application would cause the channel
-  to immediately stop dialplan execution and hangup. Now, the only time that
-  the AGI applications will cause the channel to stop dialplan execution is
-  when the channel itself requests hangup. The AGI applications now set an
-  AGISTATUS variable which will allow you to find out whether running the AGI
-  was successful or not.
-
-  Previously, there was no way to handle the case where Asterisk was unable to
-  locally execute an AGI script for some reason. In this case, dialplan
-  execution will continue as it did before, but the AGISTATUS variable will be
-  set to "FAILURE".
-
-  A locally executed AGI script can now exit with a non-zero exit code and this
-  failure will be detected by Asterisk. If an AGI script exits with a non-zero
-  exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
-  "SUCCESS".
-
-* app_voicemail: The ODBC_STORAGE capability now requires the extended table format
-  previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
-  your table format using the schema provided in doc/odbcstorage.txt
-
-* app_waitforsilence: Fixes have been made to this application which changes the 
-  default behavior with how quickly it returns. You can maintain "old-style" behavior
-  with the addition/use of a third "timeout" parameter.
-  Please consult the application documentation and make changes to your dialplan 
-  if appropriate.
-
-Manager:
-
-* After executing the 'status' manager action, the "Status" manager events
-  included the header "CallerID:" which was actually only the CallerID number,
-  and not the full CallerID string.  This header has been renamed to
-  "CallerIDNum".  For compatibility purposes, the CallerID parameter will remain
-  until after the release of 1.4, when it will be removed.  Please use the time
-  during the 1.4 release to make this transition.
-
-* The AgentConnect event now has an additional field called "BridgedChannel" 
-  which contains the unique ID of the queue member channel that is taking the 
-  call. This is useful when trying to link recording filenames back to 
-  a particular call from the queue.
-
-* app_userevent has been modified to always send Event: UserEvent with the
-  additional header UserEvent: <userspec>.  Also, the Channel and UniqueID
-  headers are not automatically sent, unless you specify them as separate
-  arguments.  Please see the application help for the new syntax.
-
-* app_meetme: Mute and Unmute events are now reported via the Manager API.
-  Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
-  are easier to use than "Action Command:". The MeetMeStopTalking event has
-  also been deprecated in favor of the already existing MeetmeTalking event
-  with a "Status" of "on" or "off" added.
-
-* OriginateFailure and OriginateSuccess events were replaced by event
-  OriginateResponse with a header named "Response" to indicate success or
-  failure
-
-Variables:
-
-* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
-  ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
-  and ${LANGUAGE} have all been deprecated in favor of their related dialplan
-  functions.  You are encouraged to move towards the associated dialplan
-  function, as these variables will be removed in a future release.
-
-* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now 
-  adjustable from cdr.conf, instead of recompiling.
-
-* OSP applications exports several new variables, ${OSPINHANDLE},
-  ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
-  ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
-  
-* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
-  created channel. This variables holds the channel name of the transferer.
-
-* The dial plan variable PRI_CAUSE will be removed from future versions 
-  of Asterisk.
-  It is replaced by adding a cause value to the hangup() application.
-
-Functions:
-
-* The function ${CHECK_MD5()} has been deprecated in favor of using an
-  expression: $[${MD5(<string>)} = ${saved_md5}].
-
-* The 'builtin' functions that used to be combined in pbx_functions.so are
-  now built as separate modules. If you are not using 'autoload=yes' in your
-  modules.conf file then you will need to explicitly load the modules that
-  contain the functions you want to use.
-
-* The ENUMLOOKUP() function with the 'c' option (for counting the number of 
-  records), but the lookup fails to match any records, the returned value will 
-  now be "0" instead of blank.
-
-* The REALTIME() function is now available in version 1.4 and app_realtime has
-  been deprecated in favor of the new function. app_realtime will be removed
-  completely with the version 1.6 release so please take the time between
-  releases to make any necessary changes
-
-* The QUEUEAGENTCOUNT() function has been deprecated in favor of
-  QUEUE_MEMBER_COUNT().
-
-The IAX2 channel:
-
-* It is possible that previous configurations depended on the order in which
-  peers and users were specified in iax.conf for forcing the order in which
-  chan_iax2 matched against them.  This behavior is going away and is considered
-  deprecated in this version.  Avoid having ambiguous peer and user entries and
-  to make things easy on yourself, always set the "username" option for users
-  so that the remote end can match on that exactly instead of trying to infer
-  which user you want based on host.
-
-  If you would like to go ahead and use the new behavior which doesn't use the
-  order in the config file to influence matching order, then change the 
-  MAX_PEER_BUCKETS define in chan_iax2.c to a value greater than one.  An
-  example is provided there.  By changing this, you will get *much* better
-  performance on systems that do a lot of peer and user lookups as they will be
-  stored in memory in a much more efficient manner.
-
-* The "mailboxdetail" option has been deprecated.  Previously, if this option
-  was not enabled, the 2 byte MSGCOUNT information element would be set to all
-  1's to indicate there there is some number of messages waiting.  With this
-  option enabled, the number of new messages were placed in one byte and the
-  number of old messages are placed in the other.  This is now the default
-  (and the only) behavior.
-
-The SIP channel:
-
-* The "incominglimit" setting is replaced by the "call-limit" setting in 
-  sip.conf.
-
-* OSP support code is removed from SIP channel to OSP applications. ospauth 
-  option in sip.conf is removed to osp.conf as authpolicy. allowguest option
-  in sip.conf cannot be set as osp anymore. 
-
-* The Asterisk RTP stack has been changed in regards to RFC2833 reception
-  and transmission. Packets will now be sent with proper duration instead of all
-  at once. If you are receiving calls from a pre-1.4 Asterisk installation you
-  will want to turn on the rfc2833compensate option. Without this option your
-  DTMF reception may act poorly.
-
-* The $SIPUSERAGENT dialplan variable is deprecated and will be removed
-  in coming versions of Asterisk. Please use the dialplan function
-  SIPCHANINFO(useragent) instead.
-
-* The ALERT_INFO dialplan variable is deprecated and will be removed
-  in coming versions of Asterisk. Please use the dialplan application
-  sipaddheader() to add the "Alert-Info" header to the outbound invite.
-
-* The "canreinvite" option has changed. canreinvite=yes used to disable
-  re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat
-  to disable re-invites when NAT=yes. This is propably what you want.
-  The settings are now: "yes", "no", "nonat", "update". Please consult
-  sip.conf.sample for detailed information.
-
-The Zap channel:
-
-* Support for MFC/R2 has been removed, as it has not been functional for some
-  time and it has no maintainer.
-
-The Agent channel:
-
-* Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
-  it provided can be done using dialplan logic, without requiring additional
-  channel and module locks (which frequently caused deadlocks). An example of
-  how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.
-
-The G726-32 codec:
-
-* It has been determined that previous versions of Asterisk used the wrong codeword
-  packing order for G726-32 data. This version supports both available packing orders,
-  and can transcode between them. It also now selects the proper order when
-  negotiating with a SIP peer based on the codec name supplied in the SDP. However,
-  there are existing devices that improperly request one order and then use another;
-  Sipura and Grandstream ATAs are known to do this, and there may be others. To
-  be able to continue to use these devices with this version of Asterisk and the
-  G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
-  to sip.conf, so that Asterisk can use the packing order expected by the device (even
-  though it requested a different order). In addition, the internal format number for
-  G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
-  result of this is that this version of Asterisk will be able to interoperate over
-  IAX2 with older versions of Asterisk, as long as this version is told to allow
-  'g726aal2' instead of 'g726' as the codec for the call.
-
-Installation:
-
-* On BSD systems, the installation directories have changed to more "FreeBSDish"
-  directories. On startup, Asterisk will look for the main configuration in 
-  /usr/local/etc/asterisk/asterisk.conf
-  If you have an old installation, you might want to remove the binaries and 
-  move the configuration files to the new locations. The following directories 
-  are now default:
-	ASTLIBDIR	/usr/local/lib/asterisk
-	ASTVARLIBDIR	/usr/local/share/asterisk
-	ASTETCDIR	/usr/local/etc/asterisk
-	ASTBINDIR	/usr/local/bin/asterisk
-	ASTSBINDIR	/usr/local/sbin/asterisk
-
-Music on Hold:
-
-* The music on hold handling has been changed in some significant ways in hopes
-  to make it work in a way that is much less confusing to users. Behavior will
-  not change if the same configuration is used from older versions of Asterisk.
-  However, there are some new configuration options that will make things work
-  in a way that makes more sense.
-
-  Previously, many of the channel drivers had an option called "musicclass" or
-  something similar. This option set what music on hold class this channel
-  would *hear* when put on hold. Some people expected (with good reason) that
-  this option was to configure what music on hold class to play when putting
-  the bridged channel on hold. This option has now been deprecated.
-
-  Two new music on hold related configuration options for channel drivers have
-  been introduced. Some channel drivers support both options, some just one,
-  and some support neither of them. Check the sample configuration files to see
-  which options apply to which channel driver.
-
-  The "mohsuggest" option specifies which music on hold class to suggest to the
-  bridged channel when putting them on hold. The only way that this class can
-  be overridden is if the bridged channel has a specific music class set that
-  was done in the dialplan using Set(CHANNEL(musicclass)=something).
-
-  The "mohinterpret" option is similar to the old "musicclass" option. It
-  specifies which music on hold class this channel would like to listen to when
-  put on hold. This music class is only effective if this channel has no music
-  class set on it from the dialplan and the bridged channel putting this one on
-  hold had no "mohsuggest" setting.
-
-  The IAX2 and Zap channel drivers have an additional feature for the
-  "mohinterpret" option. If this option is set to "passthrough", then these
-  channel drivers will pass through the HOLD message in signalling instead of
-  starting music on hold on the channel. An example for how this would be
-  useful is in an enterprise network of Asterisk servers. When one phone on one
-  server puts a phone on a different server on hold, the remote server will be
-  responsible for playing the hold music to its local phone that was put on
-  hold instead of the far end server across the network playing the music.
-
-CDR Records:
-
-* The behavior of the "clid" field of the CDR has always been that it will
-  contain the callerid ANI if it is set, or the callerid number if ANI was not
-  set.  When using the "callerid" option for various channel drivers, some
-  would set ANI and some would not.  This has been cleared up so that all
-  channel drivers set ANI.  If you would like to change the callerid number
-  on the channel from the dialplan and have that change also show up in the 
-  CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).
-
-API:
-
-* There are some API functions that were not previously prefixed with the 'ast_'
-  prefix but now are; these include the ADSI, ODBC and AGI interfaces. If you
-  have a module that uses the services provided by res_adsi, res_odbc, or
-  res_agi, you will need to add ast_ prefixes to the functions that you call
-  from those modules.
-
-Formats:
-
-* format_wav: The GAIN preprocessor definition has been changed from 2 to 0
-  in Asterisk 1.4.  This change was made in response to user complaints of
-  choppiness or the clipping of loud signal peaks.  The GAIN preprocessor
-  definition will be retained in Asterisk 1.4, but will be removed in a 
-  future release.  The use of GAIN for the increasing of voicemail message
-  volume should use the 'volgain' option in voicemail.conf
-

UPGRADE-1.6.txt

@@ -1,271 +0,0 @@
-=========================================================
-===
-=== Information for upgrading from Asterisk 1.4 to 1.6
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also includes advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-===
-=========================================================
-
-AEL:
-
-* Macros are now implemented underneath with the Gosub() application.
-  Heaven Help You if you wrote code depending on any aspect of this!
-  Previous to 1.6, macros were implemented with the Macro() app, which
-  provided a nice feature of auto-returning. The compiler will do its
-  best to insert a Return() app call at the end of your macro if you did
-  not include it, but really, you should make sure that all execution
-  paths within your macros end in "return;".
-
-* The conf2ael program is 'introduced' in this release; it is in a rather
-  crude state, but deemed useful for making a first pass at converting
-  extensions.conf code into AEL. More intelligence will come with time.
-
-Core:
-
-* The 'languageprefix' option in asterisk.conf is now deprecated, and
-  the default sound file layout for non-English sounds is the 'new
-  style' layout introduced in Asterisk 1.4 (and used by the automatic
-  sound file installer in the Makefile).
-
-* The ast_expr2 stuff has been modified to handle floating-point numbers.
-  Numbers of the format D.D are now acceptable input for the expr parser, 
-  Where D is a string of base-10 digits. All math is now done in "long double",
-  if it is available on your compiler/architecture. This was half-way between
-  a bug-fix (because the MATH func returns fp by default), and an enhancement.
-  Also, for those counting on, or needing, integer operations, a series of
-  'functions' were also added to the expr language, to allow several styles
-  of rounding/truncation, along with a set of common floating point operations,
-  like sin, cos, tan, log, pow, etc. The ability to call external functions
-  like CDR(), etc. was also added, without having to use the ${...} notation.
- 
-* The delimiter passed to applications has been changed to the comma (','), as
-  that is what people are used to using within extensions.conf.  If you are
-  using realtime extensions, you will need to translate your existing dialplan
-  to use this separator.  To use a literal comma, you need merely to escape it
-  with a backslash ('\').  Another possible side effect is that you may need to
-  remove the obscene level of backslashing that was necessary for the dialplan
-  to work correctly in 1.4 and previous versions.  This should make writing
-  dialplans less painful in the future, albeit with the pain of a one-time
-  conversion.  If you would like to avoid this conversion immediately, set
-  pbx_realtime=1.4 in the [compat] section of asterisk.conf.  After
-  transitioning, set pbx_realtime=1.6 in the same section.
-
-* For the same purpose as above, you may set res_agi=1.4 in the [compat]
-  section of asterisk.conf to continue to use the '|' delimiter in the EXEC
-  arguments of AGI applications.  After converting to use the ',' delimiter,
-  change this option to res_agi=1.6.
-
-* As a side effect of the application delimiter change, many places that used
-  to need quotes in order to get the proper meaning are no longer required.
-  You now only need to quote strings in configuration files if you literally
-  want quotation marks within a string.
-
-* Any applications run that contain the pipe symbol but not a comma symbol will
-  get a warning printed to the effect that the application delimiter has changed.
-  However, there are legitimate reasons why this might be useful in certain
-  situations, so this warning can be turned off with the dontwarn option in
-  asterisk.conf.
-
-* The logger.conf option 'rotatetimestamp' has been deprecated in favor of
-  'rotatestrategy'.  This new option supports a 'rotate' strategy that more
-  closely mimics the system logger in terms of file rotation.
-
-* The concise versions of various CLI commands are now deprecated. We recommend
-  using the manager interface (AMI) for application integration with Asterisk.
-
-Voicemail:
-
-* The voicemail configuration values 'maxmessage' and 'minmessage' have
-  been changed to 'maxsecs' and 'minsecs' to clarify their purpose and
-  to make them more distinguishable from 'maxmsgs', which sets folder
-  size.  The old variables will continue to work in this version, albeit
-  with a deprecation warning.
-
-* If you use any interface for modifying voicemail aside from the built in
-  dialplan applications, then the option "pollmailboxes" *must* be set in
-  voicemail.conf for message waiting indication (MWI) to work properly.  This
-  is because Voicemail notification is now event based instead of polling
-  based.  The channel drivers are no longer responsible for constantly manually
-  checking mailboxes for changes so that they can send MWI information to users.
-  Examples of situations that would require this option are web interfaces to
-  voicemail or an email client in the case of using IMAP storage.
-
-Applications:
-
-
-* ChanIsAvail() now has a 't' option, which allows the specified device
-  to be queried for state without consulting the channel drivers. This
-  performs mostly a 'ChanExists' sort of function.
-
-* ChannelRedirect() will not terminate the channel that fails to do a
-  channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS
-  will reflect if the attempt was successful of not.
-
-* SetCallerPres() has been replaced with the CALLERPRES() dialplan function
-  and is now deprecated.
-
-* DISA()'s fifth argument is now an options argument.  If you have previously
-  used 'NOANSWER' in this argument, you'll need to convert that to the new
-  option 'n'.
-
-* Macro() is now deprecated.  If you need subroutines, you should use the
-  Gosub()/Return() applications.  To replace MacroExclusive(), we have
-  introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK().  You may use
-  these functions in any location where you desire to ensure that only one
-  channel is executing that path at any one time.  The Macro() applications
-  are deprecated for performance reasons.  However, since Macro() has been
-  around for a long time and so many dialplans depend heavily on it, for the
-  sake of backwards compatibility it will not be removed .  It is also worth
-  noting that using both Macro() and GoSub() at the same time is _heavily_
-  discouraged.
-
-* Read() now sets a READSTATUS variable on exit.  It does NOT automatically
-  return -1 (and hangup) anymore on error.  If you want to hangup on error,
-  you need to do so explicitly in your dialplan.
-
-* Privacy() no longer uses privacy.conf, so any options must be specified
-  directly in the application arguments.
-
-* MusicOnHold application now has duration parameter which allows specifying
-  timeout in seconds.
-
-* WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold.
-
-* SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...)
-  instead.
-
-* The arguments in ExecIf changed a bit, to be more like other applications.
-  The syntax is now ExecIf(<cond>?appiftrue(args):appiffalse(args)).
-
-* The behavior of the Set application now depends upon a compatibility option,
-  set in asterisk.conf.  To use the old 1.4 behavior, which allowed Set to take
-  multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf.  To
-  use the new behavior, which permits variables to be set with embedded commas,
-  set app_set=1.6 in [compat] in asterisk.conf.  Note that you can have both
-  behaviors at the same time, if you switch to using MSet if you want the old
-  behavior.
-
-Dialplan Functions:
-
-* QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For
-  more information, issue a "show function QUEUE_MEMBER" from the CLI.
-
-CDR:
-
-* The cdr_sqlite module has been marked as deprecated in favor of
-  cdr_sqlite3_custom.  It will potentially be removed from the tree
-  after Asterisk 1.6 is released.
-
-* The cdr_odbc module now uses res_odbc to manage its connections.  The
-  username and password parameters in cdr_odbc.conf, therefore, are no
-  longer used.  The dsn parameter now points to an entry in res_odbc.conf.
-
-* The uniqueid field in the core Asterisk structure has been changed from a
-  maximum 31 character field to a 149 character field, to account for all
-  possible values the systemname prefix could be.  In the past, if the
-  systemname was too long, the uniqueid would have been truncated.
-
-* The cdr_tds module now supports all versions of FreeTDS that contain
-  the db-lib frontend.  It will also now log the userfield variable if
-  the target database table contains a column for it.
-
-Formats:
-
-* format_wav: The GAIN preprocessor definition and source code that used it
-  is removed.  This change was made in response to user complaints of
-  choppiness or the clipping of loud signal peaks.  To increase the volume
-  of voicemail messages, use the 'volgain' option in voicemail.conf
-
-Channel Drivers:
-
-* SIP: a small upgrade to support the "Record" button on the SNOM360,
-  which sends a sip INFO message with a "Record: on" or "Record: off" 
-  header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
-  requests (by default, via '*1'), then the user-configured dialpad sequence
-  is generated, and recording can be started and stopped via this button. The
-  file names and formats are all controlled via the normal mechanisms. If the
-  user has not configured the automon feature, the normal "415 Unsupported media type"
-  is returned, and nothing is done.
-
-* SIP: The "call-limit" option is marked as deprecated. It still works in this version of
-  Asterisk, but will be removed in the following version. Please use the groupcount functions
-  in the dialplan to enforce call limits. The "limitonpeer" configuration option is
-  now renamed to "counteronpeer".
-
-* SIP: The "username" option is now renamed to "defaultuser" to match "defaultip".
-  These are used only before registration to call a peer with the uri 
-	sip:defaultuser@defaultip
-  The "username" setting still work, but is deprecated and will not work in 
-  the next version of Asterisk.
-
-* chan_local.c: the comma delimiter inside the channel name has been changed to a
-  semicolon, in order to make the Local channel driver compatible with the comma
-  delimiter change in applications.
-
-* H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio"
-  to be compatible with settings in sip.conf. The "tos" and "cos" configuration
-  is deprecated and will stop working in the next release of Asterisk.
-
-* Console: A new console channel driver, chan_console, has been added to Asterisk.
-  This new module can not be loaded at the same time as chan_alsa or chan_oss.  The
-  default modules.conf only loads one of them (chan_oss by default).  So, unless you
-  have modified your modules.conf to not use the autoload option, then you will need
-  to modify modules.conf to add another "noload" line to ensure that only one of
-  these three modules gets loaded.
-
-* DAHDI: The chan_zap module that supported PSTN interfaces using
-  Zaptel has been renamed to chan_dahdi, and only supports the DAHDI
-  telephony driver package for PSTN interfaces. See the
-  Zaptel-to-DAHDI.txt file for more details on this transition.
-
-* DAHDI: The "msdstrip" option has been deprecated, as it provides no value over
-  the method of stripping digits in the dialplan using variable substring syntax.
-
-Configuration:
-
-* pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay,
-  lowcost and other is not acceptable now. Look into qos.tex for description of 
-  this parameter.
-
-* queues.conf: the queue-lessthan sound file option is no longer available, and the
-  queue-round-seconds option no longer takes '1' as a valid parameter.
-
-Manager:
-
-* Manager has been upgraded to version 1.1 with a lot of changes. 
-  Please check doc/manager_1_1.txt for information
-
-* The IAXpeers command output has been changed to more closely resemble the
-  output of the SIPpeers command.
-
-* cdr_manager now reports at the "cdr" level, not at "call"  You may need to
-   change your manager.conf to add the level to existing AMI users, if they
-   want to see the CDR events generated.
-
-* The Originate command now requires the Originate write permission.  For
-   Originate with the Application parameter, you need the additional System
-   privilege if you want to do anything that calls out to a subshell.
-
-iLBC Codec:
-
-* Previously, the Asterisk source code distribution included the iLBC
-  encoder/decoder source code, from Global IP Solutions
-  (http://www.gipscorp.com). This code is not licensed for
-  distribution, and thus has been removed from the Asterisk source
-  code distribution. If you wish to use codec_ilbc to support iLBC
-  channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh
-  script to download the source and put it in the proper place in
-  the Asterisk build tree. Once that is done you can follow your normal
-  steps of building Asterisk. You will need to run 'menuselect' and enable
-  the iLBC codec in the 'Codec  Translators' category.

UPGRADE.txt

@@ -1,88 +1,205 @@
-===========================================================
-===
-=== Information for upgrading between Asterisk 1.6 versions
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also includes advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
-===
-===========================================================
+Information for Upgrading From Previous Asterisk Releases
+=========================================================
 
-As of 1.6.0.16:
+Compiling:
 
-* The firmware for the IAXy has been removed from Asterisk.  It can be
-  downloaded from http://downloads.digium.com/pub/iaxy/.  To have Asterisk
-  install the firmware into its proper location, place the firmware in the
-  contrib/firmware/iax/ directory in the Asterisk source tree before running
-  "make install".
+* The Asterisk 1.2 source code now uses C language features
+  supported only by 'modern' C compilers.  Generally, this means GCC
+  version 3.0 or higher, although some GCC 2.96 releases will also
+  work.  Some non-GCC compilers that support C99 and the common GCC
+  extensions (including anonymous structures and unions) will also
+  work.  All releases of GCC 2.95 do _not_ have the requisite feature
+  support; systems using that compiler will need to be upgraded to
+  a more recent compiler release.
 
-* T.38 FAX error correction mode can no longer be configured in udptl.conf;
-  instead, it is configured on a per-peer (or global) basis in sip.conf, with
-  the same default as was present in udptl.conf.sample.
+Dialplan Expressions:
 
-* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
-  instead, it is either supplied by the application servicing the T.38 channel
-  (for a FAX send or receive) or calculated from the bridged endpoint's
-  maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
-  allows for overriding the value supplied by a remote endpoint, which is useful
-  when T.38 connections are made to gateways that supply incorrectly-calculated
-  maximum datagram sizes.
+* The dialplan expression parser (which handles $[ ... ] constructs)
+  has gone through a major upgrade, but has one incompatible change:
+  spaces are no longer required around expression operators, including
+  string comparisons. However, you can now use quoting to keep strings
+  together for comparison. For more details, please read the
+  doc/README.variables file, and check over your dialplan for possible
+  problems.
 
-As of 1.6.0.15:
+Agents:
 
-* There have been some changes to the IAX2 protocol to address the security
-  concerns documented in the security advisory AST-2009-006.  Please see the
-  IAX2 security document, doc/IAX2-security.pdf, for information regarding
-  backwards compatibility with versions of Asterisk that do not contain these
-  changes to IAX2.
+* The default for ackcall has been changed to "no" instead of "yes" 
+  because of a bug which caused the "yes" behavior to generally act like
+  "no".  You may need to adjust the value if your agents behave 
+  differently than you expect with respect to acknowledgement.
 
-From 1.6.0.10 to 1.6.0.11:
+* The AgentCallBackLogin application now requires a second '|' before
+  specifying an extension@context.  This is to distinguish the options
+  string from the extension, so that they do not conflict.  See
+  'show application AgentCallbackLogin' for more details.
 
-* Beginning with this release, Asterisk's internal methods of
-  negotiating T.38 (FAX over IP) sessions changed in
-  non-backwards-compatible ways. Any applications that previously used
-  AST_CONTROL_T38 control frames will have to be upgraded to use
-  AST_CONTROL_T38_PARAMETERS control frames instead; app_fax.c is a good
-  example of how to generate and respond to these frames. These changes
-  were made to solve significant T.38 interoperability problems between
-  Asterisk and various SIP/T.38 endpoints identified by many users of
-  Asterisk.
+Parking:
 
-From 1.6.0.9 to 1.6.0.10:
+* 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.
 
-* Support for Taiwanese was incorrectly supported with the "tw" language code.
-  In reality, the "tw" language code is reserved for the Twi language, native
-  to Ghana.  If you were previously using the "tw" language code, you should
-  switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
-  specific localizations.  Additionally, "mx" should be changed to "es_MX",
-  Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
-  "cs", not "cz".
+Dialing:
 
-From 1.6.0.1 to 1.6.0.2:
+* The Caller*ID of the outbound leg is now the extension that was 
+  called, rather than the Caller*ID of the inbound leg of the call.  The 
+  "o" flag for Dial can be used to restore the original behavior if 
+  desired.  Note that if you are looking for the originating callerid
+  from the manager event, there is a new manager event "Dial" which 
+  provides the source and destination channels and callerid.
 
-* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
-  API calls were added in 1.6.0, so that modules that provide multiple
-  AGI commands could register/unregister them all with a single
-  step. However, these API calls were not implemented properly, and did
-  not allow the caller to know whether registration or unregistration
-  succeeded or failed. They have been redefined to now return success
-  or failure, but this means any code using these functions will need
-  be recompiled after upgrading to a version of Asterisk containing
-  these changes. In addition, the source code using these functions
-  should be reviewed to ensure it can properly react to failure
-  of registration or unregistration of its API commands.
+IAX: 
 
-* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
-  to better match what it really does, and the argument order has been
-  changed to be consistent with other API calls that perform similar
-  operations.
+* 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:
+
+* The global option "port" in 1.0.X that is used to set which port to
+  bind to has been changed to "bindport" to be more consistent with
+  the other channel drivers and to avoid confusion with the "port"
+  option for users/peers.
+
+* The "Registry" event now uses "Username" rather than "User" for 
+  consistency with IAX.
+
+Applications:
+
+* With the addition of dialplan functions (which operate similarly
+  to variables), the SetVar application has been renamed to Set.
+
+* The CallerPres application has been removed.  Use SetCallerPres 
+  instead.  It accepts both numeric and symbolic names.
+
+* The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
+  CheckGroup have been deprecated in favor of functions.  Here is a
+  table of their replacements:
+
+  GetGroupCount([groupname][@category]	       GROUP_COUNT([groupname][@category])	Set(GROUPCOUNT=${GROUP_COUNT()})
+  GroupMatchCount(groupmatch[@category])       GROUP_MATCH_COUNT(groupmatch[@category])	Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
+  SetGroup(groupname[@category])	       GROUP([category])=groupname		Set(GROUP()=test)
+  CheckGroup(max[@category])		       N/A					GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)
+
+  Note that CheckGroup does not have a direct replacement.  There is
+  also a new function called GROUP_LIST() which will return a space
+  separated list of all of the groups set on a channel.  The GROUP()
+  function can also return the name of the group set on a channel when
+  used in a read environment.
+
+* The applications DBGet and DBPut have been deprecated in favor of
+  functions.  Here is a table of their replacements:
+
+  DBGet(foo=family/key)        Set(foo=${DB(family/key)})
+  DBPut(family/key=${foo})     Set(DB(family/key)=${foo})
+
+* The application SetLanguage has been deprecated in favor of the
+  function LANGUAGE().
+
+  SetLanguage(fr)		Set(LANGUAGE()=fr)
+
+  The LANGUAGE function can also return the currently set language:
+
+  Set(MYLANG=${LANGUAGE()})
+
+* The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
+  have been deprecated in favor of the function TIMEOUT(timeouttype):
+
+  AbsoluteTimeout(300)		Set(TIMEOUT(absolute)=300)
+  DigitTimeout(15)		Set(TIMEOUT(digit)=15)
+  ResponseTimeout(15)		Set(TIMEOUT(response)=15)
+
+  The TIMEOUT() function can also return the currently set timeouts:
+
+  Set(DTIMEOUT=${TIMEOUT(digit)})
+
+* The applications SetCIDName, SetCIDNum, and SetRDNIS have been
+  deprecated in favor of the CALLERID(datatype) function:
+
+  SetCIDName(Joe Cool)		Set(CALLERID(name)=Joe Cool)
+  SetCIDNum(2025551212)		Set(CALLERID(number)=2025551212)
+  SetRDNIS(2024561414)		Set(CALLERID(RDNIS)=2024561414)
+
+* The application Record now uses the period to separate the filename
+  from the format, rather than the colon.
+
+* The application VoiceMail now supports a 'temporary' greeting for each
+  mailbox. This greeting can be recorded by using option 4 in the
+  'mailbox options' menu, and 'change your password' option has been
+  moved to option 5.
+
+* 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
+  none of the members are available (e.g. agents not logged on).  To
+  restore the original behavior, use "leavewhenempty=strict" or 
+  "joinwhenempty=strict" instead of "=yes" for those options.
+
+* It is now possible to use multi-digit extensions in the exit context
+  for a queue (although you should not have overlapping extensions,
+  as there is no digit timeout). This means that the EXITWITHKEY event
+  in queue_log can now contain a key field with more than a single
+  character in it.
+
+Extensions:
+
+* By default, there is a new option called "autofallthrough" in
+  extensions.conf that is set to yes.  Asterisk 1.0 (and earlier) 
+  behavior was to wait for an extension to be dialed after there were no 
+  more extensions to execute.  "autofallthrough" changes this behavior
+  so that the call will immediately be terminated with BUSY,
+  CONGESTION, or HANGUP based on Asterisk's best guess.  If you are
+  writing an extension for IVR, you must use the WaitExten application
+  if "autofallthrough" is set to yes.
+
+AGI:
+
+* AGI scripts did not always get SIGHUP at the end, previously.  That 
+  behavior has been fixed.  If you do not want your script to terminate 
+  at the end of AGI being called (e.g. on a hangup) then set SIGHUP to 
+  be ignored within your application.
+
+* CallerID is reported with agi_callerid and agi_calleridname instead
+  of a single parameter holding both.
+
+Music On Hold:
+
+* The preferred format for musiconhold.conf has changed; please see the
+  sample configuration file for the new format. The existing format
+  is still supported but will generate warnings when the module is loaded.
+
+chan_modem:
+
+* All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
+  in this release, and will be removed in the next major Asterisk release.
+  Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
+  path for aopen and bestdata modem users.
+
+MeetMe:
+
+* The conference application now allows users to increase/decrease their
+  speaking volume and listening volume (independently of each other and 
+  other users); the 'admin' and 'user' menus have changed, and new sound 
+  files are included with this release. However, if a user calling in 
+  over a Zaptel channel that does NOT have hardware DTMF detection 
+  increases their speaking volume, it is likely they will no longer be 
+  able to enter/exit the menu or make any further adjustments, as the  
+  software DTMF detector will not be able to recognize the DTMF coming 
+  from their device.
+
+GetVar Manager Action:
+
+* Previously, the behavior of the GetVar manager action reported the value
+  of a variable in the following manner:
+   > name: value
+  This has been changed to a manner similar to the SetVar action and is now
+   > Variable: name
+   > Value: value

Zaptel-to-DAHDI.txt

@@ -1,53 +0,0 @@
-=========================================================
-=== Information for upgrading from Zaptel to DAHDI    ===
-=========================================================
-
-As announced in early 2008, Digium is renaming the Zaptel telephony
-interface project to DAHDI (Digium Asterisk Hardware Device Interface)
-to accommodate the desires of the owner of the Zaptel trademark for
-telephony purposes.
-
-This version of Asterisk can only be built using DAHDI, and as a
-result there are number of changes that will be visible to previous
-users of Asterisk with Zaptel.
-
-First, the modules that directly use services from DAHDI have been
-renamed; the new names are:
-
-  chan_zap.so -> chan_dahdi.so
-  app_zapbarge.so -> app_dahdibarge.so
-  app_zapras.so -> app_dahdiras.so
-  app_zapscan.so -> app_dahdiscan.so
-  codec_zap.so -> codec_dahdi.so
-
-Second, the behavior of many modules has changed due to the switch to
-DAHDI; the changes are listed below.
-
-chan_dahdi.so:
-
-  Incoming and outgoing channels managed by this module will be
-  'DAHDI' channels instead of 'Zap' channels.
-
-  All CLI commands that began with 'zap' have been renamed to 'dahdi'
-  commands.
-
-  All Asterisk Manager Interface (AMI) actions that began with 'Zap'
-  have changed to 'DAHDI' prefixes.
-
-  The ZapSendKeypadFacility dialplan application has been renamed to
-  DAHDISendKeypadFacility.
-
-  The configuration for the channel driver will be read from
-  /etc/asterisk/chan_dahdi.conf instead of /etc/asterisk/zapata.conf.
-
-app_dahdibarge.so:
-
-  The ZapBarge application has been renamed to DAHDIBarge.
-
-app_dahdiras.so:
-
-  The ZapRAS application has been renamed to DAHDIRAS.
-
-app_dahdiscan.so:
-
-  The ZapScan application has been renamed to DAHDIScan.

acl.c

@@ -0,0 +1,355 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Various sorts of access control
+ * 
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/time.h>
+#include <signal.h>
+#include <errno.h>
+#include <unistd.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#include <sys/socket.h>
+#include <netdb.h>
+#include <net/if.h>
+#include <netinet/in.h>
+#include <netinet/in_systm.h>
+#include <netinet/ip.h>
+#include <sys/ioctl.h>
+
+#if defined(__OpenBSD__) || defined(__NetBSD__) || defined(__FreeBSD__)
+#include <fcntl.h>
+#include <net/route.h>
+#endif
+
+#if defined(SOLARIS)
+#include <sys/sockio.h>
+#endif
+
+/* netinet/ip.h may not define the following (See RFCs 791 and 1349) */
+#if !defined(IPTOS_LOWCOST)
+#define       IPTOS_LOWCOST           0x02
+#endif
+
+#if !defined(IPTOS_MINCOST)
+#define       IPTOS_MINCOST           IPTOS_LOWCOST
+#endif
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/acl.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
+#include "asterisk/lock.h"
+#include "asterisk/srv.h"
+#include "asterisk/compat.h"
+
+#if defined(__OpenBSD__) || defined(__NetBSD__) || defined(__FreeBSD__)
+AST_MUTEX_DEFINE_STATIC(routeseq_lock);
+#endif
+
+struct ast_ha {
+	/* Host access rule */
+	struct in_addr netaddr;
+	struct in_addr netmask;
+	int sense;
+	struct ast_ha *next;
+};
+
+/* Default IP - if not otherwise set, don't breathe garbage */
+static struct in_addr __ourip = { 0x00000000 };
+
+struct my_ifreq {
+	char ifrn_name[IFNAMSIZ];	/* Interface name, e.g. "eth0", "ppp0", etc.  */
+	struct sockaddr_in ifru_addr;
+};
+
+/* Free HA structure */
+void ast_free_ha(struct ast_ha *ha)
+{
+	struct ast_ha *hal;
+	while(ha) {
+		hal = ha;
+		ha = ha->next;
+		free(hal);
+	}
+}
+
+/* Copy HA structure */
+static void ast_copy_ha(struct ast_ha *from, struct ast_ha *to)
+{
+	memcpy(&to->netaddr, &from->netaddr, sizeof(from->netaddr));
+	memcpy(&to->netmask, &from->netmask, sizeof(from->netmask));
+	to->sense = from->sense;
+}
+
+/* Create duplicate of ha structure */
+static struct ast_ha *ast_duplicate_ha(struct ast_ha *original)
+{
+	struct ast_ha *new_ha = malloc(sizeof(struct ast_ha));
+	/* Copy from original to new object */
+	ast_copy_ha(original, new_ha); 
+
+	return new_ha;
+}
+
+/* Create duplicate HA link list */
+/*  Used in chan_sip2 templates */
+struct ast_ha *ast_duplicate_ha_list(struct ast_ha *original)
+{
+	struct ast_ha *start=original;
+	struct ast_ha *ret = NULL;
+	struct ast_ha *link,*prev=NULL;
+
+	while (start) {
+		link = ast_duplicate_ha(start);  /* Create copy of this object */
+		if (prev)
+			prev->next = link;		/* Link previous to this object */
+
+		if (!ret) 
+			ret = link;		/* Save starting point */
+
+		start = start->next;		/* Go to next object */
+		prev = link;			/* Save pointer to this object */
+	}
+	return ret;    			/* Return start of list */
+}
+
+struct ast_ha *ast_append_ha(char *sense, char *stuff, struct ast_ha *path)
+{
+	struct ast_ha *ha = malloc(sizeof(struct ast_ha));
+	char *nm = "255.255.255.255";
+	char tmp[256];
+	struct ast_ha *prev = NULL;
+	struct ast_ha *ret;
+	int x, z;
+	unsigned int y;
+	ret = path;
+	while (path) {
+		prev = path;
+		path = path->next;
+	}
+	if (ha) {
+		ast_copy_string(tmp, stuff, sizeof(tmp));
+		nm = strchr(tmp, '/');
+		if (!nm) {
+			nm = "255.255.255.255";
+		} else {
+			*nm = '\0';
+			nm++;
+		}
+		if (!strchr(nm, '.')) {
+			if ((sscanf(nm, "%d", &x) == 1) && (x >= 0) && (x <= 32)) {
+				y = 0;
+				for (z=0;z<x;z++) {
+					y >>= 1;
+					y |= 0x80000000;
+				}
+				ha->netmask.s_addr = htonl(y);
+			}
+		} else if (!inet_aton(nm, &ha->netmask)) {
+			ast_log(LOG_WARNING, "%s is not a valid netmask\n", nm);
+			free(ha);
+			return ret;
+		}
+		if (!inet_aton(tmp, &ha->netaddr)) {
+			ast_log(LOG_WARNING, "%s is not a valid IP\n", tmp);
+			free(ha);
+			return ret;
+		}
+		ha->netaddr.s_addr &= ha->netmask.s_addr;
+		if (!strncasecmp(sense, "p", 1)) {
+			ha->sense = AST_SENSE_ALLOW;
+		} else {
+			ha->sense = AST_SENSE_DENY;
+		}
+		ha->next = NULL;
+		if (prev) {
+			prev->next = ha;
+		} else {
+			ret = ha;
+		}
+	}
+	ast_log(LOG_DEBUG, "%s/%s appended to acl for peer\n", stuff, nm);
+	return ret;
+}
+
+int ast_apply_ha(struct ast_ha *ha, struct sockaddr_in *sin)
+{
+	/* Start optimistic */
+	int res = AST_SENSE_ALLOW;
+	while (ha) {
+		char iabuf[INET_ADDRSTRLEN];
+		char iabuf2[INET_ADDRSTRLEN];
+		/* DEBUG */
+		ast_log(LOG_DEBUG,
+			"##### Testing %s with %s\n",
+			ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr),
+			ast_inet_ntoa(iabuf2, sizeof(iabuf2), ha->netaddr));
+		/* For each rule, if this address and the netmask = the net address
+		   apply the current rule */
+		if ((sin->sin_addr.s_addr & ha->netmask.s_addr) == ha->netaddr.s_addr)
+			res = ha->sense;
+		ha = ha->next;
+	}
+	return res;
+}
+
+int ast_get_ip_or_srv(struct sockaddr_in *sin, const char *value, const char *service)
+{
+	struct hostent *hp;
+	struct ast_hostent ahp;
+	char srv[256];
+	char host[256];
+	int tportno = ntohs(sin->sin_port);
+	if (inet_aton(value, &sin->sin_addr))
+		return 0;
+	if (service) {
+		snprintf(srv, sizeof(srv), "%s.%s", service, value);
+		if (ast_get_srv(NULL, host, sizeof(host), &tportno, srv) > 0) {
+			sin->sin_port = htons(tportno);
+			value = host;
+		}
+	}
+	hp = ast_gethostbyname(value, &ahp);
+	if (hp) {
+		memcpy(&sin->sin_addr, hp->h_addr, sizeof(sin->sin_addr));
+	} else {
+		ast_log(LOG_WARNING, "Unable to lookup '%s'\n", value);
+		return -1;
+	}
+	return 0;
+}
+
+int ast_str2tos(const char *value, int *tos)
+{
+	int fval;
+	if (sscanf(value, "%i", &fval) == 1)
+		*tos = fval & 0xff;
+	else if (!strcasecmp(value, "lowdelay"))
+		*tos = IPTOS_LOWDELAY;
+	else if (!strcasecmp(value, "throughput"))
+		*tos = IPTOS_THROUGHPUT;
+	else if (!strcasecmp(value, "reliability"))
+		*tos = IPTOS_RELIABILITY;
+	else if (!strcasecmp(value, "mincost"))
+		*tos = IPTOS_MINCOST;
+	else if (!strcasecmp(value, "none"))
+		*tos = 0;
+	else
+		return -1;
+	return 0;
+}
+
+int ast_get_ip(struct sockaddr_in *sin, const char *value)
+{
+	return ast_get_ip_or_srv(sin, value, NULL);
+}
+
+/* iface is the interface (e.g. eth0); address is the return value */
+int ast_lookup_iface(char *iface, struct in_addr *address) 
+{
+	int mysock, res = 0;
+	struct my_ifreq ifreq;
+
+	memset(&ifreq, 0, sizeof(ifreq));
+	ast_copy_string(ifreq.ifrn_name, iface, sizeof(ifreq.ifrn_name));
+
+	mysock = socket(PF_INET, SOCK_DGRAM, IPPROTO_IP);
+	res = ioctl(mysock, SIOCGIFADDR, &ifreq);
+
+	close(mysock);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to get IP of %s: %s\n", iface, strerror(errno));
+		memcpy((char *)address, (char *)&__ourip, sizeof(__ourip));
+		return -1;
+	} else {
+		memcpy((char *)address, (char *)&ifreq.ifru_addr.sin_addr, sizeof(ifreq.ifru_addr.sin_addr));
+		return 0;
+	}
+}
+
+int ast_ouraddrfor(struct in_addr *them, struct in_addr *us)
+{
+	int s;
+	struct sockaddr_in sin;
+	socklen_t slen;
+
+	s = socket(PF_INET, SOCK_DGRAM, 0);
+	if (s < 0) {
+		ast_log(LOG_WARNING, "Cannot create socket\n");
+		return -1;
+	}
+	sin.sin_family = AF_INET;
+	sin.sin_port = 5060;
+	sin.sin_addr = *them;
+	if (connect(s, (struct sockaddr *)&sin, sizeof(sin))) {
+		ast_log(LOG_WARNING, "Cannot connect\n");
+		close(s);
+		return -1;
+	}
+	slen = sizeof(sin);
+	if (getsockname(s, (struct sockaddr *)&sin, &slen)) {
+		ast_log(LOG_WARNING, "Cannot get socket name\n");
+		close(s);
+		return -1;
+	}
+	close(s);
+	*us = sin.sin_addr;
+	return 0;
+}
+
+int ast_find_ourip(struct in_addr *ourip, struct sockaddr_in bindaddr)
+{
+	char ourhost[MAXHOSTNAMELEN] = "";
+	struct ast_hostent ahp;
+	struct hostent *hp;
+	struct in_addr saddr;
+
+	/* just use the bind address if it is nonzero */
+	if (ntohl(bindaddr.sin_addr.s_addr)) {
+		memcpy(ourip, &bindaddr.sin_addr, sizeof(*ourip));
+		return 0;
+	}
+	/* try to use our hostname */
+	if (gethostname(ourhost, sizeof(ourhost) - 1)) {
+		ast_log(LOG_WARNING, "Unable to get hostname\n");
+	} else {
+		hp = ast_gethostbyname(ourhost, &ahp);
+		if (hp) {
+			memcpy(ourip, hp->h_addr, sizeof(*ourip));
+			return 0;
+		}
+	}
+	/* A.ROOT-SERVERS.NET. */
+	if (inet_aton("198.41.0.4", &saddr) && !ast_ouraddrfor(&saddr, ourip))
+		return 0;
+	return -1;
+}
+

aescrypt.c

@@ -31,26 +31,20 @@
  Issue Date: 26/08/2003
 
 */
-
 /*! \file
- *
- * \brief  This file contains the code for implementing encryption and decryption
- * for AES (Rijndael) for block and key sizes of 16, 24 and 32 bytes. It
- * can optionally be replaced by code written in assembler using NASM. For
- * further details see the file aesopt.h
- *
- * \author Dr Brian Gladman <brg@gladman.me.uk>
- */
+\brief  This file contains the code for implementing encryption and decryption
+ for AES (Rijndael) for block and key sizes of 16, 24 and 32 bytes. It
+ can optionally be replaced by code written in assembler using NASM. For
+ further details see the file aesopt.h
+*/
+
+#include "aesopt.h"
 
 #if defined(__cplusplus)
 extern "C"
 {
 #endif
 
-#ifndef HAVE_CRYPTO
-
-#include "aesopt.h"
-
 #define si(y,x,k,c) (s(y,c) = word_in(x, c) ^ (k)[c])
 #define so(y,x,c)   word_out(y, c, s(x,c))
 
@@ -314,8 +308,6 @@ aes_rval aes_decrypt(const void *in_blk, void *out_blk, const aes_decrypt_ctx cx
 
 #endif
 
-#endif /* !HAVE_CRYPTO */
-
 #if defined(__cplusplus)
 }
 #endif

aeskey.c

@@ -31,25 +31,19 @@
  Issue Date: 26/08/2003
 
 */
-
 /*! \file
- *
- *  \brief  This file contains the code for implementing the key schedule for AES
- *  (Rijndael) for block and key sizes of 16, 24, and 32 bytes. See aesopt.h
- *  for further details including optimisation.
- *
- * \author Dr Brian Gladman <brg@gladman.me.uk>
- */
+\brief  This file contains the code for implementing the key schedule for AES
+ (Rijndael) for block and key sizes of 16, 24, and 32 bytes. See aesopt.h
+ for further details including optimisation.
+*/
+
+#include "aesopt.h"
 
 #if defined(__cplusplus)
 extern "C"
 {
 #endif
 
-#ifndef HAVE_CRYPTO
-
-#include "aesopt.h"
-
 /* Initialise the key schedule from the user supplied key. The key
    length can be specified in bytes, with legal values of 16, 24
    and 32, or in bits, with legal values of 128, 192 and 256. These
@@ -466,8 +460,6 @@ aes_rval aes_decrypt_key(const void *in_key, int key_len, aes_decrypt_ctx cx[1])
 
 #endif
 
-#endif /* !HAVE_CRYPTO */
-
 #if defined(__cplusplus)
 }
 #endif

aestab.c

@@ -37,8 +37,6 @@ extern "C"
 {
 #endif
 
-#ifndef HAVE_CRYPTO
-
 #define DO_TABLES
 
 #include "aesopt.h"
@@ -228,8 +226,6 @@ void gen_tabs(void)
 
 #endif
 
-#endif /* !HAVE_CRYPTO */
-
 #if defined(__cplusplus)
 }
 #endif

agi/Makefile

@@ -3,7 +3,7 @@
 # 
 # Makefile for AGI-related stuff
 #
-# Copyright (C) 1999-2006, Digium
+# Copyright (C) 1999-2005, Digium
 #
 # Mark Spencer <markster@digium.com>
 #
@@ -11,43 +11,42 @@
 # the GNU General Public License
 #
 
-.PHONY: clean all uninstall
+AGIS=agi-test.agi eagi-test eagi-sphinx-test
 
-AGIS=agi-test.agi eagi-test eagi-sphinx-test jukebox.agi
+CFLAGS+=-DNO_AST_MM
 
+LIBS=
 ifeq ($(OSARCH),SunOS)
-  LIBS+=-lsocket -lnsl
+  LIBS=-lsocket -lnsl ../strcompat.o
 endif
 
-ifeq ($(OSARCH),mingw32)
-    AGIS:=
+ifeq ($(findstring BSD,${OSARCH}),BSD)
+  CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
 endif
 
-include $(ASTTOPDIR)/Makefile.rules
-
-_ASTCFLAGS+=-DSTANDALONE
-
-all: $(AGIS)
-
-strcompat.c: ../main/strcompat.c
-	@cp $< $@
-
-eagi-test: eagi-test.o strcompat.o
-
-eagi-sphinx-test: eagi-sphinx-test.o
+all: depend $(AGIS)
 
 install: all
 	mkdir -p $(DESTDIR)$(AGI_DIR)
 	for x in $(AGIS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(AGI_DIR) ; done
 
-uninstall:
-	for x in $(AGIS); do rm -f $(DESTDIR)$(AGI_DIR)/$$x ; done
+eagi-test: eagi-test.o
+	$(CC) $(CFLAGS) -o eagi-test eagi-test.o $(LIBS)
+
+eagi-sphinx-test: eagi-sphinx-test.o
+	$(CC) $(CFLAGS) -o eagi-sphinx-test eagi-sphinx-test.o $(LIBS) 
 
 clean:
-	rm -f *.so *.o look eagi-test eagi-sphinx-test
-	rm -f .*.d *.s *.i
-	rm -f strcompat.c
+	rm -f *.so *.o look .depend eagi-test eagi-sphinx-test
 
-ifneq ($(wildcard .*.d),)
-   include .*.d
+%.so : %.o
+	$(CC) -shared -Xlinker -x -o $@ $<
+
+ifneq ($(wildcard .depend),)
+include .depend
 endif
+
+depend: .depend
+
+.depend:
+	../build_tools/mkdep $(CFLAGS) `ls *.c`

agi/eagi-sphinx-test.c

@@ -6,15 +6,6 @@
  *
  */
 
-/*! \file
- * Extended AGI test application
- *
- * This code is released into public domain
- * without any warranty of any kind.
- *
- *	\ingroup agi
- */
-
 #include <stdio.h>
 #include <unistd.h>
 #include <stdlib.h>
@@ -26,10 +17,7 @@
 #include <netinet/in.h>
 #include <arpa/inet.h>
 #include <netdb.h>
-
-#include "asterisk.h"
-
-#include "asterisk/compat.h"
+#include <asterisk/compat.h>
 
 #define AUDIO_FILENO (STDERR_FILENO + 1)
 
@@ -79,9 +67,7 @@ static int read_environment(void)
 	char *val;
 	/* Read environment */
 	for(;;) {
-		if (!fgets(buf, sizeof(buf), stdin)) {
-			return -1;
-		}
+		fgets(buf, sizeof(buf), stdin);
 		if (feof(stdin))
 			return -1;
 		buf[strlen(buf) - 1] = '\0';
@@ -132,9 +118,7 @@ static char *wait_result(void)
 			return NULL;
 		}
 		if (FD_ISSET(STDIN_FILENO, &fds)) {
-			if (!fgets(astresp, sizeof(astresp), stdin)) {
-				return NULL;
-			}
+			fgets(astresp, sizeof(astresp), stdin);
 			if (feof(stdin)) {
 				fprintf(stderr, "Got hungup on apparently\n");
 				return NULL;
@@ -145,10 +129,9 @@ static char *wait_result(void)
 		}
 		if (FD_ISSET(AUDIO_FILENO, &fds)) {
 			res = read(AUDIO_FILENO, audiobuf, sizeof(audiobuf));
-			if ((res > 0) && (sphinx_sock > -1)) {
-				if (write(sphinx_sock, audiobuf, res) < 0) {
-					fprintf(stderr, "write() failed: %s\n", strerror(errno));
-				}
+			if (res > 0) {
+				if (sphinx_sock > -1) 
+					write(sphinx_sock, audiobuf, res);
 			}
 		}
 		if ((sphinx_sock > -1) && FD_ISSET(sphinx_sock, &fds)) {
@@ -223,7 +206,7 @@ int main(int argc, char *argv[])
 	connect_sphinx();
 	tmp = getenv("agi_enhanced");
 	if (tmp) {
-		if (sscanf(tmp, "%30d.%30d", &ver, &subver) != 2)
+		if (sscanf(tmp, "%d.%d", &ver, &subver) != 2)
 			ver = 0;
 	}
 	if (ver < 1) {

agi/eagi-test.c

@@ -5,28 +5,23 @@
  * with no warranty of any kind
  */
 
-#include "asterisk.h"
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <errno.h>
+#include <string.h>
+#include <sys/select.h>
+#include <asterisk/compat.h>
 
 #define AUDIO_FILENO (STDERR_FILENO + 1)
 
-/*! \file
- * Extended AGI test application
- *
- * This code is released into the public domain
- * with no warranty of any kind
- *
- * \ingroup agi
- */
-
 static int read_environment(void)
 {
 	char buf[256];
 	char *val;
 	/* Read environment */
 	for(;;) {
-		if (!fgets(buf, sizeof(buf), stdin)) {
-			return -1;
-		}
+		fgets(buf, sizeof(buf), stdin);
 		if (feof(stdin))
 			return -1;
 		buf[strlen(buf) - 1] = '\0';
@@ -70,9 +65,7 @@ static char *wait_result(void)
 			return NULL;
 		}
 		if (FD_ISSET(STDIN_FILENO, &fds)) {
-			if (!fgets(astresp, sizeof(astresp), stdin)) {
-				return NULL;
-			}
+			fgets(astresp, sizeof(astresp), stdin);
 			if (feof(stdin)) {
 				fprintf(stderr, "Got hungup on apparently\n");
 				return NULL;
@@ -156,7 +149,7 @@ int main(int argc, char *argv[])
 	}
 	tmp = getenv("agi_enhanced");
 	if (tmp) {
-		if (sscanf(tmp, "%30d.%30d", &ver, &subver) != 2)
+		if (sscanf(tmp, "%d.%d", &ver, &subver) != 2)
 			ver = 0;
 	}
 	if (ver < 1) {

agi/jukebox.agi

@@ -1,488 +0,0 @@
-#!/usr/bin/perl
-#
-# Jukebox 0.2
-#
-# A music manager for Asterisk.
-#
-# Copyright (C) 2005-2006, Justin Tunney
-#
-# Justin Tunney <jesuscyborg@gmail.com>
-#
-# This program is free software, distributed under the terms of the
-# GNU General Public License v2.
-#
-# Keep it open source pigs
-#
-# --------------------------------------------------------------------
-#
-# Uses festival to list off all your MP3 music files over a channel in
-# a hierarchical fashion.  Put this file in your agi-bin folder which
-# is located at: /var/lib/asterisk/agi-bin  Be sure to chmod +x it!
-#
-# Invocation Example:
-#   exten => 68742,1,Answer()
-#   exten => 68742,2,agi,jukebox.agi|/home/justin/Music
-#   exten => 68742,3,Hangup()
-#
-#   exten => 68742,1,Answer()
-#   exten => 68742,2,agi,jukebox.agi|/home/justin/Music|pm
-#   exten => 68742,3,Hangup()
-#
-# Options:
-#   p - Precache text2wave outputs for every possible filename.
-#       It is much better to set this option because if a caller
-#       presses a key during a cache operation, it will be ignored.
-#   m - Go back to menu after playing song
-#   g - Do not play the greeting message
-#
-# Usage Instructions:
-#   - Press '*' to go up a directory.  If you are in the root music
-#     folder you will be exitted from the script.
-#   - If you have a really long list of files, you can filter the list
-#     at any time by pressing '#' and spelling out a few letters you
-#     expect the files to start with.  For example, if you wanted to
-#     know what extension 'Requiem For A Dream' was, you'd type:
-#     '#737'.  Note, phone keypads don't include Q and Z.  Q is 7 and
-#     Z is 9.
-#
-# Notes:
-# - This AGI script uses the MP3Player command which uses the
-#   mpg123 Program.  Grab yourself a copy of this program by
-#   going to http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/
-#   Be sure to download mpg123-0.59r.tar.gz because it is known to
-#   work with Asterisk and hopefully isn't the release with that
-#   awful security problem.  If you're using Fedora Core 3 with
-#   Alsa like me, make linux-alsa isn't going to work.  Do make
-#   linux-devel and you're peachy keen.
-#
-# - You won't get nifty STDERR debug messages if you're using a
-#   remote asterisk shell.
-#
-# - For some reason, caching certain files will generate the
-#   error: 'using default diphone ax-ax for y-pau'.  Example:
-#   # echo "Depeche Mode - CUW - 05 - The Meaning of Love" | text2wave -o /var/jukeboxcache/jukeboxcache/Depeche_Mode/Depeche_Mode_-_CUW_-_05_-_The_Meaning_of_Love.mp3.ul -otype ulaw -
-#   The temporary work around is to just touch these files.
-#
-# - The background app doesn't like to get more than 2031 chars
-#   of input.
-#
-
-use strict;
-
-$|=1;
-
-# Setup some variables
-my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;
-my @masterCacheList = ();
-my $maxNumber = 10;
-
-while (<STDIN>) {
-	chomp;
-	last unless length($_);
-	if (/^agi_(\w+)\:\s+(.*)$/) {
-		$AGI{$1} = $2;
-	}
-}
-
-# setup options
-my $SHOWGREET = 1;
-my $PRECACHE = 0;
-my $MENUAFTERSONG = 0;
-
-$PRECACHE = 1 if $ARGV[1] =~ /p/;
-$MENUAFTERSONG = 1 if $ARGV[1] =~ /m/;
-$SHOWGREET = 0 if $ARGV[1] =~ /g/;
-
-# setup folders
-my $MUSIC = $ARGV[0];
-$MUSIC = &rmts($MUSIC);
-my $FESTIVALCACHE = "/var/jukeboxcache";
-if (! -e $FESTIVALCACHE) {
-	`mkdir -p -m0776 $FESTIVALCACHE`;
-}
-
-# make sure we have some essential files
-if (! -e "$FESTIVALCACHE/jukebox_greet.ul") {
-	`echo "Welcome to the Asterisk Jukebox" | text2wave -o $FESTIVALCACHE/jukebox_greet.ul -otype ulaw -`;
-}
-if (! -e "$FESTIVALCACHE/jukebox_press.ul") {
-	`echo "Press" | text2wave -o $FESTIVALCACHE/jukebox_press.ul -otype ulaw -`;
-}
-if (! -e "$FESTIVALCACHE/jukebox_for.ul") {
-	`echo "For" | text2wave -o $FESTIVALCACHE/jukebox_for.ul -otype ulaw -`;
-}
-if (! -e "$FESTIVALCACHE/jukebox_toplay.ul") {
-	`echo "To play" | text2wave -o $FESTIVALCACHE/jukebox_toplay.ul -otype ulaw -`;
-}
-if (! -e "$FESTIVALCACHE/jukebox_nonefound.ul") {
-	`echo "There were no music files found in this folder" | text2wave -o $FESTIVALCACHE/jukebox_nonefound.ul -otype ulaw -`;
-}
-if (! -e "$FESTIVALCACHE/jukebox_percent.ul") {
-	`echo "Percent" | text2wave -o $FESTIVALCACHE/jukebox_percent.ul -otype ulaw -`;
-}
-if (! -e "$FESTIVALCACHE/jukebox_generate.ul") {
-	`echo "Please wait while Astrisk Jukebox cashes the files of your music collection" | text2wave -o $FESTIVALCACHE/jukebox_generate.ul -otype ulaw -`;
-}
-if (! -e "$FESTIVALCACHE/jukebox_invalid.ul") {
-	`echo "You have entered an invalid selection" | text2wave -o $FESTIVALCACHE/jukebox_invalid.ul -otype ulaw -`;
-}
-if (! -e "$FESTIVALCACHE/jukebox_thankyou.ul") {
-	`echo "Thank you for using Astrisk Jukebox, Goodbye" | text2wave -o $FESTIVALCACHE/jukebox_thankyou.ul -otype ulaw -`;
-}
-
-# greet the user
-if ($SHOWGREET) {
-	print "EXEC Playback \"$FESTIVALCACHE/jukebox_greet\"\n";
-	my $result = <STDIN>; &check_result($result);
-}
-
-# go through the directories
-music_dir_cache() if $PRECACHE;
-music_dir_menu('/');
-
-exit 0;
-
-##########################################################################
-
-sub music_dir_menu {
-	my $dir = shift;
-
-# generate a list of mp3's and directories and assign each one it's
-# own selection number.  Then make sure that we've got a sound clip
-# for the file name
-	if (!opendir(THEDIR, rmts($MUSIC.$dir))) {
-		print STDERR "Failed to open music directory: $dir\n";
-		exit 1;
-	}
-	my @files = sort readdir THEDIR;
-	my $cnt = 1;
-	my @masterBgList = ();
-
-	foreach my $file (@files) {
-		chomp($file);
-		if ($file ne '.' && $file ne '..' && $file ne 'festivalcache') { # ignore special files
-			my $real_version = &rmts($MUSIC.$dir).'/'.$file;
-			my $cache_version = &rmts($FESTIVALCACHE.$dir).'/'.$file.'.ul';
-			my $cache_version2 = &rmts($FESTIVALCACHE.$dir).'/'.$file;
-			my $cache_version_esc = &clean_file($cache_version);
-			my $cache_version2_esc = &clean_file($cache_version2);
-
-			if (-d $real_version) {
-#		                                     0:id    1:type 2:text2wav-file      3:for-filtering             4:the-directory 5:text2wav echo
-				push(@masterBgList, [$cnt++, 1,     $cache_version2_esc, &remove_special_chars($file), $file,          "for the $file folder"]);
-			} elsif ($real_version =~ /\.mp3$/) {
-#		                                     0:id    1:type 2:text2wav-file      3:for-filtering             4:the-mp3
-				push(@masterBgList, [$cnt++, 2,     $cache_version2_esc, &remove_special_chars($file), $real_version,  "to play $file"]);
-			}
-		}
-	}
-	close(THEDIR);
-
-	my @filterList = @masterBgList;
-
-	if (@filterList == 0) {
-		print "EXEC Playback \"$FESTIVALCACHE/jukebox_nonefound\"\n";
-		my $result = <STDIN>; &check_result($result);
-		return 0;
-	}
-
-	for (;;) {
-MYCONTINUE:
-
-# play bg selections and figure out their selection
-		my $digit = '';
-		my $digitstr = '';
-		for (my $n=0; $n<@filterList; $n++) {
-			&cache_speech(&remove_file_extension($filterList[$n][5]), "$filterList[$n][2].ul") if ! -e "$filterList[$n][2].ul";
-			&cache_speech("Press $filterList[$n][0]", "$FESTIVALCACHE/jukebox_$filterList[$n][0].ul") if ! -e "$FESTIVALCACHE/jukebox_$filterList[$n][0].ul";
-			print "EXEC Background \"$filterList[$n][2]&$FESTIVALCACHE/jukebox_$filterList[$n][0]\"\n";
-			my $result = <STDIN>;
-			$digit = &check_result($result);
-			if ($digit > 0) {
-				$digitstr .= chr($digit);
-				last;
-			}
-		}
-		for (;;) {
-			print "WAIT FOR DIGIT 3000\n";
-			my $result = <STDIN>;
-			$digit = &check_result($result);
-			last if $digit <= 0;
-			$digitstr .= chr($digit);
-		}
-
-# see if it's a valid selection
-		print STDERR "Digits Entered: '$digitstr'\n";
-		exit 0 if $digitstr eq '';
-		my $found = 0;
-		goto EXITSUB if $digitstr =~ /\*/;
-
-# filter the list
-		if ($digitstr =~ /^\#\d+/) {
-			my $regexp = '';
-			for (my $n=1; $n<length($digitstr); $n++) {
-				my $d = substr($digitstr, $n, 1);
-				if ($d == 2) {
-					$regexp .= '[abc]';
-				} elsif ($d == 3) {
-					$regexp .= '[def]';
-				} elsif ($d == 4) {
-					$regexp .= '[ghi]';
-				} elsif ($d == 5) {
-					$regexp .= '[jkl]';
-				} elsif ($d == 6) {
-					$regexp .= '[mno]';
-				} elsif ($d == 7) {
-					$regexp .= '[pqrs]';
-				} elsif ($d == 8) {
-					$regexp .= '[tuv]';
-				} elsif ($d == 9) {
-					$regexp .= '[wxyz]';
-				}
-			}
-			@filterList = ();
-			for (my $n=1; $n<@masterBgList; $n++) {
-				push(@filterList, $masterBgList[$n]) if $masterBgList[$n][3] =~ /^$regexp/i;
-			}
-			goto MYCONTINUE;
-		}
-
-		for (my $n=0; $n<@masterBgList; $n++) {
-			if ($digitstr == $masterBgList[$n][0]) {
-				if ($masterBgList[$n][1] == 1) { # a folder
-					&music_dir_menu(rmts($dir).'/'.$masterBgList[$n][4]);
-					@filterList = @masterBgList;
-					goto MYCONTINUE;
-				} elsif ($masterBgList[$n][1] == 2) { # a file
-# because *'s scripting language is crunk and won't allow us to escape
-# funny filenames, we need to create a temporary symlink to the mp3
-# file
-					my $mp3 = &escape_file($masterBgList[$n][4]);
-					my $link = `mktemp`;
-					chomp($link);
-					$link .= '.mp3';
-					print STDERR "ln -s $mp3 $link\n";
-					my $cmdr = `ln -s $mp3 $link`;
-					chomp($cmdr);
-					print "Failed to create symlink to mp3: $cmdr\n" if $cmdr ne '';
-					
-					print "EXEC MP3Player \"$link\"\n";
-					my $result = <STDIN>; &check_result($result);
-
-					`rm $link`;
-					
-					if (!$MENUAFTERSONG) {
-						print "EXEC Playback \"$FESTIVALCACHE/jukebox_thankyou\"\n";
-						my $result = <STDIN>; &check_result($result);
-						exit 0;
-					} else {
-						goto MYCONTINUE;
-					}
-				}
-			}
-		}
-		print "EXEC Playback \"$FESTIVALCACHE/jukebox_invalid\"\n";
-		my $result = <STDIN>; &check_result($result);
-	}
-      EXITSUB:
-}
-
-sub cache_speech {
-	my $speech = shift;
-	my $file = shift;
-
-	my $theDir = extract_file_dir($file);
-	`mkdir -p -m0776 $theDir`;
-
-	print STDERR "echo \"$speech\" | text2wave -o $file -otype ulaw -\n";
-	my $cmdr = `echo "$speech" | text2wave -o $file -otype ulaw -`;
-	chomp($cmdr);
-	if ($cmdr =~ /using default diphone/) {
-# temporary bug work around....
-		`touch $file`;
-	} elsif ($cmdr ne '') {
-		print STDERR "Command Failed\n";
-		exit 1;
-	}
-}
-
-sub music_dir_cache {
-# generate list of text2speech files to generate
-	if (!music_dir_cache_genlist('/')) {
-		print STDERR "Horrible Dreadful Error: No Music Found in $MUSIC!";
-		exit 1;
-	}
-
-# add to list how many 'number' files we have to generate.  We can't
-# use the SayNumber app in Asterisk because we want to chain all
-# talking in one Background command.  We also want a consistent
-# voice...
-	for (my $n=1; $n<=$maxNumber; $n++) {
-		push(@masterCacheList, [3, "Press $n", "$FESTIVALCACHE/jukebox_$n.ul"]) if ! -e "$FESTIVALCACHE/jukebox_$n.ul";
-	}
-
-# now generate all these darn text2speech files
-	if (@masterCacheList > 5) {
-		print "EXEC Playback \"$FESTIVALCACHE/jukebox_generate\"\n";
-		my $result = <STDIN>; &check_result($result);
-	}
-	my $theTime = time();
-	for (my $n=0; $n < @masterCacheList; $n++) {
-		my $cmdr = '';
-		if ($masterCacheList[$n][0] == 1) { # directory
-			&cache_speech("for folder $masterCacheList[$n][1]", $masterCacheList[$n][2]);
-		} elsif ($masterCacheList[$n][0] == 2) { # file
-			&cache_speech("to play $masterCacheList[$n][1]", $masterCacheList[$n][2]);
-		} elsif ($masterCacheList[$n][0] == 3) { # number
-			&cache_speech($masterCacheList[$n][1], $masterCacheList[$n][2]);
-		}
-		if (time() >= $theTime + 30) {
-			my $percent = int($n / @masterCacheList * 100);
-			print "SAY NUMBER $percent \"\"\n";
-			my $result = <STDIN>; &check_result($result);
-			print "EXEC Playback \"$FESTIVALCACHE/jukebox_percent\"\n";
-			my $result = <STDIN>; &check_result($result);
-			$theTime = time();
-		}
-	}
-}
-
-# this function will fill the @masterCacheList of all the files that
-# need to have text2speeced ulaw files of their names generated
-sub music_dir_cache_genlist {
-	my $dir = shift;
-	if (!opendir(THEDIR, rmts($MUSIC.$dir))) {
-		print STDERR "Failed to open music directory: $dir\n";
-		exit 1;
-	}
-	my @files = sort readdir THEDIR;
-	my $foundFiles = 0;
-	my $tmpMaxNum = 0;
-	foreach my $file (@files) {
-		chomp;
-		if ($file ne '.' && $file ne '..' && $file ne 'festivalcache') { # ignore special files
-			my $real_version = &rmts($MUSIC.$dir).'/'.$file;
-			my $cache_version = &rmts($FESTIVALCACHE.$dir).'/'.$file.'.ul';
-			my $cache_version2 = &rmts($FESTIVALCACHE.$dir).'/'.$file;
-			my $cache_version_esc = &clean_file($cache_version);
-			my $cache_version2_esc = &clean_file($cache_version2);
-
-			if (-d $real_version) {
-				if (music_dir_cache_genlist(rmts($dir).'/'.$file)) {
-					$tmpMaxNum++;
-					$maxNumber = $tmpMaxNum if $tmpMaxNum > $maxNumber;
-					push(@masterCacheList, [1, $file, $cache_version_esc]) if ! -e $cache_version_esc;
-					$foundFiles = 1;
-				}
-			} elsif ($real_version =~ /\.mp3$/) {
-				$tmpMaxNum++;
-				$maxNumber = $tmpMaxNum if $tmpMaxNum > $maxNumber;
-				push(@masterCacheList, [2, &remove_file_extension($file), $cache_version_esc]) if ! -e $cache_version_esc;
-				$foundFiles = 1;
-			}
-		}
-	}
-	close(THEDIR);
-	return $foundFiles;
-}
-
-sub rmts { # remove trailing slash
-	my $hog = shift;
-	$hog =~ s/\/$//;
-	return $hog;
-}
-
-sub extract_file_name {
-	my $hog = shift;
-	$hog =~ /\/?([^\/]+)$/;
-	return $1;
-}
-
-sub extract_file_dir {
-	my $hog = shift;
-	return $hog if ! ($hog =~ /\//);
-	$hog =~ /(.*)\/[^\/]*$/;
-	return $1;
-}
-
-sub remove_file_extension {
-	my $hog = shift;
-	return $hog if ! ($hog =~ /\./);
-	$hog =~ /(.*)\.[^.]*$/;
-	return $1;
-}
-
-sub clean_file {
-	my $hog = shift;
-	$hog =~ s/\\/_/g;
-	$hog =~ s/ /_/g;
-	$hog =~ s/\t/_/g;
-	$hog =~ s/\'/_/g;
-	$hog =~ s/\"/_/g;
-	$hog =~ s/\(/_/g;
-	$hog =~ s/\)/_/g;
-	$hog =~ s/&/_/g;
-	$hog =~ s/\[/_/g;
-	$hog =~ s/\]/_/g;
-	$hog =~ s/\$/_/g;
-	$hog =~ s/\|/_/g;
-	$hog =~ s/\^/_/g;
-	return $hog;
-}
-
-sub remove_special_chars {
-	my $hog = shift;
-	$hog =~ s/\\//g;
-	$hog =~ s/ //g;
-	$hog =~ s/\t//g;
-	$hog =~ s/\'//g;
-	$hog =~ s/\"//g;
-	$hog =~ s/\(//g;
-	$hog =~ s/\)//g;
-	$hog =~ s/&//g;
-	$hog =~ s/\[//g;
-	$hog =~ s/\]//g;
-	$hog =~ s/\$//g;
-	$hog =~ s/\|//g;
-	$hog =~ s/\^//g;
-	return $hog;
-}
-
-sub escape_file {
-	my $hog = shift;
-	$hog =~ s/\\/\\\\/g;
-	$hog =~ s/ /\\ /g;
-	$hog =~ s/\t/\\\t/g;
-	$hog =~ s/\'/\\\'/g;
-	$hog =~ s/\"/\\\"/g;
-	$hog =~ s/\(/\\\(/g;
-	$hog =~ s/\)/\\\)/g;
-	$hog =~ s/&/\\&/g;
-	$hog =~ s/\[/\\\[/g;
-	$hog =~ s/\]/\\\]/g;
-	$hog =~ s/\$/\\\$/g;
-	$hog =~ s/\|/\\\|/g;
-	$hog =~ s/\^/\\\^/g;
-	return $hog;
-}
-
-sub check_result {
-	my ($res) = @_;
-	my $retval;
-	$tests++;
-	chomp $res;
-	if ($res =~ /^200/) {
-		$res =~ /result=(-?\d+)/;
-		if (!length($1)) {
-			print STDERR "FAIL ($res)\n";
-			$fail++;
-			exit 1;
-		} else {
-			print STDERR "PASS ($1)\n";
-			return $1;
-		}
-	} else {
-		print STDERR "FAIL (unexpected result '$res')\n";
-		exit 1;
-	}
-}

alaw.c

@@ -0,0 +1,100 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief u-Law to Signed linear conversion
+ * 
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/alaw.h"
+
+#define AMI_MASK 0x55
+
+static inline unsigned char linear2alaw (short int linear)
+{
+    int mask;
+    int seg;
+    int pcm_val;
+    static int seg_end[8] =
+    {
+         0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF, 0x1FFF, 0x3FFF, 0x7FFF
+    };
+    
+    pcm_val = linear;
+    if (pcm_val >= 0)
+    {
+        /* Sign (7th) bit = 1 */
+        mask = AMI_MASK | 0x80;
+    }
+    else
+    {
+        /* Sign bit = 0 */
+        mask = AMI_MASK;
+        pcm_val = -pcm_val;
+    }
+
+    /* Convert the scaled magnitude to segment number. */
+    for (seg = 0;  seg < 8;  seg++)
+    {
+        if (pcm_val <= seg_end[seg])
+	    break;
+    }
+    /* Combine the sign, segment, and quantization bits. */
+    return  ((seg << 4) | ((pcm_val >> ((seg)  ?  (seg + 3)  :  4)) & 0x0F)) ^ mask;
+}
+/*- End of function --------------------------------------------------------*/
+
+static inline short int alaw2linear (unsigned char alaw)
+{
+    int i;
+    int seg;
+
+    alaw ^= AMI_MASK;
+    i = ((alaw & 0x0F) << 4);
+    seg = (((int) alaw & 0x70) >> 4);
+    if (seg)
+        i = (i + 0x100) << (seg - 1);
+    return (short int) ((alaw & 0x80)  ?  i  :  -i);
+}
+
+unsigned char __ast_lin2a[8192];
+short __ast_alaw[256];
+
+void ast_alaw_init(void)
+{
+	int i;
+	/* 
+	 *  Set up mu-law conversion table
+	 */
+	for(i = 0;i < 256;i++)
+	   {
+	        __ast_alaw[i] = alaw2linear(i);
+	   }
+	  /* set up the reverse (mu-law) conversion table */
+	for(i = -32768; i < 32768; i++)
+	   {
+		__ast_lin2a[((unsigned short)i) >> 3] = linear2alaw(i);
+	   }
+
+}
+

apps/Makefile

@@ -3,33 +3,128 @@
 # 
 # Makefile for PBX applications
 #
-# Copyright (C) 1999-2006, Digium, Inc.
+# Copyright (C) 1999-2005, Digium
+#
+# Mark Spencer <markster@digium.com>
 #
 # This program is free software, distributed under the terms of
 # the GNU General Public License
 #
 
--include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
+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_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_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_stack.so
 
-MODULE_PREFIX=app
-MENUSELECT_CATEGORY=APPS
-MENUSELECT_DESCRIPTION=Applications
+#
+# Obsolete things...
+#
+#APPS+=app_sql_postgres.so
+#APPS+=app_sql_odbc.so
 
-MENUSELECT_OPTS_app_directory:=$(MENUSELECT_OPTS_app_voicemail)
-ifneq ($(findstring ODBC_STORAGE,$(MENUSELECT_OPTS_app_voicemail)),)
-  MENUSELECT_DEPENDS_app_voicemail+=$(MENUSELECT_DEPENDS_ODBC_STORAGE)
-  MENUSELECT_DEPENDS_app_directory+=$(MENUSELECT_DEPENDS_ODBC_STORAGE)
+#
+# Experimental things
+#
+#APPS+=app_ivrdemo.so
+#APPS+=app_skel.so
+#APPS+=app_rpt.so
+
+ifndef WITHOUT_ZAPTEL
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/zaptel.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/zaptel.h),)
+  APPS+=app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so app_page.so
 endif
-ifneq ($(findstring IMAP_STORAGE,$(MENUSELECT_OPTS_app_voicemail)),)
-  MENUSELECT_DEPENDS_app_voicemail+=$(MENUSELECT_DEPENDS_IMAP_STORAGE)
+endif # WITHOUT_ZAPTEL
+
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/osp/osp.h $(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
+APPS+=app_osplookup.so
 endif
 
-all: _all
-
-include $(ASTTOPDIR)/Makefile.moddir_rules
-
-ifneq ($(findstring $(OSARCH), mingw32 cygwin ),)
-  LIBS+= -lres_features.so -lres_ael_share.so -lres_monitor.so -lres_speech.so
-  LIBS+= -lres_smdi.so
+ifeq ($(findstring BSD,${OSARCH}),BSD)
+CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
 endif
 
+CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
+ifneq ($(shell if [[ 0x`$(CROSS_COMPILE_BIN)curl-config --vernum` -ge 0x70907 ]]; then echo "OK" ; fi),)
+  ifneq (${CURLLIBS},)
+    APPS+=app_curl.so
+  endif 
+endif
+
+ifeq (${OSARCH},CYGWIN)
+CYGSOLINK=-Wl,--out-implib=lib$@.a -Wl,--export-all-symbols
+CYGSOLIB=-L.. -L. -L../res -lasterisk.dll -lres_features.so -lres_adsi.so -lres_monitor.so
+else
+CFLAGS+=-fPIC
+APPS+=app_sms.so
+endif
+
+#
+# If you have UnixODBC you can use ODBC voicemail
+# storage
+#
+# Uncomment to use ODBC storage
+#CFLAGS+=-DUSE_ODBC_STORAGE
+# Uncomment for extended ODBC voicemail storage
+#CFLAGS+=-DEXTENDED_ODBC_STORAGE
+# See doc/README.odbcstorage for more information
+
+all: $(APPS)
+
+clean:
+	rm -f *.so *.o .depend
+
+%.so : %.o
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB}
+
+app_rpt.so : app_rpt.o
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -ltonezone
+
+install: all
+	for x in $(APPS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done
+	rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
+	rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
+
+app_curl.so: app_curl.o
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} $(CURLLIBS)
+
+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 $@ ${CYGSOLINK} $< ${CYGSOLIB} -L/usr/local/pgsql/lib -lpq
+
+app_sql_odbc.so: app_sql_odbc.o
+	$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -lodbc
+
+ifeq (SunOS,$(shell uname))
+app_chanspy.so: app_chanspy.o
+	$(CC) $(SOLINK) -o $@ $< -lrt
+endif
+
+
+ifneq ($(wildcard .depend),)
+ include .depend
+endif
+
+depend: .depend
+
+.depend:
+	../build_tools/mkdep $(CFLAGS) `ls *.c`
+
+env:
+	env

apps/app_amd.c

@@ -1,451 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2003 - 2006, Aheeva Technology.
- *
- * Claude Klimos (claude.klimos@aheeva.com)
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- *
- * A license has been granted to Digium (via disclaimer) for the use of
- * this code.
- */
-
-/*! \file
- *
- * \brief Answering machine detection
- *
- * \author Claude Klimos (claude.klimos@aheeva.com)
- */
-
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/module.h"
-#include "asterisk/lock.h"
-#include "asterisk/channel.h"
-#include "asterisk/dsp.h"
-#include "asterisk/pbx.h"
-#include "asterisk/config.h"
-#include "asterisk/app.h"
-
-
-static char *app = "AMD";
-static char *synopsis = "Attempts to detect answering machines";
-static char *descrip =
-"  AMD([initialSilence],[greeting],[afterGreetingSilence],[totalAnalysisTime]\n"
-"      ,[minimumWordLength],[betweenWordsSilence],[maximumNumberOfWords]\n"
-"      ,[silenceThreshold],[|maximumWordLength])\n"
-"  This application attempts to detect answering machines at the beginning\n"
-"  of outbound calls.  Simply call this application after the call\n"
-"  has been answered (outbound only, of course).\n"
-"  When loaded, AMD reads amd.conf and uses the parameters specified as\n"
-"  default values. Those default values get overwritten when calling AMD\n"
-"  with parameters.\n"
-"- 'initialSilence' is the maximum silence duration before the greeting. If\n"
-"   exceeded then MACHINE.\n"
-"- 'greeting' is the maximum length of a greeting. If exceeded then MACHINE.\n"
-"- 'afterGreetingSilence' is the silence after detecting a greeting.\n"
-"   If exceeded then HUMAN.\n"
-"- 'totalAnalysisTime' is the maximum time allowed for the algorithm to decide\n"
-"   on a HUMAN or MACHINE.\n"
-"- 'minimumWordLength'is the minimum duration of Voice to considered as a word.\n"
-"- 'betweenWordsSilence' is the minimum duration of silence after a word to \n"
-"   consider the audio that follows as a new word.\n"
-"- 'maximumNumberOfWords'is the maximum number of words in the greeting. \n"
-"   If exceeded then MACHINE.\n"
-"- 'silenceThreshold' is the silence threshold.\n"
-"- 'maximumWordLength' is the maximum duration of a word to accept. If exceeded then MACHINE\n"
-"This application sets the following channel variables upon completion:\n"
-"    AMDSTATUS - This is the status of the answering machine detection.\n"
-"                Possible values are:\n"
-"                MACHINE | HUMAN | NOTSURE | HANGUP\n"
-"    AMDCAUSE - Indicates the cause that led to the conclusion.\n"
-"               Possible values are:\n"
-"               TOOLONG-<%d total_time>\n"
-"               INITIALSILENCE-<%d silenceDuration>-<%d initialSilence>\n"
-"               HUMAN-<%d silenceDuration>-<%d afterGreetingSilence>\n"
-"               MAXWORDS-<%d wordsCount>-<%d maximumNumberOfWords>\n"
-"               LONGGREETING-<%d voiceDuration>-<%d greeting>\n"
-"               MAXWORDLENGTH-<%d consecutiveVoiceDuration>\n";
-
-#define STATE_IN_WORD       1
-#define STATE_IN_SILENCE    2
-
-/* Some default values for the algorithm parameters. These defaults will be overwritten from amd.conf */
-static int dfltInitialSilence       = 2500;
-static int dfltGreeting             = 1500;
-static int dfltAfterGreetingSilence = 800;
-static int dfltTotalAnalysisTime    = 5000;
-static int dfltMinimumWordLength    = 100;
-static int dfltBetweenWordsSilence  = 50;
-static int dfltMaximumNumberOfWords = 3;
-static int dfltSilenceThreshold     = 256;
-static int dfltMaximumWordLength    = 5000; /* Setting this to a large default so it is not used unless specify it in the configs or command line */
-
-/* Set to the lowest ms value provided in amd.conf or application parameters */
-static int dfltMaxWaitTimeForFrame  = 50;
-
-static void isAnsweringMachine(struct ast_channel *chan, void *data)
-{
-	int res = 0;
-	struct ast_frame *f = NULL;
-	struct ast_dsp *silenceDetector = NULL;
-	int dspsilence = 0, readFormat, framelength = 0;
-	int inInitialSilence = 1;
-	int inGreeting = 0;
-	int voiceDuration = 0;
-	int silenceDuration = 0;
-	int iTotalTime = 0;
-	int iWordsCount = 0;
-	int currentState = STATE_IN_WORD;
-	int previousState = STATE_IN_SILENCE;
-	int consecutiveVoiceDuration = 0;
-	char amdCause[256] = "", amdStatus[256] = "";
-	char *parse = ast_strdupa(data);
-
-	/* Lets set the initial values of the variables that will control the algorithm.
-	   The initial values are the default ones. If they are passed as arguments
-	   when invoking the application, then the default values will be overwritten
-	   by the ones passed as parameters. */
-	int initialSilence       = dfltInitialSilence;
-	int greeting             = dfltGreeting;
-	int afterGreetingSilence = dfltAfterGreetingSilence;
-	int totalAnalysisTime    = dfltTotalAnalysisTime;
-	int minimumWordLength    = dfltMinimumWordLength;
-	int betweenWordsSilence  = dfltBetweenWordsSilence;
-	int maximumNumberOfWords = dfltMaximumNumberOfWords;
-	int silenceThreshold     = dfltSilenceThreshold;
-	int maximumWordLength    = dfltMaximumWordLength;
-	int maxWaitTimeForFrame  = dfltMaxWaitTimeForFrame;
-
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(argInitialSilence);
-		AST_APP_ARG(argGreeting);
-		AST_APP_ARG(argAfterGreetingSilence);
-		AST_APP_ARG(argTotalAnalysisTime);
-		AST_APP_ARG(argMinimumWordLength);
-		AST_APP_ARG(argBetweenWordsSilence);
-		AST_APP_ARG(argMaximumNumberOfWords);
-		AST_APP_ARG(argSilenceThreshold);
-		AST_APP_ARG(argMaximumWordLength);
-	);
-
-	ast_verb(3, "AMD: %s %s %s (Fmt: %d)\n", chan->name ,chan->cid.cid_ani, chan->cid.cid_rdnis, chan->readformat);
-
-	/* Lets parse the arguments. */
-	if (!ast_strlen_zero(parse)) {
-		/* Some arguments have been passed. Lets parse them and overwrite the defaults. */
-		AST_STANDARD_APP_ARGS(args, parse);
-		if (!ast_strlen_zero(args.argInitialSilence))
-			initialSilence = atoi(args.argInitialSilence);
-		if (!ast_strlen_zero(args.argGreeting))
-			greeting = atoi(args.argGreeting);
-		if (!ast_strlen_zero(args.argAfterGreetingSilence))
-			afterGreetingSilence = atoi(args.argAfterGreetingSilence);
-		if (!ast_strlen_zero(args.argTotalAnalysisTime))
-			totalAnalysisTime = atoi(args.argTotalAnalysisTime);
-		if (!ast_strlen_zero(args.argMinimumWordLength))
-			minimumWordLength = atoi(args.argMinimumWordLength);
-		if (!ast_strlen_zero(args.argBetweenWordsSilence))
-			betweenWordsSilence = atoi(args.argBetweenWordsSilence);
-		if (!ast_strlen_zero(args.argMaximumNumberOfWords))
-			maximumNumberOfWords = atoi(args.argMaximumNumberOfWords);
-		if (!ast_strlen_zero(args.argSilenceThreshold))
-			silenceThreshold = atoi(args.argSilenceThreshold);
-		if (!ast_strlen_zero(args.argMaximumWordLength))
-			maximumWordLength = atoi(args.argMaximumWordLength);
-	} else {
-		ast_debug(1, "AMD using the default parameters.\n");
-	}
-
-	/* Find lowest ms value, that will be max wait time for a frame */
-	if (maxWaitTimeForFrame > initialSilence)
-		maxWaitTimeForFrame = initialSilence;
-	if (maxWaitTimeForFrame > greeting)
-		maxWaitTimeForFrame = greeting;
-	if (maxWaitTimeForFrame > afterGreetingSilence)
-		maxWaitTimeForFrame = afterGreetingSilence;
-	if (maxWaitTimeForFrame > totalAnalysisTime)
-		maxWaitTimeForFrame = totalAnalysisTime;
-	if (maxWaitTimeForFrame > minimumWordLength)
-		maxWaitTimeForFrame = minimumWordLength;
-	if (maxWaitTimeForFrame > betweenWordsSilence)
-		maxWaitTimeForFrame = betweenWordsSilence;
-
-	/* Now we're ready to roll! */
-	ast_verb(3, "AMD: initialSilence [%d] greeting [%d] afterGreetingSilence [%d] "
-		"totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silenceThreshold [%d] maximumWordLength [%d] \n",
-				initialSilence, greeting, afterGreetingSilence, totalAnalysisTime,
-				minimumWordLength, betweenWordsSilence, maximumNumberOfWords, silenceThreshold, maximumWordLength);
-
-	/* Set read format to signed linear so we get signed linear frames in */
-	readFormat = chan->readformat;
-	if (ast_set_read_format(chan, AST_FORMAT_SLINEAR) < 0 ) {
-		ast_log(LOG_WARNING, "AMD: Channel [%s]. Unable to set to linear mode, giving up\n", chan->name );
-		pbx_builtin_setvar_helper(chan , "AMDSTATUS", "");
-		pbx_builtin_setvar_helper(chan , "AMDCAUSE", "");
-		return;
-	}
-
-	/* Create a new DSP that will detect the silence */
-	if (!(silenceDetector = ast_dsp_new())) {
-		ast_log(LOG_WARNING, "AMD: Channel [%s]. Unable to create silence detector :(\n", chan->name );
-		pbx_builtin_setvar_helper(chan , "AMDSTATUS", "");
-		pbx_builtin_setvar_helper(chan , "AMDCAUSE", "");
-		return;
-	}
-
-	/* Set silence threshold to specified value */
-	ast_dsp_set_threshold(silenceDetector, silenceThreshold);
-
-	/* Now we go into a loop waiting for frames from the channel */
-	while ((res = ast_waitfor(chan, 2 * maxWaitTimeForFrame)) > -1) {
-
-		/* If we fail to read in a frame, that means they hung up */
-		if (!(f = ast_read(chan))) {
-			ast_verb(3, "AMD: Channel [%s]. HANGUP\n", chan->name);
-			ast_debug(1, "Got hangup\n");
-			strcpy(amdStatus, "HANGUP");
-			res = 1;
-			break;
-		}
-
-		if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_NULL || f->frametype == AST_FRAME_CNG) {
-			/* If the total time exceeds the analysis time then give up as we are not too sure */
-			if (f->frametype == AST_FRAME_VOICE)
-				framelength = (ast_codec_get_samples(f) / DEFAULT_SAMPLES_PER_MS);
-			else
-				framelength += 2 * maxWaitTimeForFrame;
-
-			iTotalTime += framelength;
-			if (iTotalTime >= totalAnalysisTime) {
-				ast_verb(3, "AMD: Channel [%s]. Too long...\n", chan->name );
-				ast_frfree(f);
-				strcpy(amdStatus , "NOTSURE");
-				sprintf(amdCause , "TOOLONG-%d", iTotalTime);
-				break;
-			}
-
-			/* Feed the frame of audio into the silence detector and see if we get a result */
-			if (f->frametype != AST_FRAME_VOICE)
-				dspsilence += 2 * maxWaitTimeForFrame;
-			else {
-				dspsilence = 0;
-				ast_dsp_silence(silenceDetector, f, &dspsilence);
-			}
-
-			if (dspsilence > 0) {
-				silenceDuration = dspsilence;
-				
-				if (silenceDuration >= betweenWordsSilence) {
-					if (currentState != STATE_IN_SILENCE ) {
-						previousState = currentState;
-						ast_verb(3, "AMD: Channel [%s]. Changed state to STATE_IN_SILENCE\n", chan->name);
-					}
-					/* Find words less than word duration */
-					if (consecutiveVoiceDuration < minimumWordLength && consecutiveVoiceDuration > 0){
-						ast_verb(3, "AMD: Channel [%s]. Short Word Duration: %d\n", chan->name, consecutiveVoiceDuration);
-					}
-					currentState  = STATE_IN_SILENCE;
-					consecutiveVoiceDuration = 0;
-				}
-
-				if (inInitialSilence == 1  && silenceDuration >= initialSilence) {
-					ast_verb(3, "AMD: Channel [%s]. ANSWERING MACHINE: silenceDuration:%d initialSilence:%d\n",
-						chan->name, silenceDuration, initialSilence);
-					ast_frfree(f);
-					strcpy(amdStatus , "MACHINE");
-					sprintf(amdCause , "INITIALSILENCE-%d-%d", silenceDuration, initialSilence);
-					res = 1;
-					break;
-				}
-				
-				if (silenceDuration >= afterGreetingSilence  &&  inGreeting == 1) {
-					ast_verb(3, "AMD: Channel [%s]. HUMAN: silenceDuration:%d afterGreetingSilence:%d\n",
-						chan->name, silenceDuration, afterGreetingSilence);
-					ast_frfree(f);
-					strcpy(amdStatus , "HUMAN");
-					sprintf(amdCause , "HUMAN-%d-%d", silenceDuration, afterGreetingSilence);
-					res = 1;
-					break;
-				}
-				
-			} else {
-				consecutiveVoiceDuration += framelength;
-				voiceDuration += framelength;
-
-				/* If I have enough consecutive voice to say that I am in a Word, I can only increment the
-				   number of words if my previous state was Silence, which means that I moved into a word. */
-				if (consecutiveVoiceDuration >= minimumWordLength && currentState == STATE_IN_SILENCE) {
-					iWordsCount++;
-					ast_verb(3, "AMD: Channel [%s]. Word detected. iWordsCount:%d\n", chan->name, iWordsCount);
-					previousState = currentState;
-					currentState = STATE_IN_WORD;
-				}
-				if (consecutiveVoiceDuration >= maximumWordLength){
-					ast_verb(3, "AMD: Channel [%s]. Maximum Word Length detected. [%d]\n", chan->name, consecutiveVoiceDuration);
-					ast_frfree(f);
-					strcpy(amdStatus , "MACHINE");
-					sprintf(amdCause , "MAXWORDLENGTH-%d", consecutiveVoiceDuration);
-					break;
-				}
-				if (iWordsCount >= maximumNumberOfWords) {
-					ast_verb(3, "AMD: Channel [%s]. ANSWERING MACHINE: iWordsCount:%d\n", chan->name, iWordsCount);
-					ast_frfree(f);
-					strcpy(amdStatus , "MACHINE");
-					sprintf(amdCause , "MAXWORDS-%d-%d", iWordsCount, maximumNumberOfWords);
-					res = 1;
-					break;
-				}
-
-				if (inGreeting == 1 && voiceDuration >= greeting) {
-					ast_verb(3, "AMD: Channel [%s]. ANSWERING MACHINE: voiceDuration:%d greeting:%d\n", chan->name, voiceDuration, greeting);
-					ast_frfree(f);
-					strcpy(amdStatus , "MACHINE");
-					sprintf(amdCause , "LONGGREETING-%d-%d", voiceDuration, greeting);
-					res = 1;
-					break;
-				}
-
-				if (voiceDuration >= minimumWordLength ) {
-					if (silenceDuration > 0)
-						ast_verb(3, "AMD: Channel [%s]. Detected Talk, previous silence duration: %d\n", chan->name, silenceDuration);
-					silenceDuration = 0;
-				}
-				if (consecutiveVoiceDuration >= minimumWordLength && inGreeting == 0) {
-					/* Only go in here once to change the greeting flag when we detect the 1st word */
-					if (silenceDuration > 0)
-						ast_verb(3, "AMD: Channel [%s]. Before Greeting Time:  silenceDuration: %d voiceDuration: %d\n", chan->name, silenceDuration, voiceDuration);
-					inInitialSilence = 0;
-					inGreeting = 1;
-				}
-				
-			}
-		}
-		ast_frfree(f);
-	}
-	
-	if (!res) {
-		/* It took too long to get a frame back. Giving up. */
-		ast_verb(3, "AMD: Channel [%s]. Too long...\n", chan->name);
-		strcpy(amdStatus , "NOTSURE");
-		sprintf(amdCause , "TOOLONG-%d", iTotalTime);
-	}
-
-	/* Set the status and cause on the channel */
-	pbx_builtin_setvar_helper(chan , "AMDSTATUS" , amdStatus);
-	pbx_builtin_setvar_helper(chan , "AMDCAUSE" , amdCause);
-
-	/* Restore channel read format */
-	if (readFormat && ast_set_read_format(chan, readFormat))
-		ast_log(LOG_WARNING, "AMD: Unable to restore read format on '%s'\n", chan->name);
-
-	/* Free the DSP used to detect silence */
-	ast_dsp_free(silenceDetector);
-
-	return;
-}
-
-
-static int amd_exec(struct ast_channel *chan, void *data)
-{
-	isAnsweringMachine(chan, data);
-
-	return 0;
-}
-
-static int load_config(int reload)
-{
-	struct ast_config *cfg = NULL;
-	char *cat = NULL;
-	struct ast_variable *var = NULL;
-	struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
-
-	if (!(cfg = ast_config_load("amd.conf", config_flags))) {
-		ast_log(LOG_ERROR, "Configuration file amd.conf missing.\n");
-		return -1;
-	} else if (cfg == CONFIG_STATUS_FILEUNCHANGED)
-		return 0;
-
-	cat = ast_category_browse(cfg, NULL);
-
-	while (cat) {
-		if (!strcasecmp(cat, "general") ) {
-			var = ast_variable_browse(cfg, cat);
-			while (var) {
-				if (!strcasecmp(var->name, "initial_silence")) {
-					dfltInitialSilence = atoi(var->value);
-				} else if (!strcasecmp(var->name, "greeting")) {
-					dfltGreeting = atoi(var->value);
-				} else if (!strcasecmp(var->name, "after_greeting_silence")) {
-					dfltAfterGreetingSilence = atoi(var->value);
-				} else if (!strcasecmp(var->name, "silence_threshold")) {
-					dfltSilenceThreshold = atoi(var->value);
-				} else if (!strcasecmp(var->name, "total_analysis_time")) {
-					dfltTotalAnalysisTime = atoi(var->value);
-				} else if (!strcasecmp(var->name, "min_word_length")) {
-					dfltMinimumWordLength = atoi(var->value);
-				} else if (!strcasecmp(var->name, "between_words_silence")) {
-					dfltBetweenWordsSilence = atoi(var->value);
-				} else if (!strcasecmp(var->name, "maximum_number_of_words")) {
-					dfltMaximumNumberOfWords = atoi(var->value);
-				} else if (!strcasecmp(var->name, "maximum_word_length")) {
-					dfltMaximumWordLength = atoi(var->value);
-
-				} else {
-					ast_log(LOG_WARNING, "%s: Cat:%s. Unknown keyword %s at line %d of amd.conf\n",
-						app, cat, var->name, var->lineno);
-				}
-				var = var->next;
-			}
-		}
-		cat = ast_category_browse(cfg, cat);
-	}
-
-	ast_config_destroy(cfg);
-
-	ast_verb(3, "AMD defaults: initialSilence [%d] greeting [%d] afterGreetingSilence [%d] "
-		"totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silenceThreshold [%d] maximumWordLength [%d]\n",
-		dfltInitialSilence, dfltGreeting, dfltAfterGreetingSilence, dfltTotalAnalysisTime,
-		dfltMinimumWordLength, dfltBetweenWordsSilence, dfltMaximumNumberOfWords, dfltSilenceThreshold, dfltMaximumWordLength);
-
-	return 0;
-}
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
-{
-	if (load_config(0))
-		return AST_MODULE_LOAD_DECLINE;
-	if (ast_register_application(app, amd_exec, synopsis, descrip))
-		return AST_MODULE_LOAD_FAILURE;
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
-static int reload(void)
-{
-	if (load_config(1))
-		return AST_MODULE_LOAD_DECLINE;
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Answering Machine Detection Application",
-		.load = load_module,
-		.unload = unload_module,
-		.reload = reload,
-);

apps/app_authenticate.c

@@ -17,20 +17,24 @@
  */
 
 /*! \file
- *
  * \brief Execute arbitrary authenticate commands
- *
- * \author Mark Spencer <markster@digium.com>
- *
+ * 
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+#include <stdio.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
@@ -38,189 +42,191 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/astdb.h"
 #include "asterisk/utils.h"
 
-enum {
-	OPT_ACCOUNT = (1 << 0),
-	OPT_DATABASE = (1 << 1),
-	OPT_MULTIPLE = (1 << 3),
-	OPT_REMOVE = (1 << 4),
-} auth_option_flags;
-
-AST_APP_OPTIONS(auth_app_options, {
-	AST_APP_OPTION('a', OPT_ACCOUNT),
-	AST_APP_OPTION('d', OPT_DATABASE),
-	AST_APP_OPTION('m', OPT_MULTIPLE),
-	AST_APP_OPTION('r', OPT_REMOVE),
-});
-
+static char *tdesc = "Authentication Application";
 
 static char *app = "Authenticate";
 
 static char *synopsis = "Authenticate a user";
 
 static char *descrip =
-"  Authenticate(password[,options[,maxdigits]]): This application asks the caller\n"
-"to enter a given password in order to continue dialplan execution. If the password\n"
-"begins with the '/' character, it is interpreted as a file which contains a list of\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.\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"
-"     maxdigits  - maximum acceptable number of digits. Stops reading after\n"
-"         maxdigits have been entered (without requiring the user to\n"
-"         press the '#' key).\n"
-"         Defaults to 0 - no limit - wait for the user press the '#' key.\n"
 ;
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
 static int auth_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0, retries, maxdigits;
-	char passwd[256], *prompt = "agent-pass", *argcopy = NULL;
-	struct ast_flags flags = {0};
-
-	AST_DECLARE_APP_ARGS(arglist,
-		AST_APP_ARG(password);
-		AST_APP_ARG(options);
-		AST_APP_ARG(maxdigits);
-	);
-
+	int res=0;
+	int jump = 0;
+	int retries;
+	struct localuser *u;
+	char password[256]="";
+	char passwd[256];
+	char *opts;
+	char *prompt;
+	
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Authenticate requires an argument(password)\n");
 		return -1;
 	}
+	
+	LOCAL_USER_ADD(u);
 
 	if (chan->_state != AST_STATE_UP) {
-		if ((res = ast_answer(chan)))
+		res = ast_answer(chan);
+		if (res) {
+			LOCAL_USER_REMOVE(u);
 			return -1;
+		}
 	}
-
-	argcopy = ast_strdupa(data);
-
-	AST_STANDARD_APP_ARGS(arglist, argcopy);
-
-	if (!ast_strlen_zero(arglist.options))
-		ast_app_parse_options(auth_app_options, &flags, NULL, arglist.options);
-
-	if (!ast_strlen_zero(arglist.maxdigits)) {
-		maxdigits = atoi(arglist.maxdigits);
-		if ((maxdigits<1) || (maxdigits>sizeof(passwd)-2))
-			maxdigits = sizeof(passwd) - 2;
-	} else {
-		maxdigits = sizeof(passwd) - 2;
-	}
-
+	
+	strncpy(password, data, sizeof(password) - 1);
+	opts=strchr(password, '|');
+	if (opts) {
+		*opts = 0;
+		opts++;
+	} else
+		opts = "";
+	if (strchr(opts, 'j'))
+		jump = 1;
 	/* Start asking for password */
+	prompt = "agent-pass";
 	for (retries = 0; retries < 3; retries++) {
-		if ((res = ast_app_getdata(chan, prompt, passwd, maxdigits, 0)) < 0)
+		res = ast_app_getdata(chan, prompt, passwd, sizeof(passwd) - 2, 0);
+		if (res < 0)
 			break;
-
 		res = 0;
+		if (password[0] == '/') {
+			if (strchr(opts, 'd')) {
+				char tmp[256];
+				/* Compare against a database key */
+				if (!ast_db_get(password + 1, passwd, tmp, sizeof(tmp))) {
+					/* It's a good password */
+					if (strchr(opts, 'r')) {
+						ast_db_del(password + 1, passwd);
+					}
+					break;
+				}
+			} else {
+				/* Compare against a file */
+				FILE *f;
+				f = fopen(password, "r");
+				if (f) {
+					char buf[256] = "";
+					char md5passwd[33] = "";
+					char *md5secret = NULL;
 
-		if (arglist.password[0] != '/') {
-			/* Compare against a fixed password */
-			if (!strcmp(passwd, arglist.password))
-				break;
-		} else if (ast_test_flag(&flags,OPT_DATABASE)) {
-			char tmp[256];
-			/* Compare against a database key */
-			if (!ast_db_get(arglist.password + 1, passwd, tmp, sizeof(tmp))) {
-				/* It's a good password */
-				if (ast_test_flag(&flags,OPT_REMOVE))
-					ast_db_del(arglist.password + 1, passwd);
-				break;
+					while (!feof(f)) {
+						fgets(buf, sizeof(buf), f);
+						if (!feof(f) && !ast_strlen_zero(buf)) {
+							buf[strlen(buf) - 1] = '\0';
+							if (strchr(opts, 'm')) {
+								md5secret = strchr(buf, ':');
+								if (md5secret == NULL)
+									continue;
+								*md5secret = '\0';
+								md5secret++;
+								ast_md5_hash(md5passwd, passwd);
+								if (!strcmp(md5passwd, md5secret)) {
+									if (strchr(opts, 'a'))
+										ast_cdr_setaccount(chan, buf);
+									break;
+								}
+							} else {
+								if (!strcmp(passwd, buf)) {
+									if (strchr(opts, 'a'))
+										ast_cdr_setaccount(chan, buf);
+									break;
+								}
+							}
+						}
+					}
+					fclose(f);
+					if (!ast_strlen_zero(buf)) {
+						if (strchr(opts, 'm')) {
+							if (md5secret && !strcmp(md5passwd, md5secret))
+								break;
+						} else {
+							if (!strcmp(passwd, buf))
+								break;
+						}
+					}
+				} else 
+					ast_log(LOG_WARNING, "Unable to open file '%s' for authentication: %s\n", password, strerror(errno));
 			}
 		} else {
-			/* Compare against a file */
-			FILE *f;
-			char buf[256] = "", md5passwd[33] = "", *md5secret = NULL;
-
-			if (!(f = fopen(arglist.password, "r"))) {
-				ast_log(LOG_WARNING, "Unable to open file '%s' for authentication: %s\n", arglist.password, strerror(errno));
-				continue;
-			}
-
-			for (;;) {
-				size_t len;
-
-				if (feof(f))
-					break;
-
-				if (!fgets(buf, sizeof(buf), f)) {
-					continue;
-				}
-
-				if (ast_strlen_zero(buf))
-					continue;
-
-				len = strlen(buf) - 1;
-				if (buf[len] == '\n')
-					buf[len] = '\0';
-
-				if (ast_test_flag(&flags, OPT_MULTIPLE)) {
-					md5secret = buf;
-					strsep(&md5secret, ":");
-					if (!md5secret)
-						continue;
-					ast_md5_hash(md5passwd, passwd);
-					if (!strcmp(md5passwd, md5secret)) {
-						if (ast_test_flag(&flags,OPT_ACCOUNT))
-							ast_cdr_setaccount(chan, buf);
-						break;
-					}
-				} else {
-					if (!strcmp(passwd, buf)) {
-						if (ast_test_flag(&flags, OPT_ACCOUNT))
-							ast_cdr_setaccount(chan, buf);
-						break;
-					}
-				}
-			}
-
-			fclose(f);
-
-			if (!ast_strlen_zero(buf)) {
-				if (ast_test_flag(&flags, OPT_MULTIPLE)) {
-					if (md5secret && !strcmp(md5passwd, md5secret))
-						break;
-				} else {
-					if (!strcmp(passwd, buf))
-						break;
-				}
-			}
+			/* Compare against a fixed password */
+			if (!strcmp(passwd, password)) 
+				break;
 		}
-		prompt = "auth-incorrect";
+		prompt="auth-incorrect";
 	}
-
 	if ((retries < 3) && !res) {
-		if (ast_test_flag(&flags,OPT_ACCOUNT) && !ast_test_flag(&flags,OPT_MULTIPLE))
+		if (strchr(opts, 'a') && !strchr(opts, 'm')) 
 			ast_cdr_setaccount(chan, passwd);
-		if (!(res = ast_streamfile(chan, "auth-thankyou", chan->language)))
+		res = ast_streamfile(chan, "auth-thankyou", chan->language);
+		if (!res)
 			res = ast_waitstream(chan, "");
 	} else {
-		if (!ast_streamfile(chan, "vm-goodbye", chan->language))
-			res = ast_waitstream(chan, "");
-		res = -1;
+		if (jump && ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101) == 0) {
+			res = 0;
+		} else {
+			if (!ast_streamfile(chan, "vm-goodbye", chan->language))
+				res = ast_waitstream(chan, "");
+			res = -1;
+		}
 	}
-
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+	
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
-	if (ast_register_application(app, auth_exec, synopsis, descrip))
-		return AST_MODULE_LOAD_FAILURE;
-	return AST_MODULE_LOAD_SUCCESS;
+	return ast_register_application(app, auth_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Authentication Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_cdr.c

@@ -17,47 +17,81 @@
  */
 
 /*! \file
- *
  * \brief Applications connected with CDR engine
  *
- * \author Martin Pycko <martinp@digium.com>
- *
  * \ingroup applications
  */
 
+#include <sys/types.h>
+#include <stdlib.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/channel.h"
 #include "asterisk/module.h"
+#include "asterisk/pbx.h"
 
-static char *nocdr_descrip =
+
+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_app = "NoCDR";
 static char *nocdr_synopsis = "Tell Asterisk to not maintain a CDR for the current call";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int nocdr_exec(struct ast_channel *chan, void *data)
 {
-	if (chan->cdr)
-		ast_set_flag(chan->cdr, AST_CDR_FLAG_POST_DISABLED);
+	struct localuser *u;
+	
+	LOCAL_USER_ADD(u);
+
+	if (chan->cdr) {
+		ast_cdr_free(chan->cdr);
+		chan->cdr = NULL;
+	}
+
+	LOCAL_USER_REMOVE(u);
 
 	return 0;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(nocdr_app);
+	int res;
+
+	res = ast_unregister_application(nocdr_app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
-	if (ast_register_application(nocdr_app, nocdr_exec, nocdr_synopsis, nocdr_descrip))
-		return AST_MODULE_LOAD_FAILURE;
-	return AST_MODULE_LOAD_SUCCESS;
+	return ast_register_application(nocdr_app, nocdr_exec, nocdr_synopsis, nocdr_descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Tell Asterisk to not maintain a CDR for the current call");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_chanisavail.c

@@ -18,79 +18,79 @@
 */
 
 /*! \file
- *
- * \brief Check if Channel is Available
- *
- * \author Mark Spencer <markster@digium.com>
- * \author James Golovich <james@gnuinter.net>
-
+* \brief Check if Channel is Available
+* 
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include <sys/ioctl.h>
-
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/app.h"
 #include "asterisk/devicestate.h"
+#include "asterisk/options.h"
+
+static char *tdesc = "Check channel availability";
 
 static char *app = "ChanIsAvail";
 
 static char *synopsis = "Check channel availability";
 
-static char *descrip =
-"  ChanIsAvail(Technology/resource[&Technology2/resource2...][,options]): \n"
+static char *descrip = 
+"  ChanIsAvail(Technology/resource[&Technology2/resource2...][|options]): \n"
 "This application will check to see if any of the specified channels are\n"
-"available.\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"
-"    t - Simply checks if specified channels exist in the channel list\n"
-"        (implies option s).\n"
-"Note that the AVAILSTATUS variable is used for both device state\n"
-"and cause code. It is therefore possible for it to give a value that may\n"
-"indicate a device is available when it is not. It is suggested that the\n"
-"AVAILORIGCHAN variable is used instead to see whether a device is available\n"
-"or not.\n"
-"This application sets the following channel variable upon completion:\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";
+"    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;
+
+LOCAL_USER_DECL;
 
 static int chanavail_exec(struct ast_channel *chan, void *data)
 {
-	int res=-1, inuse=-1, option_state=0, string_compare=0;
+	int res=-1, inuse=-1, option_state=0, priority_jump=0;
 	int status;
-	char *info, tmp[512], trychan[512], *peers, *tech, *number, *rest, *cur;
+	struct localuser *u;
+	char *info, tmp[512], trychan[512], *peers, *tech, *number, *rest, *cur, *options, *stringp;
 	struct ast_channel *tempchan;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(reqchans);
-		AST_APP_ARG(options);
-	);
 
 	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "ChanIsAvail requires an argument (DAHDI/1&DAHDI/2)\n");
+		ast_log(LOG_WARNING, "ChanIsAvail requires an argument (Zap/1&Zap/2)\n");
 		return -1;
 	}
 
-	info = ast_strdupa(data);
+	LOCAL_USER_ADD(u);
 
-	AST_STANDARD_APP_ARGS(args, info);
-
-	if (args.options) {
-		if (strchr(args.options, 's'))
+	info = ast_strdupa(data); 
+	stringp = info;
+	strsep(&stringp, "|");
+	options = strsep(&stringp, "|");
+	if (options) {
+		if (strchr(options, 's'))
 			option_state = 1;
-		if (strchr(args.options, 't'))
-			string_compare = 1;
+		if (strchr(options, 'j'))
+			priority_jump = 1;
 	}
-	peers = args.reqchans;
+	peers = info;
 	if (peers) {
 		cur = peers;
 		do {
@@ -104,22 +104,17 @@ static int chanavail_exec(struct ast_channel *chan, void *data)
 			number = strchr(tech, '/');
 			if (!number) {
 				ast_log(LOG_WARNING, "ChanIsAvail argument takes format ([technology]/[device])\n");
+				LOCAL_USER_REMOVE(u);
 				return -1;
 			}
 			*number = '\0';
 			number++;
 			
-			if (string_compare) {
-				/* ast_parse_device_state checks for "SIP/1234" as a channel name.
-				   ast_device_state will ask the SIP driver for the channel state. */
-
-				snprintf(trychan, sizeof(trychan), "%s/%s",cur,number);
-				status = inuse = ast_parse_device_state(trychan);
-			} else if (option_state) {
+			if (option_state) {
 				/* If the pbx says in use then don't bother trying further.
-				   This is to permit testing if someone's on a call, even if the
-				   channel can permit more calls (ie callwaiting, sip calls, etc).  */
-
+				   This is to permit testing if someone's on a call, even if the 
+	 			   channel can permit more calls (ie callwaiting, sip calls, etc).  */
+                               
 				snprintf(trychan, sizeof(trychan), "%s/%s",cur,number);
 				status = inuse = ast_device_state(trychan);
 			}
@@ -144,19 +139,47 @@ 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 (priority_jump || option_priority_jumping) {
+			if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
+				LOCAL_USER_REMOVE(u);
+				return -1;
+			}
+		}
 	}
 
+	LOCAL_USER_REMOVE(u);
 	return 0;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res = 0;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+	
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, chanavail_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Check channel availability");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_channelredirect.c

@@ -1,98 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2006, Sergey Basmanov
- *
- * 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 ChannelRedirect application
- *
- * \author Sergey Basmanov <sergey_basmanov@mail.ru>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/lock.h"
-#include "asterisk/app.h"
-#include "asterisk/features.h"
-
-static char *app = "ChannelRedirect";
-static char *synopsis = "Redirects given channel to a dialplan target.";
-static char *descrip =
-"ChannelRedirect(channel,[[context,]extension,]priority)\n"
-"  Sends the specified channel to the specified extension priority\n"
-"This application sets the following channel variables upon completion:\n"
-"  CHANNELREDIRECT_STATUS - Are set to the result of the redirection\n"
-"                           either NOCHANNEL or SUCCESS\n";
-
-static int asyncgoto_exec(struct ast_channel *chan, void *data)
-{
-	int res = -1;
-	char *info;
-	struct ast_channel *chan2 = NULL;
-
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(channel);
-		AST_APP_ARG(label);
-	);
-
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "%s requires an argument (channel,[[context,]exten,]priority)\n", app);
-		return -1;
-	}
-
-	info = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, info);
-
-	if (ast_strlen_zero(args.channel) || ast_strlen_zero(args.label)) {
-		ast_log(LOG_WARNING, "%s requires an argument (channel,[[context,]exten,]priority)\n", app);
-		return -1;
-	}
-
-	chan2 = ast_get_channel_by_name_locked(args.channel);
-	if (!chan2) {
-		ast_log(LOG_WARNING, "No such channel: %s\n", args.channel);
-		pbx_builtin_setvar_helper(chan, "CHANNELREDIRECT_STATUS", "NOCHANNEL");
-		return 0;
-	}
-
-	if (chan2->pbx) {
-		ast_set_flag(chan2, AST_FLAG_BRIDGE_HANGUP_DONT); /* don't let the after-bridge code run the h-exten */
-	}
-	res = ast_async_parseable_goto(chan2, args.label);
-	pbx_builtin_setvar_helper(chan, "CHANNELREDIRECT_STATUS", "SUCCESS");
-	ast_channel_unlock(chan2);
-
-	return res;
-}
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app, asyncgoto_exec, synopsis, descrip);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Redirects a given channel to a dialplan target");

apps/app_controlplayback.c

@@ -17,28 +17,38 @@
  */
 
 /*! \file
- *
  * \brief Trivial application to control playback of a sound file
- *
- * \author Mark Spencer <markster@digium.com>
- *
+ * 
  * \ingroup applications
  */
+ 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/app.h"
 #include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/utils.h"
+#include "asterisk/options.h"
+
+static const char *tdesc = "Control Playback Application";
 
 static const char *app = "ControlPlayback";
 
 static const char *synopsis = "Play a file with fast forward and rewind";
 
-static const char *descrip =
-"  ControlPlayback(file[,skipms[,ff[,rew[,stop[,pause[,restart,options]]]]]]]):\n"
+static const char *descrip = 
+"  ControlPlayback(file[|skipms[|ff[|rew[|stop[|pause[|restart|options]]]]]]]):\n"
 "This application will play back the given filename. By default, the '*' key\n"
 "can be used to rewind, and the '#' key can be used to fast-forward.\n"
 "Parameters:\n"
@@ -50,142 +60,131 @@ static const char *descrip =
 "  pause   - Pause playback when this DTMF digit is received.\n"
 "  restart - Restart playback when this DTMF digit is received.\n"
 "Options:\n"
-"  o(#) - Start at # ms from the beginning of the file.\n"
-"This application sets the following channel variables upon completion:\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"
-"  CPLAYBACKOFFSET -  This contains the offset in ms into the file where\n"
-"                     playback was at when it stopped.  -1 is end of file.\n"
-"  CPLAYBACKSTOPKEY - If the playback is stopped by the user this variable contains\n"
-"                     the key that was pressed.\n";
+"                     string, one of: SUCCESS | USERSTOPPED | ERROR\n";
 
-enum {
-	OPT_OFFSET = (1 << 1),
-};
+STANDARD_LOCAL_USER;
 
-enum {
-	OPT_ARG_OFFSET = 0,
-	/* must stay as the last entry ... */
-	OPT_ARG_ARRAY_LEN,
-};
-
-AST_APP_OPTIONS(cpb_opts, BEGIN_OPTIONS
-	AST_APP_OPTION_ARG('o', OPT_OFFSET, OPT_ARG_OFFSET),
-	END_OPTIONS
-);
+LOCAL_USER_DECL;
 
 static int is_on_phonepad(char key)
 {
 	return key == 35 || key == 42 || (key >= 48 && key <= 57);
 }
 
-static int is_argument(const char *haystack, int needle)
-{
-	if (ast_strlen_zero(haystack))
-		return 0;
-
-	if (strchr(haystack, needle))
-		return -1;
-
-	return 0;
-}
-
 static int controlplayback_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
+	int res = 0, priority_jump = 0;
 	int skipms = 0;
-	long offsetms = 0;
-	char offsetbuf[20];
-	char stopkeybuf[2];
+	struct localuser *u;
 	char *tmp;
-	struct ast_flags opts = { 0, };
-	char *opt_args[OPT_ARG_ARRAY_LEN];
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(filename);
-		AST_APP_ARG(skip);
-		AST_APP_ARG(fwd);
-		AST_APP_ARG(rev);
-		AST_APP_ARG(stop);
-		AST_APP_ARG(pause);
-		AST_APP_ARG(restart);
-		AST_APP_ARG(options);
-	);
-
+	int argc;
+	char *argv[8];
+	enum arg_ids {
+		arg_file = 0,
+		arg_skip = 1,
+		arg_fwd = 2,
+		arg_rev = 3,
+		arg_stop = 4,
+		arg_pause = 5,
+		arg_restart = 6,
+		options = 7,
+	};
+	
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "ControlPlayback requires an argument (filename)\n");
 		return -1;
 	}
+
+	LOCAL_USER_ADD(u);
 	
 	tmp = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, tmp);
+	memset(argv, 0, sizeof(argv));
 
-	if (args.argc < 1) {
+	argc = ast_app_separate_args(tmp, '|', argv, sizeof(argv) / sizeof(argv[0]));
+
+	if (argc < 1) {
 		ast_log(LOG_WARNING, "ControlPlayback requires an argument (filename)\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 
-	skipms = args.skip ? (atoi(args.skip) ? atoi(args.skip) : 3000) : 3000;
+	skipms = argv[arg_skip] ? atoi(argv[arg_skip]) : 3000;
+	if (!skipms)
+		skipms = 3000;
 
-	if (!args.fwd || !is_on_phonepad(*args.fwd)) {
-		char *digit = "#";
-		if (!is_argument(args.rev, *digit) && !is_argument(args.stop, *digit) && !is_argument(args.pause, *digit) && !is_argument(args.restart, *digit))
-			args.fwd = digit;
-		else
-			args.fwd = NULL;
-	}
-	if (!args.rev || !is_on_phonepad(*args.rev)) {
-		char *digit = "*";
-		if (!is_argument(args.fwd, *digit) && !is_argument(args.stop, *digit) && !is_argument(args.pause, *digit) && !is_argument(args.restart, *digit))
-			args.rev = digit;
-		else
-			args.rev = NULL;
-	}
-	ast_debug(1, "Forward key = %s, Rewind key = %s\n", args.fwd, args.rev);
-	if (args.stop && !is_on_phonepad(*args.stop))
-		args.stop = NULL;
-	if (args.pause && !is_on_phonepad(*args.pause))
-		args.pause = NULL;
-	if (args.restart && !is_on_phonepad(*args.restart))
-		args.restart = NULL;
+	if (!argv[arg_fwd] || !is_on_phonepad(*argv[arg_fwd]))
+		argv[arg_fwd] = "#";
+	if (!argv[arg_rev] || !is_on_phonepad(*argv[arg_rev]))
+		argv[arg_rev] = "*";
+	if (argv[arg_stop] && !is_on_phonepad(*argv[arg_stop]))
+		argv[arg_stop] = NULL;
+	if (argv[arg_pause] && !is_on_phonepad(*argv[arg_pause]))
+		argv[arg_pause] = NULL;
+	if (argv[arg_restart] && !is_on_phonepad(*argv[arg_restart]))
+		argv[arg_restart] = NULL;
 
-	if (args.options) {
-		ast_app_parse_options(cpb_opts, &opts, opt_args, args.options);
-		if (ast_test_flag(&opts, OPT_OFFSET))
-			offsetms = atol(opt_args[OPT_ARG_OFFSET]);
+	if (argv[options]) {
+		if (strchr(argv[options], 'j'))
+			priority_jump = 1;
 	}
 
-	res = ast_control_streamfile(chan, args.filename, args.fwd, args.rev, args.stop, args.pause, args.restart, skipms, &offsetms);
+	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 (res > 0 && args.stop && strchr(args.stop, res)) {
-		pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "USERSTOPPED");
-		snprintf(stopkeybuf, sizeof(stopkeybuf), "%c", res);
-		pbx_builtin_setvar_helper(chan, "CPLAYBACKSTOPKEY", stopkeybuf);
+	if (argv[arg_stop] && strchr(argv[arg_stop], res)) {
 		res = 0;
+		pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "USERSTOPPED");
 	} else {
 		if (res < 0) {
+			if (priority_jump || option_priority_jumping) {
+				if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
+					ast_log(LOG_WARNING, "ControlPlayback tried to jump to priority n+101 as requested, but priority didn't exist\n");
+				}
+			}
 			res = 0;
 			pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "ERROR");
 		} else
 			pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "SUCCESS");
 	}
 
-	snprintf(offsetbuf, sizeof(offsetbuf), "%ld", offsetms);
-	pbx_builtin_setvar_helper(chan, "CPLAYBACKOFFSET", offsetbuf);
+	LOCAL_USER_REMOVE(u);
 
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
 	int res;
+
 	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
 	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, controlplayback_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Control Playback Application");
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_curl.c

@@ -0,0 +1,257 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C)  2004 - 2005, Tilghman Lesher
+ *
+ * Tilghman Lesher <curl-20050919@the-tilghman.com>
+ * and Brian Wilkins <bwilkins@cfl.rr.com> (Added POST option)
+ *
+ * app_curl.c is distributed with no restrictions on usage or
+ * redistribution.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ * \brief Curl - App to load a URL
+ * 
+ * \ingroup applications
+ */
+ 
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <curl/curl.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/cli.h"
+#include "asterisk/options.h"
+#include "asterisk/module.h"
+
+static char *tdesc = "Load external URL";
+
+static char *app = "Curl";
+
+static char *synopsis = "Load an external URL";
+
+static char *descrip = 
+"  Curl(URL[|postdata]): This application will request the specified URL.\n"
+"It is mainly used for signalling external applications of an event.\n"
+"Parameters:\n"
+"  URL      - This is the external URL to request.\n"
+"  postdata - This information will be treated as POST data.\n"
+"This application will set the following variable:\n"
+"  CURL - This variable will contain the resulting page.\n"
+"This application has been deprecated in favor of the CURL function.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+struct MemoryStruct {
+	char *memory;
+	size_t size;
+};
+
+static void *myrealloc(void *ptr, size_t size)
+{
+	/* There might be a realloc() out there that doesn't like reallocing
+	   NULL pointers, so we take care of it here */
+	if (ptr)
+		return realloc(ptr, size);
+	else
+		return malloc(size);
+}
+
+static size_t WriteMemoryCallback(void *ptr, size_t size, size_t nmemb, void *data)
+{
+	register int realsize = size * nmemb;
+	struct MemoryStruct *mem = (struct MemoryStruct *)data;
+
+	mem->memory = (char *)myrealloc(mem->memory, mem->size + realsize + 1);
+	if (mem->memory) {
+		memcpy(&(mem->memory[mem->size]), ptr, realsize);
+		mem->size += realsize;
+		mem->memory[mem->size] = 0;
+	}
+	return realsize;
+}
+
+static int curl_internal(struct MemoryStruct *chunk, char *url, char *post)
+{
+	CURL *curl;
+
+	curl = curl_easy_init();
+
+	if (!curl) {
+		return -1;
+	}
+
+	curl_easy_setopt(curl, CURLOPT_URL, url);
+	curl_easy_setopt(curl, CURLOPT_WRITEFUNCTION, WriteMemoryCallback);
+	curl_easy_setopt(curl, CURLOPT_WRITEDATA, (void *)chunk);
+	curl_easy_setopt(curl, CURLOPT_USERAGENT, "asterisk-libcurl-agent/1.0");
+
+	if (post) {
+		curl_easy_setopt(curl, CURLOPT_POST, 1);
+		curl_easy_setopt(curl, CURLOPT_POSTFIELDS, post);
+	}
+
+	curl_easy_perform(curl);
+	curl_easy_cleanup(curl);
+	return 0;
+}
+
+static int curl_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u;
+	char *info, *post_data=NULL, *url;
+	struct MemoryStruct chunk = { NULL, 0 };
+	static int dep_warning = 0;
+	
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The application Curl is deprecated.  Please use the CURL() function instead.\n");
+		dep_warning = 1;
+	}
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Curl requires an argument (URL)\n");
+		return -1;
+	}
+	
+	LOCAL_USER_ADD(u);
+	
+	if ((info = ast_strdupa(data))) {
+		url = strsep(&info, "|");
+		post_data = info;
+	} else {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (! curl_internal(&chunk, url, post_data)) {
+		if (chunk.memory) {
+			chunk.memory[chunk.size] = '\0';
+			if (chunk.memory[chunk.size - 1] == 10)
+				chunk.memory[chunk.size - 1] = '\0';
+
+			pbx_builtin_setvar_helper(chan, "CURL", chunk.memory);
+
+			free(chunk.memory);
+		}
+	} else {
+		ast_log(LOG_ERROR, "Cannot allocate curl structure\n");
+		res = -1;
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static char *acf_curl_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
+	struct localuser *u;
+	char *info, *post_data=NULL, *url;
+	struct MemoryStruct chunk = { NULL, 0 };
+
+	*buf = '\0';
+	
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "CURL requires an argument (URL)\n");
+		return buf;
+	}
+
+	LOCAL_USER_ACF_ADD(u);
+
+	info = ast_strdupa(data);
+	if (!info) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return buf;
+	}
+	
+	url = strsep(&info, "|");
+	post_data = info;
+	
+	if (! curl_internal(&chunk, url, post_data)) {
+		if (chunk.memory) {
+			chunk.memory[chunk.size] = '\0';
+			if (chunk.memory[chunk.size - 1] == 10)
+				chunk.memory[chunk.size - 1] = '\0';
+
+			ast_copy_string(buf, chunk.memory, len);
+			free(chunk.memory);
+		}
+	} else {
+		ast_log(LOG_ERROR, "Cannot allocate curl structure\n");
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return buf;
+}
+
+struct ast_custom_function acf_curl = {
+	.name = "CURL",
+	.synopsis = "Retrieves the contents of a URL",
+	.syntax = "CURL(url[|post-data])",
+	.desc =
+	"  url       - URL to retrieve\n"
+	"  post-data - Optional data to send as a POST (GET is default action)\n",
+	.read = acf_curl_exec,
+};
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_custom_function_unregister(&acf_curl);
+	res |= ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+	curl_global_cleanup();
+	
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+
+	curl_global_init(CURL_GLOBAL_ALL);
+	res = ast_custom_function_register(&acf_curl);
+	res |= ast_register_application(app, curl_exec, synopsis, descrip);
+
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_cut.c

@@ -0,0 +1,473 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
+ *
+ * Tilghman Lesher <app_cut__v003@the-tilghman.com>
+ *
+ * This code is released by the author with no restrictions on usage.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ * \brief Cut application
+ *
+ * \ingroup applications
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+
+/* Maximum length of any variable */
+#define MAXRESULT	1024
+
+static char *tdesc = "Cut out information from a string";
+
+static char *app_cut = "Cut";
+
+static char *cut_synopsis = "Splits a variable's contents using the specified delimiter";
+
+static char *cut_descrip =
+"  Cut(newvar=varname,delimiter,fieldspec): This application 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"
+"This application has been deprecated in favor of the CUT function.\n";
+
+static char *app_sort = "Sort";
+static char *app_sort_synopsis = "Sorts a list of keywords and values";
+static char *app_sort_descrip =
+"  Sort(newvar=key1:val1[,key2:val2[[...],keyN:valN]]): This application will\n"
+"sort the list provided in ascending order. The result will be stored in the\n"
+"specified variable name.\n"
+"  This application has been deprecated in favor of the SORT function.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+struct sortable_keys {
+	char *key;
+	float value;
+};
+
+static int sort_subroutine(const void *arg1, const void *arg2)
+{
+	const struct sortable_keys *one=arg1, *two=arg2;
+	if (one->value < two->value) {
+		return -1;
+	} else if (one->value == two->value) {
+		return 0;
+	} else {
+		return 1;
+	}
+}
+
+#define ERROR_NOARG	(-1)
+#define ERROR_NOMEM	(-2)
+#define ERROR_USAGE	(-3)
+
+static int sort_internal(struct ast_channel *chan, char *data, char *buffer, size_t buflen)
+{
+	char *strings, *ptrkey, *ptrvalue;
+	int count=1, count2, element_count=0;
+	struct sortable_keys *sortable_keys;
+
+	memset(buffer, 0, buflen);
+
+	if (!data) {
+		return ERROR_NOARG;
+	}
+
+	strings = ast_strdupa((char *)data);
+	if (!strings) {
+		return ERROR_NOMEM;
+	}
+
+	for (ptrkey = strings; *ptrkey; ptrkey++) {
+		if (*ptrkey == '|') {
+			count++;
+		}
+	}
+
+	sortable_keys = alloca(count * sizeof(struct sortable_keys));
+	if (!sortable_keys) {
+		return ERROR_NOMEM;
+	}
+
+	memset(sortable_keys, 0, count * sizeof(struct sortable_keys));
+
+	/* Parse each into a struct */
+	count2 = 0;
+	while ((ptrkey = strsep(&strings, "|"))) {
+		ptrvalue = index(ptrkey, ':');
+		if (!ptrvalue) {
+			count--;
+			continue;
+		}
+		*ptrvalue = '\0';
+		ptrvalue++;
+		sortable_keys[count2].key = ptrkey;
+		sscanf(ptrvalue, "%f", &sortable_keys[count2].value);
+		count2++;
+	}
+
+	/* Sort the structs */
+	qsort(sortable_keys, count, sizeof(struct sortable_keys), sort_subroutine);
+
+	for (count2 = 0; count2 < count; count2++) {
+		int blen = strlen(buffer);
+		if (element_count++) {
+			strncat(buffer + blen, ",", buflen - blen - 1);
+			blen++;
+		}
+		strncat(buffer + blen, sortable_keys[count2].key, buflen - blen - 1);
+	}
+
+	return 0;
+}
+
+static int cut_internal(struct ast_channel *chan, char *data, char *buffer, size_t buflen)
+{
+	char *s, *args[3], *varname=NULL, *delimiter=NULL, *field=NULL;
+	int args_okay = 0;
+
+	memset(buffer, 0, buflen);
+
+	/* Check and parse arguments */
+	if (data) {
+		s = ast_strdupa((char *)data);
+		if (s) {
+			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;
+		}
+	}
+
+	if (args_okay) {
+		char d, ds[2];
+		char *tmp = alloca(strlen(varname) + 4);
+		char varvalue[MAXRESULT], *tmp2=varvalue;
+
+		if (tmp) {
+			snprintf(tmp, strlen(varname) + 4, "${%s}", varname);
+			memset(varvalue, 0, sizeof(varvalue));
+		} else {
+			return ERROR_NOMEM;
+		}
+
+		if (delimiter[0])
+			d = delimiter[0];
+		else
+			d = '-';
+
+		/* String form of the delimiter, for use with strsep(3) */
+		snprintf(ds, sizeof(ds), "%c", d);
+
+		pbx_substitute_variables_helper(chan, tmp, tmp2, MAXRESULT - 1);
+
+		if (tmp2) {
+			int curfieldnum = 1;
+			while ((tmp2 != NULL) && (field != NULL)) {
+				char *nextgroup = strsep(&field, "&");
+				int num1 = 0, num2 = MAXRESULT;
+				char trashchar;
+
+				if (sscanf(nextgroup, "%d-%d", &num1, &num2) == 2) {
+					/* range with both start and end */
+				} else if (sscanf(nextgroup, "-%d", &num2) == 1) {
+					/* range with end */
+					num1 = 0;
+				} else if ((sscanf(nextgroup, "%d%c", &num1, &trashchar) == 2) && (trashchar == '-')) {
+					/* range with start */
+					num2 = MAXRESULT;
+				} else if (sscanf(nextgroup, "%d", &num1) == 1) {
+					/* single number */
+					num2 = num1;
+				} else {
+					return ERROR_USAGE;
+				}
+
+				/* Get to start, if any */
+				if (num1 > 0) {
+					while ((tmp2 != (char *)NULL + 1) && (curfieldnum < num1)) {
+						tmp2 = index(tmp2, d) + 1;
+						curfieldnum++;
+					}
+				}
+
+				/* Most frequent problem is the expectation of reordering fields */
+				if ((num1 > 0) && (curfieldnum > num1)) {
+					ast_log(LOG_WARNING, "We're already past the field you wanted?\n");
+				}
+
+				/* Re-null tmp2 if we added 1 to NULL */
+				if (tmp2 == (char *)NULL + 1)
+					tmp2 = NULL;
+
+				/* Output fields until we either run out of fields or num2 is reached */
+				while ((tmp2 != NULL) && (curfieldnum <= num2)) {
+					char *tmp3 = strsep(&tmp2, ds);
+					int curlen = strlen(buffer);
+
+					if (curlen) {
+						snprintf(buffer + curlen, buflen - curlen, "%c%s", d, tmp3);
+					} else {
+						snprintf(buffer, buflen, "%s", tmp3);
+					}
+
+					curfieldnum++;
+				}
+			}
+		}
+	} else {
+		return ERROR_NOARG;
+	}
+	return 0;
+}
+
+static int sort_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *varname, *strings, result[512] = "";
+	static int dep_warning=0;
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The application Sort is deprecated.  Please use the SORT() function instead.\n");
+		dep_warning=1;
+	}
+
+	if (!data) {
+		ast_log(LOG_ERROR, "Sort() requires an argument\n");
+		return 0;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	strings = ast_strdupa((char *)data);
+	if (!strings) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+
+	varname = strsep(&strings, "=");
+	switch (sort_internal(chan, strings, result, sizeof(result))) {
+	case ERROR_NOARG:
+		ast_log(LOG_ERROR, "Sort() requires an argument\n");
+		res = 0;
+		break;
+	case ERROR_NOMEM:
+		ast_log(LOG_ERROR, "Out of memory\n");
+		res = -1;
+		break;
+	case 0:
+		pbx_builtin_setvar_helper(chan, varname, result);
+		res = 0;
+		break;
+	default:
+		ast_log(LOG_ERROR, "Unknown internal error\n");
+		res = -1;
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static int cut_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *s, *newvar=NULL, result[512];
+	static int dep_warning = 0;
+
+	LOCAL_USER_ADD(u);
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The application Cut is deprecated.  Please use the CUT() function instead.\n");
+		dep_warning=1;
+	}
+
+	/* Check and parse arguments */
+	if (data) {
+		s = ast_strdupa((char *)data);
+		if (s) {
+			newvar = strsep(&s, "=");
+		} else {
+			ast_log(LOG_ERROR, "Out of memory\n");
+			LOCAL_USER_REMOVE(u);
+			return -1;
+		}
+	}
+
+	switch (cut_internal(chan, s, result, sizeof(result))) {
+	case ERROR_NOARG:
+		ast_log(LOG_ERROR, "Cut() requires an argument\n");
+		res = 0;
+		break;
+	case ERROR_NOMEM:
+		ast_log(LOG_ERROR, "Out of memory\n");
+		res = -1;
+		break;
+	case ERROR_USAGE:
+		ast_log(LOG_ERROR, "Usage: %s\n", cut_synopsis);
+		res = 0;
+		break;
+	case 0:
+		pbx_builtin_setvar_helper(chan, newvar, result);
+		res = 0;
+		break;
+	default:
+		ast_log(LOG_ERROR, "Unknown internal error\n");
+		res = -1;
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static char *acf_sort_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
+	struct localuser *u;
+
+	LOCAL_USER_ACF_ADD(u);
+
+	switch (sort_internal(chan, data, buf, len)) {
+	case ERROR_NOARG:
+		ast_log(LOG_ERROR, "SORT() requires an argument\n");
+		break;
+	case ERROR_NOMEM:
+		ast_log(LOG_ERROR, "Out of memory\n");
+		break;
+	case 0:
+		break;
+	default:
+		ast_log(LOG_ERROR, "Unknown internal error\n");
+	}
+	LOCAL_USER_REMOVE(u);
+	return buf;
+}
+
+static char *acf_cut_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
+	struct localuser *u;
+
+	LOCAL_USER_ACF_ADD(u);
+
+	switch (cut_internal(chan, data, buf, len)) {
+	case ERROR_NOARG:
+		ast_log(LOG_ERROR, "CUT() requires an argument\n");
+		break;
+	case ERROR_NOMEM:
+		ast_log(LOG_ERROR, "Out of memory\n");
+		break;
+	case ERROR_USAGE:
+		ast_log(LOG_ERROR, "Usage: %s\n", cut_synopsis);
+		break;
+	case 0:
+		break;
+	default:
+		ast_log(LOG_ERROR, "Unknown internal error\n");
+	}
+	LOCAL_USER_REMOVE(u);
+	return buf;
+}
+
+struct ast_custom_function acf_sort = {
+	.name = "SORT",
+	.synopsis = "Sorts a list of key/vals into a list of keys, based upon the vals",
+	.syntax = "SORT(key1:val1[...][,keyN:valN])",
+	.desc =
+"Takes a comma-separated list of keys and values, each separated by a colon, and returns a\n"
+"comma-separated list of the keys, sorted by their values.  Values will be evaluated as\n"
+"floating-point numbers.\n",
+	.read = acf_sort_exec,
+};
+
+struct ast_custom_function acf_cut = {
+	.name = "CUT",
+	.synopsis = "Slices and dices strings, based upon a named delimiter.",
+	.syntax = "CUT(<varname>,<char-delim>,<range-spec>)",
+	.desc =
+"  varname    - variable you want cut\n"
+"  char-delim - defaults to '-'\n"
+"  range-spec - number of the field you want (1-based offset)\n"
+"             may also be specified as a range (with -)\n"
+"             or group of ranges and fields (with &)\n",
+	.read = acf_cut_exec,
+};
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_custom_function_unregister(&acf_cut);
+	res |= ast_custom_function_unregister(&acf_sort);
+	res |= ast_unregister_application(app_sort);
+	res |= ast_unregister_application(app_cut);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+
+	res = ast_custom_function_register(&acf_cut);
+	res |= ast_custom_function_register(&acf_sort);
+	res |= ast_register_application(app_sort, sort_exec, app_sort_synopsis, app_sort_descrip);
+	res |= ast_register_application(app_cut, cut_exec, cut_synopsis, cut_descrip);
+
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_datetime.c

@@ -0,0 +1,101 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief Time of day - Report the time of day
+ * 
+ * \ingroup applications
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/say.h"
+
+static char *tdesc = "Date and Time";
+
+static char *app = "DateTime";
+
+static char *synopsis = "Say the date and time";
+
+static char *descrip = 
+"  DateTime(): This application will say the current date and time.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int datetime_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	time_t t;
+	struct localuser *u;
+	LOCAL_USER_ADD(u);
+	time(&t);
+	if (chan->_state != AST_STATE_UP)
+		res = ast_answer(chan);
+	if (!res)
+		res = ast_say_datetime(chan, t, "", chan->language);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, datetime_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_db.c

@@ -22,53 +22,90 @@
  *
  * \brief Database access functions
  *
- * \author Mark Spencer <markster@digium.com>
- * \author Jefferson Noxon <jeff@debian.org>
- *
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <sys/types.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/options.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/astdb.h"
 #include "asterisk/lock.h"
+#include "asterisk/options.h"
+
+static char *tdesc = "Database Access Functions";
+
+static char *g_descrip =
+"  DBget(varname=family/key[|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): 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";
 
-/*! \todo XXX Remove this application after 1.4 is relased */
 static char *d_descrip =
 "  DBdel(family/key): This application will delete a key from the Asterisk\n"
-"database.\n"
-"  This application has been DEPRECATED in favor of the DB_DELETE function.\n";
+"database.\n";
 
 static char *dt_descrip =
 "  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";
 static char *d_app = "DBdel";
 static char *dt_app = "DBdeltree";
 
+static char *g_synopsis = "Retrieve a value from the database";
+static char *p_synopsis = "Store a value in the database";
 static char *d_synopsis = "Delete a key from the database";
 static char *dt_synopsis = "Delete a family or keytree from the database";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int deltree_exec(struct ast_channel *chan, void *data)
 {
 	char *argv, *family, *keytree;
+	struct localuser *u;
+
+	LOCAL_USER_ADD(u);
 
 	argv = ast_strdupa(data);
+	if (!argv) {
+		ast_log(LOG_ERROR, "Memory allocation failed\n");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
 
 	if (strchr(argv, '/')) {
 		family = strsep(&argv, "/");
 		keytree = strsep(&argv, "\0");
-		if (!family || !keytree) {
-			ast_debug(1, "Ignoring; Syntax error in argument\n");
-			return 0;
-		}
+			if (!family || !keytree) {
+				ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+				LOCAL_USER_REMOVE(u);
+				return 0;
+			}
 		if (ast_strlen_zero(keytree))
 			keytree = 0;
 	} else {
@@ -76,13 +113,19 @@ static int deltree_exec(struct ast_channel *chan, void *data)
 		keytree = 0;
 	}
 
-	if (keytree)
-		ast_verb(3, "DBdeltree: family=%s, keytree=%s\n", family, keytree);
-	else
-		ast_verb(3, "DBdeltree: family=%s\n", family);
+	if (option_verbose > 2)	{
+		if (keytree)
+			ast_verbose(VERBOSE_PREFIX_3 "DBdeltree: family=%s, keytree=%s\n", family, keytree);
+		else
+			ast_verbose(VERBOSE_PREFIX_3 "DBdeltree: family=%s\n", family);
+	}
 
-	if (ast_db_deltree(family, keytree))
-		ast_verb(3, "DBdeltree: Error deleting key from database.\n");
+	if (ast_db_deltree(family, keytree)) {
+		if (option_verbose > 2)
+			ast_verbose(VERBOSE_PREFIX_3 "DBdeltree: Error deleting key from database.\n");
+	}
+
+	LOCAL_USER_REMOVE(u);
 
 	return 0;
 }
@@ -90,50 +133,191 @@ static int deltree_exec(struct ast_channel *chan, void *data)
 static int del_exec(struct ast_channel *chan, void *data)
 {
 	char *argv, *family, *key;
-	static int deprecation_warning = 0;
+	struct localuser *u;
 
-	if (!deprecation_warning) {
-		deprecation_warning = 1;
-		ast_log(LOG_WARNING, "The DBdel application has been deprecated in favor of the DB_DELETE dialplan function!\n");
-	}
+	LOCAL_USER_ADD(u);
 
 	argv = ast_strdupa(data);
+	if (!argv) {
+		ast_log (LOG_ERROR, "Memory allocation failed\n");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
 
 	if (strchr(argv, '/')) {
 		family = strsep(&argv, "/");
 		key = strsep(&argv, "\0");
 		if (!family || !key) {
-			ast_debug(1, "Ignoring; Syntax error in argument\n");
+			ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			LOCAL_USER_REMOVE(u);
 			return 0;
 		}
-		ast_verb(3, "DBdel: family=%s, key=%s\n", family, key);
-		if (ast_db_del(family, key))
-			ast_verb(3, "DBdel: Error deleting key from database.\n");
+		if (option_verbose > 2)
+			ast_verbose(VERBOSE_PREFIX_3 "DBdel: family=%s, key=%s\n", family, key);
+		if (ast_db_del(family, key)) {
+			if (option_verbose > 2)
+				ast_verbose(VERBOSE_PREFIX_3 "DBdel: Error deleting key from database.\n");
+		}
 	} else {
-		ast_debug(1, "Ignoring, no parameters\n");
+		ast_log(LOG_DEBUG, "Ignoring, no parameters\n");
 	}
 
+	LOCAL_USER_REMOVE(u);
+	
+	return 0;
+}
+
+static int put_exec(struct ast_channel *chan, void *data)
+{
+	char *argv, *value, *family, *key;
+	static int dep_warning = 0;
+	struct localuser *u;
+
+	LOCAL_USER_ADD(u);
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "This application has been deprecated, please use the ${DB(family/key)} function instead.\n");
+		dep_warning = 1;
+	}
+	
+	argv = ast_strdupa(data);
+	if (!argv) {
+		ast_log(LOG_ERROR, "Memory allocation failed\n");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+
+	if (strchr(argv, '/') && strchr(argv, '=')) {
+		family = strsep(&argv, "/");
+		key = strsep(&argv, "=");
+		value = strsep(&argv, "\0");
+		if (!value || !family || !key) {
+			ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			LOCAL_USER_REMOVE(u);
+			return 0;
+		}
+		if (option_verbose > 2)
+			ast_verbose(VERBOSE_PREFIX_3 "DBput: family=%s, key=%s, value=%s\n", family, key, value);
+		if (ast_db_put(family, key, value)) {
+			if (option_verbose > 2)
+				ast_verbose(VERBOSE_PREFIX_3 "DBput: Error writing value to database.\n");
+		}
+
+	} else	{
+		ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
+	}
+
+	LOCAL_USER_REMOVE(u);
+
 	return 0;
 }
 
-static int unload_module(void)
+static int get_exec(struct ast_channel *chan, void *data)
+{
+	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);
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "This application has been deprecated, please use the ${DB(family/key)} function instead.\n");
+		dep_warning = 1;
+	}
+	
+	argv = ast_strdupa(data);
+	if (!argv) {
+		ast_log(LOG_ERROR, "Memory allocation failed\n");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+
+	if (strchr(argv, '=') && strchr(argv, '/')) {
+		varname = strsep(&argv, "=");
+		family = strsep(&argv, "/");
+		if (strchr((void *)&argv, '|')) {
+			key = strsep(&argv, "|");
+			options = strsep(&argv, "\0");
+		} else
+			key = strsep(&argv, "\0");
+			
+		if (!varname || !family || !key) {
+			ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			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");
+			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");
+	}
+
+	LOCAL_USER_REMOVE(u);
+
+	return 0;
+}
+
+int unload_module(void)
 {
 	int retval;
 
 	retval = ast_unregister_application(dt_app);
 	retval |= ast_unregister_application(d_app);
+	retval |= ast_unregister_application(p_app);
+	retval |= ast_unregister_application(g_app);
+
+	STANDARD_HANGUP_LOCALUSERS;
 
 	return retval;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	int retval;
 
-	retval = ast_register_application(d_app, del_exec, d_synopsis, d_descrip);
+	retval = ast_register_application(g_app, get_exec, g_synopsis, g_descrip);
+	retval |= ast_register_application(p_app, put_exec, p_synopsis, p_descrip);
+	retval |= ast_register_application(d_app, del_exec, d_synopsis, d_descrip);
 	retval |= ast_register_application(dt_app, deltree_exec, dt_synopsis, dt_descrip);
-
+	
 	return retval;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Database Access Functions");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_dictate.c

@@ -22,30 +22,38 @@
  *
  * \brief Virtual Dictation Machine Application For Asterisk
  *
- * \author Anthony Minessale II <anthmct@yahoo.com>
- *
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <sys/stat.h>	/* for mkdir */
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include <sys/stat.h>
-
-#include "asterisk/paths.h" /* use ast_config_AST_SPOOL_DIR */
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/say.h"
+#include "asterisk/lock.h"
 #include "asterisk/app.h"
 
+static char *tdesc = "Virtual Dictation Machine";
 static char *app = "Dictate";
 static char *synopsis = "Virtual Dictation Machine";
-static char *desc = "  Dictate([<base_dir>[,<filename>]])\n"
+static char *desc = "  Dictate([<base_dir>])\n"
 "Start dictation machine using optional base dir for files.\n";
 
 
+STANDARD_LOCAL_USER;
+LOCAL_USER_DECL;
+
 typedef enum {
 	DFLAG_RECORD = (1 << 0),
 	DFLAG_PLAY = (1 << 1),
@@ -61,7 +69,7 @@ typedef enum {
 
 #define ast_toggle_flag(it,flag) if(ast_test_flag(it, flag)) ast_clear_flag(it, flag); else ast_set_flag(it, flag)
 
-static int play_and_wait(struct ast_channel *chan, char *file, char *digits)
+static int play_and_wait(struct ast_channel *chan, char *file, char *digits) 
 {
 	int res = -1;
 	if (!ast_streamfile(chan, file, chan->language)) {
@@ -72,19 +80,16 @@ static int play_and_wait(struct ast_channel *chan, char *file, char *digits)
 
 static int dictate_exec(struct ast_channel *chan, void *data)
 {
-	char *path = NULL, filein[256], *filename = "";
-	char *parse;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(base);
-		AST_APP_ARG(filename);
-	);
+	char *mydata, *argv[2], *path = NULL, filein[256];
 	char dftbase[256];
 	char *base;
 	struct ast_flags flags = {0};
 	struct ast_filestream *fs;
 	struct ast_frame *f = NULL;
+	struct localuser *u;
 	int ffactor = 320 * 80,
 		res = 0,
+		argc = 0,
 		done = 0,
 		oldr = 0,
 		lastop = 0,
@@ -94,44 +99,37 @@ static int dictate_exec(struct ast_channel *chan, void *data)
 		len = 0,
 		maxlen = 0,
 		mode = 0;
-
+		
+	LOCAL_USER_ADD(u);
+	
 	snprintf(dftbase, sizeof(dftbase), "%s/dictate", ast_config_AST_SPOOL_DIR);
-	if (!ast_strlen_zero(data)) {
-		parse = ast_strdupa(data);
-		AST_STANDARD_APP_ARGS(args, parse);
-	} else
-		args.argc = 0;
-
-	if (args.argc && !ast_strlen_zero(args.base)) {
-		base = args.base;
+	if (!ast_strlen_zero(data) && (mydata = ast_strdupa(data))) {
+		argc = ast_app_separate_args(mydata, '|', argv, sizeof(argv) / sizeof(argv[0]));
+	}
+	
+	if (argc) {
+		base = argv[0];
 	} else {
 		base = dftbase;
 	}
-	if (args.argc > 1 && args.filename) {
-		filename = args.filename;
-	}
+
 	oldr = chan->readformat;
 	if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR)) < 0) {
 		ast_log(LOG_WARNING, "Unable to set to linear mode.\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 
-	if (chan->_state != AST_STATE_UP) {
-		ast_answer(chan);
-	}
+	ast_answer(chan);
 	ast_safe_sleep(chan, 200);
-	for (res = 0; !res;) {
-		if (ast_strlen_zero(filename)) {
-			if (ast_app_getdata(chan, "dictate/enter_filename", filein, sizeof(filein), 0) ||
-				ast_strlen_zero(filein)) {
-				res = -1;
-				break;
-			}
-		} else {
-			ast_copy_string(filein, filename, sizeof(filein));
-			filename = "";
+	for(res = 0; !res;) {
+		if (ast_app_getdata(chan, "dictate/enter_filename", filein, sizeof(filein), 0) || 
+			ast_strlen_zero(filein)) {
+			res = -1;
+			break;
 		}
-		ast_mkdir(base, 0755);
+		
+		mkdir(base, 0755);
 		len = strlen(base) + strlen(filein) + 2;
 		if (!path || len > maxlen) {
 			path = alloca(len);
@@ -142,7 +140,7 @@ static int dictate_exec(struct ast_channel *chan, void *data)
 		}
 
 		snprintf(path, len, "%s/%s", base, filein);
-		fs = ast_writefile(path, "raw", NULL, O_CREAT|O_APPEND, 0, AST_FILE_MODE);
+		fs = ast_writefile(path, "raw", NULL, O_CREAT|O_APPEND, 0, 0700);
 		mode = DMODE_PLAY;
 		memset(&flags, 0, sizeof(flags));
 		ast_set_flag(&flags, DFLAG_PAUSE);
@@ -172,7 +170,7 @@ static int dictate_exec(struct ast_channel *chan, void *data)
 						if (speed > 4) {
 							speed = 1;
 						}
-						res = ast_say_number(chan, speed, AST_DIGIT_ANY, chan->language, NULL);
+						res = ast_say_number(chan, speed, AST_DIGIT_ANY, chan->language, (char *) NULL);
 						break;
 					case '7':
 						samples -= ffactor;
@@ -284,7 +282,7 @@ static int dictate_exec(struct ast_channel *chan, void *data)
 				case DMODE_RECORD:
 					if (lastop != DMODE_RECORD) {
 						int oflags = O_CREAT | O_WRONLY;
-						if (ast_test_flag(&flags, DFLAG_PAUSE)) {
+						if (ast_test_flag(&flags, DFLAG_PAUSE)) {						
 							digit = play_and_wait(chan, "dictate/record_mode", AST_DIGIT_ANY);
 							if (digit == 0) {
 								digit = play_and_wait(chan, "dictate/paused", AST_DIGIT_ANY);
@@ -300,7 +298,7 @@ static int dictate_exec(struct ast_channel *chan, void *data)
 						} else {
 							oflags |= O_APPEND;
 						}
-						fs = ast_writefile(path, "raw", NULL, oflags, 0, AST_FILE_MODE);
+						fs = ast_writefile(path, "raw", NULL, oflags, 0, 0700);
 						if (ast_test_flag(&flags, DFLAG_TRUNC)) {
 							ast_seekstream(fs, 0, SEEK_SET);
 							ast_clear_flag(&flags, DFLAG_TRUNC);
@@ -322,19 +320,40 @@ static int dictate_exec(struct ast_channel *chan, void *data)
 	if (oldr) {
 		ast_set_read_format(chan, oldr);
 	}
+	LOCAL_USER_REMOVE(u);
 	return 0;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
 	int res;
+
 	res = ast_unregister_application(app);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+	
 	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, dictate_exec, synopsis, desc);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Virtual Dictation Machine");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
+

apps/app_directed_pickup.c

@@ -20,153 +20,150 @@
  *
  * \brief Directed Call Pickup Support
  *
- * \author Joshua Colp <jcolp@digium.com>
- *
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/lock.h"
 #include "asterisk/app.h"
-#include "asterisk/features.h"
-
-#define PICKUPMARK "PICKUPMARK"
 
+static const char *tdesc = "Directed Call Pickup Application";
 static const char *app = "Pickup";
 static const char *synopsis = "Directed Call Pickup";
 static const char *descrip =
-"  Pickup([extension[@context][&extension2@[context]...]]):  This application can\n"
-"pickup any ringing channel that is calling the specified extension.  If no\n"
-"context is specified, the current context will be used. If you use the special\n"
-"string \"PICKUPMARK\" for the context parameter, for example 10@PICKUPMARK,\n"
-"this application tries to find a channel which has defined a ${PICKUPMARK}\n"
-"channel variable with the same value as \"extension\" (in this example, \"10\").\n"
-"When no parameter is specified, the application will pickup a channel matching\n"
-"the pickup group of the active channel.";
+"  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";
 
-/* Perform actual pickup between two channels */
-static int pickup_do(struct ast_channel *chan, struct ast_channel *target)
-{
-	int res = 0;
+STANDARD_LOCAL_USER;
 
-	ast_debug(1, "Call pickup on '%s' by '%s'\n", target->name, chan->name);
+LOCAL_USER_DECL;
 
-	if ((res = ast_answer(chan))) {
-		ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan->name);
-		return -1;
-	}
-
-	if ((res = ast_queue_control(chan, AST_CONTROL_ANSWER))) {
-		ast_log(LOG_WARNING, "Unable to queue answer on '%s'\n", chan->name);
-		return -1;
-	}
-
-	if ((res = ast_channel_masquerade(target, chan))) {
-		ast_log(LOG_WARNING, "Unable to masquerade '%s' into '%s'\n", chan->name, target->name);
-		return -1;
-	}
-
-	return res;
-}
-
-/* Helper function that determines whether a channel is capable of being picked up */
-static int can_pickup(struct ast_channel *chan)
-{
-	if (!chan->pbx && (chan->_state == AST_STATE_RINGING || chan->_state == AST_STATE_RING || chan->_state == AST_STATE_DOWN))
-		return 1;
-	else
-		return 0;
-}
-
-/* Attempt to pick up specified extension with context */
-static int pickup_by_exten(struct ast_channel *chan, const char *exten, const char *context)
-{
-	int res = -1;
-	struct ast_channel *target = NULL;
-
-	while ((target = ast_channel_walk_locked(target))) {
-		if ((!strcasecmp(target->macroexten, exten) || !strcasecmp(target->exten, exten)) &&
-			!strcasecmp(target->dialcontext, context) &&
-			(chan != target) && can_pickup(target)) {
-			res = pickup_do(chan, target);
-			ast_channel_unlock(target);
-			break;
-		}
-		ast_channel_unlock(target);
-	}
-
-	return res;
-}
-
-/* Attempt to pick up specified mark */
-static int pickup_by_mark(struct ast_channel *chan, const char *mark)
-{
-	int res = -1;
-	const char *tmp = NULL;
-	struct ast_channel *target = NULL;
-
-	while ((target = ast_channel_walk_locked(target))) {
-		if ((tmp = pbx_builtin_getvar_helper(target, PICKUPMARK)) &&
-			!strcasecmp(tmp, mark) &&
-			can_pickup(target)) {
-			res = pickup_do(chan, target);
-			ast_channel_unlock(target);
-			break;
-		}
-		ast_channel_unlock(target);
-	}
-
-	return res;
-}
-
-/* Main application entry point */
 static int pickup_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
-	char *tmp = ast_strdupa(data);
-	char *exten = NULL, *context = NULL;
+	struct localuser *u = NULL;
+	struct ast_channel *origin = NULL, *target = NULL;
+	char *tmp = NULL, *exten = NULL, *context = NULL;
+	char workspace[256] = "";
 
 	if (ast_strlen_zero(data)) {
-		res = ast_pickup_call(chan);
-		return res;
+		ast_log(LOG_WARNING, "Pickup requires an argument (extension) !\n");
+		return -1;	
+	}
+
+	LOCAL_USER_ADD(u);
+	
+	/* Get the extension and context if present */
+	exten = data;
+	context = strchr(data, '@');
+	if (context) {
+		*context = '\0';
+		context++;
+	}
+
+	/* Find a channel to pickup */
+	origin = ast_get_channel_by_exten_locked(exten, context);
+	if (origin && origin->cdr) {
+		ast_cdr_getvar(origin->cdr, "dstchannel", &tmp, workspace,
+			       sizeof(workspace), 0);
+		if (tmp) {
+			/* We have a possible channel... now we need to find it! */
+			target = ast_get_channel_by_name_locked(tmp);
+		} else {
+			ast_log(LOG_DEBUG, "No target channel found.\n");
+			res = -1;
+		}
+		ast_mutex_unlock(&origin->lock);
+	} else {
+		if (origin)
+			ast_mutex_unlock(&origin->lock);
+		ast_log(LOG_DEBUG, "No originating channel found.\n");
 	}
 	
-	/* Parse extension (and context if there) */
-	while (!ast_strlen_zero(tmp) && (exten = strsep(&tmp, "&"))) {
-		if ((context = strchr(exten, '@')))
-			*context++ = '\0';
-		if (!ast_strlen_zero(context) && !strcasecmp(context, PICKUPMARK)) {
-			if (!pickup_by_mark(chan, exten))
-				break;
-		} else {
-			if (!pickup_by_exten(chan, exten, !ast_strlen_zero(context) ? context : chan->context))
-				break;
+	if (res)
+		goto out;
+
+	if (target && (!target->pbx) && ((target->_state == AST_STATE_RINGING) || (target->_state == AST_STATE_RING))) {
+		ast_log(LOG_DEBUG, "Call pickup on chan '%s' by '%s'\n", target->name,
+			chan->name);
+		res = ast_answer(chan);
+		if (res) {
+			ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan->name);
+			res = -1;
+			goto out;
 		}
-		ast_log(LOG_NOTICE, "No target channel found for %s.\n", exten);
+		res = ast_queue_control(chan, AST_CONTROL_ANSWER);
+		if (res) {
+			ast_log(LOG_WARNING, "Unable to queue answer on '%s'\n",
+				chan->name);
+			res = -1;
+			goto out;
+		}
+		res = ast_channel_masquerade(target, chan);
+		if (res) {
+			ast_log(LOG_WARNING, "Unable to masquerade '%s' into '%s'\n", chan->name, target->name);
+			res = -1;
+			goto out;
+		}
+	} else {
+		ast_log(LOG_DEBUG, "No call pickup possible...\n");
+		res = -1;
 	}
+	/* Done */
+ out:
+	if (target) 
+		ast_mutex_unlock(&target->lock);
+	
+	LOCAL_USER_REMOVE(u);
 
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
 	int res;
 
 	res = ast_unregister_application(app);
+	
+	STANDARD_HANGUP_LOCALUSERS;
 
 	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, pickup_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Directed Call Pickup Application");
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_disa.c

@@ -22,19 +22,23 @@
  * \brief DISA -- Direct Inward System Access Application
  *
  * \author Jim Dixon <jim@lambdatel.com>
- *
+ * 
  * \ingroup applications
  */
+ 
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <math.h>
+#include <sys/time.h>
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include <math.h>
-#include <sys/time.h>
-
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/app.h"
 #include "asterisk/indications.h"
@@ -43,62 +47,57 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/translate.h"
 #include "asterisk/ulaw.h"
 #include "asterisk/callerid.h"
-#include "asterisk/stringfields.h"
+
+static char *tdesc = "DISA (Direct Inward System Access) Application";
 
 static char *app = "DISA";
 
 static char *synopsis = "DISA (Direct Inward System Access)";
 
-static char *descrip =
-"DISA(<numeric passcode>[,<context>[,<cid>[,mailbox[,options]]]]) or\n"
-"DISA(<filename>[,,,,options])\n"
-"The DISA, Direct Inward System Access, application allows someone from \n"
-"outside the telephone switch (PBX) to obtain an \"internal\" system \n"
-"dialtone and to place calls from it as if they were placing a call from \n"
-"within the switch.\n"
-"DISA plays a dialtone. The user enters their numeric passcode, followed by\n"
-"the pound sign (#). If the passcode is correct, the user is then given\n"
-"system dialtone within <context> on which a call may be placed. If the user\n"
-"enters an invalid extension and extension \"i\" exists in the specified\n"
-"context, it will be used.\n"
-"\n"
-"If you need to present a DISA dialtone without entering a password, simply\n"
-"set <passcode> to \"no-password\".\n"
-"\n"
-"Be aware that using this may compromise the security of your PBX.\n"
-"\n"
-"The arguments to this application (in extensions.conf) allow either\n"
-"specification of a single global passcode (that everyone uses), or\n"
-"individual passcodes contained in a file.\n"
-"\n"
-"The file that contains the passcodes (if used) allows a complete\n"
-"specification of all of the same arguments available on the command\n"
-"line, with the sole exception of the options. The file may contain blank\n"
-"lines, or comments starting with \"#\" or \";\".\n"
-"\n"
-"<context> specifies the dialplan context in which the user-entered extension\n"
-"will be matched. If no context is specified, the DISA application defaults\n"
-"the context to \"disa\". Presumably a normal system will have a special\n"
-"context set up for DISA use with some or a lot of restrictions.\n"
-"\n"
-"<cid> specifies a new (different) callerid to be used for this call.\n"
-"\n"
-"<mailbox[@context]> will cause a stutter-dialtone (indication \"dialrecall\")\n"
-"to be used, if the specified mailbox contains any new messages.\n"
-"\n"
-"The following options are available:\n"
-"  n - the DISA application will not answer initially.\n"
-"  p - the extension entered will be considered complete when a '#' is entered.\n";
+static char *descrip = 
+	"DISA(<numeric passcode>[|<context>]) or disa(<filename>)\n"
+	"The DISA, Direct Inward System Access, application allows someone from \n"
+	"outside the telephone switch (PBX) to obtain an \"internal\" system \n"
+	"dialtone and to place calls from it as if they were placing a call from \n"
+	"within the switch.\n"
+	"DISA plays a dialtone. The user enters their numeric passcode, followed by\n"
+	"the pound sign (#). If the passcode is correct, the user is then given\n"
+	"system dialtone on which a call may be placed. Obviously, this type\n"
+	"of access has SERIOUS security implications, and GREAT care must be\n"
+	"taken NOT to compromise your security.\n\n"
+	"There is a possibility of accessing DISA without password. Simply\n"
+	"exchange your password with \"no-password\".\n\n"
+	"    Example: exten => s,1,DISA(no-password|local)\n\n"
+	"Be aware that using this compromises the security of your PBX.\n\n"
+	"The arguments to this application (in extensions.conf) allow either\n"
+	"specification of a single global passcode (that everyone uses), or\n"
+	"individual passcodes contained in a file. It also allow specification\n"
+	"of the context on which the user will be dialing. If no context is\n"
+	"specified, the DISA application defaults the context to \"disa\".\n"
+	"Presumably a normal system will have a special context set up\n"
+	"for DISA use with some or a lot of restrictions. \n\n"
+	"The file that contains the passcodes (if used) allows specification\n"
+	"of either just a passcode (defaulting to the \"disa\" context, or\n"
+	"passcode|context on each line of the file. The file may contain blank\n"
+	"lines, or comments starting with \"#\" or \";\". In addition, the\n"
+	"above arguments may have |new-callerid-string appended to them, to\n"
+	"specify a new (different) callerid to be used for this call, for\n"
+	"example: numeric-passcode|context|\"My Phone\" <(234) 123-4567> or \n"
+	"full-pathname-of-passcode-file|\"My Phone\" <(234) 123-4567>.  Last\n"
+	"but not least, |mailbox[@context] may be appended, which will cause\n"
+	"a stutter-dialtone (indication \"dialrecall\") to be used, if the\n"
+	"specified mailbox contains any new messages, for example:\n"
+	"numeric-passcode|context||1234 (w/a changing callerid).  Note that\n"
+	"in the case of specifying the numeric-passcode, the context must be\n"
+	"specified if the callerid is specified also.\n\n"
+	"If login is successful, the application looks up the dialed number in\n"
+	"the specified (or default) context, and executes it if found.\n"
+	"If the user enters an invalid extension and extension \"i\" (invalid) \n"
+	"exists in the context, it will be used.\n";
 
-enum {
-	NOANSWER_FLAG = (1 << 0),
-	POUND_TO_END_FLAG = (1 << 1),
-} option_flags;
+STANDARD_LOCAL_USER;
 
-AST_APP_OPTIONS(app_opts, {
-	AST_APP_OPTION('n', NOANSWER_FLAG),
-	AST_APP_OPTION('p', POUND_TO_END_FLAG),
-});
+LOCAL_USER_DECL;
 
 static void play_dialtone(struct ast_channel *chan, char *mailbox)
 {
@@ -115,140 +114,168 @@ static void play_dialtone(struct ast_channel *chan, char *mailbox)
 
 static int disa_exec(struct ast_channel *chan, void *data)
 {
-	int i = 0, j, k = 0, did_ignore = 0, special_noanswer = 0;
-	int firstdigittimeout = (chan->pbx ? chan->pbx->rtimeout * 1000 : 20000);
-	int digittimeout = (chan->pbx ? chan->pbx->dtimeout * 1000 : 10000);
-	struct ast_flags flags;
-	char *tmp, exten[AST_MAX_EXTENSION] = "", acctcode[20]="";
+	int i,j,k,x,did_ignore;
+	int firstdigittimeout = 20000;
+	int digittimeout = 10000;
+	struct localuser *u;
+	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;
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(passcode);
 		AST_APP_ARG(context);
 		AST_APP_ARG(cid);
 		AST_APP_ARG(mailbox);
-		AST_APP_ARG(options);
 	);
 
 	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "DISA requires an argument (passcode/passcode file)\n");
+		ast_log(LOG_WARNING, "disa requires an argument (passcode/passcode file)\n");
 		return -1;
 	}
 
-	ast_debug(1, "Digittimeout: %d\n", digittimeout);
-	ast_debug(1, "Responsetimeout: %d\n", firstdigittimeout);
+	LOCAL_USER_ADD(u);
+	
+	if (chan->pbx) {
+		firstdigittimeout = chan->pbx->rtimeout*1000;
+		digittimeout = chan->pbx->dtimeout*1000;
+	}
+	
+	if (ast_set_write_format(chan,AST_FORMAT_ULAW)) {
+		ast_log(LOG_WARNING, "Unable to set write format to Mu-law on %s\n",chan->name);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	if (ast_set_read_format(chan,AST_FORMAT_ULAW)) {
+		ast_log(LOG_WARNING, "Unable to set read format to Mu-law on %s\n",chan->name);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	
+	ast_log(LOG_DEBUG, "Digittimeout: %d\n", digittimeout);
+	ast_log(LOG_DEBUG, "Responsetimeout: %d\n", firstdigittimeout);
 
 	tmp = ast_strdupa(data);
+	if (!tmp) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}	
 
 	AST_STANDARD_APP_ARGS(args, tmp);
 
-	if (ast_strlen_zero(args.context))
-		args.context = "disa";
+	if (ast_strlen_zero(args.context)) 
+		args.context = "disa";	
 	if (ast_strlen_zero(args.mailbox))
 		args.mailbox = "";
-	if (!ast_strlen_zero(args.options))
-		ast_app_parse_options(app_opts, &flags, NULL, args.options);
 
-	ast_debug(1, "Mailbox: %s\n",args.mailbox);
+	ast_log(LOG_DEBUG, "Mailbox: %s\n",args.mailbox);
+	
+	if (chan->_state != AST_STATE_UP) {
+		/* answer */
+		ast_answer(chan);
+	}
+	i = k = x = 0; /* k is 0 for pswd entry, 1 for ext entry */
+	did_ignore = 0;
+	exten[0] = 0;
+	acctcode[0] = 0;
+	/* can we access DISA without password? */ 
 
-	if (!ast_test_flag(&flags, NOANSWER_FLAG)) {
-		if (chan->_state != AST_STATE_UP) {
-			/* answer */
-			ast_answer(chan);
-		}
-	} else special_noanswer = 1;
-
-	ast_debug(1, "Context: %s\n",args.context);
+	ast_log(LOG_DEBUG, "Context: %s\n",args.context);
 
 	if (!strcasecmp(args.passcode, "no-password")) {
 		k |= 1; /* We have the password */
-		ast_debug(1, "DISA no-password login success\n");
+		ast_log(LOG_DEBUG, "DISA no-password login success\n");
 	}
-
 	lastdigittime = ast_tvnow();
 
 	play_dialtone(chan, args.mailbox);
 
-	ast_set_flag(chan, AST_FLAG_END_DTMF_ONLY);
-
 	for (;;) {
 		  /* if outa time, give em reorder */
-		if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) > ((k&2) ? digittimeout : firstdigittimeout)) {
-			ast_debug(1,"DISA %s entry timeout on chan %s\n",
+		if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) > 
+		    ((k&2) ? digittimeout : firstdigittimeout))
+		{
+			ast_log(LOG_DEBUG,"DISA %s entry timeout on chan %s\n",
 				((k&1) ? "extension" : "password"),chan->name);
 			break;
 		}
-
 		if ((res = ast_waitfor(chan, -1) < 0)) {
-			ast_debug(1, "Waitfor returned %d\n", res);
+			ast_log(LOG_DEBUG, "Waitfor returned %d\n", res);
 			continue;
 		}
-
-		if (!(f = ast_read(chan))) {
-			ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
+			
+		f = ast_read(chan);
+		if (f == NULL) 
+		{
+			LOCAL_USER_REMOVE(u);
 			return -1;
 		}
-
-		if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP)) {
+		if ((f->frametype == AST_FRAME_CONTROL) &&
+		    (f->subclass == AST_CONTROL_HANGUP))
+		{
 			ast_frfree(f);
-			ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
+			LOCAL_USER_REMOVE(u);
 			return -1;
 		}
-
-		/* If the frame coming in is not DTMF, just drop it and continue */
-		if (f->frametype != AST_FRAME_DTMF) {
+		if (f->frametype == AST_FRAME_VOICE) {
+			ast_frfree(f);
+			continue;
+		}
+		  /* if not DTMF, just do it again */
+		if (f->frametype != AST_FRAME_DTMF) 
+		{
 			ast_frfree(f);
 			continue;
 		}
 
 		j = f->subclass;  /* save digit */
 		ast_frfree(f);
-
-		if (!i) {
-			k |= 2; /* We have the first digit */
+		if (i == 0) 
+		{
+			k|=2; /* We have the first digit */ 
 			ast_playtones_stop(chan);
 		}
-
 		lastdigittime = ast_tvnow();
-
-		/* got a DTMF tone */
-		if (i < AST_MAX_EXTENSION) { /* if still valid number of digits */
-			if (!(k&1)) { /* if in password state */
-				if (j == '#') { /* end of password */
+		  /* got a DTMF tone */
+		if (i < AST_MAX_EXTENSION) /* if still valid number of digits */
+		{
+			if (!(k&1)) /* if in password state */
+			{
+				if (j == '#') /* end of password */
+				{
 					  /* see if this is an integer */
-					if (sscanf(args.passcode,"%30d",&j) < 1) { /* nope, it must be a filename */
+					if (sscanf(args.passcode,"%d",&j) < 1)
+					   { /* nope, it must be a filename */
 						fp = fopen(args.passcode,"r");
-						if (!fp) {
+						if (!fp)
+						   {
 							ast_log(LOG_WARNING,"DISA password file %s not found on chan %s\n",args.passcode,chan->name);
-							ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
+							LOCAL_USER_REMOVE(u);
 							return -1;
-						}
+						   }
 						pwline[0] = 0;
-						while(fgets(pwline,sizeof(pwline) - 1,fp)) {
-							if (!pwline[0])
-								continue;
-							if (pwline[strlen(pwline) - 1] == '\n')
+						while(fgets(pwline,sizeof(pwline) - 1,fp))
+						   {
+							if (!pwline[0]) continue;
+							if (pwline[strlen(pwline) - 1] == '\n') 
 								pwline[strlen(pwline) - 1] = 0;
-							if (!pwline[0])
-								continue;
-							 /* skip comments */
-							if (pwline[0] == '#')
-								continue;
-							if (pwline[0] == ';')
-								continue;
+							if (!pwline[0]) continue;
+							  /* skip comments */
+							if (pwline[0] == '#') continue;
+							if (pwline[0] == ';') continue;
 
 							AST_STANDARD_APP_ARGS(args, pwline);
+			
+							ast_log(LOG_DEBUG, "Mailbox: %s\n",args.mailbox);
 
-							ast_debug(1, "Mailbox: %s\n",args.mailbox);
-
-							/* password must be in valid format (numeric) */
-							if (sscanf(args.passcode,"%30d", &j) < 1)
-								continue;
-							 /* if we got it */
+							  /* password must be in valid format (numeric) */
+							if (sscanf(args.passcode,"%d",&j) < 1) continue;
+							  /* if we got it */
 							if (!strcmp(exten,args.passcode)) {
 								if (ast_strlen_zero(args.context))
 									args.context = "disa";
@@ -256,17 +283,18 @@ static int disa_exec(struct ast_channel *chan, void *data)
 									args.mailbox = "";
 								break;
 							}
-						}
+						   }
 						fclose(fp);
-					}
-					/* compare the two */
-					if (strcmp(exten,args.passcode)) {
+					   }
+					  /* compare the two */
+					if (strcmp(exten,args.passcode))
+					{
 						ast_log(LOG_WARNING,"DISA on chan %s got bad password %s\n",chan->name,exten);
 						goto reorder;
 
 					}
 					 /* password good, set to dial state */
-					ast_debug(1,"DISA on chan %s password is good\n",chan->name);
+					ast_log(LOG_DEBUG,"DISA on chan %s password is good\n",chan->name);
 					play_dialtone(chan, args.mailbox);
 
 					k|=1; /* In number mode */
@@ -274,32 +302,15 @@ static int disa_exec(struct ast_channel *chan, void *data)
 					exten[sizeof(acctcode)] = 0;
 					ast_copy_string(acctcode, exten, sizeof(acctcode));
 					exten[0] = 0;
-					ast_debug(1,"Successful DISA log-in on chan %s\n", chan->name);
+					ast_log(LOG_DEBUG,"Successful DISA log-in on chan %s\n",chan->name);
 					continue;
 				}
-			} else {
-				if (j == '#') { /* end of extension .. maybe */
-					if (i == 0 && 
-							(ast_matchmore_extension(chan, args.context, "#", 1, chan->cid.cid_num) ||
-							 ast_exists_extension(chan, args.context, "#", 1, chan->cid.cid_num)) ) {
-						/* Let the # be the part of, or the entire extension */
-					} else {
-						break;
-					}
-				}
 			}
 
 			exten[i++] = j;  /* save digit */
 			exten[i] = 0;
-			if (!(k&1))
-				continue; /* if getting password, continue doing it */
-			/* if this exists */
-
-			/* user wants end of number, remove # */
-			if (ast_test_flag(&flags, POUND_TO_END_FLAG) && j == '#') {
-				exten[--i] = 0;
-				break;
-			}
+			if (!(k&1)) continue; /* if getting password, continue doing it */
+			  /* if this exists */
 
 			if (ast_ignore_pattern(args.context, exten)) {
 				play_dialtone(chan, "");
@@ -310,15 +321,13 @@ static int disa_exec(struct ast_channel *chan, void *data)
 					did_ignore = 0;
 				}
 
-			/* if can do some more, do it */
+			  /* if can do some more, do it */
 			if (!ast_matchmore_extension(chan,args.context,exten,1, chan->cid.cid_num)) {
 				break;
 			}
 		}
 	}
 
-	ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
-
 	if (k == 3) {
 		int recheck = 0;
 		struct ast_flags flags = { AST_CDR_FLAG_POSTED };
@@ -332,17 +341,18 @@ static int disa_exec(struct ast_channel *chan, void *data)
 		if (!recheck || ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
 			ast_playtones_stop(chan);
 			/* We're authenticated and have a target extension */
-			if (!ast_strlen_zero(args.cid)) {
+			if (!ast_strlen_zero(args.cid))
+			{
 				ast_callerid_split(args.cid, ourcidname, sizeof(ourcidname), ourcidnum, sizeof(ourcidnum));
 				ast_set_callerid(chan, ourcidnum, ourcidname, ourcidnum);
 			}
 
 			if (!ast_strlen_zero(acctcode))
-				ast_string_field_set(chan, accountcode, acctcode);
+				ast_copy_string(chan->accountcode, acctcode, sizeof(chan->accountcode));
 
-			if (special_noanswer) flags.flags = 0;
 			ast_cdr_reset(chan->cdr, &flags);
 			ast_explicit_goto(chan, args.context, exten, 1);
+			LOCAL_USER_REMOVE(u);
 			return 0;
 		}
 	}
@@ -350,23 +360,53 @@ static int disa_exec(struct ast_channel *chan, void *data)
 	/* Received invalid, but no "i" extension exists in the given context */
 
 reorder:
-	/* Play congestion for a bit */
-	ast_indicate(chan, AST_CONTROL_CONGESTION);
-	ast_safe_sleep(chan, 10*1000);
 
+	ast_indicate(chan,AST_CONTROL_CONGESTION);
+	/* something is invalid, give em reorder for several seconds */
+	time(&rstart);
+	while(time(NULL) < rstart + 10)
+	{
+		if (ast_waitfor(chan, -1) < 0)
+			break;
+		f = ast_read(chan);
+		if (!f)
+			break;
+		ast_frfree(f);
+	}
 	ast_playtones_stop(chan);
-
+	LOCAL_USER_REMOVE(u);
 	return -1;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, disa_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "DISA (Direct Inward System Access) Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key(void)
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_dumpchan.c

@@ -22,41 +22,53 @@
 /*! \file
  *
  * \brief Application to dump channel variables
- *
- * \author Anthony Minessale <anthmct@yahoo.com>
- *
+ * 
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
-#include "asterisk/channel.h"
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
+#include "asterisk/lock.h"
+#include "asterisk/utils.h"
 
+static char *tdesc = "Dump Info About The Calling Channel";
 static char *app = "DumpChan";
 static char *synopsis = "Dump Info About The Calling Channel";
-static char *desc =
-	"   DumpChan([<min_verbose_level>])\n"
-	"Displays information on channel and listing of all channel\n"
-	"variables. If min_verbose_level is specified, output is only\n"
-	"displayed when the verbose level is currently set to that number\n"
-	"or greater. \n";
+static char *desc = 
+"   DumpChan([<min_verbose_level>])\n"
+"Displays information on channel and listing of all channel\n"
+"variables. If min_verbose_level is specified, output is only\n"
+"displayed when the verbose level is currently set to that number\n"
+"or greater. \n";
 
+STANDARD_LOCAL_USER;
 
-static int serialize_showchan(struct ast_channel *c, char *buf, size_t size)
+LOCAL_USER_DECL;
+
+static int ast_serialize_showchan(struct ast_channel *c, char *buf, size_t size)
 {
 	struct timeval now;
-	long elapsed_seconds = 0;
-	int hour = 0, min = 0, sec = 0;
-	char cgrp[BUFSIZ/2];
-	char pgrp[BUFSIZ/2];
-	char formatbuf[BUFSIZ/2];
-
+	long elapsed_seconds=0;
+	int hour=0, min=0, sec=0;
+	char cgrp[256];
+	char pgrp[256];
+	
 	now = ast_tvnow();
-	memset(buf, 0, size);
+	memset(buf,0,size);
 	if (!c)
 		return 0;
 
@@ -67,94 +79,115 @@ static int serialize_showchan(struct ast_channel *c, char *buf, size_t size)
 		sec = elapsed_seconds % 60;
 	}
 
-	snprintf(buf,size,
-			"Name=               %s\n"
-			"Type=               %s\n"
-			"UniqueID=           %s\n"
-			"CallerIDNum=        %s\n"
-			"CallerIDName=       %s\n"
-			"DNIDDigits=         %s\n"
-			"RDNIS=              %s\n"
-			"Language=           %s\n"
-			"State=              %s (%d)\n"
-			"Rings=              %d\n"
-			"NativeFormat=       %s\n"
-			"WriteFormat=        %s\n"
-			"ReadFormat=         %s\n"
-			"RawWriteFormat=     %s\n"
-			"RawReadFormat=      %s\n"
-			"1stFileDescriptor=  %d\n"
-			"Framesin=           %d %s\n"
-			"Framesout=          %d %s\n"
-			"TimetoHangup=       %ld\n"
-			"ElapsedTime=        %dh%dm%ds\n"
-			"Context=            %s\n"
-			"Extension=          %s\n"
-			"Priority=           %d\n"
-			"CallGroup=          %s\n"
-			"PickupGroup=        %s\n"
-			"Application=        %s\n"
-			"Data=               %s\n"
-			"Blocking_in=        %s\n",
-			c->name,
-			c->tech->type,
-			c->uniqueid,
-			S_OR(c->cid.cid_num, "(N/A)"),
-			S_OR(c->cid.cid_name, "(N/A)"),
-			S_OR(c->cid.cid_dnid, "(N/A)"),
-			S_OR(c->cid.cid_rdnis, "(N/A)"),
-			c->language,
-			ast_state2str(c->_state),
-			c->_state,
-			c->rings,
-			ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->nativeformats),
-			ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->writeformat),
-			ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->readformat),
-			ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->rawwriteformat),
-			ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->rawreadformat),
-			c->fds[0], c->fin & ~DEBUGCHAN_FLAG, (c->fin & DEBUGCHAN_FLAG) ? " (DEBUGGED)" : "",
-			c->fout & ~DEBUGCHAN_FLAG, (c->fout & DEBUGCHAN_FLAG) ? " (DEBUGGED)" : "", (long)c->whentohangup,
-			hour,
-			min,
-			sec,
-			c->context,
-			c->exten,
-			c->priority,
-			ast_print_group(cgrp, sizeof(cgrp), c->callgroup),
-			ast_print_group(pgrp, sizeof(pgrp), c->pickupgroup),
-			( c->appl ? c->appl : "(N/A)" ),
-			( c-> data ? S_OR(c->data, "(Empty)") : "(None)"),
-			(ast_test_flag(c, AST_FLAG_BLOCKING) ? c->blockproc : "(Not Blocking)"));
+	snprintf(buf,size, 
+			 "Name=               %s\n"
+			 "Type=               %s\n"
+			 "UniqueID=           %s\n"
+			 "CallerID=           %s\n"
+			 "CallerIDName=       %s\n"
+			 "DNIDDigits=         %s\n"
+			 "State=              %s (%d)\n"
+			 "Rings=              %d\n"
+			 "NativeFormat=       %d\n"
+			 "WriteFormat=        %d\n"
+			 "ReadFormat=         %d\n"
+			 "1stFileDescriptor=  %d\n"
+			 "Framesin=           %d %s\n"
+			 "Framesout=          %d %s\n"
+			 "TimetoHangup=       %ld\n"
+			 "ElapsedTime=        %dh%dm%ds\n"
+			 "Context=            %s\n"
+			 "Extension=          %s\n"
+			 "Priority=           %d\n"
+			 "CallGroup=          %s\n"
+			 "PickupGroup=        %s\n"
+			 "Application=        %s\n"
+			 "Data=               %s\n"
+			 "Blocking_in=        %s\n",
+			 c->name,
+			 c->type,
+			 c->uniqueid,
+			 (c->cid.cid_num ? c->cid.cid_num : "(N/A)"),
+			 (c->cid.cid_name ? c->cid.cid_name : "(N/A)"),
+			 (c->cid.cid_dnid ? c->cid.cid_dnid : "(N/A)" ),
+			 ast_state2str(c->_state),
+			 c->_state,
+			 c->rings,
+			 c->nativeformats,
+			 c->writeformat,
+			 c->readformat,
+			 c->fds[0], c->fin & 0x7fffffff, (c->fin & 0x80000000) ? " (DEBUGGED)" : "",
+			 c->fout & 0x7fffffff, (c->fout & 0x80000000) ? " (DEBUGGED)" : "", (long)c->whentohangup,
+			 hour,
+			 min,
+			 sec,
+			 c->context,
+			 c->exten,
+			 c->priority,
+			 ast_print_group(cgrp, sizeof(cgrp), c->callgroup),
+			 ast_print_group(pgrp, sizeof(pgrp), c->pickupgroup),
+			 ( c->appl ? c->appl : "(N/A)" ),
+			 ( c-> data ? (!ast_strlen_zero(c->data) ? c->data : "(Empty)") : "(None)"),
+			 (ast_test_flag(c, AST_FLAG_BLOCKING) ? c->blockproc : "(Not Blocking)"));
 
 	return 0;
 }
 
 static int dumpchan_exec(struct ast_channel *chan, void *data)
 {
-	struct ast_str *vars = ast_str_alloca(BUFSIZ * 4); /* XXX very large! */
+	int res=0;
+	struct localuser *u;
+	char vars[1024];
 	char info[1024];
 	int level = 0;
 	static char *line = "================================================================================";
+	
+	LOCAL_USER_ADD(u);
 
-	if (!ast_strlen_zero(data))
+	if (!ast_strlen_zero(data)) {
 		level = atoi(data);
+	}
 
-	pbx_builtin_serialize_variables(chan, &vars);
-	serialize_showchan(chan, info, sizeof(info));
+	pbx_builtin_serialize_variables(chan, vars, sizeof(vars));
+	ast_serialize_showchan(chan, info, sizeof(info));
 	if (option_verbose >= level)
-		ast_verbose("\nDumping Info For Channel: %s:\n%s\nInfo:\n%s\nVariables:\n%s%s\n", chan->name, line, info, vars->str, line);
+		ast_verbose("\nDumping Info For Channel: %s:\n%s\nInfo:\n%s\nVariables:\n%s%s\n",chan->name, line, info, vars, line);
 
-	return 0;
+	LOCAL_USER_REMOVE(u);
+	
+	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, dumpchan_exec, synopsis, desc);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dump Info About The Calling Channel");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
+

apps/app_echo.c

@@ -19,69 +19,105 @@
 /*! \file
  *
  * \brief Echo application -- play back what you hear to evaluate latency
- *
- * \author Mark Spencer <markster@digium.com>
- *
+ * 
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <unistd.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/lock.h"
 #include "asterisk/file.h"
-#include "asterisk/module.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+
+static char *tdesc = "Simple Echo Application";
 
 static char *app = "Echo";
 
-static char *synopsis = "Echo audio, video, or DTMF back to the calling party";
+static char *synopsis = "Echo audio read back to the user";
 
-static char *descrip =
-"  Echo(): This application will echo any audio, video, or DTMF frames read from\n"
-"the calling channel back to itself. If the DTMF digit '#' is received, the\n"
-"application will exit.\n";
+static char *descrip = 
+"  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;
+
+LOCAL_USER_DECL;
 
 static int echo_exec(struct ast_channel *chan, void *data)
 {
-	int res = -1;
-	int format;
-
-	format = ast_best_codec(chan->nativeformats);
-	ast_set_write_format(chan, format);
-	ast_set_read_format(chan, format);
-
-	while (ast_waitfor(chan, -1) > -1) {
-		struct ast_frame *f = ast_read(chan);
-		if (!f) {
+	int res=-1;
+	struct localuser *u;
+	struct ast_frame *f;
+	LOCAL_USER_ADD(u);
+	ast_set_write_format(chan, ast_best_codec(chan->nativeformats));
+	ast_set_read_format(chan, ast_best_codec(chan->nativeformats));
+	/* Do our thing here */
+	while(ast_waitfor(chan, -1) > -1) {
+		f = ast_read(chan);
+		if (!f)
 			break;
-		}
 		f->delivery.tv_sec = 0;
 		f->delivery.tv_usec = 0;
-		if (ast_write(chan, f)) {
-			ast_frfree(f);
-			goto end;
-		}
-		if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '#')) {
-			res = 0;
-			ast_frfree(f);
-			goto end;
+		if (f->frametype == AST_FRAME_VOICE) {
+			if (ast_write(chan, f)) 
+				break;
+		} else if (f->frametype == AST_FRAME_VIDEO) {
+			if (ast_write(chan, f)) 
+				break;
+		} else if (f->frametype == AST_FRAME_DTMF) {
+			if (f->subclass == '#') {
+				res = 0;
+				break;
+			} else
+				if (ast_write(chan, f))
+					break;
 		}
 		ast_frfree(f);
 	}
-end:
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, echo_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Simple Echo Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_enumlookup.c

@@ -0,0 +1,270 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Enumlookup - lookup entry in ENUM
+ *
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <ctype.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/options.h"
+#include "asterisk/config.h"
+#include "asterisk/module.h"
+#include "asterisk/enum.h"
+#include "asterisk/utils.h"
+#include "asterisk/app.h"
+#include "asterisk/options.h"
+
+static char *tdesc = "ENUM Lookup";
+
+static char *app = "EnumLookup";
+
+static char *synopsis = "Lookup number in ENUM";
+
+static char *descrip =
+"  EnumLookup(exten[|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"
+"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"
+"  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"
+
+static char h323driver[80] = "";
+#define H323DRIVERDEFAULT "H323"
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+/*--- enumlookup_exec: Look up number in ENUM and return result */
+static int enumlookup_exec(struct ast_channel *chan, void *data)
+{
+	int res=0,priority_jump=0;
+	char tech[80];
+	char dest[80];
+	char tmp[256];
+	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");
+		return -1;
+	}
+		
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The application EnumLookup is deprecated.  Please use the ENUMLOOKUP() function instead.\n");
+		dep_warning = 1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	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);
+
+	tech[0] = '\0';
+	dest[0] = '\0';
+
+	if (args.o) {
+		if (strchr(args.o, 'j'))
+			priority_jump = 1;
+	}
+
+	res = ast_get_enum(chan, args.d, dest, sizeof(dest), tech, sizeof(tech), NULL, NULL);
+	
+	if (!res) {	/* Failed to do a lookup */
+		if (priority_jump || option_priority_jumping) {
+			/* Look for a "busy" place */
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		}
+		pbx_builtin_setvar_helper(chan, "ENUMSTATUS", "ERROR");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+	pbx_builtin_setvar_helper(chan, "ENUMSTATUS", tech);
+	/* Parse it out */
+	if (res > 0) {
+		if (!strcasecmp(tech, "SIP")) {
+			c = dest;
+			if (!strncmp(c, "sip:", 4))
+				c += 4;
+			snprintf(tmp, sizeof(tmp), "SIP/%s", c);
+			pbx_builtin_setvar_helper(chan, "ENUM", tmp);
+		} else if (!strcasecmp(tech, "h323")) {
+			c = dest;
+			if (!strncmp(c, "h323:", 5))
+				c += 5;
+			snprintf(tmp, sizeof(tmp), "%s/%s", h323driver, c);
+/* do a s!;.*!! on the H323 URI */
+			t = strchr(c,';');
+                       if (t)
+				*t = 0;
+			pbx_builtin_setvar_helper(chan, "ENUM", tmp);
+		} else if (!strcasecmp(tech, "iax")) {
+			c = dest;
+			if (!strncmp(c, "iax:", 4))
+				c += 4;
+			snprintf(tmp, sizeof(tmp), "IAX/%s", c);
+			pbx_builtin_setvar_helper(chan, "ENUM", tmp);
+		} else if (!strcasecmp(tech, "iax2")) {
+			c = dest;
+			if (!strncmp(c, "iax2:", 5))
+				c += 5;
+			snprintf(tmp, sizeof(tmp), "IAX2/%s", c);
+			pbx_builtin_setvar_helper(chan, "ENUM", tmp);
+		} else if (!strcasecmp(tech, "tel")) {
+			c = dest;
+			if (!strncmp(c, "tel:", 4))
+				c += 4;
+
+			if (c[0] != '+') {
+				ast_log(LOG_NOTICE, "tel: uri must start with a \"+\" (got '%s')\n", c);
+				res = 0;
+			} else {
+/* now copy over the number, skipping all non-digits and stop at ; or NULL */
+                               t = tmp;
+				while( *c && (*c != ';') && (t - tmp < (sizeof(tmp) - 1))) {
+					if (isdigit(*c))
+						*t++ = *c;
+					c++;
+				}
+				*t = 0;
+				pbx_builtin_setvar_helper(chan, "ENUM", tmp);
+				ast_log(LOG_NOTICE, "tel: ENUM set to \"%s\"\n", tmp);
+				if (priority_jump || option_priority_jumping) {
+					if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 51))
+						res = 0;
+				}
+			}
+		} else if (!ast_strlen_zero(tech)) {
+			ast_log(LOG_NOTICE, "Don't know how to handle technology '%s'\n", tech);
+			pbx_builtin_setvar_helper(chan, "ENUMSTATUS", "BADURI");
+			res = 0;
+		}
+	}
+
+	LOCAL_USER_REMOVE(u);
+
+	return 0;
+}
+
+/*--- load_config: Load enum.conf and find out how to handle H.323 */
+static int load_config(void)
+{
+	struct ast_config *cfg;
+	char *s;
+
+	cfg = ast_config_load(ENUM_CONFIG);
+	if (cfg) {
+		if (!(s=ast_variable_retrieve(cfg, "general", "h323driver"))) {
+			strncpy(h323driver, H323DRIVERDEFAULT, sizeof(h323driver) - 1);
+		} else {
+			strncpy(h323driver, s, sizeof(h323driver) - 1);
+		}
+		ast_config_destroy(cfg);
+		return 0;
+	}
+	ast_log(LOG_NOTICE, "No ENUM Config file, using defaults\n");
+	return 0;
+}
+
+
+/*--- unload_module: Unload this application from PBX */
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+/*--- load_module: Load this application into PBX */
+int load_module(void)
+{
+	int res;
+	
+	res = ast_register_application(app, enumlookup_exec, synopsis, descrip);
+	
+	if (!res)
+		res = load_config();
+	
+	return res;
+}
+
+/*--- reload: Reload configuration file */
+int reload(void)
+{
+	return load_config();
+}
+
+
+/*--- description: Describe module */
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
+

apps/app_eval.c

@@ -0,0 +1,127 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (c) 2004 - 2005, Tilghman Lesher.  All rights reserved.
+ *
+ * Tilghman Lesher <app_eval__v001@the-tilghman.com>
+ *
+ * This code is released by the author with no restrictions on usage.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ * \brief Eval application
+ *
+ * \author Tilghman Lesher <app_eval__v001@the-tilghman.com>
+ *
+ * \ingroup applications
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+
+/* Maximum length of any variable */
+#define MAXRESULT	1024
+
+static char *tdesc = "Reevaluates strings";
+
+static char *app_eval = "Eval";
+
+static char *eval_synopsis = "Evaluates a string";
+
+static char *eval_descrip =
+"Usage: Eval(newvar=somestring)\n"
+"  Normally Asterisk evaluates variables inline.  But what if you want to\n"
+"store variable offsets in a database, to be evaluated later?  Eval is\n"
+"the answer, by allowing a string to be evaluated twice in the dialplan,\n"
+"the first time as part of the normal dialplan, and the second using Eval.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int eval_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *s, *newvar=NULL, tmp[MAXRESULT];
+	static int dep_warning = 0;
+
+	LOCAL_USER_ADD(u);
+	
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "This application has been deprecated in favor of the dialplan function, EVAL\n");
+		dep_warning = 1;
+	}
+
+	/* Check and parse arguments */
+	if (data) {
+		s = ast_strdupa((char *)data);
+		if (s) {
+			newvar = strsep(&s, "=");
+			if (newvar && (newvar[0] != '\0')) {
+				memset(tmp, 0, MAXRESULT);
+				pbx_substitute_variables_helper(chan, s, tmp, MAXRESULT - 1);
+				pbx_builtin_setvar_helper(chan, newvar, tmp);
+			}
+		} else {
+			ast_log(LOG_ERROR, "Out of memory\n");
+			res = -1;
+		}
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app_eval);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app_eval, eval_exec, eval_synopsis, eval_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_exec.c

@@ -2,9 +2,8 @@
  * Asterisk -- An open source telephony toolkit.
  *
  * Copyright (c) 2004 - 2005, Tilghman Lesher.  All rights reserved.
- * Portions copyright (c) 2006, Philipp Dunkel.
  *
- * Tilghman Lesher <app_exec__v002@the-tilghman.com>
+ * Tilghman Lesher <app_exec__v001@the-tilghman.com>
  *
  * This code is released by the author with no restrictions on usage.
  *
@@ -20,227 +19,117 @@
  *
  * \brief Exec application
  *
- * \author Tilghman Lesher <app_exec__v002@the-tilghman.com>
- * \author Philipp Dunkel <philipp.dunkel@ebox.at>
+ * \author Tilghman Lesher <app_exec__v001@the-tilghman.com>
  *
  * \ingroup applications
  */
 
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
-#include "asterisk/app.h"
 
 /* Maximum length of any variable */
-#define MAXRESULT 1024
+#define MAXRESULT	1024
 
-/*! Note
- *
- * The key difference between these two apps is exit status.  In a
- * nutshell, Exec tries to be transparent as possible, behaving
- * in exactly the same way as if the application it calls was
- * directly invoked from the dialplan.
- *
- * TryExec, on the other hand, provides a way to execute applications
- * and catch any possible fatal error without actually fatally
- * affecting the dialplan.
- */
+static char *tdesc = "Executes applications";
 
 static char *app_exec = "Exec";
-static char *exec_synopsis = "Executes dialplan application";
+
+static char *exec_synopsis = "Executes internal application";
+
 static char *exec_descrip =
-"  Exec(appname(arguments)):\n"
-"Allows an arbitrary application to be invoked even when not\n"
-"hardcoded into the dialplan.  If the underlying application\n"
-"terminates the dialplan, or if the application cannot be found,\n"
-"Exec will terminate the dialplan.\n"
-"  To invoke external applications, see the application System.\n"
-"  If you would like to catch any error instead, see TryExec.\n";
-
-static char *app_tryexec = "TryExec";
-static char *tryexec_synopsis = "Executes dialplan application, always returning";
-static char *tryexec_descrip =
-"  TryExec(appname(arguments)):\n"
-"Allows an arbitrary application to be invoked even when not\n"
+"Usage: Exec(appname(arguments))\n"
+"  Allows an arbitrary application to be invoked even when not\n"
 "hardcoded into the dialplan. To invoke external applications\n"
-"see the application System.  Always returns to the dialplan.\n"
-"The channel variable TRYSTATUS will be set to one of:\n"
-"    SUCCESS   if the application returned zero\n"
-"    FAILED    if the application returned non-zero\n"
-"    NOAPP     if the application was not found or was not specified\n";
+"see the application System. Returns whatever value the\n"
+"app returns or a non-zero value if the app cannot be found.\n";
 
-static char *app_execif = "ExecIf";
-static char *execif_synopsis = "Executes dialplan application, conditionally";
-static char *execif_descrip =
-"  ExecIF (<expr>?<appiftrue>(<args>)[:<appiffalse>(<args>)])\n"
-"If <expr> is true, execute and return the result of <appiftrue>(<args>).\n"
-"If <expr> is true, but <appiftrue> is not found, then the application\n"
-"will return a non-zero value.\n";
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int exec_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
+	int res=0;
+	struct localuser *u;
 	char *s, *appname, *endargs, args[MAXRESULT];
 	struct ast_app *app;
 
-	if (ast_strlen_zero(data))
-		return 0;
+	LOCAL_USER_ADD(u);
 
-	s = ast_strdupa(data);
-	args[0] = 0;
-	appname = strsep(&s, "(");
-	if (s) {
-		endargs = strrchr(s, ')');
-		if (endargs)
-			*endargs = '\0';
-		pbx_substitute_variables_helper(chan, s, args, MAXRESULT - 1);
-	}
-	if (appname) {
-		app = pbx_findapp(appname);
-		if (app) {
-			res = pbx_exec(chan, app, args);
+	memset(args, 0, MAXRESULT);
+
+	/* Check and parse arguments */
+	if (data) {
+		s = ast_strdupa((char *)data);
+		if (s) {
+			appname = strsep(&s, "(");
+			if (s) {
+				endargs = strrchr(s, ')');
+				if (endargs)
+					*endargs = '\0';
+				pbx_substitute_variables_helper(chan, s, args, MAXRESULT - 1);
+			}
+			if (appname) {
+				app = pbx_findapp(appname);
+				if (app) {
+					res = pbx_exec(chan, app, args, 1);
+				} else {
+					ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
+					res = -1;
+				}
+			}
 		} else {
-			ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
+			ast_log(LOG_ERROR, "Out of memory\n");
 			res = -1;
 		}
 	}
 
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int tryexec_exec(struct ast_channel *chan, void *data)
-{
-	int res = 0;
-	char *s, *appname, *endargs, args[MAXRESULT];
-	struct ast_app *app;
-
-	if (ast_strlen_zero(data))
-		return 0;
-
-	s = ast_strdupa(data);
-	args[0] = 0;
-	appname = strsep(&s, "(");
-	if (s) {
-		endargs = strrchr(s, ')');
-		if (endargs)
-			*endargs = '\0';
-		pbx_substitute_variables_helper(chan, s, args, MAXRESULT - 1);
-	}
-	if (appname) {
-		app = pbx_findapp(appname);
-		if (app) {
-			res = pbx_exec(chan, app, args);
-			pbx_builtin_setvar_helper(chan, "TRYSTATUS", res ? "FAILED" : "SUCCESS");
-		} else {
-			ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
-			pbx_builtin_setvar_helper(chan, "TRYSTATUS", "NOAPP");
-		}
-	}
-
-	return 0;
-}
-
-static int execif_exec(struct ast_channel *chan, void *data)
-{
-	int res = 0;
-	char *truedata = NULL, *falsedata = NULL, *end, *firstcomma, *firstquestion;
-	struct ast_app *app = NULL;
-	AST_DECLARE_APP_ARGS(expr,
-		AST_APP_ARG(expr);
-		AST_APP_ARG(remainder);
-	);
-	AST_DECLARE_APP_ARGS(apps,
-		AST_APP_ARG(t);
-		AST_APP_ARG(f);
-	);
-	char *parse = ast_strdupa(data);
-
-	firstcomma = strchr(parse, ',');
-	firstquestion = strchr(parse, '?');
-
-	if ((firstcomma != NULL && firstquestion != NULL && firstcomma < firstquestion) || (firstquestion == NULL)) {
-		/* Deprecated syntax */
-		AST_DECLARE_APP_ARGS(depr,
-			AST_APP_ARG(expr);
-			AST_APP_ARG(appname);
-			AST_APP_ARG(appargs);
-		);
-		AST_STANDARD_APP_ARGS(depr, parse);
-
-		ast_log(LOG_WARNING, "Deprecated syntax found.  Please upgrade to using ExecIf(<expr>?%s(%s))\n", depr.appname, depr.appargs);
-
-		/* Make the two syntaxes look the same */
-		expr.expr = depr.expr;
-		apps.t = depr.appname;
-		apps.f = NULL;
-		truedata = depr.appargs;
-	} else {
-		/* Preferred syntax */
-
-		AST_NONSTANDARD_RAW_ARGS(expr, parse, '?');
-		if (ast_strlen_zero(expr.remainder)) {
-			ast_log(LOG_ERROR, "Usage: ExecIf(<expr>?<appiftrue>(<args>)[:<appiffalse>(<args)])\n");
-			return -1;
-		}
-
-		AST_NONSTANDARD_RAW_ARGS(apps, expr.remainder, ':');
-
-		if (apps.t && (truedata = strchr(apps.t, '('))) {
-			*truedata++ = '\0';
-			if ((end = strrchr(truedata, ')'))) {
-				*end = '\0';
-			}
-		}
-
-		if (apps.f && (falsedata = strchr(apps.f, '('))) {
-			*falsedata++ = '\0';
-			if ((end = strrchr(falsedata, ')'))) {
-				*end = '\0';
-			}
-		}
-	}
-
-	if (pbx_checkcondition(expr.expr)) {
-		if (!ast_strlen_zero(apps.t) && (app = pbx_findapp(apps.t))) {
-			res = pbx_exec(chan, app, S_OR(truedata, ""));
-		} else {
-			ast_log(LOG_WARNING, "Could not find application! (%s)\n", apps.t);
-			res = -1;
-		}
-	} else if (!ast_strlen_zero(apps.f)) {
-		if ((app = pbx_findapp(apps.f))) {
-			res = pbx_exec(chan, app, S_OR(falsedata, ""));
-		} else {
-			ast_log(LOG_WARNING, "Could not find application! (%s)\n", apps.f);
-			res = -1;
-		}
-	}
-
-	return res;
-}
-
-static int unload_module(void)
+int unload_module(void)
 {
 	int res;
 
 	res = ast_unregister_application(app_exec);
-	res |= ast_unregister_application(app_tryexec);
-	res |= ast_unregister_application(app_execif);
+
+	STANDARD_HANGUP_LOCALUSERS;
 
 	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
-	int res = ast_register_application(app_exec, exec_exec, exec_synopsis, exec_descrip);
-	res |= ast_register_application(app_tryexec, tryexec_exec, tryexec_synopsis, tryexec_descrip);
-	res |= ast_register_application(app_execif, execif_exec, execif_synopsis, execif_descrip);
+	return ast_register_application(app_exec, exec_exec, exec_synopsis, exec_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
 	return res;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Executes dialplan applications");
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_externalivr.c

@@ -22,50 +22,46 @@
 /*! \file
  *
  * \brief External IVR application interface
- *
- * \author Kevin P. Fleming <kpfleming@digium.com>
- *
- * \note Portions taken from the file-based music-on-hold work
- * created by Anthony Minessale II in res_musiconhold.c
- *
+ * 
  * \ingroup applications
  */
 
-/*** MODULEINFO
-	<depend>working_fork</depend>
- ***/
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <errno.h>
+#include <signal.h>
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include <signal.h>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
-
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/linkedlists.h"
 #include "asterisk/app.h"
-#include "asterisk/utils.h"
+#include "asterisk/options.h"
+
+static const char *tdesc = "External IVR Interface Application";
 
 static const char *app = "ExternalIVR";
 
 static const char *synopsis = "Interfaces with an external IVR application";
 
-static const char *descrip =
-"  ExternalIVR(command[,arg[,arg...]]): Forks a process to run the supplied command,\n"
+static const char *descrip = 
+"  ExternalIVR(command[|arg[|arg...]]): Forks an process to run the supplied command,\n"
 "and starts a generator on the channel. The generator's play list is\n"
 "controlled by the external application, which can add and clear entries\n"
 "via simple commands issued over its stdout. The external application\n"
 "will receive all DTMF events received on the channel, and notification\n"
 "if the channel is hung up. The application will not be forcibly terminated\n"
 "when the channel is hung up.\n"
-"See doc/externalivr.txt for a protocol specification.\n";
+"See doc/README.externalivr for a protocol specification.\n";
 
 /* XXX the parser in gcc 2.95 gets confused if you don't put a space between 'name' and the comma */
 #define ast_chan_log(level, channel, format, ...) ast_log(level, "%s: " format, channel->name , ## __VA_ARGS__)
@@ -75,8 +71,9 @@ struct playlist_entry {
 	char filename[1];
 };
 
-struct ivr_localuser {
+struct localuser {
 	struct ast_channel *chan;
+	struct localuser *next;
 	AST_LIST_HEAD(playlist, playlist_entry) playlist;
 	AST_LIST_HEAD(finishlist, playlist_entry) finishlist;
 	int abort_current_sound;
@@ -84,20 +81,17 @@ struct ivr_localuser {
 	int option_autoclear;
 };
 
+LOCAL_USER_DECL;
 
 struct gen_state {
-	struct ivr_localuser *u;
+	struct localuser *u;
 	struct ast_filestream *stream;
 	struct playlist_entry *current;
 	int sample_queue;
 };
 
-static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u, 
-              int *eivr_events_fd, int *eivr_commands_fd, int *eivr_errors_fd, 
-              const char *args);
-
-static void send_eivr_event(FILE *handle, const char event, const char *data,
-	const struct ast_channel *chan)
+static void send_child_event(FILE *handle, const char event, const char *data,
+			     const struct ast_channel *chan)
 {
 	char tmp[256];
 
@@ -108,16 +102,17 @@ static void send_eivr_event(FILE *handle, const char event, const char *data,
 	}
 
 	fprintf(handle, "%s\n", tmp);
-	if (option_debug)
-		ast_chan_log(LOG_DEBUG, chan, "sent '%s'\n", tmp);
+	ast_chan_log(LOG_DEBUG, chan, "sent '%s'\n", tmp);
 }
 
 static void *gen_alloc(struct ast_channel *chan, void *params)
 {
-	struct ivr_localuser *u = params;
+	struct localuser *u = params;
 	struct gen_state *state;
 
-	if (!(state = ast_calloc(1, sizeof(*state))))
+	state = calloc(1, sizeof(*state));
+
+	if (!state)
 		return NULL;
 
 	state->u = u;
@@ -140,15 +135,15 @@ static void gen_release(struct ast_channel *chan, void *data)
 	struct gen_state *state = data;
 
 	gen_closestream(state);
-	ast_free(data);
+	free(data);
 }
 
 /* caller has the playlist locked */
 static int gen_nextfile(struct gen_state *state)
 {
-	struct ivr_localuser *u = state->u;
+	struct localuser *u = state->u;
 	char *file_to_stream;
-
+	
 	u->abort_current_sound = 0;
 	u->playing_silence = 0;
 	gen_closestream(state);
@@ -166,7 +161,7 @@ static int gen_nextfile(struct gen_state *state)
 			ast_chan_log(LOG_WARNING, u->chan, "File '%s' could not be opened: %s\n", file_to_stream, strerror(errno));
 			if (!u->playing_silence) {
 				continue;
-			} else {
+			} else { 
 				break;
 			}
 		}
@@ -178,10 +173,10 @@ static int gen_nextfile(struct gen_state *state)
 static struct ast_frame *gen_readframe(struct gen_state *state)
 {
 	struct ast_frame *f = NULL;
-	struct ivr_localuser *u = state->u;
-
+	struct localuser *u = state->u;
+	
 	if (u->abort_current_sound ||
-		(u->playing_silence && AST_LIST_FIRST(&u->playlist))) {
+	    (u->playing_silence && AST_LIST_FIRST(&u->playlist))) {
 		gen_closestream(state);
 		AST_LIST_LOCK(&u->playlist);
 		gen_nextfile(state);
@@ -233,60 +228,13 @@ static struct ast_generator gen =
 	generate: gen_generate,
 };
 
-static void ast_eivr_getvariable(struct ast_channel *chan, char *data, char *outbuf, int outbuflen)
-{
-	/* original input data: "G,var1,var2," */
-	/* data passed as "data":  "var1,var2" */
-
-	char *inbuf, *variable;
-	const char *value;
-	int j;
-	struct ast_str *newstring = ast_str_alloca(outbuflen); 
-
-	outbuf[0] = '\0';
-
-	for (j = 1, inbuf = data; ; j++) {
-		variable = strsep(&inbuf, ",");
-		if (variable == NULL) {
-			int outstrlen = strlen(outbuf);
-			if(outstrlen && outbuf[outstrlen - 1] == ',') {
-				outbuf[outstrlen - 1] = 0;
-			}
-			break;
-		}
-		
-		value = pbx_builtin_getvar_helper(chan, variable);
-		if(!value)
-			value = "";
-		ast_str_append(&newstring, 0, "%s=%s,", variable, value);
-		ast_copy_string(outbuf, newstring->str, outbuflen);
-	}
-};
-
-static void ast_eivr_setvariable(struct ast_channel *chan, char *data)
-{
-	char *value;
-
-	char *inbuf = ast_strdupa(data), *variable;
-
-	for (variable = strsep(&inbuf, ","); variable; variable = strsep(&inbuf, ",")) {
-		ast_debug(1, "Setting up a variable: %s\n", variable);
-		/* variable contains "varname=value" */
-		value = strchr(variable, '=');
-		if (!value) {
-			value = "";
-		} else {
-			*value++ = '\0';
-		}
-		pbx_builtin_setvar_helper(chan, variable, value);
-	}
-};
-
 static struct playlist_entry *make_entry(const char *filename)
 {
 	struct playlist_entry *entry;
 
-	if (!(entry = ast_calloc(1, sizeof(*entry) + strlen(filename) + 10))) /* XXX why 10 ? */
+	entry = calloc(1, sizeof(*entry) + strlen(filename) + 10);
+
+	if (!entry)
 		return NULL;
 
 	strcpy(entry->filename, filename);
@@ -296,56 +244,65 @@ static struct playlist_entry *make_entry(const char *filename)
 
 static int app_exec(struct ast_channel *chan, void *data)
 {
+	struct localuser *u = NULL;
 	struct playlist_entry *entry;
-	int child_stdin[2] = { -1, -1 };
-	int child_stdout[2] = { -1, -1 };
-	int child_stderr[2] = { -1, -1 };
+	const char *args = data;
+	int child_stdin[2] = { 0,0 };
+	int child_stdout[2] = { 0,0 };
+	int child_stderr[2] = { 0,0 };
 	int res = -1;
 	int gen_active = 0;
 	int pid;
-	char *buf, *comma_delim_argbuf;
-	struct ivr_localuser foo = {
-		.playlist = AST_LIST_HEAD_INIT_VALUE,
-		.finishlist = AST_LIST_HEAD_INIT_VALUE,
-	};
-	struct ivr_localuser *u = &foo;
+	char *argv[32];
+	int argc = 1;
+	char *buf, *command;
+	FILE *child_commands = NULL;
+	FILE *child_errors = NULL;
+	FILE *child_events = NULL;
 	sigset_t fullset, oldset;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(cmd)[32];
-	);
+
+	LOCAL_USER_ADD(u);
 
 	sigfillset(&fullset);
 	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
 
+	AST_LIST_HEAD_INIT(&u->playlist);
+	AST_LIST_HEAD_INIT(&u->finishlist);
 	u->abort_current_sound = 0;
-	u->chan = chan;
-
-	if (ast_strlen_zero(data)) {
+	
+	if (ast_strlen_zero(args)) {
 		ast_log(LOG_WARNING, "ExternalIVR requires a command to execute\n");
-		return -1;
+		goto exit;
 	}
 
 	buf = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, buf);
+	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]));
 
-	/* copy args and replace commas with pipes */
-	comma_delim_argbuf = ast_strdupa(data);
-	
 	if (pipe(child_stdin)) {
 		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child input: %s\n", strerror(errno));
 		goto exit;
 	}
+
 	if (pipe(child_stdout)) {
 		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child output: %s\n", strerror(errno));
 		goto exit;
 	}
+
 	if (pipe(child_stderr)) {
 		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child errors: %s\n", strerror(errno));
 		goto exit;
 	}
+
 	if (chan->_state != AST_STATE_UP) {
 		ast_answer(chan);
 	}
+
 	if (ast_activate_generator(chan, &gen, u) < 0) {
 		ast_chan_log(LOG_WARNING, chan, "Failed to activate generator\n");
 		goto exit;
@@ -361,20 +318,11 @@ static int app_exec(struct ast_channel *chan, void *data)
 	if (!pid) {
 		/* child process */
 		int i;
-#ifdef HAVE_CAP
-		cap_t cap = cap_from_text("cap_net_admin-eip");
-
-		if (cap_set_proc(cap)) {
-			/* Careful with order! Logging cannot happen after we close FDs */
-			ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-		}
-		cap_free(cap);
-#endif
 
 		signal(SIGPIPE, SIG_DFL);
 		pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
 
-		if (ast_opt_high_priority)
+		if (option_highpriority)
 			ast_set_priority(0);
 
 		dup2(child_stdin[0], STDIN_FILENO);
@@ -382,290 +330,268 @@ static int app_exec(struct ast_channel *chan, void *data)
 		dup2(child_stderr[1], STDERR_FILENO);
 		for (i = STDERR_FILENO + 1; i < 1024; i++)
 			close(i);
-		execv(args.cmd[0], args.cmd);
-		fprintf(stderr, "Failed to execute '%s': %s\n", args.cmd[0], strerror(errno));
+		execv(argv[0], argv);
+		fprintf(stderr, "Failed to execute '%s': %s\n", argv[0], strerror(errno));
 		_exit(1);
 	} else {
 		/* parent process */
+		int child_events_fd = child_stdin[1];
+		int child_commands_fd = child_stdout[0];
+		int child_errors_fd = child_stderr[0];
+		struct ast_frame *f;
+		int ms;
+		int exception;
+		int ready_fd;
+		int waitfds[2] = { child_errors_fd, child_commands_fd };
+		struct ast_channel *rchan;
+
+		pthread_sigmask(SIG_SETMASK, &oldset, NULL);
 
 		close(child_stdin[0]);
-		child_stdin[0] = -1;
+		child_stdin[0] = 0;
 		close(child_stdout[1]);
-		child_stdout[1] = -1;
+		child_stdout[1] = 0;
 		close(child_stderr[1]);
-		child_stderr[1] = -1;
-		res = eivr_comm(chan, u, &child_stdin[1], &child_stdout[0], &child_stderr[0], comma_delim_argbuf);
+		child_stderr[1] = 0;
+
+		if (!(child_events = fdopen(child_events_fd, "w"))) {
+			ast_chan_log(LOG_WARNING, chan, "Could not open stream for child events\n");
+			goto exit;
+		}
+
+		if (!(child_commands = fdopen(child_commands_fd, "r"))) {
+			ast_chan_log(LOG_WARNING, chan, "Could not open stream for child commands\n");
+			goto exit;
+		}
+
+		if (!(child_errors = fdopen(child_errors_fd, "r"))) {
+			ast_chan_log(LOG_WARNING, chan, "Could not open stream for child errors\n");
+			goto exit;
+		}
+
+		setvbuf(child_events, NULL, _IONBF, 0);
+		setvbuf(child_commands, NULL, _IONBF, 0);
+		setvbuf(child_errors, NULL, _IONBF, 0);
+
+		res = 0;
+
+		while (1) {
+			if (ast_test_flag(chan, AST_FLAG_ZOMBIE)) {
+				ast_chan_log(LOG_NOTICE, chan, "Is a zombie\n");
+				res = -1;
+				break;
+			}
+
+			if (ast_check_hangup(chan)) {
+				ast_chan_log(LOG_NOTICE, chan, "Got check_hangup\n");
+				send_child_event(child_events, 'H', NULL, chan);
+				res = -1;
+				break;
+			}
+
+			ready_fd = 0;
+			ms = 100;
+			errno = 0;
+			exception = 0;
+
+			rchan = ast_waitfor_nandfds(&chan, 1, waitfds, 2, &exception, &ready_fd, &ms);
+
+			if (!AST_LIST_EMPTY(&u->finishlist)) {
+				AST_LIST_LOCK(&u->finishlist);
+				while ((entry = AST_LIST_REMOVE_HEAD(&u->finishlist, list))) {
+					send_child_event(child_events, 'F', entry->filename, chan);
+					free(entry);
+				}
+				AST_LIST_UNLOCK(&u->finishlist);
+			}
+
+			if (rchan) {
+				/* the channel has something */
+				f = ast_read(chan);
+				if (!f) {
+					ast_chan_log(LOG_NOTICE, chan, "Returned no frame\n");
+					send_child_event(child_events, 'H', NULL, chan);
+					res = -1;
+					break;
+				}
+
+				if (f->frametype == AST_FRAME_DTMF) {
+					send_child_event(child_events, f->subclass, NULL, chan);
+					if (u->option_autoclear) {
+						if (!u->abort_current_sound && !u->playing_silence)
+							send_child_event(child_events, 'T', NULL, chan);
+						AST_LIST_LOCK(&u->playlist);
+						while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+							send_child_event(child_events, 'D', entry->filename, chan);
+							free(entry);
+						}
+						if (!u->playing_silence)
+							u->abort_current_sound = 1;
+						AST_LIST_UNLOCK(&u->playlist);
+					}
+				} else if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP)) {
+					ast_chan_log(LOG_NOTICE, chan, "Got AST_CONTROL_HANGUP\n");
+					send_child_event(child_events, 'H', NULL, chan);
+					ast_frfree(f);
+					res = -1;
+					break;
+				}
+				ast_frfree(f);
+			} else if (ready_fd == child_commands_fd) {
+				char input[1024];
+
+				if (exception || feof(child_commands)) {
+					ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
+					res = -1;
+					break;
+				}
+
+				if (!fgets(input, sizeof(input), child_commands))
+					continue;
+
+				command = ast_strip(input);
+
+				ast_chan_log(LOG_DEBUG, chan, "got command '%s'\n", input);
+
+				if (strlen(input) < 4)
+					continue;
+
+				if (input[0] == 'S') {
+					if (ast_fileexists(&input[2], NULL, u->chan->language) == -1) {
+						ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
+						send_child_event(child_events, 'Z', NULL, chan);
+						strcpy(&input[2], "exception");
+					}
+					if (!u->abort_current_sound && !u->playing_silence)
+						send_child_event(child_events, 'T', NULL, chan);
+					AST_LIST_LOCK(&u->playlist);
+					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+						send_child_event(child_events, 'D', entry->filename, chan);
+						free(entry);
+					}
+					if (!u->playing_silence)
+						u->abort_current_sound = 1;
+					entry = make_entry(&input[2]);
+					if (entry)
+						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
+					AST_LIST_UNLOCK(&u->playlist);
+				} else if (input[0] == 'A') {
+					if (ast_fileexists(&input[2], NULL, u->chan->language) == -1) {
+						ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
+						send_child_event(child_events, 'Z', NULL, chan);
+						strcpy(&input[2], "exception");
+					}
+					entry = make_entry(&input[2]);
+					if (entry) {
+						AST_LIST_LOCK(&u->playlist);
+						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
+						AST_LIST_UNLOCK(&u->playlist);
+					}
+				} else if (input[0] == 'H') {
+					ast_chan_log(LOG_NOTICE, chan, "Hanging up: %s\n", &input[2]);
+					send_child_event(child_events, 'H', NULL, chan);
+					break;
+				} else if (input[0] == 'O') {
+					if (!strcasecmp(&input[2], "autoclear"))
+						u->option_autoclear = 1;
+					else if (!strcasecmp(&input[2], "noautoclear"))
+						u->option_autoclear = 0;
+					else
+						ast_chan_log(LOG_WARNING, chan, "Unknown option requested '%s'\n", &input[2]);
+				}
+			} else if (ready_fd == child_errors_fd) {
+				char input[1024];
+
+				if (exception || feof(child_errors)) {
+					ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
+					res = -1;
+					break;
+				}
+
+				if (fgets(input, sizeof(input), child_errors)) {
+					command = ast_strip(input);
+					ast_chan_log(LOG_NOTICE, chan, "stderr: %s\n", command);
+				}
+			} else if ((ready_fd < 0) && ms) { 
+				if (errno == 0 || errno == EINTR)
+					continue;
+
+				ast_chan_log(LOG_WARNING, chan, "Wait failed (%s)\n", strerror(errno));
+				break;
+			}
+		}
 	}
 
-	exit:
-	if (gen_active) {
+ exit:
+	if (gen_active)
 		ast_deactivate_generator(chan);
-	}
-	if (child_stdin[0] > -1) {
+
+	if (child_events)
+		fclose(child_events);
+
+	if (child_commands)
+		fclose(child_commands);
+
+	if (child_errors)
+		fclose(child_errors);
+
+	if (child_stdin[0])
 		close(child_stdin[0]);
-	}
-	if (child_stdin[1] > -1) {
+
+	if (child_stdin[1])
 		close(child_stdin[1]);
-	}
-	if (child_stdout[0] > -1) {
+
+	if (child_stdout[0])
 		close(child_stdout[0]);
-	}
-	if (child_stdout[1] > -1) {
+
+	if (child_stdout[1])
 		close(child_stdout[1]);
-	}
-	if (child_stderr[0] > -1) {
+
+	if (child_stderr[0])
 		close(child_stderr[0]);
-	}
-	if (child_stderr[1] > -1) {
+
+	if (child_stderr[1])
 		close(child_stderr[1]);
-	}
-	while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
-		ast_free(entry);
-	}
+
+	while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list)))
+		free(entry);
+
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }
 
-static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u, 
- 				int *eivr_events_fd, int *eivr_commands_fd, int *eivr_errors_fd, 
- 				const char *args)
+int unload_module(void)
 {
-	struct playlist_entry *entry;
-	struct ast_frame *f;
-	int ms;
- 	int exception;
- 	int ready_fd;
- 	int waitfds[2] = { *eivr_commands_fd, (eivr_errors_fd) ? *eivr_errors_fd : -1 };
- 	struct ast_channel *rchan;
- 	char *command;
- 	int res = -1;
-	int test_available_fd = -1;
-  
- 	FILE *eivr_commands = NULL;
- 	FILE *eivr_errors = NULL;
- 	FILE *eivr_events = NULL;
- 
- 	if (!(eivr_events = fdopen(*eivr_events_fd, "w"))) {
- 		ast_chan_log(LOG_WARNING, chan, "Could not open stream to send events\n");
- 		goto exit;
- 	}
- 	if (!(eivr_commands = fdopen(*eivr_commands_fd, "r"))) {
- 		ast_chan_log(LOG_WARNING, chan, "Could not open stream to receive commands\n");
- 		goto exit;
- 	}
- 	if (eivr_errors_fd) {  /*if opening a socket connection, error stream will not be used*/
- 		if (!(eivr_errors = fdopen(*eivr_errors_fd, "r"))) {
- 			ast_chan_log(LOG_WARNING, chan, "Could not open stream to receive errors\n");
- 			goto exit;
- 		}
- 	}
+	int res;
 
-	test_available_fd = open("/dev/null", O_RDONLY);
- 
- 	setvbuf(eivr_events, NULL, _IONBF, 0);
- 	setvbuf(eivr_commands, NULL, _IONBF, 0);
- 	if(eivr_errors)
-		setvbuf(eivr_errors, NULL, _IONBF, 0);
+	res = ast_unregister_application(app);
 
-	res = 0;
- 
- 	while (1) {
- 		if (ast_test_flag(chan, AST_FLAG_ZOMBIE)) {
- 			ast_chan_log(LOG_NOTICE, chan, "Is a zombie\n");
- 			res = -1;
- 			break;
- 		}
- 		if (ast_check_hangup(chan)) {
- 			ast_chan_log(LOG_NOTICE, chan, "Got check_hangup\n");
- 			send_eivr_event(eivr_events, 'H', NULL, chan);
- 			res = -1;
- 			break;
- 		}
- 
- 		ready_fd = 0;
- 		ms = 100;
- 		errno = 0;
- 		exception = 0;
- 
- 		rchan = ast_waitfor_nandfds(&chan, 1, waitfds, (eivr_errors_fd) ? 2 : 1, &exception, &ready_fd, &ms);
- 
- 		if (!AST_LIST_EMPTY(&u->finishlist)) {
- 			AST_LIST_LOCK(&u->finishlist);
- 			while ((entry = AST_LIST_REMOVE_HEAD(&u->finishlist, list))) {
- 				send_eivr_event(eivr_events, 'F', entry->filename, chan);
- 				ast_free(entry);
- 			}
- 			AST_LIST_UNLOCK(&u->finishlist);
- 		}
- 
- 		if (rchan) {
- 			/* the channel has something */
- 			f = ast_read(chan);
- 			if (!f) {
- 				ast_chan_log(LOG_NOTICE, chan, "Returned no frame\n");
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
- 				res = -1;
- 				break;
- 			}
- 			if (f->frametype == AST_FRAME_DTMF) {
- 				send_eivr_event(eivr_events, f->subclass, NULL, chan);
- 				if (u->option_autoclear) {
-  					if (!u->abort_current_sound && !u->playing_silence)
- 						send_eivr_event(eivr_events, 'T', NULL, chan);
-  					AST_LIST_LOCK(&u->playlist);
-  					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
- 						send_eivr_event(eivr_events, 'D', entry->filename, chan);
-  						ast_free(entry);
-  					}
-  					if (!u->playing_silence)
-  						u->abort_current_sound = 1;
-  					AST_LIST_UNLOCK(&u->playlist);
-  				}
- 			} else if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP)) {
- 				ast_chan_log(LOG_NOTICE, chan, "Got AST_CONTROL_HANGUP\n");
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
- 				ast_frfree(f);
- 				res = -1;
- 				break;
- 			}
- 			ast_frfree(f);
- 		} else if (ready_fd == *eivr_commands_fd) {
- 			char input[1024];
- 
- 			if (exception || (dup2(*eivr_commands_fd, test_available_fd) == -1) || feof(eivr_commands)) {
- 				ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
- 				res = -1;
-  				break;
-  			}
-  
- 			if (!fgets(input, sizeof(input), eivr_commands))
- 				continue;
- 
- 			command = ast_strip(input);
-  
- 			if (option_debug)
- 				ast_chan_log(LOG_DEBUG, chan, "got command '%s'\n", input);
-  
- 			if (strlen(input) < 4)
- 				continue;
-  
-			if (input[0] == 'P') {
- 				send_eivr_event(eivr_events, 'P', args, chan);
- 
- 			} else if (input[0] == 'S') {
- 				if (ast_fileexists(&input[2], NULL, u->chan->language) == -1) {
- 					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
- 					send_eivr_event(eivr_events, 'Z', NULL, chan);
- 					strcpy(&input[2], "exception");
- 				}
- 				if (!u->abort_current_sound && !u->playing_silence)
- 					send_eivr_event(eivr_events, 'T', NULL, chan);
- 				AST_LIST_LOCK(&u->playlist);
- 				while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
- 					send_eivr_event(eivr_events, 'D', entry->filename, chan);
- 					ast_free(entry);
- 				}
- 				if (!u->playing_silence)
- 					u->abort_current_sound = 1;
- 				entry = make_entry(&input[2]);
- 				if (entry)
- 					AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
- 				AST_LIST_UNLOCK(&u->playlist);
- 			} else if (input[0] == 'A') {
- 				if (ast_fileexists(&input[2], NULL, u->chan->language) == -1) {
- 					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
- 					send_eivr_event(eivr_events, 'Z', NULL, chan);
- 					strcpy(&input[2], "exception");
- 				}
- 				entry = make_entry(&input[2]);
- 				if (entry) {
- 					AST_LIST_LOCK(&u->playlist);
- 					AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
- 					AST_LIST_UNLOCK(&u->playlist);
- 				}
- 			} else if (input[0] == 'G') {
- 				/* A get variable message:  "G,variable1,variable2,..." */
- 				char response[2048];
+	STANDARD_HANGUP_LOCALUSERS;
 
- 				ast_chan_log(LOG_NOTICE, chan, "Getting a Variable out of the channel: %s\n", &input[2]);
- 				ast_eivr_getvariable(chan, &input[2], response, sizeof(response));
- 				send_eivr_event(eivr_events, 'G', response, chan);
- 			} else if (input[0] == 'V') {
- 				/* A set variable message:  "V,variablename=foo" */
- 				ast_chan_log(LOG_NOTICE, chan, "Setting a Variable up: %s\n", &input[2]);
- 				ast_eivr_setvariable(chan, &input[2]);
- 			} else if (input[0] == 'L') {
- 				ast_chan_log(LOG_NOTICE, chan, "Log message from EIVR: %s\n", &input[2]);
- 			} else if (input[0] == 'X') {
- 				ast_chan_log(LOG_NOTICE, chan, "Exiting ExternalIVR: %s\n", &input[2]);
- 				/*! \todo add deprecation debug message for X command here */
- 				res = 0;
- 				break;
-			} else if (input[0] == 'E') {
- 				ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
- 				send_eivr_event(eivr_events, 'E', NULL, chan);
- 				res = 0;
- 				break;
- 			} else if (input[0] == 'H') {
- 				ast_chan_log(LOG_NOTICE, chan, "Hanging up: %s\n", &input[2]);
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
- 				res = -1;
- 				break;
- 			} else if (input[0] == 'O') {
- 				if (!strcasecmp(&input[2], "autoclear"))
- 					u->option_autoclear = 1;
- 				else if (!strcasecmp(&input[2], "noautoclear"))
- 					u->option_autoclear = 0;
- 				else
- 					ast_chan_log(LOG_WARNING, chan, "Unknown option requested '%s'\n", &input[2]);
- 			}
- 		} else if (eivr_errors_fd && (ready_fd == *eivr_errors_fd)) {
- 			char input[1024];
-  
- 			if (exception || feof(eivr_errors)) {
- 				ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
- 				res = -1;
- 				break;
- 			}
- 			if (fgets(input, sizeof(input), eivr_errors)) {
- 				command = ast_strip(input);
- 				ast_chan_log(LOG_NOTICE, chan, "stderr: %s\n", command);
- 			}
- 		} else if ((ready_fd < 0) && ms) { 
- 			if (errno == 0 || errno == EINTR)
- 				continue;
- 
- 			ast_chan_log(LOG_WARNING, chan, "Wait failed (%s)\n", strerror(errno));
- 			break;
- 		}
- 	}
-
-	exit:
-	if (test_available_fd > -1) {
-		close(test_available_fd);
-	}
-	if (eivr_events) {
- 		fclose(eivr_events);
- 		*eivr_events_fd = -1;
-	}
- 	if (eivr_commands) {
-		fclose(eivr_commands);
-		*eivr_commands_fd = -1;
-	}
- 	if (eivr_errors) {
- 		fclose(eivr_errors);
-  		*eivr_errors_fd = -1;
-	}
 	return res;
- }
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, app_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "External IVR Interface Application");
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_fax.c

@@ -1,899 +0,0 @@
-/*
- * Asterisk -- A telephony toolkit for Linux.
- *
- * Simple fax applications
- * 
- * 2007-2008, Dmitry Andrianov <asterisk@dima.spb.ru>
- *
- * Code based on original implementation by Steve Underwood <steveu@coppice.org>
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License
- *
- */
-
-/*** MODULEINFO
-	 <depend>spandsp</depend>
-***/
- 
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <string.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <inttypes.h>
-#include <pthread.h>
-#include <errno.h>
-#include <tiffio.h>
-
-#include <spandsp.h>
-#ifdef HAVE_SPANDSP_EXPOSE_H
-#include <spandsp/expose.h>
-#endif
-#include <spandsp/version.h>
-
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/logger.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/app.h"
-#include "asterisk/dsp.h"
-#include "asterisk/module.h"
-#include "asterisk/manager.h"
-
-static char *app_sndfax_name = "SendFAX";
-static char *app_sndfax_synopsis = "Send a FAX";
-static char *app_sndfax_desc = 
-"  SendFAX(filename[|options]):\n"
-"Send a given TIFF file to the channel as a FAX.\n"
-"The option string may contain zero or more of the following characters:\n"
-"     'a' - makes the application behave as an answering machine\n"
-"           The default behaviour is to behave as a calling machine.\n"
-"\n"
-"This application uses following variables:\n"
-"     LOCALSTATIONID to identify itself to the remote end.\n"
-"     LOCALHEADERINFO to generate a header line on each page.\n"
-"\n"
-"This application sets the following channel variables upon completion:\n"
-"     FAXSTATUS       - status of operation:\n"
-"                           SUCCESS | FAILED\n"
-"     FAXERROR        - Error when FAILED\n"
-"     FAXMODE         - Mode used:\n"
-"                           audio | T38\n"
-"     REMOTESTATIONID - CSID of the remote side.\n"
-"     FAXPAGES        - number of pages sent.\n"
-"     FAXBITRATE      - transmition rate.\n"
-"     FAXRESOLUTION   - resolution.\n"
-"\n"
-"Returns -1 in case of user hang up or any channel error.\n"
-"Returns 0 on success.\n";
-
-static char *app_rcvfax_name = "ReceiveFAX";
-static char *app_rcvfax_synopsis = "Receive a FAX";
-static char *app_rcvfax_desc = 
-"  ReceiveFAX(filename[|options]):\n"
-"Receives a fax from the channel into the given filename overwriting\n"
-"the file if it already exists. File created will have TIFF format.\n"
-"The option string may contain zero or more of the following characters:\n"
-"     'c' -- makes the application behave as a calling machine\n"
-"            The default behaviour is to behave as an answering machine.\n"
-"\n"
-"This application uses following variables:\n"
-"     LOCALSTATIONID to identify itself to the remote end.\n"
-"     LOCALHEADERINFO to generate a header line on each page.\n"
-"\n"
-"This application sets the following channel variables upon completion:\n"
-"     FAXSTATUS       - status of operation:\n"
-"                           SUCCESS | FAILED\n"
-"     FAXERROR        - Error when FAILED\n"
-"     FAXMODE         - Mode used:\n"
-"                           audio | T38\n"
-"     REMOTESTATIONID - CSID of the remote side.\n"
-"     FAXPAGES        - number of pages sent.\n"
-"     FAXBITRATE      - transmition rate.\n"
-"     FAXRESOLUTION   - resolution.\n"
-"\n"
-"Returns -1 in case of user hang up or any channel error.\n"
-"Returns 0 on success.\n";
-
-#define MAX_SAMPLES 240
-
-/* Watchdog. I have seen situations when remote fax disconnects (because of poor line
-   quality) while SpanDSP continues staying in T30_STATE_IV_CTC state forever.
-   To avoid this, we terminate when we see that T30 state does not change for 5 minutes.
-   We also terminate application when more than 30 minutes passed regardless of
-   state changes. This is just a precaution measure - no fax should take that long */
-
-#define WATCHDOG_TOTAL_TIMEOUT	30 * 60
-#define WATCHDOG_STATE_TIMEOUT	5 * 60
-
-typedef struct {
-	struct ast_channel *chan;
-	enum ast_t38_state t38state;	/* T38 state of the channel */
-	int direction;			/* Fax direction: 0 - receiving, 1 - sending */
-	int caller_mode;
-	char *file_name;
-	struct ast_control_t38_parameters t38parameters;
-	volatile int finished;
-} fax_session;
-
-static void span_message(int level, const char *msg)
-{
-	if (level == SPAN_LOG_ERROR) {
-		ast_log(LOG_ERROR, "%s", msg);
-	} else if (level == SPAN_LOG_WARNING) {
-		ast_log(LOG_WARNING, "%s", msg);
-	} else {
-		ast_log(LOG_DEBUG, "%s", msg);
-	}
-}
-
-static int t38_tx_packet_handler(t38_core_state_t *s, void *user_data, const uint8_t *buf, int len, int count)
-{
-	struct ast_channel *chan = (struct ast_channel *) user_data;
-
-	struct ast_frame outf = {
-		.frametype = AST_FRAME_MODEM,
-		.subclass = AST_MODEM_T38,
-		.src = __FUNCTION__,
-	};
-
-	/* TODO: Asterisk does not provide means of resending the same packet multiple
-	  times so count is ignored at the moment */
-
-	AST_FRAME_SET_BUFFER(&outf, buf, 0, len);
-
-	if (ast_write(chan, &outf) < 0) {
-		ast_log(LOG_WARNING, "Unable to write frame to channel; %s\n", strerror(errno));
-		return -1;
-	}
-
-	return 0;
-}
-
-static void phase_e_handler(t30_state_t *f, void *user_data, int result)
-{
-	const char *local_ident;
-	const char *far_ident;
-	char buf[20];
-	fax_session *s = (fax_session *) user_data;
-	t30_stats_t stat;
-	int pages_transferred;
-
-	ast_debug(1, "Fax phase E handler. result=%d\n", result);
-
-	t30_get_transfer_statistics(f, &stat);
-
-	s = (fax_session *) user_data;
-
-	if (result != T30_ERR_OK) {
-		s->finished = -1;
-
-		/* FAXSTATUS is already set to FAILED */
-		pbx_builtin_setvar_helper(s->chan, "FAXERROR", t30_completion_code_to_str(result));
-
-		ast_log(LOG_WARNING, "Error transmitting fax. result=%d: %s.\n", result, t30_completion_code_to_str(result));
-
-		return;
-	}
-	
-	s->finished = 1; 
-	
-	local_ident = t30_get_tx_ident(f);
-	far_ident = t30_get_rx_ident(f);
-	pbx_builtin_setvar_helper(s->chan, "FAXSTATUS", "SUCCESS"); 
-	pbx_builtin_setvar_helper(s->chan, "FAXERROR", NULL); 
-	pbx_builtin_setvar_helper(s->chan, "REMOTESTATIONID", far_ident);
-#if SPANDSP_RELEASE_DATE >= 20090220
-	pages_transferred = (s->direction) ? stat.pages_tx : stat.pages_rx;
-#else
-	pages_transferred = stat.pages_transferred;
-#endif
-	snprintf(buf, sizeof(buf), "%d", pages_transferred);
-	pbx_builtin_setvar_helper(s->chan, "FAXPAGES", buf);
-	snprintf(buf, sizeof(buf), "%d", stat.y_resolution);
-	pbx_builtin_setvar_helper(s->chan, "FAXRESOLUTION", buf);
-	snprintf(buf, sizeof(buf), "%d", stat.bit_rate);
-	pbx_builtin_setvar_helper(s->chan, "FAXBITRATE", buf); 
-	
-	ast_debug(1, "Fax transmitted successfully.\n");
-	ast_debug(1, "  Remote station ID: %s\n", far_ident);
-	ast_debug(1, "  Pages transferred: %d\n", pages_transferred);
-	ast_debug(1, "  Image resolution:  %d x %d\n", stat.x_resolution, stat.y_resolution);
-	ast_debug(1, "  Transfer Rate:     %d\n", stat.bit_rate);
-	
-	manager_event(EVENT_FLAG_CALL,
-		      s->direction ? "FaxSent" : "FaxReceived", 
-		      "Channel: %s\r\n"
-		      "Exten: %s\r\n"
-		      "CallerID: %s\r\n"
-		      "RemoteStationID: %s\r\n"
-		      "LocalStationID: %s\r\n"
-		      "PagesTransferred: %d\r\n"
-		      "Resolution: %d\r\n"
-		      "TransferRate: %d\r\n"
-		      "FileName: %s\r\n",
-		      s->chan->name,
-		      s->chan->exten,
-		      S_OR(s->chan->cid.cid_num, ""),
-		      far_ident,
-		      local_ident,
-		      pages_transferred,
-		      stat.y_resolution,
-		      stat.bit_rate,
-		      s->file_name);
-}
-
-/* === Helper functions to configure fax === */
-
-/* Setup SPAN logging according to Asterisk debug level */
-static int set_logging(logging_state_t *state)
-{
-	int level = SPAN_LOG_WARNING + option_debug;
-
-	span_log_set_message_handler(state, span_message);
-	span_log_set_level(state, SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | level); 
-
-	return 0;
-}
-
-static void set_local_info(t30_state_t *state, fax_session *s)
-{
-	const char *x;
-
-	x = pbx_builtin_getvar_helper(s->chan, "LOCALSTATIONID");
-	if (!ast_strlen_zero(x))
-		t30_set_tx_ident(state, x);
-
-	x = pbx_builtin_getvar_helper(s->chan, "LOCALHEADERINFO");
-	if (!ast_strlen_zero(x))
-		t30_set_tx_page_header_info(state, x);
-}
-
-static void set_file(t30_state_t *state, fax_session *s)
-{
-	if (s->direction)
-		t30_set_tx_file(state, s->file_name, -1, -1);
-	else
-		t30_set_rx_file(state, s->file_name, -1);
-}
-
-static void set_ecm(t30_state_t *state, int ecm)
-{
-	t30_set_ecm_capability(state, ecm);
-	t30_set_supported_compressions(state, T30_SUPPORT_T4_1D_COMPRESSION | T30_SUPPORT_T4_2D_COMPRESSION | T30_SUPPORT_T6_COMPRESSION);
-}
-
-/* === Generator === */
-
-/* This function is only needed to return passed params so
-   generator_activate will save it to channel's generatordata */
-static void *fax_generator_alloc(struct ast_channel *chan, void *params)
-{
-	return params;
-}
-
-static int fax_generator_generate(struct ast_channel *chan, void *data, int len, int samples)
-{
-	fax_state_t *fax = (fax_state_t*) data;
-	uint8_t buffer[AST_FRIENDLY_OFFSET + MAX_SAMPLES * sizeof(uint16_t)];
-	int16_t *buf = (int16_t *) (buffer + AST_FRIENDLY_OFFSET);
-    
-	struct ast_frame outf = {
-		.frametype = AST_FRAME_VOICE,
-		.subclass = AST_FORMAT_SLINEAR,
-		.src = __FUNCTION__,
-	};
-
-	if (samples > MAX_SAMPLES) {
-		ast_log(LOG_WARNING, "Only generating %d samples, where %d requested\n", MAX_SAMPLES, samples);
-		samples = MAX_SAMPLES;
-	}
-	
-	if ((len = fax_tx(fax, buf, samples)) > 0) {
-		outf.samples = len;
-		AST_FRAME_SET_BUFFER(&outf, buffer, AST_FRIENDLY_OFFSET, len * sizeof(int16_t));
-
-		if (ast_write(chan, &outf) < 0) {
-			ast_log(LOG_WARNING, "Failed to write frame to '%s': %s\n", chan->name, strerror(errno));
-			return -1;
-		}
-	}
-
-	return 0;
-}
-
-struct ast_generator generator = {
-	alloc:		fax_generator_alloc,
-	generate: 	fax_generator_generate,
-};
-
-
-/* === Transmission === */
-
-static int transmit_audio(fax_session *s)
-{
-	int res = -1;
-	int original_read_fmt = AST_FORMAT_SLINEAR;
-	int original_write_fmt = AST_FORMAT_SLINEAR;
-	fax_state_t fax;
-	t30_state_t *t30state;
-	struct ast_frame *inf = NULL;
-	int last_state = 0;
-	struct timeval now, start, state_change;
-	enum ast_t38_state t38_state;
-	struct ast_control_t38_parameters t38_parameters = { .version = 0,
-							     .max_ifp = 800,
-							     .rate = AST_T38_RATE_14400,
-							     .rate_management = AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF,
-							     .fill_bit_removal = 1,
-/*
- * spandsp has API calls to support MMR and JBIG transcoding, but they aren't
- * implemented quite yet... so don't offer them to the remote endpoint
- *							     .transcoding_mmr = 1,
- *							     .transcoding_jbig = 1,
-*/
-	};
-
-	/* if in called party mode, try to use T.38 */
-	if (s->caller_mode == FALSE) {
-		/* check if we are already in T.38 mode (unlikely), or if we can request
-		 * a switch... if so, request it now and wait for the result, rather
-		 * than starting an audio FAX session that will have to be cancelled
-		 */
-		if ((t38_state = ast_channel_get_t38_state(s->chan)) == T38_STATE_NEGOTIATED) {
-			return 1;
-		} else if ((t38_state != T38_STATE_UNAVAILABLE) &&
-			   (t38_parameters.request_response = AST_T38_REQUEST_NEGOTIATE,
-			    (ast_indicate_data(s->chan, AST_CONTROL_T38_PARAMETERS, &t38_parameters, sizeof(t38_parameters)) == 0))) {
-			/* wait up to five seconds for negotiation to complete */
-			unsigned int timeout = 5000;
-			int ms;
-			
-			ast_debug(1, "Negotiating T.38 for receive on %s\n", s->chan->name);
-			while (timeout > 0) {
-				ms = ast_waitfor(s->chan, 1000);
-				if (ms < 0) {
-					ast_log(LOG_WARNING, "something bad happened while channel '%s' was polling.\n", s->chan->name);
-					return -1;
-				}
-				if (!ms) {
-					/* nothing happened */
-					if (timeout > 0) {
-						timeout -= 1000;
-						continue;
-					} else {
-						ast_log(LOG_WARNING, "channel '%s' timed-out during the T.38 negotiation.\n", s->chan->name);
-						break;
-					}
-				}
-				if (!(inf = ast_read(s->chan))) {
-					return -1;
-				}
-				if ((inf->frametype == AST_FRAME_CONTROL) &&
-				    (inf->subclass == AST_CONTROL_T38_PARAMETERS) &&
-				    (inf->datalen == sizeof(t38_parameters))) {
-					struct ast_control_t38_parameters *parameters = inf->data;
-					
-					switch (parameters->request_response) {
-					case AST_T38_NEGOTIATED:
-						ast_debug(1, "Negotiated T.38 for receive on %s\n", s->chan->name);
-						res = 1;
-						break;
-					case AST_T38_REFUSED:
-						ast_log(LOG_WARNING, "channel '%s' refused to negotiate T.38\n", s->chan->name);
-						break;
-					default:
-						ast_log(LOG_ERROR, "channel '%s' failed to negotiate T.38\n", s->chan->name);
-						break;
-					}
-					ast_frfree(inf);
-					if (res == 1) {
-						return 1;
-					} else {
-						break;
-					}
-				}
-				ast_frfree(inf);
-			}
-		}
-	}
-
-#if SPANDSP_RELEASE_DATE >= 20080725
-        /* for spandsp shaphots 0.0.6 and higher */
-        t30state = &fax.t30;
-#else
-        /* for spandsp release 0.0.5 */
-        t30state = &fax.t30_state;
-#endif
-
-	original_read_fmt = s->chan->readformat;
-	if (original_read_fmt != AST_FORMAT_SLINEAR) {
-		res = ast_set_read_format(s->chan, AST_FORMAT_SLINEAR);
-		if (res < 0) {
-			ast_log(LOG_WARNING, "Unable to set to linear read mode, giving up\n");
-			goto done;
-		}
-	}
-
-	original_write_fmt = s->chan->writeformat;
-	if (original_write_fmt != AST_FORMAT_SLINEAR) {
-		res = ast_set_write_format(s->chan, AST_FORMAT_SLINEAR);
-		if (res < 0) {
-			ast_log(LOG_WARNING, "Unable to set to linear write mode, giving up\n");
-			goto done;
-		}
-	}
-
-	/* Initialize T30 terminal */
-	fax_init(&fax, s->caller_mode);
-
-	/* Setup logging */
-	set_logging(&fax.logging);
-	set_logging(&t30state->logging);
-
-	/* Configure terminal */
-	set_local_info(t30state, s);
-	set_file(t30state, s);
-	set_ecm(t30state, TRUE);
-
-	fax_set_transmit_on_idle(&fax, TRUE);
-
-	t30_set_phase_e_handler(t30state, phase_e_handler, s);
-
-	start = state_change = ast_tvnow();
-
-	ast_activate_generator(s->chan, &generator, &fax);
-
-	while (!s->finished) {
-		inf = NULL;
-
-		if ((res = ast_waitfor(s->chan, 25)) < 0) {
-			ast_debug(1, "Error waiting for a frame\n");
-			break;
-		}
-
-		/* Watchdog */
-		now = ast_tvnow();
-		if (ast_tvdiff_sec(now, start) > WATCHDOG_TOTAL_TIMEOUT || ast_tvdiff_sec(now, state_change) > WATCHDOG_STATE_TIMEOUT) {
-			ast_log(LOG_WARNING, "It looks like we hung. Aborting.\n");
-			res = -1;
-			break;
-		}
-
-		if (!res) {
-			/* There was timeout waiting for a frame. Loop around and wait again */
-			continue;
-		}
-
-		/* There is a frame available. Get it */
-		res = 0;
-
-		if (!(inf = ast_read(s->chan))) {
-			ast_debug(1, "Channel hangup\n");
-			res = -1;
-			break;
-		}
-
-		ast_debug(10, "frame %d/%d, len=%d\n", inf->frametype, inf->subclass, inf->datalen);
-
-		/* Check the frame type. Format also must be checked because there is a chance
-		   that a frame in old format was already queued before we set channel format
-		   to slinear so it will still be received by ast_read */
-		if (inf->frametype == AST_FRAME_VOICE && inf->subclass == AST_FORMAT_SLINEAR) {
-			if (fax_rx(&fax, inf->data, inf->samples) < 0) {
-				/* I know fax_rx never returns errors. The check here is for good style only */
-				ast_log(LOG_WARNING, "fax_rx returned error\n");
-				res = -1;
-				break;
-			}
-			if (last_state != t30state->state) {
-				state_change = ast_tvnow();
-				last_state = t30state->state;
-			}
-		} else if ((inf->frametype == AST_FRAME_CONTROL) &&
-			   (inf->subclass == AST_CONTROL_T38_PARAMETERS)) {
-			struct ast_control_t38_parameters *parameters = inf->data;
-
-			if (parameters->request_response == AST_T38_NEGOTIATED) {
-				/* T38 switchover completed */
-				s->t38parameters = *parameters;
-				ast_debug(1, "T38 negotiated, finishing audio loop\n");
-				res = 1;
-				break;
-			} else if (parameters->request_response == AST_T38_REQUEST_NEGOTIATE) {
-				t38_parameters.request_response = AST_T38_NEGOTIATED;
-				ast_debug(1, "T38 request received, accepting\n");
-				/* Complete T38 switchover */
-				ast_indicate_data(s->chan, AST_CONTROL_T38_PARAMETERS, &t38_parameters, sizeof(t38_parameters));
-				/* Do not break audio loop, wait until channel driver finally acks switchover
-				 * with AST_T38_NEGOTIATED
-				 */
-			}
-		}
-
-		ast_frfree(inf);
-		inf = NULL;
-	}
-
-	ast_debug(1, "Loop finished, res=%d\n", res);
-
-	if (inf)
-		ast_frfree(inf);
-
-	ast_deactivate_generator(s->chan);
-
-	/* If we are switching to T38, remove phase E handler. Otherwise it will be executed
-	   by t30_terminate, display diagnostics and set status variables although no transmittion
-	   has taken place yet. */
-	if (res > 0) {
-		t30_set_phase_e_handler(t30state, NULL, NULL);
-	}
-
-	t30_terminate(t30state);
-	fax_release(&fax);
-
-done:
-	if (original_write_fmt != AST_FORMAT_SLINEAR) {
-		if (ast_set_write_format(s->chan, original_write_fmt) < 0)
-			ast_log(LOG_WARNING, "Unable to restore write format on '%s'\n", s->chan->name);
-	}
-
-	if (original_read_fmt != AST_FORMAT_SLINEAR) {
-		if (ast_set_read_format(s->chan, original_read_fmt) < 0)
-			ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", s->chan->name);
-	}
-
-	return res;
-
-}
-
-static int transmit_t38(fax_session *s)
-{
-	int res = 0;
-	t38_terminal_state_t t38;
-	struct ast_frame *inf = NULL;
-	int last_state = 0;
-	struct timeval now, start, state_change, last_frame;
-	t30_state_t *t30state;
-	t38_core_state_t *t38state;
-
-#if SPANDSP_RELEASE_DATE >= 20080725
-	/* for spandsp shaphots 0.0.6 and higher */
-	t30state = &t38.t30;
-	t38state = &t38.t38_fe.t38;
-#else
-	/* for spandsp releases 0.0.5 */
-	t30state = &t38.t30_state;
-	t38state = &t38.t38;
-#endif
-
-	/* Initialize terminal */
-	memset(&t38, 0, sizeof(t38));
-	if (t38_terminal_init(&t38, s->caller_mode, t38_tx_packet_handler, s->chan) == NULL) {
-		ast_log(LOG_WARNING, "Unable to start T.38 termination.\n");
-		res = -1;
-		goto disable_t38;
-	}
-
-	t38_set_max_datagram_size(t38state, s->t38parameters.max_ifp);
-
-	if (s->t38parameters.fill_bit_removal) {
-		t38_set_fill_bit_removal(t38state, TRUE);
-	}
-	if (s->t38parameters.transcoding_mmr) {
-		t38_set_mmr_transcoding(t38state, TRUE);
-	}
-	if (s->t38parameters.transcoding_jbig) {
-		t38_set_jbig_transcoding(t38state, TRUE);
-	}
-
-	/* Setup logging */
-	set_logging(&t38.logging);
-	set_logging(&t30state->logging);
-	set_logging(&t38state->logging);
-
-	/* Configure terminal */
-	set_local_info(t30state, s);
-	set_file(t30state, s);
-	set_ecm(t30state, TRUE);
-
-	t30_set_phase_e_handler(t30state, phase_e_handler, s);
-
-	now = start = state_change = ast_tvnow();
-
-	while (!s->finished) {
-
-		res = ast_waitfor(s->chan, 20);
-		if (res < 0)
-			break;
-		else if (res > 0)
-			res = 0;
-
-		last_frame = now;
-		now = ast_tvnow();
-		t38_terminal_send_timeout(&t38, ast_tvdiff_us(now, last_frame) / (1000000 / 8000));
-
-		inf = ast_read(s->chan);
-		if (inf == NULL) {
-			ast_debug(1, "Channel hangup\n");
-			res = -1;
-			break;
-		}
-
-		ast_debug(10, "frame %d/%d, len=%d\n", inf->frametype, inf->subclass, inf->datalen);
-
-		if (inf->frametype == AST_FRAME_MODEM && inf->subclass == AST_MODEM_T38) {
-			t38_core_rx_ifp_packet(t38state, inf->data, inf->datalen, inf->seqno);
-
-			/* Watchdog */
-			if (last_state != t30state->state) {
-				state_change = ast_tvnow();
-				last_state = t30state->state;
-			}
-		} else if (inf->frametype == AST_FRAME_CONTROL && inf->subclass == AST_CONTROL_T38_PARAMETERS) {
-			struct ast_control_t38_parameters *parameters = inf->data;
-			if (parameters->request_response == AST_T38_TERMINATED) {
-				ast_debug(1, "T38 down, finishing\n");
-				break;
-			}
-		}
-
-		ast_frfree(inf);
-		inf = NULL;
-
-		/* Watchdog */
-		if (ast_tvdiff_sec(now, start) > WATCHDOG_TOTAL_TIMEOUT || ast_tvdiff_sec(now, state_change) > WATCHDOG_STATE_TIMEOUT) {
-			ast_log(LOG_WARNING, "It looks like we hung. Aborting.\n");
-			res = -1;
-			break;
-		}
-	}
-
-	ast_debug(1, "Loop finished, res=%d\n", res);
-
-	if (inf)
-		ast_frfree(inf);
-
-	t30_terminate(t30state);
-	t38_terminal_release(&t38);
-
-disable_t38:
-	/* if we are not the caller, it's our job to shut down the T.38
-	 * session when the FAX transmisson is complete.
-	 */
-	if ((s->caller_mode == FALSE) &&
-	    (ast_channel_get_t38_state(s->chan) == T38_STATE_NEGOTIATED)) {
-		struct ast_control_t38_parameters t38_parameters = { .request_response = AST_T38_REQUEST_TERMINATE, };
-
-		if (ast_indicate_data(s->chan, AST_CONTROL_T38_PARAMETERS, &t38_parameters, sizeof(t38_parameters)) == 0) {
-			/* wait up to five seconds for negotiation to complete */
-			unsigned int timeout = 5000;
-			int ms;
-			
-			ast_debug(1, "Shutting down T.38 on %s\n", s->chan->name);
-			while (timeout > 0) {
-				ms = ast_waitfor(s->chan, 1000);
-				if (ms < 0) {
-					ast_log(LOG_WARNING, "something bad happened while channel '%s' was polling.\n", s->chan->name);
-					return -1;
-				}
-				if (!ms) {
-					/* nothing happened */
-					if (timeout > 0) {
-						timeout -= 1000;
-						continue;
-					} else {
-						ast_log(LOG_WARNING, "channel '%s' timed-out during the T.38 shutdown.\n", s->chan->name);
-						break;
-					}
-				}
-				if (!(inf = ast_read(s->chan))) {
-					return -1;
-				}
-				if ((inf->frametype == AST_FRAME_CONTROL) &&
-				    (inf->subclass == AST_CONTROL_T38_PARAMETERS) &&
-				    (inf->datalen == sizeof(t38_parameters))) {
-					struct ast_control_t38_parameters *parameters = inf->data;
-					
-					switch (parameters->request_response) {
-					case AST_T38_TERMINATED:
-						ast_debug(1, "Shut down T.38 on %s\n", s->chan->name);
-						break;
-					case AST_T38_REFUSED:
-						ast_log(LOG_WARNING, "channel '%s' refused to disable T.38\n", s->chan->name);
-						break;
-					default:
-						ast_log(LOG_ERROR, "channel '%s' failed to disable T.38\n", s->chan->name);
-						break;
-					}
-					ast_frfree(inf);
-					break;
-				}
-				ast_frfree(inf);
-			}
-		}
-	}
-
-	return res;
-}
-
-static int transmit(fax_session *s)
-{
-	int res = 0;
-
-	/* Clear all channel variables which to be set by the application.
-	   Pre-set status to error so in case of any problems we can just leave */
-	pbx_builtin_setvar_helper(s->chan, "FAXSTATUS", "FAILED"); 
-	pbx_builtin_setvar_helper(s->chan, "FAXERROR", "Channel problems"); 
-
-	pbx_builtin_setvar_helper(s->chan, "FAXMODE", NULL);
-	pbx_builtin_setvar_helper(s->chan, "REMOTESTATIONID", NULL);
-	pbx_builtin_setvar_helper(s->chan, "FAXPAGES", NULL);
-	pbx_builtin_setvar_helper(s->chan, "FAXRESOLUTION", NULL);
-	pbx_builtin_setvar_helper(s->chan, "FAXBITRATE", NULL); 
-
-	if (s->chan->_state != AST_STATE_UP) {
-		/* Shouldn't need this, but checking to see if channel is already answered
-		 * Theoretically asterisk should already have answered before running the app */
-		res = ast_answer(s->chan);
-		if (res) {
-			ast_log(LOG_WARNING, "Could not answer channel '%s'\n", s->chan->name);
-			return res;
-		}
-	}
-
-	s->t38state = ast_channel_get_t38_state(s->chan);
-	if (s->t38state != T38_STATE_NEGOTIATED) {
-		/* T38 is not negotiated on the channel yet. First start regular transmission. If it switches to T38, follow */	
-		pbx_builtin_setvar_helper(s->chan, "FAXMODE", "audio"); 
-		res = transmit_audio(s);
-		if (res > 0) {
-			/* transmit_audio reports switchover to T38. Update t38state */
-			s->t38state = ast_channel_get_t38_state(s->chan);
-			if (s->t38state != T38_STATE_NEGOTIATED) {
-				ast_log(LOG_ERROR, "Audio loop reports T38 switchover but t38state != T38_STATE_NEGOTIATED\n");
-			}
-		}
-	}
-
-	if (s->t38state == T38_STATE_NEGOTIATED) {
-		pbx_builtin_setvar_helper(s->chan, "FAXMODE", "T38"); 
-		res = transmit_t38(s);
-	}
-
-	if (res) {
-		ast_log(LOG_WARNING, "Transmission error\n");
-		res = -1;
-	} else if (s->finished < 0) {
-		ast_log(LOG_WARNING, "Transmission failed\n");
-	} else if (s->finished > 0) {
-		ast_debug(1, "Transmission finished Ok\n");
-	}
-
-	return res;
-}
-
-/* === Application functions === */
-
-static int sndfax_exec(struct ast_channel *chan, void *data)
-{
-	int res = 0;
-	char *parse;
-	fax_session session = { 0, };
-
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(file_name);
-		AST_APP_ARG(options);
-	);
-
-	if (chan == NULL) {
-		ast_log(LOG_ERROR, "Fax channel is NULL. Giving up.\n");
-		return -1;
-	}
-
-	/* 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 */
-		ast_log(LOG_ERROR, "SendFAX requires an argument (filename)\n");
-		return -1;
-	}
-
-	parse = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, parse);
-	
-	session.caller_mode = TRUE;
-
-	if (args.options) {
-		if (strchr(args.options, 'a'))
-			session.caller_mode = FALSE;
-	}
-
-	/* Done parsing */
-	session.direction = 1;
-	session.file_name = args.file_name;
-	session.chan = chan;
-	session.finished = 0;
-
-	res = transmit(&session);
-
-	return res;
-}
-
-static int rcvfax_exec(struct ast_channel *chan, void *data)
-{
-	int res = 0;
-	char *parse;
-	fax_session session;
-
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(file_name);
-		AST_APP_ARG(options);
-	);
-
-	if (chan == NULL) {
-		ast_log(LOG_ERROR, "Fax channel is NULL. Giving up.\n");
-		return -1;
-	}
-
-	/* 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 */
-		ast_log(LOG_ERROR, "ReceiveFAX requires an argument (filename)\n");
-		return -1;
-	}
-
-	parse = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, parse);
-	
-	session.caller_mode = FALSE;
-
-	if (args.options) {
-		if (strchr(args.options, 'c'))
-			session.caller_mode = TRUE;
-	}
-
-	/* Done parsing */
-	session.direction = 0;
-	session.file_name = args.file_name;
-	session.chan = chan;
-	session.finished = 0;
-
-	res = transmit(&session);
-
-	return res;
-}
-
-static int unload_module(void)
-{
-	int res;
-
-	res = ast_unregister_application(app_sndfax_name);	
-	res |= ast_unregister_application(app_rcvfax_name);	
-
-	return res;
-}
-
-static int load_module(void)
-{
-	int res ;
-
-	res = ast_register_application(app_sndfax_name, sndfax_exec, app_sndfax_synopsis, app_sndfax_desc);
-	res |= ast_register_application(app_rcvfax_name, rcvfax_exec, app_rcvfax_synopsis, app_rcvfax_desc);
-
-	/* The default SPAN message handler prints to stderr. It is something we do not want */
-	span_set_message_handler(NULL);
-
-	return res;
-}
-
-
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Simple FAX Application",
-		.load = load_module,
-		.unload = unload_module,
-		);
-
-

apps/app_festival.c

@@ -19,35 +19,33 @@
 /*! \file
  *
  * \brief Connect to festival
- *
- * \author Christos Ricudis <ricudis@itc.auth.gr>
- *
- * \extref  The Festival Speech Synthesis System - http://www.cstr.ed.ac.uk/projects/festival/
  * 
  * \ingroup applications
  */
 
-/*** MODULEINFO
-	<depend>working_fork</depend>
- ***/
+#include <sys/types.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <netdb.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#include <stdio.h>
+#include <signal.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <ctype.h>
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include <sys/socket.h>
-#include <netdb.h>
-#include <netinet/in.h>
-#include <arpa/inet.h>
-#include <signal.h>
-#include <fcntl.h>
-#include <ctype.h>
-#include <errno.h>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
-
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
@@ -55,132 +53,120 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/config.h"
 #include "asterisk/utils.h"
 #include "asterisk/lock.h"
-#include "asterisk/app.h"
+#include "asterisk/options.h"
 
 #define FESTIVAL_CONFIG "festival.conf"
-#define MAXLEN 180
-#define MAXFESTLEN 2048
+
+static char *tdesc = "Simple Festival Interface";
 
 static char *app = "Festival";
 
 static char *synopsis = "Say text to the user";
 
 static char *descrip = 
-"  Festival(text[,intkeys]):  Connect to Festival, send the argument, get back the waveform,\n"
+"  Festival(text[|intkeys]):  Connect to Festival, send the argument, get back the waveform,"
 "play it to the user, allowing any given interrupt keys to immediately terminate and return\n"
 "the value, or 'any' to allow any number back (useful in dialplan)\n";
 
+STANDARD_LOCAL_USER;
 
-static char *socket_receive_file_to_buff(int fd, int *size)
+LOCAL_USER_DECL;
+
+static char *socket_receive_file_to_buff(int fd,int *size)
 {
-	/* Receive file (probably a waveform file) from socket using
-	 * Festival key stuff technique, but long winded I know, sorry
-	 * but will receive any file without closing the stream or
-	 * using OOB data
-	 */
-	static char *file_stuff_key = "ft_StUfF_key"; /* must == Festival's key */
-	char *buff, *tmp;
-	int bufflen;
-	int n,k,i;
-	char c;
+    /* Receive file (probably a waveform file) from socket using   */
+    /* Festival key stuff technique, but long winded I know, sorry */
+    /* but will receive any file without closeing the stream or    */
+    /* using OOB data                                              */
+    static char *file_stuff_key = "ft_StUfF_key"; /* must == Festival's key */
+    char *buff;
+    int bufflen;
+    int n,k,i;
+    char c;
 
-	bufflen = 1024;
-	if (!(buff = ast_malloc(bufflen)))
-		return NULL;
-	*size = 0;
+    bufflen = 1024;
+    buff = (char *)malloc(bufflen);
+    *size=0;
 
-	for (k = 0; file_stuff_key[k] != '\0';) {
-		n = read(fd, &c, 1);
-		if (n == 0)
-			break;  /* hit stream eof before end of file */
-		if ((*size) + k + 1 >= bufflen) {
-			/* +1 so you can add a terminating NULL if you want */
-			bufflen += bufflen / 4;
-			if (!(tmp = ast_realloc(buff, bufflen))) {
-				ast_free(buff);
-				return NULL;
-			}
-			buff = tmp;
-		}
-		if (file_stuff_key[k] == c)
-			k++;
-		else if ((c == 'X') && (file_stuff_key[k+1] == '\0')) {
-			/* It looked like the key but wasn't */
-			for (i = 0; i < k; i++, (*size)++)
-				buff[*size] = file_stuff_key[i];
-			k = 0;
-			/* omit the stuffed 'X' */
-		} else {
-			for (i = 0; i < k; i++, (*size)++)
-				buff[*size] = file_stuff_key[i];
-			k = 0;
-			buff[*size] = c;
-			(*size)++;
-		}
-	}
+    for (k=0; file_stuff_key[k] != '\0';)
+    {
+        n = read(fd,&c,1);
+        if (n==0) break;  /* hit stream eof before end of file */
+        if ((*size)+k+1 >= bufflen)
+        {   /* +1 so you can add a NULL if you want */
+            bufflen += bufflen/4;
+            buff = (char *)realloc(buff,bufflen);
+        }
+        if (file_stuff_key[k] == c)
+            k++;
+        else if ((c == 'X') && (file_stuff_key[k+1] == '\0'))
+        {   /* It looked like the key but wasn't */
+            for (i=0; i < k; i++,(*size)++)
+                buff[*size] = file_stuff_key[i];
+            k=0;
+            /* omit the stuffed 'X' */
+        }
+        else
+        {
+            for (i=0; i < k; i++,(*size)++)
+                buff[*size] = file_stuff_key[i];
+            k=0;
+            buff[*size] = c;
+            (*size)++;
+        }
 
-	return buff;
+    }
+
+    return buff;
 }
 
-static int send_waveform_to_fd(char *waveform, int length, int fd)
-{
-	int res;
-	int x;
+static int send_waveform_to_fd(char *waveform, int length, int fd) {
+
+        int res;
+        int x;
 #ifdef __PPC__ 
 	char c;
 #endif
 	sigset_t fullset, oldset;
-#ifdef HAVE_CAP
-	cap_t cap;
-#endif
 
 	sigfillset(&fullset);
 	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
 
-	res = fork();
-	if (res < 0)
-		ast_log(LOG_WARNING, "Fork failed\n");
-	if (res) {
+        res = fork();
+        if (res < 0)
+                ast_log(LOG_WARNING, "Fork failed\n");
+        if (res) {
 		pthread_sigmask(SIG_SETMASK, &oldset, NULL);
-		return res;
+                return res;
 	}
-#ifdef HAVE_CAP
-	cap = cap_from_text("cap_net_admin-eip");
-
-	if (cap_set_proc(cap)) {
-		/* Careful with order! Logging cannot happen after we close FDs */
-		ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-	}
-	cap_free(cap);
-#endif
-	for (x = 0; x < 256; x++) {
-		if (x != fd)
-			close(x);
-	}
-	if (ast_opt_high_priority)
+        for (x=0;x<256;x++) {
+                if (x != fd)
+                        close(x);
+        }
+	if (option_highpriority)
 		ast_set_priority(0);
 	signal(SIGPIPE, SIG_DFL);
 	pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
+/*IAS */
 #ifdef __PPC__  
-	for (x = 0; x < length; x += 2) {
-		c = *(waveform + x + 1);
-		*(waveform + x + 1) = *(waveform + x);
-		*(waveform + x) = c;
+	for( x=0; x<length; x+=2)
+	{
+		c = *(waveform+x+1);
+		*(waveform+x+1)=*(waveform+x);
+		*(waveform+x)=c;
 	}
 #endif
 	
-	if (write(fd, waveform, length) < 0) {
-		ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
-	}
-
+	write(fd,waveform,length);
 	close(fd);
 	exit(0);
 }
 
-static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, int length, char *intkeys)
-{
-	int res = 0;
+
+static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, int length, char *intkeys) {
+	int res=0;
 	int fds[2];
+	int ms = -1;
 	int pid = -1;
 	int needed = 0;
 	int owriteformat;
@@ -189,15 +175,13 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 		struct ast_frame f;
 		char offset[AST_FRIENDLY_OFFSET];
 		char frdata[2048];
-	} myf = {
-		.f = { 0, },
-	};
-
-	if (pipe(fds)) {
-		ast_log(LOG_WARNING, "Unable to create pipe\n");
-		return -1;
-	}
-
+	} myf;
+	
+        if (pipe(fds)) {
+                 ast_log(LOG_WARNING, "Unable to create pipe\n");
+        	return -1;
+        }
+	                                                
 	/* Answer if it's not already going */
 	if (chan->_state != AST_STATE_UP)
 		ast_answer(chan);
@@ -211,13 +195,14 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 		return -1;
 	}
 	
-	res = send_waveform_to_fd(waveform, length, fds[1]);
+	res=send_waveform_to_fd(waveform,length,fds[1]);
 	if (res >= 0) {
 		pid = res;
 		/* Order is important -- there's almost always going to be mp3...  we want to prioritize the
 		   user */
 		for (;;) {
-			res = ast_waitfor(chan, 1000);
+			ms = 1000;
+			res = ast_waitfor(chan, ms);
 			if (res < 1) {
 				res = -1;
 				break;
@@ -229,7 +214,7 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 				break;
 			}
 			if (f->frametype == AST_FRAME_DTMF) {
-				ast_debug(1, "User pressed a key\n");
+				ast_log(LOG_DEBUG, "User pressed a key\n");
 				if (intkeys && strchr(intkeys, f->subclass)) {
 					res = f->subclass;
 					ast_frfree(f);
@@ -250,6 +235,7 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 					myf.f.subclass = AST_FORMAT_SLINEAR;
 					myf.f.datalen = res;
 					myf.f.samples = res / 2;
+					myf.f.mallocd = 0;
 					myf.f.offset = AST_FRIENDLY_OFFSET;
 					myf.f.src = __PRETTY_FUNCTION__;
 					myf.f.data = myf.frdata;
@@ -259,13 +245,13 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 						break;
 					}
 					if (res < needed) { /* last frame */
-						ast_debug(1, "Last frame\n");
-						res = 0;
+						ast_log(LOG_DEBUG, "Last frame\n");
+						res=0;
 						ast_frfree(f);
 						break;
 					}
 				} else {
-					ast_debug(1, "No more waveform\n");
+					ast_log(LOG_DEBUG, "No more waveform\n");
 					res = 0;
 				}
 			}
@@ -275,29 +261,34 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 	close(fds[0]);
 	close(fds[1]);
 
-#if 0
-	if (pid > -1)
-		kill(pid, SIGKILL);
-#endif
+/*	if (pid > -1) */
+/*		kill(pid, SIGKILL); */
 	if (!res && owriteformat)
 		ast_set_write_format(chan, owriteformat);
 	return res;
 }
 
+#define MAXLEN 180
+#define MAXFESTLEN 2048
+
+
+
+
 static int festival_exec(struct ast_channel *chan, void *vdata)
 {
 	int usecache;
-	int res = 0;
-	struct sockaddr_in serv_addr;
+	int res=0;
+	struct localuser *u;
+ 	struct sockaddr_in serv_addr;
 	struct hostent *serverhost;
 	struct ast_hostent ahp;
 	int fd;
 	FILE *fs;
-	const char *host;
-	const char *cachedir;
-	const char *temp;
-	const char *festivalcommand;
-	int port = 1314;
+	char *host;
+	char *cachedir;
+	char *temp;
+	char *festivalcommand;
+	int port=1314;
 	int n;
 	char ack[4];
 	char *waveform;
@@ -310,29 +301,27 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
 	char MD5Hex[33] = "";
 	char koko[4] = "";
 	char cachefile[MAXFESTLEN]="";
-	int readcache = 0;
-	int writecache = 0;
+	int readcache=0;
+	int writecache=0;
 	int strln;
 	int fdesc = -1;
 	char buffer[16384];
 	int seekpos = 0;	
 	char *data;	
+	char *intstr;
 	struct ast_config *cfg;
-	char *newfestivalcommand;
-	struct ast_flags config_flags = { 0 };
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(text);
-		AST_APP_ARG(interrupt);
-	);
 
 	if (ast_strlen_zero(vdata)) {
 		ast_log(LOG_WARNING, "festival requires an argument (text)\n");
 		return -1;
 	}
 
-	cfg = ast_config_load(FESTIVAL_CONFIG, config_flags);
+	LOCAL_USER_ADD(u);
+
+	cfg = ast_config_load(FESTIVAL_CONFIG);
 	if (!cfg) {
 		ast_log(LOG_WARNING, "No such configuration file %s\n", FESTIVAL_CONFIG);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	if (!(host = ast_variable_retrieve(cfg, "general", "host"))) {
@@ -344,226 +333,216 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
 		port = atoi(temp);
 	}
 	if (!(temp = ast_variable_retrieve(cfg, "general", "usecache"))) {
-		usecache = 0;
+		usecache=0;
 	} else {
 		usecache = ast_true(temp);
 	}
 	if (!(cachedir = ast_variable_retrieve(cfg, "general", "cachedir"))) {
 		cachedir = "/tmp/";
 	}
-
-	data = ast_strdupa(vdata);
-	AST_STANDARD_APP_ARGS(args, data);
-
 	if (!(festivalcommand = ast_variable_retrieve(cfg, "general", "festivalcommand"))) {
-		const char *startcmd = "(tts_textasterisk \"";
-		const char *endcmd = "\" 'file)(quit)\n";
-
-		strln = strlen(startcmd) + strlen(args.text) + strlen(endcmd) + 1;
-		newfestivalcommand = alloca(strln);
-		snprintf(newfestivalcommand, strln, "%s%s%s", startcmd, args.text, endcmd);
-		festivalcommand = newfestivalcommand;
-	} else { /* This else parses the festivalcommand that we're sent from the config file for \n's, etc */
-		int i, j;
-		newfestivalcommand = alloca(strlen(festivalcommand) + strlen(args.text) + 1);
-
-		for (i = 0, j = 0; i < strlen(festivalcommand); i++) {
-			if (festivalcommand[i] == '\\' && festivalcommand[i + 1] == 'n') {
-				newfestivalcommand[j++] = '\n';
-				i++;
-			} else if (festivalcommand[i] == '\\') {
-				newfestivalcommand[j++] = festivalcommand[i + 1];
-				i++;
-			} else if (festivalcommand[i] == '%' && festivalcommand[i + 1] == 's') {
-				sprintf(&newfestivalcommand[j], "%s", args.text); /* we know it is big enough */
-				j += strlen(args.text);
-				i++;
-			} else
-				newfestivalcommand[j++] = festivalcommand[i];
-		}
-		newfestivalcommand[j] = '\0';
-		festivalcommand = newfestivalcommand;
+		festivalcommand = "(tts_textasterisk \"%s\" 'file)(quit)\n";
 	}
 	
-	if (args.interrupt && !strcasecmp(args.interrupt, "any"))
-		args.interrupt = AST_DIGIT_ANY;
-
-	ast_debug(1, "Text passed to festival server : %s\n", args.text);
-	/* Connect to local festival server */
-	
-	fd = socket(AF_INET, SOCK_STREAM, IPPROTO_TCP);
-
-	if (fd < 0) {
-		ast_log(LOG_WARNING, "festival_client: can't get socket\n");
+	data = ast_strdupa(vdata);
+	if (!data) {
+		ast_log(LOG_ERROR, "Out of memery\n");
 		ast_config_destroy(cfg);
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 
-	memset(&serv_addr, 0, sizeof(serv_addr));
-
-	if ((serv_addr.sin_addr.s_addr = inet_addr(host)) == -1) {
-		/* its a name rather than an ipnum */
-		serverhost = ast_gethostbyname(host, &ahp);
-
-		if (serverhost == NULL) {
-			ast_log(LOG_WARNING, "festival_client: gethostbyname failed\n");
-			ast_config_destroy(cfg);
-			return -1;
-		}
-		memmove(&serv_addr.sin_addr, serverhost->h_addr, serverhost->h_length);
+	intstr = strchr(data, '|');
+	if (intstr) {	
+		*intstr = '\0';
+		intstr++;
+		if (!strcasecmp(intstr, "any"))
+			intstr = AST_DIGIT_ANY;
 	}
+	
+	ast_log(LOG_DEBUG, "Text passed to festival server : %s\n",(char *)data);
+	/* Connect to local festival server */
+	
+    	fd = socket(AF_INET, SOCK_STREAM, IPPROTO_TCP);
 
+    	if (fd < 0) {
+		ast_log(LOG_WARNING,"festival_client: can't get socket\n");
+		ast_config_destroy(cfg);
+		LOCAL_USER_REMOVE(u);
+        	return -1;
+	}
+        memset(&serv_addr, 0, sizeof(serv_addr));
+        if ((serv_addr.sin_addr.s_addr = inet_addr(host)) == -1) {
+	        /* its a name rather than an ipnum */
+	        serverhost = ast_gethostbyname(host, &ahp);
+	        if (serverhost == (struct hostent *)0) {
+        	    	ast_log(LOG_WARNING,"festival_client: gethostbyname failed\n");
+			ast_config_destroy(cfg);
+			LOCAL_USER_REMOVE(u);
+	            	return -1;
+        	}
+	        memmove(&serv_addr.sin_addr,serverhost->h_addr, serverhost->h_length);
+    	}
 	serv_addr.sin_family = AF_INET;
 	serv_addr.sin_port = htons(port);
 
 	if (connect(fd, (struct sockaddr *)&serv_addr, sizeof(serv_addr)) != 0) {
-		ast_log(LOG_WARNING, "festival_client: connect to server failed\n");
+		ast_log(LOG_WARNING,"festival_client: connect to server failed\n");
 		ast_config_destroy(cfg);
-		return -1;
+		LOCAL_USER_REMOVE(u);
+        	return -1;
+    	}
+    	
+    	/* Compute MD5 sum of string */
+    	MD5Init(&md5ctx);
+    	MD5Update(&md5ctx,(unsigned char const *)data,strlen(data));
+    	MD5Final(MD5Res,&md5ctx);
+		MD5Hex[0] = '\0';
+    	
+    	/* Convert to HEX and look if there is any matching file in the cache 
+    		directory */
+    	for (i=0;i<16;i++) {
+    		snprintf(koko, sizeof(koko), "%X",MD5Res[i]);
+    		strncat(MD5Hex, koko, sizeof(MD5Hex) - strlen(MD5Hex) - 1);
+    	}
+    	readcache=0;
+    	writecache=0;
+    	if (strlen(cachedir)+strlen(MD5Hex)+1<=MAXFESTLEN && (usecache==-1)) {
+    		snprintf(cachefile, sizeof(cachefile), "%s/%s", cachedir, MD5Hex);
+    		fdesc=open(cachefile,O_RDWR);
+    		if (fdesc==-1) {
+    			fdesc=open(cachefile,O_CREAT|O_RDWR,0777);
+    			if (fdesc!=-1) {
+    				writecache=1;
+    				strln=strlen((char *)data);
+    				ast_log(LOG_DEBUG,"line length : %d\n",strln);
+    				write(fdesc,&strln,sizeof(int));
+    				write(fdesc,data,strln);
+				seekpos=lseek(fdesc,0,SEEK_CUR);
+				ast_log(LOG_DEBUG,"Seek position : %d\n",seekpos);
+    			}
+    		} else {
+    			read(fdesc,&strln,sizeof(int));
+    			ast_log(LOG_DEBUG,"Cache file exists, strln=%d, strlen=%d\n",strln,(int)strlen((char *)data));
+    			if (strlen((char *)data)==strln) {
+    				ast_log(LOG_DEBUG,"Size OK\n");
+    				read(fdesc,&bigstring,strln);
+	    			bigstring[strln] = 0;
+				if (strcmp(bigstring,data)==0) { 
+	    				readcache=1;
+	    			} else {
+	    				ast_log(LOG_WARNING,"Strings do not match\n");
+	    			}
+	    		} else {
+	    			ast_log(LOG_WARNING,"Size mismatch\n");
+	    		}
+	    	}
 	}
-
-	/* Compute MD5 sum of string */
-	MD5Init(&md5ctx);
-	MD5Update(&md5ctx, (unsigned char *)args.text, strlen(args.text));
-	MD5Final(MD5Res, &md5ctx);
-	MD5Hex[0] = '\0';
-
-	/* Convert to HEX and look if there is any matching file in the cache 
-		directory */
-	for (i = 0; i < 16; i++) {
-		snprintf(koko, sizeof(koko), "%X", MD5Res[i]);
-		strncat(MD5Hex, koko, sizeof(MD5Hex) - strlen(MD5Hex) - 1);
-	}
-	readcache = 0;
-	writecache = 0;
-	if (strlen(cachedir) + strlen(MD5Hex) + 1 <= MAXFESTLEN && (usecache == -1)) {
-		snprintf(cachefile, sizeof(cachefile), "%s/%s", cachedir, MD5Hex);
-		fdesc = open(cachefile, O_RDWR);
-		if (fdesc == -1) {
-			fdesc = open(cachefile, O_CREAT | O_RDWR, AST_FILE_MODE);
-			if (fdesc != -1) {
-				writecache = 1;
-				strln = strlen(args.text);
-				ast_debug(1, "line length : %d\n", strln);
-    				if (write(fdesc,&strln,sizeof(int)) < 0) {
-					ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
-				}
-    				if (write(fdesc,data,strln) < 0) {
-					ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
-				}
-				seekpos = lseek(fdesc, 0, SEEK_CUR);
-				ast_debug(1, "Seek position : %d\n", seekpos);
-			}
-		} else {
-    			if (read(fdesc,&strln,sizeof(int)) != sizeof(int)) {
-				ast_log(LOG_WARNING, "read() failed: %s\n", strerror(errno));
-			}
-			ast_debug(1, "Cache file exists, strln=%d, strlen=%d\n", strln, (int)strlen(args.text));
-			if (strlen(args.text) == strln) {
-				ast_debug(1, "Size OK\n");
-    				if (read(fdesc,&bigstring,strln) != strln) {
-					ast_log(LOG_WARNING, "read() failed: %s\n", strerror(errno));
-				}
-				bigstring[strln] = 0;
-				if (strcmp(bigstring, args.text) == 0) { 
-					readcache = 1;
-				} else {
-					ast_log(LOG_WARNING, "Strings do not match\n");
-				}
-			} else {
-				ast_log(LOG_WARNING, "Size mismatch\n");
-			}
-		}
-	}
-
-	if (readcache == 1) {
+    			
+	if (readcache==1) {
 		close(fd);
-		fd = fdesc;
-		ast_debug(1, "Reading from cache...\n");
+		fd=fdesc;
+		ast_log(LOG_DEBUG,"Reading from cache...\n");
 	} else {
-		ast_debug(1, "Passing text to festival...\n");
-		fs = fdopen(dup(fd), "wb");
-
-		fprintf(fs, "%s", festivalcommand);
+		ast_log(LOG_DEBUG,"Passing text to festival...\n");
+    		fs=fdopen(dup(fd),"wb");
+		fprintf(fs,festivalcommand,(char *)data);
 		fflush(fs);
 		fclose(fs);
 	}
 	
 	/* Write to cache and then pass it down */
-	if (writecache == 1) {
-		ast_debug(1, "Writing result to cache...\n");
-		while ((strln = read(fd, buffer, 16384)) != 0) {
-			if (write(fdesc,buffer,strln) < 0) {
-				ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
-			}
+	if (writecache==1) {
+		ast_log(LOG_DEBUG,"Writing result to cache...\n");
+		while ((strln=read(fd,buffer,16384))!=0) {
+			write(fdesc,buffer,strln);
 		}
 		close(fd);
 		close(fdesc);
-		fd = open(cachefile, O_RDWR);
-		lseek(fd, seekpos, SEEK_SET);
+		fd=open(cachefile,O_RDWR);
+		lseek(fd,seekpos,SEEK_SET);
 	}
 	
-	ast_debug(1, "Passing data to channel...\n");
-
+	ast_log(LOG_DEBUG,"Passing data to channel...\n");
+	
 	/* Read back info from server */
 	/* This assumes only one waveform will come back, also LP is unlikely */
 	wave = 0;
 	do {
-		int read_data;
-		for (n = 0; n < 3; ) {
-			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\n");
-				close(fd);
-				ast_config_destroy(cfg);
-				return -1;
-			}
-			n += read_data;
-		}
+               int read_data;
+		for (n=0; n < 3; )
+               {
+                       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\n");
+			       close(fd);
+			       ast_config_destroy(cfg);
+			       LOCAL_USER_REMOVE(u);
+                               return -1;
+                       }
+                       n += read_data;
+               }
 		ack[3] = '\0';
-		if (strcmp(ack, "WV\n") == 0) {         /* receive a waveform */
-			ast_debug(1, "Festival WV command\n");
-			if ((waveform = socket_receive_file_to_buff(fd, &filesize))) {
-				res = send_waveform_to_channel(chan, waveform, filesize, args.interrupt);
-				ast_free(waveform);
-			}
-			break;
-		} else if (strcmp(ack, "LP\n") == 0) {   /* receive an s-expr */
-			ast_debug(1, "Festival LP command\n");
-			if ((waveform = socket_receive_file_to_buff(fd, &filesize))) {
-				waveform[filesize] = '\0';
-				ast_log(LOG_WARNING, "Festival returned LP : %s\n", waveform);
-				ast_free(waveform);
-			}
-		} else if (strcmp(ack, "ER\n") == 0) {    /* server got an error */
-			ast_log(LOG_WARNING, "Festival returned ER\n");
-			res = -1;
+		if (strcmp(ack,"WV\n") == 0) {         /* receive a waveform */
+			ast_log(LOG_DEBUG,"Festival WV command\n");
+			waveform = socket_receive_file_to_buff(fd,&filesize);
+			res = send_waveform_to_channel(chan,waveform,filesize, intstr);
+			free(waveform);
 			break;
 		}
-	} while (strcmp(ack, "OK\n") != 0);
+		else if (strcmp(ack,"LP\n") == 0) {   /* receive an s-expr */
+			ast_log(LOG_DEBUG,"Festival LP command\n");
+			waveform = socket_receive_file_to_buff(fd,&filesize);
+			waveform[filesize]='\0';
+			ast_log(LOG_WARNING,"Festival returned LP : %s\n",waveform);
+			free(waveform);
+		} else if (strcmp(ack,"ER\n") == 0) {    /* server got an error */
+			ast_log(LOG_WARNING,"Festival returned ER\n");
+			res=-1;
+			break;
+    		}
+	} while (strcmp(ack,"OK\n") != 0);
 	close(fd);
 	ast_config_destroy(cfg);
+	LOCAL_USER_REMOVE(u);
+	return res;
+
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
 	return res;
 }
 
-static int unload_module(void)
+int load_module(void)
 {
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
-{
-	struct ast_flags config_flags = { 0 };
-	struct ast_config *cfg = ast_config_load(FESTIVAL_CONFIG, config_flags);
-	if (!cfg) {
-		ast_log(LOG_WARNING, "No such configuration file %s\n", FESTIVAL_CONFIG);
-		return AST_MODULE_LOAD_DECLINE;
-	}
-	ast_config_destroy(cfg);
+	
 	return ast_register_application(app, festival_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Simple Festival Interface");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_flash.c

@@ -18,50 +18,58 @@
 
 /*! \file
  *
- * \brief App to flash a DAHDI trunk
- *
- * \author Mark Spencer <markster@digium.com>
+ * \brief App to flash a zap trunk
  * 
  * \ingroup applications
  */
  
-/*** MODULEINFO
-	<depend>dahdi</depend>
- ***/
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+#ifdef __linux__
+#include <linux/zaptel.h>
+#else
+#include <zaptel.h>
+#endif /* __linux__ */
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include <sys/ioctl.h>
-#include <dahdi/user.h>
-
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/translate.h"
 #include "asterisk/image.h"
+#include "asterisk/options.h"
+
+static char *tdesc = "Flash zap trunk application";
 
 static char *app = "Flash";
 
-static char *synopsis = "Flashes a DAHDI Trunk";
+static char *synopsis = "Flashes a Zap Trunk";
 
 static char *descrip = 
-"Performs a flash on a DAHDI trunk.  This can be used\n"
-"to access features provided on an incoming analogue circuit\n"
-"such as conference and call waiting. Use with SendDTMF() to\n"
-"perform external transfers\n";
+"  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 oths application will only work on Zap trunks.\n";
 
+STANDARD_LOCAL_USER;
 
-static inline int dahdi_wait_event(int fd)
+LOCAL_USER_DECL;
+
+static inline int zt_wait_event(int fd)
 {
-	/* Avoid the silly dahdi_waitevent which ignores a bunch of events */
+	/* Avoid the silly zt_waitevent which ignores a bunch of events */
 	int i,j=0;
-	i = DAHDI_IOMUX_SIGEVENT;
-	if (ioctl(fd, DAHDI_IOMUX, &i) == -1) return -1;
-	if (ioctl(fd, DAHDI_GETEVENT, &j) == -1) return -1;
+	i = ZT_IOMUX_SIGEVENT;
+	if (ioctl(fd, ZT_IOMUX, &i) == -1) return -1;
+	if (ioctl(fd, ZT_GETEVENT, &j) == -1) return -1;
 	return j;
 }
 
@@ -69,45 +77,65 @@ static int flash_exec(struct ast_channel *chan, void *data)
 {
 	int res = -1;
 	int x;
-	struct dahdi_params dahdip;
-
-	if (strcasecmp(chan->tech->type, "DAHDI")) {
-		ast_log(LOG_WARNING, "%s is not a DAHDI channel\n", chan->name);
-		return -1;
-	}
-	
-	memset(&dahdip, 0, sizeof(dahdip));
-	res = ioctl(chan->fds[0], DAHDI_GET_PARAMS, &dahdip);
-	if (!res) {
-		if (dahdip.sigtype & __DAHDI_SIG_FXS) {
-			x = DAHDI_FLASH;
-			res = ioctl(chan->fds[0], DAHDI_HOOK, &x);
-			if (!res || (errno == EINPROGRESS)) {
-				if (res) {
-					/* Wait for the event to finish */
-					dahdi_wait_event(chan->fds[0]);
-				}
-				res = ast_safe_sleep(chan, 1000);
-				ast_verb(3, "Flashed channel %s\n", chan->name);
+	struct localuser *u;
+	struct zt_params ztp;
+	LOCAL_USER_ADD(u);
+	if (!strcasecmp(chan->type, "Zap")) {
+		memset(&ztp, 0, sizeof(ztp));
+		res = ioctl(chan->fds[0], ZT_GET_PARAMS, &ztp);
+		if (!res) {
+			if (ztp.sigtype & __ZT_SIG_FXS) {
+				x = ZT_FLASH;
+				res = ioctl(chan->fds[0], ZT_HOOK, &x);
+				if (!res || (errno == EINPROGRESS)) {
+					if (res) {
+						/* Wait for the event to finish */
+						zt_wait_event(chan->fds[0]);
+					}
+					res = ast_safe_sleep(chan, 1000);
+					if (option_verbose > 2)
+						ast_verbose(VERBOSE_PREFIX_3 "Flashed channel %s\n", chan->name);
+				} else
+					ast_log(LOG_WARNING, "Unable to flash channel %s: %s\n", chan->name, strerror(errno));
 			} else
-				ast_log(LOG_WARNING, "Unable to flash channel %s: %s\n", chan->name, strerror(errno));
+				ast_log(LOG_WARNING, "%s is not an FXO Channel\n", chan->name);
 		} else
-			ast_log(LOG_WARNING, "%s is not an FXO Channel\n", chan->name);
+			ast_log(LOG_WARNING, "Unable to get parameters of %s: %s\n", chan->name, strerror(errno));
 	} else
-		ast_log(LOG_WARNING, "Unable to get parameters of %s: %s\n", chan->name, strerror(errno));
+		ast_log(LOG_WARNING, "%s is not a Zap channel\n", chan->name);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
 
 	return res;
 }
 
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, flash_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Flash channel application");
+char *description(void)
+{
+	return tdesc;
+}
 
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_forkcdr.c

@@ -18,142 +18,42 @@
 /*! \file
  *
  * \brief Fork CDR application
- *
- * \author Anthony Minessale anthmct@yahoo.com
- *
- * \note Development of this app Sponsored/Funded by TAAN Softworks Corp
  * 
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/cdr.h"
-#include "asterisk/app.h"
 #include "asterisk/module.h"
 
+static char *tdesc = "Fork The CDR into 2 separate entities.";
 static char *app = "ForkCDR";
 static char *synopsis = 
 "Forks the Call Data Record";
 static char *descrip = 
 "  ForkCDR([options]):  Causes the Call Data Record to fork an additional\n"
-"cdr record starting from the time of the fork call. This new cdr record will\n"
-"be linked to end of the list of cdr records attached to the channel. The original CDR\n"
-"has a LOCKED flag set, which forces most cdr operations to skip it, except\n"
-"for the functions that set the answer and end times, which ignore the LOCKED\n"
-"flag. This allows all the cdr records in the channel to be 'ended' together\n"
-"when the channel is closed.\n"
-"The CDR() func (when setting CDR values) normally ignores the LOCKED flag also,\n"
-"but has options to vary its behavior. The 'T' option (described below), can\n"
-"override this behavior, but beware the risks.\n"
-"\n"
-"Detailed Behavior Description:\n"
-"First, this app finds the last cdr record in the list, and makes\n"
-"a copy of it. This new copy will be the newly forked cdr record.\n"
-"Next, this new record is linked to the end of the cdr record list.\n"
-"Next, The new cdr record is RESET (unless you use an option to prevent this)\n"
-"This means that:\n"
-"   1. All flags are unset on the cdr record\n"
-"   2. the start, end, and answer times are all set to zero.\n"
-"   3. the billsec and duration fields are set to zero.\n"
-"   4. the start time is set to the current time.\n"
-"   5. the disposition is set to NULL.\n"
-"Next, unless you specified the 'v' option, all variables will be\n"
-"removed from the original cdr record. Thus, the 'v' option allows\n"
-"any CDR variables to be replicated to all new forked cdr records.\n"
-"Without the 'v' option, the variables on the original are effectively\n"
-"moved to the new forked cdr record.\n"
-"Next, if the 's' option is set, the provided variable and value\n"
-"are set on the original cdr record.\n"
-"Next, if the 'a' option is given, and the original cdr record has an\n"
-"answer time set, then the new forked cdr record will have its answer\n"
-"time set to its start time. If the old answer time were carried forward,\n"
-"the answer time would be earlier than the start time, giving strange\n"
-"duration and billsec times.\n"
-"Next, if the 'd' option was specified, the disposition is copied from\n"
-"the original cdr record to the new forked cdr.\n"
-"Next, if the 'D' option was specified, the destination channel field\n"
-"in the new forked CDR is erased.\n"
-"Next, if the 'e' option was specified, the 'end' time for the original\n"
-"cdr record is set to the current time. Future hang-up or ending events\n"
-"will not override this time stamp.\n"
-"Next, If the 'A' option is specified, the original cdr record will have\n"
-"it ANS_LOCKED flag set, which prevent future answer events\n"
-"from updating the original cdr record's disposition. Normally, an\n"
-"'ANSWERED' event would mark all cdr records in the chain as 'ANSWERED'.\n"
-"Next, if the 'T' option is specified, the original cdr record will have\n"
-"its 'DONT_TOUCH' flag set, which will force the cdr_answer, cdr_end, and\n"
-"cdr_setvar functions to leave that cdr record alone.\n"
-"And, last but not least, the original cdr record has its LOCKED flag\n"
-"set. Almost all internal CDR functions (except for the funcs that set\n"
-"the end, and answer times, and set a variable) will honor this flag\n"
-"and leave a LOCKED cdr record alone.\n"
-"This means that the newly created forked cdr record will be affected\n"
-"by events transpiring within Asterisk, with the previously noted\n"
-"exceptions.\n"
-"  Options:\n"
-"    a - update the answer time on the NEW CDR just after it's been inited.\n"
-"         The new CDR may have been answered already. The reset that forkcdr\n"
-"         does will erase the answer time. This will bring it back, but\n"
-"         the answer time will be a copy of the fork/start time. It will\n"
-"         only do this if the initial cdr was indeed already answered.\n"
-"    A - Lock the original CDR against the answer time being updated.\n"
-"         This will allow the disposition on the original CDR to remain the same.\n"
-"    d - Copy the disposition forward from the old cdr, after the \n"
-"         init.\n"
-"    D - Clear the dstchannel on the new CDR after reset.\n"
-"    e - end the original CDR. Do this after all the neccessary data\n"
-"         is copied from the original CDR to the new forked CDR.\n"
-"    R -  do NOT reset the new cdr.\n"
-"    s(name=val) - Set the CDR var 'name' in the original CDR, with value\n"
-"                  'val'.\n"
-"    T -  Mark the original CDR with a DONT_TOUCH flag. setvar, answer, and end\n"
-"          cdr funcs will obey this flag; normally they don't honor the LOCKED\n"
-"          flag set on the original CDR record.\n"
-"          Beware-- using this flag may cause CDR's not to have their end times\n"
-"          updated! It is suggested that if you specify this flag, you might\n"
-"          wish to use the 'e' flag as well!\n"
-"    v  - When the new CDR is forked, it gets a copy of the vars attached\n"
-"         to the current CDR. The vars attached to the original CDR are removed\n"
-"         unless this option is specified.\n";
+	"cdr record starting from the time of the fork call\n"
+"If the option 'v' is passed all cdr variables will be passed along also.\n"
+"";
 
 
-enum {
-	OPT_SETANS =            (1 << 0),
-	OPT_SETDISP =           (1 << 1),
-	OPT_RESETDEST =         (1 << 2),
-	OPT_ENDCDR =            (1 << 3),
-	OPT_NORESET =           (1 << 4),
-	OPT_KEEPVARS =          (1 << 5),
-	OPT_VARSET =            (1 << 6),
-	OPT_ANSLOCK =           (1 << 7),
-	OPT_DONTOUCH =          (1 << 8),
-};
+STANDARD_LOCAL_USER;
 
-enum {
-	OPT_ARG_VARSET = 0,
-	/* note: this entry _MUST_ be the last one in the enum */
-	OPT_ARG_ARRAY_SIZE,
-};
+LOCAL_USER_DECL;
 
-AST_APP_OPTIONS(forkcdr_exec_options, {
-	AST_APP_OPTION('a', OPT_SETANS),
-	AST_APP_OPTION('A', OPT_ANSLOCK),
-	AST_APP_OPTION('d', OPT_SETDISP),
-	AST_APP_OPTION('D', OPT_RESETDEST),
-	AST_APP_OPTION('e', OPT_ENDCDR),
-	AST_APP_OPTION('R', OPT_NORESET),
-	AST_APP_OPTION_ARG('s', OPT_VARSET, OPT_ARG_VARSET),
-	AST_APP_OPTION('T', OPT_DONTOUCH),
-	AST_APP_OPTION('v', OPT_KEEPVARS),
-});
-
-static void ast_cdr_fork(struct ast_channel *chan, struct ast_flags optflags, char *set) 
+static void ast_cdr_fork(struct ast_channel *chan) 
 {
 	struct ast_cdr *cdr;
 	struct ast_cdr *newcdr;
@@ -168,84 +68,64 @@ static void ast_cdr_fork(struct ast_channel *chan, struct ast_flags optflags, ch
 		return;
 	
 	ast_cdr_append(cdr, newcdr);
-
-	if (!ast_test_flag(&optflags, OPT_NORESET))
-		ast_cdr_reset(newcdr, &flags);
-		
+	ast_cdr_reset(newcdr, &flags);
+	
 	if (!ast_test_flag(cdr, AST_CDR_FLAG_KEEP_VARS))
 		ast_cdr_free_vars(cdr, 0);
 	
-	if (!ast_strlen_zero(set)) {
-		char *varname = ast_strdupa(set), *varval;
-		varval = strchr(varname,'=');
-		if (varval) {
-			*varval = 0;
-			varval++;
-			ast_cdr_setvar(cdr, varname, varval, 0);
-		}
-	}
-	
-	if (ast_test_flag(&optflags, OPT_SETANS) && !ast_tvzero(cdr->answer))
-		newcdr->answer = newcdr->start;
-
-	if (ast_test_flag(&optflags, OPT_SETDISP))
-		newcdr->disposition = cdr->disposition;
-	
-	if (ast_test_flag(&optflags, OPT_RESETDEST))
-		newcdr->dstchannel[0] = 0;
-	
-	if (ast_test_flag(&optflags, OPT_ENDCDR))
-		ast_cdr_end(cdr);
-
-	if (ast_test_flag(&optflags, OPT_ANSLOCK))
-		ast_set_flag(cdr, AST_CDR_FLAG_ANSLOCKED);
-	
-	if (ast_test_flag(&optflags, OPT_DONTOUCH))
-		ast_set_flag(cdr, AST_CDR_FLAG_DONT_TOUCH);
-		
 	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;
-	char *argcopy = NULL;
-	struct ast_flags flags = {0};
-	char *opts[OPT_ARG_ARRAY_SIZE];
-	AST_DECLARE_APP_ARGS(arglist,
-		AST_APP_ARG(options);
-	);
+	struct localuser *u;
 
 	if (!chan->cdr) {
 		ast_log(LOG_WARNING, "Channel does not have a CDR\n");
 		return 0;
 	}
 
-	argcopy = ast_strdupa(data);
+	LOCAL_USER_ADD(u);
 
-	AST_STANDARD_APP_ARGS(arglist, argcopy);
-
-	opts[OPT_ARG_VARSET] = 0;
-
-	if (!ast_strlen_zero(arglist.options))
-		ast_app_parse_options(forkcdr_exec_options, &flags, opts, arglist.options);
-	
 	if (!ast_strlen_zero(data))
-		ast_set2_flag(chan->cdr, ast_test_flag(&flags, OPT_KEEPVARS), AST_CDR_FLAG_KEEP_VARS);
+		ast_set2_flag(chan->cdr, strchr(data, 'v'), AST_CDR_FLAG_KEEP_VARS);
 	
-	ast_cdr_fork(chan, flags, opts[OPT_ARG_VARSET]);
+	ast_cdr_fork(chan);
 
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, forkcdr_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Fork The CDR into 2 separate entities");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_getcpeid.c

@@ -19,31 +19,41 @@
 /*! \file
  *
  * \brief Get ADSI CPE ID
- *
- * \author Mark Spencer <markster@digium.com>
  * 
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <unistd.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/adsi.h"
+#include "asterisk/options.h"
+
+static char *tdesc = "Get ADSI CPE ID";
 
 static char *app = "GetCPEID";
 
 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 dahdi.conf for on-hook operations.\n";
+"  GetCPEID: Obtains and displays ADSI CPE ID and other information in order\n"
+"to properly setup zapata.conf for on-hook operations.\n";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int cpeid_setstatus(struct ast_channel *chan, char *stuff[], int voice)
 {
@@ -53,54 +63,60 @@ static int cpeid_setstatus(struct ast_channel *chan, char *stuff[], int voice)
 	for (x=0;x<4;x++)
 		tmp[x] = stuff[x];
 	tmp[4] = NULL;
-	return ast_adsi_print(chan, tmp, justify, voice);
+	return adsi_print(chan, tmp, justify, voice);
 }
 
 static int cpeid_exec(struct ast_channel *chan, void *idata)
 {
 	int res=0;
+	struct localuser *u;
 	unsigned char cpeid[4];
 	int gotgeometry = 0;
 	int gotcpeid = 0;
 	int width, height, buttons;
-	char *data[4];
-	unsigned int x;
+	char data[4][80];
+	char *stuff[4];
 
-	for (x = 0; x < 4; x++)
-		data[x] = alloca(80);
-
-	strcpy(data[0], "** CPE Info **");
-	strcpy(data[1], "Identifying CPE...");
-	strcpy(data[2], "Please wait...");
-	res = ast_adsi_load_session(chan, NULL, 0, 1);
+	LOCAL_USER_ADD(u);
+	stuff[0] = data[0];
+	stuff[1] = data[1];
+	stuff[2] = data[2];
+	stuff[3] = data[3];
+	memset(data, 0, sizeof(data));
+	strncpy(stuff[0], "** CPE Info **", sizeof(data[0]) - 1);
+	strncpy(stuff[1], "Identifying CPE...", sizeof(data[1]) - 1);
+	strncpy(stuff[2], "Please wait...", sizeof(data[2]) - 1);
+	res = adsi_load_session(chan, NULL, 0, 1);
 	if (res > 0) {
-		cpeid_setstatus(chan, data, 0);
-		res = ast_adsi_get_cpeid(chan, cpeid, 0);
+		cpeid_setstatus(chan, stuff, 0);
+		res = adsi_get_cpeid(chan, cpeid, 0);
 		if (res > 0) {
 			gotcpeid = 1;
-			ast_verb(3, "Got CPEID of '%02x:%02x:%02x:%02x' on '%s'\n", cpeid[0], cpeid[1], cpeid[2], cpeid[3], chan->name);
+			if (option_verbose > 2)
+				ast_verbose(VERBOSE_PREFIX_3 "Got CPEID of '%02x:%02x:%02x:%02x' on '%s'\n", cpeid[0], cpeid[1], cpeid[2], cpeid[3], chan->name);
 		}
 		if (res > -1) {
-			strcpy(data[1], "Measuring CPE...");
-			strcpy(data[2], "Please wait...");
-			cpeid_setstatus(chan, data, 0);
-			res = ast_adsi_get_cpeinfo(chan, &width, &height, &buttons, 0);
+			strncpy(stuff[1], "Measuring CPE...", sizeof(data[1]) - 1);
+			strncpy(stuff[2], "Please wait...", sizeof(data[2]) - 1);
+			cpeid_setstatus(chan, stuff, 0);
+			res = adsi_get_cpeinfo(chan, &width, &height, &buttons, 0);
 			if (res > -1) {
-				ast_verb(3, "CPE has %d lines, %d columns, and %d buttons on '%s'\n", height, width, buttons, chan->name);
+				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "CPE has %d lines, %d columns, and %d buttons on '%s'\n", height, width, buttons, chan->name);
 				gotgeometry = 1;
 			}
 		}
 		if (res > -1) {
 			if (gotcpeid)
-				snprintf(data[1], 80, "CPEID: %02x:%02x:%02x:%02x", cpeid[0], cpeid[1], cpeid[2], cpeid[3]);
+				snprintf(stuff[1], sizeof(data[1]), "CPEID: %02x:%02x:%02x:%02x", cpeid[0], cpeid[1], cpeid[2], cpeid[3]);
 			else
-				strcpy(data[1], "CPEID Unknown");
+				strncpy(stuff[1], "CPEID Unknown", sizeof(data[1]) - 1);
 			if (gotgeometry) 
-				snprintf(data[2], 80, "Geom: %dx%d, %d buttons", width, height, buttons);
+				snprintf(stuff[2], sizeof(data[2]), "Geom: %dx%d, %d buttons", width, height, buttons);
 			else
-				strcpy(data[2], "Geometry unknown");
-			strcpy(data[3], "Press # to exit");
-			cpeid_setstatus(chan, data, 1);
+				strncpy(stuff[2], "Geometry unknown", sizeof(data[2]) - 1);
+			strncpy(stuff[3], "Press # to exit", sizeof(data[3]) - 1);
+			cpeid_setstatus(chan, stuff, 1);
 			for(;;) {
 				res = ast_waitfordigit(chan, 1000);
 				if (res < 0)
@@ -110,21 +126,42 @@ static int cpeid_exec(struct ast_channel *chan, void *idata)
 					break;
 				}
 			}
-			ast_adsi_unload_session(chan);
+			adsi_unload_session(chan);
 		}
 	}
-
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, cpeid_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Get ADSI CPE ID");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_groupcount.c

@@ -0,0 +1,338 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Group Manipulation Applications
+ *
+ * \ingroup applications
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <sys/types.h>
+#include <regex.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/utils.h"
+#include "asterisk/cli.h"
+#include "asterisk/app.h"
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int group_count_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	int count;
+	struct localuser *u;
+	char group[80] = "";
+	char category[80] = "";
+	char ret[80] = "";
+	char *grp;
+	static int deprecation_warning = 0;
+
+	LOCAL_USER_ADD(u);
+
+	if (!deprecation_warning) {
+	        ast_log(LOG_WARNING, "The GetGroupCount application has been deprecated, please use the GROUP_COUNT function.\n");
+		deprecation_warning = 1;
+	}
+
+	ast_app_group_split_group(data, group, sizeof(group), category, sizeof(category));
+
+	if (ast_strlen_zero(group)) {
+		grp = pbx_builtin_getvar_helper(chan, category);
+		if (!ast_strlen_zero(grp))
+			ast_copy_string(group, grp, sizeof(group));
+	}
+
+	count = ast_app_group_get_count(group, category);
+	snprintf(ret, sizeof(ret), "%d", count);
+	pbx_builtin_setvar_helper(chan, "GROUPCOUNT", ret);
+
+	LOCAL_USER_REMOVE(u);
+
+	return res;
+}
+
+static int group_match_count_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	int count;
+	struct localuser *u;
+	char group[80] = "";
+	char category[80] = "";
+	char ret[80] = "";
+	static int deprecation_warning = 0;
+
+	LOCAL_USER_ADD(u);
+
+	if (!deprecation_warning) {
+	        ast_log(LOG_WARNING, "The GetGroupMatchCount application has been deprecated, please use the GROUP_MATCH_COUNT function.\n");
+		deprecation_warning = 1;
+	}
+
+	ast_app_group_split_group(data, group, sizeof(group), category, sizeof(category));
+
+	if (!ast_strlen_zero(group)) {
+		count = ast_app_group_match_get_count(group, category);
+		snprintf(ret, sizeof(ret), "%d", count);
+		pbx_builtin_setvar_helper(chan, "GROUPCOUNT", ret);
+	}
+
+	LOCAL_USER_REMOVE(u);
+
+	return res;
+}
+
+static int group_set_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u;
+	static int deprecation_warning = 0;
+
+	LOCAL_USER_ADD(u);
+	
+	if (!deprecation_warning) {
+	        ast_log(LOG_WARNING, "The SetGroup application has been deprecated, please use the GROUP() function.\n");
+		deprecation_warning = 1;
+	}
+
+	if (ast_app_group_set_channel(chan, data))
+		ast_log(LOG_WARNING, "SetGroup requires an argument (group name)\n");
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static int group_check_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	int max, count;
+	struct localuser *u;
+	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][|options])\n");
+		return 0;
+	}
+
+	if (!(parse = ast_strdupa(data))) {
+		ast_log(LOG_WARNING, "Memory Error!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
+	if (ast_strlen_zero(args.max)) {
+		ast_log(LOG_WARNING, "CheckGroup requires an argument(max[@category][|options])\n");
+		return res;
+	}
+
+  	ast_app_group_split_group(args.max, limit, sizeof(limit), category, sizeof(category));
+
+ 	if ((sscanf(limit, "%d", &max) == 1) && (max > -1)) {
+		count = ast_app_group_get_count(pbx_builtin_getvar_helper(chan, category), category);
+		if (count > max) {
+			pbx_builtin_setvar_helper(chan, "CHECKGROUPSTATUS", "OVERMAX");
+			if (priority_jump || option_priority_jumping) {
+				if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
+					res = -1;
+			}
+		} else
+			pbx_builtin_setvar_helper(chan, "CHECKGROUPSTATUS", "OK");
+	} else
+		ast_log(LOG_WARNING, "CheckGroup requires a positive integer argument (max)\n");
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static int group_show_channels(int fd, int argc, char *argv[])
+{
+#define FORMAT_STRING  "%-25s  %-20s  %-20s\n"
+
+	int numchans = 0;
+	struct ast_group_info *gi = NULL;
+	regex_t regexbuf;
+	int havepattern = 0;
+
+	if (argc < 3 || argc > 4)
+		return RESULT_SHOWUSAGE;
+	
+	if (argc == 4) {
+		if (regcomp(&regexbuf, argv[3], REG_EXTENDED | REG_NOSUB))
+			return RESULT_SHOWUSAGE;
+		havepattern = 1;
+	}
+
+	ast_cli(fd, FORMAT_STRING, "Channel", "Group", "Category");
+
+	ast_app_group_list_lock();
+
+	gi = ast_app_group_list_head();
+	while (gi) {
+		if (!havepattern || !regexec(&regexbuf, gi->group, 0, NULL, 0)) {
+			ast_cli(fd, FORMAT_STRING, gi->chan->name, gi->group, (ast_strlen_zero(gi->category) ? "(default)" : gi->category));
+			numchans++;
+		}
+		gi = AST_LIST_NEXT(gi, list);
+	}
+
+	ast_app_group_list_unlock();
+
+	if (havepattern)
+		regfree(&regexbuf);
+
+	ast_cli(fd, "%d active channel%s\n", numchans, (numchans != 1) ? "s" : "");
+	return RESULT_SUCCESS;
+#undef FORMAT_STRING
+}
+
+static char *tdesc = "Group Management Routines";
+
+static char *app_group_count = "GetGroupCount";
+static char *app_group_set = "SetGroup";
+static char *app_group_check = "CheckGroup";
+static char *app_group_match_count = "GetGroupMatchCount";
+
+static char *group_count_synopsis = "Get the channel count of a group";
+static char *group_set_synopsis = "Set the channel's group";
+static char *group_check_synopsis = "Check the channel count of a group against a limit";
+static char *group_match_count_synopsis = "Get the channel count of all groups that match a pattern";
+
+static char *group_count_descrip =
+"Usage: GetGroupCount([groupname][@category])\n"
+"  Calculates the group count for the specified group, or uses\n"
+"the current channel's group if not specifed (and non-empty).\n"
+"Stores result in GROUPCOUNT. \n"
+"Note: This application has been deprecated, please use the function\n"
+"GROUP_COUNT.\n";
+
+static char *group_set_descrip =
+"Usage: SetGroup(groupname[@category])\n"
+"  Sets the channel group to the specified value.  Equivalent to\n"
+"Set(GROUP=group).  Always returns 0.\n";
+
+static char *group_check_descrip =
+"Usage: CheckGroup(max[@category][|options])\n"
+"  Checks that the current number of total channels in the\n"
+"current channel's group does not exceed 'max'.  If the number\n"
+"does not exceed 'max', we continue to the next step. \n"
+" The option string may contain zero of the following character:\n"
+"	'j' -- jump to n+101 priority if the number does in fact exceed max,\n"
+"              and priority n+101 exists. Execuation then continues at that\n"
+"	       step, otherwise -1 is returned.\n"
+" This application sets the following channel variable upon successful completion:\n"
+"	CHECKGROUPSTATUS  The status of the check that the current channel's\n"
+"			  group does not exceed 'max'. It's value is one of\n"
+"		OK | OVERMAX \n";	
+
+static char *group_match_count_descrip =
+"Usage: GetGroupMatchCount(groupmatch[@category])\n"
+"  Calculates the group count for all groups that match the specified\n"
+"pattern. Uses standard regular expression matching (see regex(7)).\n"
+"Stores result in GROUPCOUNT.  Always returns 0.\n"
+"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"
+"       Lists all currently active channels with channel group(s) specified.\n       Optional regular expression pattern is matched to group names for each channel.\n";
+
+static struct ast_cli_entry  cli_show_channels =
+	{ { "group", "show", "channels", NULL }, group_show_channels, "Show active channels with group(s)", show_channels_usage};
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_cli_unregister(&cli_show_channels);
+	res |= ast_unregister_application(app_group_count);
+	res |= ast_unregister_application(app_group_set);
+	res |= ast_unregister_application(app_group_check);
+	res |= ast_unregister_application(app_group_match_count);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+
+	res = ast_register_application(app_group_count, group_count_exec, group_count_synopsis, group_count_descrip);
+	res |= ast_register_application(app_group_set, group_set_exec, group_set_synopsis, group_set_descrip);
+	res |= ast_register_application(app_group_check, group_check_exec, group_check_synopsis, group_check_descrip);
+	res |= ast_register_application(app_group_match_count, group_match_count_exec, group_match_count_synopsis, group_match_count_descrip);
+	res |= ast_cli_register(&cli_show_channels);
+	
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_hasnewvoicemail.c

@@ -0,0 +1,363 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Changes Copyright (c) 2004 - 2005 Todd Freeman <freeman@andrews.edu>
+ * 
+ * 95% based on HasNewVoicemail by:
+ * 
+ * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
+ * 
+ * Tilghman Lesher <asterisk-hasnewvoicemail-app@the-tilghman.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief HasVoicemail application
+ *
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <dirent.h>
+#include <sys/types.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/utils.h"
+#include "asterisk/app.h"
+#include "asterisk/options.h"
+#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 with the right options set";
+static char *hasvoicemail_descrip =
+"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"
+"  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 with the right options set";
+static char *hasnewvoicemail_descrip =
+"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;
+
+#ifdef USE_ODBC_STORAGE
+static int hasvoicemail_internal(const char *context, const char *mailbox, const char *folder)
+{
+	int nummsgs = 0;
+	int res;
+	SQLHSTMT stmt;
+	char sql[256];
+	char rowdata[20];
+
+	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;
+		}
+		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 *input, *varname = NULL, *vmbox, *context = "default";
+	char *vmfolder;
+	int vmcount = 0;
+	static int dep_warning = 0;
+	int priority_jump = 0;
+	char tmp[12];
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(vmbox);
+		AST_APP_ARG(varname);
+		AST_APP_ARG(options);
+	);
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "The applications HasVoicemail and HasNewVoicemail have been deprecated.  Please use the VMCOUNT() function instead.\n");
+		dep_warning = 1;
+	}
+	
+	if (!data) {
+		ast_log(LOG_WARNING, "HasVoicemail requires an argument (vm-box[/folder][@context][|varname[|options]])\n");
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	input = ast_strdupa((char *)data);
+	if (! input) {
+		ast_log(LOG_ERROR, "Out of memory error\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args, input);
+
+	if ((vmbox = strsep(&args.vmbox, "@")))
+		if (!ast_strlen_zero(args.vmbox))
+			context = args.vmbox;
+	if (!vmbox)
+		vmbox = args.vmbox;
+
+	vmfolder = strchr(vmbox, '/');
+	if (vmfolder) {
+		*vmfolder = '\0';
+		vmfolder++;
+	} else {
+		vmfolder = "INBOX";
+	}
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
+	vmcount = hasvoicemail_internal(context, vmbox, vmfolder);
+	/* Set the count in the channel variable */
+	if (varname) {
+		snprintf(tmp, sizeof(tmp), "%d", vmcount);
+		pbx_builtin_setvar_helper(chan, varname, tmp);
+	}
+
+	if (vmcount > 0) {
+		/* Branch to the next extension */
+		if (priority_jump || option_priority_jumping) {
+			if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) 
+				ast_log(LOG_WARNING, "VM box %s@%s has new voicemail, but extension %s, priority %d doesn't exist\n", vmbox, context, chan->exten, chan->priority + 101);
+		}
+	}
+
+	snprintf(tmp, sizeof(tmp), "%d", vmcount);
+	pbx_builtin_setvar_helper(chan, "HASVMSTATUS", tmp);
+	
+	LOCAL_USER_REMOVE(u);
+
+	return 0;
+}
+
+static char *acf_vmcount_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
+	struct localuser *u;
+	char *args, *context, *box, *folder;
+
+	LOCAL_USER_ACF_ADD(u);
+
+	buf[0] = '\0';
+
+	args = ast_strdupa(data);
+	if (!args) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return buf;
+	}
+
+	box = strsep(&args, "|");
+	if (strchr(box, '@')) {
+		context = box;
+		box = strsep(&context, "@");
+	} else {
+		context = "default";
+	}
+
+	if (args) {
+		folder = args;
+	} else {
+		folder = "INBOX";
+	}
+
+	snprintf(buf, len, "%d", hasvoicemail_internal(context, box, folder));
+
+	LOCAL_USER_REMOVE(u);
+	
+	return buf;
+}
+
+struct ast_custom_function acf_vmcount = {
+	.name = "VMCOUNT",
+	.synopsis = "Counts the voicemail in a specified mailbox",
+	.syntax = "VMCOUNT(vmbox[@context][|folder])",
+	.desc =
+	"  context - defaults to \"default\"\n"
+	"  folder  - defaults to \"INBOX\"\n",
+	.read = acf_vmcount_exec,
+};
+
+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;
+	
+	res = ast_custom_function_unregister(&acf_vmcount);
+	res |= ast_unregister_application(app_hasvoicemail);
+	res |= ast_unregister_application(app_hasnewvoicemail);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+	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);
+
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_ices.c

@@ -19,40 +19,37 @@
 /*! \file
  *
  * \brief Stream to an icecast server via ICES (see contrib/asterisk-ices.xml)
- *
- * \author Mark Spencer <markster@digium.com>
  * 
- * \extref ICES - http://www.icecast.org/ices.php
- *
  * \ingroup applications
  */
  
-/*** MODULEINFO
-	<depend>working_fork</depend>
- ***/
+#include <string.h>
+#include <stdio.h>
+#include <signal.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/time.h>
+#include <errno.h>
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include <signal.h>
-#include <fcntl.h>
-#include <sys/time.h>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
-
-#include "asterisk/paths.h"	/* use ast_config_AST_CONFIG_DIR */
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/frame.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/translate.h"
+#include "asterisk/options.h"
 
-#define path_BIN "/usr/bin/"
-#define path_LOCAL "/usr/local/bin/"
+#define ICES "/usr/bin/ices"
+#define LOCAL_ICES "/usr/local/bin/ices"
+
+static char *tdesc = "Encode and Stream via icecast and ices";
 
 static char *app = "ICES";
 
@@ -61,19 +58,17 @@ 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 contrib/asterisk-ices.xml). \n"
-"\n"
-"- ICES version 2 cient and server required.\n";
+"for ices (see examples/asterisk-ices.conf). \n";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int icesencode(char *filename, int fd)
 {
 	int res;
 	int x;
 	sigset_t fullset, oldset;
-#ifdef HAVE_CAP
-	cap_t cap;
-#endif
 
 	sigfillset(&fullset);
 	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
@@ -90,46 +85,27 @@ static int icesencode(char *filename, int fd)
 	signal(SIGPIPE, SIG_DFL);
 	pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
 
-#ifdef HAVE_CAP
-	cap = cap_from_text("cap_net_admin-eip");
-
-	if (cap_set_proc(cap)) {
-		/* Careful with order! Logging cannot happen after we close FDs */
-		ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-	}
-	cap_free(cap);
-#endif
-
-	if (ast_opt_high_priority)
+	if (option_highpriority)
 		ast_set_priority(0);
 	dup2(fd, STDIN_FILENO);
 	for (x=STDERR_FILENO + 1;x<1024;x++) {
 		if ((x != STDIN_FILENO) && (x != STDOUT_FILENO))
 			close(x);
 	}
-
-	/* Most commonly installed in /usr/local/bin 
-	 * But many places has it in /usr/bin 
-	 * As a last-ditch effort, try to use PATH
-	 */
-	execl(path_LOCAL "ices2", "ices", filename, (char *)NULL);
-	execl(path_BIN "ices2", "ices", filename, (char *)NULL);
-	execlp("ices2", "ices", filename, (char *)NULL);
-
-	ast_debug(1, "Couldn't find ices version 2, attempting to use ices version 1.");
-
-	execl(path_LOCAL "ices", "ices", filename, (char *)NULL);
-	execl(path_BIN "ices", "ices", filename, (char *)NULL);
+	/* Most commonly installed in /usr/local/bin */
+	execl(ICES, "ices", filename, (char *)NULL);
+	/* But many places has it in /usr/bin */
+	execl(LOCAL_ICES, "ices", filename, (char *)NULL);
+	/* As a last-ditch effort, try to use PATH */
 	execlp("ices", "ices", filename, (char *)NULL);
-
-	ast_log(LOG_WARNING, "Execute of ices failed, could not find command.\n");
-	close(fd);
+	ast_log(LOG_WARNING, "Execute of ices failed\n");
 	_exit(0);
 }
 
 static int ices_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
+	int res=0;
+	struct localuser *u;
 	int fds[2];
 	int ms = -1;
 	int pid = -1;
@@ -144,11 +120,14 @@ static int ices_exec(struct ast_channel *chan, void *data)
 		ast_log(LOG_WARNING, "ICES requires an argument (configfile.xml)\n");
 		return -1;
 	}
+
+	LOCAL_USER_ADD(u);
 	
 	last = ast_tv(0, 0);
 	
 	if (pipe(fds)) {
 		ast_log(LOG_WARNING, "Unable to create pipe\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	flags = fcntl(fds[1], F_GETFL);
@@ -163,6 +142,7 @@ static int ices_exec(struct ast_channel *chan, void *data)
 		close(fds[0]);
 		close(fds[1]);
 		ast_log(LOG_WARNING, "Answer failed!\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 
@@ -172,30 +152,32 @@ static int ices_exec(struct ast_channel *chan, void *data)
 		close(fds[0]);
 		close(fds[1]);
 		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	if (((char *)data)[0] == '/')
-		ast_copy_string(filename, (char *) data, sizeof(filename));
+		strncpy(filename, (char *)data, sizeof(filename) - 1);
 	else
-		snprintf(filename, sizeof(filename), "%s/%s", ast_config_AST_CONFIG_DIR, (char *)data);
+		snprintf(filename, sizeof(filename), "%s/%s", (char *)ast_config_AST_CONFIG_DIR, (char *)data);
 	/* Placeholder for options */		
 	c = strchr(filename, '|');
 	if (c)
 		*c = '\0';	
 	res = icesencode(filename, fds[0]);
+	close(fds[0]);
 	if (res >= 0) {
 		pid = res;
 		for (;;) {
 			/* Wait for audio, and stream */
 			ms = ast_waitfor(chan, -1);
 			if (ms < 0) {
-				ast_debug(1, "Hangup detected\n");
+				ast_log(LOG_DEBUG, "Hangup detected\n");
 				res = -1;
 				break;
 			}
 			f = ast_read(chan);
 			if (!f) {
-				ast_debug(1, "Null frame == hangup() detected\n");
+				ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
 				res = -1;
 				break;
 			}
@@ -213,7 +195,6 @@ static int ices_exec(struct ast_channel *chan, void *data)
 			ast_frfree(f);
 		}
 	}
-	close(fds[0]);
 	close(fds[1]);
 	
 	if (pid > -1)
@@ -221,17 +202,40 @@ static int ices_exec(struct ast_channel *chan, void *data)
 	if (!res && oreadformat)
 		ast_set_read_format(chan, oreadformat);
 
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, ices_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Encode and Stream via icecast and ices");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_image.c

@@ -19,62 +19,129 @@
 /*! \file
  *
  * \brief App to transmit an image
- *
- * \author Mark Spencer <markster@digium.com>
  * 
  * \ingroup applications
  */
  
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
+#include "asterisk/translate.h"
 #include "asterisk/image.h"
+#include "asterisk/app.h"
+#include "asterisk/options.h"
+
+static char *tdesc = "Image Transmission Application";
 
 static char *app = "SendImage";
 
 static char *synopsis = "Send an image file";
 
 static char *descrip = 
-"  SendImage(filename): Sends an image on a channel.\n"
-"If the channel supports image transport but the image send fails, the channel\n"
-"will be hung up.  Otherwise, the dialplan continues execution.  This\n"
-"application sets the following channel variable upon completion:\n"
-"   SENDIMAGESTATUS  The status is the result of the attempt, one of:\n"
-"                    OK | NOSUPPORT \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;
+
+LOCAL_USER_DECL;
 
 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);
+	);
+	
+	LOCAL_USER_ADD(u);
 
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "SendImage requires an argument (filename)\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 (ast_strlen_zero(args.filename)) {
+		ast_log(LOG_WARNING, "SendImage requires an argument (filename[|options])\n");
+		return -1;
+	}
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
 	if (!ast_supports_images(chan)) {
 		/* Does not support transport */
+		if (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;
 	}
 
-	if (!(res = ast_send_image(chan, data)))
+	res = ast_send_image(chan, args.filename);
+	
+	if (!res)
 		pbx_builtin_setvar_helper(chan, "SENDIMAGESTATUS", "OK");
-		
+	
+	LOCAL_USER_REMOVE(u);
+	
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res; 
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, sendimage_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Image Transmission Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_intercom.c

@@ -0,0 +1,234 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Use /dev/dsp as an intercom.
+ * 
+ * \ingroup applications
+ */
+ 
+#include <stdio.h>
+#include <unistd.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+#include <string.h>
+#include <stdlib.h>
+#include <sys/time.h>
+#include <netinet/in.h>
+
+#if defined(__linux__)
+#include <linux/soundcard.h>
+#elif defined(__FreeBSD__)
+#include <sys/soundcard.h>
+#else
+#include <soundcard.h>
+#endif
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/frame.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+
+#ifdef __OpenBSD__
+#define DEV_DSP "/dev/audio"
+#else
+#define DEV_DSP "/dev/dsp"
+#endif
+
+/* Number of 32 byte buffers -- each buffer is 2 ms */
+#define BUFFER_SIZE 32
+
+static char *tdesc = "Intercom using /dev/dsp for output";
+
+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.  User can terminate\n"with a DTMF tone, or by hangup.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+AST_MUTEX_DEFINE_STATIC(sound_lock);
+static int sound = -1;
+
+static int write_audio(short *data, int len)
+{
+	int res;
+	struct audio_buf_info info;
+	ast_mutex_lock(&sound_lock);
+	if (sound < 0) {
+		ast_log(LOG_WARNING, "Sound device closed?\n");
+		ast_mutex_unlock(&sound_lock);
+		return -1;
+	}
+    if (ioctl(sound, SNDCTL_DSP_GETOSPACE, &info)) {
+		ast_log(LOG_WARNING, "Unable to read output space\n");
+		ast_mutex_unlock(&sound_lock);
+        return -1;
+    }
+	res = write(sound, data, len);
+	ast_mutex_unlock(&sound_lock);
+	return res;
+}
+
+static int create_audio(void)
+{
+	int fmt, desired, res, fd;
+	fd = open(DEV_DSP, O_WRONLY);
+	if (fd < 0) {
+		ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
+		close(fd);
+		return -1;
+	}
+	fmt = AFMT_S16_LE;
+	res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
+		close(fd);
+		return -1;
+	}
+	fmt = 0;
+	res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+		close(fd);
+		return -1;
+	}
+	/* 8000 Hz desired */
+	desired = 8000;
+	fmt = desired;
+	res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+		close(fd);
+		return -1;
+	}
+	if (fmt != desired) {
+		ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
+	}
+#if 1
+	/* 2 bytes * 15 units of 2^5 = 32 bytes per buffer */
+	fmt = ((BUFFER_SIZE) << 16) | (0x0005);
+	res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
+	}
+#endif
+	sound = fd;
+	return 0;
+}
+
+static int intercom_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u;
+	struct ast_frame *f;
+	int oreadformat;
+	LOCAL_USER_ADD(u);
+	/* Remember original read format */
+	oreadformat = chan->readformat;
+	/* Set mode to signed linear */
+	res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set format to signed linear on channel %s\n", chan->name);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	/* Read packets from the channel */
+	while(!res) {
+		res = ast_waitfor(chan, -1);
+		if (res > 0) {
+			res = 0;
+			f = ast_read(chan);
+			if (f) {
+				if (f->frametype == AST_FRAME_DTMF) {
+					ast_frfree(f);
+					break;
+				} else {
+					if (f->frametype == AST_FRAME_VOICE) {
+						if (f->subclass == AST_FORMAT_SLINEAR) {
+							res = write_audio(f->data, f->datalen);
+							if (res > 0)
+								res = 0;
+						} else
+							ast_log(LOG_DEBUG, "Unable to handle non-signed linear frame (%d)\n", f->subclass);
+					} 
+				}
+				ast_frfree(f);
+			} else
+				res = -1;
+		}
+	}
+	
+	if (!res)
+		ast_set_read_format(chan, oreadformat);
+
+	LOCAL_USER_REMOVE(u);
+
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	if (sound > -1)
+		close(sound);
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	if (create_audio())
+		return -1;
+	return ast_register_application(app, intercom_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_ivrdemo.c

@@ -19,21 +19,21 @@
 /*! \file
  *
  * \brief IVR Demo application
- *
- * \author Mark Spencer <markster@digium.com>
  * 
  * \ingroup applications
  */
 
-/*** MODULEINFO
-	<defaultenabled>no</defaultenabled>
- ***/
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
@@ -82,16 +82,22 @@ AST_IVR_DECLARE_MENU(ivr_demo, "IVR Demo Main Menu", 0,
 	{ NULL },
 });
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int skel_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
+	struct localuser *u;
 	
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "skel requires an argument (filename)\n");
 		return -1;
 	}
 	
+	LOCAL_USER_ADD(u);
+
 	/* Do our thing here */
 
 	if (chan->_state != AST_STATE_UP)
@@ -99,17 +105,40 @@ static int skel_exec(struct ast_channel *chan, void *data)
 	if (!res)
 		res = ast_ivr_menu_run(chan, &ivr_demo, data);
 	
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+	
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+	
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, skel_exec, tdesc, synopsis);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "IVR Demo Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_jack.c

@@ -1,976 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2007 - 2008, Russell Bryant
- *
- * Russell Bryant <russell@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief Jack Application
- *
- * \author Russell Bryant <russell@digium.com>
- *
- * This is an application to connect an Asterisk channel to an input
- * and output jack port so that the audio can be processed through
- * another application, or to play audio from another application.
- *
- * \arg http://www.jackaudio.org/
- *
- * \note To install libresample, check it out of the following repository:
- * <code>$ svn co http://svn.digium.com/svn/thirdparty/libresample/trunk</code>
- *
- * \ingroup applications
- */
-
-/*** MODULEINFO
-	<depend>jack</depend>
-	<depend>resample</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <limits.h>
-
-#include <jack/jack.h>
-#include <jack/ringbuffer.h>
-
-#include <libresample.h>
-
-#include "asterisk/module.h"
-#include "asterisk/channel.h"
-#include "asterisk/strings.h"
-#include "asterisk/lock.h"
-#include "asterisk/app.h"
-#include "asterisk/pbx.h"
-#include "asterisk/audiohook.h"
-
-#define RESAMPLE_QUALITY 1
-
-#define RINGBUFFER_SIZE 16384
-
-/*! \brief Common options between the Jack() app and JACK_HOOK() function */
-#define COMMON_OPTIONS \
-"    s(<name>) - Connect to the specified jack server name.\n" \
-"    i(<name>) - Connect the output port that gets created to the specified\n" \
-"                jack input port.\n" \
-"    o(<name>) - Connect the input port that gets created to the specified\n" \
-"                jack output port.\n" \
-"    n         - Do not automatically start the JACK server if it is not already\n" \
-"                running.\n"
-
-static char *jack_app = "JACK";
-static char *jack_synopsis = 
-"JACK (Jack Audio Connection Kit) Application";
-static char *jack_desc = 
-"JACK([options])\n"
-"  When this application is executed, two jack ports will be created; one input\n"
-"and one output.  Other applications can be hooked up to these ports to access\n"
-"the audio coming from, or being sent to the channel.\n"
-"  Valid options:\n"
-COMMON_OPTIONS
-"";
-
-struct jack_data {
-	AST_DECLARE_STRING_FIELDS(
-		AST_STRING_FIELD(server_name);
-		AST_STRING_FIELD(connect_input_port);
-		AST_STRING_FIELD(connect_output_port);
-	);
-	jack_client_t *client;
-	jack_port_t *input_port;
-	jack_port_t *output_port;
-	jack_ringbuffer_t *input_rb;
-	jack_ringbuffer_t *output_rb;
-	void *output_resampler;
-	double output_resample_factor;
-	void *input_resampler;
-	double input_resample_factor;
-	unsigned int stop:1;
-	unsigned int has_audiohook:1;
-	unsigned int no_start_server:1;
-	/*! Only used with JACK_HOOK */
-	struct ast_audiohook audiohook;
-};
-
-static const struct {
-	jack_status_t status;
-	const char *str;
-} jack_status_table[] = {
-	{ JackFailure,        "Failure" },
-	{ JackInvalidOption,  "Invalid Option" },
-	{ JackNameNotUnique,  "Name Not Unique" },
-	{ JackServerStarted,  "Server Started" },
-	{ JackServerFailed,   "Server Failed" },
-	{ JackServerError,    "Server Error" },
-	{ JackNoSuchClient,   "No Such Client" },
-	{ JackLoadFailure,    "Load Failure" },
-	{ JackInitFailure,    "Init Failure" },
-	{ JackShmFailure,     "Shared Memory Access Failure" },
-	{ JackVersionError,   "Version Mismatch" },
-};
-
-static const char *jack_status_to_str(jack_status_t status)
-{
-	int i;
-
-	for (i = 0; i < ARRAY_LEN(jack_status_table); i++) {
-		if (jack_status_table[i].status == status)
-			return jack_status_table[i].str;
-	}
-
-	return "Unknown Error";
-}
-
-static void log_jack_status(const char *prefix, jack_status_t status)
-{
-	struct ast_str *str = ast_str_alloca(512);
-	int i, first = 0;
-
-	for (i = 0; i < (sizeof(status) * 8); i++) {
-		if (!(status & (1 << i)))
-			continue;
-
-		if (!first) {
-			ast_str_set(&str, 0, "%s", jack_status_to_str((1 << i)));
-			first = 1;
-		} else
-			ast_str_append(&str, 0, ", %s", jack_status_to_str((1 << i)));
-	}
-	
-	ast_log(LOG_NOTICE, "%s: %s\n", prefix, str->str);
-}
-
-static int alloc_resampler(struct jack_data *jack_data, int input)
-{
-	double from_srate, to_srate, jack_srate;
-	void **resampler;
-	double *resample_factor;
-
-	if (input && jack_data->input_resampler)
-		return 0;
-
-	if (!input && jack_data->output_resampler)
-		return 0;
-
-	jack_srate = jack_get_sample_rate(jack_data->client);
-
-	/* XXX Hard coded 8 kHz */
-
-	to_srate = input ? 8000.0 : jack_srate; 
-	from_srate = input ? jack_srate : 8000.0;
-
-	resample_factor = input ? &jack_data->input_resample_factor : 
-		&jack_data->output_resample_factor;
-
-	if (from_srate == to_srate) {
-		/* Awesome!  The jack sample rate is the same as ours.
-		 * Resampling isn't needed. */
-		*resample_factor = 1.0;
-		return 0;
-	}
-
-	*resample_factor = to_srate / from_srate;
-
-	resampler = input ? &jack_data->input_resampler :
-		&jack_data->output_resampler;
-
-	if (!(*resampler = resample_open(RESAMPLE_QUALITY, 
-		*resample_factor, *resample_factor))) {
-		ast_log(LOG_ERROR, "Failed to open %s resampler\n", 
-			input ? "input" : "output");
-		return -1;
-	}
-
-	return 0;
-}
-
-/*!
- * \brief Handle jack input port
- *
- * Read nframes number of samples from the input buffer, resample it
- * if necessary, and write it into the appropriate ringbuffer. 
- */
-static void handle_input(void *buf, jack_nframes_t nframes, 
-	struct jack_data *jack_data)
-{
-	short s_buf[nframes];
-	float *in_buf = buf;
-	size_t res;
-	int i;
-	size_t write_len = sizeof(s_buf);
-
-	if (jack_data->input_resampler) {
-		int total_in_buf_used = 0;
-		int total_out_buf_used = 0;
-		float f_buf[nframes + 1];
-
-		memset(f_buf, 0, sizeof(f_buf));
-
-		while (total_in_buf_used < nframes) {
-			int in_buf_used;
-			int out_buf_used;
-
-			out_buf_used = resample_process(jack_data->input_resampler,
-				jack_data->input_resample_factor,
-				&in_buf[total_in_buf_used], nframes - total_in_buf_used,
-				0, &in_buf_used,
-				&f_buf[total_out_buf_used], ARRAY_LEN(f_buf) - total_out_buf_used);
-
-			if (out_buf_used < 0)
-				break;
-
-			total_out_buf_used += out_buf_used;
-			total_in_buf_used += in_buf_used;
-	
-			if (total_out_buf_used == ARRAY_LEN(f_buf)) {
-				ast_log(LOG_ERROR, "Output buffer filled ... need to increase its size, "
-					"nframes '%d', total_out_buf_used '%d'\n", nframes, total_out_buf_used);
-				break;
-			}
-		}
-
-		for (i = 0; i < total_out_buf_used; i++)
-			s_buf[i] = f_buf[i] * (SHRT_MAX / 1.0);
-		
-		write_len = total_out_buf_used * sizeof(int16_t);
-	} else {
-		/* No resampling needed */
-
-		for (i = 0; i < nframes; i++)
-			s_buf[i] = in_buf[i] * (SHRT_MAX / 1.0);
-	}
-
-	res = jack_ringbuffer_write(jack_data->input_rb, (const char *) s_buf, write_len);
-	if (res != write_len) {
-		ast_debug(2, "Tried to write %d bytes to the ringbuffer, but only wrote %d\n",
-			(int) sizeof(s_buf), (int) res);
-	}
-}
-
-/*!
- * \brief Handle jack output port
- *
- * Read nframes number of samples from the ringbuffer and write it out to the
- * output port buffer.
- */
-static void handle_output(void *buf, jack_nframes_t nframes, 
-	struct jack_data *jack_data)
-{
-	size_t res, len;
-
-	len = nframes * sizeof(float);
-
-	res = jack_ringbuffer_read(jack_data->output_rb, buf, len);
-
-	if (len != res) {
-		ast_debug(2, "Wanted %d bytes to send to the output port, "
-			"but only got %d\n", (int) len, (int) res);
-	}
-}
-
-static int jack_process(jack_nframes_t nframes, void *arg)
-{
-	struct jack_data *jack_data = arg;
-	void *input_port_buf, *output_port_buf;
-
-	if (!jack_data->input_resample_factor)
-		alloc_resampler(jack_data, 1);
-
-	input_port_buf = jack_port_get_buffer(jack_data->input_port, nframes);
-	handle_input(input_port_buf, nframes, jack_data);
-
-	output_port_buf = jack_port_get_buffer(jack_data->output_port, nframes);
-	handle_output(output_port_buf, nframes, jack_data);
-
-	return 0;
-}
-
-static void jack_shutdown(void *arg)
-{
-	struct jack_data *jack_data = arg;
-
-	jack_data->stop = 1;
-}
-
-static struct jack_data *destroy_jack_data(struct jack_data *jack_data)
-{
-	if (jack_data->input_port) {
-		jack_port_unregister(jack_data->client, jack_data->input_port);
-		jack_data->input_port = NULL;
-	}
-
-	if (jack_data->output_port) {
-		jack_port_unregister(jack_data->client, jack_data->output_port);
-		jack_data->output_port = NULL;
-	}
-
-	if (jack_data->client) {
-		jack_client_close(jack_data->client);
-		jack_data->client = NULL;
-	}
-
-	if (jack_data->input_rb) {
-		jack_ringbuffer_free(jack_data->input_rb);
-		jack_data->input_rb = NULL;
-	}
-
-	if (jack_data->output_rb) {
-		jack_ringbuffer_free(jack_data->output_rb);
-		jack_data->output_rb = NULL;
-	}
-
-	if (jack_data->output_resampler) {
-		resample_close(jack_data->output_resampler);
-		jack_data->output_resampler = NULL;
-	}
-	
-	if (jack_data->input_resampler) {
-		resample_close(jack_data->input_resampler);
-		jack_data->input_resampler = NULL;
-	}
-
-	if (jack_data->has_audiohook)
-		ast_audiohook_destroy(&jack_data->audiohook);
-
-	ast_string_field_free_memory(jack_data);
-
-	ast_free(jack_data);
-
-	return NULL;
-}
-
-static int init_jack_data(struct ast_channel *chan, struct jack_data *jack_data)
-{
-	const char *chan_name;
-	jack_status_t status = 0;
-	jack_options_t jack_options = JackNullOption;
-
-	ast_channel_lock(chan);
-	chan_name = ast_strdupa(chan->name);
-	ast_channel_unlock(chan);
-
-	if (!(jack_data->output_rb = jack_ringbuffer_create(RINGBUFFER_SIZE)))
-		return -1;
-
-	if (!(jack_data->input_rb = jack_ringbuffer_create(RINGBUFFER_SIZE)))
-		return -1;
-
-	if (jack_data->no_start_server)
-		jack_options |= JackNoStartServer;
-
-	if (!ast_strlen_zero(jack_data->server_name)) {
-		jack_options |= JackServerName;
-		jack_data->client = jack_client_open(chan_name, jack_options, &status,
-			jack_data->server_name);
-	} else {
-		jack_data->client = jack_client_open(chan_name, jack_options, &status);
-	}
-
-	if (status)
-		log_jack_status("Client Open Status", status);
-
-	if (!jack_data->client)
-		return -1;
-
-	jack_data->input_port = jack_port_register(jack_data->client, "input",
-		JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput | JackPortIsTerminal, 0);
-	if (!jack_data->input_port) {
-		ast_log(LOG_ERROR, "Failed to create input port for jack port\n");
-		return -1;
-	}
-
-	jack_data->output_port = jack_port_register(jack_data->client, "output",
-		JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput | JackPortIsTerminal, 0);
-	if (!jack_data->output_port) {
-		ast_log(LOG_ERROR, "Failed to create output port for jack port\n");
-		return -1;
-	}
-
-	if (jack_set_process_callback(jack_data->client, jack_process, jack_data)) {
-		ast_log(LOG_ERROR, "Failed to register process callback with jack client\n");
-		return -1;
-	}
-
-	jack_on_shutdown(jack_data->client, jack_shutdown, jack_data);
-
-	if (jack_activate(jack_data->client)) {
-		ast_log(LOG_ERROR, "Unable to activate jack client\n");
-		return -1;
-	}
-
-	while (!ast_strlen_zero(jack_data->connect_input_port)) {
-		const char **ports;
-		int i;
-
-		ports = jack_get_ports(jack_data->client, jack_data->connect_input_port,
-			NULL, JackPortIsInput);
-
-		if (!ports) {
-			ast_log(LOG_ERROR, "No input port matching '%s' was found\n",
-				jack_data->connect_input_port);
-			break;
-		}
-
-		for (i = 0; ports[i]; i++) {
-			ast_debug(1, "Found port '%s' that matched specified input port '%s'\n",
-				ports[i], jack_data->connect_input_port);
-		}
-
-		if (jack_connect(jack_data->client, jack_port_name(jack_data->output_port), ports[0])) {
-			ast_log(LOG_ERROR, "Failed to connect '%s' to '%s'\n", ports[0],
-				jack_port_name(jack_data->output_port));
-		} else {
-			ast_debug(1, "Connected '%s' to '%s'\n", ports[0],
-				jack_port_name(jack_data->output_port));
-		}
-
-		free((void *) ports);
-
-		break;
-	}
-
-	while (!ast_strlen_zero(jack_data->connect_output_port)) {
-		const char **ports;
-		int i;
-
-		ports = jack_get_ports(jack_data->client, jack_data->connect_output_port,
-			NULL, JackPortIsOutput);
-
-		if (!ports) {
-			ast_log(LOG_ERROR, "No output port matching '%s' was found\n",
-				jack_data->connect_output_port);
-			break;
-		}
-
-		for (i = 0; ports[i]; i++) {
-			ast_debug(1, "Found port '%s' that matched specified output port '%s'\n",
-				ports[i], jack_data->connect_output_port);
-		}
-
-		if (jack_connect(jack_data->client, ports[0], jack_port_name(jack_data->input_port))) {
-			ast_log(LOG_ERROR, "Failed to connect '%s' to '%s'\n", ports[0],
-				jack_port_name(jack_data->input_port));
-		} else {
-			ast_debug(1, "Connected '%s' to '%s'\n", ports[0],
-				jack_port_name(jack_data->input_port));
-		}
-
-		free((void *) ports);
-
-		break;
-	}
-
-	return 0;
-}
-
-static int queue_voice_frame(struct jack_data *jack_data, struct ast_frame *f)
-{
-	float f_buf[f->samples * 8];
-	size_t f_buf_used = 0;
-	int i;
-	int16_t *s_buf = f->data;
-	size_t res;
-
-	memset(f_buf, 0, sizeof(f_buf));
-
-	if (!jack_data->output_resample_factor)
-		alloc_resampler(jack_data, 0);
-
-	if (jack_data->output_resampler) {
-		float in_buf[f->samples];
-		int total_in_buf_used = 0;
-		int total_out_buf_used = 0;
-
-		memset(in_buf, 0, sizeof(in_buf));
-
-		for (i = 0; i < f->samples; i++)
-			in_buf[i] = s_buf[i] * (1.0 / SHRT_MAX);
-
-		while (total_in_buf_used < ARRAY_LEN(in_buf)) {
-			int in_buf_used;
-			int out_buf_used;
-
-			out_buf_used = resample_process(jack_data->output_resampler, 
-				jack_data->output_resample_factor,
-				&in_buf[total_in_buf_used], ARRAY_LEN(in_buf) - total_in_buf_used, 
-				0, &in_buf_used, 
-				&f_buf[total_out_buf_used], ARRAY_LEN(f_buf) - total_out_buf_used);
-
-			if (out_buf_used < 0)
-				break;
-
-			total_out_buf_used += out_buf_used;
-			total_in_buf_used += in_buf_used;
-
-			if (total_out_buf_used == ARRAY_LEN(f_buf)) {
-				ast_log(LOG_ERROR, "Output buffer filled ... need to increase its size\n");
-				break;
-			}
-		}
-
-		f_buf_used = total_out_buf_used;
-		if (f_buf_used > ARRAY_LEN(f_buf))
-			f_buf_used = ARRAY_LEN(f_buf);
-	} else {
-		/* No resampling needed */
-
-		for (i = 0; i < f->samples; i++)
-			f_buf[i] = s_buf[i] * (1.0 / SHRT_MAX);
-
-		f_buf_used = f->samples;
-	}
-
-	res = jack_ringbuffer_write(jack_data->output_rb, (const char *) f_buf, f_buf_used * sizeof(float));
-	if (res != (f_buf_used * sizeof(float))) {
-		ast_debug(2, "Tried to write %d bytes to the ringbuffer, but only wrote %d\n",
-			(int) (f_buf_used * sizeof(float)), (int) res);
-	}
-
-	return 0;
-}
-
-/*!
- * \brief handle jack audio
- *
- * \param[in]  chan The Asterisk channel to write the frames to if no output frame
- *             is provided.
- * \param[in]  jack_data This is the jack_data struct that contains the input
- *             ringbuffer that audio will be read from.
- * \param[out] out_frame If this argument is non-NULL, then assuming there is
- *             enough data avilable in the ringbuffer, the audio in this frame
- *             will get replaced with audio from the input buffer.  If there is
- *             not enough data available to read at this time, then the frame
- *             data gets zeroed out.
- *
- * Read data from the input ringbuffer, which is the properly resampled audio
- * that was read from the jack input port.  Write it to the channel in 20 ms frames,
- * or fill up an output frame instead if one is provided.
- *
- * \return Nothing.
- */
-static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_data,
-	struct ast_frame *out_frame)
-{	
-	short buf[160];
-	struct ast_frame f = {
-		.frametype = AST_FRAME_VOICE,
-		.subclass = AST_FORMAT_SLINEAR,
-		.src = "JACK",
-		.data = buf,
-		.datalen = sizeof(buf),
-		.samples = ARRAY_LEN(buf),
-	};
-
-	for (;;) {
-		size_t res, read_len;
-		char *read_buf;
-
-		read_len = out_frame ? out_frame->datalen : sizeof(buf);
-		read_buf = out_frame ? out_frame->data : buf;
-
-		res = jack_ringbuffer_read_space(jack_data->input_rb);
-
-		if (res < read_len) {
-			/* Not enough data ready for another frame, move on ... */
-			if (out_frame) {
-				ast_debug(1, "Sending an empty frame for the JACK_HOOK\n");
-				memset(out_frame->data, 0, out_frame->datalen);
-			}
-			break;
-		}
-
-		res = jack_ringbuffer_read(jack_data->input_rb, (char *) read_buf, read_len);
-
-		if (res < read_len) {
-			ast_log(LOG_ERROR, "Error reading from ringbuffer, even though it said there was enough data\n");
-			break;
-		}
-
-		if (out_frame) {
-			/* If an output frame was provided, then we just want to fill up the
-			 * buffer in that frame and return. */
-			break;
-		}
-
-		ast_write(chan, &f);
-	}
-}
-
-enum {
-	OPT_SERVER_NAME =    (1 << 0),
-	OPT_INPUT_PORT =     (1 << 1),
-	OPT_OUTPUT_PORT =    (1 << 2),
-	OPT_NOSTART_SERVER = (1 << 3),
-};
-
-enum {
-	OPT_ARG_SERVER_NAME,
-	OPT_ARG_INPUT_PORT,
-	OPT_ARG_OUTPUT_PORT,
-	/* Must be the last element */
-	OPT_ARG_ARRAY_SIZE,
-};
-
-AST_APP_OPTIONS(jack_exec_options, BEGIN_OPTIONS
-	AST_APP_OPTION_ARG('s', OPT_SERVER_NAME, OPT_ARG_SERVER_NAME),
-	AST_APP_OPTION_ARG('i', OPT_INPUT_PORT, OPT_ARG_INPUT_PORT),
-	AST_APP_OPTION_ARG('o', OPT_OUTPUT_PORT, OPT_ARG_OUTPUT_PORT),
-	AST_APP_OPTION('n', OPT_NOSTART_SERVER),
-END_OPTIONS );
-
-static struct jack_data *jack_data_alloc(void)
-{
-	struct jack_data *jack_data;
-
-	if (!(jack_data = ast_calloc(1, sizeof(*jack_data))))
-		return NULL;
-	
-	if (ast_string_field_init(jack_data, 32)) {
-		ast_free(jack_data);
-		return NULL;
-	}
-
-	return jack_data;
-}
-
-/*!
- * \note This must be done before calling init_jack_data().
- */
-static int handle_options(struct jack_data *jack_data, const char *__options_str)
-{
-	struct ast_flags options = { 0, };
-	char *option_args[OPT_ARG_ARRAY_SIZE];
-	char *options_str;
-
-	options_str = ast_strdupa(__options_str);
-
-	ast_app_parse_options(jack_exec_options, &options, option_args, options_str);
-
-	if (ast_test_flag(&options, OPT_SERVER_NAME)) {
-		if (!ast_strlen_zero(option_args[OPT_ARG_SERVER_NAME]))
-			ast_string_field_set(jack_data, server_name, option_args[OPT_ARG_SERVER_NAME]);
-		else {
-			ast_log(LOG_ERROR, "A server name must be provided with the s() option\n");
-			return -1;
-		}
-	}
-
-	if (ast_test_flag(&options, OPT_INPUT_PORT)) {
-		if (!ast_strlen_zero(option_args[OPT_ARG_INPUT_PORT]))
-			ast_string_field_set(jack_data, connect_input_port, option_args[OPT_ARG_INPUT_PORT]);
-		else {
-			ast_log(LOG_ERROR, "A name must be provided with the i() option\n");
-			return -1;
-		}
-	}
-
-	if (ast_test_flag(&options, OPT_OUTPUT_PORT)) {
-		if (!ast_strlen_zero(option_args[OPT_ARG_OUTPUT_PORT]))
-			ast_string_field_set(jack_data, connect_output_port, option_args[OPT_ARG_OUTPUT_PORT]);
-		else {
-			ast_log(LOG_ERROR, "A name must be provided with the o() option\n");
-			return -1;
-		}
-	}
-
-	jack_data->no_start_server = ast_test_flag(&options, OPT_NOSTART_SERVER) ? 1 : 0;
-
-	return 0;
-}
-
-static int jack_exec(struct ast_channel *chan, void *data)
-{
-	struct jack_data *jack_data;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(options);
-	);
-
-	if (!(jack_data = jack_data_alloc()))
-		return -1;
-
-	args.options = data;
-
-	if (!ast_strlen_zero(args.options) && handle_options(jack_data, args.options)) {
-		destroy_jack_data(jack_data);
-		return -1;
-	}
-
-	if (init_jack_data(chan, jack_data)) {
-		destroy_jack_data(jack_data);
-		return -1;
-	}
-
-	if (ast_set_read_format(chan, AST_FORMAT_SLINEAR)) {
-		destroy_jack_data(jack_data);
-		return -1;
-	}
-
-	if (ast_set_write_format(chan, AST_FORMAT_SLINEAR)) {
-		destroy_jack_data(jack_data);
-		return -1;
-	}
-
-	while (!jack_data->stop) {
-		struct ast_frame *f;
-
-		ast_waitfor(chan, -1);
-
-		f = ast_read(chan);
-		if (!f) {
-			jack_data->stop = 1;
-			continue;
-		}
-
-		switch (f->frametype) {
-		case AST_FRAME_CONTROL:
-			if (f->subclass == AST_CONTROL_HANGUP)
-				jack_data->stop = 1;
-			break;
-		case AST_FRAME_VOICE:
-			queue_voice_frame(jack_data, f);
-		default:
-			break;
-		}
-
-		ast_frfree(f);
-
-		handle_jack_audio(chan, jack_data, NULL);
-	}
-
-	jack_data = destroy_jack_data(jack_data);
-
-	return 0;
-}
-
-static void jack_hook_ds_destroy(void *data)
-{
-	struct jack_data *jack_data = data;
-
-	destroy_jack_data(jack_data);
-}
-
-static const struct ast_datastore_info jack_hook_ds_info = {
-	.type = "JACK_HOOK",
-	.destroy = jack_hook_ds_destroy,
-};
-
-static int jack_hook_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, 
-	struct ast_frame *frame, enum ast_audiohook_direction direction)
-{
-	struct ast_datastore *datastore;
-	struct jack_data *jack_data;
-
-	if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
-		return 0;
-
-	if (direction != AST_AUDIOHOOK_DIRECTION_READ)
-		return 0;
-
-	if (frame->frametype != AST_FRAME_VOICE)
-		return 0;
-
-	if (frame->subclass != AST_FORMAT_SLINEAR) {
-		ast_log(LOG_WARNING, "Expected frame in SLINEAR for the audiohook, but got format %d\n",
-			frame->subclass);
-		return 0;
-	}
-
-	ast_channel_lock(chan);
-
-	if (!(datastore = ast_channel_datastore_find(chan, &jack_hook_ds_info, NULL))) {
-		ast_log(LOG_ERROR, "JACK_HOOK datastore not found for '%s'\n", chan->name);
-		ast_channel_unlock(chan);
-		return -1;
-	}
-
-	jack_data = datastore->data;
-
-	queue_voice_frame(jack_data, frame);
-
-	handle_jack_audio(chan, jack_data, frame);
-
-	ast_channel_unlock(chan);
-
-	return 0;
-}
-
-static int enable_jack_hook(struct ast_channel *chan, char *data)
-{
-	struct ast_datastore *datastore;
-	struct jack_data *jack_data = NULL;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(mode);
-		AST_APP_ARG(options);
-	);
-
-	AST_STANDARD_APP_ARGS(args, data);
-
-	ast_channel_lock(chan);
-
-	if ((datastore = ast_channel_datastore_find(chan, &jack_hook_ds_info, NULL))) {
-		ast_log(LOG_ERROR, "JACK_HOOK already enabled for '%s'\n", chan->name);
-		goto return_error;
-	}
-
-	if (ast_strlen_zero(args.mode) || strcasecmp(args.mode, "manipulate")) {
-		ast_log(LOG_ERROR, "'%s' is not a supported mode.  Only manipulate is supported.\n", 
-			S_OR(args.mode, "<none>"));
-		goto return_error;
-	}
-
-	if (!(jack_data = jack_data_alloc()))
-		goto return_error;
-
-	if (!ast_strlen_zero(args.options) && handle_options(jack_data, args.options))
-		goto return_error;
-
-	if (init_jack_data(chan, jack_data))
-		goto return_error;
-
-	if (!(datastore = ast_channel_datastore_alloc(&jack_hook_ds_info, NULL)))
-		goto return_error;
-
-	jack_data->has_audiohook = 1;
-	ast_audiohook_init(&jack_data->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "JACK_HOOK");
-	jack_data->audiohook.manipulate_callback = jack_hook_callback;
-
-	datastore->data = jack_data;
-
-	if (ast_audiohook_attach(chan, &jack_data->audiohook))
-		goto return_error;
-
-	if (ast_channel_datastore_add(chan, datastore))
-		goto return_error;
-
-	ast_channel_unlock(chan);
-
-	return 0;
-
-return_error:
-	ast_channel_unlock(chan);
-
-	if (jack_data)
-		destroy_jack_data(jack_data);
-
-	return -1;
-}
-
-static int disable_jack_hook(struct ast_channel *chan)
-{
-	struct ast_datastore *datastore;
-	struct jack_data *jack_data;
-
-	ast_channel_lock(chan);
-
-	if (!(datastore = ast_channel_datastore_find(chan, &jack_hook_ds_info, NULL))) {
-		ast_channel_unlock(chan);
-		ast_log(LOG_WARNING, "No JACK_HOOK found to disable\n");
-		return -1;
-	}
-
-	ast_channel_datastore_remove(chan, datastore);
-
-	jack_data = datastore->data;
-	ast_audiohook_detach(&jack_data->audiohook);
-
-	/* Keep the channel locked while we destroy the datastore, so that we can
-	 * ensure that all of the jack stuff is stopped just in case another frame
-	 * tries to come through the audiohook callback. */
-	ast_channel_datastore_free(datastore);
-
-	ast_channel_unlock(chan);
-
-	return 0;
-}
-
-static int jack_hook_write(struct ast_channel *chan, const char *cmd, char *data, 
-	const char *value)
-{
-	int res;
-
-	if (!strcasecmp(value, "on"))
-		res = enable_jack_hook(chan, data);
-	else if (!strcasecmp(value, "off"))
-		res = disable_jack_hook(chan);
-	else {
-		ast_log(LOG_ERROR, "'%s' is not a valid value for JACK_HOOK()\n", value);	
-		res = -1;
-	}
-
-	return res;
-}
-
-static struct ast_custom_function jack_hook_function = {
-	.name = "JACK_HOOK",
-	.synopsis = "Enable a jack hook on a channel",
-	.syntax = "JACK_HOOK(<mode>,[options])",
-	.desc =
-	"   The JACK_HOOK allows turning on or off jack connectivity to this channel.\n"
-	"When the JACK_HOOK is turned on, jack ports will get created that allow\n"
-	"access to the audio stream for this channel.  The mode specifies which mode\n"
-	"this hook should run in.  A mode must be specified when turning the JACK_HOOK.\n"
-	"on.  However, all arguments are optional when turning it off.\n"
-	"\n"
-	"   Valid modes are:\n"
-#if 0
-	/* XXX TODO */
-	"    spy -        Create a read-only audio hook.  Only an output jack port will\n"
-	"                 get created.\n"
-	"    whisper -    Create a write-only audio hook.  Only an input jack port will\n"
-	"                 get created.\n"
-#endif
-	"    manipulate - Create a read/write audio hook.  Both an input and an output\n"
-	"                 jack port will get created.  Audio from the channel will be\n"
-	"                 sent out the output port and will be replaced by the audio\n"
-	"                 coming in on the input port as it gets passed on.\n"
-	"\n"
-	"   Valid options are:\n"
-	COMMON_OPTIONS
-	"\n"
-	" Examples:\n"
-	"   To turn on the JACK_HOOK,\n"
-	"     Set(JACK_HOOK(manipulate,i(pure_data_0:input0)o(pure_data_0:output0))=on)\n"
-	"   To turn off the JACK_HOOK,\n"
-	"     Set(JACK_HOOK()=off)\n"
-	"",
-	.write = jack_hook_write,
-};
-
-static int unload_module(void)
-{
-	int res;
-
-	res = ast_unregister_application(jack_app);
-	res |= ast_custom_function_unregister(&jack_hook_function);
-
-	return res;
-}
-
-static int load_module(void)
-{
-	if (ast_register_application(jack_app, jack_exec, jack_synopsis, jack_desc))
-		return AST_MODULE_LOAD_DECLINE;
-
-	if (ast_custom_function_register(&jack_hook_function)) {
-		ast_unregister_application(jack_app);
-		return AST_MODULE_LOAD_DECLINE;
-	}
-
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "JACK Interface");

apps/app_lookupblacklist.c

@@ -0,0 +1,141 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to lookup the callerid number, and see if it is blacklisted
+ *
+ * \ingroup applications
+ * 
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/callerid.h"
+#include "asterisk/astdb.h"
+#include "asterisk/options.h"
+
+static char *tdesc = "Look up Caller*ID name/number from blacklist database";
+
+static char *app = "LookupBlacklist";
+
+static char *synopsis = "Look up Caller*ID name/number from blacklist database";
+
+static char *descrip =
+  "  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;
+
+LOCAL_USER_DECL;
+
+static int
+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 (!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 (option_verbose > 2)
+				ast_log (LOG_NOTICE,"Blacklisted name \"%s\" found\n",chan->cid.cid_name);
+			bl = 1;
+		}
+	}
+
+	if (bl) {
+		if (priority_jump || option_priority_jumping) 
+			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
+		pbx_builtin_setvar_helper(chan, "LOOKUPBLSTATUS", "FOUND");
+	} else
+		pbx_builtin_setvar_helper(chan, "LOOKUPBLSTATUS", "NOTFOUND");	
+
+	LOCAL_USER_REMOVE(u);
+
+	return 0;
+}
+
+int unload_module (void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
+}
+
+int load_module (void)
+{
+	return ast_register_application (app, lookupblacklist_exec, synopsis,descrip);
+}
+
+char *description (void)
+{
+	return tdesc;
+}
+
+int usecount (void)
+{
+	int res;
+	STANDARD_USECOUNT (res);
+	return res;
+}
+
+char *key ()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_lookupcidname.c

@@ -0,0 +1,120 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief App to set callerid name from database, based on directory number
+ * 
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/callerid.h"
+#include "asterisk/astdb.h"
+
+static char *tdesc = "Look up CallerID Name from local database";
+
+static char *app = "LookupCIDName";
+
+static char *synopsis = "Look up CallerID Name from local database";
+
+static char *descrip =
+  "  LookupCIDName: Looks up the Caller*ID number on the active\n"
+  "channel in the Asterisk database (family 'cidname') and sets the\n"
+  "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.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int
+lookupcidname_exec (struct ast_channel *chan, void *data)
+{
+  char dbname[64];
+  struct localuser *u;
+
+  LOCAL_USER_ADD (u);
+  if (chan->cid.cid_num) {
+	if (!ast_db_get ("cidname", chan->cid.cid_num, dbname, sizeof (dbname))) {
+		ast_set_callerid (chan, NULL, dbname, NULL);
+		  if (option_verbose > 2)
+		    ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID name to %s\n",
+				 dbname);
+	}
+  }
+  LOCAL_USER_REMOVE (u);
+  return 0;
+}
+
+int
+unload_module (void)
+{
+	int res;
+
+	res = ast_unregister_application (app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int
+load_module (void)
+{
+  return ast_register_application (app, lookupcidname_exec, synopsis,
+				   descrip);
+}
+
+char *
+description (void)
+{
+  return tdesc;
+}
+
+int
+usecount (void)
+{
+  int res;
+  STANDARD_USECOUNT (res);
+  return res;
+}
+
+char *
+key ()
+{
+  return ASTERISK_GPL_KEY;
+}

apps/app_macro.c

@@ -19,20 +19,26 @@
 /*! \file
  *
  * \brief Dial plan macro Implementation
- *
- * \author Mark Spencer <markster@digium.com>
  * 
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <sys/types.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
+#include "asterisk/options.h"
 #include "asterisk/config.h"
 #include "asterisk/utils.h"
 #include "asterisk/lock.h"
@@ -42,46 +48,36 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 /* special result value used to force macro exit */
 #define MACRO_EXIT_RESULT 1024
 
-#define WAITEXTENWARNING "Use of the application WaitExten within a macro will not function as expected.\n" \
-	"Please use the Read application in order to read DTMF from a channel currently\n" \
-	"executing a macro.\n"
+static char *tdesc = "Extension Macros";
 
 static char *descrip =
-"  Macro(macroname,arg1,arg2...): Executes a macro using the context\n"
+"  Macro(macroname|arg1|arg2...): Executes a macro using the context\n"
 "'macro-<macroname>', jumping to the 's' extension of that context and\n"
 "executing each step, then returning when the steps end. \n"
-"The calling extension, context, and priority are stored in ${MACRO_EXTEN},\n"
+"The calling extension, context, and priority are stored in ${MACRO_EXTEN}, \n"
 "${MACRO_CONTEXT} and ${MACRO_PRIORITY} respectively.  Arguments become\n"
 "${ARG1}, ${ARG2}, etc in the macro context.\n"
 "If you Goto out of the Macro context, the Macro will terminate and control\n"
 "will be returned at the location of the Goto.\n"
 "If ${MACRO_OFFSET} is set at termination, Macro will attempt to continue\n"
 "at priority MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.\n"
+"Extensions: While a macro is being executed, it becomes the current context.\n"
+"            This means that if a hangup occurs, for instance, that the macro\n"
+"            will be searched for an 'h' extension, NOT the context from which\n"
+"            the macro was called. So, make sure to define all appropriate\n"
+"            extensions in your macro! (you can use 'catch' in AEL) \n"
 "WARNING: Because of the way Macro is implemented (it executes the priorities\n"
-"         contained within it via sub-engine), and a fixed per-thread memory\n"
-"         stack allowance, macros are limited to 7 levels of nesting (macro\n"
-"         calling macro calling macro, etc.). It may be possible that\n"
-"         stack-intensive applications in deeply nested macros could cause\n"
-"         Asterisk to crash earlier than this limit. It is advised that if you\n"
-"         need to deeply nest macro calls, that you use the Gosub application\n"
-"         (now allows arguments like a Macro) with explict Return() calls\n"
-"         instead.\n"
-WAITEXTENWARNING;
+"         contained within it via sub-engine), and a fixed per-thread\n"
+"         memory stack allowance, macros are limited to 7 levels\n"
+"         of nesting (macro calling macro calling macro, etc.); It\n"
+"         may be possible that stack-intensive applications in deeply nested\n"
+"         macros could cause asterisk to crash earlier than this limit.\n";
 
 static char *if_descrip =
-"  MacroIf(<expr>?macroname_a[,arg1][:macroname_b[,arg1]])\n"
+"  MacroIf(<expr>?macroname_a[|arg1][:macroname_b[|arg1]])\n"
 "Executes macro defined in <macroname_a> if <expr> is true\n"
 "(otherwise <macroname_b> if provided)\n"
-"Arguments and return values as in application Macro()\n"
-WAITEXTENWARNING;
-
-static char *exclusive_descrip =
-"  MacroExclusive(macroname,arg1,arg2...):\n"
-"Executes macro defined in the context 'macro-macroname'\n"
-"Only one call at a time may run the macro.\n"
-"(we'll wait if another call is busy executing in the Macro)\n"
-"Arguments and return values as in application Macro()\n"
-WAITEXTENWARNING;
+"Arguments and return values as in application macro()\n";
 
 static char *exit_descrip =
 "  MacroExit():\n"
@@ -92,38 +88,15 @@ static char *exit_descrip =
 
 static char *app = "Macro";
 static char *if_app = "MacroIf";
-static char *exclusive_app = "MacroExclusive";
 static char *exit_app = "MacroExit";
 
 static char *synopsis = "Macro Implementation";
 static char *if_synopsis = "Conditional Macro Implementation";
-static char *exclusive_synopsis = "Exclusive Macro Implementation";
 static char *exit_synopsis = "Exit From Macro";
 
-static void macro_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan);
+STANDARD_LOCAL_USER;
 
-struct ast_datastore_info macro_ds_info = {
-	.type = "MACRO",
-	.chan_fixup = macro_fixup,
-};
-
-static void macro_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan)
-{
-	int i;
-	char varname[10];
-	pbx_builtin_setvar_helper(new_chan, "MACRO_DEPTH", "0");
-	pbx_builtin_setvar_helper(new_chan, "MACRO_CONTEXT", NULL);
-	pbx_builtin_setvar_helper(new_chan, "MACRO_EXTEN", NULL);
-	pbx_builtin_setvar_helper(new_chan, "MACRO_PRIORITY", NULL);
-	pbx_builtin_setvar_helper(new_chan, "MACRO_OFFSET", NULL);
-	for (i = 1; i < 100; i++) {
-		snprintf(varname, sizeof(varname), "ARG%d", i);
-		while (pbx_builtin_getvar_helper(new_chan, varname)) {
-			/* Kill all levels of arguments */
-			pbx_builtin_setvar_helper(new_chan, varname, NULL);
-		}
-	}
-}
+LOCAL_USER_DECL;
 
 static struct ast_exten *find_matching_priority(struct ast_context *c, const char *exten, int priority, const char *callerid)
 {
@@ -160,9 +133,8 @@ static struct ast_exten *find_matching_priority(struct ast_context *c, const cha
 	return NULL;
 }
 
-static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
+static int macro_exec(struct ast_channel *chan, void *data)
 {
-	const char *s;
 	char *tmp;
 	char *cur, *rest;
 	char *macro;
@@ -176,53 +148,47 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
 	int oldpriority, gosub_level = 0;
 	char pc[80], depthc[12];
 	char oldcontext[AST_MAX_CONTEXT] = "";
-	const char *inhangupc;
+	char *offsets, *s, *inhangupc;
 	int offset, depth = 0, maxdepth = 7;
 	int setmacrocontext=0;
-	int autoloopflag, inhangup = 0;
+	int autoloopflag, dead = 0, inhangup = 0;
   
 	char *save_macro_exten;
 	char *save_macro_context;
 	char *save_macro_priority;
 	char *save_macro_offset;
-	struct ast_datastore *macro_store = ast_channel_datastore_find(chan, &macro_ds_info, NULL);
+	struct localuser *u;
  
 	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "Macro() requires arguments. See \"core show application macro\" for help.\n");
+		ast_log(LOG_WARNING, "Macro() requires arguments. See \"show application macro\" for help.\n");
 		return -1;
 	}
 
-	do {
-		if (macro_store) {
-			break;
-		}
-		if (!(macro_store = ast_channel_datastore_alloc(&macro_ds_info, NULL))) {
-			ast_log(LOG_WARNING, "Unable to allocate new datastore.\n");
-			break;
-		}
-		/* Just the existence of this datastore is enough. */
-		macro_store->inheritance = DATASTORE_INHERIT_FOREVER;
-		ast_channel_datastore_add(chan, macro_store);
-	} while (0);
+	LOCAL_USER_ADD(u);
 
 	/* does the user want a deeper rabbit hole? */
 	s = pbx_builtin_getvar_helper(chan, "MACRO_RECURSION");
 	if (s)
-		sscanf(s, "%30d", &maxdepth);
+		sscanf(s, "%d", &maxdepth);
 
 	/* Count how many levels deep the rabbit hole goes */
-	s = pbx_builtin_getvar_helper(chan, "MACRO_DEPTH");
-	if (s)
-		sscanf(s, "%30d", &depth);
+	tmp = pbx_builtin_getvar_helper(chan, "MACRO_DEPTH");
+	if (tmp) {
+		sscanf(tmp, "%d", &depth);
+	} else {
+		depth = 0;
+	}
+
 	/* Used for detecting whether to return when a Macro is called from another Macro after hangup */
 	if (strcmp(chan->exten, "h") == 0)
 		pbx_builtin_setvar_helper(chan, "MACRO_IN_HANGUP", "1");
 	inhangupc = pbx_builtin_getvar_helper(chan, "MACRO_IN_HANGUP");
 	if (!ast_strlen_zero(inhangupc))
-		sscanf(inhangupc, "%30d", &inhangup);
+		sscanf(inhangupc, "%d", &inhangup);
 
 	if (depth >= maxdepth) {
 		ast_log(LOG_ERROR, "Macro():  possible infinite loop detected.  Returning early.\n");
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
 	snprintf(depthc, sizeof(depthc), "%d", depth + 1);
@@ -230,32 +196,21 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
 
 	tmp = ast_strdupa(data);
 	rest = tmp;
-	macro = strsep(&rest, ",");
+	macro = strsep(&rest, "|");
 	if (ast_strlen_zero(macro)) {
 		ast_log(LOG_WARNING, "Invalid macro name specified\n");
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
-
 	snprintf(fullmacro, sizeof(fullmacro), "macro-%s", macro);
 	if (!ast_exists_extension(chan, fullmacro, "s", 1, chan->cid.cid_num)) {
-		if (!ast_context_find(fullmacro)) 
+  		if (!ast_context_find(fullmacro)) 
 			ast_log(LOG_WARNING, "No such context '%s' for macro '%s'\n", fullmacro, macro);
 		else
-			ast_log(LOG_WARNING, "Context '%s' for macro '%s' lacks 's' extension, priority 1\n", fullmacro, macro);
+	  		ast_log(LOG_WARNING, "Context '%s' for macro '%s' lacks 's' extension, priority 1\n", fullmacro, macro);
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
-
-	/* If we are to run the macro exclusively, take the mutex */
-	if (exclusive) {
-		ast_debug(1, "Locking macrolock for '%s'\n", fullmacro);
-		ast_autoservice_start(chan);
-		if (ast_context_lockmacro(fullmacro)) {
-			ast_log(LOG_WARNING, "Failed to lock macro '%s' as in-use\n", fullmacro);
-			ast_autoservice_stop(chan);
-			return 0;
-		}
-		ast_autoservice_stop(chan);
-	}
 	
 	/* Save old info */
 	oldpriority = chan->priority;
@@ -269,17 +224,25 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
 	}
 	argc = 1;
 	/* Save old macro variables */
-	save_macro_exten = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_EXTEN"));
+	save_macro_exten = pbx_builtin_getvar_helper(chan, "MACRO_EXTEN");
+	if (save_macro_exten) 
+		save_macro_exten = strdup(save_macro_exten);
 	pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", oldexten);
 
-	save_macro_context = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_CONTEXT"));
+	save_macro_context = pbx_builtin_getvar_helper(chan, "MACRO_CONTEXT");
+	if (save_macro_context)
+		save_macro_context = strdup(save_macro_context);
 	pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", oldcontext);
 
-	save_macro_priority = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_PRIORITY"));
+	save_macro_priority = pbx_builtin_getvar_helper(chan, "MACRO_PRIORITY");
+	if (save_macro_priority) 
+		save_macro_priority = strdup(save_macro_priority);
 	snprintf(pc, sizeof(pc), "%d", oldpriority);
 	pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", pc);
   
-	save_macro_offset = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"));
+	save_macro_offset = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET");
+	if (save_macro_offset) 
+		save_macro_offset = strdup(save_macro_offset);
 	pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", NULL);
 
 	/* Setup environment for new run */
@@ -288,14 +251,13 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
 	ast_copy_string(chan->context, fullmacro, sizeof(chan->context));
 	chan->priority = 1;
 
-	while((cur = strsep(&rest, ",")) && (argc < MAX_ARGS)) {
-		const char *s;
+	while((cur = strsep(&rest, "|")) && (argc < MAX_ARGS)) {
   		/* Save copy of old arguments if we're overwriting some, otherwise
 	   	let them pass through to the other macro */
   		snprintf(varname, sizeof(varname), "ARG%d", argc);
-		s = pbx_builtin_getvar_helper(chan, varname);
-		if (s)
-			oldargs[argc] = ast_strdup(s);
+		oldargs[argc] = pbx_builtin_getvar_helper(chan, varname);
+		if (oldargs[argc])
+			oldargs[argc] = strdup(oldargs[argc]);
 		pbx_builtin_setvar_helper(chan, varname, cur);
 		argc++;
 	}
@@ -304,17 +266,16 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
 	while(ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
 		struct ast_context *c;
 		struct ast_exten *e;
-		int foundx;
 		runningapp[0] = '\0';
 		runningdata[0] = '\0';
 
 		/* What application will execute? */
-		if (ast_rdlock_contexts()) {
+		if (ast_lock_contexts()) {
 			ast_log(LOG_WARNING, "Failed to lock contexts list\n");
 		} else {
 			for (c = ast_walk_contexts(NULL), e = NULL; c; c = ast_walk_contexts(c)) {
 				if (!strcmp(ast_get_context_name(c), chan->context)) {
-					if (ast_rdlock_context(c)) {
+					if (ast_lock_context(c)) {
 						ast_log(LOG_WARNING, "Unable to lock context?\n");
 					} else {
 						e = find_matching_priority(c, chan->exten, chan->priority, chan->cid.cid_num);
@@ -333,55 +294,65 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
 		/* Reset the macro depth, if it was changed in the last iteration */
 		pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
 
-		if ((res = ast_spawn_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num, &foundx,1))) {
+		if ((res = ast_spawn_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num))) {
 			/* Something bad happened, or a hangup has been requested. */
 			if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F')) ||
 		    	(res == '*') || (res == '#')) {
 				/* Just return result as to the previous application as if it had been dialed */
-				ast_debug(1, "Oooh, got something to jump out with ('%c')!\n", res);
+				ast_log(LOG_DEBUG, "Oooh, got something to jump out with ('%c')!\n", res);
 				break;
 			}
 			switch(res) {
 			case MACRO_EXIT_RESULT:
 				res = 0;
 				goto out;
+			case AST_PBX_KEEPALIVE:
+				if (option_debug)
+					ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited KEEPALIVE in macro %s on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
+				if (option_verbose > 1)
+					ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited KEEPALIVE in macro '%s' on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
+				goto out;
+				break;
 			default:
-				ast_debug(2, "Spawn extension (%s,%s,%d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
-				ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
+				if (option_debug)
+					ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
+				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;
 			}
 		}
 
-		ast_debug(1, "Executed application: %s\n", runningapp);
+		ast_log(LOG_DEBUG, "Executed application: %s\n", runningapp);
 
 		if (!strcasecmp(runningapp, "GOSUB")) {
 			gosub_level++;
-			ast_debug(1, "Incrementing gosub_level\n");
+			ast_log(LOG_DEBUG, "Incrementing gosub_level\n");
 		} else if (!strcasecmp(runningapp, "GOSUBIF")) {
-			char tmp2[1024], *cond, *app, *app2 = tmp2;
+			char tmp2[1024] = "", *cond, *app, *app2 = tmp2;
 			pbx_substitute_variables_helper(chan, runningdata, tmp2, sizeof(tmp2) - 1);
 			cond = strsep(&app2, "?");
 			app = strsep(&app2, ":");
 			if (pbx_checkcondition(cond)) {
 				if (!ast_strlen_zero(app)) {
 					gosub_level++;
-					ast_debug(1, "Incrementing gosub_level\n");
+					ast_log(LOG_DEBUG, "Incrementing gosub_level\n");
 				}
 			} else {
 				if (!ast_strlen_zero(app2)) {
 					gosub_level++;
-					ast_debug(1, "Incrementing gosub_level\n");
+					ast_log(LOG_DEBUG, "Incrementing gosub_level\n");
 				}
 			}
 		} else if (!strcasecmp(runningapp, "RETURN")) {
 			gosub_level--;
-			ast_debug(1, "Decrementing gosub_level\n");
+			ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
 		} else if (!strcasecmp(runningapp, "STACKPOP")) {
 			gosub_level--;
-			ast_debug(1, "Decrementing gosub_level\n");
+			ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
 		} else if (!strncasecmp(runningapp, "EXEC", 4)) {
 			/* Must evaluate args to find actual app */
-			char tmp2[1024], *tmp3 = NULL;
+			char tmp2[1024] = "", *tmp3 = NULL;
 			pbx_substitute_variables_helper(chan, runningdata, tmp2, sizeof(tmp2) - 1);
 			if (!strcasecmp(runningapp, "EXECIF")) {
 				tmp3 = strchr(tmp2, '|');
@@ -393,82 +364,85 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
 				tmp3 = tmp2;
 
 			if (tmp3)
-				ast_debug(1, "Last app: %s\n", tmp3);
+				ast_log(LOG_DEBUG, "Last app: %s\n", tmp3);
 
 			if (tmp3 && !strncasecmp(tmp3, "GOSUB", 5)) {
 				gosub_level++;
-				ast_debug(1, "Incrementing gosub_level\n");
+				ast_log(LOG_DEBUG, "Incrementing gosub_level\n");
 			} else if (tmp3 && !strncasecmp(tmp3, "RETURN", 6)) {
 				gosub_level--;
-				ast_debug(1, "Decrementing gosub_level\n");
+				ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
 			} else if (tmp3 && !strncasecmp(tmp3, "STACKPOP", 8)) {
 				gosub_level--;
-				ast_debug(1, "Decrementing gosub_level\n");
+				ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
 			}
 		}
 
 		if (gosub_level == 0 && strcasecmp(chan->context, fullmacro)) {
-			ast_verb(2, "Channel '%s' jumping out of macro '%s'\n", chan->name, macro);
+			if (option_verbose > 1)
+				ast_verbose(VERBOSE_PREFIX_2 "Channel '%s' jumping out of macro '%s'\n", chan->name, macro);
 			break;
 		}
 
 		/* don't stop executing extensions when we're in "h" */
-		if (ast_check_hangup(chan) && !inhangup) {
-			ast_debug(1, "Extension %s, macroexten %s, priority %d returned normally even though call was hung up\n", chan->exten, chan->macroexten, chan->priority);
+		if (chan->_softhangup && !inhangup) {
+			ast_log(LOG_DEBUG, "Extension %s, macroexten %s, priority %d returned normally even though call was hung up\n",
+				chan->exten, chan->macroexten, chan->priority);
 			goto out;
 		}
 		chan->priority++;
   	}
 	out:
-
-	/* Don't let the channel change now. */
-	ast_channel_lock(chan);
-
 	/* Reset the depth back to what it was when the routine was entered (like if we called Macro recursively) */
 	snprintf(depthc, sizeof(depthc), "%d", depth);
-	pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
-	ast_set2_flag(chan, autoloopflag, AST_FLAG_IN_AUTOLOOP);
+	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++) {
   		/* Restore old arguments and delete ours */
 		snprintf(varname, sizeof(varname), "ARG%d", x);
   		if (oldargs[x]) {
-			pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
-			ast_free(oldargs[x]);
-		} else {
+			if (!dead)
+				pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
+			free(oldargs[x]);
+		} else if (!dead) {
 			pbx_builtin_setvar_helper(chan, varname, NULL);
 		}
   	}
 
 	/* Restore macro variables */
-	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 (!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)
-		ast_free(save_macro_exten);
+		free(save_macro_exten);
 	if (save_macro_context)
-		ast_free(save_macro_context);
+		free(save_macro_context);
 	if (save_macro_priority)
-		ast_free(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));
-		if (!(ast_check_hangup(chan) & AST_SOFTHANGUP_ASYNCGOTO)) {
+		if (!(chan->_softhangup & AST_SOFTHANGUP_ASYNCGOTO)) {
 			/* Copy the extension, so long as we're not in softhangup, where we could be given an asyncgoto */
-			const char *offsets;
 			ast_copy_string(chan->exten, oldexten, sizeof(chan->exten));
 			if ((offsets = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"))) {
 				/* Handle macro offset if it's set by checking the availability of step n + offset + 1, otherwise continue
 			   	normally if there is any problem */
-				if (sscanf(offsets, "%30d", &offset) == 1) {
+				if (sscanf(offsets, "%d", &offset) == 1) {
 					if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + offset + 1, chan->cid.cid_num)) {
 						chan->priority += offset;
 					}
@@ -477,40 +451,28 @@ static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
 		}
 	}
 
-	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)
-		ast_free(save_macro_offset);
-
-	/* Unlock the macro */
-	if (exclusive) {
-		ast_debug(1, "Unlocking macrolock for '%s'\n", fullmacro);
-		if (ast_context_unlockmacro(fullmacro)) {
-			ast_log(LOG_ERROR, "Failed to unlock macro '%s' - that isn't good\n", fullmacro);
-			res = 0;
-		}
-	}
-	ast_channel_unlock(chan);
-
+		free(save_macro_offset);
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int macro_exec(struct ast_channel *chan, void *data)
-{
-	return _macro_exec(chan, data, 0);
-}
-
-static int macroexclusive_exec(struct ast_channel *chan, void *data)
-{
-	return _macro_exec(chan, data, 1);
-}
-
 static int macroif_exec(struct ast_channel *chan, void *data) 
 {
 	char *expr = NULL, *label_a = NULL, *label_b = NULL;
 	int res = 0;
+	struct localuser *u;
 
-	if (!(expr = ast_strdupa(data)))
+	LOCAL_USER_ADD(u);
+
+	expr = ast_strdupa(data);
+	if (!expr) {
+		ast_log(LOG_ERROR, "Out of Memory!\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
+	}
 
 	if ((label_a = strchr(expr, '?'))) {
 		*label_a = '\0';
@@ -526,6 +488,8 @@ static int macroif_exec(struct ast_channel *chan, void *data)
 	} else
 		ast_log(LOG_WARNING, "Invalid Syntax.\n");
 
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }
 			
@@ -534,28 +498,43 @@ static int macro_exit_exec(struct ast_channel *chan, void *data)
 	return MACRO_EXIT_RESULT;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
 	int res;
 
 	res = ast_unregister_application(if_app);
 	res |= ast_unregister_application(exit_app);
 	res |= ast_unregister_application(app);
-	res |= ast_unregister_application(exclusive_app);
+
+	STANDARD_HANGUP_LOCALUSERS;
 
 	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	int res;
 
 	res = ast_register_application(exit_app, macro_exit_exec, exit_synopsis, exit_descrip);
 	res |= ast_register_application(if_app, macroif_exec, if_synopsis, if_descrip);
-	res |= ast_register_application(exclusive_app, macroexclusive_exec, exclusive_synopsis, exclusive_descrip);
 	res |= ast_register_application(app, macro_exec, synopsis, descrip);
 
 	return res;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Extension Macros");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_math.c

@@ -0,0 +1,298 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2004 - 2005, Andy Powell 
+ *
+ * Updated by Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief A simple math application
+ * 
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <sys/time.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <time.h>
+#include <dirent.h>
+#include <ctype.h>
+#include <sys/file.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/options.h"
+#include "asterisk/config.h"
+#include "asterisk/say.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+#include "asterisk/manager.h"
+#include "asterisk/localtime.h"
+#include "asterisk/cli.h"
+#include "asterisk/utils.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+
+static char *tdesc = "Basic Math Functions";
+
+static char *app_math = "Math";
+
+static char *math_synopsis = "Performs Mathematical Functions";
+
+static char *math_descrip =
+"Math(returnvar,<number1><op><number 2>\n\n"
+"Perform floating point calculation on number 1 to number 2 and \n"
+"store the result in returnvar.  Valid ops are: \n"
+"    +,-,/,*,%,<,>,>=,<=,==\n"
+"and behave as their C equivalents.\n"
+"  This application has been deprecated in favor of the MATH function.\n";
+
+#define ADDFUNCTION 0
+#define DIVIDEFUNCTION 1
+#define MULTIPLYFUNCTION 2
+#define SUBTRACTFUNCTION 3
+#define MODULUSFUNCTION 4
+
+#define GTFUNCTION 5
+#define LTFUNCTION 6
+#define GTEFUNCTION 7
+#define LTEFUNCTION 8
+#define EQFUNCTION 9
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int math_exec(struct ast_channel *chan, void *data) 
+{
+	float fnum1;
+	float fnum2;
+	float ftmp = 0;
+	char *op;
+	int iaction=-1;
+	static int deprecation_warning = 0;
+
+	/* dunno, big calulations :D */
+	char user_result[30];
+
+	char *s;
+	char *mvar, *mvalue1, *mvalue2=NULL;
+		
+	struct localuser *u;
+
+	if (!deprecation_warning) {
+		ast_log(LOG_WARNING, "Math() is deprecated, please use Set(var=${MATH(...)} instead.\n");
+		deprecation_warning = 1;
+	}
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "No parameters passed. !\n");
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	s = ast_strdupa(data);
+	if (!s) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	mvar = strsep(&s, "|");
+	mvalue1 = strsep(&s, "|");
+	
+	if ((op = strchr(mvalue1, '+'))) {
+		iaction = ADDFUNCTION;
+		*op = '\0';
+	} else if ((op = strchr(mvalue1, '-'))) {
+		iaction = SUBTRACTFUNCTION;
+		*op = '\0';
+	} else if ((op = strchr(mvalue1, '*'))) {
+		iaction = MULTIPLYFUNCTION;
+		*op = '\0';
+	} else if ((op = strchr(mvalue1, '/'))) {
+		iaction = DIVIDEFUNCTION;
+		*op = '\0';
+	} else if ((op = strchr(mvalue1, '>'))) {
+		iaction = GTFUNCTION;
+		*op = '\0';
+		if (*(op+1) == '=') {
+			op++;
+			*op = '\0';
+			iaction = GTEFUNCTION;
+		}
+	} else if ((op = strchr(mvalue1, '<'))) {
+		iaction = LTFUNCTION;
+		*op = '\0';
+		if (*(op+1) == '=') {
+			op++;
+			*op = '\0';
+			iaction = LTEFUNCTION;
+		}
+	} else if ((op = strchr(mvalue1, '='))) {
+		iaction = GTFUNCTION;
+		*op = '\0';
+		if (*(op+1) == '=') {
+			op++;
+			*op = '\0';
+			iaction = EQFUNCTION;
+		} else
+			op = NULL;
+	} 
+	
+	if (op) 
+		mvalue2 = op + 1;
+		
+	if (!mvar || !mvalue1 || !mvalue2) {
+		ast_log(LOG_WARNING, "Supply all the parameters - just this once, please\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (!strcmp(mvar,"")) {
+		ast_log(LOG_WARNING, "No return variable set.\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (sscanf(mvalue1, "%f", &fnum1) != 1) {
+		ast_log(LOG_WARNING, "'%s' is not a valid number\n", mvalue1);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (sscanf(mvalue2, "%f", &fnum2) != 1) {
+		ast_log(LOG_WARNING, "'%s' is not a valid number\n", mvalue2);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	switch (iaction) {
+	case ADDFUNCTION :
+		ftmp = fnum1 + fnum2;
+		break;
+	case DIVIDEFUNCTION :
+		if (fnum2 <=0)
+			ftmp = 0; /* can't do a divide by 0 */
+		else
+			ftmp = (fnum1 / fnum2);
+		break;
+	case MULTIPLYFUNCTION :
+		ftmp = (fnum1 * fnum2);
+		break;
+	case SUBTRACTFUNCTION :
+		ftmp = (fnum1 - fnum2);
+		break;
+	case MODULUSFUNCTION : {
+		int inum1 = fnum1;
+		int inum2 = fnum2;
+			
+		ftmp = (inum1 % inum2);
+		
+		break;
+		}
+	case GTFUNCTION :
+		if (fnum1 > fnum2)
+			strcpy(user_result, "TRUE");
+		else
+			strcpy(user_result, "FALSE");
+		break;
+	case LTFUNCTION :
+		if (fnum1 < fnum2)
+			strcpy(user_result, "TRUE");
+		else
+			strcpy(user_result, "FALSE");
+		break;
+	case GTEFUNCTION :
+		if (fnum1 >= fnum2)
+			strcpy(user_result, "TRUE");
+		else
+			strcpy(user_result, "FALSE");
+		break;
+	case LTEFUNCTION :
+		if (fnum1 <= fnum2)
+			strcpy(user_result, "TRUE");
+		else
+			strcpy(user_result, "FALSE");
+		break;					
+	case EQFUNCTION :
+		if (fnum1 == fnum2)
+			strcpy(user_result, "TRUE");
+		else
+			strcpy(user_result, "FALSE");
+		break;
+	default :
+		ast_log(LOG_WARNING, "Something happened that neither of us should be proud of %d\n", iaction);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if (iaction < GTFUNCTION || iaction > EQFUNCTION) 
+		snprintf(user_result,sizeof(user_result),"%f",ftmp);
+		
+	pbx_builtin_setvar_helper(chan, mvar, user_result);	
+	
+	LOCAL_USER_REMOVE(u);
+	return 0;
+}
+
+int unload_module(void)
+{
+	int res;
+	
+	res = ast_unregister_application(app_math);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_application(app_math, math_exec, math_synopsis, math_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
+
+/* Fading everything to black and blue... */

apps/app_md5.c

@@ -0,0 +1,205 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2005, Olle E. Johansson, Edvina.net
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+
+/*! \file
+ *
+ * \brief MD5 checksum application
+ * 
+ * \todo Remove this deprecated application in 1.3dev
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/utils.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/app.h"
+
+static char *tdesc_md5 = "MD5 checksum applications";
+static char *app_md5 = "MD5";
+static char *desc_md5 = "Calculate MD5 checksum";
+static char *synopsis_md5 = 
+"  MD5(<var>=<string>): Calculates a MD5 checksum on <string>.\n"
+"Returns hash value in a channel variable. \n";
+
+static char *app_md5check = "MD5Check";
+static char *desc_md5check = "Check MD5 checksum";
+static char *synopsis_md5check = 
+"  MD5Check(<md5hash>|<string>[|options]): Calculates a MD5 checksum on <string>\n"
+"and compares it with the hash. Returns 0 if <md5hash> is correct for <string>.\n"
+"The option string may contain zero or more of the following characters:\n"
+"	'j' -- jump to priority n+101 if the hash and string do not match \n"
+"This application sets the following channel variable upon completion:\n"
+"	CHECKMD5STATUS	The status of the MD5 check, one of the following\n"
+"		MATCH | NOMATCH\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+/*--- md5_exec: Calculate MD5 checksum (hash) on given string and
+	return it in channel variable ---*/
+static int md5_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *varname= NULL; /* Variable to set */
+	char *string = NULL; /* String to calculate on */
+	char retvar[50]; /* Return value */
+	static int dep_warning = 0;
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "This application has been deprecated, please use the MD5 function instead.\n");
+		dep_warning = 1;
+	}	
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Syntax: md5(<varname>=<string>) - missing argument!\n");
+		return -1;
+	}
+	
+	LOCAL_USER_ADD(u);
+
+	memset(retvar,0, sizeof(retvar));
+	string = ast_strdupa(data);
+	varname = strsep(&string,"=");
+	if (ast_strlen_zero(varname)) {
+		ast_log(LOG_WARNING, "Syntax: md5(<varname>=<string>) - missing argument!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	ast_md5_hash(retvar, string);
+	pbx_builtin_setvar_helper(chan, varname, retvar);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+/*--- md5check_exec: Calculate MD5 checksum and compare it with
+	existing checksum. ---*/
+static int md5check_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *string = NULL; /* String to calculate on */
+	char newhash[50]; /* Return value */
+	static int dep_warning = 0;
+	int priority_jump = 0;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(md5hash);
+		AST_APP_ARG(string);
+		AST_APP_ARG(options);
+	);
+
+	if (!dep_warning) {
+		ast_log(LOG_WARNING, "This application has been deprecated, please use the CHECK_MD5 function instead.\n");
+		dep_warning = 1;
+	}
+	
+	LOCAL_USER_ADD(u);
+
+	if (!(string = ast_strdupa(data))) {
+		ast_log(LOG_WARNING, "Memory Error!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args, string);
+
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
+	if (ast_strlen_zero(args.md5hash) || ast_strlen_zero(args.string)) {
+		ast_log(LOG_WARNING, "Syntax: MD5Check(<md5hash>|<string>[|options]) - missing argument!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	memset(newhash,0, sizeof(newhash));
+
+	ast_md5_hash(newhash, args.string);
+	if (!strcmp(newhash, args.md5hash)) {	/* Verification ok */
+		if (option_debug > 2)
+			ast_log(LOG_DEBUG, "MD5 verified ok: %s -- %s\n", args.md5hash, args.string);
+		pbx_builtin_setvar_helper(chan, "CHECKMD5STATUS", "MATCH");
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+	if (option_debug > 2)
+		ast_log(LOG_DEBUG, "ERROR: MD5 not verified: %s -- %s\n", args.md5hash, args.string);
+	pbx_builtin_setvar_helper(chan, "CHECKMD5STATUS", "NOMATCH");		
+	if (priority_jump || option_priority_jumping) {
+		if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
+			if (option_debug > 2)
+				ast_log(LOG_DEBUG, "Can't jump to exten+101 (e%s,p%d), sorry\n", chan->exten,chan->priority+101);
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app_md5);
+	res |= ast_unregister_application(app_md5check);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+
+	res = ast_register_application(app_md5check, md5check_exec, desc_md5check, synopsis_md5check);
+	res |= ast_register_application(app_md5, md5_exec, desc_md5, synopsis_md5);
+	
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc_md5;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_milliwatt.c

@@ -19,144 +19,152 @@
 /*! \file
  *
  * \brief Digital Milliwatt Test
- *
- * \author Mark Spencer <markster@digium.com>
  * 
  * \ingroup applications
  */
 
-/*** MODULEINFO
-	<depend>res_indications</depend>
- ***/
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <errno.h>
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
-#include "asterisk/indications.h"
+#include "asterisk/module.h"
+
+static char *tdesc = "Digital Milliwatt (mu-law) Test Application";
 
 static char *app = "Milliwatt";
 
-static char *synopsis = "Generate a Constant 1004Hz tone at 0dbm (mu-law)";
+static char *synopsis = "Generate a Constant 1000Hz tone at 0dbm (mu-law)";
 
 static char *descrip = 
-"   Milliwatt([options]): Generate a Constant 1004Hz tone at 0dbm.\n"
-"Previous versions of this application generated the tone at 1000Hz.  If for\n"
-"some reason you would prefer that behavior, supply the 'o' option to get the\n"
-"old behavior.\n"
-"";
+"Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law)\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static char digital_milliwatt[] = {0x1e,0x0b,0x0b,0x1e,0x9e,0x8b,0x8b,0x9e} ;
 
 static void *milliwatt_alloc(struct ast_channel *chan, void *params)
 {
-	return ast_calloc(1, sizeof(int));
+int	*indexp;
+	indexp = malloc(sizeof(int));
+	if (indexp == NULL) return(NULL);
+	*indexp = 0;
+	return(indexp);
 }
 
 static void milliwatt_release(struct ast_channel *chan, void *data)
 {
-	ast_free(data);
+	free(data);
 	return;
 }
 
 static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int samples)
 {
+	struct ast_frame wf;
 	unsigned char buf[AST_FRIENDLY_OFFSET + 640];
-	const int maxsamples = ARRAY_LEN(buf);
-	int i, *indexp = (int *) data;
-	struct ast_frame wf = {
-		.frametype = AST_FRAME_VOICE,
-		.subclass = AST_FORMAT_ULAW,
-		.offset = AST_FRIENDLY_OFFSET,
-		.data = buf + AST_FRIENDLY_OFFSET,
-		.src = __FUNCTION__,
-	};
+	int i,*indexp = (int *) data;
 
-	/* Instead of len, use samples, because channel.c generator_force
-	* generate(chan, tmp, 0, 160) ignores len. In any case, len is
-	* a multiple of samples, given by number of samples times bytes per
-	* sample. In the case of ulaw, len = samples. for signed linear
-	* len = 2 * samples */
-	if (samples > maxsamples) {
-		ast_log(LOG_WARNING, "Only doing %d samples (%d requested)\n", maxsamples, samples);
-		samples = maxsamples;
+	if (len + AST_FRIENDLY_OFFSET > 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;
 	}
-
-	len = samples * sizeof (buf[0]);
+	wf.frametype = AST_FRAME_VOICE;
+	wf.subclass = AST_FORMAT_ULAW;
+	wf.offset = AST_FRIENDLY_OFFSET;
+	wf.mallocd = 0;
+	wf.data = buf + AST_FRIENDLY_OFFSET;
 	wf.datalen = len;
-	wf.samples = samples;
-
+	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++) {
+	for(i = 0; i < len; i++)
+	{
 		buf[AST_FRIENDLY_OFFSET + i] = digital_milliwatt[(*indexp)++];
 		*indexp &= 7;
 	}
-
-	if (ast_write(chan,&wf) < 0) {
+	if (ast_write(chan,&wf) < 0)
+	{
 		ast_log(LOG_WARNING,"Failed to write frame to '%s': %s\n",chan->name,strerror(errno));
 		return -1;
 	}
-
 	return 0;
 }
 
-static struct ast_generator milliwattgen = {
+static struct ast_generator milliwattgen = 
+{
 	alloc: milliwatt_alloc,
 	release: milliwatt_release,
 	generate: milliwatt_generate,
-};
-
-static int old_milliwatt_exec(struct ast_channel *chan)
-{
-	ast_set_write_format(chan, AST_FORMAT_ULAW);
-	ast_set_read_format(chan, AST_FORMAT_ULAW);
-
-	if (chan->_state != AST_STATE_UP) {
-		ast_answer(chan);
-	}
-
-	if (ast_activate_generator(chan,&milliwattgen,"milliwatt") < 0) {
-		ast_log(LOG_WARNING,"Failed to activate generator on '%s'\n",chan->name);
-		return -1;
-	}
-
-	while (!ast_safe_sleep(chan, 10000))
-		;
-
-	ast_deactivate_generator(chan);
-
-	return -1;
-}
+} ;
 
 static int milliwatt_exec(struct ast_channel *chan, void *data)
 {
-	const char *options = data;
-	int res = -1;
 
-	if (!ast_strlen_zero(options) && strchr(options, 'o')) {
-		return old_milliwatt_exec(chan);
+	struct localuser *u;
+	LOCAL_USER_ADD(u);
+	ast_set_write_format(chan, AST_FORMAT_ULAW);
+	ast_set_read_format(chan, AST_FORMAT_ULAW);
+	if (chan->_state != AST_STATE_UP)
+	{
+		ast_answer(chan);
 	}
-
-	res = ast_playtones_start(chan, 23255, "1004/1000", 0);
-
-	while (!res) {
-		res = ast_safe_sleep(chan, 10000);
+	if (ast_activate_generator(chan,&milliwattgen,"milliwatt") < 0)
+	{
+		ast_log(LOG_WARNING,"Failed to activate generator on '%s'\n",chan->name);
+		LOCAL_USER_REMOVE(u);
+		return -1;
 	}
+	while(!ast_safe_sleep(chan, 10000));
+	ast_deactivate_generator(chan);
+	LOCAL_USER_REMOVE(u);
+	return -1;
+}
+
+int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
 
 	return res;
 }
 
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, milliwatt_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Digital Milliwatt (mu-law) Test Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_mixmonitor.c

@@ -2,7 +2,7 @@
  * Asterisk -- An open source telephony toolkit.
  *
  * Copyright (C) 2005, Anthony Minessale II
- * Copyright (C) 2005 - 2006, Digium, Inc.
+ * Copyright (C) 2005, Digium, Inc.
  *
  * Mark Spencer <markster@digium.com>
  * Kevin P. Fleming <kpfleming@digium.com>
@@ -22,50 +22,47 @@
  */
 
 /*! \file
- *
  * \brief MixMonitor() - Record a call and mix the audio during the recording
  * \ingroup applications
- *
- * \author Mark Spencer <markster@digium.com>
- * \author Kevin P. Fleming <kpfleming@digium.com>
- *
- * \note Based on app_muxmon.c provided by
- * Anthony Minessale II <anthmct@yahoo.com>
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include "asterisk/paths.h"	/* use ast_config_AST_MONITOR_DIR */
 #include "asterisk/file.h"
-#include "asterisk/audiohook.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/chanspy.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
+#include "asterisk/lock.h"
 #include "asterisk/cli.h"
+#include "asterisk/options.h"
 #include "asterisk/app.h"
-#include "asterisk/channel.h"
+#include "asterisk/linkedlists.h"
 
 #define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
 
+static const char *tdesc = "Mixed Audio Monitoring Application";
 static const char *app = "MixMonitor";
 static const char *synopsis = "Record a call and mix the audio during the recording";
 static const char *desc = ""
-"  MixMonitor(<file>.<ext>[,<options>[,<command>]]):\n"
+"  MixMonitor(<file>.<ext>[|<options>[|<command>]])\n\n"
 "Records the audio on the current channel to the specified file.\n"
 "If the filename is an absolute path, uses that path, otherwise\n"
 "creates the file in the configured monitoring directory from\n"
-"asterisk.conf.  Use of StopMixMonitor is required to guarantee\n"
-"the audio file is available for processing during dialplan execution.\n\n"
+"asterisk.conf.\n\n"
 "Valid options:\n"
 " a      - Append to the file instead of overwriting it.\n"
 " b      - Only save audio to the file while the channel is bridged.\n"
 "          Note: Does not include conferences or sounds played to each bridged\n"
 "                party.\n"
-"          Note: If you utilize this option inside a Local channel, you must\n"
-"                 make sure the Local channel is not optimized away. To do this,\n"
-"                 be sure to call your Local channel with the '/n' option.\n"
-"                 For example: Dial(Local/start@mycontext/n)\n"
 " v(<x>) - Adjust the heard volume by a factor of <x> (range -4 to 4)\n"	
 " V(<x>) - Adjust the spoken volume by a factor of <x> (range -4 to 4)\n"	
 " W(<x>) - Adjust the both heard and spoken volumes by a factor of <x>\n"
@@ -76,24 +73,18 @@ static const char *desc = ""
 "The variable MIXMONITOR_FILENAME will contain the filename used to record.\n"
 "";
 
-static const char *stop_app = "StopMixMonitor";
-static const char *stop_synopsis = "Stop recording a call through MixMonitor";
-static const char *stop_desc = ""
-"  StopMixMonitor():\n"
-"Stop recording a call through MixMonitor, and free the recording's file handle.\n"
-"";
+STANDARD_LOCAL_USER;
 
-struct module_symbols *me;
+LOCAL_USER_DECL;
 
 static const char *mixmonitor_spy_type = "MixMonitor";
 
 struct mixmonitor {
-	struct ast_audiohook audiohook;
+	struct ast_channel_spy spy;
 	char *filename;
 	char *post_process;
 	char *name;
 	unsigned int flags;
-	struct mixmonitor_ds *mixmonitor_ds;
 };
 
 enum {
@@ -119,94 +110,17 @@ AST_APP_OPTIONS(mixmonitor_opts, {
 	AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
 });
 
-/* This structure is used as a means of making sure that our pointer to
- * the channel we are monitoring remains valid. This is very similar to 
- * what is used in app_chanspy.c.
- */
-struct mixmonitor_ds {
-	struct ast_channel *chan;
-	/* These condition variables are used to be sure that the channel
-	 * hangup code completes before the mixmonitor thread attempts to
-	 * free this structure. The combination of a bookean flag and a
-	 * ast_cond_t ensure that no matter what order the threads run in,
-	 * we are guaranteed to never have the waiting thread block forever
-	 * in the case that the signaling thread runs first.
-	 */
-	unsigned int destruction_ok;
-	ast_cond_t destruction_condition;
-	ast_mutex_t lock;
-
-	/* The filestream is held in the datastore so it can be stopped
-	 * immediately during stop_mixmonitor or channel destruction. */
-	int fs_quit;
-	struct ast_filestream *fs;
-	struct ast_audiohook *audiohook;
-};
-
- /*!
-  * \internal
-  * \pre mixmonitor_ds must be locked before calling this function
-  */
-static void mixmonitor_ds_close_fs(struct mixmonitor_ds *mixmonitor_ds)
+static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy) 
 {
-	if (mixmonitor_ds->fs) {
-		ast_closestream(mixmonitor_ds->fs);
-		mixmonitor_ds->fs = NULL;
-		mixmonitor_ds->fs_quit = 1;
-		ast_verb(2, "MixMonitor close filestream\n");
-	}
-}
-
-static void mixmonitor_ds_destroy(void *data)
-{
-	struct mixmonitor_ds *mixmonitor_ds = data;
-
-	ast_mutex_lock(&mixmonitor_ds->lock);
-	mixmonitor_ds->chan = NULL;
-	mixmonitor_ds->audiohook = NULL;
-	mixmonitor_ds->destruction_ok = 1;
-	ast_cond_signal(&mixmonitor_ds->destruction_condition);
-	ast_mutex_unlock(&mixmonitor_ds->lock);
-}
-
-static void mixmonitor_ds_chan_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan)
-{
-	struct mixmonitor_ds *mixmonitor_ds = data;
-
-	ast_mutex_lock(&mixmonitor_ds->lock);
-	mixmonitor_ds->chan = new_chan;
-	ast_mutex_unlock(&mixmonitor_ds->lock);
-}
-
-static struct ast_datastore_info mixmonitor_ds_info = {
-	.type = "mixmonitor",
-	.destroy = mixmonitor_ds_destroy,
-	.chan_fixup = mixmonitor_ds_chan_fixup,
-};
-
-static void destroy_monitor_audiohook(struct mixmonitor *mixmonitor)
-{
-	if (mixmonitor->mixmonitor_ds) {
-		ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
-		mixmonitor->mixmonitor_ds->audiohook = NULL;
-		ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
-	}
-	/* kill the audiohook.*/
-	ast_audiohook_lock(&mixmonitor->audiohook);
-	ast_audiohook_detach(&mixmonitor->audiohook);
-	ast_audiohook_unlock(&mixmonitor->audiohook);
-	ast_audiohook_destroy(&mixmonitor->audiohook);
-}
-
-static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook) 
-{
-	struct ast_channel *peer = NULL;
-	int res = 0;
+	struct ast_channel *peer;
+	int res;
 
 	if (!chan)
 		return -1;
 
-	ast_audiohook_attach(chan, audiohook);
+	ast_mutex_lock(&chan->lock);
+	res = ast_channel_spy_add(chan, spy);
+	ast_mutex_unlock(&chan->lock);
 
 	if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan)))
 		ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);	
@@ -216,139 +130,97 @@ static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook)
 
 #define SAMPLES_PER_FRAME 160
 
-static void mixmonitor_free(struct mixmonitor *mixmonitor)
-{
-	if (mixmonitor) {
-		if (mixmonitor->mixmonitor_ds) {
-			ast_mutex_destroy(&mixmonitor->mixmonitor_ds->lock);
-			ast_cond_destroy(&mixmonitor->mixmonitor_ds->destruction_condition);
-			ast_free(mixmonitor->mixmonitor_ds);
-		}
-		ast_free(mixmonitor);
-	}
-}
-
 static void *mixmonitor_thread(void *obj) 
 {
 	struct mixmonitor *mixmonitor = obj;
-	struct ast_filestream **fs = NULL;
+	struct ast_frame *f = NULL;
+	struct ast_filestream *fs = NULL;
 	unsigned int oflags;
 	char *ext;
 	int errflag = 0;
+	
+	
+	STANDARD_INCREMENT_USECOUNT;
+	
+	if (option_verbose > 1)
+		ast_verbose(VERBOSE_PREFIX_2 "Begin MixMonitor Recording %s\n", mixmonitor->name);
+	
+	ast_mutex_lock(&mixmonitor->spy.lock);
 
-	ast_verb(2, "Begin MixMonitor Recording %s\n", mixmonitor->name);
+	while (mixmonitor->spy.chan) {
+		struct ast_frame *next;
+		int write;
 
-	fs = &mixmonitor->mixmonitor_ds->fs;
-
-	/* The audiohook must enter and exit the loop locked */
-	ast_audiohook_lock(&mixmonitor->audiohook);
-	while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING && !mixmonitor->mixmonitor_ds->fs_quit) {
-		struct ast_frame *fr = NULL;
-
-		if (!(fr = ast_audiohook_read_frame(&mixmonitor->audiohook, SAMPLES_PER_FRAME, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR))) {
-			ast_audiohook_trigger_wait(&mixmonitor->audiohook);
-
-			if (mixmonitor->audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
+		ast_channel_spy_trigger_wait(&mixmonitor->spy);
+		
+		if (!mixmonitor->spy.chan || mixmonitor->spy.status != CHANSPY_RUNNING)
+			break;
+		
+		while (1) {
+			if (!(f = ast_channel_spy_read_frame(&mixmonitor->spy, SAMPLES_PER_FRAME)))
 				break;
-			}
-			continue;
-		}
 
-		/* audiohook lock is not required for the next block.
-		 * Unlock it, but remember to lock it before looping or exiting */
-		ast_audiohook_unlock(&mixmonitor->audiohook);
+			write = (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) ||
+				 ast_bridged_channel(mixmonitor->spy.chan));
 
-		ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
-		if (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) || (mixmonitor->mixmonitor_ds->chan && ast_bridged_channel(mixmonitor->mixmonitor_ds->chan))) {
-			/* Initialize the file if not already done so */
-			if (!*fs && !errflag && !mixmonitor->mixmonitor_ds->fs_quit) {
-				oflags = O_CREAT | O_WRONLY;
-				oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
+			/* it is possible for ast_channel_spy_read_frame() to return a chain
+			   of frames if a queue flush was necessary, so process them
+			*/
+			for (; f; f = next) {
+				next = f->next;
+				if (write && errflag == 0) {
+					if (!fs) {
+						/* Determine creation flags and filename plus extension for filestream */
+						oflags = O_CREAT | O_WRONLY;
+						oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
 
-				if ((ext = strrchr(mixmonitor->filename, '.')))
-					*(ext++) = '\0';
-				else
-					ext = "raw";
+						if ((ext = strrchr(mixmonitor->filename, '.')))
+							*(ext++) = '\0';
+						else
+							ext = "raw";
 
-				if (!(*fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) {
-					ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
-					errflag = 1;
-				}
-			}
-
-			/* Write out the frame(s) */
-			if (*fs) {
-				struct ast_frame *cur;
-
-				for (cur = fr; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
-					ast_writestream(*fs, cur);
+						/* Move onto actually creating the filestream */
+						if (!(fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) {
+							ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
+							errflag = 1;
+						}
+
+					}
+					if (fs)
+						ast_writestream(fs, f);
 				}
+				ast_frfree(f);
 			}
 		}
-		ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
-
-		/* All done! free it. */
-		ast_frame_free(fr, 0);
-		ast_audiohook_lock(&mixmonitor->audiohook);
 	}
-	ast_audiohook_unlock(&mixmonitor->audiohook);
 
-	/* Datastore cleanup.  close the filestream and wait for ds destruction */
-	ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
-	mixmonitor_ds_close_fs(mixmonitor->mixmonitor_ds);
-	if (!mixmonitor->mixmonitor_ds->destruction_ok) {
-		ast_cond_wait(&mixmonitor->mixmonitor_ds->destruction_condition, &mixmonitor->mixmonitor_ds->lock);
-	}
-	ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
+	ast_mutex_unlock(&mixmonitor->spy.lock);
+	
+	ast_channel_spy_free(&mixmonitor->spy);
 
-	/* kill the audiohook */
-	destroy_monitor_audiohook(mixmonitor);
+	if (option_verbose > 1)
+		ast_verbose(VERBOSE_PREFIX_2 "End MixMonitor Recording %s\n", mixmonitor->name);
 
 	if (mixmonitor->post_process) {
-		ast_verb(2, "Executing [%s]\n", mixmonitor->post_process);
+		if (option_verbose > 2)
+			ast_verbose(VERBOSE_PREFIX_2 "Executing [%s]\n", mixmonitor->post_process);
 		ast_safe_system(mixmonitor->post_process);
 	}
+		
+	if (fs)
+		ast_closestream(fs);
+
+	free(mixmonitor);
+
+	STANDARD_DECREMENT_USECOUNT;
 
-	ast_verb(2, "End MixMonitor Recording %s\n", mixmonitor->name);
-	mixmonitor_free(mixmonitor);
 	return NULL;
 }
 
-static int setup_mixmonitor_ds(struct mixmonitor *mixmonitor, struct ast_channel *chan)
-{
-	struct ast_datastore *datastore = NULL;
-	struct mixmonitor_ds *mixmonitor_ds;
-
-	if (!(mixmonitor_ds = ast_calloc(1, sizeof(*mixmonitor_ds)))) {
-		return -1;
-	}
-	
-	ast_mutex_init(&mixmonitor_ds->lock);
-	ast_cond_init(&mixmonitor_ds->destruction_condition, NULL);
-
-	if (!(datastore = ast_channel_datastore_alloc(&mixmonitor_ds_info, NULL))) {
-		ast_mutex_destroy(&mixmonitor_ds->lock);
-		ast_cond_destroy(&mixmonitor_ds->destruction_condition);
-		ast_free(mixmonitor_ds);
-		return -1;
-	}
-
-	/* No need to lock mixmonitor_ds since this is still operating in the channel's thread */
-	mixmonitor_ds->chan = chan;
-	mixmonitor_ds->audiohook = &mixmonitor->audiohook;
-	datastore->data = mixmonitor_ds;
-
-	ast_channel_lock(chan);
-	ast_channel_datastore_add(chan, datastore);
-	ast_channel_unlock(chan);
-
-	mixmonitor->mixmonitor_ds = mixmonitor_ds;
-	return 0;
-}
-
 static void launch_monitor_thread(struct ast_channel *chan, const char *filename, unsigned int flags,
 				  int readvol, int writevol, const char *post_process) 
 {
+	pthread_attr_t attr;
 	pthread_t thread;
 	struct mixmonitor *mixmonitor;
 	char postprocess2[1024] = "";
@@ -356,7 +228,6 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename
 
 	len = sizeof(*mixmonitor) + strlen(chan->name) + strlen(filename) + 2;
 
-	postprocess2[0] = 0;
 	/* If a post process system command is given attach it to the structure */
 	if (!ast_strlen_zero(post_process)) {
 		char *p1, *p2;
@@ -367,28 +238,20 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename
 				*p2 = '$';
 			}
 		}
+
 		pbx_substitute_variables_helper(chan, p1, postprocess2, sizeof(postprocess2) - 1);
 		if (!ast_strlen_zero(postprocess2))
 			len += strlen(postprocess2) + 1;
 	}
 
 	/* Pre-allocate mixmonitor structure and spy */
-	if (!(mixmonitor = ast_calloc(1, len))) {
-		return;
-	}
-
-	/* Setup the actual spy before creating our thread */
-	if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type)) {
-		mixmonitor_free(mixmonitor);
+	if (!(mixmonitor = calloc(1, len))) {
+		ast_log(LOG_ERROR, "Memory Error!\n");
 		return;
 	}
 
 	/* Copy over flags and channel name */
 	mixmonitor->flags = flags;
-	if (setup_mixmonitor_ds(mixmonitor, chan)) {
-		mixmonitor_free(mixmonitor);
-		return;
-	}
 	mixmonitor->name = (char *) mixmonitor + sizeof(*mixmonitor);
 	strcpy(mixmonitor->name, chan->name);
 	if (!ast_strlen_zero(postprocess2)) {
@@ -399,29 +262,45 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename
 	mixmonitor->filename = (char *) mixmonitor + sizeof(*mixmonitor) + strlen(chan->name) + 1;
 	strcpy(mixmonitor->filename, filename);
 
-	ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_SYNC);
+	/* Setup the actual spy before creating our thread */
+	ast_set_flag(&mixmonitor->spy, CHANSPY_FORMAT_AUDIO);
+	ast_set_flag(&mixmonitor->spy, CHANSPY_MIXAUDIO);
+	mixmonitor->spy.type = mixmonitor_spy_type;
+	mixmonitor->spy.status = CHANSPY_RUNNING;
+	mixmonitor->spy.read_queue.format = AST_FORMAT_SLINEAR;
+	mixmonitor->spy.write_queue.format = AST_FORMAT_SLINEAR;
+	if (readvol) {
+		ast_set_flag(&mixmonitor->spy, CHANSPY_READ_VOLADJUST);
+		mixmonitor->spy.read_vol_adjustment = readvol;
+	}
+	if (writevol) {
+		ast_set_flag(&mixmonitor->spy, CHANSPY_WRITE_VOLADJUST);
+		mixmonitor->spy.write_vol_adjustment = writevol;
+	}
+	ast_mutex_init(&mixmonitor->spy.lock);
 
-	if (readvol)
-		mixmonitor->audiohook.options.read_volume = readvol;
-	if (writevol)
-		mixmonitor->audiohook.options.write_volume = writevol;
-
-	if (startmon(chan, &mixmonitor->audiohook)) {
+	if (startmon(chan, &mixmonitor->spy)) {
 		ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
-			mixmonitor_spy_type, chan->name);
-		ast_audiohook_destroy(&mixmonitor->audiohook);
-		mixmonitor_free(mixmonitor);
+			mixmonitor->spy.type, chan->name);
+		/* Since we couldn't add ourselves - bail out! */
+		ast_mutex_destroy(&mixmonitor->spy.lock);
+		free(mixmonitor);
 		return;
 	}
 
-	ast_pthread_create_detached_background(&thread, NULL, mixmonitor_thread, mixmonitor);
+	pthread_attr_init(&attr);
+	pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
+	ast_pthread_create(&thread, &attr, mixmonitor_thread, mixmonitor);
+	pthread_attr_destroy(&attr);
+
 }
 
 static int mixmonitor_exec(struct ast_channel *chan, void *data)
 {
 	int x, readvol = 0, writevol = 0;
+	struct localuser *u;
 	struct ast_flags flags = {0};
-	char *parse, *tmp, *slash;
+	char *parse;
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(filename);
 		AST_APP_ARG(options);
@@ -433,12 +312,19 @@ static int mixmonitor_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 
-	parse = ast_strdupa(data);
+	LOCAL_USER_ADD(u);
+
+	if (!(parse = ast_strdupa(data))) {
+		ast_log(LOG_WARNING, "Memory Error!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
 
 	AST_STANDARD_APP_ARGS(args, parse);
 	
 	if (ast_strlen_zero(args.filename)) {
 		ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 
@@ -450,7 +336,7 @@ static int mixmonitor_exec(struct ast_channel *chan, void *data)
 		if (ast_test_flag(&flags, MUXFLAG_READVOLUME)) {
 			if (ast_strlen_zero(opts[OPT_ARG_READVOLUME])) {
 				ast_log(LOG_WARNING, "No volume level was provided for the heard volume ('v') option.\n");
-			} else if ((sscanf(opts[OPT_ARG_READVOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
+			} 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);
@@ -460,7 +346,7 @@ static int mixmonitor_exec(struct ast_channel *chan, void *data)
 		if (ast_test_flag(&flags, MUXFLAG_WRITEVOLUME)) {
 			if (ast_strlen_zero(opts[OPT_ARG_WRITEVOLUME])) {
 				ast_log(LOG_WARNING, "No volume level was provided for the spoken volume ('V') option.\n");
-			} else if ((sscanf(opts[OPT_ARG_WRITEVOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
+			} 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);
@@ -470,7 +356,7 @@ static int mixmonitor_exec(struct ast_channel *chan, void *data)
 		if (ast_test_flag(&flags, MUXFLAG_VOLUME)) {
 			if (ast_strlen_zero(opts[OPT_ARG_VOLUME])) {
 				ast_log(LOG_WARNING, "No volume level was provided for the combined volume ('W') option.\n");
-			} else if ((sscanf(opts[OPT_ARG_VOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
+			} 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);
@@ -487,114 +373,82 @@ static int mixmonitor_exec(struct ast_channel *chan, void *data)
 		args.filename = build;
 	}
 
-	tmp = ast_strdupa(args.filename);
-	if ((slash = strrchr(tmp, '/')))
-		*slash = '\0';
-	ast_mkdir(tmp, 0777);
-
 	pbx_builtin_setvar_helper(chan, "MIXMONITOR_FILENAME", args.filename);
 	launch_monitor_thread(chan, args.filename, flags.flags, readvol, writevol, args.post_process);
 
-	return 0;
-}
-
-static int stop_mixmonitor_exec(struct ast_channel *chan, void *data)
-{
-	struct ast_datastore *datastore = NULL;
-
-	ast_channel_lock(chan);
-	ast_audiohook_detach_source(chan, mixmonitor_spy_type);
-	if ((datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info, NULL))) {
-		struct mixmonitor_ds *mixmonitor_ds = datastore->data;
-
-		ast_mutex_lock(&mixmonitor_ds->lock);
-
-		/* closing the filestream here guarantees the file is avaliable to the dialplan
-	 	 * after calling StopMixMonitor */
-		mixmonitor_ds_close_fs(mixmonitor_ds);
-
-		/* The mixmonitor thread may be waiting on the audiohook trigger.
-		 * In order to exit from the mixmonitor loop before waiting on channel
-		 * destruction, poke the audiohook trigger. */
-		if (mixmonitor_ds->audiohook) {
-			ast_audiohook_lock(mixmonitor_ds->audiohook);
-			ast_cond_signal(&mixmonitor_ds->audiohook->trigger);
-			ast_audiohook_unlock(mixmonitor_ds->audiohook);
-			mixmonitor_ds->audiohook = NULL;
-		}
-
-		ast_mutex_unlock(&mixmonitor_ds->lock);
-
-		/* Remove the datastore so the monitor thread can exit */
-		if (!ast_channel_datastore_remove(chan, datastore)) {
-			ast_channel_datastore_free(datastore);
-		}
-	}
-	ast_channel_unlock(chan);
+	LOCAL_USER_REMOVE(u);
 
 	return 0;
 }
 
-static char *handle_cli_mixmonitor(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+static int mixmonitor_cli(int fd, int argc, char **argv) 
 {
 	struct ast_channel *chan;
 
-	switch (cmd) {
-	case CLI_INIT:
-		e->command = "mixmonitor [start|stop]";
-		e->usage =
-			"Usage: mixmonitor <start|stop> <chan_name> [args]\n"
-			"       The optional arguments are passed to the MixMonitor\n"
-			"       application when the 'start' command is used.\n";
-		return NULL;
-	case CLI_GENERATE:
-		return ast_complete_channels(a->line, a->word, a->pos, a->n, 2);
+	if (argc < 3)
+		return RESULT_SHOWUSAGE;
+
+	if (!(chan = ast_get_channel_by_name_prefix_locked(argv[2], strlen(argv[2])))) {
+		ast_cli(fd, "No channel matching '%s' found.\n", argv[2]);
+		return RESULT_SUCCESS;
 	}
 
-	if (a->argc < 3)
-		return CLI_SHOWUSAGE;
+	if (!strcasecmp(argv[1], "start"))
+		mixmonitor_exec(chan, argv[3]);
+	else if (!strcasecmp(argv[1], "stop"))
+		ast_channel_spy_stop_by_type(chan, mixmonitor_spy_type);
 
-	if (!(chan = ast_get_channel_by_name_prefix_locked(a->argv[2], strlen(a->argv[2])))) {
-		ast_cli(a->fd, "No channel matching '%s' found.\n", a->argv[2]);
-		/* Technically this is a failure, but we don't want 2 errors printing out */
-		return CLI_SUCCESS;
-	}
+	ast_mutex_unlock(&chan->lock);
 
-	if (!strcasecmp(a->argv[1], "start")) {
-		mixmonitor_exec(chan, a->argv[3]);
-		ast_channel_unlock(chan);
-	} else {
-		ast_channel_unlock(chan);
-		ast_audiohook_detach_source(chan, mixmonitor_spy_type);
-	}
-
-	return CLI_SUCCESS;
+	return RESULT_SUCCESS;
 }
 
-static struct ast_cli_entry cli_mixmonitor[] = {
-	AST_CLI_DEFINE(handle_cli_mixmonitor, "Execute a MixMonitor command")
+
+static struct ast_cli_entry cli_mixmonitor = {
+	{ "mixmonitor", NULL, NULL },
+	mixmonitor_cli, 
+	"Execute a MixMonitor command",
+	"mixmonitor <start|stop> <chan_name> [<args>]\n"
 };
 
-static int unload_module(void)
+
+int unload_module(void)
 {
 	int res;
 
-	ast_cli_unregister_multiple(cli_mixmonitor, sizeof(cli_mixmonitor) / sizeof(struct ast_cli_entry));
-	res = ast_unregister_application(stop_app);
+	res = ast_cli_unregister(&cli_mixmonitor);
 	res |= ast_unregister_application(app);
 	
+	STANDARD_HANGUP_LOCALUSERS;
+
 	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	int res;
 
-	ast_cli_register_multiple(cli_mixmonitor, sizeof(cli_mixmonitor) / sizeof(struct ast_cli_entry));
-	res = ast_register_application(app, mixmonitor_exec, synopsis, desc);
-	res |= ast_register_application(stop_app, stop_mixmonitor_exec, stop_synopsis, stop_desc);
+	res = ast_cli_register(&cli_mixmonitor);
+	res |= ast_register_application(app, mixmonitor_exec, synopsis, desc);
 
 	return res;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mixed Audio Monitoring Application");
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_morsecode.c

@@ -1,161 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (c) 2006, Tilghman Lesher.  All rights reserved.
- *
- * Tilghman Lesher <app_morsecode__v001@the-tilghman.com>
- *
- * This code is released by the author with no restrictions on usage.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- */
-
-/*! \file
- *
- * \brief Morsecode application
- *
- * \author Tilghman Lesher <app_morsecode__v001@the-tilghman.com>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/indications.h"
-
-static char *app_morsecode = "Morsecode";
-
-static char *morsecode_synopsis = "Plays morse code";
-
-static char *morsecode_descrip =
-"  Morsecode(<string>):\n"
-"Plays the Morse code equivalent of the passed string.  If the variable\n"
-"MORSEDITLEN is set, it will use that value for the length (in ms) of the dit\n"
-"(defaults to 80).  Additionally, if MORSETONE is set, it will use that tone\n"
-"(in Hz).  The tone default is 800.\n";
-
-
-static char *morsecode[] = {
-	"", "", "", "", "", "", "", "", "", "", "", "", "", "", "", "", /*  0-15 */
-	"", "", "", "", "", "", "", "", "", "", "", "", "", "", "", "", /* 16-31 */
-	" ",      /* 32 - <space> */
-	".-.-.-", /* 33 - ! */
-	".-..-.", /* 34 - " */
-	"",       /* 35 - # */
-	"",       /* 36 - $ */
-	"",       /* 37 - % */
-	"",       /* 38 - & */
-	".----.", /* 39 - ' */
-	"-.--.-", /* 40 - ( */
-	"-.--.-", /* 41 - ) */
-	"",       /* 42 - * */
-	"",       /* 43 - + */
-	"--..--", /* 44 - , */
-	"-....-", /* 45 - - */
-	".-.-.-", /* 46 - . */
-	"-..-.",  /* 47 - / */
-	"-----", ".----", "..---", "...--", "....-", ".....", "-....", "--...", "---..", "----.", /* 48-57 - 0-9 */
-	"---...", /* 58 - : */
-	"-.-.-.", /* 59 - ; */
-	"",       /* 60 - < */
-	"-...-",  /* 61 - = */
-	"",       /* 62 - > */
-	"..--..", /* 63 - ? */
-	".--.-.", /* 64 - @ */
-	".-", "-...", "-.-.", "-..", ".", "..-.", "--.", "....", "..", ".---", "-.-", ".-..", "--",
-	"-.", "---", ".--.", "--.-", ".-.", "...", "-", "..-", "...-", ".--", "-..-", "-.--", "--..",
-	"-.--.-", /* 91 - [ (really '(') */
-	"-..-.",  /* 92 - \ (really '/') */
-	"-.--.-", /* 93 - ] (really ')') */
-	"",       /* 94 - ^ */
-	"..--.-", /* 95 - _ */
-	".----.", /* 96 - ` */
-	".-", "-...", "-.-.", "-..", ".", "..-.", "--.", "....", "..", ".---", "-.-", ".-..", "--",
-	"-.", "---", ".--.", "--.-", ".-.", "...", "-", "..-", "...-", ".--", "-..-", "-.--", "--..",
-	"-.--.-", /* 123 - { (really '(') */
-	"",       /* 124 - | */
-	"-.--.-", /* 125 - } (really ')') */
-	"-..-.",  /* 126 - ~ (really bar) */
-	". . .",  /* 127 - <del> (error) */
-};
-
-static void playtone(struct ast_channel *chan, int tone, int len)
-{
-	char dtmf[20];
-	snprintf(dtmf, sizeof(dtmf), "%d/%d", tone, len);
-	ast_playtones_start(chan, 0, dtmf, 0);
-	ast_safe_sleep(chan, len);
-	ast_playtones_stop(chan);
-}
-
-static int morsecode_exec(struct ast_channel *chan, void *data)
-{
-	int res=0, ditlen, tone;
-	char *digit;
-	const char *ditlenc, *tonec;
-
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "Syntax: Morsecode(<string>) - no argument found\n");
-		return 0;
-	}
-
-	/* Use variable MORESEDITLEN, if set (else 80) */
-	ditlenc = pbx_builtin_getvar_helper(chan, "MORSEDITLEN");
-	if (ast_strlen_zero(ditlenc) || (sscanf(ditlenc, "%30d", &ditlen) != 1)) {
-		ditlen = 80;
-	}
-
-	/* Use variable MORSETONE, if set (else 800) */
-	tonec = pbx_builtin_getvar_helper(chan, "MORSETONE");
-	if (ast_strlen_zero(tonec) || (sscanf(tonec, "%30d", &tone) != 1)) {
-		tone = 800;
-	}
-
-	for (digit = data; *digit; digit++) {
-		int digit2 = *digit;
-		char *dahdit;
-		if (digit2 < 0) {
-			continue;
-		}
-		for (dahdit = morsecode[digit2]; *dahdit; dahdit++) {
-			if (*dahdit == '-') {
-				playtone(chan, tone, 3 * ditlen);
-			} else if (*dahdit == '.') {
-				playtone(chan, tone, 1 * ditlen);
-			} else {
-				/* Account for ditlen of silence immediately following */
-				playtone(chan, 0, 2 * ditlen);
-			}
-
-			/* Pause slightly between each dit and dah */
-			playtone(chan, 0, 1 * ditlen);
-		}
-		/* Pause between characters */
-		playtone(chan, 0, 2 * ditlen);
-	}
-
-	return res;
-}
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app_morsecode);
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app_morsecode, morsecode_exec, morsecode_synopsis, morsecode_descrip);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Morse code");

apps/app_mp3.c

@@ -19,55 +19,55 @@
 /*! \file
  *
  * \brief Silly application to play an MP3 file -- uses mpg123
- *
- * \author Mark Spencer <markster@digium.com>
  * 
  * \ingroup applications
  */
  
-/*** MODULEINFO
-	<depend>working_fork</depend>
- ***/
+#include <string.h>
+#include <stdio.h>
+#include <signal.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/time.h>
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include <sys/time.h>
-#include <signal.h>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
-
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/frame.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/translate.h"
+#include "asterisk/options.h"
 
 #define LOCAL_MPG_123 "/usr/local/bin/mpg123"
 #define MPG_123 "/usr/bin/mpg123"
 
+static char *tdesc = "Silly MP3 Application";
+
 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"
+"  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;
+
+LOCAL_USER_DECL;
 
 static int mp3play(char *filename, int fd)
 {
 	int res;
 	int x;
 	sigset_t fullset, oldset;
-#ifdef HAVE_CAP
-	cap_t cap;
-#endif
 
 	sigfillset(&fullset);
 	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
@@ -79,23 +79,15 @@ static int mp3play(char *filename, int fd)
 		pthread_sigmask(SIG_SETMASK, &oldset, NULL);
 		return res;
 	}
-#ifdef HAVE_CAP
-	cap = cap_from_text("cap_net_admin-eip");
-
-	if (cap_set_proc(cap)) {
-		/* Careful with order! Logging cannot happen after we close FDs */
-		ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-	}
-	cap_free(cap);
-#endif
-	if (ast_opt_high_priority)
+	if (option_highpriority)
 		ast_set_priority(0);
 	signal(SIGPIPE, SIG_DFL);
 	pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
 
 	dup2(fd, STDOUT_FILENO);
 	for (x=STDERR_FILENO + 1;x<256;x++) {
-		close(x);
+		if (x != STDOUT_FILENO)
+			close(x);
 	}
 	/* Execute mpg123, but buffer if it's a net connection */
 	if (!strncasecmp(filename, "http://", 7)) {
@@ -124,7 +116,7 @@ static int timed_read(int fd, void *data, int datalen, int timeout)
 	struct pollfd fds[1];
 	fds[0].fd = fd;
 	fds[0].events = POLLIN;
-	res = ast_poll(fds, 1, timeout);
+	res = poll(fds, 1, timeout);
 	if (res < 1) {
 		ast_log(LOG_NOTICE, "Poll timed out/errored out with %d\n", res);
 		return -1;
@@ -136,6 +128,7 @@ static int timed_read(int fd, void *data, int datalen, int timeout)
 static int mp3_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
+	struct localuser *u;
 	int fds[2];
 	int ms = -1;
 	int pid = -1;
@@ -147,17 +140,18 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 		struct ast_frame f;
 		char offset[AST_FRIENDLY_OFFSET];
 		short frdata[160];
-	} myf = {
-		.f = { 0, },
-	};
-
+	} myf;
+	
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "MP3 Playback requires an argument (filename)\n");
 		return -1;
 	}
 
+	LOCAL_USER_ADD(u);
+
 	if (pipe(fds)) {
 		ast_log(LOG_WARNING, "Unable to create pipe\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	
@@ -167,6 +161,7 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 	res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
 	if (res < 0) {
 		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	
@@ -201,7 +196,7 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 						break;
 					}
 				} else {
-					ast_debug(1, "No more mp3\n");
+					ast_log(LOG_DEBUG, "No more mp3\n");
 					res = 0;
 					break;
 				}
@@ -209,19 +204,19 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 			} else {
 				ms = ast_waitfor(chan, ms);
 				if (ms < 0) {
-					ast_debug(1, "Hangup detected\n");
+					ast_log(LOG_DEBUG, "Hangup detected\n");
 					res = -1;
 					break;
 				}
 				if (ms) {
 					f = ast_read(chan);
 					if (!f) {
-						ast_debug(1, "Null frame == hangup() detected\n");
+						ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
 						res = -1;
 						break;
 					}
 					if (f->frametype == AST_FRAME_DTMF) {
-						ast_debug(1, "User pressed a key\n");
+						ast_log(LOG_DEBUG, "User pressed a key\n");
 						ast_frfree(f);
 						res = 0;
 						break;
@@ -238,18 +233,41 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 		kill(pid, SIGKILL);
 	if (!res && owriteformat)
 		ast_set_write_format(chan, owriteformat);
+
+	LOCAL_USER_REMOVE(u);
 	
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+	
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, mp3_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Silly MP3 Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_nbscat.c

@@ -19,35 +19,32 @@
 /*! \file
  *
  * \brief Silly application to play an NBScat file -- uses nbscat8k
- *
- * \author Mark Spencer <markster@digium.com>
- *  
+ * 
  * \ingroup applications
  */
  
-/*** MODULEINFO
-	<depend>working_fork</depend>
- ***/
+#include <string.h>
+#include <stdio.h>
+#include <signal.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/time.h>
+#include <sys/socket.h>
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
-#include <fcntl.h>
-#include <sys/time.h>
-#include <sys/socket.h>
-#include <signal.h>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
-
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/frame.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/translate.h"
+#include "asterisk/options.h"
 
 #define LOCAL_NBSCAT "/usr/local/bin/nbscat8k"
 #define NBSCAT "/usr/bin/nbscat8k"
@@ -56,23 +53,25 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #define AF_LOCAL AF_UNIX
 #endif
 
+static char *tdesc = "Silly NBS Stream Application";
+
 static char *app = "NBScat";
 
 static char *synopsis = "Play an NBS local stream";
 
 static char *descrip = 
-"  NBScat(): Executes nbscat to listen to the local NBS stream.\n"
-"User can exit by pressing any key.\n";
+"  NBScat: Executes nbscat to listen to the local NBS stream.\n"
+"User can exit by pressing any key\n.";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int NBScatplay(int fd)
 {
 	int res;
 	int x;
 	sigset_t fullset, oldset;
-#ifdef HAVE_CAP
-	cap_t cap;
-#endif
 
 	sigfillset(&fullset);
 	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
@@ -87,16 +86,7 @@ static int NBScatplay(int fd)
 	signal(SIGPIPE, SIG_DFL);
 	pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
 
-#ifdef HAVE_CAP
-	cap = cap_from_text("cap_net_admin-eip");
-
-	if (cap_set_proc(cap)) {
-		/* Careful with order! Logging cannot happen after we close FDs */
-		ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-	}
-	cap_free(cap);
-#endif
-	if (ast_opt_high_priority)
+	if (option_highpriority)
 		ast_set_priority(0);
 
 	dup2(fd, STDOUT_FILENO);
@@ -117,7 +107,7 @@ static int timed_read(int fd, void *data, int datalen)
 	struct pollfd fds[1];
 	fds[0].fd = fd;
 	fds[0].events = POLLIN;
-	res = ast_poll(fds, 1, 2000);
+	res = poll(fds, 1, 2000);
 	if (res < 1) {
 		ast_log(LOG_NOTICE, "Selected timed out/errored out with %d\n", res);
 		return -1;
@@ -129,6 +119,7 @@ static int timed_read(int fd, void *data, int datalen)
 static int NBScat_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
+	struct localuser *u;
 	int fds[2];
 	int ms = -1;
 	int pid = -1;
@@ -141,8 +132,11 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 		short frdata[160];
 	} myf;
 	
+	LOCAL_USER_ADD(u);
+
 	if (socketpair(AF_LOCAL, SOCK_STREAM, 0, fds)) {
 		ast_log(LOG_WARNING, "Unable to create socketpair\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	
@@ -152,6 +146,7 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 	res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
 	if (res < 0) {
 		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 	
@@ -183,7 +178,7 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 						break;
 					}
 				} else {
-					ast_debug(1, "No more mp3\n");
+					ast_log(LOG_DEBUG, "No more mp3\n");
 					res = 0;
 					break;
 				}
@@ -191,19 +186,19 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 			} else {
 				ms = ast_waitfor(chan, ms);
 				if (ms < 0) {
-					ast_debug(1, "Hangup detected\n");
+					ast_log(LOG_DEBUG, "Hangup detected\n");
 					res = -1;
 					break;
 				}
 				if (ms) {
 					f = ast_read(chan);
 					if (!f) {
-						ast_debug(1, "Null frame == hangup() detected\n");
+						ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
 						res = -1;
 						break;
 					}
 					if (f->frametype == AST_FRAME_DTMF) {
-						ast_debug(1, "User pressed a key\n");
+						ast_log(LOG_DEBUG, "User pressed a key\n");
 						ast_frfree(f);
 						res = 0;
 						break;
@@ -221,17 +216,40 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 	if (!res && owriteformat)
 		ast_set_write_format(chan, owriteformat);
 
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, NBScat_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Silly NBS Stream Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_page.c

@@ -20,154 +20,179 @@
  *
  * \brief page() - Paging application
  *
- * \author Mark Spencer <markster@digium.com>
- *
  * \ingroup applications
  */
 
-/*** MODULEINFO
-	<depend>dahdi</depend>
-	<depend>app_meetme</depend>
- ***/
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
 
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/options.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/file.h"
 #include "asterisk/app.h"
 #include "asterisk/chanvars.h"
-#include "asterisk/utils.h"
-#include "asterisk/devicestate.h"
-#include "asterisk/dial.h"
+
+
+static const char *tdesc = "Page Multiple Phones";
 
 static const char *app_page= "Page";
 
 static const char *page_synopsis = "Pages phones";
 
 static const char *page_descrip =
-"Page(Technology/Resource&Technology2/Resource2[,options])\n"
+"Page(Technology/Resource&Technology2/Resource2[|options])\n"
 "  Places outbound calls to the given technology / resource and dumps\n"
 "them into a conference bridge as muted participants.  The original\n"
 "caller is dumped into the conference as a speaker and the room is\n"
 "destroyed when the original caller leaves.  Valid options are:\n"
 "        d - full duplex audio\n"
-"        q - quiet, do not play beep to caller\n"
-"        r - record the page into a file (see 'r' for app_meetme)\n"
-"        s - only dial channel if devicestate says it is not in use\n";
+"        q - quiet, do not play beep to caller\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 enum {
 	PAGE_DUPLEX = (1 << 0),
 	PAGE_QUIET = (1 << 1),
-	PAGE_RECORD = (1 << 2),
-	PAGE_SKIP = (1 << 3),
 } page_opt_flags;
 
 AST_APP_OPTIONS(page_opts, {
 	AST_APP_OPTION('d', PAGE_DUPLEX),
 	AST_APP_OPTION('q', PAGE_QUIET),
-	AST_APP_OPTION('r', PAGE_RECORD),
-	AST_APP_OPTION('s', PAGE_SKIP),
 });
 
+struct calloutdata {
+	char cidnum[64];
+	char cidname[64];
+	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, cd->variables, NULL, NULL);
+	free(cd);
+	return NULL;
+}
+
+static void launch_page(struct ast_channel *chan, const char *meetmeopts, const char *tech, const char *resource)
+{
+	struct calloutdata *cd;
+	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));
+	if (cd) {
+		memset(cd, 0, sizeof(struct calloutdata));
+		ast_copy_string(cd->cidnum, chan->cid.cid_num ? chan->cid.cid_num : "", sizeof(cd->cidnum));
+		ast_copy_string(cd->cidname, chan->cid.cid_name ? chan->cid.cid_name : "", sizeof(cd->cidname));
+		ast_copy_string(cd->tech, tech, sizeof(cd->tech));
+		ast_copy_string(cd->resource, resource, sizeof(cd->resource));
+		ast_copy_string(cd->meetmeopts, meetmeopts, sizeof(cd->meetmeopts));
+
+		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)) {
+			ast_log(LOG_WARNING, "Unable to create paging thread: %s\n", strerror(errno));
+			free(cd);
+		}
+		pthread_attr_destroy(&attr);
+	}
+}
 
 static int page_exec(struct ast_channel *chan, void *data)
 {
-	char *options, *tech, *resource, *tmp, *tmp2;
-	char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0];
+	struct localuser *u;
+	char *options;
+	char *tech, *resource;
+	char meetmeopts[80];
 	struct ast_flags flags = { 0 };
-	unsigned int confid = ast_random();
+	unsigned int confid = rand();
 	struct ast_app *app;
-	int res = 0, pos = 0, i = 0;
-	struct ast_dial **dial_list;
-	unsigned int num_dials;
+	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");
 		return -1;
 	}
 
+	LOCAL_USER_ADD(u);
+
 	if (!(app = pbx_findapp("MeetMe"))) {
 		ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	};
 
 	options = ast_strdupa(data);
+	if (!options) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
 
 	ast_copy_string(originator, chan->name, sizeof(originator));
 	if ((tmp = strchr(originator, '-')))
 		*tmp = '\0';
 
-	tmp = strsep(&options, ",");
+	tmp = strsep(&options, "|");
 	if (options)
-		ast_app_parse_options(page_opts, &flags, opts, options);
+		ast_app_parse_options(page_opts, &flags, NULL, options);
 
-	snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
-		(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
+	snprintf(meetmeopts, sizeof(meetmeopts), "%ud|%sqxdw(5)", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m");
 
-	/* Count number of extensions in list by number of ampersands + 1 */
-	num_dials = 1;
-	tmp2 = tmp;
-	while (*tmp2) {
-		if (*tmp2 == '&') {
-			num_dials++;
-		}
-		tmp2++;
-	}
-
-	if (!(dial_list = ast_calloc(num_dials, sizeof(struct ast_dial *)))) {
-		ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(struct ast_dial *) * num_dials));
-		return -1;
-	}
-
-	/* Go through parsing/calling each device */
 	while ((tech = strsep(&tmp, "&"))) {
-		int state = 0;
-		struct ast_dial *dial = NULL;
-
 		/* don't call the originating device */
 		if (!strcasecmp(tech, originator))
 			continue;
 
-		/* If no resource is available, continue on */
-		if (!(resource = strchr(tech, '/'))) {
+		if ((resource = strchr(tech, '/'))) {
+			*resource++ = '\0';
+			launch_page(chan, meetmeopts, tech, resource);
+		} else {
 			ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
-			continue;
 		}
-
-		/* Ensure device is not in use if skip option is enabled */
-		if (ast_test_flag(&flags, PAGE_SKIP)) {
-			state = ast_device_state(tech);
-			if (state == AST_DEVICE_UNKNOWN) {
-				ast_log(LOG_WARNING, "Destination '%s' has device state '%s'. Paging anyway.\n", tech, devstate2str(state));
-			} else if (state != AST_DEVICE_NOT_INUSE) {
-				ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, devstate2str(state));
-				continue;
-			}
-		}
-
-		*resource++ = '\0';
-
-		/* Create a dialing structure */
-		if (!(dial = ast_dial_create())) {
-			ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
-			continue;
-		}
-
-		/* Append technology and resource */
-		ast_dial_append(dial, tech, resource);
-
-		/* Set ANSWER_EXEC as global option */
-		ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
-
-		/* Run this dial in async mode */
-		ast_dial_run(dial, chan, 1);
-
-		/* Put in our dialing array */
-		dial_list[pos++] = dial;
 	}
 
 	if (!ast_test_flag(&flags, PAGE_QUIET)) {
@@ -175,39 +200,47 @@ static int page_exec(struct ast_channel *chan, void *data)
 		if (!res)
 			res = ast_waitstream(chan, "");
 	}
-
 	if (!res) {
-		snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
-			(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
-		pbx_exec(chan, app, meetmeopts);
+		snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%sqxd", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t");
+		pbx_exec(chan, app, meetmeopts, 1);
 	}
 
-	/* Go through each dial attempt cancelling, joining, and destroying */
-	for (i = 0; i < pos; i++) {
-		struct ast_dial *dial = dial_list[i];
-
-		/* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
-		ast_dial_join(dial);
-
-		/* Hangup all channels */
-		ast_dial_hangup(dial);
-
-		/* Destroy dialing structure */
-		ast_dial_destroy(dial);
-	}
+	LOCAL_USER_REMOVE(u);
 
 	return -1;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app_page);
+	int res;
+
+	res =  ast_unregister_application(app_page);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
+char *description(void)
+{
+	return (char *) tdesc;
+}
 
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_parkandannounce.c

@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 1999 - 2006, Digium, Inc.
+ * Copyright (C) 1999 - 2005, Digium, Inc.
  *
  * Mark Spencer <markster@digium.com>
  *
@@ -22,123 +22,186 @@
 /*! \file
  *
  * \brief ParkAndAnnounce application for Asterisk
- *
- * \author Ben Miller <bgmiller@dccinc.com>
- * \arg With TONS of help from Mark!
- *
+ * 
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <sys/types.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/features.h"
+#include "asterisk/options.h"
+#include "asterisk/logger.h"
 #include "asterisk/say.h"
 #include "asterisk/lock.h"
-#include "asterisk/utils.h"
-#include "asterisk/app.h"
+
+static char *tdesc = "Call Parking and Announce Application";
 
 static char *app = "ParkAndAnnounce";
 
 static char *synopsis = "Park and Announce";
 
 static char *descrip =
-"  ParkAndAnnounce(announce:template,timeout,dial[,return_context]):\n"
-"Park a call into the parkinglot and announce the call to another channel.\n"
-"\n"
-"announce template: Colon-separated list of files to announce.  The word PARKED\n"
-"                   will be replaced by a say_digits of the extension in which\n"
-"                   the call is parked.\n"
-"timeout:           Time in seconds before the call returns into the return\n"
-"                   context.\n"
-"dial:              The app_dial style resource to call to make the\n"
-"                   announcement.  Console/dsp calls the console.\n"
-"return_context:    The goto-style label to jump the call back into after\n"
-"                   timeout.  Default <priority+1>.\n"
-"\n"
-"The variable ${PARKEDAT} will contain the parking extension into which the\n"
-"call was placed.  Use with the Local channel to allow the dialplan to make\n"
-"use of this information.\n";
+"  ParkAndAnnounce(announce:template|timeout|dial|[return_context]):\n"
+"Park a call into the parkinglot and announce the call over the console.\n"
+"announce template: colon separated list of files to announce, the word PARKED\n"
+"                   will be replaced by a say_digits of the ext the call is parked in\n"
+"timeout: time in seconds before the call returns into the return context.\n"
+"dial: The app_dial style resource to call to make the announcement. Console/dsp calls the console.\n"
+"return_context: the goto style label to jump the call back into after timeout. default=prio+1\n";
 
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
 static int parkandannounce_exec(struct ast_channel *chan, void *data)
 {
-	int res = -1;
-	int lot, timeout = 0, dres;
-	char *dialtech, *tmp[100], buf[13];
-	int looptemp, i;
-	char *s;
+	int res=0;
+	char *return_context;
+	int l, lot, timeout = 0, dres;
+	char *working, *context, *exten, *priority, *dial, *dialtech, *dialstr;
+	char *template, *tpl_working, *tpl_current;
+	char *tmp[100];
+	int looptemp=0,i=0;
+	char *s,*orig_s;
 
 	struct ast_channel *dchan;
-	struct outgoing_helper oh = { 0, };
+	struct outgoing_helper oh;
 	int outstate;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(template);
-		AST_APP_ARG(timeout);
-		AST_APP_ARG(dial);
-		AST_APP_ARG(return_context);
-	);
+
+	struct localuser *u;
+
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "ParkAndAnnounce requires arguments: (announce:template|timeout|dial|[return_context])\n");
 		return -1;
 	}
   
-	s = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, s);
+	LOCAL_USER_ADD(u);
 
-	if (args.timeout)
-		timeout = atoi(args.timeout) * 1000;
-
-	if (ast_strlen_zero(args.dial)) {
-		ast_log(LOG_WARNING, "PARK: A dial resource must be specified i.e: Console/dsp or DAHDI/g1/5551212\n");
+	l=strlen(data)+2;
+	orig_s=malloc(l);
+	if(!orig_s) {
+		ast_log(LOG_WARNING, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
+	s=orig_s;
+	strncpy(s,data,l);
 
-	dialtech = strsep(&args.dial, "/");
-	ast_verb(3, "Dial Tech,String: (%s,%s)\n", dialtech, args.dial);
-
-	if (!ast_strlen_zero(args.return_context)) {
-		ast_clear_flag(chan, AST_FLAG_IN_AUTOLOOP);
-		ast_parseable_goto(chan, args.return_context);
+	template=strsep(&s,"|");
+	if(! template) {
+		ast_log(LOG_WARNING, "PARK: An announce template must be defined\n");
+		free(orig_s);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+  
+	if(s) {
+		timeout = atoi(strsep(&s, "|"));
+		timeout *= 1000;
+	}
+	dial=strsep(&s, "|");
+	if(!dial) {
+		ast_log(LOG_WARNING, "PARK: A dial resource must be specified i.e: Console/dsp or Zap/g1/5551212\n");
+		free(orig_s);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	} else {
+		dialtech=strsep(&dial, "/");
+		dialstr=dial;
+		ast_verbose( VERBOSE_PREFIX_3 "Dial Tech,String: (%s,%s)\n", dialtech,dialstr);
 	}
 
-	ast_verb(3, "Return Context: (%s,%s,%d) ID: %s\n", chan->context, chan->exten, chan->priority, chan->cid.cid_num);
-		if (!ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
-		ast_verb(3, "Warning: Return Context Invalid, call will return to default|s\n");
+	return_context = s;
+  
+	if(return_context != NULL) {
+		/* set the return context. Code borrowed from the Goto builtin */
+    
+		working = return_context;
+		context = strsep(&working, "|");
+		exten = strsep(&working, "|");
+		if(!exten) {
+			/* Only a priority in this one */
+			priority = context;
+			exten = NULL;
+			context = NULL;
+		} else {
+			priority = strsep(&working, "|");
+			if(!priority) {
+				/* Only an extension and priority in this one */
+				priority = exten;
+				exten = context;
+				context = NULL;
 		}
+	}
+	if(atoi(priority) < 0) {
+		ast_log(LOG_WARNING, "Priority '%s' must be a number > 0\n", priority);
+		free(orig_s);
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	/* At this point we have a priority and maybe an extension and a context */
+	chan->priority = atoi(priority);
+	if(exten && strcasecmp(exten, "BYEXTENSION"))
+		strncpy(chan->exten, exten, sizeof(chan->exten)-1);
+	if(context)
+		strncpy(chan->context, context, sizeof(chan->context)-1);
+	} else {  /* increment the priority by default*/
+		chan->priority++;
+	}
 
+	if(option_verbose > 2) {
+		ast_verbose( VERBOSE_PREFIX_3 "Return Context: (%s,%s,%d) ID: %s\n", chan->context,chan->exten, chan->priority, chan->cid.cid_num);
+		if(!ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
+			ast_verbose( VERBOSE_PREFIX_3 "Warning: Return Context Invalid, call will return to default|s\n");
+		}
+	}
+  
 	/* we are using masq_park here to protect * from touching the channel once we park it.  If the channel comes out of timeout
 	before we are done announcing and the channel is messed with, Kablooeee.  So we use Masq to prevent this.  */
 
-	res = ast_masq_park_call(chan, NULL, timeout, &lot);
-	if (res == -1)
-		return res;
+	ast_masq_park_call(chan, NULL, timeout, &lot);
 
-	ast_verb(3, "Call Parking Called, lot: %d, timeout: %d, context: %s\n", lot, timeout, args.return_context);
+	res=-1; 
 
-	/* Now place the call to the extension */
+	ast_verbose( VERBOSE_PREFIX_3 "Call Parking Called, lot: %d, timeout: %d, context: %s\n", lot, timeout, return_context);
 
-	snprintf(buf, sizeof(buf), "%d", lot);
+	/* Now place the call to the extention */
+
+	memset(&oh, 0, sizeof(oh));
 	oh.parent_channel = chan;
-	oh.vars = ast_variable_new("_PARKEDAT", buf, "");
-	dchan = __ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, args.dial, 30000, &outstate, chan->cid.cid_num, chan->cid.cid_name, &oh);
+	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) {
-			ast_verb(4, "Channel %s was answered.\n", dchan->name);
+	if(dchan) {
+		if(dchan->_state == AST_STATE_UP) {
+			if(option_verbose > 3)
+				ast_verbose(VERBOSE_PREFIX_4 "Channel %s was answered.\n", dchan->name);
 		} else {
-			ast_verb(4, "Channel %s was never answered.\n", dchan->name);
+			if(option_verbose > 3)
+				ast_verbose(VERBOSE_PREFIX_4 "Channel %s was never answered.\n", dchan->name);
         			ast_log(LOG_WARNING, "PARK: Channel %s was never answered for the announce.\n", dchan->name);
 			ast_hangup(dchan);
+			free(orig_s);
+			LOCAL_USER_REMOVE(u);
 			return -1;
 		}
 	} else {
 		ast_log(LOG_WARNING, "PARK: Unable to allocate announce channel.\n");
+		free(orig_s);
+		LOCAL_USER_REMOVE(u);
 		return -1; 
 	}
 
@@ -146,22 +209,24 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
 
 	/* now we have the call placed and are ready to play stuff to it */
 
-	ast_verb(4, "Announce Template:%s\n", args.template);
+	ast_verbose(VERBOSE_PREFIX_4 "Announce Template:%s\n", template);
 
-	for (looptemp = 0; looptemp < ARRAY_LEN(tmp); looptemp++) {
-		if ((tmp[looptemp] = strsep(&args.template, ":")) != NULL)
-			continue;
-		else
-			break;
+	tpl_working = template;
+	tpl_current=strsep(&tpl_working, ":");
+
+	while(tpl_current && looptemp < sizeof(tmp)) {
+		tmp[looptemp]=tpl_current;
+		looptemp++;
+		tpl_current=strsep(&tpl_working,":");
 	}
 
-	for (i = 0; i < looptemp; i++) {
-		ast_verb(4, "Announce:%s\n", tmp[i]);
-		if (!strcmp(tmp[i], "PARKED")) {
+	for(i=0; i<looptemp; i++) {
+		ast_verbose(VERBOSE_PREFIX_4 "Announce:%s\n", tmp[i]);
+		if(!strcmp(tmp[i], "PARKED")) {
 			ast_say_digits(dchan, lot, "", dchan->language);
 		} else {
 			dres = ast_streamfile(dchan, tmp[i], dchan->language);
-			if (!dres) {
+			if(!dres) {
 				dres = ast_waitstream(dchan, "");
 			} else {
 				ast_log(LOG_WARNING, "ast_streamfile of %s failed on %s\n", tmp[i], dchan->name);
@@ -172,19 +237,45 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
 
 	ast_stopstream(dchan);  
 	ast_hangup(dchan);
+	free(orig_s);
+	
+	LOCAL_USER_REMOVE(u);
 	
 	return res;
 }
 
-static int unload_module(void)
+
+
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	/* return ast_register_application(app, park_exec); */
 	return ast_register_application(app, parkandannounce_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Call Parking and Announce Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_pickupchan.c

@@ -1,174 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2008, Gary Cook
- *
- * 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 Pickup a ringing channel
- *
- * \author Gary Cook
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#include <unistd.h>
-
-#include "asterisk/file.h"
-#include "asterisk/logger.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/lock.h"
-#include "asterisk/app.h"
-#include "asterisk/options.h"
-
-static const char *app = "PickupChan";
-static const char *synopsis = "Pickup a ringing channel";
-static const char *descrip =
-"  PickupChan(channel[&channel...]): This application can pickup any ringing channel\n";
-
-/*! \todo This application should return a result code, like PICKUPRESULT */
-
-/*! \brief Helper function that determines whether a channel is capable of being picked up */
-static int can_pickup(struct ast_channel *chan)
-{
-	ast_debug(3, "Checking Pickup '%s' state '%s ( %d )'\n", chan->name, ast_state2str(chan->_state), chan->_state);
-
-	if (!chan->pbx && (chan->_state == AST_STATE_RINGING || chan->_state == AST_STATE_RING)) {
-		return 1;
-	} else {
-		return 0;
-	}
-}
-
-/*! \brief Helper Function to walk through ALL channels checking NAME and STATE */
-static struct ast_channel *my_ast_get_channel_by_name_locked(char *channame)
-{
-	struct ast_channel *chan;
-	char *chkchan = alloca(strlen(channame) + 2);
-
-	/* need to append a '-' for the comparison so we check full channel name,
-	 * i.e SIP/hgc- , use a temporary variable so original stays the same for
-	 * debugging.
-	 */
-	strcpy(chkchan, channame);
-	strcat(chkchan, "-");
-
-	for (chan = ast_walk_channel_by_name_prefix_locked(NULL, channame, strlen(channame));
-		 chan;
-		 chan = ast_walk_channel_by_name_prefix_locked(chan, channame, strlen(channame))) {
-		if (!strncasecmp(chan->name, chkchan, strlen(chkchan)) && can_pickup(chan))
-			return chan;
-		ast_channel_unlock(chan);
-	}
-	return NULL;
-}
-
-/*! \brief Perform actual pickup between two channels */
-static int pickup_do(struct ast_channel *chan, struct ast_channel *target)
-{
-	int res = 0;
-
-	ast_debug(3, "Call pickup on '%s' by '%s'\n", target->name, chan->name);
-
-	if ((res = ast_answer(chan))) {
-		ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan->name);
-		return -1;
-	}
-
-	if ((res = ast_queue_control(chan, AST_CONTROL_ANSWER))) {
-		ast_log(LOG_WARNING, "Unable to queue answer on '%s'\n", chan->name);
-		return -1;
-	}
-
-	if ((res = ast_channel_masquerade(target, chan))) {
-		ast_log(LOG_WARNING, "Unable to masquerade '%s' into '%s'\n", chan->name, target->name);
-		return -1;
-	}
-
-	return res;
-}
-
-/*! \brief Attempt to pick up specified channel named , does not use context */
-static int pickup_by_channel(struct ast_channel *chan, char *pickup)
-{
-	int res = 0;
-	struct ast_channel *target;
-
-	if (!(target = my_ast_get_channel_by_name_locked(pickup)))
-		return -1;
-
-	/* Just check that we are not picking up the SAME as target */
-	if (chan->name != target->name && chan != target) {
-		res = pickup_do(chan, target);
-		ast_channel_unlock(target);
-	}
-
-	return res;
-}
-
-/*! \brief Main application entry point */
-static int pickupchan_exec(struct ast_channel *chan, void *data)
-{
-	int res = 0;
-	struct ast_module_user *u = NULL;
-	char *tmp = ast_strdupa(data);
-	char *pickup = NULL, *context = NULL;
-
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "Pickup requires an argument (channel)!\n");
-		return -1;	
-	}
-
-	u = ast_module_user_add(chan);
-
-	/* Parse channel (and ignore context if there) */
-	while (!ast_strlen_zero(tmp) && (pickup = strsep(&tmp, "&"))) {
-		if ((context = strchr(pickup , '@'))) {
-			*context++ = '\0';
-		}
-		if (!strncasecmp(chan->name, pickup , strlen(pickup))) {
-			ast_log(LOG_NOTICE, "Cannot pickup your own channel %s.\n", pickup);
-		} else {
-			if (!pickup_by_channel(chan, pickup)) {
-				break;
-			}
-			ast_log(LOG_NOTICE, "No target channel found for %s.\n", pickup);
-		}
-	}
-
-	ast_module_user_remove(u);
-
-	return res;
-}
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app, pickupchan_exec, synopsis, descrip);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Channel Pickup Application");

apps/app_playback.c

@@ -19,388 +19,64 @@
 /*! \file
  *
  * \brief Trivial application to playback a sound file
- *
- * \author Mark Spencer <markster@digium.com>
  * 
  * \ingroup applications
  */
  
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/utils.h"
+#include "asterisk/options.h"
 #include "asterisk/app.h"
-/* This file provides config-file based 'say' functions, and implenents
- * some CLI commands.
- */
-#include "asterisk/say.h"	/* provides config-file based 'say' functions */
-#include "asterisk/cli.h"
+
+static char *tdesc = "Sound File Playback Application";
 
 static char *app = "Playback";
 
 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 comma.\n"
-"The 'skip' option causes the playback of the message to be skipped if the channel\n"
+"  Playback(filename[&filename2...][|option]):  Plays back given filenames (do not put\n"
+"extension). Options may also be included following a pipe symbol. The 'skip'\n"
+"option causes the playback of the message to be skipped if the channel\n"
 "is not in the 'up' state (i.e. it hasn't been  answered  yet). If 'skip' is \n"
 "specified, the application will return immediately should the channel not be\n"
 "off hook.  Otherwise, unless 'noanswer' is specified, the channel will\n"
 "be answered before the sound is played. Not all channels support playing\n"
-"messages while still on hook.\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"
-"See Also: Background (application) -- for playing soundfiles that are interruptible\n"
-"          WaitExten (application) -- wait for digits from caller, optionally play music on hold\n"
 ;
 
+STANDARD_LOCAL_USER;
 
-static struct ast_config *say_cfg = NULL;
-/* save the say' api calls.
- * The first entry is NULL if we have the standard source,
- * otherwise we are sourcing from here.
- * 'say load [new|old]' will enable the new or old method, or report status
- */
-static const void *say_api_buf[40];
-static const char *say_old = "old";
-static const char *say_new = "new";
-
-static void save_say_mode(const void *arg)
-{
-	int i = 0;
-	say_api_buf[i++] = arg;
-
-        say_api_buf[i++] = ast_say_number_full;
-        say_api_buf[i++] = ast_say_enumeration_full;
-        say_api_buf[i++] = ast_say_digit_str_full;
-        say_api_buf[i++] = ast_say_character_str_full;
-        say_api_buf[i++] = ast_say_phonetic_str_full;
-        say_api_buf[i++] = ast_say_datetime;
-        say_api_buf[i++] = ast_say_time;
-        say_api_buf[i++] = ast_say_date;
-        say_api_buf[i++] = ast_say_datetime_from_now;
-        say_api_buf[i++] = ast_say_date_with_format;
-}
-
-static void restore_say_mode(void *arg)
-{
-	int i = 0;
-	say_api_buf[i++] = arg;
-
-        ast_say_number_full = say_api_buf[i++];
-        ast_say_enumeration_full = say_api_buf[i++];
-        ast_say_digit_str_full = say_api_buf[i++];
-        ast_say_character_str_full = say_api_buf[i++];
-        ast_say_phonetic_str_full = say_api_buf[i++];
-        ast_say_datetime = say_api_buf[i++];
-        ast_say_time = say_api_buf[i++];
-        ast_say_date = say_api_buf[i++];
-        ast_say_datetime_from_now = say_api_buf[i++];
-        ast_say_date_with_format = say_api_buf[i++];
-}
-
-/* 
- * Typical 'say' arguments in addition to the date or number or string
- * to say. We do not include 'options' because they may be different
- * in recursive calls, and so they are better left as an external
- * parameter.
- */
-typedef struct {
-        struct ast_channel *chan;
-        const char *ints;
-        const char *language;
-        int audiofd;
-        int ctrlfd;
-} say_args_t;
-
-static int s_streamwait3(const say_args_t *a, const char *fn)
-{
-        int res = ast_streamfile(a->chan, fn, a->language);
-        if (res) {
-                ast_log(LOG_WARNING, "Unable to play message %s\n", fn);
-                return res;
-        }
-        res = (a->audiofd  > -1 && a->ctrlfd > -1) ?
-                ast_waitstream_full(a->chan, a->ints, a->audiofd, a->ctrlfd) :
-                ast_waitstream(a->chan, a->ints);
-        ast_stopstream(a->chan);
-        return res;  
-}
-
-/*
- * the string is 'prefix:data' or prefix:fmt:data'
- * with ':' being invalid in strings.
- */
-static int do_say(say_args_t *a, const char *s, const char *options, int depth)
-{
-	struct ast_variable *v;
-	char *lang, *x, *rule = NULL;
-	int ret = 0;   
-	struct varshead head = { .first = NULL, .last = NULL };
-	struct ast_var_t *n;
-
-	ast_debug(2, "string <%s> depth <%d>\n", s, depth);
-	if (depth++ > 10) {
-		ast_log(LOG_WARNING, "recursion too deep, exiting\n");
-		return -1;
-	} else if (!say_cfg) {
-		ast_log(LOG_WARNING, "no say.conf, cannot spell '%s'\n", s);
-		return -1;
-	}
-
-	/* scan languages same as in file.c */
-	if (a->language == NULL)
-		a->language = "en";     /* default */
-	ast_debug(2, "try <%s> in <%s>\n", s, a->language);
-	lang = ast_strdupa(a->language);
-	for (;;) {
-		for (v = ast_variable_browse(say_cfg, lang); v ; v = v->next) {
-			if (ast_extension_match(v->name, s)) {
-				rule = ast_strdupa(v->value);
-				break;
-			}
-		}
-		if (rule)
-			break;
-		if ( (x = strchr(lang, '_')) )
-			*x = '\0';      /* try without suffix */
-		else if (strcmp(lang, "en"))
-			lang = "en";    /* last resort, try 'en' if not done yet */
-		else
-			break;
-	}
-	if (!rule)
-		return 0;
-
-	/* skip up to two prefixes to get the value */
-	if ( (x = strchr(s, ':')) )
-		s = x + 1;
-	if ( (x = strchr(s, ':')) )
-		s = x + 1;
-	ast_debug(2, "value is <%s>\n", s);
-	n = ast_var_assign("SAY", s);
-	AST_LIST_INSERT_HEAD(&head, n, entries);
-
-	/* scan the body, one piece at a time */
-	while ( !ret && (x = strsep(&rule, ",")) ) { /* exit on key */
-		char fn[128];
-		const char *p, *fmt, *data; /* format and data pointers */
-
-		/* prepare a decent file name */
-		x = ast_skip_blanks(x);
-		ast_trim_blanks(x);
-
-		/* replace variables */
-		pbx_substitute_variables_varshead(&head, x, fn, sizeof(fn));
-		ast_debug(2, "doing [%s]\n", fn);
-
-		/* locate prefix and data, if any */
-		fmt = strchr(fn, ':');
-		if (!fmt || fmt == fn)	{	/* regular filename */
-			ret = s_streamwait3(a, fn);
-			continue;
-		}
-		fmt++;
-		data = strchr(fmt, ':');	/* colon before data */
-		if (!data || data == fmt) {	/* simple prefix-fmt */
-			ret = do_say(a, fn, options, depth);
-			continue;
-		}
-		/* prefix:fmt:data */
-		for (p = fmt; p < data && ret <= 0; p++) {
-			char fn2[sizeof(fn)];
-			if (*p == ' ' || *p == '\t')	/* skip blanks */
-				continue;
-			if (*p == '\'') {/* file name - we trim them */
-				char *y;
-				strcpy(fn2, ast_skip_blanks(p+1));	/* make a full copy */
-				y = strchr(fn2, '\'');
-				if (!y) {
-					p = data;	/* invalid. prepare to end */
-					break;
-				}
-				*y = '\0';
-				ast_trim_blanks(fn2);
-				p = strchr(p+1, '\'');
-				ret = s_streamwait3(a, fn2);
-			} else {
-				int l = fmt-fn;
-				strcpy(fn2, fn); /* copy everything */
-				/* after prefix, append the format */
-				fn2[l++] = *p;
-				strcpy(fn2 + l, data);
-				ret = do_say(a, fn2, options, depth);
-			}
-			
-			if (ret) {
-				break;
-			}
-		}
-	}
-	ast_var_delete(n);
-	return ret;
-}
-
-static int say_full(struct ast_channel *chan, const char *string,
-        const char *ints, const char *lang, const char *options,
-        int audiofd, int ctrlfd)
-{
-        say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
-        return do_say(&a, string, options, 0);
-}
-
-static int say_number_full(struct ast_channel *chan, int num,
-	const char *ints, const char *lang, const char *options,
-	int audiofd, int ctrlfd)
-{
-	char buf[64];
-        say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
-	snprintf(buf, sizeof(buf), "num:%d", num);
-        return do_say(&a, buf, options, 0);
-}
-
-static int say_enumeration_full(struct ast_channel *chan, int num,
-	const char *ints, const char *lang, const char *options,
-	int audiofd, int ctrlfd)
-{
-	char buf[64];
-        say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
-	snprintf(buf, sizeof(buf), "enum:%d", num);
-        return do_say(&a, buf, options, 0);
-}
-
-static int say_date_generic(struct ast_channel *chan, time_t t,
-	const char *ints, const char *lang, const char *format, const char *timezone, const char *prefix)
-{
-	char buf[128];
-	struct ast_tm tm;
-	struct timeval tv = { t, 0 };
-        say_args_t a = { chan, ints, lang, -1, -1 };
-	if (format == NULL)
-		format = "";
-
-	ast_localtime(&tv, &tm, NULL);
-	snprintf(buf, sizeof(buf), "%s:%s:%04d%02d%02d%02d%02d.%02d-%d-%3d",
-		prefix,
-		format,
-		tm.tm_year+1900,
-		tm.tm_mon+1,
-		tm.tm_mday,
-		tm.tm_hour,
-		tm.tm_min,
-		tm.tm_sec,
-		tm.tm_wday,
-		tm.tm_yday);
-        return do_say(&a, buf, NULL, 0);
-}
-
-static int say_date_with_format(struct ast_channel *chan, time_t t,
-	const char *ints, const char *lang, const char *format, const char *timezone)
-{
-	return say_date_generic(chan, t, ints, lang, format, timezone, "datetime");
-}
-
-static int say_date(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
-{
-	return say_date_generic(chan, t, ints, lang, "", NULL, "date");
-}
-
-static int say_time(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
-{
-	return say_date_generic(chan, t, ints, lang, "", NULL, "time");
-}
-
-static int say_datetime(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
-{
-	return say_date_generic(chan, t, ints, lang, "", NULL, "datetime");
-}
-
-/*
- * remap the 'say' functions to use those in this file
- */
-static int say_init_mode(const char *mode) {
-	if (!strcmp(mode, say_new)) {
-		if (say_cfg == NULL) {
-			ast_log(LOG_ERROR, "There is no say.conf file to use new mode\n");
-			return -1;
-		}
-		save_say_mode(say_new);
-		ast_say_number_full = say_number_full;
-
-		ast_say_enumeration_full = say_enumeration_full;
-#if 0
-		ast_say_digits_full = say_digits_full;
-		ast_say_digit_str_full = say_digit_str_full;
-		ast_say_character_str_full = say_character_str_full;
-		ast_say_phonetic_str_full = say_phonetic_str_full;
-		ast_say_datetime_from_now = say_datetime_from_now;
-#endif
-		ast_say_datetime = say_datetime;
-		ast_say_time = say_time;
-		ast_say_date = say_date;
-		ast_say_date_with_format = say_date_with_format;
-	} else if (!strcmp(mode, say_old) && say_api_buf[0] == say_new) {
-		restore_say_mode(NULL);
-	} else if (strcmp(mode, say_old)) {
-		ast_log(LOG_WARNING, "unrecognized mode %s\n", mode);
-		return -1;
-	}
-	
-	return 0;
-}
-
-static char *__say_cli_init(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
-	const char *old_mode = say_api_buf[0] ? say_new : say_old;
-	char *mode;
-	switch (cmd) {
-	case CLI_INIT:
-		e->command = "say load [new|old]";
-		e->usage = 
-			"Usage: say load [new|old]\n"
-			"       say load\n"
-			"           Report status of current say mode\n"
-			"       say load new\n"
-			"           Set say method, configured in say.conf\n"
-			"       say load old\n"
-			"           Set old say method, coded in asterisk core\n";
-		return NULL;
-	case CLI_GENERATE:
-		return NULL;
-	}
-	if (a->argc == 2) {
-		ast_cli(a->fd, "say mode is [%s]\n", old_mode);
-		return CLI_SUCCESS;
-	} else if (a->argc != e->args)
-		return CLI_SHOWUSAGE;
-	mode = a->argv[2];
-	if (!strcmp(mode, old_mode))
-		ast_cli(a->fd, "say mode is %s already\n", mode);
-	else
-		if (say_init_mode(mode) == 0)
-			ast_cli(a->fd, "setting say mode from %s to %s\n", old_mode, mode);
-
-	return CLI_SUCCESS;
-}
-
-static struct ast_cli_entry cli_playback[] = {
-	AST_CLI_DEFINE(__say_cli_init, "Set or show the say mode"),
-};
+LOCAL_USER_DECL;
 
 static int playback_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
-	int mres = 0;
-	char *tmp;
+	int res = 0, mres = 0;
+	struct localuser *u;
+	char *tmp = NULL;
 	int option_skip=0;
-	int option_say=0;
 	int option_noanswer = 0;
-
+	char *front = NULL, *back = NULL;
+	int priority_jump = 0;
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(filenames);
 		AST_APP_ARG(options);
@@ -411,117 +87,95 @@ static int playback_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 
+	LOCAL_USER_ADD(u);
+
 	tmp = ast_strdupa(data);
+	if (!tmp) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;	
+	}
+
 	AST_STANDARD_APP_ARGS(args, tmp);
 
 	if (args.options) {
 		if (strcasestr(args.options, "skip"))
 			option_skip = 1;
-		if (strcasestr(args.options, "say"))
-			option_say = 1;
 		if (strcasestr(args.options, "noanswer"))
 			option_noanswer = 1;
-	} 
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+	
 	if (chan->_state != AST_STATE_UP) {
 		if (option_skip) {
 			/* At the user's option, skip if the line is not up */
-			goto done;
-		} else if (!option_noanswer) {
+			pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "SUCCESS");
+			LOCAL_USER_REMOVE(u);
+			return 0;
+		} else if (!option_noanswer)
 			/* Otherwise answer unless we're supposed to send this while on-hook */
 			res = ast_answer(chan);
-		}
 	}
 	if (!res) {
-		char *back = args.filenames;
-		char *front;
-
 		ast_stopstream(chan);
-		while (!res && (front = strsep(&back, "&"))) {
-			if (option_say)
-				res = say_full(chan, front, "", chan->language, NULL, -1, -1);
-			else
-				res = ast_streamfile(chan, front, chan->language);
+		front = tmp;
+		while (!res && front) {
+			if ((back = strchr(front, '&'))) {
+				*back = '\0';
+				back++;
+			}
+			res = ast_streamfile(chan, front, chan->language);
 			if (!res) { 
 				res = ast_waitstream(chan, "");	
 				ast_stopstream(chan);
 			} else {
 				ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
+				if (priority_jump || option_priority_jumping)
+					ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
 				res = 0;
 				mres = 1;
 			}
+			front = back;
 		}
 	}
-done:
-	pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", mres ? "FAILED" : "SUCCESS");
+	if (mres)
+		pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "FAILED");
+	else
+		pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "SUCCESS");
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int reload(void)
-{
-	struct ast_variable *v;
-	struct ast_flags config_flags = { CONFIG_FLAG_FILEUNCHANGED };
-	struct ast_config *newcfg;
-
-	if ((newcfg = ast_config_load("say.conf", config_flags)) == CONFIG_STATUS_FILEUNCHANGED)
-		return 0;
-
-	if (say_cfg) {
-		ast_config_destroy(say_cfg);
-		ast_log(LOG_NOTICE, "Reloading say.conf\n");
-		say_cfg = newcfg;
-	}
-
-	if (say_cfg) {
-		for (v = ast_variable_browse(say_cfg, "general"); v ; v = v->next) {
-    			if (ast_extension_match(v->name, "mode")) {
-				say_init_mode(v->value);
-				break;
-			}
-		}
-	}
-	
-	/*
-	 * XXX here we should sort rules according to the same order
-	 * we have in pbx.c so we have the same matching behaviour.
-	 */
-	return 0;
-}
-
-static int unload_module(void)
+int unload_module(void)
 {
 	int res;
 
 	res = ast_unregister_application(app);
 
-	ast_cli_unregister_multiple(cli_playback, sizeof(cli_playback) / sizeof(struct ast_cli_entry));
-
-	if (say_cfg)
-		ast_config_destroy(say_cfg);
+	STANDARD_HANGUP_LOCALUSERS;
 
 	return res;	
 }
 
-static int load_module(void)
+int load_module(void)
 {
-	struct ast_variable *v;
-	struct ast_flags config_flags = { 0 };
-
-	say_cfg = ast_config_load("say.conf", config_flags);
-	if (say_cfg) {
-		for (v = ast_variable_browse(say_cfg, "general"); v ; v = v->next) {
-    			if (ast_extension_match(v->name, "mode")) {
-				say_init_mode(v->value);
-				break;
-			}
-		}
-	}
-
-        ast_cli_register_multiple(cli_playback, sizeof(cli_playback) / sizeof(struct ast_cli_entry));
 	return ast_register_application(app, playback_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Sound File Playback Application",
-		.load = load_module,
-		.unload = unload_module,
-		.reload = reload,
-	       );
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_privacy.c

@@ -19,12 +19,14 @@
 /*! \file
  *
  * \brief Block all calls without Caller*ID, require phone # to be entered
- *
- * \author Mark Spencer <markster@digium.com>
  * 
  * \ingroup applications
  */
 
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
@@ -32,6 +34,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
 #include "asterisk/utils.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
@@ -41,24 +45,40 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/app.h"
 #include "asterisk/config.h"
 
+#define PRIV_CONFIG "privacy.conf"
+
+static char *tdesc = "Require phone number to be entered, if no CallerID sent";
+
 static char *app = "PrivacyManager";
 
 static char *synopsis = "Require phone number to be entered, if no CallerID sent";
 
 static char *descrip =
-  "  PrivacyManager([maxretries][,minlength]): If no Caller*ID \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 'maxretries' attempts to do so.\n"
+  "enter their phone number. The caller is given 3 attempts to do so.\n"
   "The application does nothing if Caller*ID was received on the channel.\n"
+  "  Configuration file privacy.conf contains two variables:\n"
   "   maxretries  default 3  -maximum number of attempts the caller is allowed \n"
   "               to input a callerid.\n"
   "   minlength   default 10 -minimum allowable digits in the input callerid number.\n"
+  "If you don't want to use the config file and have an i/o operation with\n"
+  "every call, you can also specify maxretries and minlength as application\n"
+  "parameters. Doing so supercedes any values set in privacy.conf.\n"
+  "The option string may contain the following character: \n"
+  "  'j' -- jump to n+101 priority after <maxretries> failed attempts to collect\n"
+  "         the minlength number of digits.\n"
   "The application sets the following channel variable upon completion: \n"
   "PRIVACYMGRSTATUS  The status of the privacy manager's attempt to collect \n"
   "                  a phone number from the user. A text string that is either:\n" 
   "          SUCCESS | FAILED \n"
 ;
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+
 
 static int privacy_exec (struct ast_channel *chan, void *data)
 {
@@ -67,51 +87,89 @@ static int privacy_exec (struct ast_channel *chan, void *data)
 	int maxretries = 3;
 	int minlength = 10;
 	int x = 0;
+	char *s;
 	char phone[30];
+	struct localuser *u;
+	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)) {
-		ast_verb(3, "CallerID Present: Skipping\n");
+		if (option_verbose > 2)
+			ast_verbose (VERBOSE_PREFIX_3 "CallerID Present: Skipping\n");
 	} else {
 		/*Answer the channel if it is not already*/
 		if (chan->_state != AST_STATE_UP) {
-			if ((res = ast_answer(chan)))
+			res = ast_answer(chan);
+			if (res) {
+				LOCAL_USER_REMOVE(u);
 				return -1;
+			}
 		}
 
-		if (!ast_strlen_zero(data)) {
+		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, "%30d", &x) == 1)
+				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, "%30d", &x) == 1)
+				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;
 
 		}		
 
-		/* Play unidentified call */
+		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);
 		if (!res)
 			res = ast_streamfile(chan, "privacy-unident", chan->language);
 		if (!res)
 			res = ast_waitstream(chan, "");
 
-		/* Ask for 10 digit number, give 3 attempts */
+		/*Ask for 10 digit number, give 3 attempts*/
 		for (retries = 0; retries < maxretries; retries++) {
 			if (!res)
 				res = ast_streamfile(chan, "privacy-prompt", chan->language);
@@ -124,7 +182,7 @@ static int privacy_exec (struct ast_channel *chan, void *data)
 			if (res < 0)
 				break;
 
-			/* Make sure we get at least digits */
+			/*Make sure we get at least digits*/
 			if (strlen(phone) >= minlength ) 
 				break;
 			else {
@@ -134,7 +192,7 @@ static int privacy_exec (struct ast_channel *chan, void *data)
 			}
 		}
 		
-		/* Got a number, play sounds and send them on their way */
+		/*Got a number, play sounds and send them on their way*/
 		if ((retries < maxretries) && res >= 0 ) {
 			res = ast_streamfile(chan, "privacy-thankyou", chan->language);
 			if (!res)
@@ -147,25 +205,58 @@ static int privacy_exec (struct ast_channel *chan, void *data)
 			 */
 			chan->cid.cid_pres &= (AST_PRES_UNAVAILABLE ^ 0xFF);
 
-			ast_verb(3, "Changed Caller*ID to %s, callerpres to %d\n",phone,chan->cid.cid_pres);
-
+			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 {
+			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);
 	}
 
-	return 0;
+  LOCAL_USER_REMOVE (u);
+  return 0;
 }
 
-static int unload_module(void)
+int
+unload_module (void)
 {
-	return ast_unregister_application (app);
+	int res;
+
+	res = ast_unregister_application (app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }
 
-static int load_module(void)
+int
+load_module (void)
 {
-	return ast_register_application(app, privacy_exec, synopsis, descrip);
+  return ast_register_application (app, privacy_exec, synopsis,
+				   descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Require phone number to be entered, if no CallerID sent");
+char *
+description (void)
+{
+  return tdesc;
+}
+
+int
+usecount (void)
+{
+  int res;
+  STANDARD_USECOUNT (res);
+  return res;
+}
+
+char *
+key ()
+{
+  return ASTERISK_GPL_KEY;
+}

apps/app_random.c

@@ -0,0 +1,126 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (c) 2003 - 2005 Tilghman Lesher.  All rights reserved.
+ *
+ * Tilghman Lesher <asterisk__app_random__200508@the-tilghman.com>
+ *
+ * This code is released by the author with no restrictions on usage or distribution.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ *
+ * \brief Random application
+ *
+ * \author Tilghman Lesher <asterisk__app_random__200508@the-tilghman.com>
+ * \ingroup applications
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+
+static char *tdesc = "Random goto";
+
+static char *app_random = "Random";
+
+static char *random_synopsis = "Conditionally branches, based upon a probability";
+
+static char *random_descrip =
+"Random([probability]:[[context|]extension|]priority)\n"
+"  probability := INTEGER in the range 1 to 100\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static char random_state[256];
+
+static int random_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+
+	char *s;
+	char *prob;
+	int probint;
+	
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "Random requires an argument ([probability]:[[context|]extension|]priority)\n");
+		return -1;
+	}
+	
+	LOCAL_USER_ADD(u);
+
+	s = ast_strdupa(data);
+	if (!s) {
+		ast_log(LOG_ERROR, "Out of memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	prob = strsep(&s,":");
+	if ((!prob) || (sscanf(prob, "%d", &probint) != 1))
+		probint = 0;
+
+	if ((random() % 100) + probint >= 100) {
+		res = ast_parseable_goto(chan, s);
+		if (option_verbose > 2)
+			ast_verbose( VERBOSE_PREFIX_3 "Random branches to (%s,%s,%d)\n",
+				chan->context,chan->exten, chan->priority+1);
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+	
+	res = ast_unregister_application(app_random);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
+}
+
+int load_module(void)
+{
+	initstate((getppid() * 65535 + getpid()) % RAND_MAX, random_state, 256);
+	return ast_register_application(app_random, random_exec, random_synopsis, random_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	/* Don't allow unload, since rand(3) depends upon this module being here. */
+	return 1;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_read.c

@@ -19,213 +19,217 @@
 /*! \file
  *
  * \brief Trivial application to read a variable
- *
- * \author Mark Spencer <markster@digium.com>
  * 
  * \ingroup applications
  */
  
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/lock.h"
 #include "asterisk/file.h"
-#include "asterisk/pbx.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
+#include "asterisk/pbx.h"
 #include "asterisk/app.h"
 #include "asterisk/module.h"
-#include "asterisk/indications.h"
+#include "asterisk/translate.h"
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
 
-enum {
-	OPT_SKIP = (1 << 0),
-	OPT_INDICATION = (1 << 1),
-	OPT_NOANSWER = (1 << 2),
-} read_option_flags;
-
-AST_APP_OPTIONS(read_app_options, {
-	AST_APP_OPTION('s', OPT_SKIP),
-	AST_APP_OPTION('i', OPT_INDICATION),
-	AST_APP_OPTION('n', OPT_NOANSWER),
-});
+static char *tdesc = "Read Variable Application";
 
 static char *app = "Read";
 
 static char *synopsis = "Read a variable";
 
 static char *descrip = 
-"  Read(variable[,filename[&filename2...]][,maxdigits][,option][,attempts][,timeout])\n\n"
+"  Read(variable[|filename][|maxdigits][|option][|attempts][|timeout])\n\n"
 "Reads a #-terminated string of digits a certain number of times from the\n"
 "user in to the given variable.\n"
-"  filename   -- file(s) to play before reading digits or tone with option i\n"
+"  filename   -- file to play before reading digits.\n"
 "  maxdigits  -- maximum acceptable number of digits. Stops reading after\n"
 "                maxdigits have been entered (without requiring the user to\n"
 "                press the '#' key).\n"
 "                Defaults to 0 - no limit - wait for the user press the '#' key.\n"
 "                Any value below 0 means the same. Max accepted value is 255.\n"
-"  option     -- options are 's' , 'i', 'n'\n"
-"                's' to return immediately if the line is not up,\n"
-"                'i' to play  filename as an indication tone from your indications.conf\n"
-"                'n' to read digits even if the line is not up.\n"
+"  option     -- may be 'skip' to return immediately if the line is not up,\n"
+"                or 'noanswer' to read digits even if the line is not up.\n"
 "  attempts   -- if greater than 1, that many attempts will be made in the \n"
 "                event no data is entered.\n"
-"  timeout    -- The number of seconds to wait for a digit response. If greater\n"
-"                than 0, that value will override the default timeout. Can be floating point.\n"
-"This application sets the following channel variable upon completion:\n"
-"    READSTATUS - This is the status of the read operation.\n"
-"                 Possible values are:\n"
-"                 OK | ERROR | HANGUP | INTERRUPTED | SKIPPED | TIMEOUT\n";
+"  timeout    -- if greater than 0, that value will override the default timeout.\n\n"
+"Read should disconnect if the function fails or errors out.\n";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 #define ast_next_data(instr,ptr,delim) if((ptr=strchr(instr,delim))) { *(ptr) = '\0' ; ptr++;}
 
 static int read_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
-	char tmp[256] = "";
-	int maxdigits = 255;
-	int tries = 1, to = 0, x = 0;
-	double tosec;
+	struct localuser *u;
+	char tmp[256];
+	char *timeout = NULL;
+	char *varname = NULL;
+	char *filename = NULL;
+	char *loops;
+	char *maxdigitstr=NULL;
+	char *options=NULL;
+	int option_skip = 0;
+	int option_noanswer = 0;
+	int maxdigits=255;
+	int tries = 1;
+	int to = 0;
+	int x = 0;
 	char *argcopy = NULL;
-	struct tone_zone_sound *ts = NULL;
-	struct ast_flags flags = {0};
-	const char *status = "ERROR";
+	char *args[8];
 
-	AST_DECLARE_APP_ARGS(arglist,
-		AST_APP_ARG(variable);
-		AST_APP_ARG(filename);
-		AST_APP_ARG(maxdigits);
-		AST_APP_ARG(options);
-		AST_APP_ARG(attempts);
-		AST_APP_ARG(timeout);
-	);
-	
-	pbx_builtin_setvar_helper(chan, "READSTATUS", status);
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Read requires an argument (variable)\n");
-		return 0;
+		return -1;
 	}
+
+	LOCAL_USER_ADD(u);
 	
 	argcopy = ast_strdupa(data);
-
-	AST_STANDARD_APP_ARGS(arglist, argcopy);
-
-	if (!ast_strlen_zero(arglist.options)) {
-		ast_app_parse_options(read_app_options, &flags, NULL, arglist.options);
+	if (!argcopy) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -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;
+	}
+
+	varname = args[x++];
+	filename = args[x++];
+	maxdigitstr = args[x++];
+	options = args[x++];
+	loops = args[x++];
+	timeout = args[x++];
 	
-	if (!ast_strlen_zero(arglist.attempts)) {
-		tries = atoi(arglist.attempts);
-		if (tries <= 0)
+	if (options) { 
+		if (!strcasecmp(options, "skip"))
+			option_skip = 1;
+		else if (!strcasecmp(options, "noanswer"))
+			option_noanswer = 1;
+		else {
+			if (strchr(options, 's'))
+				option_skip = 1;
+			if (strchr(options, 'n'))
+				option_noanswer = 1;
+		}
+	}
+
+	if(loops) {
+		tries = atoi(loops);
+		if(tries <= 0)
 			tries = 1;
 	}
 
-	if (!ast_strlen_zero(arglist.timeout)) {
-		tosec = atof(arglist.timeout);
-		if (tosec <= 0)
+	if(timeout) {
+		to = atoi(timeout);
+		if (to <= 0)
 			to = 0;
 		else
-			to = tosec * 1000.0;
+			to *= 1000;
 	}
 
-	if (ast_strlen_zero(arglist.filename)) {
-		arglist.filename = NULL;
+	if (ast_strlen_zero(filename)) 
+		filename = NULL;
+	if (maxdigitstr) {
+		maxdigits = atoi(maxdigitstr);
+		if ((maxdigits<1) || (maxdigits>255)) {
+    			maxdigits = 255;
+		} else if (option_verbose > 2)
+			ast_verbose(VERBOSE_PREFIX_3 "Accepting a maximum of %d digits.\n", maxdigits);
 	}
-	if (!ast_strlen_zero(arglist.maxdigits)) {
-		maxdigits = atoi(arglist.maxdigits);
-		if ((maxdigits < 1) || (maxdigits > 255)) {
-			maxdigits = 255;
-		} else
-			ast_verb(3, "Accepting a maximum of %d digits.\n", maxdigits);
-	}
-	if (ast_strlen_zero(arglist.variable)) {
-		ast_log(LOG_WARNING, "Invalid! Usage: Read(variable[,filename][,maxdigits][,option][,attempts][,timeout])\n\n");
-		return 0;
-	}
-	if (ast_test_flag(&flags, OPT_INDICATION)) {
-		if (! ast_strlen_zero(arglist.filename)) {
-			ts = ast_get_indication_tone(chan->zone, arglist.filename);
-		}
+	if (ast_strlen_zero(varname)) {
+		ast_log(LOG_WARNING, "Invalid! Usage: Read(variable[|filename][|maxdigits][|option][|attempts][|timeout])\n\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
 	}
+	
 	if (chan->_state != AST_STATE_UP) {
-		if (ast_test_flag(&flags, OPT_SKIP)) {
+		if (option_skip) {
 			/* At the user's option, skip if the line is not up */
-			pbx_builtin_setvar_helper(chan, arglist.variable, "");
-			pbx_builtin_setvar_helper(chan, "READSTATUS", "SKIPPED");
+			pbx_builtin_setvar_helper(chan, varname, "\0");
+			LOCAL_USER_REMOVE(u);
 			return 0;
-		} else if (!ast_test_flag(&flags, OPT_NOANSWER)) {
+		} else if (!option_noanswer) {
 			/* Otherwise answer unless we're supposed to read while on-hook */
 			res = ast_answer(chan);
 		}
 	}
 	if (!res) {
-		while (tries && !res) {
+		while(tries && !res) {
 			ast_stopstream(chan);
-			if (ts && ts->data[0]) {
-				if (!to)
-					to = chan->pbx ? chan->pbx->rtimeout * 1000 : 6000;
-				res = ast_playtones_start(chan, 0, ts->data, 0);
-				for (x = 0; x < maxdigits; ) {
-					res = ast_waitfordigit(chan, to);
-					ast_playtones_stop(chan);
-					if (res < 1) {
-						if (res == 0)
-							status = "TIMEOUT";
-						tmp[x]='\0';
-						break;
-					}
-					tmp[x++] = res;
-					if (tmp[x-1] == '#') {
-						tmp[x-1] = '\0';
-						status = "OK";
-						break;
-					}
-					if (x >= maxdigits) {
-						status = "OK";
-					}
-				}
-			} else {
-				res = ast_app_getdata(chan, arglist.filename, tmp, maxdigits, to);
-				if (res == AST_GETDATA_COMPLETE || res == AST_GETDATA_EMPTY_END_TERMINATED)
-					status = "OK";
-				else if (res == AST_GETDATA_TIMEOUT)
-					status = "TIMEOUT";
-				else if (res == AST_GETDATA_INTERRUPTED)
-					status = "INTERRUPTED";
-			}
+			res = ast_app_getdata(chan, filename, tmp, maxdigits, to);
 			if (res > -1) {
-				pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
+				pbx_builtin_setvar_helper(chan, varname, tmp);
 				if (!ast_strlen_zero(tmp)) {
-					ast_verb(3, "User entered '%s'\n", tmp);
+					if (option_verbose > 2)
+						ast_verbose(VERBOSE_PREFIX_3 "User entered '%s'\n", tmp);
 					tries = 0;
 				} else {
 					tries--;
-					if (tries)
-						ast_verb(3, "User entered nothing, %d chance%s left\n", tries, (tries != 1) ? "s" : "");
-					else
-						ast_verb(3, "User entered nothing.\n");
+					if (option_verbose > 2) {
+						if (tries)
+							ast_verbose(VERBOSE_PREFIX_3 "User entered nothing, %d chance%s left\n", tries, (tries != 1) ? "s" : "");
+						else
+							ast_verbose(VERBOSE_PREFIX_3 "User entered nothing.\n");
+					}
 				}
 				res = 0;
 			} else {
-				pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
-				ast_verb(3, "User disconnected\n");
+				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "User disconnected\n");
 			}
 		}
 	}
-
-	if (ast_check_hangup(chan))
-		status = "HANGUP";
-	pbx_builtin_setvar_helper(chan, "READSTATUS", status);
-	return 0;
+	LOCAL_USER_REMOVE(u);
+	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, read_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Read Variable Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_readexten.c

@@ -1,258 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2007 Dave Chappell
- *
- * David Chappell <David.Chappell@trincoll.edu>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Trivial application to read an extension into a variable
- *
- * \author David Chappell <David.Chappell@trincoll.edu>
- * 
- * \ingroup applications
- */
- 
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/pbx.h"
-#include "asterisk/app.h"
-#include "asterisk/module.h"
-#include "asterisk/indications.h"
-#include "asterisk/channel.h"
-
-enum {
-	OPT_SKIP = (1 << 0),
-	OPT_INDICATION = (1 << 1),
-	OPT_NOANSWER = (1 << 2),
-} readexten_option_flags;
-
-AST_APP_OPTIONS(readexten_app_options, {
-	AST_APP_OPTION('s', OPT_SKIP),
-	AST_APP_OPTION('i', OPT_INDICATION),
-	AST_APP_OPTION('n', OPT_NOANSWER),
-});
-
-static char *app = "ReadExten";
-
-static char *synopsis = "Read an extension into a variable";
-
-static char *descrip = 
-"  ReadExten(<variable>[,[<filename>][,[<context>][,[<option>][,<timeout>]]]])\n\n"
-"Reads a #-terminated string of digits from the user into the given variable.\n"
-"  filename  file to play before reading digits or tone with option i\n"
-"  context   context in which to match extensions\n"
-"  option    options are:\n"
-"              s - Return immediately if the channel is not answered,\n"
-"              i - Play filename as an indication tone from your\n"
-"                  indications.conf\n"
-"              n - Read digits even if the channel is not answered.\n"
-"  timeout   An integer number of seconds to wait for a digit response. If\n"
-"            greater than 0, that value will override the default timeout.\n\n"
-"ReadExten will set READEXTENSTATUS on exit with one of the following statuses:\n"
-"  OK        A valid extension exists in ${variable}\n"
-"  TIMEOUT   No extension was entered in the specified time\n"
-"  INVALID   An invalid extension, ${INVALID_EXTEN}, was entered\n"
-"  SKIP      Line was not up and the option 's' was specified\n"
-"  ERROR     Invalid arguments were passed\n";
-
-
-static int readexten_exec(struct ast_channel *chan, void *data)
-{
-	int res = 0;
-	char exten[256] = "";
-	int maxdigits = sizeof(exten) - 1;
-	int timeout = 0, digit_timeout = 0, x = 0;
-	char *argcopy = NULL, *status = "";
-	struct tone_zone_sound *ts = NULL;
-	struct ast_flags flags = {0};
-
-	 AST_DECLARE_APP_ARGS(arglist,
-		AST_APP_ARG(variable);
-		AST_APP_ARG(filename);
-		AST_APP_ARG(context);
-		AST_APP_ARG(options);
-		AST_APP_ARG(timeout);
-	);
-	
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "ReadExten requires at least one argument\n");
-		pbx_builtin_setvar_helper(chan, "READEXTENSTATUS", "ERROR");
-		return 0;
-	}
-
-	argcopy = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(arglist, argcopy);
-
-	if (ast_strlen_zero(arglist.variable)) {
-		ast_log(LOG_WARNING, "Invalid! Usage: ReadExten(variable[|filename][|context][|options][|timeout])\n\n");
-		pbx_builtin_setvar_helper(chan, "READEXTENSTATUS", "ERROR");
-		return 0;
-	}
-
-	if (ast_strlen_zero(arglist.filename))
-		arglist.filename = NULL;
-
-	if (ast_strlen_zero(arglist.context))
-		arglist.context = chan->context;
-
-	if (!ast_strlen_zero(arglist.options))
-		ast_app_parse_options(readexten_app_options, &flags, NULL, arglist.options);
-
-	if (!ast_strlen_zero(arglist.timeout)) {
-		timeout = atoi(arglist.timeout);
-		if (timeout > 0)
-			timeout *= 1000;
-	}
-
-	if (timeout <= 0)
-		timeout = chan->pbx ? chan->pbx->rtimeout * 1000 : 10000;
-
-	if (digit_timeout <= 0)
-		digit_timeout = chan->pbx ? chan->pbx->dtimeout * 1000 : 5000;
-
-	if (ast_test_flag(&flags, OPT_INDICATION) && !ast_strlen_zero(arglist.filename))
-		ts = ast_get_indication_tone(chan->zone, arglist.filename);
-
-	do {
-		if (chan->_state != AST_STATE_UP) {
-			if (ast_test_flag(&flags, OPT_SKIP)) {
-				/* At the user's option, skip if the line is not up */
-				pbx_builtin_setvar_helper(chan, arglist.variable, "");
-				status = "SKIP";
-				break;
-			} else if (!ast_test_flag(&flags, OPT_NOANSWER)) {
-				/* Otherwise answer unless we're supposed to read while on-hook */
-				res = ast_answer(chan);
-			}
-		}
-
-		if (res < 0) {
-			status = "HANGUP";
-			break;
-		}
-
-		ast_playtones_stop(chan);
-		ast_stopstream(chan);
-
-		if (ts && ts->data[0])
-			res = ast_playtones_start(chan, 0, ts->data, 0);
-		else if (arglist.filename)
-			res = ast_streamfile(chan, arglist.filename, chan->language);
-
-		for (x = 0; x < maxdigits; x++) {
-			ast_debug(3, "extension so far: '%s', timeout: %d\n", exten, timeout);
-			res = ast_waitfordigit(chan, timeout);
-
-			ast_playtones_stop(chan);
-			ast_stopstream(chan);
-			timeout = digit_timeout;
-
-			if (res < 1) {		/* timeout expired or hangup */
-				if (ast_check_hangup(chan))
-					status = "HANGUP";
-				else
-					status = "TIMEOUT";
-				break;
-			} else if (res == '#') {
-				break;
-			}
-
-			exten[x] = res;
-			if (!ast_matchmore_extension(chan, arglist.context, exten, 1 /* priority */, chan->cid.cid_num)) {
-				break;
-			}
-		}
-
-		if (!ast_strlen_zero(status))
-			break;
-
-		if (ast_exists_extension(chan, arglist.context, exten, 1, chan->cid.cid_num)) {
-			ast_debug(3, "User entered valid extension '%s'\n", exten);
-			pbx_builtin_setvar_helper(chan, arglist.variable, exten);
-			status = "OK";
-		} else {
-			ast_debug(3, "User dialed invalid extension '%s' in context '%s' on %s\n", exten, arglist.context, chan->name);
-			pbx_builtin_setvar_helper(chan, "INVALID_EXTEN", exten);
-			status = "INVALID";
-		}
-	} while (0);
-
-	pbx_builtin_setvar_helper(chan, "READEXTENSTATUS", status);
-
-	return status[0] == 'H' ? -1 : 0;
-}
-
-static int acf_isexten_exec(struct ast_channel *chan, const char *cmd, char *parse, char *buffer, size_t buflen)
-{
-	int priority_int;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(context);
-		AST_APP_ARG(extension);
-		AST_APP_ARG(priority);
-	);
-
-	AST_STANDARD_APP_ARGS(args, parse);
-
-	if (ast_strlen_zero(args.context))
-		args.context = chan->context;
-
-	if (ast_strlen_zero(args.extension)) {
-		ast_log(LOG_WARNING, "Syntax: VALID_EXTEN([<context>],<extension>[,<priority>]) - missing argument <extension>!\n");
-		return -1;
-	}
-
-	if (ast_strlen_zero(args.priority))
-		priority_int = 1;
-	else
-		priority_int = atoi(args.priority);
-
-	if (ast_exists_extension(chan, args.context, args.extension, priority_int, chan->cid.cid_num))
-	    ast_copy_string(buffer, "1", buflen);
-	else
-	    ast_copy_string(buffer, "0", buflen);
-
-	return 0;
-}
-
-static struct ast_custom_function acf_isexten = {
-	.name = "VALID_EXTEN",
-	.synopsis = "Determine whether an extension exists or not",
-	.syntax = "VALID_EXTEN([<context>],<extension>[,<priority>])",
-	.desc =
-"Returns a true value if the indicated context, extension, and priority exist.\n"
-"Context defaults to the current context, priority defaults to 1.\n",
-	.read = acf_isexten_exec,
-};
-
-static int unload_module(void)
-{
-	int res = ast_unregister_application(app);
-	res |= ast_custom_function_unregister(&acf_isexten);
-
-	return res;	
-}
-
-static int load_module(void)
-{
-	int res = ast_register_application(app, readexten_exec, synopsis, descrip);
-	res |= ast_custom_function_register(&acf_isexten);
-	return res;
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Read and evaluate extension validity");

apps/app_readfile.c

@@ -20,60 +20,76 @@
  *
  * \brief ReadFile application -- Reads in a File for you.
  *
- * \author Matt O'Gorman <mogorman@digium.com>
- *
  * \ingroup applications
  */
 
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/app.h"
 #include "asterisk/module.h"
 
+static char *tdesc = "Stores output of file into a variable";
+
 static char *app_readfile = "ReadFile";
 
-static char *readfile_synopsis = "Read the contents of a text file into a channel variable";
+static char *readfile_synopsis = "ReadFile(varname=file,length)";
 
 static char *readfile_descrip =
 "ReadFile(varname=file,length)\n"
-"  varname  - Result stored here.\n"
-"  file     - The name of the file to read.\n"
-"  length   - Maximum number of characters to capture.\n";
+"  Varname - Result stored here.\n"
+"  File - The name of the file to read.\n"
+"  Length - Maximum number of characters to capture.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 
 static int readfile_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
+	struct localuser *u;
 	char *s, *varname=NULL, *file=NULL, *length=NULL, *returnvar=NULL;
 	int len=0;
-	static int deprecation_warning = 0;
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "ReadFile require an argument!\n");
 		return -1;
 	}
 
+	LOCAL_USER_ADD(u);
+
 	s = ast_strdupa(data);
+	if (!s) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
 
 	varname = strsep(&s, "=");
-	file = strsep(&s, ",");
+	file = strsep(&s, "|");
 	length = s;
 
-	if (deprecation_warning++ % 10 == 0)
-		ast_log(LOG_WARNING, "ReadFile has been deprecated in favor of Set(%s=${FILE(%s,0,%s)})\n", varname, file, length);
-
 	if (!varname || !file) {
 		ast_log(LOG_ERROR, "No file or variable specified!\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
 
 	if (length) {
-		if ((sscanf(length, "%30d", &len) != 1) || (len < 0)) {
+		if ((sscanf(length, "%d", &len) != 1) || (len < 0)) {
 			ast_log(LOG_WARNING, "%s is not a positive number, defaulting length to max\n", length);
 			len = 0;
 		}
@@ -87,21 +103,42 @@ static int readfile_exec(struct ast_channel *chan, void *data)
 				ast_log(LOG_WARNING, "%s is longer than %d, and %d \n", file, len, (int)strlen(returnvar));
 		}
 		pbx_builtin_setvar_helper(chan, varname, returnvar);
-		ast_free(returnvar);
+		free(returnvar);
 	}
-
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app_readfile);
+	int res;
+
+	res = ast_unregister_application(app_readfile);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app_readfile, readfile_exec, readfile_synopsis, readfile_descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Stores output of file into a variable");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_realtime.c

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

apps/app_record.c

@@ -20,28 +20,37 @@
  *
  * \brief Trivial application to record a sound file
  *
- * \author Matthew Fredrickson <creslin@digium.com>
- *
  * \ingroup applications
  */
  
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/dsp.h"
+#include "asterisk/utils.h"
+#include "asterisk/options.h"
 #include "asterisk/app.h"
-#include "asterisk/channel.h"
-#include "asterisk/dsp.h"	/* use dsp routines for silence detection */
+
+static char *tdesc = "Trivial Record Application";
 
 static char *app = "Record";
 
 static char *synopsis = "Record to a file";
 
 static char *descrip = 
-"  Record(filename.format,silence[,maxduration][,options])\n\n"
+"  Record(filename.format|silence[|maxduration][|options])\n\n"
 "Records from the channel into a given filename. If the file exists it will\n"
 "be overwritten.\n"
 "- 'format' is the format of the file type to be recorded (wav, gsm, etc).\n"
@@ -50,52 +59,35 @@ static char *descrip =
 "or 0 there is no maximum.\n"
 "- 'options' may contain any of the following letters:\n"
 "     'a' : append to existing recording rather than replacing\n"
-"     'k' : keep recorded file upon hangup\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 (DTMF) instead of default '#'\n"
-"     'x' : ignore all terminator keys (DTMF) and keep recording until hangup\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. A channel variable\n"
 "named RECORDED_FILE will also be set, which contains the final filemname.\n\n"
-"Use 'core show file formats' to see the available formats on your system\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"
 "If the user should hangup during a recording, all data will be lost and the\n"
 "application will teminate. \n";
 
-enum {
-	OPTION_APPEND = (1 << 0),
-	OPTION_NOANSWER = (1 << 1),
-	OPTION_QUIET = (1 << 2),
-	OPTION_SKIP = (1 << 3),
-	OPTION_STAR_TERMINATE = (1 << 4),
-	OPTION_IGNORE_TERMINATE = (1 << 5),
-	OPTION_KEEP = (1 << 6),
-	FLAG_HAS_PERCENT = (1 << 7),
-};
+STANDARD_LOCAL_USER;
 
-AST_APP_OPTIONS(app_opts,{
-	AST_APP_OPTION('a', OPTION_APPEND),
-	AST_APP_OPTION('k', OPTION_KEEP),	
-	AST_APP_OPTION('n', OPTION_NOANSWER),
-	AST_APP_OPTION('q', OPTION_QUIET),
-	AST_APP_OPTION('s', OPTION_SKIP),
-	AST_APP_OPTION('t', OPTION_STAR_TERMINATE),
-	AST_APP_OPTION('x', OPTION_IGNORE_TERMINATE),
-});
+LOCAL_USER_DECL;
 
 static int record_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
 	int count = 0;
-	char *ext = NULL, *opts[0];
-	char *parse, *dir, *file;
+	int percentflag = 0;
+	char *filename, *ext = NULL, *silstr, *maxstr, *options;
+	char *vdata, *p;
 	int i = 0;
 	char tmp[256];
 
-	struct ast_filestream *s = NULL;
+	struct ast_filestream *s = '\0';
+	struct localuser *u;
 	struct ast_frame *f = NULL;
 	
 	struct ast_dsp *sildet = NULL;   	/* silence detector dsp */
@@ -105,18 +97,15 @@ static int record_exec(struct ast_channel *chan, void *data)
 	int gotsilence = 0;		/* did we timeout for silence? */
 	int maxduration = 0;		/* max duration of recording in milliseconds */
 	int gottimeout = 0;		/* did we timeout for maxduration exceeded? */
+	int option_skip = 0;
+	int option_noanswer = 0;
+	int option_append = 0;
 	int terminator = '#';
+	int option_quiet = 0;
 	int rfmt = 0;
-	int ioflags;
+	int flags;
 	int waitres;
 	struct ast_silence_generator *silgen = NULL;
-	struct ast_flags flags = { 0, };
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(filename);
-		AST_APP_ARG(silence);
-		AST_APP_ARG(maxduration);
-		AST_APP_ARG(options);
-	);
 	
 	/* 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 */
@@ -124,17 +113,28 @@ static int record_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 
-	parse = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, parse);
-	if (args.argc == 4)
-		ast_app_parse_options(app_opts, &flags, opts, args.options);
+	LOCAL_USER_ADD(u);
 
-	if (!ast_strlen_zero(args.filename)) {
-		if (strstr(args.filename, "%d"))
-			ast_set_flag(&flags, FLAG_HAS_PERCENT);
-		ext = strrchr(args.filename, '.'); /* to support filename with a . in the filename, not format */
+	/* Yay for strsep being easy */
+	vdata = ast_strdupa(data);
+	if (!vdata) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	p = vdata;
+	filename = strsep(&p, "|");
+	silstr = strsep(&p, "|");
+	maxstr = strsep(&p, "|");	
+	options = strsep(&p, "|");
+	
+	if (filename) {
+		if (strstr(filename, "%d"))
+			percentflag = 1;
+		ext = strrchr(filename, '.'); /* to support filename with a . in the filename, not format */
 		if (!ext)
-			ext = strchr(args.filename, ':');
+			ext = strchr(filename, ':');
 		if (ext) {
 			*ext = '\0';
 			ext++;
@@ -142,43 +142,59 @@ static int record_exec(struct ast_channel *chan, void *data)
 	}
 	if (!ext) {
 		ast_log(LOG_WARNING, "No extension specified to filename!\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
 	}
-	if (args.silence) {
-		if ((sscanf(args.silence, "%30d", &i) == 1) && (i > -1)) {
+	if (silstr) {
+		if ((sscanf(silstr, "%d", &i) == 1) && (i > -1)) {
 			silence = i * 1000;
-		} else if (!ast_strlen_zero(args.silence)) {
-			ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", args.silence);
+		} else if (!ast_strlen_zero(silstr)) {
+			ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", silstr);
 		}
 	}
 	
-	if (args.maxduration) {
-		if ((sscanf(args.maxduration, "%30d", &i) == 1) && (i > -1))
+	if (maxstr) {
+		if ((sscanf(maxstr, "%d", &i) == 1) && (i > -1))
 			/* Convert duration to milliseconds */
 			maxduration = i * 1000;
-		else if (!ast_strlen_zero(args.maxduration))
-			ast_log(LOG_WARNING, "'%s' is not a valid maximum duration\n", args.maxduration);
+		else if (!ast_strlen_zero(maxstr))
+			ast_log(LOG_WARNING, "'%s' is not a valid maximum duration\n", maxstr);
 	}
-
-	if (ast_test_flag(&flags, OPTION_STAR_TERMINATE))
-		terminator = '*';
-	if (ast_test_flag(&flags, OPTION_IGNORE_TERMINATE))
-		terminator = '\0';
-
+	if (options) {
+		/* Retain backwards compatibility with old style options */
+		if (!strcasecmp(options, "skip"))
+			option_skip = 1;
+		else if (!strcasecmp(options, "noanswer"))
+			option_noanswer = 1;
+		else {
+			if (strchr(options, 's'))
+				option_skip = 1;
+			if (strchr(options, 'n'))
+				option_noanswer = 1;
+			if (strchr(options, 'a'))
+				option_append = 1;
+			if (strchr(options, 't'))
+				terminator = '*';
+			if (strchr(options, 'q'))
+				option_quiet = 1;
+		}
+	}
+	
 	/* done parsing */
-
+	
 	/* these are to allow the use of the %d in the config file for a wild card of sort to
 	  create a new file with the inputed name scheme */
-	if (ast_test_flag(&flags, FLAG_HAS_PERCENT)) {
+	if (percentflag) {
 		AST_DECLARE_APP_ARGS(fname,
 			AST_APP_ARG(piece)[100];
 		);
-		char *tmp2 = ast_strdupa(args.filename);
+		char *tmp2 = ast_strdupa(filename);
 		char countstring[15];
 		int i;
 
 		/* Separate each piece out by the format specifier */
-		AST_NONSTANDARD_APP_ARGS(fname, tmp2, '%');
+		/* AST_NONSTANDARD_APP_ARGS(fname, tmp2, '%'); */
+		fname.argc = ast_app_separate_args(tmp2, '%', fname.argv, (sizeof(fname) - sizeof(fname.argc)) / sizeof(fname.argv[0]));
 		do {
 			int tmplen;
 			/* First piece has no leading percent, so it's copied verbatim */
@@ -199,28 +215,31 @@ static int record_exec(struct ast_channel *chan, void *data)
 				ast_copy_string(tmp + tmplen, &(fname.piece[i][1]), sizeof(tmp) - tmplen);
 			}
 			count++;
-		} while (ast_fileexists(tmp, ext, chan->language) > 0);
+		} while ( ast_fileexists(tmp, ext, chan->language) != -1 );
 		pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
 	} else
-		ast_copy_string(tmp, args.filename, sizeof(tmp));
+		strncpy(tmp, filename, sizeof(tmp)-1);
 	/* end of routine mentioned */
-
+	
+	
+	
 	if (chan->_state != AST_STATE_UP) {
-		if (ast_test_flag(&flags, OPTION_SKIP)) {
+		if (option_skip) {
 			/* At the user's option, skip if the line is not up */
+			LOCAL_USER_REMOVE(u);
 			return 0;
-		} else if (!ast_test_flag(&flags, OPTION_NOANSWER)) {
+		} else if (!option_noanswer) {
 			/* Otherwise answer unless we're supposed to record while on-hook */
 			res = ast_answer(chan);
 		}
 	}
-
+	
 	if (res) {
 		ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
 		goto out;
 	}
-
-	if (!ast_test_flag(&flags, OPTION_QUIET)) {
+	
+	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) {
@@ -230,47 +249,44 @@ static int record_exec(struct ast_channel *chan, void *data)
 		}
 		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);
 	} 
-
-	/* Create the directory if it does not exist. */
-	dir = ast_strdupa(tmp);
-	if ((file = strrchr(dir, '/')))
-		*file++ = '\0';
-	ast_mkdir (dir, 0777);
-
-	ioflags = ast_test_flag(&flags, OPTION_APPEND) ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
-	s = ast_writefile(tmp, ext, NULL, ioflags, 0, AST_FILE_MODE);
-
+		
+		
+	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", args.filename);
+		ast_log(LOG_WARNING, "Could not create file %s\n", filename);
 		goto out;
 	}
 
-	if (ast_opt_transmit_silence)
+	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) {
@@ -279,7 +295,7 @@ static int record_exec(struct ast_channel *chan, void *data)
 			}
 			maxduration = waitres;
 		}
-
+		
 		f = ast_read(chan);
 		if (!f) {
 			res = -1;
@@ -287,13 +303,13 @@ static int record_exec(struct ast_channel *chan, void *data)
 		}
 		if (f->frametype == AST_FRAME_VOICE) {
 			res = ast_writestream(s, f);
-
+			
 			if (res) {
 				ast_log(LOG_WARNING, "Problem writing frame\n");
 				ast_frfree(f);
 				break;
 			}
-
+			
 			if (silence > 0) {
 				dspsilence = 0;
 				ast_dsp_silence(sildet, f, &dspsilence);
@@ -311,7 +327,7 @@ static int record_exec(struct ast_channel *chan, void *data)
 			}
 		} 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);
@@ -325,15 +341,12 @@ static int record_exec(struct ast_channel *chan, void *data)
 		ast_frfree(f);
 	}
 	if (!f) {
-		ast_debug(1, "Got hangup\n");
+		ast_log(LOG_DEBUG, "Got hangup\n");
 		res = -1;
-		if (!ast_test_flag(&flags, OPTION_KEEP)) {
-			ast_filedelete(args.filename, NULL);
-		}
 	}
-
+			
 	if (gotsilence) {
-		ast_stream_rewind(s, silence - 1000);
+		ast_stream_rewind(s, silence-1000);
 		ast_truncstream(s);
 	} else if (!gottimeout) {
 		/* Strip off the last 1/4 second of it */
@@ -344,8 +357,8 @@ static int record_exec(struct ast_channel *chan, void *data)
 
 	if (silgen)
 		ast_channel_stop_silence_generator(chan, silgen);
-
-out:
+	
+ out:
 	if ((silence > 0) && rfmt) {
 		res = ast_set_read_format(chan, rfmt);
 		if (res)
@@ -354,17 +367,40 @@ out:
 			ast_dsp_free(sildet);
 	}
 
+	LOCAL_USER_REMOVE(u);
+
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+
+	res = ast_unregister_application(app);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, record_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Trivial Record Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_sayunixtime.c

@@ -1,8 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (c) 2003, 2006 Tilghman Lesher.  All rights reserved.
- * Copyright (c) 2006 Digium, Inc.
+ * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
  *
  * Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com>
  *
@@ -19,22 +18,29 @@
 /*! \file
  *
  * \brief SayUnixTime application
- *
- * \author Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com>
  * 
  * \ingroup applications
  */
 
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/options.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/say.h"
-#include "asterisk/app.h"
+
+
+static char *tdesc = "Say time";
 
 static char *app_sayunixtime = "SayUnixTime";
 static char *app_datetime = "DateTime";
@@ -42,64 +48,94 @@ static char *app_datetime = "DateTime";
 static char *sayunixtime_synopsis = "Says a specified time in a custom format";
 
 static char *sayunixtime_descrip =
-"SayUnixTime([unixtime][,[timezone][,format]])\n"
-"  unixtime  - time, in seconds since Jan 1, 1970.  May be negative.\n"
+"SayUnixTime([unixtime][|[timezone][|format]])\n"
+"  unixtime: time, in seconds since Jan 1, 1970.  May be negative.\n"
 "              defaults to now.\n"
-"  timezone  - timezone, see /usr/share/zoneinfo for a list.\n"
+"  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"
+"  format:   a format the time is to be said in.  See voicemail.conf.\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"
+"DateTime([unixtime][|[timezone][|format]])\n"
+"  unixtime: time, in seconds since Jan 1, 1970.  May be negative.\n"
 "              defaults to now.\n"
-"  timezone  - timezone, see /usr/share/zoneinfo for a list.\n"
+"  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"
+"  format:   a format the time is to be said in.  See voicemail.conf.\n"
 "              defaults to \"ABdY 'digits/at' IMp\"\n";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int sayunixtime_exec(struct ast_channel *chan, void *data)
 {
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(timeval);
-		AST_APP_ARG(timezone);
-		AST_APP_ARG(format);
-	);
-	char *parse;
-	int res = 0;
+	int res=0;
+	struct localuser *u;
+	char *s,*zone=NULL,*timec,*format;
 	time_t unixtime;
+	struct timeval tv;
 	
-	if (!data)
-		return 0;
+	LOCAL_USER_ADD(u);
 
-	parse = ast_strdupa(data);
+	tv = ast_tvnow();
+	unixtime = (time_t)tv.tv_sec;
 
-	AST_STANDARD_APP_ARGS(args, parse);
+	if( !strcasecmp(chan->language, "da" ) ) {
+		format = "A dBY HMS";
+	} else if ( !strcasecmp(chan->language, "de" ) ) {
+		format = "A dBY HMS";
+	} else {
+		format = "ABdY 'digits/at' IMp";
+	} 
 
-	ast_get_time_t(args.timeval, &unixtime, time(NULL), NULL);
+	if (data) {
+		s = data;
+		s = ast_strdupa(s);
+		if (s) {
+			timec = strsep(&s,"|");
+			if ((timec) && (*timec != '\0')) {
+				long timein;
+				if (sscanf(timec,"%ld",&timein) == 1) {
+					unixtime = (time_t)timein;
+				}
+			}
+			if (s) {
+				zone = strsep(&s,"|");
+				if (zone && (*zone == '\0'))
+					zone = NULL;
+				if (s) {
+					format = s;
+				}
+			}
+		} else {
+			ast_log(LOG_ERROR, "Out of memory error\n");
+		}
+	}
 
-	if (chan->_state != AST_STATE_UP)
+	if (chan->_state != AST_STATE_UP) {
 		res = ast_answer(chan);
-
+	}
 	if (!res)
-		res = ast_say_date_with_format(chan, unixtime, AST_DIGIT_ANY,
-					       chan->language, args.format, args.timezone);
+		res = ast_say_date_with_format(chan, unixtime, AST_DIGIT_ANY, chan->language, format, zone);
 
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
 	int res;
 	
 	res = ast_unregister_application(app_sayunixtime);
 	res |= ast_unregister_application(app_datetime);
+
+	STANDARD_HANGUP_LOCALUSERS;
 	
 	return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	int res;
 	
@@ -109,4 +145,19 @@ static int load_module(void)
 	return res;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Say time");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_senddtmf.c

@@ -19,111 +19,111 @@
 /*! \file
  *
  * \brief App to send DTMF digits
- *
- * \author Mark Spencer <markster@digium.com>
  * 
  * \ingroup applications
  */
  
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
 #include "asterisk/app.h"
-#include "asterisk/manager.h"
-#include "asterisk/channel.h"
+
+static char *tdesc = "Send DTMF digits Application";
 
 static char *app = "SendDTMF";
 
 static char *synopsis = "Sends arbitrary DTMF digits";
 
 static char *descrip = 
-" SendDTMF(digits[,[timeout_ms][,duration_ms]]): Sends DTMF digits on a channel. \n"
-" Accepted digits: 0-9, *#abcd, (default .25s pause between digits)\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"
-" Optional Params: \n"
-"   timeout_ms: pause between digits.\n"
-"   duration_ms: duration of each digit.\n";
+" encounters an error.\n";
 
-static int senddtmf_exec(struct ast_channel *chan, void *vdata)
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int senddtmf_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
-	char *data;
-	int timeout = 0, duration = 0;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(digits);
-		AST_APP_ARG(timeout);
-		AST_APP_ARG(duration);
-	);
+	struct localuser *u;
+	char *digits = NULL, *to = NULL;
+	int timeout = 250;
 
-	if (ast_strlen_zero(vdata)) {
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "SendDTMF requires an argument (digits or *#aAbBcCdD)\n");
 		return 0;
 	}
 
-	data = ast_strdupa(vdata);
-	AST_STANDARD_APP_ARGS(args, data);
+	LOCAL_USER_ADD(u);
 
-	if (!ast_strlen_zero(args.timeout))
-		timeout = atoi(args.timeout);
-	if (!ast_strlen_zero(args.duration))
-		duration = atoi(args.duration);
-	res = ast_dtmf_stream(chan, NULL, args.digits, timeout <= 0 ? 250 : timeout, duration);
+	digits = ast_strdupa(data);
+	if (!digits) {
+		ast_log(LOG_ERROR, "Out of Memory!\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	if ((to = strchr(digits,'|'))) {
+		*to = '\0';
+		to++;
+		timeout = atoi(to);
+	}
+		
+	if(timeout <= 0)
+		timeout = 250;
+
+	res = ast_dtmf_stream(chan,NULL,digits,timeout);
+		
+	LOCAL_USER_REMOVE(u);
 
 	return res;
 }
 
-static char mandescr_playdtmf[] =
-"Description: Plays a dtmf digit on the specified channel.\n"
-"Variables: (all are required)\n"
-"	Channel: Channel name to send digit to\n"
-"	Digit: The dtmf digit to play\n";
-
-static int manager_play_dtmf(struct mansession *s, const struct message *m)
-{
-	const char *channel = astman_get_header(m, "Channel");
-	const char *digit = astman_get_header(m, "Digit");
-	struct ast_channel *chan = ast_get_channel_by_name_locked(channel);
-	
-	if (!chan) {
-		astman_send_error(s, m, "Channel not specified");
-		return 0;
-	}
-	if (ast_strlen_zero(digit)) {
-		astman_send_error(s, m, "No digit specified");
-		ast_channel_unlock(chan);
-		return 0;
-	}
-
-	ast_senddigit(chan, *digit, 0);
-
-	ast_channel_unlock(chan);
-	astman_send_ack(s, m, "DTMF successfully queued");
-	
-	return 0;
-}
-
-static int unload_module(void)
+int unload_module(void)
 {
 	int res;
 
 	res = ast_unregister_application(app);
-	res |= ast_manager_unregister("PlayDTMF");
+
+	STANDARD_HANGUP_LOCALUSERS;
 
 	return res;	
 }
 
-static int load_module(void)
+int load_module(void)
+{
+	return ast_register_application(app, senddtmf_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
 {
 	int res;
-
-	res = ast_manager_register2( "PlayDTMF", EVENT_FLAG_CALL, manager_play_dtmf, "Play DTMF signal on a specific channel.", mandescr_playdtmf );
-	res |= ast_register_application(app, senddtmf_exec, synopsis, descrip);
-
+	STANDARD_USECOUNT(res);
 	return res;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Send DTMF digits Application");
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_sendtext.c

@@ -19,81 +19,138 @@
 /*! \file
  *
  * \brief App to transmit a text message
- *
- * \author Mark Spencer <markster@digium.com>
  * 
- * \note Requires support of sending text messages from channel driver
+ * Requires support of sending text messages from channel driver
  *
  * \ingroup applications
  */
  
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
+#include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/image.h"
+#include "asterisk/options.h"
 #include "asterisk/app.h"
 
+static const char *tdesc = "Send Text Applications";
+
 static const char *app = "SendText";
 
 static const char *synopsis = "Send a Text Message";
 
 static const char *descrip = 
-"  SendText(text): Sends text to current channel (callee).\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"
 "      FAILURE      Transmission failed\n"
 "      UNSUPPORTED  Text transmission not supported by channel\n"
 "\n"
-"At this moment, text is supposed to be 7 bit ASCII in most channels.\n";
+"At this moment, text is supposed to be 7 bit ASCII in most channels.\n"
+"The option string many contain the following character:\n"
+"'j' -- jump to n+101 priority if the channel doesn't support\n"
+"       text transport\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 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);	
 
-	/* NOT ast_strlen_zero, because some protocols (e.g. SIP) MUST be able to
-	 * send a zero-length message. */
-	if (!data) {
-		ast_log(LOG_WARNING, "SendText requires an argument (text)\n");
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "SendText requires an argument (text[|options])\n");
+		LOCAL_USER_REMOVE(u);
 		return -1;
-	} else
+	} else {
 		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);
 
-	ast_channel_lock(chan);
+	if (args.options) {
+		if (strchr(args.options, 'j'))
+			priority_jump = 1;
+	}
+
+	ast_mutex_lock(&chan->lock);
 	if (!chan->tech->send_text) {
-		ast_channel_unlock(chan);
+		ast_mutex_unlock(&chan->lock);
 		/* Does not support transport */
-		pbx_builtin_setvar_helper(chan, "SENDTEXTSTATUS", status);
+		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_channel_unlock(chan);
+	ast_mutex_unlock(&chan->lock);
 	res = ast_sendtext(chan, args.text);
 	if (!res)
 		status = "SUCCESS";
 	pbx_builtin_setvar_helper(chan, "SENDTEXTSTATUS", status);
+	LOCAL_USER_REMOVE(u);
 	return 0;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	return ast_unregister_application(app);
+	int res;
+	
+	res = ast_unregister_application(app);
+	
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;	
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, sendtext_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Send Text Applications");
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_setcallerid.c

@@ -18,19 +18,22 @@
 
 /*! \file
  *
- * \brief App to set callerid presentation
- *
- * \author Mark Spencer <markster@digium.com>
+ * \brief App to set callerid
  * 
  * \ingroup applications
  */
  
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
 #include "asterisk.h"
 
 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/lock.h"
 #include "asterisk/file.h"
+#include "asterisk/logger.h"
 #include "asterisk/channel.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
@@ -42,6 +45,9 @@ static char *app2 = "SetCallerPres";
 
 static char *synopsis2 = "Set CallerID Presentation";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static char *descrip2 = 
 "  SetCallerPres(presentation): Set Caller*ID presentation on a call.\n"
@@ -61,37 +67,112 @@ static char *descrip2 =
 
 static int setcallerid_pres_exec(struct ast_channel *chan, void *data)
 {
+	struct localuser *u;
 	int pres = -1;
-	static int deprecated = 0;
 
-	if (!deprecated) {
-		deprecated = 1;
-		ast_log(LOG_WARNING, "SetCallerPres is deprecated.  Please use Set(CALLERPRES()=%s) instead.\n", (char *)data);
-	}
-
-	/* For interface consistency, permit the argument to be specified as a number */
-	if (sscanf(data, "%30d", &pres) != 1 || pres < 0 || pres > 255 || (pres & 0x9c)) {
-		pres = ast_parse_caller_presentation(data);
-	}
+	LOCAL_USER_ADD(u);
+	
+	pres = ast_parse_caller_presentation(data);
 
 	if (pres < 0) {
 		ast_log(LOG_WARNING, "'%s' is not a valid presentation (see 'show application SetCallerPres')\n",
 			(char *) data);
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
 	
 	chan->cid.cid_pres = pres;
+	LOCAL_USER_REMOVE(u);
 	return 0;
 }
 
-static int unload_module(void)
+
+
+static char *tdesc = "Set CallerID Application";
+
+static char *app = "SetCallerID";
+
+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. \n";
+
+static int setcallerid_exec(struct ast_channel *chan, void *data)
 {
-	return ast_unregister_application(app2);
+	int res = 0;
+	char *tmp = NULL;
+	char name[256];
+	char num[256];
+	struct localuser *u;
+	char *opt;
+	int anitoo = 0;
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "SetCallerID requires an argument!\n");
+		return 0;
+	}
+	
+	LOCAL_USER_ADD(u);
+	
+	tmp = ast_strdupa(data);
+	if (!tmp) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+	
+	opt = strchr(tmp, '|');
+	if (opt) {
+		*opt = '\0';
+		opt++;
+		if (*opt == 'a')
+			anitoo = 1;
+	}
+	
+	ast_callerid_split(tmp, name, sizeof(name), num, sizeof(num));
+	ast_set_callerid(chan, num, name, anitoo ? num : NULL);
+
+	LOCAL_USER_REMOVE(u);
+	
+	return res;
 }
 
-static int load_module(void)
+int unload_module(void)
 {
-	return ast_register_application(app2, setcallerid_pres_exec, synopsis2, descrip2);
+	int res;
+
+	res = ast_unregister_application(app2);
+	res |= ast_unregister_application(app);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return res;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Set CallerID Presentation Application");
+int load_module(void)
+{
+	int res;
+	
+	res = ast_register_application(app2, setcallerid_pres_exec, synopsis2, descrip2);
+	res |= ast_register_application(app, setcallerid_exec, synopsis, descrip);
+
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}