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/1.0.10@212958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

.cvsignore

@@ -1,4 +0,0 @@
-asterisk
-build.h
-ast_expr.c
-.version

BUGS

@@ -1,48 +1,23 @@
-* EVERYTHING MARKED WITH "XXX" IN THE SOURCE REPRESENTS A BUG!  Sometimes
-  these bugs are in asterisk, and sometimes they relate to the products
-  that asterisk uses.
+Asterisk 1.0-RC-1 Known Major Bugs
+==================================
+* Some people still have issues with H.323
+* QuickNet driver still not entirely stable
 
-* In general Asterisk is a very new program, and there are liable to be
-  many bugs yet to be discovered, so if you think you've found one, please
-  be sure to report it.
+Asterisk Bug Tracking Information
+=================================
 
-* When you flip to call waiting on a tormenta channel while you have a
-  three way call up, the parties in the three way cannot hear one another
-  in the general case.
+To learn about and report Asterisk bugs or make feature
+requests, please visit the official Asterisk Bug Tracker
+at:
 
-* No auto-reload in chan_zap yet
+	http://bugs.digium.com
 
-* Must be able to call park with flash-hook transfer
+For more information on using the bug tracker, or to 
+learn how you can contribute by acting as a bug marshal
+please see:
 
-======================================================================
-Short report on the voicemail system
-======================================================================
-Stuff We Need:
+	http://www.digium.com/bugtracker.html
 
--Date/Time (conversion on the fly for different locales)
--A more fleshed/emphasized Main Menu
--Mailbox Options
-	-useful for allowing user to set certain options
--Notification of new vs. old messages
--Notification of first and last messages (old and new)
-	and a return to the Main Menu @ the end
-**-Better handling of lack of user input, specifically...
-	infinite loops...
-	currently found in:  vm-instructions
-			     vm-msginstruct
-	System MUST disconnect user for inactivity
--Mid message menu w/
-	pause/unpause
-	seeking
-	callback option
-	option to get caller's number if available
-	option to leave message directly on caller's voicemail
-		if he/she has account on system
--Also redesign the End Of Message Menu
--Efficienty Question...
-	Better to always rename msgs to being @ 0001
-	or...
-	Better to append new msgs numerically @ the
-	end and use software to traverse them in 
-	order...saving cpu cycles on renaming files
-	..could get messy w/ lots of users
+Thank you!
+
+Mark

CHANGES

@@ -1,267 +0,0 @@
-Asterisk 0.4.0
- -- Merge and edit Nick's FXO dial support
- -- Reengineer SIP registration (outbound)
- -- Support call pickup on SIP and compatibly with ZAP
- -- Support 302 Redirect on SIP
- -- Management interface improvements
- -- Add "hint" support
- -- Improve call forwarding using new "Local" channel driver.
- -- Add "Local" channel
- -- Substantial SIP enhancements including retransmissions
- -- Enforce case sensitivity on extension/context names
- -- Add monitor support (Thanks, Mahmut)
- -- Add experimental "trunk" option to IAX2 for high density VoIP
- -- Add experimental "debug channel" command
- -- Add 'C' flag to dial command to reset call detail record (handy for calling cards)
- -- Add NAT and dynamic support to MGCP
- -- Allow selection of in-band, out-of-band, or INFO based DTMF
- -- Add contributed "*80" support to blacklist numbers (Thanks James!)
- -- Add "NAT" option to sip user, peer, friend
- -- Add experimental "IAX2" protocol
- -- Change special variable "EXTEN-n" to "EXTEN:n" to follow Bash syntax
- -- Add "Enhanced" AGI with audio pass-through (voice recognition anyone?)
- -- Choose best priority from codec from allow/disallow
- -- Reject SIP calls to self
- -- Allow SIP registration to provide an alternative contact
- -- Make HOLD on SIP make use of asterisk MOH
- -- Add supervised transfer (tested with Pingtel only)
- -- Allow maxexpirey and defaultexpirey to be runtime configurable for SIP
- -- Preliminary codec 13 support (RFC3389)
- -- Add app_authenticate for general purpose authentication
- -- Optimize RTP and smoother
- -- Create special variable "EXTEN-n" where it is extension stripped by n MSD
- -- Fix uninitialized frame pointer in channel.c
- -- Add global variables support under [globals] of extensions.conf
- -- Add macro support (show application Macro)
- -- Allow [123-5] etc in extensions
- -- Allow format of App(arg1,arg2,...) instead of just App,arg1|arg2 in dialplan
- -- Add message waiting indicator to SIP
- -- Fix double free bug in channel.c
-Asterisk 0.3.0
- -- Add fastfoward, rewind, seek, and truncate functions to streams
- -- Support registration
- -- Add G729 format
- -- Permit applications to return a digit indicating new extension
- -- Change "SHUTDOWN" to "STOP" in commands
- -- SIP "Hold" fixes and VXML URI support
- -- New chan_zap with 160 sample chunk size
- -- Add DTMF, MF, and Fax tone detector to dsp routines
- -- Allow overlap dialing (inbound) on PRI
- -- Enable tone detection with PRI
- -- Add special information tone detection
- -- Add Asterisk DB support
- -- Add pulse dialing
- -- Re-record all system prompts
- -- Change "timelen" to samples for better accuracy
- -- Move to editline, eliminating readline dependency
- -- Add peer "poke" support to SIP and IAX
- -- Add experimental call progress detection
- -- Add SIP authentication (digest)
- -- Add RDNIS
- -- Reroute faxes to "fax" extension
- -- Create ISDN/modem group concept
- -- Centralize indication
- -- Add initial MGCP support
- -- SIP debugging cleanup
- -- SIP reload
- -- SIP commands (show channels, etc)
- -- Add optional busy detection
- -- Add Visual Message Waiting Indicator (MDMF and SDMF)
- -- Add ambiguous extension matching
- -- Add *69
- -- Major SIP enhancements from SIPit
- -- Rewrite of ZAP CLASS features using subchannels
- -- Enhanced call parking
- -- Add extended outgoing spool support (pbx_spool)
-Asterisk 0.2.0
- -- Outbound origination API
- -- Call management improvements
- -- Add Do Not Disturb (*78, *79)
- -- Add agents
- -- Document variables
- -- Add transfer capability on the console
- -- Add SpeeX codec translator
- -- Add call queues
- -- Add setcallerid functionality (AGI, application)
- -- Add special variables ${CALLERID}, ${EXTEN}, ${CONTEXT}, ${PRIORITY}
- -- Don't echo cancel on pure TDM connections by default
- -- Implement Async GOTO
- -- Differentiate softhangups
- -- Add date/time
-Asterisk 0.1.12
- -- Fix for Big Endian machines
- -- MySQL CDR Engine
- -- Various SIP fixes and enhancements
- -- Add "zapateller application and arbitrary tone pairs
- -- Don't always start at "s"
- -- Separate linear mode for pseudo and real
- -- Add initial RTP and SIP support (no jitter buffer yet, unknown stability)
- -- Add 'h' extension, executed on hangup
- -- Add duration timer to message info
- -- Add web based voicemail checking ("make webvmail")
- -- Add ast_queue_frame function and eliminate frame pipes in most drivers
- -- Centralize host access (and possibly future ACL's)
- -- Add Caller*ID on PhoneJack (Thanks Nathan)
- -- Add "safe_asterisk" wrapper script to auto-restart Asterisk
- -- Indicate ringback on chan_phone
- -- Add answer confirmation (press '#' to confirm answer)
- -- Add distinctive ring support (e.g. Dial,Zap/4r2)
- -- Add ANSI/vt100 color support
- -- Make parking configurable through parking.conf
- -- Fix the empty voicemail problem
- -- Add Music On Hold
- -- Add ADSI Compiler (app_adsiprog)
- -- Extensive DISA re-work to improve tone generation
- -- Reset all idle channels every 10 minutes on a PRI
- -- Reset channels which are hungup with "channel in use"
- -- Implement VNAK support in chan_iax
- -- Fix chan_oss to support proper hangups and autoanswer
- -- Make shutdown properly hangup channels
- -- Add idling capability to chan_zap for idle-net
- -- Add "MeetMe" conferencing app (app_meetme)
- -- Add timing information to include
-Asterisk 0.1.11
- -- Add ISDN RAS capability
- -- Add stutter dialtone to Chan Zap
- -- Add "#include" capability to config files.
- -- Add call-forward variable to Chan Zap (*72, *73)
- -- Optimize IAX flow when transfer isn't possible
- -- Allow transmission of ANI over IAX
-Asterisk 0.1.10
- -- Make ast_readstring parameter be the max # of digits, not the max size with \0
- -- Make up any missing messages on the fly
- -- Add support for specific DTMF interruption to saying numbers
- -- Add new "u" and "b" options to condense busy/unavail handling
- -- Add support for RSA authentication on IAX calls
- -- Add support for ADSI compatible CPE
- -- Outgoing call queue
- -- Remote dialplan fixes for Quicknet
- -- Added AGI commands supporting TDD functions (RECEIVE CHAR & TDD MODE)
- -- Added TDD support (send/receive text in chan_zap)
- -- Fix all strncpy references
- -- Implement CSV CDR backend
- -- Implement Call Detail Records
-Asterisk 0.1.9
- -- Implement IAX quelching
- -- Allow Caller*ID to be overridden and suggested
- -- Configure defaults to use IAXTEL
- -- Allow remote dialplan polling via IAX
- -- Eliminate ast_longest_extension
- -- Implement dialplan request/reply
- -- Let peers have allow/disallow for codecs
- -- Change allow/deny to permit/deny in IAX
- -- Allow dialplan entries to match Caller*ID as well
- -- Added AGI (Asterisk Gateway Interface) scripting interface (app_agi)
- -- Added chan_zap for zapata telephony kernel interface, removed chan_tor
- -- Add convenience functions
- -- Fix race condition in channel hangup
- -- Fix memory leaks in both asterisk and iax frame allocations
- -- Add "iax show stats" command and -DTRACE_FRAMES (for frame tracing)
- -- Add DISA application (Thanks to Jim Dixon)
- -- Add IAX transfer support
- -- Add URL and HTML transmission
- -- Add application for sending images
- -- Add RedHat RPM spec file and build capability
- -- Fix GSM WAV file format bug
- -- Move ignorepat to main dialplan
- -- Add ability to specificy TOS bits in IAX
- -- Allow username:password in IAX strings
- -- Updates to PhoneJack interface
- -- Allow "servermail" in voicemail.conf to override e-mail in "from" line
- -- Add 'skip' option to app_playback
- -- Reject IAX calls on unknown extensions
- -- Fix version stuff
-Asterisk 0.1.8
- -- Keep track of version information
- -- Add -f to cause Asterisk not to fork
- -- Keep important information in voicemail .txt file
- -- Adtran Voice over Frame Relay updates
- -- Implement option setting/querying of channel drivers
- -- IAX performance improvements and protocol fixes
- -- Substantial enhancement of console channel driver
- -- Add IAX registration.  Now IAX can dynamically register
- -- Add flash-hook transfer on tormenta channels
- -- Added Three Way Calling on tormenta channels
- -- Start on concept of zombie channel
- -- Add Call Waiting CallerID
- -- Keep track of who registeres contexts, includes, and extensions
- -- Added Call Waiting(tm), *67, *70, and *82 codes
- -- Move parked calls into "parkedcalls" context by default
- -- Allow dialplan to be displayed
- -- Allow "=>" instead of just "=" to make instantiation clearer
- -- Asterisk forks if called with no arguments
- -- Add remote control by running asterisk -vvvc
- -- Adjust verboseness with "set verbose" now
- -- No longer requires libaudiofile
- -- Install beep
- -- Make PBX Config module reload extensions on SIGHUP
- -- Allow modules to be reloaded when SIGHUP is received
- -- Variables now contain line numbers
- -- Make dialer send in band signalling
- -- Add record application
- -- Added PRI signalling to Tormenta driver
- -- Allow use of BYEXTENSION in "Goto"
- -- Allow adjustment of gains on tormenta channels
- -- Added raw PCM file format support
- -- Add U-law translator
- -- Fix DTMF handling in bridge code
- -- Fix access control with IAX
-* Asterisk 0.1.7
- -- Update configuration files and add some missing sounds
- -- Added ability to include one context in another
- -- Rewrite of PBX switching
- -- Major mods to dialler application
- -- Added Caller*ID spill reception
- -- Added Dialogic VOX file format support
- -- Added ADPCM Codec
- -- Add Tormenta driver (RBS signalling)
- -- Add Caller*ID spill creation
- -- Rewrite of translation layer entirely
- -- Add ability to run PBX without additional thread
-* Asterisk 0.1.6
- -- Make app_dial handle a lack of translators smoothly
- -- Add ISDN4Linux support -- dtmf is weird...
- -- Minor bug fixes
-* Asterisk 0.1.5
- -- Fix a small mistake in IAX
- -- Fix the QuickNet driver to work with newer cards
-* Asterisk 0.1.4
- -- Update VoFR some more
- -- Fix the QuickNet driver to work with LineJack
- -- Add ability to pass images for IAX.
-* Asterisk 0.1.3
- -- Update VoFR for latest sangoma code
- -- Update QuickNet Driver
- -- Add text message handling
- -- Fix transfers to use "default" if not in current context
- -- Add call parking
- -- Improve format/content negotiation
- -- Added support for multiple languages
- -- Bug fixes, as always...
-* Asterisk 0.1.2
- -- Updated README file with a "Getting Started" section
- -- Added sample sounds and configuration files.
- -- Added LPC10 very low bandwidth (low quality) compression
- -- Enhanced translation selection mechanism.
- -- Enhanced IAX jitter buffer, improved reliability
- -- Support echo cancelation on PhoneJack
- -- Updated PhoneJack driver to std. Telephony interface
- -- Added app_echo for evaluating VoIP latency
- -- Added app_system to execute arbitrary programs
- -- Updated sample configuration files
- -- Added OSS channel driver (full duplex only)
- -- Added IAX implementation
- -- Fixed some deadlocks.
- -- A whole bunch of bug fixes
-* Asterisk 0.1.1
- -- Revised translator, fixed some general race conditions throughout *
- -- Made dialer somewhat more aware of incompatible voice channels
- -- Added Voice Modem driver and A/Open Modem Driver stub
- -- Added MP3 decoder channel
- -- Added Microsoft WAV49 support
- -- Revised License -- Pure GPL, nothing else
- -- Modified Copyright statement since code is still currently owned by author
- -- Added RAW GSM headerless data format
- -- Innumerable bug fixes
-* Asterisk 0.1.0
- -- Initial Release

CREDITS

@@ -5,12 +5,16 @@ Asterisk:
 
 Pilosoft, Inc. - for supporting ADSI development in Asterisk
 
+Asterlink, Inc. - for supporting broad Asterisk development
+
 GFS - for supporting ALSA development
 
 Telesthetic - for supporting SIP development
 
 Christos Ricudis - for substantial code contributions
 
+Paul Bagyenda, Digital Solutions - for initial Voicetronix driver development
+
 === WISHLIST CONTRIBUTERS ===
 Jeremy McNamara - SpeeX support
 Nick Seraphin - RDNIS support
@@ -22,6 +26,8 @@ Wasim - Hangup detect
 PhoneJack and Linejack card to the project.  (http://www.quicknet.net)
 
 === MISCELLANEOUS PATCHES ===
+Anthony Minessale - Countless big and small fixes, and relentless forward push
+       anthmct@yahoo.com              http://www.asterlink.com
 James Golovich - Innumerable contributions
        You can find him and asterisk-perl at http://asterisk.gnuinter.net
 Andre Bierwirth - Extension hints and status
@@ -30,6 +36,8 @@ 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@esoft.pf             http://www.esoft.pf
+William Jordan / Vonage - MySQL enhancmenets to Voicemail
+	wjordan@vonage.com
 Jac Kersing - Various fixes
 Steven Critchfield - Seek and Trunc functions for playback and recording
        critch@basesys.com
@@ -40,10 +48,25 @@ Mahmut Fettahlioglu - Audio recording, music-on-hold changes, alaw file
        format, and various fixes. Can be contacted at mahmut@oa.com.au
 James Dennis - Cisco SIP compatibility patches to work with SIP service
        providers. Can be contacted at asterisk@jdennis.net
+Tilghman Lesher - - Route lookup code; ast_localtime(), ast_say_date_with_format(); 
+                    GotoIfTime, Random, SayUnixTime, HasNewVoicemail, and Cut applications, 
+		    along with various other patches. http://asterisk.drunkcoder.com/
+Jayson Vantuyl - Manager protocol changes, various other bugs.
+	jvantuyl@computingedge.net
+Thorsten Lockert - OpenBSD, FreeBSD ports, making MacOS X port run on 10.3,
+	dialplan include verification, route lookup on OpenBSD, various other
+	bugs. tholo@sigmasoft.com
+Brian West - ODBC support and Bug Marshaling  
+William Waites - syslog support, SIP NAT traversal for SIP-UA. ww@styx.org
+Rich Murphey - Porting to FreeBSD, NetBSD, OpenBSD, and Darwin.
+	rich@whiteoaklabs.com  http://whiteoaklabs.com
 
 === OTHER CONTRIBUTIONS ===
 John Todd - Monkey sounds and associated teletorture prompt
 
+=== HOLD MUSIC ===
+Music provided by www.opsound.org
+
 === OTHER SOURCE CODE IN ASTERISK ===
 
 I did not implement the codecs in asterisk.  Here is the copyright on the

ChangeLog

@@ -0,0 +1,640 @@
+ NOTE: Corrections or additions to the ChangeLog may be submitted to
+       http://bugs.digium.com.  Documentation and formatting fixes are not
+       not listed here.  A complete listing of changes is available through
+       the Asterisk-CVS mailing list hosted at http://lists.digium.com.
+
+Asterisk 1.0.10
+
+ -- chan_local
+    -- In releases 1.0.8 and 1.0.9, the Local channels that are created would
+       not be masqueraded into the new channel type.  This has now been fixed.
+ -- chan_sip
+    -- The 'insecure' options have been changed to support matching peersby IP
+       only, not requiring authentication on incoming invites, or both. Before,
+       to not require authentication on incoming invites also required matching
+       peers based on IP only.
+ -- chan_zap
+    -- Before, call waiting could occur during the initial ringing on the line.
+       This has now been fixed.
+ -- app_disa
+    -- We will now not set the accountcode if one is not supplied. 
+ -- app_meetme
+    -- If the first caller into a conference hangs up while being prompted for
+       the conference pin number, the conference will no longer be held open.
+ -- app_userevent
+    -- Events created with this application were indicated as a "call" event
+       instead of a "user" event.  This made the "user" event permissions
+       not work correctly.
+ -- app_voicemail
+    -- When using the externpass option for voicemail, the password will be
+       immediately updated in memory as well, instead of having to wait for
+       the next time the configuration is reloaded. 
+ -- app_zapras
+    -- We now ensure buffer policy is restored after RAS is done with a channel.
+       This could cause audio problems on the channel after zapras is done
+       with it. 
+ -- res_agi
+    -- We now unmask the SIGHUP signal before executing an AGI script.  This
+       fixes problems where some AGI scripts would continue running long after
+       the call is over.
+ -- extensions
+    -- A potential crash has been fixed when calling LEN() to get the length of
+       a string that was 80 characters or larger.
+ -- logger
+    -- The Asterisk logger will automatically detect when a log file needs to
+       be rotated.  However, this feature could put Asterisk in a nasty loop
+       that would result in a crash.
+ -- general
+    -- Added man pages for astgenkey, autosupport, and safe_asterisk
+
+Asterisk 1.0.9
+
+ -- fix bug in callerid matching in the dialplan that was introduced in 1.0.8
+
+Asterisk 1.0.8
+
+ -- chan_zap
+    -- Asterisk will now also look in the regular context for the fax extension
+       while executing a macro.  Previously, for this to work, the fax extension
+       would have to be included in the macro definition.
+    -- On some systems, ALERTING will be sent after PROCEEDING, so code has been
+       added to account for this case.
+    -- If no extension is specified on an overlap call, the 's' extension will 
+       be used.
+ -- chan_sip
+    -- We no longer send a "to" tag on "100 Trying" messages, as it is 
+       inappropriate to do so.
+    -- We now respond correctly to an invite for T.38 with a "488 Not acceptable
+       here"
+    -- We now discard saved tags on 401/407 responses in case the provider we're
+       talking to tries to pull a dirty trick on us and change it.
+    -- rtptimeout options will now be correctly set on a peer basis rather than
+       only global
+ -- chan_mgcp
+    -- Fixed setting of accountcode
+    -- Fixed where *67 to block callerid only worked for first call
+ -- chan_agent
+    -- We now will not pass audio until the agent has acked the call if the 
+       configuration
+       is set up for the agent to do so.
+ -- chan_alsa
+    -- Fixed problems with the unloading of this module
+ -- res_agi
+    -- A fix has been added to prevent calls from being hung up when more than 
+       one call is executing an AGI script calling the GET DATA command.
+    -- AGI scripts will now continue to run even if a file was not found with
+       the GET DATA command.
+    -- When calling SAY NUMBER with a number like 09, we will now say "nine" 
+       instead of "zero"
+ -- app_dial
+    -- There was a problem where text frames would not be forwarded before the
+       channel has been answered. 
+ -- app_disa
+    -- Fixed the timeout used when no password is set
+ -- app_queue
+    -- Distinctive ring has been fixed to work for queue members
+  -- rtp
+    -- Fixed a logic error when setting the "rtpchecksums" option
+ -- say.c
+    -- A problem has been fixed with saying the date in Spanish.
+ -- Makefile
+    -- A line was missing for the autosupport script that caused "make rpm" to 
+       fail
+ -- format_wav_gsm
+    -- Fixed a problem with wav formatting that prevented files from being 
+       played in some media players
+ -- pbx_spool
+    -- Fixed if the last line of text in a file for the call spool did not 
+       contain a new line, it would not be processed
+ -- logger
+    -- Fixed the logger so that color escape sequences wouldn't be sent to the 
+       logs
+ -- format_sln
+    -- A lot of changes were made to correctly handle signed linear format on
+       big endian machines
+ -- asterisk.conf
+     -- fix 'highpriority' option for asterisk.conf
+
+Asterisk 1.0.7
+
+ -- chan_sip
+    -- The fix for some codec availibility issues in 1.0.6 caused music on hold
+       problems, but has now been fixed.
+ -- chan_skinny
+    -- A check has been added to avoid a crash.
+ -- chan_iax2
+    -- A feature has been added to CVS head to have the option of sending 
+       timestamps with trunk frames.  It is not supported in 1.0, but a change 
+       has been made so that it will at least not choke if sent trunk
+       timestamps.
+ -- app_voicemail
+    -- Some checks have been added to avoid a crash.
+ -- speex
+    -- The path /usr/include/speex has been added for a place to look for the 
+       speex header.
+
+Asterisk 1.0.6
+
+ -- chan_iax2:
+    -- Fixed a bug dealing with a division by zero that could cause a crash
+ -- chan_sip:
+    -- Behavior was changed so that when a registration fails due to DNS 
+       resolution issues, a retry will be attempted in 20 seconds.
+    -- Peer settings were not reset to null values when reloading the 
+       configuration file. Behavior has been changed so that these values are 
+       now cleared.
+    -- 'restrictcid' now properly works on MySQL peers.
+    -- Only use the default callerid if it has been specified.
+    -- Asterisk was not sending the same From: line in SIP messages during 
+       certain times. Fixed to make sure it stays the same. This makes some 
+       providers happier, to a working state.
+    -- Certain circumstances involving a blank callerid caused asterisk to 
+       segmentation fault.
+    -- There was a problem incorrectly matching codec availablity when global 
+       preferences were different from that of the user.  To fix this, 
+       processing of SDP data has been moved to after determining who the call 
+       is coming from.
+    -- Asterisk would run out of RTP ports while waiting for SUBSCRIBE's to 
+       expire even though an RTP port isn't needed in this case.  This has been
+       fixed by releasing the ports early.
+ -- chan_zap:
+    -- During a certain scenario when using flash and '#' transfers you would 
+       hear the other person and the music they were hearing. This has been 
+       fixed.
+    -- A fix for a compilation issue with gcc4 was added.
+ -- chan_modem_bestdata:
+    -- A fix for a compilation issue with gcc4 was added.
+ -- format_g729:
+    -- Treat a 10-byte read as an end of file indication instead of an error. 
+       Some G729 encoders like to put 10-bytes at the end to indicate this.
+ -- res_features:
+    -- During certain situations when parking a call, both endpoints would get 
+       musiconhold. This has been fixed so the individual who parked the call 
+       will hear the digits and not musiconhold.
+ -- app_dial:
+    -- DIALEDPEERNUMBER is now being set, so if you attempted to use it in the 
+       past and failed, it should work now.
+    -- A callerid change caused many headaches, this has been reversed to the 
+       original 1.0 behavior.
+    -- A crash caused with the combination of the 'g' option and # transfer was
+       fixed.
+ -- app_voicemail:
+    -- If two people hit the voicemail system at the same time, and were leaving
+       a message the second message was overwriting the first. This has been 
+       fixed so that each one is distinct and will not overwrite eachother.
+ -- cdr_tds:
+    -- If the server you were using was going down, it had the potential to 
+       bring your asterisk server down with it. Extra stuff has been added so 
+       as to bring in more error/connection checking.
+ -- cdr_pgsql:
+    -- This will now attempt to reconnect after a connection problem.
+ -- IAXY firmware:
+    -- This has been updated to version 23.  It includes a fix for lost
+       registrations.
+ -- internals
+    -- Behavior was changed for 'show codec <number>' to make it more intuitive.
+    -- DNS failures and asterisk do not get along too well, this is not totally
+       the case anymore.
+    -- Asterisk will now handle DNS failures at startup more gracefully, and 
+       won't crash and burn
+    -- Choosing to append to a wave file would render the outputted wave file 
+       corrupt. Appending now works again.
+    -- If you failed to define certain keys, asterisk had the potential to crash
+       when seeing if you had used them.
+    -- Attempting to use such things as ${EXTEN:-1} gave a wrong return value. 
+       However, this was never a documented feature...
+
+Asterisk 1.0.5
+
+ -- chan_zap
+    -- fix a callerid bug introduced in 1.0.4
+ -- app_queue
+    -- fix some penalty behavior
+
+Asterisk 1.0.4
+
+ -- general
+    -- fix memory leak evident with extensive use of variables
+    -- update IAXy firmware to version 22
+       -- enable some special write protection
+       -- enable outbound DTMF
+    -- fix seg fault with incorrect usage of SetVar
+    -- other minor fixes including typos and doc updates
+ -- chan_sip
+   -- fix codecs to not be case sensitive
+   -- Re-use auth credentials
+   -- fix MWI when using type=friend
+   -- fix global NAT option
+ -- chan_agent / chan_local
+   -- fix incorrect use count
+ -- chan_zap
+   -- Allow CID rings to be configured in zapata.conf
+      -- no more patching needed for UK CID
+ -- app_macro 
+    -- allow Macros to exit with '*' or '#' like regular extension processing
+ -- app_voicemail
+   -- don't allow '#' as a password
+   -- add option to save voicemail before going to the operator
+   -- fix global operator=yes
+ -- app_read
+   -- return 0 instead of -1 if user enters nothing
+ -- res_agi
+    -- don't exit AGI when file not found to stream
+    -- send script parameter when using FastAGI
+ 
+Asterisk 1.0.3
+
+ -- chan_zap
+    -- fix seg fault when doing *0 to flash a trunk
+ -- rtp
+    -- seg fault fix
+ -- chan_sip
+    -- fix to prevent seg fault when attempting a transfer
+    -- fix bug with supervised transfers
+    -- fix codec preferences
+ -- chan_h323
+    -- fix compilation problem
+ -- chan_iax2
+   -- avoid a deadlock related to a static config of a BUNCH of peers
+ -- cdr_pgsql
+    -- fix memory leak when reading config
+ -- Numerous other minor bug fixes
+
+Asterisk 1.0.2
+
+ -- Major bugfix release
+
+Asterisk 1.0.1
+
+ -- Added AGI over TCP support
+ -- Add ability to purge callers from queue if no agents are logged in
+ -- Fix inband PRI indication detection
+ -- Fix for MGCP - always request digits if no RTP stream
+ -- Fixed seg fault for ast_control_streamfile
+ -- Make pick-up extension configurable via features.conf 
+ -- Numerous other bug fixes
+
+Asterisk 1.0.0
+ -- Use Q.931 standard cause codes for asterisk cause codes
+ -- Bug fixes from the bug tracker
+Asterisk 1.0-RC2 
+ -- Additional CDR backends
+ -- Allow muted to reconnect
+ -- Call parking improvements (including SIP parking support)
+ -- Added licensed hold music from opsound.org
+ -- GR-303 and Zap improvements
+ -- More bug fixes from the bug tracker
+ -- Improved FreeBSD/OpenBSD/MacOS X support
+Asterisk 1.0-RC1
+ -- Innumerable bug fixes and features from the bug tracker
+ -- Added Open Settlement Protocol (OSP) support
+ -- Added Non-facility Associated Signalling (NFAS) Support
+ -- Added alarm Monitoring support
+ -- Added new MeetMe options
+ -- Added GR-303 Support
+ -- Added trunk groups
+ -- ADPCM Standardization
+ -- Numerous bug fixes
+ -- Add IAX2 Firmware Support
+ -- Add G.726 support
+ -- Add ices/icecast support
+ -- Numerous bug fixes
+Asterisk 0.7.2
+ -- Countless small bug fixes from bug tracker
+ -- DSP Fixes
+ -- Fix unloading of Zaptel
+ -- Pass Caller*ID/ANI properly on call forwarding
+ -- Add indication for Italy
+Asterisk 0.7.1
+ -- Fixed timed include context's and GotoIfTime
+ -- Fixed chan_h323 it now gets remote ip properly instead of 127.0.0.1
+Asterisk 0.7.0
+ -- Removed MP3 format and codec
+ -- Can now load and unload SIP,IAX,IAX2,H323 channels without core
+ -- Fixed various compiler warnings and clean up source tree
+ -- Preliminary AES Support
+ -- Fix SIP REINVITE
+ -- Outbound SIP registration behind NAT using externip
+ -- More CLI documentation and clean up
+ -- Pin numbers on MeeMe
+ -- Dynamic MeetMe conferences are more consistent with static conferences
+ -- Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP}, ${ACCONTCODE}
+ -- ODBC support for logging CDRs
+ -- Indications for Norway and New Zeland
+ -- Major redesign of app_voicemail
+ -- Syslog support
+ -- Reload logfiles with CLI command 'logger reload' and rotate logs with "logger rotate'
+ -- Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now appear on remote console 
+ -- Properly reaping any zombie processes
+ -- Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail, ZapScan, Random, ResetCDR, NoCDR
+ -- Make PRI Hangup Cause available to the dialplan
+ -- Verify included contexts in extensions.conf
+ -- Add DESTDIR support for building RPMs and packages
+ -- Do route lookups on OpenBSD
+ -- Add support for building on FreeBSD and OS X
+ -- Add support for PostgreSQL in Voicemail
+ -- Translate SIP hangup cause to PRI hangup cause where needed
+ -- Better support for MOH in IAX2
+ -- Fix SIP problem where channels were not removed on BYE
+ -- Display codecs by name
+ -- Remove MySQL and put PGSql instead for licensing reasons
+ -- Better capability matching in SIP
+ -- Full IBR4 compliance for chan_zap
+ -- More flexible CDR handling
+ -- Distinguish between BUSY and FAILURE on outbound calls
+ -- Add initial support for SCCP via chan_skinny
+ -- Better support for Future Group B signaling
+Asterisk 0.5.0
+ -- Retain IAX2 and SIP registrations past shutdown/crash and restart
+ -- True data mode bridging when possible
+ -- H.323 build improvements
+ -- Agent Callback-login support
+ -- RFC2833 Improvements
+ -- Add thread debugging
+ -- Add optional pedantic SIP checking for Pingtel
+ -- Allow extension names, include context, switch to use global vars.
+ -- Allow variables in extensions.conf to reference previously defined ones
+ -- Merge voicemail enhancements (app_voicemail2)
+ -- Add multiple queueing strategies
+ -- Merge support for 'T'
+ -- Allow pending agent calling (Agent/:1)
+ -- Add groupings to agents.conf
+ -- Add video support to IAX2
+ -- Zaptel optimize playback
+ -- Add video support to SIP
+ -- Make RTP ports configurable
+ -- Add RDNIS support to SIP and IAX2
+ -- Add transfer app (implement in SIP and IAX2)
+ -- Make voicemail segmentable by context (app_voicemail2)
+ -- Major restructuring of voicemail (app_voicemail2)
+ -- Add initial ENUM support
+ -- Add malloc debugging support
+ -- Add preliminary Voicetronix support
+ -- Add iLBC codec
+Asterisk 0.4.0
+ -- Merge and edit Nick's FXO dial support
+ -- Reengineer SIP registration (outbound)
+ -- Support call pickup on SIP and compatibly with ZAP
+ -- Support 302 Redirect on SIP
+ -- Management interface improvements
+ -- Add "hint" support
+ -- Improve call forwarding using new "Local" channel driver.
+ -- Add "Local" channel
+ -- Substantial SIP enhancements including retransmissions
+ -- Enforce case sensitivity on extension/context names
+ -- Add monitor support (Thanks, Mahmut)
+ -- Add experimental "trunk" option to IAX2 for high density VoIP
+ -- Add experimental "debug channel" command
+ -- Add 'C' flag to dial command to reset call detail record (handy for calling cards)
+ -- Add NAT and dynamic support to MGCP
+ -- Allow selection of in-band, out-of-band, or INFO based DTMF
+ -- Add contributed "*80" support to blacklist numbers (Thanks James!)
+ -- Add "NAT" option to sip user, peer, friend
+ -- Add experimental "IAX2" protocol
+ -- Change special variable "EXTEN-n" to "EXTEN:n" to follow Bash syntax
+ -- Add "Enhanced" AGI with audio pass-through (voice recognition anyone?)
+ -- Choose best priority from codec from allow/disallow
+ -- Reject SIP calls to self
+ -- Allow SIP registration to provide an alternative contact
+ -- Make HOLD on SIP make use of asterisk MOH
+ -- Add supervised transfer (tested with Pingtel only)
+ -- Allow maxexpirey and defaultexpirey to be runtime configurable for SIP
+ -- Preliminary codec 13 support (RFC3389)
+ -- Add app_authenticate for general purpose authentication
+ -- Optimize RTP and smoother
+ -- Create special variable "EXTEN-n" where it is extension stripped by n MSD
+ -- Fix uninitialized frame pointer in channel.c
+ -- Add global variables support under [globals] of extensions.conf
+ -- Add macro support (show application Macro)
+ -- Allow [123-5] etc in extensions
+ -- Allow format of App(arg1,arg2,...) instead of just App,arg1|arg2 in dialplan
+ -- Add message waiting indicator to SIP
+ -- Fix double free bug in channel.c
+Asterisk 0.3.0
+ -- Add fastfoward, rewind, seek, and truncate functions to streams
+ -- Support registration
+ -- Add G729 format
+ -- Permit applications to return a digit indicating new extension
+ -- Change "SHUTDOWN" to "STOP" in commands
+ -- SIP "Hold" fixes and VXML URI support
+ -- New chan_zap with 160 sample chunk size
+ -- Add DTMF, MF, and Fax tone detector to dsp routines
+ -- Allow overlap dialing (inbound) on PRI
+ -- Enable tone detection with PRI
+ -- Add special information tone detection
+ -- Add Asterisk DB support
+ -- Add pulse dialing
+ -- Re-record all system prompts
+ -- Change "timelen" to samples for better accuracy
+ -- Move to editline, eliminating readline dependency
+ -- Add peer "poke" support to SIP and IAX
+ -- Add experimental call progress detection
+ -- Add SIP authentication (digest)
+ -- Add RDNIS
+ -- Reroute faxes to "fax" extension
+ -- Create ISDN/modem group concept
+ -- Centralize indication
+ -- Add initial MGCP support
+ -- SIP debugging cleanup
+ -- SIP reload
+ -- SIP commands (show channels, etc)
+ -- Add optional busy detection
+ -- Add Visual Message Waiting Indicator (MDMF and SDMF)
+ -- Add ambiguous extension matching
+ -- Add *69
+ -- Major SIP enhancements from SIPit
+ -- Rewrite of ZAP CLASS features using subchannels
+ -- Enhanced call parking
+ -- Add extended outgoing spool support (pbx_spool)
+Asterisk 0.2.0
+ -- Outbound origination API
+ -- Call management improvements
+ -- Add Do Not Disturb (*78, *79)
+ -- Add agents
+ -- Document variables
+ -- Add transfer capability on the console
+ -- Add SpeeX codec translator
+ -- Add call queues
+ -- Add setcallerid functionality (AGI, application)
+ -- Add special variables ${CALLERID}, ${EXTEN}, ${CONTEXT}, ${PRIORITY}
+ -- Don't echo cancel on pure TDM connections by default
+ -- Implement Async GOTO
+ -- Differentiate softhangups
+ -- Add date/time
+Asterisk 0.1.12
+ -- Fix for Big Endian machines
+ -- MySQL CDR Engine
+ -- Various SIP fixes and enhancements
+ -- Add "zapateller application and arbitrary tone pairs
+ -- Don't always start at "s"
+ -- Separate linear mode for pseudo and real
+ -- Add initial RTP and SIP support (no jitter buffer yet, unknown stability)
+ -- Add 'h' extension, executed on hangup
+ -- Add duration timer to message info
+ -- Add web based voicemail checking ("make webvmail")
+ -- Add ast_queue_frame function and eliminate frame pipes in most drivers
+ -- Centralize host access (and possibly future ACL's)
+ -- Add Caller*ID on PhoneJack (Thanks Nathan)
+ -- Add "safe_asterisk" wrapper script to auto-restart Asterisk
+ -- Indicate ringback on chan_phone
+ -- Add answer confirmation (press '#' to confirm answer)
+ -- Add distinctive ring support (e.g. Dial,Zap/4r2)
+ -- Add ANSI/vt100 color support
+ -- Make parking configurable through parking.conf
+ -- Fix the empty voicemail problem
+ -- Add Music On Hold
+ -- Add ADSI Compiler (app_adsiprog)
+ -- Extensive DISA re-work to improve tone generation
+ -- Reset all idle channels every 10 minutes on a PRI
+ -- Reset channels which are hungup with "channel in use"
+ -- Implement VNAK support in chan_iax
+ -- Fix chan_oss to support proper hangups and autoanswer
+ -- Make shutdown properly hangup channels
+ -- Add idling capability to chan_zap for idle-net
+ -- Add "MeetMe" conferencing app (app_meetme)
+ -- Add timing information to include
+Asterisk 0.1.11
+ -- Add ISDN RAS capability
+ -- Add stutter dialtone to Chan Zap
+ -- Add "#include" capability to config files.
+ -- Add call-forward variable to Chan Zap (*72, *73)
+ -- Optimize IAX flow when transfer isn't possible
+ -- Allow transmission of ANI over IAX
+Asterisk 0.1.10
+ -- Make ast_readstring parameter be the max # of digits, not the max size with \0
+ -- Make up any missing messages on the fly
+ -- Add support for specific DTMF interruption to saying numbers
+ -- Add new "u" and "b" options to condense busy/unavail handling
+ -- Add support for RSA authentication on IAX calls
+ -- Add support for ADSI compatible CPE
+ -- Outgoing call queue
+ -- Remote dialplan fixes for Quicknet
+ -- Added AGI commands supporting TDD functions (RECEIVE CHAR & TDD MODE)
+ -- Added TDD support (send/receive text in chan_zap)
+ -- Fix all strncpy references
+ -- Implement CSV CDR backend
+ -- Implement Call Detail Records
+Asterisk 0.1.9
+ -- Implement IAX quelching
+ -- Allow Caller*ID to be overridden and suggested
+ -- Configure defaults to use IAXTEL
+ -- Allow remote dialplan polling via IAX
+ -- Eliminate ast_longest_extension
+ -- Implement dialplan request/reply
+ -- Let peers have allow/disallow for codecs
+ -- Change allow/deny to permit/deny in IAX
+ -- Allow dialplan entries to match Caller*ID as well
+ -- Added AGI (Asterisk Gateway Interface) scripting interface (app_agi)
+ -- Added chan_zap for zapata telephony kernel interface, removed chan_tor
+ -- Add convenience functions
+ -- Fix race condition in channel hangup
+ -- Fix memory leaks in both asterisk and iax frame allocations
+ -- Add "iax show stats" command and -DTRACE_FRAMES (for frame tracing)
+ -- Add DISA application (Thanks to Jim Dixon)
+ -- Add IAX transfer support
+ -- Add URL and HTML transmission
+ -- Add application for sending images
+ -- Add RedHat RPM spec file and build capability
+ -- Fix GSM WAV file format bug
+ -- Move ignorepat to main dialplan
+ -- Add ability to specificy TOS bits in IAX
+ -- Allow username:password in IAX strings
+ -- Updates to PhoneJack interface
+ -- Allow "servermail" in voicemail.conf to override e-mail in "from" line
+ -- Add 'skip' option to app_playback
+ -- Reject IAX calls on unknown extensions
+ -- Fix version stuff
+Asterisk 0.1.8
+ -- Keep track of version information
+ -- Add -f to cause Asterisk not to fork
+ -- Keep important information in voicemail .txt file
+ -- Adtran Voice over Frame Relay updates
+ -- Implement option setting/querying of channel drivers
+ -- IAX performance improvements and protocol fixes
+ -- Substantial enhancement of console channel driver
+ -- Add IAX registration.  Now IAX can dynamically register
+ -- Add flash-hook transfer on tormenta channels
+ -- Added Three Way Calling on tormenta channels
+ -- Start on concept of zombie channel
+ -- Add Call Waiting CallerID
+ -- Keep track of who registeres contexts, includes, and extensions
+ -- Added Call Waiting(tm), *67, *70, and *82 codes
+ -- Move parked calls into "parkedcalls" context by default
+ -- Allow dialplan to be displayed
+ -- Allow "=>" instead of just "=" to make instantiation clearer
+ -- Asterisk forks if called with no arguments
+ -- Add remote control by running asterisk -vvvc
+ -- Adjust verboseness with "set verbose" now
+ -- No longer requires libaudiofile
+ -- Install beep
+ -- Make PBX Config module reload extensions on SIGHUP
+ -- Allow modules to be reloaded when SIGHUP is received
+ -- Variables now contain line numbers
+ -- Make dialer send in band signalling
+ -- Add record application
+ -- Added PRI signalling to Tormenta driver
+ -- Allow use of BYEXTENSION in "Goto"
+ -- Allow adjustment of gains on tormenta channels
+ -- Added raw PCM file format support
+ -- Add U-law translator
+ -- Fix DTMF handling in bridge code
+ -- Fix access control with IAX
+* Asterisk 0.1.7
+ -- Update configuration files and add some missing sounds
+ -- Added ability to include one context in another
+ -- Rewrite of PBX switching
+ -- Major mods to dialler application
+ -- Added Caller*ID spill reception
+ -- Added Dialogic VOX file format support
+ -- Added ADPCM Codec
+ -- Add Tormenta driver (RBS signalling)
+ -- Add Caller*ID spill creation
+ -- Rewrite of translation layer entirely
+ -- Add ability to run PBX without additional thread
+* Asterisk 0.1.6
+ -- Make app_dial handle a lack of translators smoothly
+ -- Add ISDN4Linux support -- dtmf is weird...
+ -- Minor bug fixes
+* Asterisk 0.1.5
+ -- Fix a small mistake in IAX
+ -- Fix the QuickNet driver to work with newer cards
+* Asterisk 0.1.4
+ -- Update VoFR some more
+ -- Fix the QuickNet driver to work with LineJack
+ -- Add ability to pass images for IAX.
+* Asterisk 0.1.3
+ -- Update VoFR for latest sangoma code
+ -- Update QuickNet Driver
+ -- Add text message handling
+ -- Fix transfers to use "default" if not in current context
+ -- Add call parking
+ -- Improve format/content negotiation
+ -- Added support for multiple languages
+ -- Bug fixes, as always...
+* Asterisk 0.1.2
+ -- Updated README file with a "Getting Started" section
+ -- Added sample sounds and configuration files.
+ -- Added LPC10 very low bandwidth (low quality) compression
+ -- Enhanced translation selection mechanism.
+ -- Enhanced IAX jitter buffer, improved reliability
+ -- Support echo cancelation on PhoneJack
+ -- Updated PhoneJack driver to std. Telephony interface
+ -- Added app_echo for evaluating VoIP latency
+ -- Added app_system to execute arbitrary programs
+ -- Updated sample configuration files
+ -- Added OSS channel driver (full duplex only)
+ -- Added IAX implementation
+ -- Fixed some deadlocks.
+ -- A whole bunch of bug fixes
+* Asterisk 0.1.1
+ -- Revised translator, fixed some general race conditions throughout *
+ -- Made dialer somewhat more aware of incompatible voice channels
+ -- Added Voice Modem driver and A/Open Modem Driver stub
+ -- Added MP3 decoder channel
+ -- Added Microsoft WAV49 support
+ -- Revised License -- Pure GPL, nothing else
+ -- Modified Copyright statement since code is still currently owned by author
+ -- Added RAW GSM headerless data format
+ -- Innumerable bug fixes
+* Asterisk 0.1.0
+ -- Initial Release

HARDWARE

@@ -16,11 +16,11 @@ Zaptel compatible hardware
    * Wildcard X100P - Single FXO interface connects to Loopstart phone 
      line
 
-   * Wildcard T400P - Quad T1 interface connects to four T1/PRI 
+   * 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 - Quad E1 interface connects to four E1/PRI (or PRA)
-     interfaces.  Supports PRA/PRI, EuroISDN voice and PPP, FR, 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.
@@ -31,6 +31,12 @@ Zaptel compatible hardware
    * 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
 ==============================
 

Makefile

@@ -13,36 +13,104 @@
 
 .EXPORT_ALL_VARIABLES:
 
+# Create OPTIONS variable
+OPTIONS=
+
+OSARCH=$(shell uname -s)
+
+ifeq (${OSARCH},Linux)
+PROC=$(shell uname -m)
+ifeq ($(PROC),x86_64)
+# You must have GCC 3.4 to use k8, otherwise use athlon
+PROC=k8
+#PROC=athlon
+OPTIONS+=-m64
+endif
+ifeq ($(PROC),sparc64)
+#The problem with sparc is the best stuff is in newer versions of gcc (post 3.0) only.
+#This works for even old (2.96) versions of gcc and provides a small boost either way.
+#A ultrasparc cpu is really v9 but the stock debian stable 3.0 gcc doesn't support it.
+#So we go lowest common available by gcc and go a step down, still a step up from
+#the default as we now have a better instruction set to work with. - Belgarath
+PROC=ultrasparc
+OPTIONS+=$(shell if $(CC) -mtune=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-mtune=$(PROC)"; fi)
+OPTIONS+=$(shell if $(CC) -mcpu=v8 -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-mcpu=v8"; fi)
+OPTIONS+=-fomit-frame-pointer
+endif
+
+MPG123TARG=linux
+endif
+
+ifeq ($(findstring BSD,${OSARCH}),BSD)
+PROC=$(shell uname -m)
+endif
+
 # Pentium Pro Optimize
 #PROC=i686
-# Pentium Optimize
+
+# Pentium & VIA processors optimize
 #PROC=i586
+
 #PROC=k6
 #PROC=ppc
-PROC=$(shell uname -m)
+
+PWD=$(shell pwd)
 
 ######### More GSM codec optimization
 ######### Uncomment to enable MMXTM optimizations for x86 architecture CPU's
 ######### which support MMX instructions.  This should be newer pentiums,
 ######### ppro's, etc, as well as the AMD K6 and K7.  
-K6OPT  = #-DK6OPT
+#K6OPT  = -DK6OPT
 
 #Tell gcc to optimize the asterisk's code
-OPTIMIZE=-O6
+OPTIMIZE+=-O6
 
 #Include debug symbols in the executables (-g) and profiling info (-pg)
 DEBUG=-g #-pg
 
+# If you are running a radio application, define RADIO_RELAX so that the DTMF
+# will be received more reliably
+#OPTIONS += -DRADIO_RELAX
+
+# If you don't have a lot of memory (e.g. embedded Asterisk), uncomment the
+# following to reduce the size of certain static buffers
+#OPTIONS += -DLOW_MEMORY
+
 # Optional debugging parameters
-DEBUG_THREADS = #-DDO_CRASH -DDEBUG_THREADS
+DEBUG_THREADS = #-DDEBUG_THREADS #-DDO_CRASH 
 
 # Uncomment next one to enable ast_frame tracing (for debugging)
 TRACE_FRAMES = #-DTRACE_FRAMES
 
+# Uncomment next one to enable malloc debugging
+# You can view malloc debugging with:
+#   *CLI> show memory allocations [filename]
+#   *CLI> show memory summary [filename]
+#
+MALLOC_DEBUG = #-include $(PWD)/include/asterisk/astmm.h
+
 # Where to install asterisk after compiling
 # Default -> leave empty
 INSTALL_PREFIX=
 
+# Staging directory
+# Files are copied here temporarily during the install process
+# For example, make DESTDIR=/tmp/asterisk woud put things in
+# /tmp/asterisk/etc/asterisk
+DESTDIR=
+
+# Original busydetect routine
+BUSYDETECT = #-DBUSYDETECT
+
+# Improved busydetect routine, comment the previous one if you use this one
+BUSYDETECT+= -DBUSYDETECT_MARTIN 
+# Detect the busy signal looking only at tone lengths
+# For example if you have 3 beeps 100ms tone, 100ms silence separated by 500 ms of silence
+BUSYDETECT+= #-DBUSYDETECT_TONEONLY
+# Inforce the detection of busy singal (get rid of false hangups)
+# Don't use together with -DBUSYDETECT_TONEONLY
+BUSYDETECT+= #-DBUSYDETECT_COMPARE_TONE_AND_SILENCE
+
 ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk
 ASTVARLIBDIR=$(INSTALL_PREFIX)/var/lib/asterisk
 ASTETCDIR=$(INSTALL_PREFIX)/etc/asterisk
@@ -53,7 +121,7 @@ ASTCONFPATH=$(ASTETCDIR)/asterisk.conf
 ASTBINDIR=$(INSTALL_PREFIX)/usr/bin
 ASTSBINDIR=$(INSTALL_PREFIX)/usr/sbin
 ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run
-
+ASTMANDIR=$(INSTALL_PREFIX)/usr/share/man
 
 MODULES_DIR=$(ASTLIBDIR)/modules
 AGI_DIR=$(ASTVARLIBDIR)/agi-bin
@@ -61,12 +129,42 @@ AGI_DIR=$(ASTVARLIBDIR)/agi-bin
 INCLUDE=-Iinclude -I../include
 CFLAGS=-pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE #-DMAKE_VALGRIND_HAPPY
 CFLAGS+=$(OPTIMIZE)
+
+ifneq ($(PROC),ultrasparc)
 CFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
+endif
+
 CFLAGS+=$(shell if uname -m | grep -q ppc; then echo "-fsigned-char"; fi)
+CFLAGS+=$(shell if [ -f /usr/include/osp/osp.h ]; then echo "-DOSP_SUPPORT -I/usr/include/osp" ; fi)
+
+ifeq (${OSARCH},FreeBSD)
+OSVERSION=$(shell make -V OSVERSION -f /usr/share/mk/bsd.port.subdir.mk)
+CFLAGS+=$(shell if test ${OSVERSION} -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
+LIBS+=$(shell if test  ${OSVERSION} -lt 502102 ; then echo "-lc_r"; else echo "-pthread"; fi)
+INCLUDE+=-I/usr/local/include
+CFLAGS+=$(shell if [ -d /usr/local/include/spandsp ]; then echo "-I/usr/local/include/spandsp"; fi)
+MPG123TARG=freebsd
+endif # FreeBSD
+
+ifeq (${OSARCH},NetBSD)
+CFLAGS+=-pthread
+INCLUDE+=-I/usr/local/include -I/usr/pkg/include
+MPG123TARG=netbsd
+endif
+
+ifeq (${OSARCH},OpenBSD)
+CFLAGS+=-pthread
+endif
+
+#Uncomment this to use the older DSP routines
+#CFLAGS+=-DOLD_DSP_ROUTINES
+
+CFLAGS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo "-DZAPTEL_OPTIMIZATIONS"; fi)
+CFLAGS+=$(shell if [ -f /usr/local/include/zaptel.h ]; then echo "-DZAPTEL_OPTIMIZATIONS"; fi)
 
 LIBEDIT=editline/libedit.a
 
-ASTERISKVERSION=$(shell if [ -f .version ]; then cat .version; fi)
+ASTERISKVERSION=$(shell if [ -f .version ]; then cat .version; else if [ -d CVS ]; then if [ -f CVS/Tag ] ; then echo "CVS-`sed 's/^T//g' CVS/Tag`-`date +"%D-%T"`"; else echo "CVS-HEAD-`date +"%D-%T"`"; fi; fi; fi)
 HTTPDIR=$(shell if [ -d /var/www ]; then echo "/var/www"; else echo "/home/httpd"; fi)
 RPMVERSION=$(shell if [ -f .version ]; then sed 's/[-\/:]/_/g' .version; else echo "unknown" ; fi)
 CFLAGS+=-DASTERISK_VERSION=\"$(ASTERISKVERSION)\"
@@ -83,14 +181,58 @@ CFLAGS+=-DASTAGIDIR=\"$(AGI_DIR)\"
 
 CFLAGS+= $(DEBUG_THREADS)
 CFLAGS+= $(TRACE_FRAMES)
+CFLAGS+= $(MALLOC_DEBUG)
+CFLAGS+= $(BUSYDETECT)
+CFLAGS+= $(OPTIONS)
 CFLAGS+=# -fomit-frame-pointer 
-SUBDIRS=res channels pbx apps codecs formats agi cdr astman
-LIBS=-ldl -lpthread -lncurses -lm  #-lnjamd
+SUBDIRS=res channels pbx apps codecs formats agi cdr astman stdtime
+ifeq (${OSARCH},Linux)
+LIBS=-ldl -lpthread
+endif
+LIBS+=-lncurses -lm
+ifeq (${OSARCH},Linux)
+LIBS+=-lresolv  #-lnjamd
+endif
+ifeq (${OSARCH},Darwin)
+LIBS+=-lresolv
+endif
+ifeq (${OSARCH},FreeBSD)
+LIBS+=-lcrypto
+endif
+ifeq (${OSARCH},NetBSD)
+LIBS+=-lpthread -lcrypto -lm -L/usr/local/lib -L/usr/pkg/lib -lncurses
+endif
+ifeq (${OSARCH},OpenBSD)
+LIBS=-lcrypto -lpthread -lm -lncurses
+endif
+LIBS+=-lssl
 OBJS=io.o sched.o logger.o frame.o loader.o config.o channel.o \
 	translate.o file.o say.o pbx.o cli.o md5.o term.o \
 	ulaw.o alaw.o callerid.o fskmodem.o image.o app.o \
 	cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o \
-	dsp.o chanvars.o indications.o autoservice.o db.o privacy.o
+	dsp.o chanvars.o indications.o autoservice.o db.o privacy.o \
+	astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o \
+	utils.o 
+ifeq (${OSARCH},Darwin)
+ASTLINK=-Wl,-dynamic
+SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
+OBJS+= poll.o
+CFLAGS+=-DPOLLCOMPAT
+else
+ASTLINK=-Wl,-E 
+SOLINK=-shared -Xlinker -x
+endif
+
+ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/sys/poll.h),)
+  OBJS+= poll.o
+  CFLAGS+=-DPOLLCOMPAT
+endif
+
+ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/dlfcn.h),)
+  OBJS+= dhfcn.o
+  CFLAGS+=-DDLFCNCOMPAT
+endif
+
 CC=gcc
 INSTALL=install
 
@@ -100,32 +242,34 @@ _all: all
 	@echo " + cannot be run before being installed by   +"  
 	@echo " + running:                                  +"  
 	@echo " +                                           +"
-	@echo " +               make install                +"  
+	@echo " +               $(MAKE) install                +"  
 	@echo " +-------------------------------------------+"  
 
-all: asterisk subdirs
+all: depend asterisk subdirs 
 
 editline/config.h:
-	@if [ -d editline ]; then \
-		cd editline && unset CFLAGS LIBS && ./configure ; \
-	else \
-		echo "You need to do a cvs update -d not just cvs update"; \
-		exit 1; \
-	fi
+	cd editline && unset CFLAGS LIBS && ./configure ; \
 
-editline/libedit.a: editline/config.h
-	make -C editline libedit.a
+editline/libedit.a: FORCE
+	cd editline && unset CFLAGS LIBS && test -f config.h || ./configure
+	$(MAKE) -C editline libedit.a
 
-db1-ast/libdb1.a: 
+db1-ast/libdb1.a: FORCE
 	@if [ -d db1-ast ]; then \
-		make -C db1-ast libdb1.a ; \
+		$(MAKE) -C db1-ast libdb1.a ; \
 	else \
 		echo "You need to do a cvs update -d not just cvs update"; \
 		exit 1; \
 	fi
 
+ifneq ($(wildcard .depend),)
+include .depend
+endif
+
+.PHONY: _version
+
 _version: 
-	if [ -d CVS ] && ! [ -f .version ]; then echo "CVS-`date +"%D-%T"`" > .version; fi 
+	if [ -d CVS ] && ! [ -f .version ]; then echo $(ASTERISKVERSION) > .version; fi 
 
 .version: _version
 
@@ -134,79 +278,152 @@ _version:
 
 ast_expr.o: ast_expr.c
 
+cli.o: cli.c build.h
+
+asterisk.o: asterisk.c build.h
+
+manpage: asterisk.8.gz
+
+asterisk.8.gz: asterisk.sgml
+	rm -f asterisk.8
+	docbook2man asterisk.sgml
+	mv ./*.8 asterisk.8
+	gzip asterisk.8
+
+ifneq ($(strip $(ASTERISKVERSION)),)
+build.h: .version
+	./make_build_h
+else
 build.h:
 	./make_build_h
+endif
 
-asterisk: .version build.h editline/libedit.a db1-ast/libdb1.a $(OBJS)
-	gcc -o asterisk -rdynamic $(OBJS) $(LIBS) $(LIBEDIT) db1-ast/libdb1.a
+stdtime/libtime.a: FORCE
+	@if [ -d stdtime ]; then \
+		$(MAKE) -C stdtime libtime.a ; \
+	else \
+		echo "You need to do a cvs update -d not just cvs update"; \
+		exit 1; \
+	fi
+
+asterisk: editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a $(OBJS)
+	$(CC) $(DEBUG) -o asterisk $(ASTLINK) $(OBJS) $(LIBEDIT) db1-ast/libdb1.a stdtime/libtime.a $(LIBS)
+
+muted: muted.o
+	$(CC) -o muted muted.o
 
 subdirs: 
 	for x in $(SUBDIRS); do $(MAKE) -C $$x || exit 1 ; done
 
 clean:
 	for x in $(SUBDIRS); do $(MAKE) -C $$x clean || exit 1 ; done
-	rm -f *.o *.so asterisk
+	rm -f *.o *.so asterisk .depend
 	rm -f build.h 
 	rm -f ast_expr.c
-	@if [ -e editline/Makefile ]; then make -C editline clean ; fi
-	make -C db1-ast clean
+	@if [ -e editline/Makefile ]; then $(MAKE) -C editline distclean ; fi
+	@if [ -d mpg123-0.59r ]; then $(MAKE) -C mpg123-0.59r clean; fi	
+	$(MAKE) -C db1-ast clean
+	$(MAKE) -C stdtime clean
 
 datafiles: all
-	mkdir -p $(ASTVARLIBDIR)/sounds/digits
+	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/digits
 	for x in sounds/digits/*.gsm; do \
 		if grep -q "^%`basename $$x`%" sounds.txt; then \
-			install $$x $(ASTVARLIBDIR)/sounds/digits ; \
+			install -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds/digits ; \
 		else \
 			echo "No description for $$x"; \
 			exit 1; \
 		fi; \
 	done
-	for x in sounds/vm-* sounds/transfer* sounds/pbx-* sounds/ss-* sounds/beep* sounds/dir-* sounds/conf-* sounds/agent-* sounds/invalid* sounds/tt-* sounds/auth-* sounds/privacy-*; do \
+	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/letters
+	for x in sounds/letters/*.gsm; do \
 		if grep -q "^%`basename $$x`%" sounds.txt; then \
-			install $$x $(ASTVARLIBDIR)/sounds ; \
+			install -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds/letters ; \
 		else \
 			echo "No description for $$x"; \
 			exit 1; \
 		fi; \
 	done
-	mkdir -p $(ASTVARLIBDIR)/mohmp3
-	mkdir -p $(ASTVARLIBDIR)/images
-	for x in images/*.jpg; do \
-		install $$x $(ASTVARLIBDIR)/images ; \
+	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/phonetic
+	for x in sounds/phonetic/*.gsm; do \
+		if grep -q "^%`basename $$x`%" sounds.txt; then \
+			install -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds/phonetic ; \
+		else \
+			echo "No description for $$x"; \
+			exit 1; \
+		fi; \
 	done
-	mkdir -p $(AGI_DIR)
+	for x in sounds/vm-* sounds/transfer* sounds/pbx-* sounds/ss-* sounds/beep* sounds/dir-* sounds/conf-* sounds/agent-* sounds/invalid* sounds/tt-* sounds/auth-* sounds/privacy-* sounds/queue-* sounds/hello-*; do \
+		if grep -q "^%`basename $$x`%" sounds.txt; then \
+			install -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds ; \
+		else \
+			echo "No description for $$x"; \
+			exit 1; \
+		fi; \
+	done
+	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/mohmp3
+	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/images
+	for x in images/*.jpg; do \
+		install -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/images ; \
+	done
+	mkdir -p $(DESTDIR)$(AGI_DIR)
 
 update: 
 	@if [ -d CVS ]; then \
 		echo "Updating from CVS..." ; \
-		cvs update -d; \
+		cvs -q -z3 update -Pd; \
 		rm -f .version; \
 	else \
 		echo "Not CVS";  \
 	fi
 
 bininstall: all
-	mkdir -p $(MODULES_DIR)
-	mkdir -p $(ASTSBINDIR)
-	mkdir -p $(ASTETCDIR)
-	mkdir -p $(ASTBINDIR)
-	mkdir -p $(ASTSBINDIR)
-	mkdir -p $(ASTVARRUNDIR)
-	install -m 755 asterisk $(ASTSBINDIR)/
-	install -m 755 astgenkey $(ASTSBINDIR)/
-	install -m 755 safe_asterisk $(ASTSBINDIR)/
+	mkdir -p $(DESTDIR)$(MODULES_DIR)
+	mkdir -p $(DESTDIR)$(ASTSBINDIR)
+	mkdir -p $(DESTDIR)$(ASTETCDIR)
+	mkdir -p $(DESTDIR)$(ASTBINDIR)
+	mkdir -p $(DESTDIR)$(ASTSBINDIR)
+	mkdir -p $(DESTDIR)$(ASTVARRUNDIR)
+	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/voicemail
+	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/tmp
+	install -m 755 asterisk $(DESTDIR)$(ASTSBINDIR)/
+	install -m 755 contrib/scripts/astgenkey $(DESTDIR)$(ASTSBINDIR)/
+	install -m 755 contrib/scripts/autosupport $(DESTDIR)$(ASTSBINDIR)/
+	if [ ! -f $(DESTDIR)$(ASTSBINDIR)/safe_asterisk ]; then \
+		install -m 755 contrib/scripts/safe_asterisk $(DESTDIR)$(ASTSBINDIR)/ ;\
+	fi
 	for x in $(SUBDIRS); do $(MAKE) -C $$x install || exit 1 ; done
-	install -d $(ASTHEADERDIR)
-	install include/asterisk/*.h $(ASTHEADERDIR)
-	rm -f $(ASTVARLIBDIR)/sounds/vm
-	mkdir -p $(ASTSPOOLDIR)/vm
-	rm -f $(ASTMODULESDIR)/chan_ixj.so
-	rm -f $(ASTMODULESDIR)/chan_tor.so
-	mkdir -p $(ASTVARLIBDIR)/sounds
-	mkdir -p $(ASTLOGDIR)/cdr-csv
-	mkdir -p $(ASTVARLIBDIR)/keys
-	install -m 644 keys/iaxtel.pub $(ASTVARLIBDIR)/keys
-	( cd $(ASTVARLIBDIR)/sounds  ; ln -s $(ASTSPOOLDIR)/vm . )
+	install -d $(DESTDIR)$(ASTHEADERDIR)
+	install -m 644 include/asterisk/*.h $(DESTDIR)$(ASTHEADERDIR)
+	rm -f $(DESTDIR)$(ASTVARLIBDIR)/sounds/vm
+	rm -f $(DESTDIR)$(ASTVARLIBDIR)/sounds/voicemail
+	if [ ! -h $(DESTDIR)$(ASTSPOOLDIR)/vm ] && [ -d $(DESTDIR)$(ASTSPOOLDIR)/vm ]; then \
+		mv $(DESTDIR)$(ASTSPOOLDIR)/vm $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default; \
+	else \
+		mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default; \
+		rm -f $(DESTDIR)$(ASTSPOOLDIR)/vm; \
+	fi
+	ln -s $(ASTSPOOLDIR)/voicemail/default $(DESTDIR)$(ASTSPOOLDIR)/vm
+	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds
+	mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-csv
+	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/keys
+	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/firmware
+	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/firmware/iax
+	mkdir -p $(DESTDIR)$(ASTMANDIR)/man8
+	install -m 644 keys/iaxtel.pub $(DESTDIR)$(ASTVARLIBDIR)/keys
+	install -m 644 keys/freeworlddialup.pub $(DESTDIR)$(ASTVARLIBDIR)/keys
+	install -m 644 asterisk.8.gz $(DESTDIR)$(ASTMANDIR)/man8
+	install -m 644 contrib/scripts/astgenkey.8 $(DESTDIR)$(ASTMANDIR)/man8
+	install -m 644 contrib/scripts/autosupport.8 $(DESTDIR)$(ASTMANDIR)/man8
+	install -m 644 contrib/scripts/safe_asterisk.8 $(DESTDIR)$(ASTMANDIR)/man8
+	if [ -d contrib/firmware/iax ]; then \
+		install -m 644 contrib/firmware/iax/iaxy.bin $(DESTDIR)$(ASTVARLIBDIR)/firmware/iax/iaxy.bin; \
+	else \
+		echo "You need to do cvs update -d not just cvs update" ; \
+	fi 
+	( cd $(DESTDIR)$(ASTVARLIBDIR)/sounds  ; ln -s $(ASTSPOOLDIR)/vm . )
+	( cd $(DESTDIR)$(ASTVARLIBDIR)/sounds  ; ln -s $(ASTSPOOLDIR)/voicemail . )
+	if [ -f mpg123-0.59r/mpg123 ]; then $(MAKE) -C mpg123-0.59r install; fi
 	@echo " +---- Asterisk Installation Complete -------+"  
 	@echo " +                                           +"
 	@echo " +    YOU MUST READ THE SECURITY DOCUMENT    +"
@@ -216,76 +433,90 @@ bininstall: all
 	@echo " + configuration files (overwriting any      +"
 	@echo " + existing config files), run:              +"  
 	@echo " +                                           +"
-	@echo " +               make samples                +"
+	@echo " +               $(MAKE) samples                +"
 	@echo " +                                           +"
 	@echo " +-----------------  or ---------------------+"
 	@echo " +                                           +"
 	@echo " + You can go ahead and install the asterisk +"
 	@echo " + program documentation now or later run:   +"
 	@echo " +                                           +"
-	@echo " +              make progdocs                +"
+	@echo " +              $(MAKE) progdocs                +"
 	@echo " +                                           +"
 	@echo " + **Note** This requires that you have      +"
 	@echo " + doxygen installed on your local system    +"
 	@echo " +-------------------------------------------+"
+	@echo " +                                           +"
+	@echo " + ** NOTE FOR DOWNGRADING FROM CVS HEAD **  +"
+	@echo " +                                           +"
+	@echo " + If you are downgrading from CVS HEAD to   +"
+	@echo " + a stable release, remember to delete      +"
+	@echo " + everything from your asterisk modules     +"
+	@echo " + directory (/usr/lib/asterisk/modules/)    +"
+	@echo " + and the asterisk header directory         +"
+	@echo " + (/usr/include/asterisk/)                  +"
+	@echo " + before doing a '$(MAKE) install'.            +"
+	@echo " +                                           +"
+	@echo " +-------------------------------------------+"
+
 
 install: all datafiles bininstall
 
 upgrade: all bininstall
 
 adsi: all
-	mkdir -p $(ASTETCDIR)
+	mkdir -p $(DESTDIR)$(ASTETCDIR)
 	for x in configs/*.adsi; do \
-		if ! [ -f $(ASTETCDIRX)/$$x ]; then \
-			install -m 644 $$x $(ASTETCDIR)/`basename $$x` ; \
+		if ! [ -f $(DESTDIR)$(ASTETCDIRX)/$$x ]; then \
+			install -m 644 $$x $(DESTDIR)$(ASTETCDIR)/`basename $$x` ; \
 		fi ; \
 	done
 
 samples: all datafiles adsi
-	mkdir -p $(ASTETCDIR)
+	mkdir -p $(DESTDIR)$(ASTETCDIR)
 	for x in configs/*.sample; do \
-		if [ -f $(ASTETCDIR)/`basename $$x .sample` ]; then \
-			mv -f $(ASTETCDIR)/`basename $$x .sample` $(ASTETCDIR)/`basename $$x .sample`.old ; \
+		if [ -f $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` ]; then \
+			mv -f $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample`.old ; \
 		fi ; \
-		install $$x $(ASTETCDIR)/`basename $$x .sample` ;\
+		install -m 644 $$x $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` ;\
 	done
-	echo "[directories]" > $(ASTETCDIR)/asterisk.conf
-	echo "astetcdir => $(ASTETCDIR)" >> $(ASTETCDIR)/asterisk.conf
-	echo "astmoddir => $(MODULES_DIR)" >> $(ASTETCDIR)/asterisk.conf
-	echo "astvarlibdir => $(ASTVARLIBDIR)" >> $(ASTETCDIR)/asterisk.conf
-	echo "astagidir => $(AGI_DIR)" >> $(ASTETCDIR)/asterisk.conf
-	echo "astspooldir => $(ASTSPOOLDIR)" >> $(ASTETCDIR)/asterisk.conf
-	echo "astrundir => $(ASTVARRUNDIR)" >> $(ASTETCDIR)/asterisk.conf
-	echo "astlogdir => $(ASTLOGDIR)" >> $(ASTETCDIR)/asterisk.conf
+	echo "[directories]" > $(DESTDIR)$(ASTETCDIR)/asterisk.conf
+	echo "astetcdir => $(ASTETCDIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf
+	echo "astmoddir => $(MODULES_DIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf
+	echo "astvarlibdir => $(ASTVARLIBDIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf
+	echo "astagidir => $(AGI_DIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf
+	echo "astspooldir => $(ASTSPOOLDIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf
+	echo "astrundir => $(ASTVARRUNDIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf
+	echo "astlogdir => $(ASTLOGDIR)" >> $(DESTDIR)$(ASTETCDIR)/asterisk.conf
 	for x in sounds/demo-*; do \
 		if grep -q "^%`basename $$x`%" sounds.txt; then \
-			install $$x $(ASTVARLIBDIR)/sounds ; \
+			install -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds ; \
 		else \
 			echo "No description for $$x"; \
 			exit 1; \
 		fi; \
 	done
 	for x in sounds/*.mp3; do \
-		install $$x $(ASTVARLIBDIR)/mohmp3 ; \
+		install -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/mohmp3 ; \
 	done
-	mkdir -p $(ASTSPOOLDIR)/vm/1234/INBOX
-	:> $(ASTVARLIBDIR)/sounds/vm/1234/unavail.gsm
+	rm -f $(DESTDIR)$(ASTVARLIBDIR)/mohmp3/sample-hold.mp3
+	mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/INBOX
+	:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm
 	for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
-		cat $(ASTVARLIBDIR)/sounds/$$x.gsm >> $(ASTVARLIBDIR)/sounds/vm/1234/unavail.gsm ; \
+		cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm ; \
 	done
-	:> $(ASTVARLIBDIR)/sounds/vm/1234/busy.gsm
+	:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/busy.gsm
 	for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isonphone; do \
-		cat $(ASTVARLIBDIR)/sounds/$$x.gsm >> $(ASTVARLIBDIR)/sounds/vm/1234/busy.gsm ; \
+		cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/busy.gsm ; \
 	done
 
 webvmail:
-	@[ -d $(HTTPDIR) ] || ( echo "No HTTP directory" && exit 1 )
-	@[ -d $(HTTPDIR)/html ] || ( echo "No http directory" && exit 1 )
-	@[ -d $(HTTPDIR)/cgi-bin ] || ( echo "No cgi-bin directory" && exit 1 )
-	install -m 4755 -o root -g root vmail.cgi $(HTTPDIR)/cgi-bin/vmail.cgi
-	mkdir -p $(HTTPDIR)/html/_asterisk
+	@[ -d $(DESTDIR)$(HTTPDIR) ] || ( echo "No HTTP directory" && exit 1 )
+	@[ -d $(DESTDIR)$(HTTPDIR)/html ] || ( echo "No http directory" && exit 1 )
+	@[ -d $(DESTDIR)$(HTTPDIR)/cgi-bin ] || ( echo "No cgi-bin directory" && exit 1 )
+	install -m 4755 -o root -g root contrib/scripts/vmail.cgi $(DESTDIR)$(HTTPDIR)/cgi-bin/vmail.cgi
+	mkdir -p $(DESTDIR)$(HTTPDIR)/html/_asterisk
 	for x in images/*.gif; do \
-		install -m 644 $$x $(HTTPDIR)/html/_asterisk/; \
+		install -m 644 $$x $(DESTDIR)$(HTTPDIR)/html/_asterisk/; \
 	done
 	@echo " +--------- Asterisk Web Voicemail ----------+"  
 	@echo " +                                           +"
@@ -297,38 +528,61 @@ webvmail:
 	@echo " +-------------------------------------------+"  
 
 mailbox:
-	./addmailbox 
-	
+	./contrib/scripts/addmailbox 
 
 rpm: __rpm
 
 __rpm: _version
 	rm -rf /tmp/asterisk ; \
 	mkdir -p /tmp/asterisk/redhat/RPMS/i386 ; \
-	make INSTALL_PREFIX=/tmp/asterisk install ; \
-	make INSTALL_PREFIX=/tmp/asterisk samples ; \
+	$(MAKE) DESTDIR=/tmp/asterisk install ; \
+	$(MAKE) DESTDIR=/tmp/asterisk samples ; \
 	mkdir -p /tmp/asterisk/etc/rc.d/init.d ; \
 	cp -f redhat/asterisk /tmp/asterisk/etc/rc.d/init.d/ ; \
-	cp -f redhat/rpmrc /tmp/asterisk/ ; \
-	cp -f redhat/rpmmacros /tmp/asterisk/ ; \
-	sed "s/Version:/Version: $(RPMVERSION)/g" redhat/asterisk.spec > /tmp/asterisk/asterisk.spec ; \
-	rpm --rcfile /usr/lib/rpm/rpmrc:/tmp/asterisk/rpmrc -bb /tmp/asterisk/asterisk.spec ; \
-	mv /tmp/asterisk/redhat/RPMS/i386/asterisk* ./ ; \
-	rm -rf /tmp/asterisk
+	sed "s/^Version:.*/Version: $(RPMVERSION)/g" redhat/asterisk.spec > asterisk.spec ; \
+	rpmbuild --rcfile /usr/lib/rpm/rpmrc:redhat/rpmrc -bb asterisk.spec
 
 progdocs:
-	doxygen asterisk-ng-doxygen
+	doxygen contrib/asterisk-ng-doxygen
+
+mpg123:
+	@wget -V >/dev/null || (echo "You need wget" ; false )
+	[ -f mpg123-0.59r.tar.gz ] || wget http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz
+	[ -d mpg123-0.59r ] || tar xfz mpg123-0.59r.tar.gz
+	$(MAKE) -C mpg123-0.59r $(MPG123TARG)
 
 config:
 	if [ -d /etc/rc.d/init.d ]; then \
-		install -m 755 init.asterisk /etc/rc.d/init.d/asterisk; \
+		install -m 755 contrib/init.d/rc.redhat.asterisk /etc/rc.d/init.d/asterisk; \
 		/sbin/chkconfig --add asterisk; \
 	elif [ -d /etc/init.d ]; then \
 		install -m 755 init.asterisk /etc/init.d/asterisk; \
 	fi 
 
-	
 dont-optimize:
-	make OPTIMIZE= K6OPT= install
+	$(MAKE) OPTIMIZE= K6OPT= install
 
 valgrind: dont-optimize
+
+depend: .depend
+	for x in $(SUBDIRS); do $(MAKE) -C $$x depend || exit 1 ; done
+
+.depend:
+	@if ! which mpg123 &>/dev/null ; then \
+		echo "*** You don't have mpg123 installed. You're going to need ***";\
+		echo "***       it if you want MusicOnHold                      ***";\
+	elif ! mpg123 --longhelp 2>&1 | grep .59r &>/dev/null ; then \
+			echo "*************************************************************";\
+			echo "*** You have the WRONG version of mpg123... you need .59r ***";\
+			echo "*** Use 'make mpg123' to get the right verison            ***";\
+			echo "*************************************************************";\
+	fi
+	./mkdep ${CFLAGS} `ls *.c`
+
+FORCE:
+
+%_env:
+	$(MAKE) -C $(shell echo $@ | sed "s/_env//g") env
+
+env:
+	env

README

@@ -1,6 +1,6 @@
 The Asterisk Open Source PBX
-by Mark Spencer <markster@linux-support.net>
-Copyright (C) 2001, Linux Support Services, Inc.
+by Mark Spencer <markster@digium.com>
+Copyright (C) 2001-2004 Digium
 ================================================================
 * SECURITY
   It is imperative that you read and fully understand the contents of
@@ -12,54 +12,61 @@ sense, middleware between Internet and telephony channels on the bottom,
 and Internet and telephony applications at the top.  For more information
 on the project itself, please visit the Asterisk home page at:
 
-           http://www.asteriskpbx.com
+           http://www.asterisk.org
+
+In addition you'll find lot's of information compiled by the Asterisk
+community on this Wiki:
+
+           http://www.voip-info.org/wiki-Asterisk
 
 * LICENSING
   Asterisk is distributed under GNU General Public License.  The GPL also
 must apply to all loadable modules as well, except as defined below.
 
-  Linux Support Services, Inc. retains copyright to all of the core
-Asterisk system, and therefore can grant, at its sole discression, the
-ability for companies, individuals, or organizations to create proprietary
+  Digium, Inc. (formerly Linux Support Services) retains copyright to all 
+of the core Asterisk system, and therefore can grant, at its sole discretion, 
+the ability for companies, individuals, or organizations to create proprietary
 or Open Source (but non-GPL'd) modules which may be dynamically linked at
 runtime with the portions of Asterisk which fall under our copyright
-umbrella, or are distributed under more flexible licenses than GPL.  At
-this time (5/21/2001) the only component of Asterisk which is covered
-under GPL and not under our Copyright is the Xing MP3 decoder.
+umbrella, or are distributed under more flexible licenses than GPL.  
+
 
   If you wish to use our code in other GPL programs, don't worry -- there
 is no requirement that you provide the same exemption in your GPL'd
 products (although if you've written a module for Asterisk we would
-strongly encourage you to make the same excemption that we do).
+strongly encourage you to make the same exemption that we do).
 
   Specific permission is also granted to OpenSSL and OpenH323 to link to
 Asterisk.
 
   If you have any questions, whatsoever, regarding our licensing policy,
 please contact us.
+
+  Modules that are GPL-licensed and not available under Digium's 
+licensing scheme are added to the Asterisk-addons CVS module.
   
 * REQUIRED COMPONENTS
 
 == Linux ==
   Currently, the Asterisk Open Source PBX is only known to run on the
 Linux OS, although it may be portable to other UNIX-like operating systems
-as well.
+(like FreeBSD) as well. 
 
 
 * GETTING STARTED
 
-First, be sure you've got supported hardware.  To use Asterisk right now,
-you will need one of the following:
+First, be sure you've got supported hardware (but note that you don't need ANY hardware, not even a soundcard) to install and run Asterisk. Supported are:
 
-	* All Wildcard (tm) products from LSS (www.linux-support.net)
+	* All Wildcard (tm) products from Digium (www.digium.com)
 	* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
 	* Full Duplex Sound Card supported by Linux
 	* Adtran Atlas 800 Plus
 	* ISDN4Linux compatible ISDN card
 	* Tormenta Dual T1 card (www.bsdtelephony.com.mx)
 
-Assuming you have one of these (most likely the third) you're ready to 
-proceed:
+Hint: CAPI compatible ISDN cards can be run using the add-on channel chan_capi.
+
+So let's proceed:
 
 1) Run "make"
 2) Run "make install"
@@ -69,7 +76,7 @@ the sample PBX, with demonstration extensions, etc.  If so, run:
 
 	"make samples"
 
-Doing so will overwrite any existing config files you have.
+Doing so will overwrite any existing config files you have. If you are lacking a soundcard you won't be able to use the DIAL command on the console, though.
 
 Finally, you can launch Asterisk with:
 
@@ -129,9 +136,78 @@ parameters.  For example, the line "channel => 25-47" creates objects for
 the channels 25 through 47 of the tormenta card, obtaining the settings
 from the variables specified above.
 
+* SPECIAL NOTE ON TIME
+  
+  Those using SIP phones should be aware the Asterisk is sensitive to
+large jumps in time.  Manually changing the system time using date(1)
+(or other similar commands) may cause SIP registrations and other
+internal processes to fail.  If your system cannot keep accurate time
+by itself use NTP (http://www.ntp.org/) to keep the system clock
+synchronized to "real time".  NTP is designed to keep the system clock
+synchronized by speeding up or slowing down the system clock until it
+is synchronized to "real time" rather than by jumping the time and
+causing discontinuities. Most Linux distributions include precompiled
+versions of NTP.  Beware of some time synchronization methods that get
+the correct real time periodically and then manually set the system
+clock.
+
+  Apparent time changes due to daylight savings time are just that,
+apparent.  The use of daylight savings time in a Linux system is
+purely a user interface issue and does not affect the operation of the
+Linux kernel or Asterisk.  The system clock on Linux kernels operates
+on UTC.  UTC does not use daylight savings time.
+   
+  Also note that this issue is separate from the clocking of TDM
+channels, and is known to at least affect SIP registrations.
+
+* FILE DESCRIPTORS
+
+  Depending on the size of your system and your configuration,
+Asterisk can consume a large number of file descriptors.  In UNIX,
+file descriptors are used for more than just files on disk.  File
+descriptors are also used for handling network communication
+(e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and
+digital trunk hardware).  Asterisk accesses many on-disk files for
+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 approxiately 150
+SIP calls simultaneously.  To change the number of file descriptors
+follow the instructions for your system below:
+
+== 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:
+
+root            soft    nofile          4096
+root            hard    nofile          8196
+asterisk        soft    nofile          4096
+asterisk        hard    nofile          8196
+
+(adjust the numbers to taste).  You may need to reboot the system for
+these changes to take effect.
+
+== Generic UNIX System ==
+
+  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
 
+See the doc directory for more documentation.
+
 Finally, you may wish to visit the web site and join the mailing list if
 you're interested in getting more information.
 
-Mark
+   http://www.asterisk.org/index.php?menu=support
+
+Welcome to the growing worldwide community of Asterisk users!
+
+Mark Spencer
+
+

README.cdr

@@ -1,14 +0,0 @@
-Asterisk now generates Call Detail Records.  See include/asterisk/cdr.h for
-all the fields which are recorded.  By default, records in comma-separated
-values will be created in /var/log/asterisk/cdr-csv.  You can specify 
-account codes and AMA (Automated Machine Accounting) flags on a per-channel
-(Zaptel et al) or per-user (IAX) basis to help with accounting.  Look
-at the top of cdr/cdr_csv.c to see the format for the records.  
-
-ONE IMPORTANT NOTE: If you are trying to collect records on IAX to IAX calls
-you need to be aware that by default, IAX will attempt to transfer calls
-in this situation (if DTMF is not required).  When the transfer is completed
-the call is dumped from the middle machine and thus the call detail records
-will report a short call time.  If you want detailed records you must
-turn off IAX transfer, but unless your servers are very close together, you
-will definitely get a latency hit from doing so.

README.festival

@@ -1,22 +0,0 @@
-
-app_festival is an application that allows one to send text-to-speech commands
-to a background festival server, and to obtain the resulting waveform which
-gets sent down to the respective channel. app_festival also employs a waveform 
-cache, so invariant text-to-speech strings ("Please press 1 for instructions") 
-do not need to be dynamically generated all the time. 
-
-You need : 
-
-1) festival, patched to produce 8khz waveforms on output. Patch for Festival
-1.4.2 RELEASE are included. The patch adds a new command to festival 
-(asterisk_tts). 
-
-2) My patches to asterisk that provide variable substitution and quoting to 
-the Asterisk Extension Logic. This is not really a requirement, but without
-this, app_festival is mostly useless (you could very well use prerecorded
-voices for static information). 
-
-3) Before running asterisk, you have to run festival-server with a command 
-like : 
-
-/usr/local/festival/bin/festival --server > /dev/null 2>&1 &

README.iax

@@ -1,326 +0,0 @@
-Inter-Asterisk eXchange Protocol
-================================
-
-INTRODUCTION
-------------
-
-This document is intended as an introduction to the Inter-Asterisk 
-eXchange (or simply IAX) protocol.  It provides both a theoretical 
-background and practical information on its use.
-
-WHY IAX
--------
-The first question most people are thinking at this point is "Why do you 
-need another VoIP protocol?  Why didn't you just use SIP or H.323?"
-
-Well, the answer is a fairly complicated one, but in a nutshell it's like
-this...  Asterisk is intended as a very flexible and powerful
-communications tool.  As such, the primary feature we need from a VoIP
-protocol is the ability to meet our own goals with Asterisk, and one with
-enough flexibility that we could use it as a kind of laboratory for
-inventing and implementing new concepts in the field.  Neither H.323 or
-SIP fit the roles we needed, so we developed our own protocol, which,
-while not standards based, provides a number of advantages over both SIP
-and H.323, some of which are:
-
-	* Interoperability with NAT/PAT/Masquerade firewalls
-	     IAX seamlessly interoperates through all sorts of NAT and PAT
-             and other firewalls, including the ability to place and 
-             receive calls, and transfer calls to other stations.
-
-	* High performance, low overhead protocol
-	     When running on low-bandwidth connections, or when running 
-	     large numbers of calls, optimized bandwidth utilization is 
-	     imperitive.  IAX uses only 4 bytes of overhead
-
-	* Internationalization support
-	     IAX transmits language information, so that remote PBX 
-	     content can be delivered in the native language of the
-	     calling party.
-
-	* Remote dialplan polling
-	     IAX allows a PBX or IP phone to poll the availability of a 
-	     number from a remote server.  This allows PBX dialplans to 
-	     be centralized.
-
-	* Flexible authentication
-	     IAX supports cleartext, md5, and RSA authentication, 
-	     providing flexible security models for outgoing calls and 
-	     registration services.
-	
-	* Multimedia protocol
-	     IAX supports the transmission of voice, video, images, text, 
-	     HTML, DTMF, and URL's.  Voice menus can be presented in both
-	     audibly and visually.
-
-	* Call statistic gathering
-	     IAX gathers statistics about network performance (including 
-	     latency and jitter, as well as providing end-to-end latency
-	     measurement.
-
-	* Call parameter communication
-	     Caller*ID, requested extension, requested context, etc are
-	     all communicated through the call.
-
-	* Single socket design
-	     IAX's single socket design allows up to 32768 calls to be 
-	     multiplexed.
-	
-While we value the importance of standards based (i.e. SIP) call handling, 
-hopefully this will provide a reasonable explanation of why we developed 
-IAX rather than starting with SIP.
-
-CONFIG FILE CONVENTIONS
------------------------
-Lines beginning with '>' represent lines which might appear in an actual 
-configuration file.  The '>' is used to help separate them from the 
-descriptive text and should not actually be included in the file itself.
-
-Lines within []'s by themselves represent section labels within the 
-configuration file.  like this:
-
-> [mysection]
-
-Options are set using the "=" sign, for example
-
-> myoption = value
-
-Sometimes an option will have a number of discrete values which it can 
-take.  In that case, in the documentation, the options will be listed 
-within square brackets (the "[" and "]" ones) separated by the pipe symbol 
-("|").  For example:
-
-> myoption = [value1|value2|value3]
-
-means the option "myoption" can be assigned a value of "value1", "value2", 
-or "value3".
-
-Objects, or pseudo-objects are instantiated using the "=>" construct.  For 
-example:
-
-> myobject => parameter
-
-creates an object called "myobject" with some parameter whose definition
-would be specific to that object.  Note that the config file parser
-considers "=>" and "=" to be equivalent and their use is purely to make
-configuration files more readable and easier to "humanly parse".
-
-The comment character in Asterisk configuration files is the semicolon 
-";".  The reason it is not "#" is because the "#" symbol can be used as 
-parts of extensions and it didn't seem like a good idea to have to escape 
-it.
-
-IAX CONFIGURATION IN ASTERISK
------------------------------
-
-Like everything else in Asterisk, IAX's configuration lies in 
-/etc/asterisk -- specifically /etc/asterisk/iax.conf
-
-The IAX configuration file is a collection of sections, each of which
-(with the exception of the "general" section) represents an entity within 
-the IAX scope.
-
-------------
-
-The first section is typically the "general" section.  In this area, 
-a number of parameters which affect the entire system are configured.  
-Specifically, the default codecs, port and address, jitter behavior, TOS 
-bits, and registrations.
-
-The first line of the "general" section is always:
-
-> [general]
-
-Following the first line are a number of other possibilities:
-
-> port = <portnum>
-
-This sets the port that IAX will bind to.  The default IAX port number is 
-5036.  It is recommended that this value not be altered in general.
-
-> bindaddr = <ipaddr>
-
-This allows you to bind IAX to a specific local IP address instead of
-binding to all addresses.  This could be used to enhance security if, for
-example, you only wanted IAX to be available to users on your LAN.
-
-> bandwidth = [low|medium|high]
-
-The bandwidth selection initializes the codec selection to appropriate
-values for given bandwidths.  The "high" selection enables all codecs and
-is recommended only for 10Mbps or higher connections.  The "medium"
-bandwidth eliminates signed linear, Mu-law and A-law codecs, leaving only
-the codecs which are 32kbps and smaller (with MP3 as a special case).  It
-can be used with broadband connections if desired. "low" eliminates ADPCM
-and MP3 formats, leaving only the G.723.1, GSM, and LPC10.
-
-> allow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]
-> disallow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]
-
-The "allow" and "disallow" allow you to fine tune the codec selection 
-beyond the initial bandwidth selection on a codec-by-codec basis.  
-
-The recommended configuration is to select "low" bandwidth and then 
-disallow the LPC10 codec just because it doesn't sound very good. 
-
-> jitterbuffer = [yes|no]
-> dropcount = <dropamount>
-> maxjitterbuffer = <max>
-> maxexcessbuffer = <max>
-
-These parameters control the operation of the jitter buffer.  The 
-jitterbuffer should always be enabled unless you expect all your 
-connections to be over a LAN.  The drop count is the maximum number of 
-voice packets to allow to drop (out of 100).  Useful values are 3-10.  The 
-maxjitterbuffer is the maximum amount of jitter buffer to permit to be 
-used.  The "maxexcessbuffer" is the maximum amount of excess jitter buffer 
-that is permitted before the jitter buffer is slowly shrunk to eliminate 
-latency.
-
-> accountcode = <code>
-> amaflags = [default|omit|billing|documentation]
-
-These parameters affect call detail record generation.  The first sets the 
-account code for records received with IAX.  The account code can be 
-overridden on a per-user basis for incoming calls (see below).  The 
-amaflags controls how the record is labeled ("omit" causes no record to be 
-written.  "billing" and "documentation" label the records as billing or 
-documentation records respectively, and "default" selects the system 
-default.
-
-> tos = [lowdelay|throughput|reliability|mincost|none]
-
-IAX can optionally set the TOS (Type of Service) bits to specified values 
-to help improve performance in routing.  The recommended value is 
-"lowdelay", which many routers (including any Linux routers with 2.4 
-kernels that have not been altered with ip tables) will give priority to 
-these packets, improving voice quality.
-
-> register => <name>[:<secret>]@<host>[:port]
-
-Any number of registery entries may be instantiated in the general 
-section.  Registration allows Asterisk to notify a remote Asterisk server 
-(with a fixed address) what our current address is.  In order for 
-registration to work, the remote Asterisk server will need to have a 
-dynamic peer entry with the same name (and secret if provided).  
-
-The name is a required field, and is the remote peer name that we wish to 
-identify ourselves as.  A secret may be provided as well.  The secret is 
-generally a shared password between the local server and the remote 
-server.  However, if the secret is in square brackets ([]'s) then it is 
-interpreted as the name of a key to use.  In that case, the local Asterisk 
-server must have the *private* key (/var/lib/asterisk/keys/<name>.key) and 
-the remote server will have to have the corresponding public key.
-
-The "host" is a required field and is the hostname or IP address of the 
-remote Asterisk server.  The port specification is optional and is by 
-default 5036 if not specified.
-
--------------
-
-The following sections, after "general" define either users, peers or
-friends.  A "user" is someone who connects to us.  A "peer" is someone
-that we connect to.  A "friend" is simply shorthand for creating a "user" 
-and "peer" with identical parameters (i.e. someone who can contact us and 
-who we contact). 
-
-> [identifier]
-
-The section begins with the identifier in square brackets.  The identifier 
-should be an alphanumeric string.
-
-> type = [user|peer|friend]
-
-This line tells Asterisk how to interpret this entity.  Users are things 
-that connect to us, while peers are people we connect to, and a friend is 
-shorthand for creating a user and a peer with identical information
-
-----------------
-User fields:
-
-> context = <context>
-
-One or more context lines may be specified in a user, thus giving the user 
-access to place calls in the given contexts.  Contexts are used by 
-Asterisk to divide dialing plans into logical units each with the ability 
-to have numbers interpreted differently, have their own security model, 
-auxilliary switch handling, and include other contexts.  Most users are 
-given access to the default context.  Trusted users could be given access 
-to the local context for example.
-
-> permit = <ipaddr>/<netmask>
-> deny = <ipaddr>/<netmask>
-
-Permit and deny rules may be applied to users, allowing them to connect 
-from certain IP addresses and not others.  The permit and deny rules are 
-interpreted in sequence and all are evaluated on a given IP address, with 
-the final result being the decision.  For example:
-
-> permit = 0.0.0.0/0.0.0.0
-> deny = 192.168.0.0/255.255.255.0
-
-would deny anyone in 192.168.0.0 with a netmask of 24 bits (class C), 
-whereas:
-
-> deny = 192.168.0.0/255.255.255.0
-> permit = 0.0.0.0/0.0.0.0
-
-would not deny anyone since the final rule would permit anyone, thsu 
-overriding the denial.  
-
-If no permit/deny rules are listed, it is assumed that someone may connect 
-from anywhere.
-
-> callerid = <callerid>
-
-You may override the Caller*ID information passed by a user to you (if 
-they choose to send it) in order that it always be accurate from the 
-perspective of your server.
-
-> auth = [md5|plaintext|rsa]
-
-You may select which authentication methods are permitted to be used by 
-the user to authenticate to us.  Multiple methods may be specified, 
-separated by commas.  If md5 or plaintext authentication is selected, a 
-secret must be provided.  If RSA authentication is specified, then one or 
-more key names must be specifed with "inkeys"
-
-If no secret is specified and no authentication method is specified, then 
-no authentication will be required.
-
-> secret = <secret>
-
-The "secret" line specifies the shared secret for md5 and plaintext 
-authentication methods.  It is never suggested to use plaintext except in 
-some cases for debugging.
-
-> inkeys = key1[:key2...]
-
-The "inkeys" line specifies which keys we can use to authenticate the 
-remote peer.  If the peer's challenge passes with any of the given keys, 
-then we accept its authentication.  The key files live in 
-/var/lib/asterisk/keys/<name>.pub and are *public keys*.  Public keys are 
-not typically DES3 encrypted and thus do not usually need initialization.
-
----------------
-Peer configuration
-
-> allow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]
-> disallow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]
-
-The "allow" and "disallow" may be used to enable or disable specific codec 
-support on a per-peer basis.  
-
-> host = [<ipaddr>|dynamic]
-
-The host line specifies the hostname or IP address of the remote host, or 
-may be the word "dynamic" signifying that the host will register with us 
-(see register => in the general section above).
-
-> defaultip = <ipaddr>
-
-If the host uses dynamic registration, Asterisk may still be given a 
-default IP address to use when dynamic registration has not been performed 
-or has timed out.
-
-

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

README.variables

@@ -1,138 +0,0 @@
-GENERAL ENCHANCEMENTS TO EXTENSION LOGIC : 
-
-QUOTING: 
-
-exten => s,5,BackGround,blabla
-
-The parameter (blabla) can be quoted ("blabla"). In this case, a 
-comma does not terminate the field. 
-
-Also, characters special to variable substitution, expression evaluation, etc
-(see below), can be quoted. For example, to literally use a $ on the 
-string "$1231", quote it with a preceeding \. Special characters that must
-be quoted to be used, are [ ] $ " \. (to write \ itself, use \\). 
-
-VARIABLES: 
-
-Parameter strings can include variables. Variable names are arbitrary strings. 
-They are stored in the respective channel structure. 
-
-To set a variable to a particular value, do : 
-
-;exten => 1,2,SetVar,varname=value
-
-You can substitute the value of a variable everywhere using ${variablename}.
-For example, to stringwise append $lala to $blabla and store result in $koko, 
-do: 
-
-;exten => 1,2,SetVar,koko=${blabla}${lala}
-
-There are also the following special variables: 
-
-${CALLERID}	Caller ID
-${CALLERIDNAME}	Caller ID Name only
-${CALLERIDNUM}	Caller ID Number only
-${EXTEN}	Current extension
-${CONTEXT}      Current context
-${PRIORITY}	Current priority
-${CHANNEL}	Current channel name
-
-There are two reference modes - reference by value and reference by name. 
-To refer to a variable with its name (as an argument to a function that 
-requires a variable), just write the name. To refer to the variable's value, 
-enclose it inside ${}. For example, SetVar takes as the first argument 
-(before the =) a variable name, so: 
-
-;exten => 1,2,SetVar,koko=lala
-;exten => 1,3,SetVar,${koko}=blabla
-
-stores to the variable "koko" the value "lala" and to variable "lala" the 
-value "blabla". 
-
-In fact, everything contained ${here} is just replaced with the value of 
-the variable "here". 
-
-EXPRESSIONS: 
-
-Everything contained inside a bracket pair prefixed by a $ (like $[this]) is 
-considered as an expression and it is evaluated. Evaluation works similar to 
-(but is done on a later stage than) variable substitution: the expression 
-(including the square brackets) is replaced by the result of the expression 
-evaluation. The arguments and operands of the expression MUST BE separated 
-with spaces (take care NOT to leave ANY spaces between opening and closing 
-square brackets and the first and last arguments). 
-
-For example, after the sequence: 
-
-exten => 1,1,SetVar,"lala=$[1 + 2]";
-exten => 1,2,SetVar,"koko=$[2 * ${lala}]";
-
-the value of variable koko is "6".
-
-Operators are listed below in order of increasing precedence.  Operators
-with equal precedence are grouped within { } symbols.
-
-     expr1 | expr2
-             Return the evaluation of expr1 if it is neither an empty string
-             nor zero; otherwise, returns the evaluation of expr2.
-
-     expr1 & expr2
-             Return the evaluation of expr1 if neither expression evaluates to
-             an empty string or zero; otherwise, returns zero.
-
-     expr1 {=, >, >=, <, <=, !=} expr2
-             Return the results of integer comparison if both arguments are
-             integers; otherwise, returns the results of string comparison
-             using the locale-specific collation sequence.  The result of each
-             comparison is 1 if the specified relation is true, or 0 if the
-             relation is false.
-
-     expr1 {+, -} expr2
-             Return the results of addition or subtraction of integer-valued
-             arguments.
-
-     expr1 {*, /, %} expr2
-             Return the results of multiplication, integer division, or
-             remainder of integer-valued arguments.
-
-     expr1 : expr2
-             The `:' operator matches expr1 against expr2, which must be a
-             regular expression.  The regular expression is anchored to the
-             beginning of  the string with an implicit `^'.
-
-             If the match succeeds and the pattern contains at least one regu-
-             lar expression subexpression `\(...\)', the string correspond-
-             ing to `\1' is returned; otherwise the matching operator
-             returns the number of characters matched.  If the match fails and
-             the pattern contains a regular expression subexpression the null
-             string is returned; otherwise 0.
-
-Parentheses are used for grouping in the usual manner.
-
-The parser must be parsed with bison (bison is REQUIRED - yacc cannot 
-produce pure parsers, which are reentrant) 
-
-CONDITIONALS
-
-There is one conditional operator - the conditional goto : 
-
-;exten => 1,2,gotoif,condition?label1:label2
-
-If condition is true go to label1, else go to label2. Labels are interpreted
-exactly as in the normal goto command.
-
-"condition" is just a string. If the string is empty or "0", the condition
-is considered to be false, if it's anything else, the condition is true. 
-This is designed to be used together with the expression syntax described 
-above, eg : 
-
-exten => 1,2,gotoif,$[${CALLERID} = 123456]?2|1:3|1
-
-
-Example of use : 
-
-exten => s,2,SetVar,"vara=1"
-exten => s,3,SetVar,"varb=$[${vara} + 2]"
-exten => s,4,SetVar,"varc=$[${varb} * 2]"
-exten => s,5,GotoIf,"$[${varc} = 6]?99|1:s|6";
-

SECURITY

@@ -25,15 +25,15 @@ the "default" context within other private contexts by using:
 in the appropriate section.  A well designed PBX might look like this:
 
 [longdistance]
-exten => _91NXXNXXXXXX,1,Dial,Tor/g2/BYEXTENSION
+exten => _91NXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1})
 include => local
 
 [local]
-exten => _9NXXNXXX,1,Dial,Tor/g2/BYEXTENSION
+exten => _9NXXNXXX,1,Dial(Zap/g2/${EXTEN:1})
 include => default
 
 [default]
-exten => 6123,Dial,Tor/1
+exten => 6123,Dial(Zap/1)
 
 
 DON'T FORGET TO TAKE THE DEMO CONTEXT OUT OF YOUR DEFAULT CONTEXT.  There

acl.c

@@ -13,7 +13,6 @@
 
 #include <stdio.h>
 #include <stdlib.h>
-#include <pthread.h>
 #include <string.h>
 #include <sys/time.h>
 #include <signal.h>
@@ -21,12 +20,23 @@
 #include <unistd.h>
 #include <asterisk/acl.h>
 #include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/utils.h>
+#include <asterisk/lock.h>
 #include <arpa/inet.h>
 #include <sys/socket.h>
 #include <netdb.h>
+#include <net/if.h>
+#include <netinet/in_systm.h>
+#include <netinet/ip.h>
+#include <sys/ioctl.h>
+#if defined(__OpenBSD__) || defined(__NetBSD__) || defined(__FreeBSD__)
+#include <fcntl.h>
+#include <net/route.h>
+
+AST_MUTEX_DEFINE_STATIC(routeseq_lock);
+#endif
 
-#define AST_SENSE_DENY			0
-#define AST_SENSE_ALLOW			1
 
 struct ast_ha {
 	/* Host access rule */
@@ -36,6 +46,15 @@ struct ast_ha {
 	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;
@@ -46,33 +65,89 @@ void ast_free_ha(struct ast_ha *ha)
 	}
 }
 
+/* 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;
+	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) {
-		char *stringp=NULL;
-		stringp=stuff;
-		strsep(&stringp, "/");
-		nm = strsep(&stringp, "/");
+		strncpy(tmp, stuff, sizeof(tmp) - 1);
+		nm = strchr(tmp, '/');
 		if (!nm)
 			nm = "255.255.255.255";
-		if (!inet_aton(stuff, &ha->netaddr)) {
-			ast_log(LOG_WARNING, "%s not a valid IP\n", stuff);
-			free(ha);
-			return NULL;
+		else {
+			*nm = '\0';
+			nm++;
 		}
-		if (!inet_aton(nm, &ha->netmask)) {
+		if (!strchr(nm, '.')) {
+			if ((sscanf(nm, "%i", &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 not a valid netmask\n", nm);
 			free(ha);
-			return NULL;
+			return path;
+		}
+		if (!inet_aton(tmp, &ha->netaddr)) {
+			ast_log(LOG_WARNING, "%s not a valid IP\n", tmp);
+			free(ha);
+			return path;
 		}
 		ha->netaddr.s_addr &= ha->netmask.s_addr;
 		if (!strncasecmp(sense, "p", 1)) {
@@ -86,7 +161,8 @@ struct ast_ha *ast_append_ha(char *sense, char *stuff, struct ast_ha *path)
 		else
 			ret = ha;
 	}
-	return NULL;
+	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)
@@ -94,6 +170,13 @@ 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))
@@ -106,7 +189,8 @@ int ast_apply_ha(struct ast_ha *ha, struct sockaddr_in *sin)
 int ast_get_ip(struct sockaddr_in *sin, char *value)
 {
 	struct hostent *hp;
-	hp = gethostbyname(value);
+	struct ast_hostent ahp;
+	hp = ast_gethostbyname(value, &ahp);
 	if (hp) {
 		memcpy(&sin->sin_addr, hp->h_addr, sizeof(sin->sin_addr));
 	} else {
@@ -116,8 +200,210 @@ int ast_get_ip(struct sockaddr_in *sin, char *value)
 	return 0;
 }
 
-int inaddrcmp(struct sockaddr_in *sin1, struct sockaddr_in *sin2)
-{
-	return ((sin1->sin_addr.s_addr != sin2->sin_addr.s_addr )
-			|| (sin1->sin_port != sin2->sin_port));
+/* 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));
+	strncpy(ifreq.ifrn_name,iface,sizeof(ifreq.ifrn_name) - 1);
+
+	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)
+{
+#if defined(__OpenBSD__) || defined(__NetBSD__) || defined(__FreeBSD__)
+	struct sockaddr_in *sin;
+	struct sockaddr *sa;
+	struct {
+		struct	rt_msghdr m_rtm;
+		char	m_space[512];
+	} m_rtmsg;
+	char iabuf[INET_ADDRSTRLEN];
+	char *cp, *p;
+	int i, l, s, seq, flags;
+	pid_t pid = getpid();
+	static int routeseq;	/* Protected by "routeseq_lock" mutex */
+
+	p = ast_strdupa(ast_inet_ntoa(iabuf, sizeof(iabuf), *them));
+	memset(us, 0, sizeof(struct in_addr));
+
+	memset(&m_rtmsg, 0, sizeof(m_rtmsg));
+	m_rtmsg.m_rtm.rtm_type = RTM_GET;
+	m_rtmsg.m_rtm.rtm_version = RTM_VERSION;
+	ast_mutex_lock(&routeseq_lock);
+	seq = ++routeseq;
+	ast_mutex_unlock(&routeseq_lock);
+	m_rtmsg.m_rtm.rtm_seq = seq;
+	m_rtmsg.m_rtm.rtm_addrs = RTA_DST | RTA_IFA;
+	m_rtmsg.m_rtm.rtm_msglen = sizeof(struct rt_msghdr) + sizeof(struct sockaddr_in);
+	sin = (struct sockaddr_in *)m_rtmsg.m_space;
+	sin->sin_family = AF_INET;
+	sin->sin_len = sizeof(struct sockaddr_in);
+	sin->sin_addr = *them;
+
+	if ((s = socket(PF_ROUTE, SOCK_RAW, AF_UNSPEC)) < 0) {
+		ast_log(LOG_ERROR, "Error opening routing socket\n");
+		return -1;
+	}
+	flags = fcntl(s, F_GETFL);
+	fcntl(s, F_SETFL, flags | O_NONBLOCK);
+	if (write(s, (char *)&m_rtmsg, m_rtmsg.m_rtm.rtm_msglen) < 0) {
+		ast_log(LOG_ERROR, "Error writing to routing socket: %s\n", strerror(errno));
+		close(s);
+		return -1;
+	}
+	do {
+		l = read(s, (char *)&m_rtmsg, sizeof(m_rtmsg));
+	} while (l > 0 && (m_rtmsg.m_rtm.rtm_seq != seq || m_rtmsg.m_rtm.rtm_pid != pid));
+	if (l < 0) {
+		if (errno != EAGAIN)
+			ast_log(LOG_ERROR, "Error reading from routing socket\n");
+		close(s);
+		return -1;
+	}
+	close(s);
+
+	if (m_rtmsg.m_rtm.rtm_version != RTM_VERSION) {
+		ast_log(LOG_ERROR, "Unsupported route socket protocol version\n");
+		return -1;
+	}
+
+	if (m_rtmsg.m_rtm.rtm_msglen != l)
+		ast_log(LOG_WARNING, "Message length mismatch, in packet %d, returned %d\n",
+				m_rtmsg.m_rtm.rtm_msglen, l);
+
+	if (m_rtmsg.m_rtm.rtm_errno) {
+		ast_log(LOG_ERROR, "RTM_GET got %s (%d)\n",
+				strerror(m_rtmsg.m_rtm.rtm_errno), m_rtmsg.m_rtm.rtm_errno);
+		return -1;
+	}
+
+	cp = (char *)m_rtmsg.m_space;
+	if (m_rtmsg.m_rtm.rtm_addrs)
+		for (i = 1; i; i <<= 1)
+			if (m_rtmsg.m_rtm.rtm_addrs & i) {
+				sa = (struct sockaddr *)cp;
+				if (i == RTA_IFA && sa->sa_family == AF_INET) {
+					sin = (struct sockaddr_in *)sa;
+					*us = sin->sin_addr;
+					ast_log(LOG_DEBUG, "Found route to %s, output from our address %s.\n", p, ast_inet_ntoa(iabuf, sizeof(iabuf), *us));
+					return 0;
+				}
+				cp += sa->sa_len > 0 ?
+					  (1 + ((sa->sa_len - 1) | (sizeof(long) - 1))) :
+					  sizeof(long);
+			}
+
+	ast_log(LOG_DEBUG, "No route found for address %s!\n", p);
+	return -1;
+#else
+	FILE *PROC;
+	unsigned int remote_ip;
+	int res = 1;
+	char line[256];
+	remote_ip = them->s_addr;
+	
+	PROC = fopen("/proc/net/route","r");
+	if (!PROC) {
+		bzero(us,sizeof(struct in_addr));
+		return -1;
+	}
+	/* First line contains headers */
+	fgets(line,sizeof(line),PROC);
+
+	while (!feof(PROC)) {
+		char iface[256];
+		unsigned int dest, gateway, mask;
+		int i,fieldnum;
+		char *fields[40];
+
+		fgets(line,sizeof(line),PROC);
+
+		fieldnum = 0;
+		for (i=0;i<sizeof(line);i++) {
+			char *offset;
+
+			fields[fieldnum++] = line + i;
+			offset = strchr(line + i,'\t');
+			if (offset == NULL) {
+				/* Exit loop */
+				break;
+			} else if (fieldnum >= 9) {
+				/* Short-circuit: can't break at 8, since the end of field 7 is figured when fieldnum=8 */
+				break;
+			} else {
+				*offset = '\0';
+				i = offset - line;
+			}
+		}
+		if (fieldnum >= 8) {
+
+			sscanf(fields[0],"%s",iface);
+			sscanf(fields[1],"%x",&dest);
+			sscanf(fields[2],"%x",&gateway);
+			sscanf(fields[7],"%x",&mask);
+#if 0
+			{ char iabuf[INET_ADDRSTRLEN]; 
+			printf("Addr: %s %08x Dest: %08x Mask: %08x\n", ast_inet_ntoa(iabuf, sizeof(iabuf), *them), remote_ip, dest, mask); }
+#endif		
+			/* Looks simple, but here is the magic */
+			if (((remote_ip & mask) ^ dest) == 0) {
+				res = ast_lookup_iface(iface,us);
+				break;
+			}
+		}
+	}
+	fclose(PROC);
+	if (res == 1) {
+		ast_log(LOG_WARNING, "Yikes!  No default route?!!\n");
+		bzero(us,sizeof(struct in_addr));
+		return -2;
+	} else if (res) {
+		/* We've already warned in subroutine */
+		return -1;
+ 	}
+	return 0;
+#endif
+}
+
+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;
+}
+

addmailbox

@@ -1,20 +0,0 @@
-#!/bin/sh
-VMHOME=/var/spool/asterisk/vm
-SNDHOME=/var/lib/asterisk/sounds
-echo -n "Enter mailbox number: "
-read mailbox
-mkdir -p ${VMHOME}/${mailbox}
-mkdir -p ${VMHOME}/${mailbox}/INBOX
-cat ${SNDHOME}/vm-theperson.gsm > ${VMHOME}/${mailbox}/unavail.gsm
-cat ${SNDHOME}/vm-theperson.gsm > ${VMHOME}/${mailbox}/busy.gsm
-cat ${SNDHOME}/vm-extension.gsm > ${VMHOME}/${mailbox}/greet.gsm
-nums=`echo $mailbox | sed 's/./ \0/g'`
-for x in $nums; do
-	cat ${SNDHOME}/digits/${x}.gsm >> ${VMHOME}/${mailbox}/unavail.gsm
-	cat ${SNDHOME}/digits/${x}.gsm >> ${VMHOME}/${mailbox}/busy.gsm
-	cat ${SNDHOME}/digits/${x}.gsm >> ${VMHOME}/${mailbox}/greet.gsm
-done
-
-cat ${SNDHOME}/vm-isunavail.gsm >> ${VMHOME}/${mailbox}/unavail.gsm
-cat ${SNDHOME}/vm-isonphone.gsm >> ${VMHOME}/${mailbox}/busy.gsm
-

aescrypt.c

@@ -0,0 +1,311 @@
+/*
+ ---------------------------------------------------------------------------
+ Copyright (c) 2003, Dr Brian Gladman <brg@gladman.me.uk>, Worcester, UK.
+ All rights reserved.
+
+ LICENSE TERMS
+
+ The free distribution and use of this software in both source and binary
+ form is allowed (with or without changes) provided that:
+
+   1. distributions of this source code include the above copyright
+      notice, this list of conditions and the following disclaimer;
+
+   2. distributions in binary form include the above copyright
+      notice, this list of conditions and the following disclaimer
+      in the documentation and/or other associated materials;
+
+   3. the copyright holder's name is not used to endorse products
+      built using this software without specific written permission.
+
+ ALTERNATIVELY, provided that this notice is retained in full, this product
+ may be distributed under the terms of the GNU General Public License (GPL),
+ in which case the provisions of the GPL apply INSTEAD OF those given above.
+
+ DISCLAIMER
+
+ This software is provided 'as is' with no explicit or implied warranties
+ in respect of its properties, including, but not limited to, correctness
+ and/or fitness for purpose.
+ ---------------------------------------------------------------------------
+ Issue Date: 26/08/2003
+
+ 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
+
+#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))
+
+#if defined(ARRAYS)
+#define locals(y,x)     x[4],y[4]
+#else
+#define locals(y,x)     x##0,x##1,x##2,x##3,y##0,y##1,y##2,y##3
+#endif
+
+#define l_copy(y, x)    s(y,0) = s(x,0); s(y,1) = s(x,1); \
+                        s(y,2) = s(x,2); s(y,3) = s(x,3);
+#define state_in(y,x,k) si(y,x,k,0); si(y,x,k,1); si(y,x,k,2); si(y,x,k,3)
+#define state_out(y,x)  so(y,x,0); so(y,x,1); so(y,x,2); so(y,x,3)
+#define round(rm,y,x,k) rm(y,x,k,0); rm(y,x,k,1); rm(y,x,k,2); rm(y,x,k,3)
+
+#if defined(ENCRYPTION) && !defined(AES_ASM)
+
+/* Visual C++ .Net v7.1 provides the fastest encryption code when using
+   Pentium optimiation with small code but this is poor for decryption
+   so we need to control this with the following VC++ pragmas
+*/
+
+#if defined(_MSC_VER)
+#pragma optimize( "s", on )
+#endif
+
+/* Given the column (c) of the output state variable, the following
+   macros give the input state variables which are needed in its
+   computation for each row (r) of the state. All the alternative
+   macros give the same end values but expand into different ways
+   of calculating these values.  In particular the complex macro
+   used for dynamically variable block sizes is designed to expand
+   to a compile time constant whenever possible but will expand to
+   conditional clauses on some branches (I am grateful to Frank
+   Yellin for this construction)
+*/
+
+#define fwd_var(x,r,c)\
+ ( r == 0 ? ( c == 0 ? s(x,0) : c == 1 ? s(x,1) : c == 2 ? s(x,2) : s(x,3))\
+ : r == 1 ? ( c == 0 ? s(x,1) : c == 1 ? s(x,2) : c == 2 ? s(x,3) : s(x,0))\
+ : r == 2 ? ( c == 0 ? s(x,2) : c == 1 ? s(x,3) : c == 2 ? s(x,0) : s(x,1))\
+ :          ( c == 0 ? s(x,3) : c == 1 ? s(x,0) : c == 2 ? s(x,1) : s(x,2)))
+
+#if defined(FT4_SET)
+#undef  dec_fmvars
+#define fwd_rnd(y,x,k,c)    (s(y,c) = (k)[c] ^ four_tables(x,t_use(f,n),fwd_var,rf1,c))
+#elif defined(FT1_SET)
+#undef  dec_fmvars
+#define fwd_rnd(y,x,k,c)    (s(y,c) = (k)[c] ^ one_table(x,upr,t_use(f,n),fwd_var,rf1,c))
+#else
+#define fwd_rnd(y,x,k,c)    (s(y,c) = (k)[c] ^ fwd_mcol(no_table(x,t_use(s,box),fwd_var,rf1,c)))
+#endif
+
+#if defined(FL4_SET)
+#define fwd_lrnd(y,x,k,c)   (s(y,c) = (k)[c] ^ four_tables(x,t_use(f,l),fwd_var,rf1,c))
+#elif defined(FL1_SET)
+#define fwd_lrnd(y,x,k,c)   (s(y,c) = (k)[c] ^ one_table(x,ups,t_use(f,l),fwd_var,rf1,c))
+#else
+#define fwd_lrnd(y,x,k,c)   (s(y,c) = (k)[c] ^ no_table(x,t_use(s,box),fwd_var,rf1,c))
+#endif
+
+aes_rval aes_encrypt(const void *in_blk, void *out_blk, const aes_encrypt_ctx cx[1])
+{   aes_32t         locals(b0, b1);
+    const aes_32t   *kp = cx->ks;
+#ifdef dec_fmvars
+    dec_fmvars; /* declare variables for fwd_mcol() if needed */
+#endif
+
+    aes_32t nr = (kp[45] ^ kp[52] ^ kp[53] ? kp[52] : 14);
+
+#ifdef AES_ERR_CHK
+    if(   (nr != 10 || !(kp[0] | kp[3] | kp[4])) 
+       && (nr != 12 || !(kp[0] | kp[5] | kp[6]))
+       && (nr != 14 || !(kp[0] | kp[7] | kp[8])) )
+        return aes_error;
+#endif
+
+    state_in(b0, in_blk, kp);
+
+#if (ENC_UNROLL == FULL)
+
+    switch(nr)
+    {
+    case 14:
+        round(fwd_rnd,  b1, b0, kp + 1 * N_COLS);
+        round(fwd_rnd,  b0, b1, kp + 2 * N_COLS);
+        kp += 2 * N_COLS;
+    case 12:
+        round(fwd_rnd,  b1, b0, kp + 1 * N_COLS);
+        round(fwd_rnd,  b0, b1, kp + 2 * N_COLS);
+        kp += 2 * N_COLS;
+    case 10:
+        round(fwd_rnd,  b1, b0, kp + 1 * N_COLS);
+        round(fwd_rnd,  b0, b1, kp + 2 * N_COLS);
+        round(fwd_rnd,  b1, b0, kp + 3 * N_COLS);
+        round(fwd_rnd,  b0, b1, kp + 4 * N_COLS);
+        round(fwd_rnd,  b1, b0, kp + 5 * N_COLS);
+        round(fwd_rnd,  b0, b1, kp + 6 * N_COLS);
+        round(fwd_rnd,  b1, b0, kp + 7 * N_COLS);
+        round(fwd_rnd,  b0, b1, kp + 8 * N_COLS);
+        round(fwd_rnd,  b1, b0, kp + 9 * N_COLS);
+        round(fwd_lrnd, b0, b1, kp +10 * N_COLS);
+    }
+
+#else
+
+#if (ENC_UNROLL == PARTIAL)
+    {   aes_32t    rnd;
+        for(rnd = 0; rnd < (nr >> 1) - 1; ++rnd)
+        {
+            kp += N_COLS;
+            round(fwd_rnd, b1, b0, kp);
+            kp += N_COLS;
+            round(fwd_rnd, b0, b1, kp);
+        }
+        kp += N_COLS;
+        round(fwd_rnd,  b1, b0, kp);
+#else
+    {   aes_32t    rnd;
+        for(rnd = 0; rnd < nr - 1; ++rnd)
+        {
+            kp += N_COLS;
+            round(fwd_rnd, b1, b0, kp);
+            l_copy(b0, b1);
+        }
+#endif
+        kp += N_COLS;
+        round(fwd_lrnd, b0, b1, kp);
+    }
+#endif
+
+    state_out(out_blk, b0);
+#ifdef AES_ERR_CHK
+    return aes_good;
+#endif
+}
+
+#endif
+
+#if defined(DECRYPTION) && !defined(AES_ASM)
+
+/* Visual C++ .Net v7.1 provides the fastest encryption code when using
+   Pentium optimiation with small code but this is poor for decryption
+   so we need to control this with the following VC++ pragmas
+*/
+
+#if defined(_MSC_VER)
+#pragma optimize( "t", on )
+#endif
+
+/* Given the column (c) of the output state variable, the following
+   macros give the input state variables which are needed in its
+   computation for each row (r) of the state. All the alternative
+   macros give the same end values but expand into different ways
+   of calculating these values.  In particular the complex macro
+   used for dynamically variable block sizes is designed to expand
+   to a compile time constant whenever possible but will expand to
+   conditional clauses on some branches (I am grateful to Frank
+   Yellin for this construction)
+*/
+
+#define inv_var(x,r,c)\
+ ( r == 0 ? ( c == 0 ? s(x,0) : c == 1 ? s(x,1) : c == 2 ? s(x,2) : s(x,3))\
+ : r == 1 ? ( c == 0 ? s(x,3) : c == 1 ? s(x,0) : c == 2 ? s(x,1) : s(x,2))\
+ : r == 2 ? ( c == 0 ? s(x,2) : c == 1 ? s(x,3) : c == 2 ? s(x,0) : s(x,1))\
+ :          ( c == 0 ? s(x,1) : c == 1 ? s(x,2) : c == 2 ? s(x,3) : s(x,0)))
+
+#if defined(IT4_SET)
+#undef  dec_imvars
+#define inv_rnd(y,x,k,c)    (s(y,c) = (k)[c] ^ four_tables(x,t_use(i,n),inv_var,rf1,c))
+#elif defined(IT1_SET)
+#undef  dec_imvars
+#define inv_rnd(y,x,k,c)    (s(y,c) = (k)[c] ^ one_table(x,upr,t_use(i,n),inv_var,rf1,c))
+#else
+#define inv_rnd(y,x,k,c)    (s(y,c) = inv_mcol((k)[c] ^ no_table(x,t_use(i,box),inv_var,rf1,c)))
+#endif
+
+#if defined(IL4_SET)
+#define inv_lrnd(y,x,k,c)   (s(y,c) = (k)[c] ^ four_tables(x,t_use(i,l),inv_var,rf1,c))
+#elif defined(IL1_SET)
+#define inv_lrnd(y,x,k,c)   (s(y,c) = (k)[c] ^ one_table(x,ups,t_use(i,l),inv_var,rf1,c))
+#else
+#define inv_lrnd(y,x,k,c)   (s(y,c) = (k)[c] ^ no_table(x,t_use(i,box),inv_var,rf1,c))
+#endif
+
+aes_rval aes_decrypt(const void *in_blk, void *out_blk, const aes_decrypt_ctx cx[1])
+{   aes_32t        locals(b0, b1);
+#ifdef dec_imvars
+    dec_imvars; /* declare variables for inv_mcol() if needed */
+#endif
+
+    aes_32t nr = (cx->ks[45] ^ cx->ks[52] ^ cx->ks[53] ? cx->ks[52] : 14);
+    const aes_32t *kp = cx->ks + nr * N_COLS;
+
+#ifdef AES_ERR_CHK
+    if(   (nr != 10 || !(cx->ks[0] | cx->ks[3] | cx->ks[4])) 
+       && (nr != 12 || !(cx->ks[0] | cx->ks[5] | cx->ks[6]))
+       && (nr != 14 || !(cx->ks[0] | cx->ks[7] | cx->ks[8])) )
+        return aes_error;
+#endif
+
+    state_in(b0, in_blk, kp);
+
+#if (DEC_UNROLL == FULL)
+
+    switch(nr)
+    {
+    case 14:
+        round(inv_rnd,  b1, b0, kp -  1 * N_COLS);
+        round(inv_rnd,  b0, b1, kp -  2 * N_COLS);
+        kp -= 2 * N_COLS;
+    case 12:
+        round(inv_rnd,  b1, b0, kp -  1 * N_COLS);
+        round(inv_rnd,  b0, b1, kp -  2 * N_COLS);
+        kp -= 2 * N_COLS;
+    case 10:
+        round(inv_rnd,  b1, b0, kp -  1 * N_COLS);
+        round(inv_rnd,  b0, b1, kp -  2 * N_COLS);
+        round(inv_rnd,  b1, b0, kp -  3 * N_COLS);
+        round(inv_rnd,  b0, b1, kp -  4 * N_COLS);
+        round(inv_rnd,  b1, b0, kp -  5 * N_COLS);
+        round(inv_rnd,  b0, b1, kp -  6 * N_COLS);
+        round(inv_rnd,  b1, b0, kp -  7 * N_COLS);
+        round(inv_rnd,  b0, b1, kp -  8 * N_COLS);
+        round(inv_rnd,  b1, b0, kp -  9 * N_COLS);
+        round(inv_lrnd, b0, b1, kp - 10 * N_COLS);
+    }
+
+#else
+
+#if (DEC_UNROLL == PARTIAL)
+    {   aes_32t    rnd;
+        for(rnd = 0; rnd < (nr >> 1) - 1; ++rnd)
+        {
+            kp -= N_COLS;
+            round(inv_rnd, b1, b0, kp);
+            kp -= N_COLS;
+            round(inv_rnd, b0, b1, kp);
+        }
+        kp -= N_COLS;
+        round(inv_rnd, b1, b0, kp);
+#else
+    {   aes_32t    rnd;
+        for(rnd = 0; rnd < nr - 1; ++rnd)
+        {
+            kp -= N_COLS;
+            round(inv_rnd, b1, b0, kp);
+            l_copy(b0, b1);
+        }
+#endif
+        kp -= N_COLS;
+        round(inv_lrnd, b0, b1, kp);
+    }
+#endif
+
+    state_out(out_blk, b0);
+#ifdef AES_ERR_CHK
+    return aes_good;
+#endif
+}
+
+#endif
+
+#if defined(__cplusplus)
+}
+#endif

aeskey.c

@@ -0,0 +1,463 @@
+/*
+ ---------------------------------------------------------------------------
+ Copyright (c) 2003, Dr Brian Gladman <brg@gladman.me.uk>, Worcester, UK.
+ All rights reserved.
+
+ LICENSE TERMS
+
+ The free distribution and use of this software in both source and binary
+ form is allowed (with or without changes) provided that:
+
+   1. distributions of this source code include the above copyright
+      notice, this list of conditions and the following disclaimer;
+
+   2. distributions in binary form include the above copyright
+      notice, this list of conditions and the following disclaimer
+      in the documentation and/or other associated materials;
+
+   3. the copyright holder's name is not used to endorse products
+      built using this software without specific written permission.
+
+ ALTERNATIVELY, provided that this notice is retained in full, this product
+ may be distributed under the terms of the GNU General Public License (GPL),
+ in which case the provisions of the GPL apply INSTEAD OF those given above.
+
+ DISCLAIMER
+
+ This software is provided 'as is' with no explicit or implied warranties
+ in respect of its properties, including, but not limited to, correctness
+ and/or fitness for purpose.
+ ---------------------------------------------------------------------------
+ Issue Date: 26/08/2003
+
+ 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
+
+/* 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
+   values correspond with Nk values of 4, 6 and 8 respectively.
+
+   The following macros implement a single cycle in the key
+   schedule generation process. The number of cycles needed
+   for each cx->n_col and nk value is:
+
+    nk =             4  5  6  7  8
+    ------------------------------
+    cx->n_col = 4   10  9  8  7  7
+    cx->n_col = 5   14 11 10  9  9
+    cx->n_col = 6   19 15 12 11 11
+    cx->n_col = 7   21 19 16 13 14
+    cx->n_col = 8   29 23 19 17 14
+*/
+
+#define ke4(k,i) \
+{   k[4*(i)+4] = ss[0] ^= ls_box(ss[3],3) ^ t_use(r,c)[i]; k[4*(i)+5] = ss[1] ^= ss[0]; \
+    k[4*(i)+6] = ss[2] ^= ss[1]; k[4*(i)+7] = ss[3] ^= ss[2]; \
+}
+#define kel4(k,i) \
+{   k[4*(i)+4] = ss[0] ^= ls_box(ss[3],3) ^ t_use(r,c)[i]; k[4*(i)+5] = ss[1] ^= ss[0]; \
+    k[4*(i)+6] = ss[2] ^= ss[1]; k[4*(i)+7] = ss[3] ^= ss[2]; \
+}
+
+#define ke6(k,i) \
+{   k[6*(i)+ 6] = ss[0] ^= ls_box(ss[5],3) ^ t_use(r,c)[i]; k[6*(i)+ 7] = ss[1] ^= ss[0]; \
+    k[6*(i)+ 8] = ss[2] ^= ss[1]; k[6*(i)+ 9] = ss[3] ^= ss[2]; \
+    k[6*(i)+10] = ss[4] ^= ss[3]; k[6*(i)+11] = ss[5] ^= ss[4]; \
+}
+#define kel6(k,i) \
+{   k[6*(i)+ 6] = ss[0] ^= ls_box(ss[5],3) ^ t_use(r,c)[i]; k[6*(i)+ 7] = ss[1] ^= ss[0]; \
+    k[6*(i)+ 8] = ss[2] ^= ss[1]; k[6*(i)+ 9] = ss[3] ^= ss[2]; \
+}
+
+#define ke8(k,i) \
+{   k[8*(i)+ 8] = ss[0] ^= ls_box(ss[7],3) ^ t_use(r,c)[i]; k[8*(i)+ 9] = ss[1] ^= ss[0]; \
+    k[8*(i)+10] = ss[2] ^= ss[1]; k[8*(i)+11] = ss[3] ^= ss[2]; \
+    k[8*(i)+12] = ss[4] ^= ls_box(ss[3],0); k[8*(i)+13] = ss[5] ^= ss[4]; \
+    k[8*(i)+14] = ss[6] ^= ss[5]; k[8*(i)+15] = ss[7] ^= ss[6]; \
+}
+#define kel8(k,i) \
+{   k[8*(i)+ 8] = ss[0] ^= ls_box(ss[7],3) ^ t_use(r,c)[i]; k[8*(i)+ 9] = ss[1] ^= ss[0]; \
+    k[8*(i)+10] = ss[2] ^= ss[1]; k[8*(i)+11] = ss[3] ^= ss[2]; \
+}
+
+#if defined(ENCRYPTION_KEY_SCHEDULE)
+
+#if defined(AES_128) || defined(AES_VAR)
+
+aes_rval aes_encrypt_key128(const void *in_key, aes_encrypt_ctx cx[1])
+{   aes_32t    ss[4];
+
+    cx->ks[0] = ss[0] = word_in(in_key, 0);
+    cx->ks[1] = ss[1] = word_in(in_key, 1);
+    cx->ks[2] = ss[2] = word_in(in_key, 2);
+    cx->ks[3] = ss[3] = word_in(in_key, 3);
+
+#if ENC_UNROLL == NONE
+    {   aes_32t i;
+
+        for(i = 0; i < ((11 * N_COLS - 1) / 4); ++i)
+            ke4(cx->ks, i);
+    }
+#else
+    ke4(cx->ks, 0);  ke4(cx->ks, 1);
+    ke4(cx->ks, 2);  ke4(cx->ks, 3);
+    ke4(cx->ks, 4);  ke4(cx->ks, 5);
+    ke4(cx->ks, 6);  ke4(cx->ks, 7);
+    ke4(cx->ks, 8); kel4(cx->ks, 9);
+#endif
+
+    /* cx->ks[45] ^ cx->ks[52] ^ cx->ks[53] is zero for a 256 bit       */
+    /* key and must be non-zero for 128 and 192 bits keys   */
+    cx->ks[53] = cx->ks[45] = 0;
+    cx->ks[52] = 10;
+#ifdef AES_ERR_CHK
+    return aes_good;
+#endif
+}
+
+#endif
+
+#if defined(AES_192) || defined(AES_VAR)
+
+aes_rval aes_encrypt_key192(const void *in_key, aes_encrypt_ctx cx[1])
+{   aes_32t    ss[6];
+
+    cx->ks[0] = ss[0] = word_in(in_key, 0);
+    cx->ks[1] = ss[1] = word_in(in_key, 1);
+    cx->ks[2] = ss[2] = word_in(in_key, 2);
+    cx->ks[3] = ss[3] = word_in(in_key, 3);
+    cx->ks[4] = ss[4] = word_in(in_key, 4);
+    cx->ks[5] = ss[5] = word_in(in_key, 5);
+
+#if ENC_UNROLL == NONE
+    {   aes_32t i;
+
+        for(i = 0; i < (13 * N_COLS - 1) / 6; ++i)
+            ke6(cx->ks, i);
+    }
+#else
+    ke6(cx->ks, 0);  ke6(cx->ks, 1);
+    ke6(cx->ks, 2);  ke6(cx->ks, 3);
+    ke6(cx->ks, 4);  ke6(cx->ks, 5);
+    ke6(cx->ks, 6); kel6(cx->ks, 7);
+#endif
+
+    /* cx->ks[45] ^ cx->ks[52] ^ cx->ks[53] is zero for a 256 bit       */
+    /* key and must be non-zero for 128 and 192 bits keys   */
+    cx->ks[53] = cx->ks[45];
+    cx->ks[52] = 12;
+#ifdef AES_ERR_CHK
+    return aes_good;
+#endif
+}
+
+#endif
+
+#if defined(AES_256) || defined(AES_VAR)
+
+aes_rval aes_encrypt_key256(const void *in_key, aes_encrypt_ctx cx[1])
+{   aes_32t    ss[8];
+
+    cx->ks[0] = ss[0] = word_in(in_key, 0);
+    cx->ks[1] = ss[1] = word_in(in_key, 1);
+    cx->ks[2] = ss[2] = word_in(in_key, 2);
+    cx->ks[3] = ss[3] = word_in(in_key, 3);
+    cx->ks[4] = ss[4] = word_in(in_key, 4);
+    cx->ks[5] = ss[5] = word_in(in_key, 5);
+    cx->ks[6] = ss[6] = word_in(in_key, 6);
+    cx->ks[7] = ss[7] = word_in(in_key, 7);
+
+#if ENC_UNROLL == NONE
+    {   aes_32t i;
+
+        for(i = 0; i < (15 * N_COLS - 1) / 8; ++i)
+            ke8(cx->ks,  i);
+    }
+#else
+    ke8(cx->ks, 0); ke8(cx->ks, 1);
+    ke8(cx->ks, 2); ke8(cx->ks, 3);
+    ke8(cx->ks, 4); ke8(cx->ks, 5);
+    kel8(cx->ks, 6);
+#endif
+#ifdef AES_ERR_CHK
+    return aes_good;
+#endif
+}
+
+#endif
+
+#if defined(AES_VAR)
+
+aes_rval aes_encrypt_key(const void *in_key, int key_len, aes_encrypt_ctx cx[1])
+{
+    switch(key_len)
+    {
+#ifdef AES_ERR_CHK
+    case 16: case 128: return aes_encrypt_key128(in_key, cx);
+    case 24: case 192: return aes_encrypt_key192(in_key, cx);
+    case 32: case 256: return aes_encrypt_key256(in_key, cx);
+    default: return aes_error;
+#else
+    case 16: case 128: aes_encrypt_key128(in_key, cx); return;
+    case 24: case 192: aes_encrypt_key192(in_key, cx); return;
+    case 32: case 256: aes_encrypt_key256(in_key, cx); return;
+#endif
+    }
+}
+
+#endif
+
+#endif
+
+#if defined(DECRYPTION_KEY_SCHEDULE)
+
+#if DEC_ROUND == NO_TABLES
+#define ff(x)   (x)
+#else
+#define ff(x)   inv_mcol(x)
+#ifdef  dec_imvars
+#define d_vars  dec_imvars
+#endif
+#endif
+
+#if 1
+#define kdf4(k,i) \
+{   ss[0] = ss[0] ^ ss[2] ^ ss[1] ^ ss[3]; ss[1] = ss[1] ^ ss[3]; ss[2] = ss[2] ^ ss[3]; ss[3] = ss[3]; \
+    ss[4] = ls_box(ss[(i+3) % 4], 3) ^ t_use(r,c)[i]; ss[i % 4] ^= ss[4]; \
+    ss[4] ^= k[4*(i)];   k[4*(i)+4] = ff(ss[4]); ss[4] ^= k[4*(i)+1]; k[4*(i)+5] = ff(ss[4]); \
+    ss[4] ^= k[4*(i)+2]; k[4*(i)+6] = ff(ss[4]); ss[4] ^= k[4*(i)+3]; k[4*(i)+7] = ff(ss[4]); \
+}
+#define kd4(k,i) \
+{   ss[4] = ls_box(ss[(i+3) % 4], 3) ^ t_use(r,c)[i]; ss[i % 4] ^= ss[4]; ss[4] = ff(ss[4]); \
+    k[4*(i)+4] = ss[4] ^= k[4*(i)]; k[4*(i)+5] = ss[4] ^= k[4*(i)+1]; \
+    k[4*(i)+6] = ss[4] ^= k[4*(i)+2]; k[4*(i)+7] = ss[4] ^= k[4*(i)+3]; \
+}
+#define kdl4(k,i) \
+{   ss[4] = ls_box(ss[(i+3) % 4], 3) ^ t_use(r,c)[i]; ss[i % 4] ^= ss[4]; \
+    k[4*(i)+4] = (ss[0] ^= ss[1]) ^ ss[2] ^ ss[3]; k[4*(i)+5] = ss[1] ^ ss[3]; \
+    k[4*(i)+6] = ss[0]; k[4*(i)+7] = ss[1]; \
+}
+#else
+#define kdf4(k,i) \
+{   ss[0] ^= ls_box(ss[3],3) ^ t_use(r,c)[i]; k[4*(i)+ 4] = ff(ss[0]); ss[1] ^= ss[0]; k[4*(i)+ 5] = ff(ss[1]); \
+    ss[2] ^= ss[1]; k[4*(i)+ 6] = ff(ss[2]); ss[3] ^= ss[2]; k[4*(i)+ 7] = ff(ss[3]); \
+}
+#define kd4(k,i) \
+{   ss[4] = ls_box(ss[3],3) ^ t_use(r,c)[i]; \
+    ss[0] ^= ss[4]; ss[4] = ff(ss[4]); k[4*(i)+ 4] = ss[4] ^= k[4*(i)]; \
+    ss[1] ^= ss[0]; k[4*(i)+ 5] = ss[4] ^= k[4*(i)+ 1]; \
+    ss[2] ^= ss[1]; k[4*(i)+ 6] = ss[4] ^= k[4*(i)+ 2]; \
+    ss[3] ^= ss[2]; k[4*(i)+ 7] = ss[4] ^= k[4*(i)+ 3]; \
+}
+#define kdl4(k,i) \
+{   ss[0] ^= ls_box(ss[3],3) ^ t_use(r,c)[i]; k[4*(i)+ 4] = ss[0]; ss[1] ^= ss[0]; k[4*(i)+ 5] = ss[1]; \
+    ss[2] ^= ss[1]; k[4*(i)+ 6] = ss[2]; ss[3] ^= ss[2]; k[4*(i)+ 7] = ss[3]; \
+}
+#endif
+
+#define kdf6(k,i) \
+{   ss[0] ^= ls_box(ss[5],3) ^ t_use(r,c)[i]; k[6*(i)+ 6] = ff(ss[0]); ss[1] ^= ss[0]; k[6*(i)+ 7] = ff(ss[1]); \
+    ss[2] ^= ss[1]; k[6*(i)+ 8] = ff(ss[2]); ss[3] ^= ss[2]; k[6*(i)+ 9] = ff(ss[3]); \
+    ss[4] ^= ss[3]; k[6*(i)+10] = ff(ss[4]); ss[5] ^= ss[4]; k[6*(i)+11] = ff(ss[5]); \
+}
+#define kd6(k,i) \
+{   ss[6] = ls_box(ss[5],3) ^ t_use(r,c)[i]; \
+    ss[0] ^= ss[6]; ss[6] = ff(ss[6]); k[6*(i)+ 6] = ss[6] ^= k[6*(i)]; \
+    ss[1] ^= ss[0]; k[6*(i)+ 7] = ss[6] ^= k[6*(i)+ 1]; \
+    ss[2] ^= ss[1]; k[6*(i)+ 8] = ss[6] ^= k[6*(i)+ 2]; \
+    ss[3] ^= ss[2]; k[6*(i)+ 9] = ss[6] ^= k[6*(i)+ 3]; \
+    ss[4] ^= ss[3]; k[6*(i)+10] = ss[6] ^= k[6*(i)+ 4]; \
+    ss[5] ^= ss[4]; k[6*(i)+11] = ss[6] ^= k[6*(i)+ 5]; \
+}
+#define kdl6(k,i) \
+{   ss[0] ^= ls_box(ss[5],3) ^ t_use(r,c)[i]; k[6*(i)+ 6] = ss[0]; ss[1] ^= ss[0]; k[6*(i)+ 7] = ss[1]; \
+    ss[2] ^= ss[1]; k[6*(i)+ 8] = ss[2]; ss[3] ^= ss[2]; k[6*(i)+ 9] = ss[3]; \
+}
+
+#define kdf8(k,i) \
+{   ss[0] ^= ls_box(ss[7],3) ^ t_use(r,c)[i]; k[8*(i)+ 8] = ff(ss[0]); ss[1] ^= ss[0]; k[8*(i)+ 9] = ff(ss[1]); \
+    ss[2] ^= ss[1]; k[8*(i)+10] = ff(ss[2]); ss[3] ^= ss[2]; k[8*(i)+11] = ff(ss[3]); \
+    ss[4] ^= ls_box(ss[3],0); k[8*(i)+12] = ff(ss[4]); ss[5] ^= ss[4]; k[8*(i)+13] = ff(ss[5]); \
+    ss[6] ^= ss[5]; k[8*(i)+14] = ff(ss[6]); ss[7] ^= ss[6]; k[8*(i)+15] = ff(ss[7]); \
+}
+#define kd8(k,i) \
+{   aes_32t g = ls_box(ss[7],3) ^ t_use(r,c)[i]; \
+    ss[0] ^= g; g = ff(g); k[8*(i)+ 8] = g ^= k[8*(i)]; \
+    ss[1] ^= ss[0]; k[8*(i)+ 9] = g ^= k[8*(i)+ 1]; \
+    ss[2] ^= ss[1]; k[8*(i)+10] = g ^= k[8*(i)+ 2]; \
+    ss[3] ^= ss[2]; k[8*(i)+11] = g ^= k[8*(i)+ 3]; \
+    g = ls_box(ss[3],0); \
+    ss[4] ^= g; g = ff(g); k[8*(i)+12] = g ^= k[8*(i)+ 4]; \
+    ss[5] ^= ss[4]; k[8*(i)+13] = g ^= k[8*(i)+ 5]; \
+    ss[6] ^= ss[5]; k[8*(i)+14] = g ^= k[8*(i)+ 6]; \
+    ss[7] ^= ss[6]; k[8*(i)+15] = g ^= k[8*(i)+ 7]; \
+}
+#define kdl8(k,i) \
+{   ss[0] ^= ls_box(ss[7],3) ^ t_use(r,c)[i]; k[8*(i)+ 8] = ss[0]; ss[1] ^= ss[0]; k[8*(i)+ 9] = ss[1]; \
+    ss[2] ^= ss[1]; k[8*(i)+10] = ss[2]; ss[3] ^= ss[2]; k[8*(i)+11] = ss[3]; \
+}
+
+#if defined(AES_128) || defined(AES_VAR)
+
+aes_rval aes_decrypt_key128(const void *in_key, aes_decrypt_ctx cx[1])
+{   aes_32t    ss[5];
+#ifdef  d_vars
+        d_vars;
+#endif
+    cx->ks[0] = ss[0] = word_in(in_key, 0);
+    cx->ks[1] = ss[1] = word_in(in_key, 1);
+    cx->ks[2] = ss[2] = word_in(in_key, 2);
+    cx->ks[3] = ss[3] = word_in(in_key, 3);
+
+#if DEC_UNROLL == NONE
+    {   aes_32t i;
+
+        for(i = 0; i < (11 * N_COLS - 1) / 4; ++i)
+            ke4(cx->ks, i);
+#if !(DEC_ROUND == NO_TABLES)
+        for(i = N_COLS; i < 10 * N_COLS; ++i)
+            cx->ks[i] = inv_mcol(cx->ks[i]);
+#endif
+    }
+#else
+    kdf4(cx->ks, 0);  kd4(cx->ks, 1);
+     kd4(cx->ks, 2);  kd4(cx->ks, 3);
+     kd4(cx->ks, 4);  kd4(cx->ks, 5);
+     kd4(cx->ks, 6);  kd4(cx->ks, 7);
+     kd4(cx->ks, 8); kdl4(cx->ks, 9);
+#endif
+
+    /* cx->ks[45] ^ cx->ks[52] ^ cx->ks[53] is zero for a 256 bit       */
+    /* key and must be non-zero for 128 and 192 bits keys   */
+    cx->ks[53] = cx->ks[45] = 0;
+    cx->ks[52] = 10;
+#ifdef AES_ERR_CHK
+    return aes_good;
+#endif
+}
+
+#endif
+
+#if defined(AES_192) || defined(AES_VAR)
+
+aes_rval aes_decrypt_key192(const void *in_key, aes_decrypt_ctx cx[1])
+{   aes_32t    ss[7];
+#ifdef  d_vars
+        d_vars;
+#endif
+    cx->ks[0] = ss[0] = word_in(in_key, 0);
+    cx->ks[1] = ss[1] = word_in(in_key, 1);
+    cx->ks[2] = ss[2] = word_in(in_key, 2);
+    cx->ks[3] = ss[3] = word_in(in_key, 3);
+
+#if DEC_UNROLL == NONE
+    cx->ks[4] = ss[4] = word_in(in_key, 4);
+    cx->ks[5] = ss[5] = word_in(in_key, 5);
+    {   aes_32t i;
+
+        for(i = 0; i < (13 * N_COLS - 1) / 6; ++i)
+            ke6(cx->ks, i);
+#if !(DEC_ROUND == NO_TABLES)
+        for(i = N_COLS; i < 12 * N_COLS; ++i)
+            cx->ks[i] = inv_mcol(cx->ks[i]);
+#endif
+    }
+#else
+    cx->ks[4] = ff(ss[4] = word_in(in_key, 4));
+    cx->ks[5] = ff(ss[5] = word_in(in_key, 5));
+    kdf6(cx->ks, 0); kd6(cx->ks, 1);
+    kd6(cx->ks, 2);  kd6(cx->ks, 3);
+    kd6(cx->ks, 4);  kd6(cx->ks, 5);
+    kd6(cx->ks, 6); kdl6(cx->ks, 7);
+#endif
+
+    /* cx->ks[45] ^ cx->ks[52] ^ cx->ks[53] is zero for a 256 bit       */
+    /* key and must be non-zero for 128 and 192 bits keys   */
+    cx->ks[53] = cx->ks[45];
+    cx->ks[52] = 12;
+#ifdef AES_ERR_CHK
+    return aes_good;
+#endif
+}
+
+#endif
+
+#if defined(AES_256) || defined(AES_VAR)
+
+aes_rval aes_decrypt_key256(const void *in_key, aes_decrypt_ctx cx[1])
+{   aes_32t    ss[8];
+#ifdef  d_vars
+        d_vars;
+#endif
+    cx->ks[0] = ss[0] = word_in(in_key, 0);
+    cx->ks[1] = ss[1] = word_in(in_key, 1);
+    cx->ks[2] = ss[2] = word_in(in_key, 2);
+    cx->ks[3] = ss[3] = word_in(in_key, 3);
+
+#if DEC_UNROLL == NONE
+    cx->ks[4] = ss[4] = word_in(in_key, 4);
+    cx->ks[5] = ss[5] = word_in(in_key, 5);
+    cx->ks[6] = ss[6] = word_in(in_key, 6);
+    cx->ks[7] = ss[7] = word_in(in_key, 7);
+    {   aes_32t i;
+
+        for(i = 0; i < (15 * N_COLS - 1) / 8; ++i)
+            ke8(cx->ks,  i);
+#if !(DEC_ROUND == NO_TABLES)
+        for(i = N_COLS; i < 14 * N_COLS; ++i)
+            cx->ks[i] = inv_mcol(cx->ks[i]);
+#endif
+    }
+#else
+    cx->ks[4] = ff(ss[4] = word_in(in_key, 4));
+    cx->ks[5] = ff(ss[5] = word_in(in_key, 5));
+    cx->ks[6] = ff(ss[6] = word_in(in_key, 6));
+    cx->ks[7] = ff(ss[7] = word_in(in_key, 7));
+    kdf8(cx->ks, 0); kd8(cx->ks, 1);
+    kd8(cx->ks, 2);  kd8(cx->ks, 3);
+    kd8(cx->ks, 4);  kd8(cx->ks, 5);
+    kdl8(cx->ks, 6);
+#endif
+#ifdef AES_ERR_CHK
+    return aes_good;
+#endif
+}
+
+#endif
+
+#if defined(AES_VAR)
+
+aes_rval aes_decrypt_key(const void *in_key, int key_len, aes_decrypt_ctx cx[1])
+{
+    switch(key_len)
+    {
+#ifdef AES_ERR_CHK
+    case 16: case 128: return aes_decrypt_key128(in_key, cx);
+    case 24: case 192: return aes_decrypt_key192(in_key, cx);
+    case 32: case 256: return aes_decrypt_key256(in_key, cx);
+    default: return aes_error;
+#else
+    case 16: case 128: aes_decrypt_key128(in_key, cx); return;
+    case 24: case 192: aes_decrypt_key192(in_key, cx); return;
+    case 32: case 256: aes_decrypt_key256(in_key, cx); return;
+#endif
+    }
+}
+
+#endif
+
+#endif
+
+#if defined(__cplusplus)
+}
+#endif

aestab.c

@@ -0,0 +1,232 @@
+/*
+ ---------------------------------------------------------------------------
+ Copyright (c) 2003, Dr Brian Gladman <brg@gladman.me.uk>, Worcester, UK.
+ All rights reserved.
+
+ LICENSE TERMS
+
+ The free distribution and use of this software in both source and binary
+ form is allowed (with or without changes) provided that:
+
+   1. distributions of this source code include the above copyright
+      notice, this list of conditions and the following disclaimer;
+
+   2. distributions in binary form include the above copyright
+      notice, this list of conditions and the following disclaimer
+      in the documentation and/or other associated materials;
+
+   3. the copyright holder's name is not used to endorse products
+      built using this software without specific written permission.
+
+ ALTERNATIVELY, provided that this notice is retained in full, this product
+ may be distributed under the terms of the GNU General Public License (GPL),
+ in which case the provisions of the GPL apply INSTEAD OF those given above.
+
+ DISCLAIMER
+
+ This software is provided 'as is' with no explicit or implied warranties
+ in respect of its properties, including, but not limited to, correctness
+ and/or fitness for purpose.
+ ---------------------------------------------------------------------------
+ Issue Date: 26/08/2003
+
+*/
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+#define DO_TABLES
+
+#include "aesopt.h"
+
+#if defined(FIXED_TABLES)
+
+/* implemented in case of wrong call for fixed tables */
+
+void gen_tabs(void)
+{
+}
+
+#else   /* dynamic table generation */
+
+#if !defined(FF_TABLES)
+
+/*  Generate the tables for the dynamic table option
+
+    It will generally be sensible to use tables to compute finite
+    field multiplies and inverses but where memory is scarse this
+    code might sometimes be better. But it only has effect during
+    initialisation so its pretty unimportant in overall terms.
+*/
+
+/*  return 2 ^ (n - 1) where n is the bit number of the highest bit
+    set in x with x in the range 1 < x < 0x00000200.   This form is
+    used so that locals within fi can be bytes rather than words
+*/
+
+static aes_08t hibit(const aes_32t x)
+{   aes_08t r = (aes_08t)((x >> 1) | (x >> 2));
+
+    r |= (r >> 2);
+    r |= (r >> 4);
+    return (r + 1) >> 1;
+}
+
+/* return the inverse of the finite field element x */
+
+static aes_08t fi(const aes_08t x)
+{   aes_08t p1 = x, p2 = BPOLY, n1 = hibit(x), n2 = 0x80, v1 = 1, v2 = 0;
+
+    if(x < 2) return x;
+
+    for(;;)
+    {
+        if(!n1) return v1;
+
+        while(n2 >= n1)
+        {
+            n2 /= n1; p2 ^= p1 * n2; v2 ^= v1 * n2; n2 = hibit(p2);
+        }
+
+        if(!n2) return v2;
+
+        while(n1 >= n2)
+        {
+            n1 /= n2; p1 ^= p2 * n1; v1 ^= v2 * n1; n1 = hibit(p1);
+        }
+    }
+}
+
+#endif
+
+/* The forward and inverse affine transformations used in the S-box */
+
+#define fwd_affine(x) \
+    (w = (aes_32t)x, w ^= (w<<1)^(w<<2)^(w<<3)^(w<<4), 0x63^(aes_08t)(w^(w>>8)))
+
+#define inv_affine(x) \
+    (w = (aes_32t)x, w = (w<<1)^(w<<3)^(w<<6), 0x05^(aes_08t)(w^(w>>8)))
+
+static int init = 0;
+
+void gen_tabs(void)
+{   aes_32t  i, w;
+
+#if defined(FF_TABLES)
+
+    aes_08t  pow[512], log[256];
+
+    if(init) return;
+    /*  log and power tables for GF(2^8) finite field with
+        WPOLY as modular polynomial - the simplest primitive
+        root is 0x03, used here to generate the tables
+    */
+
+    i = 0; w = 1;
+    do
+    {
+        pow[i] = (aes_08t)w;
+        pow[i + 255] = (aes_08t)w;
+        log[w] = (aes_08t)i++;
+        w ^=  (w << 1) ^ (w & 0x80 ? WPOLY : 0);
+    }
+    while (w != 1);
+
+#else
+    if(init) return;
+#endif
+
+    for(i = 0, w = 1; i < RC_LENGTH; ++i)
+    {
+        t_set(r,c)[i] = bytes2word(w, 0, 0, 0);
+        w = f2(w);
+    }
+
+    for(i = 0; i < 256; ++i)
+    {   aes_08t    b;
+
+        b = fwd_affine(fi((aes_08t)i));
+        w = bytes2word(f2(b), b, b, f3(b));
+
+#ifdef  SBX_SET
+        t_set(s,box)[i] = b;
+#endif
+
+#ifdef  FT1_SET                 /* tables for a normal encryption round */
+        t_set(f,n)[i] = w;
+#endif
+#ifdef  FT4_SET
+        t_set(f,n)[0][i] = w;
+        t_set(f,n)[1][i] = upr(w,1);
+        t_set(f,n)[2][i] = upr(w,2);
+        t_set(f,n)[3][i] = upr(w,3);
+#endif
+        w = bytes2word(b, 0, 0, 0);
+
+#ifdef  FL1_SET                 /* tables for last encryption round (may also   */
+        t_set(f,l)[i] = w;        /* be used in the key schedule)                 */
+#endif
+#ifdef  FL4_SET
+        t_set(f,l)[0][i] = w;
+        t_set(f,l)[1][i] = upr(w,1);
+        t_set(f,l)[2][i] = upr(w,2);
+        t_set(f,l)[3][i] = upr(w,3);
+#endif
+
+#ifdef  LS1_SET                 /* table for key schedule if t_set(f,l) above is    */
+        t_set(l,s)[i] = w;      /* not of the required form                     */
+#endif
+#ifdef  LS4_SET
+        t_set(l,s)[0][i] = w;
+        t_set(l,s)[1][i] = upr(w,1);
+        t_set(l,s)[2][i] = upr(w,2);
+        t_set(l,s)[3][i] = upr(w,3);
+#endif
+
+        b = fi(inv_affine((aes_08t)i));
+        w = bytes2word(fe(b), f9(b), fd(b), fb(b));
+
+#ifdef  IM1_SET                 /* tables for the inverse mix column operation  */
+        t_set(i,m)[b] = w;
+#endif
+#ifdef  IM4_SET
+        t_set(i,m)[0][b] = w;
+        t_set(i,m)[1][b] = upr(w,1);
+        t_set(i,m)[2][b] = upr(w,2);
+        t_set(i,m)[3][b] = upr(w,3);
+#endif
+
+#ifdef  ISB_SET
+        t_set(i,box)[i] = b;
+#endif
+#ifdef  IT1_SET                 /* tables for a normal decryption round */
+        t_set(i,n)[i] = w;
+#endif
+#ifdef  IT4_SET
+        t_set(i,n)[0][i] = w;
+        t_set(i,n)[1][i] = upr(w,1);
+        t_set(i,n)[2][i] = upr(w,2);
+        t_set(i,n)[3][i] = upr(w,3);
+#endif
+        w = bytes2word(b, 0, 0, 0);
+#ifdef  IL1_SET                 /* tables for last decryption round */
+        t_set(i,l)[i] = w;
+#endif
+#ifdef  IL4_SET
+        t_set(i,l)[0][i] = w;
+        t_set(i,l)[1][i] = upr(w,1);
+        t_set(i,l)[2][i] = upr(w,2);
+        t_set(i,l)[3][i] = upr(w,3);
+#endif
+    }
+    init = 1;
+}
+
+#endif
+
+#if defined(__cplusplus)
+}
+#endif
+

agi/.cvsignore

@@ -1,2 +0,0 @@
-eagi-test
-eagi-sphinx-test

agi/Makefile

@@ -15,19 +15,29 @@ AGIS=agi-test.agi eagi-test eagi-sphinx-test
 
 CFLAGS+=
 
-all: $(AGIS)
+all: depend $(AGIS)
 
 install: all
-	for x in $(AGIS); do $(INSTALL) -m 755 $$x $(AGI_DIR) ; done
+	mkdir -p $(DESTDIR)$(AGI_DIR)
+	for x in $(AGIS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(AGI_DIR) ; done
 
 eagi-test: eagi-test.o
-	$(CC) -o eagi-test eagi-test.o
-	
+	$(CC) $(CFLAGS) -o eagi-test eagi-test.o
+
 eagi-sphinx-test: eagi-sphinx-test.o
-	$(CC) -o eagi-sphinx-test eagi-sphinx-test.o
+	$(CC) $(CFLAGS) -o eagi-sphinx-test eagi-sphinx-test.o
 
 clean:
-	rm -f *.so *.o look
+	rm -f *.so *.o look .depend eagi-test eagi-sphinx-test
 
 %.so : %.o
 	$(CC) -shared -Xlinker -x -o $@ $<
+
+ifneq ($(wildcard .depend),)
+include .depend
+endif
+
+depend: .depend
+
+.depend:
+	../mkdep $(CFLAGS) `ls *.c`

agi/agi-test.agi

@@ -1,5 +1,11 @@
 #!/usr/bin/perl
+use strict;
+
 $|=1;
+
+# Setup some variables
+my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;
+
 while(<STDIN>) {
 	chomp;
 	last unless length($_);
@@ -9,7 +15,7 @@ while(<STDIN>) {
 }
 
 print STDERR "AGI Environment Dump:\n";
-foreach $i (sort keys %AGI) {
+foreach my $i (sort keys %AGI) {
 	print STDERR " -- $i = $AGI{$i}\n";
 }
 
@@ -35,38 +41,38 @@ sub checkresult {
 
 print STDERR "1.  Testing 'sendfile'...";
 print "STREAM FILE beep \"\"\n";
-$result = <STDIN>;
-checkresult($result);
+my $result = <STDIN>;
+&checkresult($result);
 
 print STDERR "2.  Testing 'sendtext'...";
 print "SEND TEXT \"hello world\"\n";
-$result = <STDIN>;
-checkresult($result);
+my $result = <STDIN>;
+&checkresult($result);
 
 print STDERR "3.  Testing 'sendimage'...";
 print "SEND IMAGE asterisk-image\n";
-$result = <STDIN>;
-checkresult($result);
+my $result = <STDIN>;
+&checkresult($result);
 
 print STDERR "4.  Testing 'saynumber'...";
 print "SAY NUMBER 192837465 \"\"\n";
-$result = <STDIN>;
-checkresult($result);
+my $result = <STDIN>;
+&checkresult($result);
 
 print STDERR "5.  Testing 'waitdtmf'...";
 print "WAIT FOR DIGIT 1000\n";
-$result = <STDIN>;
-checkresult($result);
+my $result = <STDIN>;
+&checkresult($result);
 
 print STDERR "6.  Testing 'record'...";
 print "RECORD FILE testagi gsm 1234 3000\n";
-$result = <STDIN>;
-checkresult($result);
+my $result = <STDIN>;
+&checkresult($result);
 
 print STDERR "6a.  Testing 'record' playback...";
 print "STREAM FILE testagi \"\"\n";
-$result = <STDIN>;
-checkresult($result);
+my $result = <STDIN>;
+&checkresult($result);
 
 print STDERR "================== Complete ======================\n";
 print STDERR "$tests tests completed, $pass passed, $fail failed\n";

agi/eagi-sphinx-test.c

@@ -13,9 +13,9 @@
 #include <string.h>
 #include <sys/select.h>
 #include <fcntl.h>
-#include <arpa/inet.h>
 #include <sys/socket.h>
 #include <netinet/in.h>
+#include <arpa/inet.h>
 #include <netdb.h>
 
 #define AUDIO_FILENO (STDERR_FILENO + 1)
@@ -44,7 +44,7 @@ static int connect_sphinx(void)
 	sin.sin_family = AF_INET;
 	sin.sin_port = htons(SPHINX_PORT);
 	memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
-	if (connect(sphinx_sock, &sin, sizeof(sin))) {
+	if (connect(sphinx_sock, (struct sockaddr *)&sin, sizeof(sin))) {
 		fprintf(stderr, "Unable to connect on socket: %s\n", strerror(errno));
 		close(sphinx_sock);
 		sphinx_sock = -1;

agi/fastagi-test

@@ -0,0 +1,94 @@
+#!/usr/bin/perl
+use strict;
+use Socket;
+use Carp;
+use IO::Handle;
+
+my $port = 4573;
+
+$|=1;
+
+# Setup some variables
+my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;
+
+sub checkresult {
+	my ($res) = @_;
+	my $retval;
+	$tests++;
+	chomp $res;
+	if ($res =~ /^200/) {
+		$res =~ /result=(-?\d+)/;
+		if (!length($1)) {
+			print STDERR "FAIL ($res)\n";
+			$fail++;
+		} else {
+			print STDERR "PASS ($1)\n";
+			$pass++;
+		}
+	} else {
+		print STDERR "FAIL (unexpected result '$res')\n";
+		$fail++;
+	}
+}
+
+socket(SERVER, PF_INET, SOCK_STREAM, 0);
+setsockopt(SERVER, SOL_SOCKET, SO_REUSEADDR, pack("l", 1));
+bind(SERVER, sockaddr_in($port, INADDR_ANY)) || die("can't bind\n");
+listen(SERVER, SOMAXCONN);
+
+for(;;) {
+	my $raddr = accept(CLIENT, SERVER);
+	my ($s, $p) = sockaddr_in($raddr);
+	CLIENT->autoflush(1);
+	while(<CLIENT>) {
+		chomp;
+		last unless length($_);
+		if (/^agi_(\w+)\:\s+(.*)$/) {
+			$AGI{$1} = $2;
+		}
+	}
+	print STDERR "AGI Environment Dump from $s:$p --\n";
+	foreach my $i (sort keys %AGI) {
+		print STDERR " -- $i = $AGI{$i}\n";
+	}
+
+	print STDERR "1.  Testing 'sendfile'...";
+	print CLIENT "STREAM FILE beep \"\"\n";
+	my $result = <CLIENT>;
+	&checkresult($result);
+
+	print STDERR "2.  Testing 'sendtext'...";
+	print CLIENT "SEND TEXT \"hello world\"\n";
+	my $result = <CLIENT>;
+	&checkresult($result);
+
+	print STDERR "3.  Testing 'sendimage'...";
+	print CLIENT "SEND IMAGE asterisk-image\n";
+	my $result = <CLIENT>;
+	&checkresult($result);
+
+	print STDERR "4.  Testing 'saynumber'...";
+	print CLIENT "SAY NUMBER 192837465 \"\"\n";
+	my $result = <CLIENT>;
+	&checkresult($result);
+
+	print STDERR "5.  Testing 'waitdtmf'...";
+	print CLIENT "WAIT FOR DIGIT 1000\n";
+	my $result = <CLIENT>;
+	&checkresult($result);
+
+	print STDERR "6.  Testing 'record'...";
+	print CLIENT "RECORD FILE testagi gsm 1234 3000\n";
+	my $result = <CLIENT>;
+	&checkresult($result);
+
+	print STDERR "6a.  Testing 'record' playback...";
+	print CLIENT "STREAM FILE testagi \"\"\n";
+	my $result = <CLIENT>;
+	&checkresult($result);
+	close(CLIENT);
+	print STDERR "================== Complete ======================\n";
+	print STDERR "$tests tests completed, $pass passed, $fail failed\n";
+	print STDERR "==================================================\n";
+}
+

app.c

@@ -1,11 +1,11 @@
 /*
  * Asterisk -- A telephony toolkit for Linux.
  *
- * Channel Management
+ * Convenient Application Routines
  * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999-2004, Digium, Inc.
  *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
  *
  * This program is free software, distributed under the terms of
  * the GNU General Public License
@@ -13,7 +13,6 @@
 
 #include <stdio.h>
 #include <stdlib.h>
-#include <pthread.h>
 #include <string.h>
 #include <sys/time.h>
 #include <signal.h>
@@ -21,27 +20,35 @@
 #include <unistd.h>
 #include <dirent.h>
 #include <asterisk/channel.h>
+#include <asterisk/pbx.h>
 #include <asterisk/file.h>
 #include <asterisk/app.h>
 #include <asterisk/dsp.h>
 #include <asterisk/logger.h>
 #include <asterisk/options.h>
+#include <asterisk/utils.h>
+#include <asterisk/lock.h>
 #include "asterisk.h"
 #include "astconf.h"
 
+#define MAX_OTHER_FORMATS 10
+
 /* set timeout to 0 for "standard" timeouts. Set timeout to -1 for 
    "ludicrous time" (essentially never times out) */
 int ast_app_getdata(struct ast_channel *c, char *prompt, char *s, int maxlen, int timeout)
 {
 	int res,to,fto;
 	/* XXX Merge with full version? XXX */
+	if (maxlen)
+		s[0] = '\0';
 	if (prompt) {
 		res = ast_streamfile(c, prompt, c->language);
 		if (res < 0)
 			return res;
 	}
-	fto = 6000;
-	to = 2000;
+	fto = c->pbx ? c->pbx->rtimeout * 1000 : 6000;
+	to = c->pbx ? c->pbx->dtimeout * 1000 : 2000;
+
 	if (timeout > 0) fto = to = timeout;
 	if (timeout < 0) fto = to = 1000000000;
 	res = ast_readstring(c, s, maxlen, to, fto, "#");
@@ -129,6 +136,13 @@ int ast_app_getvoice(struct ast_channel *c, char *dest, char *dstfmt, char *prom
 					ast_frfree(f);
 					break;
 				}
+				res = ast_writestream(writer, f);
+				if (res < 0) {
+					ast_log(LOG_WARNING, "Failed to write to stream at %s!\n", dest);
+					ast_frfree(f);
+					break;
+				}
+					
 			}
 			ast_frfree(f);
 		}
@@ -148,23 +162,31 @@ int ast_app_has_voicemail(const char *mailbox)
 	char fn[256];
 	char tmp[256]="";
 	char *mb, *cur;
+	char *context;
 	int ret;
 	/* If no mailbox, return immediately */
-	if (!strlen(mailbox))
+	if (ast_strlen_zero(mailbox))
 		return 0;
 	if (strchr(mailbox, ',')) {
-		strncpy(tmp, mailbox, sizeof(tmp));
+		strncpy(tmp, mailbox, sizeof(tmp) - 1);
 		mb = tmp;
 		ret = 0;
-		while((cur = strsep(&mb, ", "))) {
-			if (strlen(cur)) {
+		while((cur = strsep(&mb, ","))) {
+			if (!ast_strlen_zero(cur)) {
 				if (ast_app_has_voicemail(cur))
 					return 1; 
 			}
 		}
 		return 0;
 	}
-	snprintf(fn, sizeof(fn), "%s/vm/%s/INBOX", (char *)ast_config_AST_SPOOL_DIR, mailbox);
+	strncpy(tmp, mailbox, sizeof(tmp) - 1);
+	context = strchr(tmp, '@');
+	if (context) {
+		*context = '\0';
+		context++;
+	} else
+		context = "default";
+	snprintf(fn, sizeof(fn), "%s/voicemail/%s/%s/INBOX", (char *)ast_config_AST_SPOOL_DIR, context, tmp);
 	dir = opendir(fn);
 	if (!dir)
 		return 0;
@@ -185,21 +207,22 @@ int ast_app_messagecount(const char *mailbox, int *newmsgs, int *oldmsgs)
 	char fn[256];
 	char tmp[256]="";
 	char *mb, *cur;
+	char *context;
 	int ret;
 	if (newmsgs)
 		*newmsgs = 0;
 	if (oldmsgs)
 		*oldmsgs = 0;
 	/* If no mailbox, return immediately */
-	if (!strlen(mailbox))
+	if (ast_strlen_zero(mailbox))
 		return 0;
 	if (strchr(mailbox, ',')) {
 		int tmpnew, tmpold;
-		strncpy(tmp, mailbox, sizeof(tmp));
+		strncpy(tmp, mailbox, sizeof(tmp) - 1);
 		mb = tmp;
 		ret = 0;
 		while((cur = strsep(&mb, ", "))) {
-			if (strlen(cur)) {
+			if (!ast_strlen_zero(cur)) {
 				if (ast_app_messagecount(cur, newmsgs ? &tmpnew : NULL, oldmsgs ? &tmpold : NULL))
 					return -1;
 				else {
@@ -212,8 +235,15 @@ int ast_app_messagecount(const char *mailbox, int *newmsgs, int *oldmsgs)
 		}
 		return 0;
 	}
+	strncpy(tmp, mailbox, sizeof(tmp) - 1);
+	context = strchr(tmp, '@');
+	if (context) {
+		*context = '\0';
+		context++;
+	} else
+		context = "default";
 	if (newmsgs) {
-		snprintf(fn, sizeof(fn), "%s/vm/%s/INBOX", (char *)ast_config_AST_SPOOL_DIR, mailbox);
+		snprintf(fn, sizeof(fn), "%s/voicemail/%s/%s/INBOX", (char *)ast_config_AST_SPOOL_DIR, context, tmp);
 		dir = opendir(fn);
 		if (dir) {
 			while ((de = readdir(dir))) {
@@ -226,7 +256,7 @@ int ast_app_messagecount(const char *mailbox, int *newmsgs, int *oldmsgs)
 		}
 	}
 	if (oldmsgs) {
-		snprintf(fn, sizeof(fn), "%s/vm/%s/Old", (char *)ast_config_AST_SPOOL_DIR, mailbox);
+		snprintf(fn, sizeof(fn), "%s/voicemail/%s/%s/Old", (char *)ast_config_AST_SPOOL_DIR, context, tmp);
 		dir = opendir(fn);
 		if (dir) {
 			while ((de = readdir(dir))) {
@@ -240,3 +270,751 @@ int ast_app_messagecount(const char *mailbox, int *newmsgs, int *oldmsgs)
 	}
 	return 0;
 }
+
+int ast_dtmf_stream(struct ast_channel *chan,struct ast_channel *peer,char *digits,int between) 
+{
+	char *ptr=NULL;
+	int res=0;
+	struct ast_frame f;
+	if (!between)
+		between = 100;
+
+	if (peer)
+		res = ast_autoservice_start(peer);
+
+	if (!res) {
+		res = ast_waitfor(chan,100);
+		if (res > -1) {
+			for (ptr=digits; *ptr; ptr++) {
+				if (*ptr == 'w') {
+					res = ast_safe_sleep(chan, 500);
+					if (res) 
+						break;
+					continue;
+				}
+				memset(&f, 0, sizeof(f));
+				f.frametype = AST_FRAME_DTMF;
+				f.subclass = *ptr;
+				f.src = "ast_dtmf_stream";
+				if (strchr("0123456789*#abcdABCD",*ptr)==NULL) {
+					ast_log(LOG_WARNING, "Illegal DTMF character '%c' in string. (0-9*#aAbBcCdD allowed)\n",*ptr);
+				} else {
+					res = ast_write(chan, &f);
+					if (res) 
+						break;
+					/* pause between digits */
+					res = ast_safe_sleep(chan,between);
+					if (res) 
+						break;
+				}
+			}
+		}
+		if (peer)
+			res = ast_autoservice_stop(peer);
+	}
+	return res;
+}
+
+struct linear_state {
+	int fd;
+	int autoclose;
+	int allowoverride;
+	int origwfmt;
+};
+
+static void linear_release(struct ast_channel *chan, void *params)
+{
+	struct linear_state *ls = params;
+	if (ls->origwfmt && ast_set_write_format(chan, ls->origwfmt)) {
+		ast_log(LOG_WARNING, "Unable to restore channel '%s' to format '%d'\n", chan->name, ls->origwfmt);
+	}
+	if (ls->autoclose)
+		close(ls->fd);
+	free(params);
+}
+
+static int linear_generator(struct ast_channel *chan, void *data, int len, int samples)
+{
+	struct ast_frame f;
+	short buf[2048 + AST_FRIENDLY_OFFSET / 2];
+	struct linear_state *ls = data;
+	int res;
+	len = samples * 2;
+	if (len > sizeof(buf) - AST_FRIENDLY_OFFSET) {
+		ast_log(LOG_WARNING, "Can't generate %d bytes of data!\n" ,len);
+		len = sizeof(buf) - AST_FRIENDLY_OFFSET;
+	}
+	memset(&f, 0, sizeof(f));
+	res = read(ls->fd, buf + AST_FRIENDLY_OFFSET/2, len);
+	if (res > 0) {
+		f.frametype = AST_FRAME_VOICE;
+		f.subclass = AST_FORMAT_SLINEAR;
+		f.data = buf + AST_FRIENDLY_OFFSET/2;
+		f.datalen = res;
+		f.samples = res / 2;
+		f.offset = AST_FRIENDLY_OFFSET;
+		ast_write(chan, &f);
+		if (res == len)
+			return 0;
+	}
+	return -1;
+}
+
+static void *linear_alloc(struct ast_channel *chan, void *params)
+{
+	struct linear_state *ls;
+	/* In this case, params is already malloc'd */
+	if (params) {
+		ls = params;
+		if (ls->allowoverride)
+			chan->writeinterrupt = 1;
+		else
+			chan->writeinterrupt = 0;
+		ls->origwfmt = chan->writeformat;
+		if (ast_set_write_format(chan, AST_FORMAT_SLINEAR)) {
+			ast_log(LOG_WARNING, "Unable to set '%s' to linear format (write)\n", chan->name);
+			free(ls);
+			ls = params = NULL;
+		}
+	}
+	return params;
+}
+
+static struct ast_generator linearstream = 
+{
+	alloc: linear_alloc,
+	release: linear_release,
+	generate: linear_generator,
+};
+
+int ast_linear_stream(struct ast_channel *chan, const char *filename, int fd, int allowoverride)
+{
+	struct linear_state *lin;
+	char tmpf[256] = "";
+	int res = -1;
+	int autoclose = 0;
+	if (fd < 0) {
+		if (!filename || ast_strlen_zero(filename))
+			return -1;
+		autoclose = 1;
+		if (filename[0] == '/') 
+			strncpy(tmpf, filename, sizeof(tmpf) - 1);
+		else
+			snprintf(tmpf, sizeof(tmpf), "%s/%s/%s", (char *)ast_config_AST_VAR_DIR, "sounds", filename);
+		fd = open(tmpf, O_RDONLY);
+		if (fd < 0){
+			ast_log(LOG_WARNING, "Unable to open file '%s': %s\n", tmpf, strerror(errno));
+			return -1;
+		}
+	}
+	lin = malloc(sizeof(struct linear_state));
+	if (lin) {
+		memset(lin, 0, sizeof(lin));
+		lin->fd = fd;
+		lin->allowoverride = allowoverride;
+		lin->autoclose = autoclose;
+		res = ast_activate_generator(chan, &linearstream, lin);
+	}
+	return res;
+}
+
+int ast_control_streamfile(struct ast_channel *chan, char *file, char *fwd, char *rev, char *stop, char *pause, int skipms) 
+{
+	struct timeval started, ended;
+	long elapsed = 0,last_elapsed =0;
+	char *breaks=NULL;
+	char *end=NULL;
+	int blen=2;
+	int res=0;
+
+	if (stop)
+		blen += strlen(stop);
+	if (pause)
+		blen += strlen(pause);
+
+	if (blen > 2) {
+		breaks = alloca(blen + 1);
+		breaks[0] = '\0';
+		strcat(breaks, stop);
+		strcat(breaks, pause);
+	}
+	if (chan->_state != AST_STATE_UP)
+		res = ast_answer(chan);
+
+	if (chan)
+		ast_stopstream(chan);
+
+
+	if (file) {
+		if ((end = strchr(file,':'))) {
+			if (!strcasecmp(end, ":end")) {
+				*end = '\0';
+				end++;
+			}
+		}
+	}
+
+	for (;;) {
+		gettimeofday(&started,NULL);
+
+		if (chan)
+			ast_stopstream(chan);
+		res = ast_streamfile(chan, file, chan->language);
+		if (!res) {
+			if (end) {
+				ast_seekstream(chan->stream, 0, SEEK_END);
+				end=NULL;
+			}
+			res = 1;
+			if (elapsed) {
+				ast_stream_fastforward(chan->stream, elapsed);
+				last_elapsed = elapsed - 200;
+			}
+			if (res)
+				res = ast_waitstream_fr(chan, breaks, fwd, rev, skipms);
+			else
+				break;
+		}
+
+		if (res < 1)
+			break;
+
+		if (pause != NULL && strchr(pause, res)) {
+			gettimeofday(&ended, NULL);
+			elapsed = (((ended.tv_sec * 1000) + ended.tv_usec / 1000) - ((started.tv_sec * 1000) + started.tv_usec / 1000) + last_elapsed);
+			for(;;) {
+				if (chan)
+					ast_stopstream(chan);
+				res = ast_waitfordigit(chan, 1000);
+				if (res == 0)
+					continue;
+				else if (res == -1 || strchr(pause, res) || (stop && strchr(stop, res)))
+					break;
+			}
+			if (res == *pause) {
+				res = 0;
+				continue;
+			}
+		}
+		if (res == -1)
+			break;
+
+		/* if we get one of our stop chars, return it to the calling function */
+		if (stop && strchr(stop, res)) {
+			/* res = 0; */
+			break;
+		}
+	}
+	if (chan)
+		ast_stopstream(chan);
+
+	return res;
+}
+
+int ast_play_and_wait(struct ast_channel *chan, char *fn)
+{
+	int d;
+	d = ast_streamfile(chan, fn, chan->language);
+	if (d)
+		return d;
+	d = ast_waitstream(chan, AST_DIGIT_ANY);
+	ast_stopstream(chan);
+	return d;
+}
+
+static int global_silence_threshold = 128;
+static int global_maxsilence = 0;
+
+int ast_play_and_record(struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int silencethreshold, int maxsilence, const char *path)
+{
+	int d;
+	char *fmts;
+	char comment[256];
+	int x, fmtcnt=1, res=-1,outmsg=0;
+	struct ast_frame *f;
+	struct ast_filestream *others[MAX_OTHER_FORMATS];
+	char *sfmt[MAX_OTHER_FORMATS];
+	char *stringp=NULL;
+	time_t start, end;
+	struct ast_dsp *sildet=NULL;   	/* silence detector dsp */
+	int totalsilence = 0;
+	int dspsilence = 0;
+	int gotsilence = 0;		/* did we timeout for silence? */
+	int rfmt=0;
+
+	if (silencethreshold < 0)
+		silencethreshold = global_silence_threshold;
+
+	if (maxsilence < 0)
+		maxsilence = global_maxsilence;
+
+	/* barf if no pointer passed to store duration in */
+	if (duration == NULL) {
+		ast_log(LOG_WARNING, "Error play_and_record called without duration pointer\n");
+		return -1;
+	}
+
+	ast_log(LOG_DEBUG,"play_and_record: %s, %s, '%s'\n", playfile ? playfile : "<None>", recordfile, fmt);
+	snprintf(comment,sizeof(comment),"Playing %s, Recording to: %s on %s\n", playfile ? playfile : "<None>", recordfile, chan->name);
+
+	if (playfile) {
+		d = ast_play_and_wait(chan, playfile);
+		if (d > -1)
+			d = ast_streamfile(chan, "beep",chan->language);
+		if (!d)
+			d = ast_waitstream(chan,"");
+		if (d < 0)
+			return -1;
+	}
+
+	fmts = ast_strdupa(fmt);
+
+	stringp=fmts;
+	strsep(&stringp, "|");
+	ast_log(LOG_DEBUG,"Recording Formats: sfmts=%s\n", fmts);
+	sfmt[0] = ast_strdupa(fmts);
+
+	while((fmt = strsep(&stringp, "|"))) {
+		if (fmtcnt > MAX_OTHER_FORMATS - 1) {
+			ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app_voicemail.c\n");
+			break;
+		}
+		sfmt[fmtcnt++] = ast_strdupa(fmt);
+	}
+
+	time(&start);
+	end=start;  /* pre-initialize end to be same as start in case we never get into loop */
+	for (x=0;x<fmtcnt;x++) {
+		others[x] = ast_writefile(recordfile, sfmt[x], comment, O_TRUNC, 0, 0700);
+		ast_verbose( VERBOSE_PREFIX_3 "x=%i, open writing:  %s format: %s, %p\n", x, recordfile, sfmt[x], others[x]);
+
+		if (!others[x]) {
+			break;
+		}
+	}
+
+	if (path)
+		ast_unlock_path(path);
+
+	
+	if (maxsilence > 0) {
+		sildet = ast_dsp_new(); /* Create the silence detector */
+		if (!sildet) {
+			ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
+			return -1;
+		}
+		ast_dsp_set_threshold(sildet, silencethreshold);
+		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");
+			ast_dsp_free(sildet);
+			return -1;
+		}
+	}
+
+	if (x == fmtcnt) {
+	/* Loop forever, writing the packets we read to the writer(s), until
+	   we read a # or get a hangup */
+		f = NULL;
+		for(;;) {
+		 	res = ast_waitfor(chan, 2000);
+			if (!res) {
+				ast_log(LOG_DEBUG, "One waitfor failed, trying another\n");
+				/* Try one more time in case of masq */
+			 	res = ast_waitfor(chan, 2000);
+				if (!res) {
+					ast_log(LOG_WARNING, "No audio available on %s??\n", chan->name);
+					res = -1;
+				}
+			}
+
+			if (res < 0) {
+				f = NULL;
+				break;
+			}
+			f = ast_read(chan);
+			if (!f)
+				break;
+			if (f->frametype == AST_FRAME_VOICE) {
+				/* write each format */
+				for (x=0;x<fmtcnt;x++) {
+					res = ast_writestream(others[x], f);
+				}
+
+				/* Silence Detection */
+				if (maxsilence > 0) {
+					dspsilence = 0;
+					ast_dsp_silence(sildet, f, &dspsilence);
+					if (dspsilence)
+						totalsilence = dspsilence;
+					else
+						totalsilence = 0;
+
+					if (totalsilence > maxsilence) {
+						/* Ended happily with silence */
+                                        	if (option_verbose > 2)
+                                                	ast_verbose( VERBOSE_PREFIX_3 "Recording automatically stopped after a silence of %d seconds\n", totalsilence/1000);
+						ast_frfree(f);
+						gotsilence = 1;
+						outmsg=2;
+						break;
+					}
+				}
+				/* Exit on any error */
+				if (res) {
+					ast_log(LOG_WARNING, "Error writing frame\n");
+					ast_frfree(f);
+					break;
+				}
+			} else if (f->frametype == AST_FRAME_VIDEO) {
+				/* Write only once */
+				ast_writestream(others[0], f);
+			} else if (f->frametype == AST_FRAME_DTMF) {
+				if (f->subclass == '#') {
+					if (option_verbose > 2)
+						ast_verbose( VERBOSE_PREFIX_3 "User ended message by pressing %c\n", f->subclass);
+					res = '#';
+					outmsg = 2;
+					ast_frfree(f);
+					break;
+				}
+			}
+				if (f->subclass == '0') {
+				/* Check for a '0' during message recording also, in case caller wants operator */
+					if (option_verbose > 2)
+						ast_verbose(VERBOSE_PREFIX_3 "User cancelled by pressing %c\n", f->subclass);
+					res = '0';
+					outmsg = 0;
+					ast_frfree(f);
+					break;
+				}
+			if (maxtime) {
+				time(&end);
+				if (maxtime < (end - start)) {
+					if (option_verbose > 2)
+						ast_verbose( VERBOSE_PREFIX_3 "Took too long, cutting it short...\n");
+					outmsg = 2;
+					res = 't';
+					ast_frfree(f);
+					break;
+				}
+			}
+			ast_frfree(f);
+		}
+		if (end == start) time(&end);
+		if (!f) {
+			if (option_verbose > 2)
+				ast_verbose( VERBOSE_PREFIX_3 "User hung up\n");
+			res = -1;
+			outmsg=1;
+		}
+	} else {
+		ast_log(LOG_WARNING, "Error creating writestream '%s', format '%s'\n", recordfile, sfmt[x]);
+	}
+
+	*duration = end - start;
+
+	for (x=0;x<fmtcnt;x++) {
+		if (!others[x])
+			break;
+		if (res > 0) {
+			if (totalsilence)
+				ast_stream_rewind(others[x], totalsilence-200);
+			else
+				ast_stream_rewind(others[x], 200);
+		}
+		ast_truncstream(others[x]);
+		ast_closestream(others[x]);
+	}
+	if (rfmt) {
+		if (ast_set_read_format(chan, rfmt)) {
+			ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_getformatname(rfmt), chan->name);
+		}
+	}
+	if (outmsg > 1) {
+		/* Let them know recording is stopped */
+		if(!ast_streamfile(chan, "auth-thankyou", chan->language))
+			ast_waitstream(chan, "");
+	}
+	if (sildet)
+		ast_dsp_free(sildet);
+	return res;
+}
+
+int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int beep, int silencethreshold, int maxsilence)
+{
+	int d = 0;
+	char *fmts;
+	char comment[256];
+	int x, fmtcnt=1, res=-1,outmsg=0;
+	struct ast_frame *f;
+	struct ast_filestream *others[MAX_OTHER_FORMATS];
+	struct ast_filestream *realfiles[MAX_OTHER_FORMATS];
+	char *sfmt[MAX_OTHER_FORMATS];
+	char *stringp=NULL;
+	time_t start, end;
+	struct ast_dsp *sildet;   	/* silence detector dsp */
+	int totalsilence = 0;
+	int dspsilence = 0;
+	int gotsilence = 0;		/* did we timeout for silence? */
+	int rfmt=0;	
+	char prependfile[80];
+	
+	if (silencethreshold < 0)
+		silencethreshold = global_silence_threshold;
+
+	if (maxsilence < 0)
+		maxsilence = global_maxsilence;
+
+	/* barf if no pointer passed to store duration in */
+	if (duration == NULL) {
+		ast_log(LOG_WARNING, "Error play_and_prepend called without duration pointer\n");
+		return -1;
+	}
+
+	ast_log(LOG_DEBUG,"play_and_prepend: %s, %s, '%s'\n", playfile ? playfile : "<None>", recordfile, fmt);
+	snprintf(comment,sizeof(comment),"Playing %s, Recording to: %s on %s\n", playfile ? playfile : "<None>", recordfile, chan->name);
+
+	if (playfile || beep) {	
+		if (!beep)
+			d = ast_play_and_wait(chan, playfile);
+		if (d > -1)
+			d = ast_streamfile(chan, "beep",chan->language);
+		if (!d)
+			d = ast_waitstream(chan,"");
+		if (d < 0)
+			return -1;
+	}
+	strncpy(prependfile, recordfile, sizeof(prependfile) -1);	
+	strncat(prependfile, "-prepend", sizeof(prependfile) - strlen(prependfile) - 1);
+			
+	fmts = ast_strdupa(fmt);
+	
+	stringp=fmts;
+	strsep(&stringp, "|");
+	ast_log(LOG_DEBUG,"Recording Formats: sfmts=%s\n", fmts);	
+	sfmt[0] = ast_strdupa(fmts);
+	
+	while((fmt = strsep(&stringp, "|"))) {
+		if (fmtcnt > MAX_OTHER_FORMATS - 1) {
+			ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app_voicemail.c\n");
+			break;
+		}
+		sfmt[fmtcnt++] = ast_strdupa(fmt);
+	}
+
+	time(&start);
+	end=start;  /* pre-initialize end to be same as start in case we never get into loop */
+	for (x=0;x<fmtcnt;x++) {
+		others[x] = ast_writefile(prependfile, sfmt[x], comment, O_TRUNC, 0, 0700);
+		ast_verbose( VERBOSE_PREFIX_3 "x=%i, open writing:  %s format: %s, %p\n", x, prependfile, sfmt[x], others[x]);
+		if (!others[x]) {
+			break;
+		}
+	}
+	
+	sildet = ast_dsp_new(); /* Create the silence detector */
+	if (!sildet) {
+		ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
+		return -1;
+	}
+	ast_dsp_set_threshold(sildet, silencethreshold);
+
+	if (maxsilence > 0) {
+		rfmt = chan->readformat;
+		res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
+		if (res < 0) {
+			ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
+			return -1;
+		}
+	}
+						
+	if (x == fmtcnt) {
+	/* Loop forever, writing the packets we read to the writer(s), until
+	   we read a # or get a hangup */
+		f = NULL;
+		for(;;) {
+		 	res = ast_waitfor(chan, 2000);
+			if (!res) {
+				ast_log(LOG_DEBUG, "One waitfor failed, trying another\n");
+				/* Try one more time in case of masq */
+			 	res = ast_waitfor(chan, 2000);
+				if (!res) {
+					ast_log(LOG_WARNING, "No audio available on %s??\n", chan->name);
+					res = -1;
+				}
+			}
+			
+			if (res < 0) {
+				f = NULL;
+				break;
+			}
+			f = ast_read(chan);
+			if (!f)
+				break;
+			if (f->frametype == AST_FRAME_VOICE) {
+				/* write each format */
+				for (x=0;x<fmtcnt;x++) {
+					if (!others[x])
+						break;
+					res = ast_writestream(others[x], f);
+				}
+				
+				/* Silence Detection */
+				if (maxsilence > 0) {
+					dspsilence = 0;
+					ast_dsp_silence(sildet, f, &dspsilence);
+					if (dspsilence)
+						totalsilence = dspsilence;
+					else
+						totalsilence = 0;
+					
+					if (totalsilence > maxsilence) {
+					/* Ended happily with silence */
+					if (option_verbose > 2) 
+						ast_verbose( VERBOSE_PREFIX_3 "Recording automatically stopped after a silence of %d seconds\n", totalsilence/1000);
+					ast_frfree(f);
+					gotsilence = 1;
+					outmsg=2;
+					break;
+					}
+				}
+				/* Exit on any error */
+				if (res) {
+					ast_log(LOG_WARNING, "Error writing frame\n");
+					ast_frfree(f);
+					break;
+				}
+			} else if (f->frametype == AST_FRAME_VIDEO) {
+				/* Write only once */
+				ast_writestream(others[0], f);
+			} else if (f->frametype == AST_FRAME_DTMF) {
+				/* stop recording with any digit */
+				if (option_verbose > 2) 
+					ast_verbose( VERBOSE_PREFIX_3 "User ended message by pressing %c\n", f->subclass);
+				res = 't';
+				outmsg = 2;
+				ast_frfree(f);
+				break;
+			}
+			if (maxtime) {
+				time(&end);
+				if (maxtime < (end - start)) {
+					if (option_verbose > 2)
+						ast_verbose( VERBOSE_PREFIX_3 "Took too long, cutting it short...\n");
+					res = 't';
+					outmsg=2;
+					ast_frfree(f);
+					break;
+				}
+			}
+			ast_frfree(f);
+		}
+		if (end == start) time(&end);
+		if (!f) {
+			if (option_verbose > 2) 
+				ast_verbose( VERBOSE_PREFIX_3 "User hung up\n");
+			res = -1;
+			outmsg=1;
+#if 0
+			/* delete all the prepend files */
+			for (x=0;x<fmtcnt;x++) {
+				if (!others[x])
+					break;
+				ast_closestream(others[x]);
+				ast_filedelete(prependfile, sfmt[x]);
+			}
+#endif
+		}
+	} else {
+		ast_log(LOG_WARNING, "Error creating writestream '%s', format '%s'\n", prependfile, sfmt[x]); 
+	}
+	*duration = end - start;
+#if 0
+	if (outmsg > 1) {
+#else
+	if (outmsg) {
+#endif
+		struct ast_frame *fr;
+		for (x=0;x<fmtcnt;x++) {
+			snprintf(comment, sizeof(comment), "Opening the real file %s.%s\n", recordfile, sfmt[x]);
+			realfiles[x] = ast_readfile(recordfile, sfmt[x], comment, O_RDONLY, 0, 0);
+			if (!others[x] || !realfiles[x])
+				break;
+			if (totalsilence)
+				ast_stream_rewind(others[x], totalsilence-200);
+			else
+				ast_stream_rewind(others[x], 200);
+			ast_truncstream(others[x]);
+			/* add the original file too */
+			while ((fr = ast_readframe(realfiles[x]))) {
+				ast_writestream(others[x],fr);
+			}
+			ast_closestream(others[x]);
+			ast_closestream(realfiles[x]);
+			ast_filerename(prependfile, recordfile, sfmt[x]);
+#if 0
+			ast_verbose("Recording Format: sfmts=%s, prependfile %s, recordfile %s\n", sfmt[x],prependfile,recordfile);
+#endif
+			ast_filedelete(prependfile, sfmt[x]);
+		}
+	}
+	if (rfmt) {
+		if (ast_set_read_format(chan, rfmt)) {
+			ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_getformatname(rfmt), chan->name);
+		}
+	}
+	if (outmsg) {
+		if (outmsg > 1) {
+			/* Let them know it worked */
+			ast_streamfile(chan, "auth-thankyou", chan->language);
+			ast_waitstream(chan, "");
+		}
+	}	
+	return res;
+}
+
+int ast_lock_path(const char *path)
+{
+	char *s;
+	char *fs;
+	int res;
+	int fd;
+	time_t start;
+	s = alloca(strlen(path) + 10);
+	fs = alloca(strlen(path) + 20);
+	if (!fs || !s) {
+		ast_log(LOG_WARNING, "Out of memory!\n");
+		return -1;
+	}
+	snprintf(fs, strlen(path) + 19, "%s/%s-%08x", path, ".lock", rand());
+	fd = open(fs, O_WRONLY | O_CREAT | O_EXCL, 0600);
+	if (fd < 0) {
+		fprintf(stderr, "Unable to create lock file: %s\n", strerror(errno));
+		return -1;
+	}
+	close(fd);
+	snprintf(s, strlen(path) + 9, "%s/%s", path, ".lock");
+	time(&start);
+	while (((res = link(fs, s)) < 0) && (errno == EEXIST) && (time(NULL) - start < 5))
+		usleep(1);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to lock path '%s': %s\n", path, strerror(errno));
+	}
+	unlink(fs);
+	ast_log(LOG_DEBUG, "Locked path '%s'\n", path);
+	return res;
+}
+
+int ast_unlock_path(const char *path)
+{
+	char *s;
+	s = alloca(strlen(path) + 10);
+	if (!s)
+		return -1;
+	snprintf(s, strlen(path) + 9, "%s/%s", path, ".lock");
+	ast_log(LOG_DEBUG, "Unlocked path '%s'\n", path);
+	return unlink(s);
+}
+

apps/Makefile

@@ -11,52 +11,94 @@
 # the GNU General Public License
 #
 
+USE_MYSQL_VM_INTERFACE=0
+USE_POSTGRES_VM_INTERFACE=0
+
 #APPS=app_dial.so app_playback.so app_directory.so app_intercom.so app_mp3.so 
-APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_intercom.so app_mp3.so \
+APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
      app_system.so app_echo.so app_record.so app_image.so app_url.so app_disa.so \
-     app_agi.so app_qcall.so app_adsiprog.so app_getcpeid.so app_milliwatt.so \
-     app_zapateller.so app_datetime.so app_setcallerid.so app_festival.so \
+     app_qcall.so app_adsiprog.so app_getcpeid.so app_milliwatt.so \
+     app_zapateller.so app_setcallerid.so app_festival.so \
      app_queue.so app_senddtmf.so app_parkandannounce.so app_striplsd.so \
      app_setcidname.so app_lookupcidname.so app_substring.so app_macro.so \
      app_authenticate.so app_softhangup.so app_lookupblacklist.so \
-     app_waitforring.so app_privacy.so app_db.so app_chanisavail.so
+     app_waitforring.so app_privacy.so app_db.so app_chanisavail.so \
+     app_enumlookup.so app_transfer.so app_setcidnum.so app_cdr.so \
+     app_hasnewvoicemail.so app_sayunixtime.so app_cut.so app_read.so \
+     app_setcdruserfield.so app_random.so app_ices.so app_eval.so \
+     app_nbscat.so app_sendtext.so app_exec.so app_sms.so \
+     app_groupcount.so app_txtcidname.so app_controlplayback.so \
+     app_talkdetect.so app_alarmreceiver.so app_userevent.so app_verbose.so \
+     app_test.so app_forkcdr.so
+
+ifneq (${OSARCH},Darwin)
+APPS+=app_intercom.so
+endif
 
 #APPS+=app_sql_postgres.so
 #APPS+=app_sql_odbc.so
+#APPS+=app_rpt.so
 
-APPS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo "app_zapras.so app_meetme.so app_flash.so app_zapbarge.so" ; fi)
-#APPS+=$(shell if [ -f /usr/include/zap.h ]; then echo "app_rpt.so" ; fi)
+APPS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo "app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so" ; fi)
+APPS+=$(shell if [ -f /usr/local/include/zaptel.h ]; then echo "app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so" ; fi)
+APPS+=$(shell if [ -f /usr/include/osp/osp.h ]; then echo "app_osplookup.so" ; fi)
 
-CFLAGS+=
+CFLAGS+=-fPIC
+
+ifeq ($(USE_POSTGRES_VM_INTERFACE),1)
+CFLAGS+=-DUSEPOSTGRESVM
+endif
+
+ifeq ($(USE_MYSQL_VM_INTERFACE),1)
+CFLAGS+=-DUSEMYSQLVM
+endif
 
 all: $(APPS)
 
 clean:
-	rm -f *.so *.o look
+	rm -f *.so *.o look .depend
 
 %.so : %.o
-	$(CC) -shared -Xlinker -x -o $@ $<
+	$(CC) $(SOLINK) -o $@ $<
 
 app_rpt.so : app_rpt.o
-	$(CC) -shared -Xlinker -x -o $@ $< -ltonezone
+	$(CC) $(SOLINK) -o $@ $< -ltonezone
 
 install: all
-	for x in $(APPS); do $(INSTALL) -m 755 $$x $(MODULES_DIR) ; done
+	for x in $(APPS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done
+	rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
 
-app_todd.o: app_todd.c
-	gcc -pipe -O6 -g  -Iinclude -I../include -D_REENTRANT -march=i586 -DDO_CRASH -c -o  app_todd.o app_todd.c
-
-app_todd.so: app_todd.o
-	$(CC) -shared -Xlinker -x -o $@ $< -L/usr/local/ssl/lib -lssl -lcrypto
+app_voicemail.so : app_voicemail.o
+ifeq ($(USE_MYSQL_VM_INTERFACE),1)
+	$(CC) $(SOLINK) -o $@ $(MLFLAGS) $< -L/usr/lib/mysql -lmysqlclient -lz
+else
+ifeq ($(USE_POSTGRES_VM_INTERFACE),1)
+	$(CC) $(SOLINK) -o $@ $(MLFLAGS) $< -lpq
+else
+	$(CC) $(SOLINK) -o $@ $(MLFLAGS) $< 
+endif
+endif
 
 app_sql_postgres.o: app_sql_postgres.c
-	$(CC) -pipe -I/usr/local/pgsql/include -Wall -Wmissing-prototypes -Wmissing-declarations -O6 -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -march=i686  -DASTERISK_VERSION=\"CVS-07/21/02-14:49:14\" -DDO_CRASH -DDEBUG_THREADS     -c -o app_sql_postgres.o app_sql_postgres.c
+	$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres.o app_sql_postgres.c
 
 app_sql_postgres.so: app_sql_postgres.o
-	$(CC) -shared -Xlinker -x -o $@ $< -L/usr/local/pgsql/lib -lpq
+	$(CC) $(SOLINK) -o $@ $< -L/usr/local/pgsql/lib -lpq
 
 app_sql_odbc.so: app_sql_odbc.o
-	$(CC) -shared -Xlinker -x -o $@ $< -lodbc
+	$(CC) $(SOLINK) -o $@ $< -lodbc
 
 look:	look.c
-	gcc -pipe -O6 -g look.c -o look -lncurses
+	$(CC) -pipe -O6 -g look.c -o look -lncurses
+
+ifneq ($(wildcard .depend),)
+include .depend
+endif
+
+depend: .depend
+
+.depend:
+	../mkdep $(CFLAGS) `ls *.c`
+
+env:
+	env

apps/app_adsiprog.c

@@ -11,6 +11,7 @@
  * the GNU General Public License
  */
 
+#include <sys/types.h>
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
@@ -18,6 +19,8 @@
 #include <asterisk/module.h>
 #include <asterisk/adsi.h>
 #include <asterisk/options.h>
+#include <asterisk/utils.h>
+#include <asterisk/lock.h>
 #include <netinet/in.h>
 #include <stdlib.h>
 #include <unistd.h>
@@ -27,8 +30,6 @@
 #include <stdio.h>
 #include <errno.h>
 
-#include <pthread.h>
-
 #include "../asterisk.h"
 #include "../astconf.h"
 
@@ -177,7 +178,7 @@ static int process_token(void *out, char *src, int maxlen, int argtype)
 			maxlen = strlen(src) - 1;
 		memcpy(out, src, maxlen);
 		((char *)out)[maxlen] = '\0';
-	} else if (strlen(src) && (src[0] == '\\')) {
+	} else if (!ast_strlen_zero(src) && (src[0] == '\\')) {
 		if (!(argtype & ARG_NUMBER))
 			return -1;
 		/* Octal value */
@@ -197,7 +198,7 @@ static int process_token(void *out, char *src, int maxlen, int argtype)
 			/* Convert */
 			*((unsigned int *)out) = htonl(*((unsigned int *)out));
 		}
-	} else if ((strlen(src) && isdigit(src[0]))) {
+	} else if ((!ast_strlen_zero(src) && isdigit(src[0]))) {
 		if (!(argtype & ARG_NUMBER))
 			return -1;
 		/* Hex value */
@@ -665,7 +666,7 @@ static int showdisplay(char *buf, char *name, int id, char *args, struct adsi_sc
 	}
 				   
 	buf[0] = id;
-	buf[1] = (cmd << 6) | (disp->id & 0x2f); 
+	buf[1] = (cmd << 6) | (disp->id & 0x3f); 
 	buf[2] = ((line & 0x1f) << 3) | (flag & 0x7);
 	return 3;
 }
@@ -694,6 +695,18 @@ static int digitdirect(char *buf, char *name, int id, char *args, struct adsi_sc
 	return 2;
 }
 
+static int clearcbone(char *buf, char *name, int id, char *args, struct adsi_script *istate, char *script, int lineno)
+{
+	char *tok;
+	tok = get_token(&args, script, lineno);
+	if (tok)
+		ast_log(LOG_WARNING, "CLEARCB1 requires no arguments ('%s') at line %d of %s\n", tok, lineno, script);
+
+	buf[0] = id;
+	buf[1] = 0;
+	return 2;
+}
+
 static int digitcollect(char *buf, char *name, int id, char *args, struct adsi_script *istate, char *script, int lineno)
 {
 	char *tok;
@@ -830,6 +843,7 @@ static struct adsi_key_cmd kcmds[] = {
 	{ "VOICEMODE", 0x93 },
 	/* Display call buffer 'n' */
 	/* Clear call buffer 'n' */
+	{ "CLEARCB1", 0x95, clearcbone },
 	{ "DIGITCOLLECT", 0x96, digitcollect },
 	{ "DIGITDIRECT", 0x96, digitdirect },
 	{ "CLEAR", 0x97 },
@@ -987,7 +1001,7 @@ static int adsi_process(struct adsi_script *state, char *buf, char *script, int
 				break;
 			}
 			if (process_token(vname, args, sizeof(vname) - 1, ARG_STRING)) {
-				ast_log(LOG_WARNING "'%s' is not a valid token for a KEY name at line %d of %s\n", args, lineno, script);
+				ast_log(LOG_WARNING, "'%s' is not a valid token for a KEY name at line %d of %s\n", args, lineno, script);
 				break;
 			}
 			state->key = getkeybyname(state, vname, script, lineno);
@@ -1010,7 +1024,7 @@ static int adsi_process(struct adsi_script *state, char *buf, char *script, int
 				break;
 			}
 			if (process_token(tmp, args, sizeof(tmp) - 1, ARG_STRING)) {
-				ast_log(LOG_WARNING "'%s' is not a valid token for a KEY short name at line %d of %s\n", args, lineno, script);
+				ast_log(LOG_WARNING, "'%s' is not a valid token for a KEY short name at line %d of %s\n", args, lineno, script);
 				break;
 			}
 			args = get_token(&buf, script, lineno);
@@ -1025,7 +1039,7 @@ static int adsi_process(struct adsi_script *state, char *buf, char *script, int
 					break;
 				}
 				if (process_token(tmp2, args, sizeof(tmp2) - 1, ARG_STRING)) {
-					ast_log(LOG_WARNING "'%s' is not a valid token for a KEY long name at line %d of %s\n", args, lineno, script);
+					ast_log(LOG_WARNING, "'%s' is not a valid token for a KEY long name at line %d of %s\n", args, lineno, script);
 					break;
 				}
 			} else {
@@ -1065,7 +1079,7 @@ static int adsi_process(struct adsi_script *state, char *buf, char *script, int
 				break;
 			}
 			if (process_token(vname, args, sizeof(vname) - 1, ARG_STRING)) {
-				ast_log(LOG_WARNING "'%s' is not a valid token for a KEY name at line %d of %s\n", args, lineno, script);
+				ast_log(LOG_WARNING, "'%s' is not a valid token for a KEY name at line %d of %s\n", args, lineno, script);
 				break;
 			}
 			state->sub = getsubbyname(state, vname, script, lineno);
@@ -1103,7 +1117,7 @@ static int adsi_process(struct adsi_script *state, char *buf, char *script, int
 				break;
 			}
 			if (process_token(vname, args, sizeof(vname) - 1, ARG_STRING)) {
-				ast_log(LOG_WARNING "'%s' is not a valid token for a STATE name at line %d of %s\n", args, lineno, script);
+				ast_log(LOG_WARNING, "'%s' is not a valid token for a STATE name at line %d of %s\n", args, lineno, script);
 				break;
 			}
 			if (getstatebyname(state, vname, script, lineno, 0)) {
@@ -1118,7 +1132,7 @@ static int adsi_process(struct adsi_script *state, char *buf, char *script, int
 				break;
 			}
 			if (process_token(vname, args, sizeof(vname) - 1, ARG_STRING)) {
-				ast_log(LOG_WARNING "'%s' is not a valid token for a FLAG name at line %d of %s\n", args, lineno, script);
+				ast_log(LOG_WARNING, "'%s' is not a valid token for a FLAG name at line %d of %s\n", args, lineno, script);
 				break;
 			}
 			if (getflagbyname(state, vname, script, lineno, 0)) {
@@ -1135,7 +1149,7 @@ static int adsi_process(struct adsi_script *state, char *buf, char *script, int
 				break;
 			}
 			if (process_token(vname, args, sizeof(vname) - 1, ARG_STRING)) {
-				ast_log(LOG_WARNING "'%s' is not a valid token for a KEY name at line %d of %s\n", args, lineno, script);
+				ast_log(LOG_WARNING, "'%s' is not a valid token for a KEY name at line %d of %s\n", args, lineno, script);
 				break;
 			}
 			if (getdisplaybyname(state, vname, script, lineno, 0)) {
@@ -1349,7 +1363,7 @@ static struct adsi_script *compile_script(char *script)
 			/* Strip comments */
 			if (c)
 				*c = '\0';
-			if (strlen(buf))
+			if (!ast_strlen_zero(buf))
 				adsi_process(scr, buf, script, lineno);
 		}
 	}
@@ -1434,7 +1448,7 @@ static int adsi_prog(struct ast_channel *chan, char *script)
 	bytes = 0;
 	/* Start with key definitions */
 	for (x=0;x<scr->numkeys;x++) {
-		if (bytes + scr->keys[x].retstrlen > 254) {
+		if (bytes + scr->keys[x].retstrlen > 253) {
 			/* Send what we've collected so far */
 			if (adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DOWNLOAD)) {
 				ast_log(LOG_WARNING, "Unable to send chunk ending at %d\n", x);
@@ -1458,7 +1472,7 @@ static int adsi_prog(struct ast_channel *chan, char *script)
 	bytes = 0;
 	/* Continue with the display messages */
 	for (x=0;x<scr->numdisplays;x++) {
-		if (bytes + scr->displays[x].datalen > 254) {
+		if (bytes + scr->displays[x].datalen > 253) {
 			/* Send what we've collected so far */
 			if (adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DOWNLOAD)) {
 				ast_log(LOG_WARNING, "Unable to send chunk ending at %d\n", x);
@@ -1482,7 +1496,7 @@ static int adsi_prog(struct ast_channel *chan, char *script)
 	bytes = 0;
 	/* Send subroutines */
 	for (x=0;x<scr->numsubs;x++) {
-		if (bytes + scr->subs[x].datalen > 254) {
+		if (bytes + scr->subs[x].datalen > 253) {
 			/* Send what we've collected so far */
 			if (adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DOWNLOAD)) {
 				ast_log(LOG_WARNING, "Unable to send chunk ending at %d\n", x);
@@ -1526,7 +1540,7 @@ static int adsi_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
 	struct localuser *u;
-	if (!data || !strlen(data))
+	if (!data || ast_strlen_zero(data))
 		data = "asterisk.adsi";
 	LOCAL_USER_ADD(u);
 	if (!adsi_available(chan)) {

apps/app_alarmreceiver.c

@@ -0,0 +1,858 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Central Station Alarm receiver for Ademco Contact ID  
+ * 
+ * Copyright (C) 2004 Steve Rodgers
+ *
+ * Steve Rodgers <hwstar@rodgers.sdcoxmail.com>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ *
+ * *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** 
+ *
+ * Use at your own risk. Please consult the GNU GPL license document included with Asterisk details. *
+ *
+ * *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING ***
+ *
+ */ 
+ 
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/ulaw.h>
+#include <asterisk/options.h>
+#include <asterisk/app.h>
+#include <asterisk/dsp.h>
+#include <asterisk/config.h>
+#include <asterisk/localtime.h>
+#include <asterisk/callerid.h>
+#include <asterisk/astdb.h>
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <math.h>
+#include <sys/wait.h>
+#include <unistd.h>
+#include <sys/time.h>
+
+#define ALMRCV_CONFIG "alarmreceiver.conf"
+#define ADEMCO_CONTACT_ID "ADEMCO_CONTACT_ID"
+
+struct event_node{
+	char data[17];
+	struct event_node *next;
+};
+
+typedef struct event_node event_node_t;
+
+static char *tdesc = "Alarm Receiver for Asterisk";
+
+static char *app = "AlarmReceiver";
+
+static char *synopsis = "Provide support for receving alarm reports from a burglar or fire alarm panel";
+static char *descrip =
+"Alarm receiver application for Asterisk. Only 1 signalling format is supported at this time:\n"
+"Ademco Contact ID. This application should be called whenever there is an alarm panel calling in\n"
+"to dump its events. The application will handshake with the alarm panel, and receive events,\n"
+"validate them, handshake them, and store them until the panel hangs up. Once the panel hangs up,\n"
+"the application will run the command line specified by the eventcmd setting in alarmreceiver.conf\n"
+"and pipe the events to the standard input of the application. Alarmreceiver.conf also contains settings\n"
+"for DTMF timing, and for the loudness of the acknowledgement tones.\n";
+
+/* Config Variables */
+
+static int fdtimeout = 2000;
+static int sdtimeout = 200; 
+static int toneloudness = 4096;
+static int log_individual_events = 0;
+static char event_spool_dir[128] = {'\0'};
+static char event_app[128] = {'\0'};
+static char db_family[128] = {'\0'};
+static char time_stamp_format[128] = {"%a %b %d, %Y @ %H:%M:%S %Z"};
+
+
+/* Misc variables */
+
+	
+static char event_file[14] = "/event-XXXXXX";
+
+
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+/*
+* Attempt to access a database variable and increment it,
+* provided that the user defined db-family in alarmreceiver.conf
+* The alarmreceiver app will write statistics to a few variables
+* in this family if it is defined. If the new key doesn't exist in the
+* family, then create it and set its value to 1.
+*/
+
+static void database_increment( char *key )
+{
+	int res = 0;
+	unsigned v;
+	char value[16];
+	
+	
+	if(!strlen(db_family))
+		return; /* If not defined, don't do anything */
+	
+	res = ast_db_get(db_family, key, value, sizeof(value) - 1);
+	
+	if(res){
+		if(option_verbose >= 4)
+			ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Creating database entry %s and setting to 1\n", key);
+		/* Guess we have to create it */
+		res = ast_db_put(db_family, key, "1");
+		return;
+	}
+	
+	sscanf(value, "%u", &v);
+	v++;
+	
+	if(option_verbose >= 4)
+		ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: New value for %s: %u\n", key, v);
+		
+	snprintf(value, sizeof(value), "%u", v);
+	
+	res = ast_db_put(db_family, key, value);
+	
+	if((res)&&(option_verbose >= 4))
+		ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: database_increment write error");
+	
+	return;	
+}
+
+
+/*
+* Build a MuLaw data block for a single frequency tone
+*/
+
+static void make_tone_burst(unsigned char *data, float freq, float loudness, int len, int *x)
+{
+	int     i;
+	float   val;
+                                                                                                                                    
+        for(i = 0; i < len; i++){
+		val = loudness * sin((freq * 2.0 * M_PI * (*x)++)/8000.0);
+		data[i] = AST_LIN2MU((int)val);
+	}
+
+	/* wrap back around from 8000 */
+
+	if (*x >= 8000) *x = 0;
+	return;
+}
+
+/*
+* Send a single tone burst for a specifed duration and frequency. 
+* Returns 0 if successful
+*/
+
+static int send_tone_burst(struct ast_channel *chan, float freq, int duration, int tldn)
+{
+	int res = 0;
+	int i = 0;
+	int x = 0;
+	struct ast_frame *f, wf;
+	
+	struct {
+		unsigned char offset[AST_FRIENDLY_OFFSET];
+		unsigned char buf[640];
+	} tone_block;
+
+	for(;;)
+	{
+	
+		if (ast_waitfor(chan, -1) < 0){
+			res = -1;
+			break;
+		}
+		
+		f = ast_read(chan);
+		if (!f){
+			res = -1;
+			break;
+		}
+		
+		if (f->frametype == AST_FRAME_VOICE) {
+			wf.frametype = AST_FRAME_VOICE;
+			wf.subclass = AST_FORMAT_ULAW;
+			wf.offset = AST_FRIENDLY_OFFSET;
+			wf.mallocd = 0;
+			wf.data = tone_block.buf;
+			wf.datalen = f->datalen;
+			wf.samples = wf.datalen;
+			
+			make_tone_burst(tone_block.buf, freq, (float) tldn, wf.datalen, &x);
+
+			i += wf.datalen / 8;
+			if (i > duration) {
+				break;
+			}
+			if (ast_write(chan, &wf)){
+				if(option_verbose >= 4)
+					ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Failed to write frame on %s\n", chan->name);
+				ast_log(LOG_WARNING, "AlarmReceiver Failed to write frame on %s\n",chan->name);
+				res = -1;
+				break;
+			}
+		}
+		
+		ast_frfree(f);
+	}
+	return res;
+}
+
+/*
+* Return the difference in milliseconds between two timeval structs
+*/
+
+static int ms_diff(struct timeval *tv1, struct timeval *tv2){
+
+	int	ms;
+	
+	ms = (tv1->tv_sec - tv2->tv_sec) * 1000;
+	ms += (tv1->tv_usec - tv2->tv_usec) / 1000;
+	
+	return(ms);
+}
+
+/*
+* Receive a string of DTMF digits where the length of the digit string is known in advance. Do not give preferential
+* treatment to any digit value, and allow separate time out values to be specified for the first digit and all subsequent
+* digits.
+*
+* Returns 0 if all digits successfully received.
+* Returns 1 if a digit time out occurred
+* Returns -1 if the caller hung up or there was a channel error.
+*
+*/
+
+static int receive_dtmf_digits(struct ast_channel *chan, char *digit_string, int length, int fdto, int sdto)
+{
+	int res = 0;
+	int i = 0;
+	int r;
+	struct ast_frame *f;
+	struct timeval now, lastdigittime;
+	
+	gettimeofday(&lastdigittime,NULL);
+	for(;;){
+		gettimeofday(&now,NULL);
+		
+		  /* if outa time, leave */
+		if (ms_diff(&now,&lastdigittime) > 
+		    ((i > 0) ? sdto : fdto)){
+			if(option_verbose >= 4)
+				ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: DTMF Digit Timeout on %s\n", chan->name);
+				
+			ast_log(LOG_DEBUG,"AlarmReceiver: DTMF timeout on chan %s\n",chan->name);
+				
+			res = 1;
+			break;
+		}
+		
+		if ((r = ast_waitfor(chan, -1) < 0)) {
+			ast_log(LOG_DEBUG, "Waitfor returned %d\n", r);
+			continue;
+		}
+			
+		f = ast_read(chan);
+		
+		if (f == NULL){
+			res = -1;
+			break;
+		}
+		
+		/* If they hung up, leave */
+		if ((f->frametype == AST_FRAME_CONTROL) &&
+		    (f->subclass == AST_CONTROL_HANGUP)){
+			ast_frfree(f);
+			res = -1;
+			break;
+		}
+		
+		/* if not DTMF, just do it again */
+		if (f->frametype != AST_FRAME_DTMF){
+			ast_frfree(f);
+			continue;
+		}
+
+		digit_string[i++] = f->subclass;  /* save digit */
+		
+		ast_frfree(f);
+		
+		/* If we have all the digits we expect, leave */
+		if(i >= length)
+			break;
+		
+		gettimeofday(&lastdigittime,NULL);
+	}
+	
+	digit_string[i] = '\0'; /* Nul terminate the end of the digit string */
+	return res;
+
+}
+
+/*
+* Write the metadata to the log file
+*/
+
+static int write_metadata( FILE *logfile, char *signalling_type, struct ast_channel *chan)
+{
+	int res = 0;
+	time_t t;
+	struct tm now;
+	char *cl,*cn;
+	char workstring[80];
+	char timestamp[80];
+	
+	/* Extract the caller ID location */
+	
+	strncpy(workstring, chan->callerid, sizeof(workstring) - 1);
+	workstring[sizeof(workstring) - 1] = '\0';
+	
+	ast_callerid_parse(workstring, &cn, &cl);
+	if (cl) 
+		ast_shrink_phone_number(cl);
+                
+
+	/* Get the current time */
+		
+	time(&t);
+	ast_localtime(&t, &now, NULL);
+	
+	/* Format the time */
+	
+	strftime(timestamp, sizeof(timestamp), time_stamp_format, &now); 
+
+	
+	res = fprintf(logfile, "\n\n[metadata]\n\n");
+	
+	if(res >= 0)
+		res = fprintf(logfile, "PROTOCOL=%s\n", signalling_type);
+		
+	if(res >= 0)	
+		res = fprintf(logfile, "CALLINGFROM=%s\n", (!cl) ? "<unknown>" : cl);
+		
+	if(res >- 0)
+		res = fprintf(logfile, "CALLERNAME=%s\n", (!cn) ? "<unknown>" : cn);
+		
+	if(res >= 0)
+		res = fprintf(logfile, "TIMESTAMP=%s\n\n", timestamp);
+	
+	if(res >= 0)
+		res = fprintf(logfile, "[events]\n\n");
+	
+	if(res < 0){
+		ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: can't write metadata\n");	
+		
+		ast_log(LOG_DEBUG,"AlarmReceiver: can't write metadata\n");
+	}
+	else
+		res = 0;
+
+	return res;
+}
+
+/*
+* Write a single event to the log file
+*/
+
+static int write_event( FILE *logfile,  event_node_t *event)
+{
+	int res = 0;
+
+	if( fprintf(logfile, "%s\n", event->data) < 0)
+		res = -1;
+			
+	return res;
+}
+
+
+/*
+* If we are configured to log events, do so here.
+*
+*/
+
+static int log_events(struct ast_channel *chan,  char *signalling_type, event_node_t *event)
+{
+
+	int res = 0;
+	char workstring[sizeof(event_spool_dir)+sizeof(event_file)] = "";
+	int fd;
+	FILE *logfile;
+	event_node_t *elp = event;
+	
+	if(strlen(event_spool_dir)){
+		
+		/* Make a template */
+		
+		strncpy(workstring, event_spool_dir, sizeof(workstring) - 1);
+		strncat(workstring, event_file, sizeof(workstring) - strlen(workstring) - 1);
+		
+		/* Make the temporary file */
+		
+		fd = mkstemp(workstring);
+		
+		if(fd == -1){
+			ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: can't make temporary file\n");	
+			ast_log(LOG_DEBUG,"AlarmReceiver: can't make temporary file\n");
+			res = -1;
+		}
+		
+		if(!res){
+			logfile = fdopen(fd, "w");
+			if(logfile){
+				/* Write the file */
+				res = write_metadata(logfile, signalling_type, chan);
+				if(!res)
+					while((!res) && (elp != NULL)){
+						res = write_event(logfile, elp);
+						elp = elp->next;
+					}
+				if(!res){
+					if(fflush(logfile) == EOF)
+						res = -1;
+					if(!res){
+						if(fclose(logfile) == EOF)
+							res = -1;
+					}				
+				}
+			}
+			else
+				res = -1;
+		}
+	}
+
+	return res;	
+}
+
+/*
+* This function implements the logic to receive the Ademco contact ID  format.
+*
+* The function will return 0 when the caller hangs up, else a -1 if there was a problem.
+*/
+
+static int receive_ademco_contact_id( struct ast_channel *chan, void *data, int fdto, int sdto, int tldn, event_node_t **ehead)
+{
+	int i,j;
+	int res = 0;
+	int checksum;
+	char event[17];
+	event_node_t *enew, *elp;
+	int got_some_digits = 0;
+	int events_received = 0;
+	int ack_retries = 0;
+	
+	static char digit_map[15] = "0123456789*#ABC";
+        static unsigned char digit_weights[15] = {10,1,2,3,4,5,6,7,8,9,11,12,13,14,15};
+                                                                                                                      
+	database_increment("calls-received");
+
+	/* Wait for first event */
+
+	if(option_verbose >= 4)
+		ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Waiting for first event from panel\n");
+
+	while(res >= 0){
+
+		if(got_some_digits == 0){
+
+	        	/* Send ACK tone sequence */
+                        
+		                                                                                                                    
+        		if(option_verbose >= 4)
+                		ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Sending 1400Hz 100ms burst (ACK)\n");
+                                                                                                                                            
+                                                                                                                                            
+        		res = send_tone_burst(chan, 1400.0, 100, tldn);
+                                                                                                                                            
+        		if(!res)
+                		res = ast_safe_sleep(chan, 100);
+                                                                                                                                            
+        		if(!res){
+                		if(option_verbose >= 4)
+                        		ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Sending 2300Hz 100ms burst (ACK)\n");
+                                                                                                                                            
+                		res = send_tone_burst(chan, 2300.0, 100, tldn);
+        		}
+                                                                                                                                            
+		}
+
+		if( res >= 0)
+			res = receive_dtmf_digits(chan, event, sizeof(event) - 1, fdto, sdto);
+		
+		if (res < 0){
+		
+			if(events_received == 0)
+				/* Hangup with no events received should be logged in the DB */
+				database_increment("no-events-received");
+			else{
+				if(ack_retries){
+					if(option_verbose >= 4)
+						ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: ACK retries during this call: %d\n", ack_retries);
+					
+					database_increment("ack-retries");
+				}
+			}
+			if(option_verbose >= 4)
+				ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: App exiting...\n");
+			res = -1;
+			break;
+		}
+		
+		if(res != 0){
+			 /* Didn't get all of the digits */
+			if(option_verbose >= 2)
+				ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Incomplete string: %s, trying again...\n", event);
+
+			if(!got_some_digits){
+				got_some_digits = (strlen(event)) ? 1 : 0;
+				ack_retries++;
+			}
+			continue;	
+		}		
+		
+		got_some_digits = 1;
+
+		if(option_verbose >= 2)
+			ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Received Event %s\n", event);
+		ast_log(LOG_DEBUG, "AlarmReceiver: Received event: %s\n", event);
+		
+		/* Calculate checksum */
+		
+		for(j = 0, checksum = 0; j < 16; j++){
+			for(i = 0 ; i < sizeof(digit_map) ; i++){
+				if(digit_map[i] == event[j])
+					break;
+			}
+			
+			if(i == 16)
+				break;
+				
+			checksum += digit_weights[i];
+		}
+		
+		if(i == 16){
+			if(option_verbose >= 2)
+				ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Bad DTMF character %c, trying again\n", event[j]);
+			continue; /* Bad character */
+		}
+
+		/* Checksum is mod(15) of the total */
+
+		checksum = checksum % 15;
+
+		if(checksum){
+			database_increment("checksum-errors");
+			if(option_verbose >= 2){
+				ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Nonzero checksum\n");
+			ast_log(LOG_DEBUG, "AlarmReceiver: Nonzero checksum\n");
+			continue;
+			}
+		}
+
+		/* Check the message type for correctness */
+
+		if(strncmp(event + 4, "18", 2)){
+			if(strncmp(event + 4, "98", 2)){
+				database_increment("format-errors");
+				if(option_verbose >= 2)
+					ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Wrong message type\n");
+				ast_log(LOG_DEBUG, "AlarmReceiver: Wrong message type\n");
+			continue;
+			}
+		}
+
+		events_received++;
+		
+		/* Queue the Event */
+
+		if((enew = malloc(sizeof(event_node_t))) == NULL){
+			if(option_verbose >= 1)
+				ast_verbose(VERBOSE_PREFIX_1 "AlarmReceiver: Failed to allocate memory\n");
+			ast_log(LOG_WARNING, "AlarmReceiver Failed to allocate memory\n");
+			res = -1;
+                        break;
+		}
+
+		memset(enew, 0, sizeof(event_node_t));
+		
+		enew->next = NULL;
+		strncpy(enew->data, event, sizeof(enew->data) - 1);
+
+		/*
+		* Insert event onto end of list
+		*/
+		
+		if(*ehead == NULL){
+			*ehead = enew;
+		}
+		else{
+			for(elp = *ehead; elp->next != NULL; elp = elp->next)
+			;
+			
+			elp->next = enew;
+		}
+		
+		if(res > 0)
+			res = 0;
+		
+		/* Let the user have the option of logging the single event before sending the kissoff tone */
+
+		if((res == 0) && (log_individual_events))
+			res = log_events(chan, ADEMCO_CONTACT_ID, enew);
+	
+		/* Wait 200 msec before sending kissoff */	
+			
+		if(res == 0)	
+			res = ast_safe_sleep(chan, 200);
+
+		/* Send the kissoff tone */
+
+		if(res == 0)		
+			res = send_tone_burst(chan, 1400.0, 900, tldn);
+	}
+
+	
+	return res;
+}
+
+
+/*
+* This is the main function called by Asterisk Core whenever the App is invoked in the extension logic.
+* This function will always return 0.
+*/
+
+static int alarmreceiver_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u;
+	event_node_t *elp, *efree;
+	char signalling_type[64] = "";
+
+	event_node_t *event_head = NULL;
+
+	LOCAL_USER_ADD(u);
+
+	/* Set write and read formats to ULAW */
+
+	if(option_verbose >= 4)
+		ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Setting read and write formats to ULAW\n");
+
+	if (ast_set_write_format(chan,AST_FORMAT_ULAW)){
+		ast_log(LOG_WARNING, "AlarmReceiver: Unable to set write format to Mu-law on %s\n",chan->name);
+		return -1;
+	}
+	
+	if (ast_set_read_format(chan,AST_FORMAT_ULAW)){
+		ast_log(LOG_WARNING, "AlarmReceiver: Unable to set read format to Mu-law on %s\n",chan->name);
+		return -1;
+	}
+
+	/* Set default values for this invokation of the application */
+	
+	strncpy(signalling_type, ADEMCO_CONTACT_ID, sizeof(signalling_type) - 1);
+
+
+	/* Answer the channel if it is not already */
+
+	if(option_verbose >= 4)
+		ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Answering channel\n");
+
+	if (chan->_state != AST_STATE_UP) {
+	
+		res = ast_answer(chan);
+		
+		if (res) {
+			LOCAL_USER_REMOVE(u);
+			return -1;
+		}
+	}
+
+	/* Wait for the connection to settle post-answer */
+
+	if(option_verbose >= 4)
+		ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Waiting for connection to stabilize\n");
+
+	res = ast_safe_sleep(chan, 1250);
+
+	/* Attempt to receive the events */
+
+	if(!res){
+	
+		/* Determine the protocol to receive in advance */
+		/* Note: Ademco contact is the only one supported at this time */
+		/* Others may be added later */
+		
+		if(!strcmp(signalling_type, ADEMCO_CONTACT_ID))
+			receive_ademco_contact_id(chan, data, fdtimeout, sdtimeout, toneloudness, &event_head);
+		else
+			res = -1;
+	}
+	
+		
+	
+	/* Events queued by receiver, write them all out here if so configured */
+
+	if((!res) && (log_individual_events == 0)){
+		res = log_events(chan, signalling_type, event_head);
+
+	}
+
+	/*
+	* Do we exec a command line at the end?
+	*/
+	
+	if((!res) && (strlen(event_app)) && (event_head)){
+		ast_log(LOG_DEBUG,"Alarmreceiver: executing: %s\n", event_app);
+		ast_safe_system(event_app);
+	}
+
+	/*
+	* Free up the data allocated in our linked list
+	*/
+		
+	for(elp = event_head; (elp != NULL);){
+		efree = elp;
+		elp = elp->next;
+		free(efree);
+	}
+
+
+	LOCAL_USER_REMOVE(u);
+
+	return 0;
+}
+
+/* 
+* Load the configuration from the configuration file
+*/
+
+static int load_config(void)
+{
+	struct ast_config *cfg;
+	char *p;
+
+	/* Read in the config file */
+
+	cfg = ast_load(ALMRCV_CONFIG);
+                                                                                                                                  
+	if(!cfg){
+	
+		if(option_verbose >= 4)
+			ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: No config file\n");
+	}
+	else{
+
+		
+		p = ast_variable_retrieve(cfg, "general", "eventcmd");
+		
+		if(p){
+			strncpy(event_app, p, sizeof(event_app) - 1);
+			event_app[sizeof(event_app) - 1] = '\0';
+		}
+		
+		p = ast_variable_retrieve(cfg, "general", "loudness");
+		if(p){
+			toneloudness = atoi(p);
+			if(toneloudness < 100)
+				toneloudness = 100;
+			if(toneloudness > 8192)
+				toneloudness = 8192;
+		}
+		p = ast_variable_retrieve(cfg, "general", "fdtimeout");
+		if(p){
+			fdtimeout = atoi(p);
+			if(fdtimeout < 1000)
+				fdtimeout = 1000;
+			if(fdtimeout > 10000)
+				fdtimeout = 10000;	
+		}
+		
+		p = ast_variable_retrieve(cfg, "general", "sdtimeout");
+		if(p){
+			sdtimeout = atoi(p);
+			if(sdtimeout < 110)
+				sdtimeout = 110;
+			if(sdtimeout > 4000)
+				sdtimeout = 4000;			
+
+		}
+		
+		p = ast_variable_retrieve(cfg, "general", "logindividualevents");
+		if(p){
+			log_individual_events = ast_true(p);
+
+		}
+		
+		p = ast_variable_retrieve(cfg, "general", "eventspooldir");
+			
+		if(p){
+			strncpy(event_spool_dir, p, sizeof(event_spool_dir) - 1);
+			event_spool_dir[sizeof(event_spool_dir) - 1] = '\0';
+		}
+		
+		p = ast_variable_retrieve(cfg, "general", "timestampformat");
+			
+		if(p){
+			strncpy(time_stamp_format, p, sizeof(time_stamp_format) - 1);
+			time_stamp_format[sizeof(time_stamp_format) - 1] = '\0';
+		}
+
+		p = ast_variable_retrieve(cfg, "general", "db-family");
+                                                                                                                                            
+		if(p){
+			strncpy(db_family, p, sizeof(db_family) - 1);
+			db_family[sizeof(db_family) - 1] = '\0';
+		}
+		ast_destroy(cfg);
+	}
+	return 0;
+
+}
+
+/*
+* These functions are required to implement an Asterisk App.
+*/
+
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	load_config();
+	return ast_register_application(app, alarmreceiver_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_authenticate.c

@@ -18,14 +18,14 @@
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
 #include <asterisk/app.h>
+#include <asterisk/astdb.h>
+#include <asterisk/utils.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <errno.h>
 #include <stdlib.h>
 #include <stdio.h>
-#include <pthread.h>
-
 
 static char *tdesc = "Authentication Application";
 
@@ -41,7 +41,11 @@ static char *descrip =
 "an optional set of opions may be provided by concatenating any\n"
 "of the following letters:\n"
 "     a - Set account code to the password that is entered\n"
+"     d - Interpret path as database key, not literal file\n"
+"     r - Remove database key upon successful entry (valid with 'd' only)\n"
 "\n"
+"When using a database key, the value associated with the key can be\n"
+"anything.\n"
 "Returns 0 if the user enters a valid password within three\n"
 "tries, or -1 otherwise (or on hangup).\n";
 
@@ -58,7 +62,7 @@ static int auth_exec(struct ast_channel *chan, void *data)
 	char passwd[256];
 	char *opts;
 	char *prompt;
-	if (!data || !strlen(data)) {
+	if (!data || ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Authenticate requires an argument(password)\n");
 		return -1;
 	}
@@ -85,24 +89,36 @@ static int auth_exec(struct ast_channel *chan, void *data)
 			break;
 		res = 0;
 		if (password[0] == '/') {
-			/* Compare against a file */
-			FILE *f;
-			f = fopen(password, "r");
-			if (f) {
-				char buf[256] = "";
-				while(!feof(f)) {
-					fgets(buf, sizeof(buf), f);
-					if (!feof(f) && strlen(buf)) {
-						buf[strlen(buf) - 1] = '\0';
-						if (strlen(buf) && !strcmp(passwd, buf))
-							break;
+			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);
 					}
-				}
-				fclose(f);
-				if (strlen(buf) && !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;
+				f = fopen(password, "r");
+				if (f) {
+					char buf[256] = "";
+					while(!feof(f)) {
+						fgets(buf, sizeof(buf), f);
+						if (!feof(f) && !ast_strlen_zero(buf)) {
+							buf[strlen(buf) - 1] = '\0';
+							if (!ast_strlen_zero(buf) && !strcmp(passwd, buf))
+								break;
+						}
+					}
+					fclose(f);
+					if (!ast_strlen_zero(buf) && !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 fixed password */
 			if (!strcmp(passwd, password)) 

apps/app_cdr.c

@@ -0,0 +1,66 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Applictions connected with CDR engine
+ * 
+ * Copyright (C) 2003, Digium
+ *
+ * Martin Pycko <martinp@digium.com>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <sys/types.h>
+#include <asterisk/channel.h>
+#include <asterisk/module.h>
+#include <asterisk/pbx.h>
+#include <stdlib.h>
+
+
+static char *tdesc = "Make sure asterisk doesn't save CDR for a certain call";
+
+static char *nocdr_descrip = "NoCDR(): makes sure there won't be any CDR written for a certain call";
+static char *nocdr_app = "NoCDR";
+static char *nocdr_synopsis = "Make sure asterisk doesn't save CDR for a certain call";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int nocdr_exec(struct ast_channel *chan, void *data)
+{
+	if (chan->cdr) {
+		ast_cdr_free(chan->cdr);
+		chan->cdr = NULL;
+	}
+	return 0;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(nocdr_app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(nocdr_app, nocdr_exec, nocdr_synopsis, nocdr_descrip);
+}
+
+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

@@ -27,8 +27,6 @@
 #include <stdlib.h>
 #include <sys/ioctl.h>
 
-#include <pthread.h>
-
 static char *tdesc = "Check if channel is available";
 
 static char *app = "ChanIsAvail";
@@ -38,12 +36,14 @@ static char *synopsis = "Check if channel is available";
 static char *descrip = 
 "  ChanIsAvail(Technology/resource[&Technology2/resource2...]): \n"
 "Checks is any of the requested channels are available.  If none\n"
-"of the requested channels are available the new priority will\n"
-"be n+101 (unless such a priority does not exist, in which case\n"
-"ChanIsAvail will return -1.  If any of the requested channels\n"
-"are available, the next priority will be n+1, the channel variable\n"
-"${AVAILCHAN} will be set to the name of the available channel and\n"
-"the ChanIsAvail app will return 0.\n";
+"of the requested channels are available the new priority will be\n"
+"n+101 (unless such a priority does not exist or on error, in which\n"
+"case ChanIsAvail will return -1).\n"
+"If any of the requested channels are available, the next priority will be n+1,\n"
+"the channel variable ${AVAILCHAN} will be set to the name of the available channel\n"
+"and the ChanIsAvail app will return 0.\n"
+"${AVAILORIGCHAN} is the canonical channel name that was used to create the channel.\n"
+"${AVAILSTATUS} is the status code for the channel.\n";
 
 STANDARD_LOCAL_USER;
 
@@ -53,7 +53,7 @@ static int chanavail_exec(struct ast_channel *chan, void *data)
 {
 	int res=-1;
 	struct localuser *u;
-	char info[256], *peers, *tech, *number, *rest, *cur;
+	char info[512], tmp[512], *peers, *tech, *number, *rest, *cur;
 	struct ast_channel *tempchan;
 
 	if (!data) {
@@ -62,7 +62,7 @@ static int chanavail_exec(struct ast_channel *chan, void *data)
 	}
 	LOCAL_USER_ADD(u);
 
-	strncpy(info, (char *)data, strlen((char *)data) + AST_MAX_EXTENSION-1);
+	strncpy(info, (char *)data, sizeof(info)-1);
 	peers = info;
 	if (peers) {
 		cur = peers;
@@ -76,13 +76,16 @@ static int chanavail_exec(struct ast_channel *chan, void *data)
 			tech = cur;
 			number = strchr(tech, '/');
 			if (!number) {
-				ast_log(LOG_WARNING, "ChanIsAvail argument takes format (Zap/[device])\n");
-				continue;
+				ast_log(LOG_WARNING, "ChanIsAvail argument takes format ([technology]/[device])\n");
+				return -1;
 			}
 			*number = '\0';
 			number++;
 			if ((tempchan = ast_request(tech, chan->nativeformats, number))) {
 					pbx_builtin_setvar_helper(chan, "AVAILCHAN", tempchan->name);
+					/* Store the originally used channel too */
+					snprintf(tmp, sizeof(tmp), "%s/%s", tech, number);
+					pbx_builtin_setvar_helper(chan, "AVAILORIGCHAN", tmp);
 					ast_hangup(tempchan);
 					tempchan = NULL;
 					res = 1;
@@ -94,6 +97,7 @@ static int chanavail_exec(struct ast_channel *chan, void *data)
 
 	if (res < 1) {
 		pbx_builtin_setvar_helper(chan, "AVAILCHAN", "");
+		pbx_builtin_setvar_helper(chan, "AVAILORIGCHAN", "");
 		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
 			chan->priority+=100;
 		else

apps/app_controlplayback.c

@@ -0,0 +1,138 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Trivial application to control playback a sound file
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/app.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/utils.h>
+#include <string.h>
+#include <stdlib.h>
+
+static char *tdesc = "Control Playback Application";
+
+static char *app = "ControlPlayback";
+
+static char *synopsis = "Play a file with fast forward and rewind";
+
+static char *descrip = 
+"ControlPlayback(filename[|skipms[|ffchar[|rewchar[|stopchar[|pausechr]]]]]):\n"
+"  Plays  back  a  given  filename (do not put extension). Options may also\n"
+"  be included following a pipe symbol.  You can use * and # to rewind and\n"
+"  fast forward the playback specified. If 'stopchar' is added the file will\n"
+"  terminate playback when 'stopchar' is pressed. Returns -1 if the channel\n"
+"  was hung up, or if the file does not exist. Returns 0 otherwise.\n\n"
+"  Example:  exten => 1234,1,ControlPlayback(file|4000|*|#|1|0)\n\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int is_on_phonepad(char key)
+{
+	return (key == 35 || key == 42 || (key >= 48 && key <= 57)) ? 1 : 0;
+}
+
+static int controlplayback_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	int skipms = 0;
+	struct localuser *u;
+	char tmp[256];
+	char *skip = NULL, *fwd = NULL, *rev = NULL, *stop = NULL, *pause = NULL, *file = NULL;
+
+	if (!data || ast_strlen_zero((char *)data)) {
+		ast_log(LOG_WARNING, "ControlPlayback requires an argument (filename)\n");
+		return -1;
+	}
+
+	strncpy(tmp, (char *)data, sizeof(tmp)-1);
+	file = tmp;
+
+	if ((skip=strchr(tmp,'|'))) {
+		*skip++ = '\0';
+		fwd=strchr(skip,'|');
+		if (fwd) {
+			*fwd++ = '\0';
+			rev = strchr(fwd,'|');
+			if (rev) {
+				*rev++ = '\0';
+				stop = strchr(rev,'|');
+				if (stop) {
+					*stop++ = '\0';
+					pause = strchr(stop,'|');
+					if (pause) {
+						*pause++ = '\0';
+					}
+				}
+			}
+		}
+	}
+
+	skipms = skip ? atoi(skip) : 3000;
+	if (!skipms)
+		skipms = 3000;
+
+	if (!fwd || !is_on_phonepad(*fwd))
+		fwd = "#";
+	if (!rev || !is_on_phonepad(*rev))
+		rev = "*";
+	if (stop && !is_on_phonepad(*stop))
+		stop = NULL;
+	if (pause && !is_on_phonepad(*pause))
+		pause = NULL;
+
+	LOCAL_USER_ADD(u);
+
+	res = ast_control_streamfile(chan, file, fwd, rev, stop, pause, skipms);
+
+	LOCAL_USER_REMOVE(u);
+	
+	/* If we stopped on one of our stop keys, return 0  */
+	if(stop && strchr(stop, res)) 
+		res = 0;
+	
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, controlplayback_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_cut.c

@@ -0,0 +1,198 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Cut application
+ *
+ * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
+ *
+ * Tilghman Lesher <app_cut__v003@the-tilghman.com>
+ *
+ * $Id$
+ *
+ * This code is released by the author with no restrictions on usage.
+ *
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+
+/* Maximum length of any variable */
+#define MAXRESULT	1024
+
+static char *tdesc = "Cuts up variables";
+
+static char *app_cut = "Cut";
+
+static char *cut_synopsis = "Splits a variable's content using the specified delimiter";
+
+static char *cut_descrip =
+"Usage: Cut(newvar=varname,delimiter,fieldspec)\n"
+"  newvar    - new variable created from result string\n"
+"  varname   - variable you want cut\n"
+"  delimiter - defaults to '-'\n"
+"  fieldspec - number of the field you want (1-based offset)\n"
+"            may also be specified as a range (with -)\n"
+"            or group of ranges and fields (with &)\n" 
+"  Returns 0 or -1 on hangup or error.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int cut_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *s, *newvar=NULL, *varname=NULL, *delimiter=NULL, *field=NULL;
+	int args_okay = 0;
+
+	LOCAL_USER_ADD(u);
+
+	/* Check and parse arguments */
+	if (data) {
+		s = ast_strdupa((char *)data);
+		if (s) {
+			newvar = strsep(&s, "=");
+			if (newvar && (newvar[0] != '\0')) {
+				varname = strsep(&s, "|");
+				if (varname && (varname[0] != '\0')) {
+					delimiter = strsep(&s, "|");
+					if (delimiter) {
+						field = strsep(&s, "|");
+						if (field) {
+							args_okay = 1;
+						}
+					}
+				}
+			}
+		} else {
+			ast_log(LOG_ERROR, "Out of memory\n");
+			res = -1;
+		}
+	}
+
+	if (args_okay) {
+		char d, ds[2];
+		char *tmp = alloca(strlen(varname) + 4);
+		char varvalue[MAXRESULT], *tmp2=varvalue;
+		char retstring[MAXRESULT];
+
+		memset(retstring, 0, MAXRESULT);
+
+		if (tmp) {
+			snprintf(tmp, strlen(varname) + 4, "${%s}", varname);
+			memset(varvalue, 0, sizeof(varvalue));
+		} else {
+			ast_log(LOG_ERROR, "Out of memory");
+			return -1;
+		}
+
+		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 {
+					ast_log(LOG_ERROR, "Cut(): Illegal range '%s'\n", nextgroup);
+					ast_log(LOG_ERROR, "Usage: %s\n", cut_synopsis);
+					return -1;
+				}
+
+				/* 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, "Cut(): 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(retstring);
+
+					if (strlen(retstring)) {
+						snprintf(retstring + curlen, MAXRESULT - curlen, "%c%s", d, tmp3);
+					} else {
+						snprintf(retstring, MAXRESULT, "%s", tmp3);
+					}
+
+					curfieldnum++;
+				}
+			}
+		}
+
+		pbx_builtin_setvar_helper(chan, newvar, retstring);
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app_cut);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app_cut, cut_exec, cut_synopsis, cut_descrip);
+}
+
+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

@@ -23,9 +23,6 @@
 #include <string.h>
 #include <stdlib.h>
 
-#include <pthread.h>
-
-
 static char *tdesc = "Date and Time";
 
 static char *app = "DateTime";

apps/app_db.c

@@ -13,39 +13,38 @@
  * the GNU General Public License
  */
 
+#include <sys/types.h>
 #include <asterisk/options.h>
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
-#include <asterisk/pbx.h>
 #include <asterisk/astdb.h>
+#include <asterisk/lock.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
-
 
 static char *tdesc = "Database access functions for Asterisk extension logic";
 
 static char *g_descrip =
-  "  DBget(varname=family/key): Retrieves a value from the Asterisk\n"
-  "database and stores it in the given variable.  Always returns 0.  If the\n"
-  "requested key is not found, jumps to priority n+101 if available.\n";
+	"  DBget(varname=family/key): Retrieves a value from the Asterisk\n"
+	"database and stores it in the given variable.  Always returns 0.  If the\n"
+	"requested key is not found, jumps to priority n+101 if available.\n";
 
 static char *p_descrip =
-  "  DBput(family/key=value): Stores the given value in the Asterisk\n"
-  "database.  Always returns 0.\n";
+	"  DBput(family/key=value): Stores the given value in the Asterisk\n"
+	"database.  Always returns 0.\n";
 
 static char *d_descrip =
-  "  DBdel(family/key): Deletes a key from the Asterisk database.  Always\n"
-  "returns 0.\n";
+	"  DBdel(family/key): Deletes a key from the Asterisk database.  Always\n"
+	"returns 0.\n";
 
 static char *dt_descrip =
-  "  DBdeltree(family[/keytree]): Deletes a family or keytree from the Asterisk\n"
-  "database.  Always returns 0.\n";
+	"  DBdeltree(family[/keytree]): Deletes a family or keytree from the Asterisk\n"
+	"database.  Always returns 0.\n";
 
 static char *g_app = "DBget";
 static char *p_app = "DBput";
@@ -61,255 +60,196 @@ STANDARD_LOCAL_USER;
 
 LOCAL_USER_DECL;
 
-
-static int
-deltree_exec (struct ast_channel *chan, void *data)
+static int deltree_exec (struct ast_channel *chan, void *data)
 {
-  int arglen;
-  char *argv, *family, *keytree;
+	int arglen;
+	char *argv, *family, *keytree;
 
-  arglen = strlen (data);
-  argv = alloca (arglen + 1);
-  if (!argv)			/* Why would this fail? */
-    {
-      ast_log (LOG_DEBUG, "Memory allocation failed\n");
-      return 0;
-    }
-  memcpy (argv, data, arglen + 1);
-
-  if (strchr (argv, '/')) {
-    {
-      family = strsep (&argv, "/");
-      keytree = strsep (&argv, "\0");
-      if (!family || !keytree)
-	{
-	  ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
-	  return 0;
+	arglen = strlen (data);
+	argv = alloca (arglen + 1);
+	if (!argv) {	/* Why would this fail? */
+		ast_log (LOG_DEBUG, "Memory allocation failed\n");
+		return 0;
 	}
-      if (!strlen (keytree))
-	keytree = 0;
-    }
-  } else
-    {
-      family = argv;
-      keytree = 0;
-    }
+	memcpy (argv, data, arglen + 1);
 
-  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))
-    {
-      if (option_verbose > 2)
-	ast_verbose (VERBOSE_PREFIX_3
-		     "DBdeltree: Error deleting key from database.\n");
-    }
-
-  return 0;
-}
-
-
-static int
-del_exec (struct ast_channel *chan, void *data)
-{
-  int arglen;
-  char *argv, *family, *key;
-
-  arglen = strlen (data);
-  argv = alloca (arglen + 1);
-  if (!argv)			/* Why would this fail? */
-    {
-      ast_log (LOG_DEBUG, "Memory allocation failed\n");
-      return 0;
-    }
-  memcpy (argv, data, arglen + 1);
-
-  if (strchr (argv, '/'))
-    {
-      family = strsep (&argv, "/");
-      key = strsep (&argv, "\0");
-      if (!family || !key)
-	{
-	  ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
-	  return 0;
-	}
-      if (option_verbose > 2)
-	ast_verbose (VERBOSE_PREFIX_3
-		     "DBdel: family=%s, key=%s\n", family, key);
-      if (ast_db_del (family, key))
-	{
-	  if (option_verbose > 2)
-	    ast_verbose (VERBOSE_PREFIX_3
-			 "DBdel: Error deleting key from database.\n");
-	}
-    }
-  else
-    {
-      ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
-    }
-
-  return 0;
-}
-
-static int
-put_exec (struct ast_channel *chan, void *data)
-{
-  int arglen;
-  char *argv, *value, *family, *key;
-
-  arglen = strlen (data);
-  argv = alloca (arglen + 1);
-  if (!argv)			/* Why would this fail? */
-    {
-      ast_log (LOG_DEBUG, "Memory allocation failed\n");
-      return 0;
-    }
-  memcpy (argv, data, arglen + 1);
-
-  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");
-	  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");
+	if (strchr (argv, '/')) {
+		family = strsep (&argv, "/");
+		keytree = strsep (&argv, "\0");
+			if (!family || !keytree) {
+				ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+				return 0;
+			}
+		if (!strlen (keytree))
+			keytree = 0;
+	} else {
+		family = argv;
+		keytree = 0;
 	}
 
-    }
-  else
-    {
-      ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
-    }
-
-  return 0;
-}
-
-static int
-get_exec (struct ast_channel *chan, void *data)
-{
-  int arglen;
-  char *argv, *varname, *family, *key;
-  char dbresult[256];
-
-  arglen = strlen (data);
-  argv = alloca (arglen + 1);
-  if (!argv)			/* Why would this fail? */
-    {
-      ast_log (LOG_DEBUG, "Memory allocation failed\n");
-      return 0;
-    }
-  memcpy (argv, data, arglen + 1);
-
-  if (strchr (argv, '=') && strchr (argv, '/'))
-    {
-      varname = strsep (&argv, "=");
-      family = strsep (&argv, "/");
-      key = strsep (&argv, "\0");
-      if (!varname || !family || !key)
-	{
-	  ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
-	  return 0;
-	}
-      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);
-	}
-      else
-	{
-	  if (option_verbose > 2)
-	    ast_verbose (VERBOSE_PREFIX_3
-			 "DBget: Value not found in database.\n");
-	  /* Send the call to n+101 priority, where n is the current priority */
-	  if (ast_exists_extension
-	      (chan, chan->context, chan->exten, chan->priority + 101,
-	       chan->callerid))
-	    chan->priority += 100;
+	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);
 	}
 
-    }
-  else
-    {
-      ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
-    }
+	if (ast_db_deltree (family, keytree)) {
+		if (option_verbose > 2)
+			ast_verbose (VERBOSE_PREFIX_3 "DBdeltree: Error deleting key from database.\n");
+	}
 
-  return 0;
+	return 0;
 }
 
-int
-unload_module (void)
+static int del_exec (struct ast_channel *chan, void *data)
 {
-  int retval;
+	int arglen;
+	char *argv, *family, *key;
 
-  STANDARD_HANGUP_LOCALUSERS;
-  retval = ast_unregister_application (dt_app);
-  retval |= ast_unregister_application (d_app);
-  retval |= ast_unregister_application (p_app);
-  retval |= ast_unregister_application (g_app);
+	arglen = strlen (data);
+	argv = alloca (arglen + 1);
+	if (!argv) {	/* Why would this fail? */
+		ast_log (LOG_DEBUG, "Memory allocation failed\n");
+		return 0;
+	}
+	memcpy (argv, data, arglen + 1);
 
-  return retval;
+	if (strchr (argv, '/')) {
+		family = strsep (&argv, "/");
+		key = strsep (&argv, "\0");
+		if (!family || !key) {
+			ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			return 0;
+		}
+		if (option_verbose > 2)
+			ast_verbose (VERBOSE_PREFIX_3 "DBdel: family=%s, key=%s\n", family, key);
+		if (ast_db_del (family, key)) {
+			if (option_verbose > 2)
+				ast_verbose (VERBOSE_PREFIX_3 "DBdel: Error deleting key from database.\n");
+		}
+	} else {
+		ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
+	}
+	return 0;
 }
 
-int
-load_module (void)
+static int put_exec (struct ast_channel *chan, void *data)
 {
-  int retval;
+	int arglen;
+	char *argv, *value, *family, *key;
 
-  retval = ast_register_application (g_app, get_exec, g_synopsis, g_descrip);
-  if (!retval)
-    retval =
-      ast_register_application (p_app, put_exec, p_synopsis, p_descrip);
-  if (!retval)
-    retval =
-      ast_register_application (d_app, del_exec, d_synopsis, d_descrip);
-  if (!retval)
-    retval =
-      ast_register_application (dt_app, deltree_exec, dt_synopsis,
-				dt_descrip);
+	arglen = strlen (data);
+	argv = alloca (arglen + 1);
+	if (!argv) {	/* Why would this fail? */
+		ast_log (LOG_DEBUG, "Memory allocation failed\n");
+		return 0;
+	}
+	memcpy (argv, data, arglen + 1);
 
-  return retval;
+	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");
+			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");
+	}
+	return 0;
 }
 
-char *
-description (void)
+static int get_exec (struct ast_channel *chan, void *data)
 {
-  return tdesc;
+	int arglen;
+	char *argv, *varname, *family, *key;
+	char dbresult[256];
+
+	arglen = strlen (data);
+	argv = alloca (arglen + 1);
+	if (!argv) {	/* Why would this fail? */
+		ast_log (LOG_DEBUG, "Memory allocation failed\n");
+		return 0;
+	}
+	memcpy (argv, data, arglen + 1);
+
+	if (strchr (argv, '=') && strchr (argv, '/')) {
+		varname = strsep (&argv, "=");
+		family = strsep (&argv, "/");
+		key = strsep (&argv, "\0");
+		if (!varname || !family || !key) {
+			ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
+			return 0;
+		}
+		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);
+		} else {
+			if (option_verbose > 2)
+			ast_verbose (VERBOSE_PREFIX_3 "DBget: Value not found in database.\n");
+			/* Send the call to n+101 priority, where n is the current priority */
+			if (ast_exists_extension (chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+				chan->priority += 100;
+		}
+
+	} else {
+		ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
+	}
+	return 0;
 }
 
-int
-usecount (void)
+int unload_module (void)
 {
-  int res;
-  STANDARD_USECOUNT (res);
-  return res;
+	int retval;
+
+	STANDARD_HANGUP_LOCALUSERS;
+	retval = ast_unregister_application (dt_app);
+	retval |= ast_unregister_application (d_app);
+	retval |= ast_unregister_application (p_app);
+	retval |= ast_unregister_application (g_app);
+
+	return retval;
 }
 
-char *
-key ()
+int load_module (void)
 {
-  return ASTERISK_GPL_KEY;
+	int retval;
+
+	retval = ast_register_application (g_app, get_exec, g_synopsis, g_descrip);
+	if (!retval)
+		retval = ast_register_application (p_app, put_exec, p_synopsis, p_descrip);
+	if (!retval)
+		retval = ast_register_application (d_app, del_exec, d_synopsis, d_descrip);
+	if (!retval)
+		retval = ast_register_application (dt_app, deltree_exec, dt_synopsis, dt_descrip);
+	return retval;
+}
+
+char *description (void)
+{
+	return tdesc;
+}
+
+int usecount (void)
+{
+	int res;
+	STANDARD_USECOUNT (res);
+	return res;
+}
+
+char *key ()
+{
+	return ASTERISK_GPL_KEY;
 }

apps/app_directory.c

@@ -19,10 +19,10 @@
 #include <asterisk/module.h>
 #include <asterisk/config.h>
 #include <asterisk/say.h>
+#include <asterisk/utils.h>
 #include <string.h>
 #include <ctype.h>
 #include <stdlib.h>
-#include <pthread.h>
 #include <stdio.h>
 #include "../asterisk.h"
 #include "../astconf.h"
@@ -32,12 +32,15 @@ static char *app = "Directory";
 
 static char *synopsis = "Provide directory of voicemail extensions";
 static char *descrip =
-"  Directory(context): Presents the user with a directory of extensions from\n"
-"which they  may  select  by name. The  list  of  names  and  extensions  is\n"
-"discovered from  voicemail.conf. The  context  argument  is  required,  and\n"
-"specifies  the  context  in  which to interpret the extensions\n. Returns 0\n"
-"unless the user hangs up. It  also sets up the channel on exit to enter the\n"
-"extension the user selected.\n";
+"  Directory(vm-context[|dial-context[|options]]): Presents the user with a directory\n"
+"of extensions from which they  may  select  by name. The  list  of  names \n"
+"and  extensions  is discovered from  voicemail.conf. The  vm-context  argument\n"
+"is required, and specifies  the  context  of voicemail.conf to use.  The\n"
+"dial-context is the context to use for dialing the users, and defaults to\n"
+"the vm-context if unspecified. The 'f' option causes the directory to match\n"
+"based on the first name in voicemail.conf instead of the last name.\n"
+"Returns 0 unless the user hangs up. It  also sets up the channel on exit\n"
+"to enter the extension the user selected.\n";
 
 /* For simplicity, I'm keeping the format compatible with the voicemail config,
    but i'm open to suggestions for isolating it */
@@ -112,7 +115,6 @@ static char *convert(char *lastname)
 			case 'Z':
 				tmp[lcount++] = '9';
 				break;
-			default:
 			}
 			lastname++;
 		}
@@ -121,17 +123,108 @@ static char *convert(char *lastname)
 	return tmp;
 }
 
-static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *context, char digit)
+/* play name of mailbox owner.
+ * returns:  -1 for bad or missing extension
+ *           '1' for selected entry from directory
+ *           '*' for skipped entry from directory
+ */
+static int play_mailbox_owner(struct ast_channel *chan, char *context, char *dialcontext, char *ext, char *name) {
+	int res = 0;
+	int loop = 3;
+	char fn[256];
+	char fn2[256];
+
+	/* Check for the VoiceMail2 greeting first */
+	snprintf(fn, sizeof(fn), "%s/voicemail/%s/%s/greet",
+		(char *)ast_config_AST_SPOOL_DIR, context, ext);
+
+	/* Otherwise, check for an old-style Voicemail greeting */
+	snprintf(fn2, sizeof(fn2), "%s/vm/%s/greet",
+		(char *)ast_config_AST_SPOOL_DIR, ext);
+
+	if (ast_fileexists(fn, NULL, chan->language) > 0) {
+		res = ast_streamfile(chan, fn, chan->language);
+		if (!res) {
+			res = ast_waitstream(chan, AST_DIGIT_ANY);
+		}
+		ast_stopstream(chan);
+	} else if (ast_fileexists(fn2, NULL, chan->language) > 0) {
+		res = ast_streamfile(chan, fn2, chan->language);
+		if (!res) {
+			res = ast_waitstream(chan, AST_DIGIT_ANY);
+		}
+		ast_stopstream(chan);
+	} else {
+		res = ast_say_character_str(chan, !ast_strlen_zero(name) ? name : ext,
+					AST_DIGIT_ANY, chan->language);
+	}
+
+	while (loop) {
+		if (!res) {
+			res = ast_streamfile(chan, "dir-instr", chan->language);
+		}
+		if (!res) {
+			res = ast_waitstream(chan, AST_DIGIT_ANY);
+		}
+		if (!res) {
+			res = ast_waitfordigit(chan, 3000);
+		}
+		ast_stopstream(chan);
+	
+		if (res > -1) {
+			switch (res) {
+				case '1':
+					/* Name selected */
+					loop = 0;
+					if (ast_exists_extension(chan,dialcontext,ext,1,chan->callerid)) {
+						strncpy(chan->exten, ext, sizeof(chan->exten)-1);
+						chan->priority = 0;
+						strncpy(chan->context, dialcontext, sizeof(chan->context)-1);
+					} else {
+						ast_log(LOG_WARNING,
+							"Can't find extension '%s' in context '%s'.  "
+							"Did you pass the wrong context to Directory?\n",
+							ext, dialcontext);
+						res = -1;
+					}
+					break;
+	
+				case '*':   
+					/* Skip to next match in list */
+					loop = 0;
+					break;
+	
+				default:
+					/* Not '1', or '*', so decrement number of tries */
+					res = 0;
+					loop--;
+					break;
+			} /* end switch */
+		} /* end if */
+		else {
+			/* User hungup, so jump out now */
+			loop = 0;
+		}
+	} /* end while */
+
+	return(res);
+}
+
+static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *context, char *dialcontext, char digit, int last)
 {
 	/* Read in the first three digits..  "digit" is the first digit, already read */
 	char ext[NUMDIGITS + 1];
+	char name[80] = "";
 	struct ast_variable *v;
 	int res;
 	int found=0;
+	int lastuserchoice = 0;
 	char *start, *pos, *conv,*stringp=NULL;
-	char fn[256];
-	if (!context || !strlen(context)) {
-		ast_log(LOG_WARNING, "Directory must be called with an argument (context in which to interpret extensions)\n");
+
+	if (!context || ast_strlen_zero(context)) {
+		ast_log(LOG_WARNING,
+			"Directory must be called with an argument "
+			"(context in which to interpret extensions)\n");
 		return -1;
 	}
 	memset(ext, 0, sizeof(ext));
@@ -151,8 +244,9 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
 					strsep(&stringp, ",");
 					pos = strsep(&stringp, ",");
 					if (pos) {
+						strncpy(name, pos, sizeof(name) - 1);
 						/* Grab the last name */
-						if (strrchr(pos, ' '))
+						if (last && strrchr(pos,' '))
 							pos = strrchr(pos, ' ') + 1;
 						conv = convert(pos);
 						if (conv) {
@@ -170,51 +264,46 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
 				}
 				v = v->next;
 			}
+
 			if (v) {
 				/* We have a match -- play a greeting if they have it */
-				snprintf(fn, sizeof(fn), "%s/vm/%s/greet", (char *)ast_config_AST_SPOOL_DIR, v->name);
-				if (ast_fileexists(fn, NULL, chan->language) > 0) {
-					res = ast_streamfile(chan, fn, chan->language);
-					if (!res)
-						res = ast_waitstream(chan, AST_DIGIT_ANY);
-					ast_stopstream(chan);
-				} else {
-					res = ast_say_digit_str(chan, v->name, AST_DIGIT_ANY, chan->language);
-				}
-ahem:
-				if (!res)
-					res = ast_streamfile(chan, "dir-instr", chan->language);
-				if (!res)
-					res = ast_waitstream(chan, AST_DIGIT_ANY);
-				if (!res)
-					res = ast_waitfordigit(chan, 3000);
-				ast_stopstream(chan);
-				if (res > -1) {
-					if (res == '1') {
-						strncpy(chan->exten, v->name, sizeof(chan->exten)-1);
+				res = play_mailbox_owner(chan, context, dialcontext, v->name, name);
+				switch (res) {
+					case -1:
+						/* user pressed '1' but extension does not exist, or
+						 * user hungup
+						 */
+						lastuserchoice = 0;
+						break;
+					case '1':
+						/* user pressed '1' and extensions exists */
+						lastuserchoice = res;
+						strncpy(chan->context, dialcontext, sizeof(chan->context) - 1);
+						strncpy(chan->exten, v->name, sizeof(chan->exten) - 1);
 						chan->priority = 0;
-						strncpy(chan->context, context, sizeof(chan->context)-1);
+						break;
+					case '*':
+						/* user pressed '*' to skip something found */
+						lastuserchoice = res;
 						res = 0;
 						break;
-					} else if (res == '*') {
-						res = 0;
-						v = v->next;
-					} else {
-						res = 0;
-						goto ahem;
-					}
+					default:
+						break;
 				}
-			} else {
-				if (found) 
-					res = ast_streamfile(chan, "dir-nomore", chan->language);
-				else
-					res = ast_streamfile(chan, "dir-nomatch", chan->language);
-				if (!res)
-					res = 1;
-				return res;
+				v = v->next;
 			}
 		}
-		
+
+		if (lastuserchoice != '1') {
+			if (found) 
+				res = ast_streamfile(chan, "dir-nomore", chan->language);
+			else
+				res = ast_streamfile(chan, "dir-nomatch", chan->language);
+			if (!res)
+				res = 1;
+			return res;
+		}
+		return 0;
 	}
 	return res;
 }
@@ -224,8 +313,11 @@ static int directory_exec(struct ast_channel *chan, void *data)
 	int res = 0;
 	struct localuser *u;
 	struct ast_config *cfg;
+	int last = 1;
+	char *context, *dialcontext, *dirintro, *options;
+
 	if (!data) {
-		ast_log(LOG_WARNING, "directory requires an argument (context)\n");
+		ast_log(LOG_WARNING, "directory requires an argument (context[,dialcontext])\n");
 		return -1;
 	}
 	cfg = ast_load(DIRECTORY_CONFIG);
@@ -235,15 +327,40 @@ static int directory_exec(struct ast_channel *chan, void *data)
 	}
 	LOCAL_USER_ADD(u);
 top:
+	context = ast_strdupa(data);
+	dialcontext = strchr(context, '|');
+	if (dialcontext) {
+		*dialcontext = '\0';
+		dialcontext++;
+		options = strchr(dialcontext, '|');
+		if (options) {
+			*options = '\0';
+			options++; 
+			if (strchr(options, 'f'))
+				last = 0;
+		}
+	} else	
+		dialcontext = context;
+	dirintro = ast_variable_retrieve(cfg, context, "directoryintro");
+	if (!dirintro || ast_strlen_zero(dirintro))
+		dirintro = ast_variable_retrieve(cfg, "general", "directoryintro");
+	if (!dirintro || ast_strlen_zero(dirintro)) {
+		if (last)
+			dirintro = "dir-intro";	
+		else
+			dirintro = "dir-intro-fn";
+	}
+	if (chan->_state != AST_STATE_UP) 
+		res = ast_answer(chan);
 	if (!res)
-		res = ast_streamfile(chan, "dir-intro", chan->language);
+		res = ast_streamfile(chan, dirintro, chan->language);
 	if (!res)
 		res = ast_waitstream(chan, AST_DIGIT_ANY);
 	ast_stopstream(chan);
 	if (!res)
 		res = ast_waitfordigit(chan, 5000);
 	if (res > 0) {
-		res = do_directory(chan, cfg, (char *)data, res);
+		res = do_directory(chan, cfg, context, dialcontext, res, last);
 		if (res > 0) {
 			res = ast_waitstream(chan, AST_DIGIT_ANY);
 			ast_stopstream(chan);

apps/app_disa.c

@@ -7,6 +7,8 @@
  *
  * Jim Dixon <jim@lambdatel.com>
  *
+ * Made only slightly more sane by Mark Spencer <markster@digium.com>
+ *
  * This program is free software, distributed under the terms of
  * the GNU General Public License
  */
@@ -15,20 +17,18 @@
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
+#include <asterisk/indications.h>
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
 #include <asterisk/translate.h>
 #include <asterisk/ulaw.h>
+#include <asterisk/utils.h>
 #include <string.h>
 #include <stdlib.h>
 #include <stdio.h>
 #include <math.h>
-#include <pthread.h>
 #include <sys/time.h>
 
-/*
-#define	TONE_BLOCK_SIZE 320
-*/
 
 static char *tdesc = "DISA (Direct Inward System Access) Application";
 
@@ -81,26 +81,8 @@ STANDARD_LOCAL_USER;
 
 LOCAL_USER_DECL;
 
-static float loudness=4096.0;
-
-int firstdigittimeout = 20000; /* 20 seconds first digit timeout */
-int digittimeout = 10000; /* 10 seconds subsequent digit timeout */
-
-static void make_tone_block(unsigned char *data, float f1, float f2, int len, int *x)
-{
-int	i;
-float	val;
-
-	for(i = 0; i < len; i++)
-	{
-		val = loudness * sin((f1 * 2.0 * M_PI * (*x))/8000.0);
-		val += loudness * sin((f2 * 2.0 * M_PI * (*x)++)/8000.0);
-		data[i] = AST_LIN2MU((int)val);
-	 }		
-	  /* wrap back around from 8000 */
-	if (*x >= 8000) *x = 0;
-	return;
-}
+static int firstdigittimeout = 20000; /* 20 seconds first digit timeout */
+static int digittimeout = 10000; /* 10 seconds subsequent digit timeout */
 
 static int ms_diff(struct timeval *tv1, struct timeval *tv2)
 {
@@ -111,16 +93,26 @@ int	ms;
 	return(ms);
 }
 
+static void play_dialtone(struct ast_channel *chan)
+{
+	const struct tone_zone_sound *ts = NULL;
+	ts = ast_get_indication_tone(chan->zone, "dial");
+	if (ts)
+		ast_playtones_start(chan, 0, ts->data, 0);
+	else
+		ast_tonepair_start(chan, 350, 440, 0, 0);
+}
+
 static int disa_exec(struct ast_channel *chan, void *data)
 {
-	int i,j,k,x;
+	int i,j,k,x,did_ignore;
 	struct localuser *u;
-	char tmp[256],arg2[256],exten[AST_MAX_EXTENSION],acctcode[20];
-	unsigned char tone_block[640];
+	char tmp[256],arg2[256]="",exten[AST_MAX_EXTENSION],acctcode[20]="";
 	char *ourcontext,*ourcallerid;
-	struct ast_frame *f,wf;
+	struct ast_frame *f;
 	struct timeval lastout, now, lastdigittime;
 	int res;
+	time_t rstart;
 	FILE *fp;
 	char *stringp=NULL;
 
@@ -145,7 +137,7 @@ static int disa_exec(struct ast_channel *chan, void *data)
 	ourcontext = strsep(&stringp, "|");
 	/* if context specified, save 2nd arg and parse third */
 	if (ourcontext) {
-		strcpy(arg2,ourcontext);
+		strncpy(arg2,ourcontext, sizeof(arg2) - 1);
 		ourcallerid = strsep(&stringp,"|");
 	}
 	  /* if context not specified, use "disa" */
@@ -161,25 +153,32 @@ static int disa_exec(struct ast_channel *chan, void *data)
 		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? */ 
+
+	ast_log(LOG_DEBUG, "Context: %s\n",ourcontext);
+
 	if (!strcasecmp(tmp, "no-password"))
-	{
-		k = 1;
+	{;
+		k |= 1; /* We have the password */
 		ast_log(LOG_DEBUG, "DISA no-password login success\n");
 	}
 	gettimeofday(&lastdigittime,NULL);
+
+	play_dialtone(chan);
+
 	for(;;)
 	{
 		gettimeofday(&now,NULL);
 		  /* if outa time, give em reorder */
 		if (ms_diff(&now,&lastdigittime) > 
-		    ((k) ? digittimeout : firstdigittimeout))
+		    ((k&2) ? digittimeout : firstdigittimeout))
 		{
 			ast_log(LOG_DEBUG,"DISA %s entry timeout on chan %s\n",
-				((k) ? "extension" : "password"),chan->name);
-			goto reorder;
+				((k&1) ? "extension" : "password"),chan->name);
+			break;
 		}
 		if ((res = ast_waitfor(chan, -1) < 0)) {
 			ast_log(LOG_DEBUG, "Waitfor returned %d\n", res);
@@ -200,24 +199,7 @@ static int disa_exec(struct ast_channel *chan, void *data)
 			return -1;
 		}
 		if (f->frametype == AST_FRAME_VOICE) {
-			if (!i || (ast_ignore_pattern(ourcontext, exten) && k)) {
-				wf.frametype = AST_FRAME_VOICE;
-				wf.subclass = AST_FORMAT_ULAW;
-				wf.offset = AST_FRIENDLY_OFFSET;
-				wf.mallocd = 0;
-				wf.data = tone_block;
-				wf.datalen = f->datalen;
-				make_tone_block(tone_block, 350, 440, f->datalen, &x);
-				wf.samples = wf.datalen;
-				ast_frfree(f);
-			    if (ast_write(chan, &wf)) 
-				{
-			        ast_log(LOG_WARNING, "DISA Failed to write frame on %s\n",chan->name);
-					LOCAL_USER_REMOVE(u);
-					return -1;
-				}
-			} else
-				ast_frfree(f);
+			ast_frfree(f);
 			continue;
 		}
 		  /* if not DTMF, just do it again */
@@ -229,12 +211,16 @@ static int disa_exec(struct ast_channel *chan, void *data)
 
 		j = f->subclass;  /* save digit */
 		ast_frfree(f);
-		
+		if (i == 0) 
+		{
+			k|=2; /* We have the first digit */ 
+			ast_playtones_stop(chan);
+		}
 		gettimeofday(&lastdigittime,NULL);
 		  /* got a DTMF tone */
 		if (i < AST_MAX_EXTENSION) /* if still valid number of digits */
 		{
-			if (!k) /* if in password state */
+			if (!(k&1)) /* if in password state */
 			{
 				if (j == '#') /* end of password */
 				{
@@ -282,80 +268,76 @@ static int disa_exec(struct ast_channel *chan, void *data)
 
 					}
 					 /* password good, set to dial state */
-						ast_log(LOG_WARNING,"DISA on chan %s password is good\n",chan->name);
-					k = 1;
+					ast_log(LOG_DEBUG,"DISA on chan %s password is good\n",chan->name);
+					play_dialtone(chan);
+
+					k|=1; /* In number mode */
 					i = 0;  /* re-set buffer pointer */
 					exten[sizeof(acctcode)] = 0;
-					strcpy(acctcode,exten);
+					strncpy(acctcode,exten, sizeof(acctcode) - 1);
 					exten[0] = 0;
 					ast_log(LOG_DEBUG,"Successful DISA log-in on chan %s\n",chan->name);
 					continue;
 				}
 			}
+
 			exten[i++] = j;  /* save digit */
 			exten[i] = 0;
-			if (!k) continue; /* if getting password, continue doing it */
+			if (!(k&1)) continue; /* if getting password, continue doing it */
 			  /* if this exists */
-			if (ast_exists_extension(chan,ourcontext,exten,1, chan->callerid))
-			{
-				if (ourcallerid && *ourcallerid)
-				{
-					if (chan->callerid) free(chan->callerid);
-					chan->callerid = strdup(ourcallerid);
+
+			if (ast_ignore_pattern(ourcontext, exten)) {
+				play_dialtone(chan);
+				did_ignore = 1;
+			} else
+				if (did_ignore) {
+					ast_playtones_stop(chan);
+					did_ignore = 0;
 				}
-				strcpy(chan->exten,exten);
-				strcpy(chan->context,ourcontext);
-				strcpy(chan->accountcode,acctcode);
-				chan->priority = 0;
-				ast_cdr_init(chan->cdr,chan);
-				LOCAL_USER_REMOVE(u);
-				return 0;
-			}
+
 			  /* if can do some more, do it */
-			if (ast_canmatch_extension(chan,ourcontext,exten,1, chan->callerid)) continue;
-		}
-reorder:
-		
-		/* something is invalid, give em reorder forever */
-		x = 0;
-		k = 0;	/* k = 0 means busy tone, k = 1 means silence) */
-		i = 0;	/* Number of samples we've done */
-		for(;;)
-		{
-			if (ast_waitfor(chan, -1) < 0)
+			if (!ast_matchmore_extension(chan,ourcontext,exten,1, chan->callerid)) {
 				break;
-			f = ast_read(chan);
-			if (!f)
-				break;
-			if (f->frametype == AST_FRAME_VOICE) {
-				wf.frametype = AST_FRAME_VOICE;
-				wf.subclass = AST_FORMAT_ULAW;
-				wf.offset = AST_FRIENDLY_OFFSET;
-				wf.mallocd = 0;
-				wf.data = tone_block;
-				wf.datalen = f->datalen;
-				wf.samples = wf.datalen;
-				if (k) 
-					memset(tone_block, 0x7f, wf.datalen);
-				else
-					make_tone_block(tone_block,480.0, 620.0,wf.datalen, &x);
-				i += wf.datalen / 8;
-				if (i > 250) {
-					i = 0;
-					k = !k;
-				}
-			    if (ast_write(chan, &wf)) 
-				{
-			        ast_log(LOG_WARNING, "DISA Failed to write frame on %s\n",chan->name);
-					LOCAL_USER_REMOVE(u);
-					return -1;
-				}
 			}
-			ast_frfree(f);
 		}
-		LOCAL_USER_REMOVE(u);
-		return -1;
 	}
+
+	if ((k==3) && (ast_exists_extension(chan,ourcontext,exten,1, chan->callerid)))
+	{
+		ast_playtones_stop(chan);
+		/* We're authenticated and have a valid extension */
+		if (ourcallerid && *ourcallerid)
+		{
+			if (chan->callerid) free(chan->callerid);
+			chan->callerid = strdup(ourcallerid);
+		}
+		strncpy(chan->exten, exten, sizeof(chan->exten) - 1);
+		strncpy(chan->context, ourcontext, sizeof(chan->context) - 1);
+		if (!ast_strlen_zero(acctcode))
+			strncpy(chan->accountcode, acctcode, sizeof(chan->accountcode) - 1);
+		chan->priority = 0;
+		ast_cdr_init(chan->cdr,chan);
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+
+reorder:
+
+	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;
 }
 
 int unload_module(void)
@@ -381,7 +363,7 @@ int usecount(void)
 	return res;
 }
 
-char *key()
+char *key(void)
 {
 	return ASTERISK_GPL_KEY;
 }

apps/app_echo.c

@@ -22,9 +22,6 @@
 #include <string.h>
 #include <stdlib.h>
 
-#include <pthread.h>
-
-
 static char *tdesc = "Simple Echo Application";
 
 static char *app = "Echo";
@@ -52,9 +49,14 @@ static int echo_exec(struct ast_channel *chan, void *data)
 		f = ast_read(chan);
 		if (!f)
 			break;
+		f->delivery.tv_sec = 0;
+		f->delivery.tv_usec = 0;
 		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;

apps/app_enumlookup.c

@@ -0,0 +1,203 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Time of day - Report the time of day
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/options.h>
+#include <asterisk/config.h>
+#include <asterisk/module.h>
+#include <asterisk/enum.h>
+#include <asterisk/utils.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+#include <ctype.h>
+
+static char *tdesc = "ENUM Lookup";
+
+static char *app = "EnumLookup";
+
+static char *synopsis = "Lookup number in ENUM";
+
+static char *descrip = 
+"  EnumLookup(exten):  Looks up an extension via ENUM and sets\n"
+"the variable 'ENUM'. For VoIP URIs this variable will \n"
+"look like 'TECHNOLOGY/URI' with the appropriate technology.\n"
+"Returns -1 on hangup, or 0 on completion regardless of whether the \n"
+"lookup was successful. \n"
+"Currently, the enumservices SIP, H323, IAX, IAX2 and TEL are recognized. \n"
+"A good SIP, H323, IAX or IAX2 entry will result in normal priority handling, \n"
+"whereas a good TEL entry will increase the priority by 51 (if existing).\n"
+"If the lookup was *not* successful and there exists a priority n + 101,\n"
+"then that priority will be taken next.\n" ;
+
+#define ENUM_CONFIG "enum.conf"
+
+static char h323driver[80] = "";
+#define H323DRIVERDEFAULT "H323"
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int enumlookup_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	char tech[80];
+	char dest[80];
+	char tmp[256];
+	char *c,*t;
+	struct localuser *u;
+	if (!data || ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "EnumLookup requires an argument (extension)\n");
+		res = 1;
+	}
+	LOCAL_USER_ADD(u);
+	if (!res) {
+		res = ast_get_enum(chan, data, dest, sizeof(dest), tech, sizeof(tech));
+		printf("ENUM got '%d'\n", res);
+	}
+	LOCAL_USER_REMOVE(u);
+	/* 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 (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 51, chan->callerid))
+					chan->priority += 50;
+				else
+					res = 0;
+			}
+		} else if (!ast_strlen_zero(tech)) {
+			ast_log(LOG_NOTICE, "Don't know how to handle technology '%s'\n", tech);
+			res = 0;
+		}
+	}
+	if (!res) {
+		/* Look for a "busy" place */
+		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+			chan->priority += 100;
+	} else if (res > 0)
+		res = 0;
+	return res;
+}
+
+static int load_config(void)
+{
+	struct ast_config *cfg;
+	char *s;
+
+	cfg = ast_load(ENUM_CONFIG);
+	if (cfg) {
+		if (!(s=ast_variable_retrieve(cfg, "general", "h323driver"))) {
+			strncpy(h323driver, H323DRIVERDEFAULT, sizeof(h323driver) - 1);
+		} else {
+			strncpy(h323driver, s, sizeof(h323driver) - 1);
+		}
+		ast_destroy(cfg);
+		return 0;
+	}
+	ast_log(LOG_NOTICE, "No ENUM Config file, using defaults\n");
+	return 0;
+}
+
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	int res;
+	res = ast_register_application(app, enumlookup_exec, synopsis, descrip);
+	if (res)
+		return(res);
+	if ((res=load_config())) {
+		return(res);
+	}
+	return(0);
+}
+
+int reload(void)
+{
+	return(load_config());
+}
+
+
+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 +1,101 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Eval application
+ *
+ * Copyright (c) 2004 Tilghman Lesher.  All rights reserved.
+ *
+ * Tilghman Lesher <app_eval__v001@the-tilghman.com>
+ *
+ * $Id$
+ *
+ * This code is released by the author with no restrictions on usage.
+ *
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+
+/* 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];
+
+	LOCAL_USER_ADD(u);
+
+	/* 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)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app_eval);
+}
+
+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

@@ -0,0 +1,113 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Exec application
+ *
+ * Copyright (c) 2004 Tilghman Lesher.  All rights reserved.
+ *
+ * Tilghman Lesher <app_exec__v001@the-tilghman.com>
+ *
+ * $Id$
+ *
+ * This code is released by the author with no restrictions on usage.
+ *
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+
+/* Maximum length of any variable */
+#define MAXRESULT	1024
+
+static char *tdesc = "Executes applications";
+
+static char *app_exec = "Exec";
+
+static char *exec_synopsis = "Exec(Appname(arguments))";
+
+static char *exec_descrip =
+"Exec(appname(arguments))\n"
+"  Allows an arbitrary application to be invoked even when not\n"
+"hardcoded into the dialplan.  Returns whatever value the\n"
+"app returns or -2 when the app cannot be found.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int exec_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *s, *appname, *endargs, args[MAXRESULT];
+	struct ast_app *app;
+
+	LOCAL_USER_ADD(u);
+
+	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 = -2;
+				}
+			}
+		} else {
+			ast_log(LOG_ERROR, "Out of memory\n");
+			res = -1;
+		}
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app_exec);
+}
+
+int load_module(void)
+{
+	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;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_festival.c

@@ -5,12 +5,13 @@
  * 
  * Copyright (C) 2002, Christos Ricudis
  *
- * Christos Ricudis <ricudis@paiko.gr>
+ * Christos Ricudis <ricudis@itc.auth.gr>
  *
  * This program is free software, distributed under the terms of
  * the GNU General Public License
  */
 
+#include <sys/types.h>
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
@@ -18,6 +19,8 @@
 #include <asterisk/module.h>
 #include <asterisk/md5.h>
 #include <asterisk/config.h>
+#include <asterisk/utils.h>
+#include <asterisk/lock.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
@@ -33,7 +36,6 @@
 #include <unistd.h>
 #include <fcntl.h>
 #include <ctype.h>
-#include <pthread.h>
 
 
 #define FESTIVAL_CONFIG "festival.conf"
@@ -45,8 +47,9 @@ static char *app = "Festival";
 static char *synopsis = "Say text to the user";
 
 static char *descrip = 
-"  Festival():  Connect to Festival, send the argument, get back the waveform,"
-"play it to the user.\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;
 
@@ -104,6 +107,10 @@ static int send_waveform_to_fd(char *waveform, int length, int fd) {
 
         int res;
         int x;
+#ifdef __PPC__ 
+	char c;
+#endif
+
         res = fork();
         if (res < 0)
                 ast_log(LOG_WARNING, "Fork failed\n");
@@ -113,39 +120,44 @@ static int send_waveform_to_fd(char *waveform, int length, int fd) {
                 if (x != fd)
                         close(x);
         }
+//IAS
+#ifdef __PPC__  
+	for( x=0; x<length; x+=2)
+	{
+		c = *(waveform+x+1);
+		*(waveform+x+1)=*(waveform+x);
+		*(waveform+x)=c;
+	}
+#endif
+	
 	write(fd,waveform,length);
-	write(fd,"a",1);
 	close(fd);
 	exit(0);
 }
 
 
-static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, int length) {
+static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, int length, char *intkeys) {
 	int res=0;
 	int fds[2];
-	int rfds[1 + AST_MAX_FDS];
 	int ms = -1;
 	int pid = -1;
-	int us;
-	int exception;
+	int needed = 0;
 	int owriteformat;
-	struct timeval tv;
-	struct timeval last;
 	struct ast_frame *f;
-	int x;
 	struct myframe {
 		struct ast_frame f;
 		char offset[AST_FRIENDLY_OFFSET];
-		char frdata[160];
+		char frdata[2048];
 	} myf;
-	last.tv_usec = 0;
-	last.tv_sec = 0;
 	
         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);
 	ast_stopstream(chan);
 
 	owriteformat = chan->writeformat;
@@ -160,53 +172,36 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 		pid = res;
 		/* Order is important -- there's almost always going to be mp3...  we want to prioritize the
 		   user */
-		rfds[AST_MAX_FDS] = fds[0];
 		for (;;) {
-			CHECK_BLOCKING(chan);
-			for (x=0;x<AST_MAX_FDS;x++) 
-				rfds[x] = chan->fds[x];
-  			res = ast_waitfor_n_fd(rfds, AST_MAX_FDS+1, &ms, &exception);
-			chan->blocking = 0;
+			ms = 1000;
+			res = ast_waitfor(chan, ms);
 			if (res < 1) {
-				ast_log(LOG_DEBUG, "Hangup detected\n");
 				res = -1;
 				break;
 			}
-			for(x=0;x<AST_MAX_FDS;x++) 
-				if (res == chan->fds[x])
-					break;
-
-			if (x < AST_MAX_FDS) {
-				if (exception)
-					chan->exception = 1;
-				f = ast_read(chan);
-				if (!f) {
-					ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
-					res = -1;
-					break;
-				}
-				if (f->frametype == AST_FRAME_DTMF) {
-					ast_log(LOG_DEBUG, "User pressed a key\n");
+			f = ast_read(chan);
+			if (!f) {
+				ast_log(LOG_WARNING, "Null frame == hangup() detected\n");
+				res = -1;
+				break;
+			}
+			if (f->frametype == AST_FRAME_DTMF) {
+				ast_log(LOG_DEBUG, "User pressed a key\n");
+				if (intkeys && strchr(intkeys, f->subclass)) {
+					res = f->subclass;
 					ast_frfree(f);
-					res = 0;
 					break;
 				}
-				ast_frfree(f);
-			} else if (res == fds[0]) {
-				gettimeofday(&tv, NULL);
-				if (last.tv_sec || last.tv_usec) {
-					/* We should wait at least a frame length */
-					us = sizeof(myf.frdata) / 16 * 1000;
-					/* Subtract 1,000,000 us for each second late we've passed */
-					us -= (tv.tv_sec - last.tv_sec) * 1000000;
-					/* And one for each us late we've passed */
-					us -= (tv.tv_usec - last.tv_usec);
-					/* Sleep that long if needed */
-					if (us > 0)
-						usleep(us);
+			}
+			if (f->frametype == AST_FRAME_VOICE) {
+				/* Treat as a generator */
+				needed = f->samples * 2;
+				if (needed > sizeof(myf.frdata)) {
+					ast_log(LOG_WARNING, "Only able to deliver %d of %d requested samples\n",
+						(int)sizeof(myf.frdata) / 2, needed/2);
+					needed = sizeof(myf.frdata);
 				}
-				last = tv;
-				res = read(fds[0], myf.frdata, sizeof(myf.frdata));
+				res = read(fds[0], myf.frdata, needed);
 				if (res > 0) {
 					myf.f.frametype = AST_FRAME_VOICE;
 					myf.f.subclass = AST_FORMAT_SLINEAR;
@@ -220,20 +215,17 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 						res = -1;
 						break;
 					}
-					if (res < sizeof(myf.frdata)) { // last frame
-						ast_log(LOG_WARNING, "Last frame\n");
+					if (res < needed) { // last frame
+						ast_log(LOG_DEBUG, "Last frame\n");
 						res=0;
 						break;
 					}
 				} else {
-					ast_log(LOG_WARNING, "No more waveform\n");
+					ast_log(LOG_DEBUG, "No more waveform\n");
 					res = 0;
 				}
-			} else {
-				ast_log(LOG_DEBUG, "HuhHHH?\n");
-				res = -1;
-				break;
 			}
+			ast_frfree(f);
 		}
 	}
 	close(fds[0]);
@@ -251,13 +243,14 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 
 
 
-static int festival_exec(struct ast_channel *chan, void *data)
+static int festival_exec(struct ast_channel *chan, void *vdata)
 {
 	int usecache;
 	int res=0;
 	struct localuser *u;
  	struct sockaddr_in serv_addr;
 	struct hostent *serverhost;
+	struct ast_hostent ahp;
 	int fd;
 	FILE *fs;
 	char *host;
@@ -274,15 +267,17 @@ static int festival_exec(struct ast_channel *chan, void *data)
 	int i;
 	struct MD5Context md5ctx;
 	unsigned char MD5Res[16];
-	char MD5Hex[32];
-	char koko[4];
-	char cachefile[MAXFESTLEN];
+	char MD5Hex[33] = "";
+	char koko[4] = "";
+	char cachefile[MAXFESTLEN]="";
 	int readcache=0;
 	int writecache=0;
 	int strln;
-	int fdesc;
+	int fdesc = -1;
 	char buffer[16384];
-	int seekpos;	
+	int seekpos = 0;	
+	char data[256] = "";
+	char *intstr;
 	
 	struct ast_config *cfg;
 	cfg = ast_load(FESTIVAL_CONFIG);
@@ -301,9 +296,7 @@ static int festival_exec(struct ast_channel *chan, void *data)
 	if (!(temp = ast_variable_retrieve(cfg, "general", "usecache"))) {
 		usecache=0;
 	} else {
-		if (strcasecmp(temp,"yes")==0) {
-			usecache=1;
-		}
+		usecache = ast_true(temp);
 	}
 	if (!(cachedir = ast_variable_retrieve(cfg, "general", "cachedir"))) {
 		cachedir = "/tmp/";
@@ -311,29 +304,36 @@ static int festival_exec(struct ast_channel *chan, void *data)
 	if (!(festivalcommand = ast_variable_retrieve(cfg, "general", "festivalcommand"))) {
 		festivalcommand = "(tts_textasterisk \"%s\" 'file)(quit)\n";
 	}
-	
-		
-
-	if (!data) {
+	if (!vdata || ast_strlen_zero(vdata)) {
 		ast_log(LOG_WARNING, "festival requires an argument (text)\n");
+		ast_destroy(cfg);
 		return -1;
 	}
+	strncpy(data, vdata, sizeof(data) - 1);
+	if ((intstr = strchr(data, '|'))) {
+		*intstr = '\0';
+		intstr++;
+		if (!strcasecmp(intstr, "any"))
+			intstr = AST_DIGIT_ANY;
+	}
 	LOCAL_USER_ADD(u);
-	ast_log(LOG_WARNING, "Text passed to festival server : %s\n",(char *)data);
+	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_destroy(cfg);
         	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 = gethostbyname(host);
+	        serverhost = ast_gethostbyname(host, &ahp);
 	        if (serverhost == (struct hostent *)0) {
         	    	ast_log(LOG_WARNING,"festival_client: gethostbyname failed\n");
+			ast_destroy(cfg);
 	            	return -1;
         	}
 	        memmove(&serv_addr.sin_addr,serverhost->h_addr, serverhost->h_length);
@@ -343,6 +343,7 @@ static int festival_exec(struct ast_channel *chan, void *data)
 
 	if (connect(fd, (struct sockaddr *)&serv_addr, sizeof(serv_addr)) != 0) {
 		ast_log(LOG_WARNING,"festival_client: connect to server failed\n");
+		ast_destroy(cfg);
         	return -1;
     	}
     	
@@ -350,37 +351,38 @@ static int festival_exec(struct ast_channel *chan, void *data)
     	MD5Init(&md5ctx);
     	MD5Update(&md5ctx,(unsigned char const *)data,strlen(data));
     	MD5Final(MD5Res,&md5ctx);
-    	strcpy(MD5Hex,"");
+		MD5Hex[0] = '\0';
     	
     	/* Convert to HEX and look if there is any matching file in the cache 
     		directory */
     	for (i=0;i<16;i++) {
-    		sprintf(koko,"%X",MD5Res[i]);
-    		strcat(MD5Hex,koko);
+    		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)) {
-    		sprintf(cachefile,"%s/%s",cachedir,MD5Hex);
+    	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,0);
+    			fdesc=open(cachefile,O_CREAT|O_RDWR,0777);
     			if (fdesc!=-1) {
     				writecache=1;
     				strln=strlen((char *)data);
-    				ast_log(LOG_WARNING,"line length : %d\n",strln);
+    				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_WARNING,"Seek position : %d\n",seekpos);
+				ast_log(LOG_DEBUG,"Seek position : %d\n",seekpos);
     			}
     		} else {
     			read(fdesc,&strln,sizeof(int));
-    			ast_log(LOG_WARNING,"Cache file exists, strln=%d, strlen=%d\n",strln,strlen((char *)data));
+    			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_WARNING,"Size OK\n");
+    				ast_log(LOG_DEBUG,"Size OK\n");
     				read(fdesc,&bigstring,strln);
-	    			if (strcmp(bigstring,data)==0) { 
+	    			bigstring[strln] = 0;
+				if (strcmp(bigstring,data)==0) { 
 	    				readcache=1;
 	    			} else {
 	    				ast_log(LOG_WARNING,"Strings do not match\n");
@@ -394,9 +396,9 @@ static int festival_exec(struct ast_channel *chan, void *data)
 	if (readcache==1) {
 		close(fd);
 		fd=fdesc;
-		ast_log(LOG_WARNING,"Reading from cache...\n");
+		ast_log(LOG_DEBUG,"Reading from cache...\n");
 	} else {
-		ast_log(LOG_WARNING,"Passing text to festival...\n");
+		ast_log(LOG_DEBUG,"Passing text to festival...\n");
     		fs=fdopen(dup(fd),"wb");
 		fprintf(fs,festivalcommand,(char *)data);
 		fflush(fs);
@@ -405,7 +407,7 @@ static int festival_exec(struct ast_channel *chan, void *data)
 	
 	/* Write to cache and then pass it down */
 	if (writecache==1) {
-		ast_log(LOG_WARNING,"Writing result to cache...\n");
+		ast_log(LOG_DEBUG,"Writing result to cache...\n");
 		while ((strln=read(fd,buffer,16384))!=0) {
 			write(fdesc,buffer,strln);
 		}
@@ -415,7 +417,7 @@ static int festival_exec(struct ast_channel *chan, void *data)
 		lseek(fd,seekpos,SEEK_SET);
 	}
 	
-	ast_log(LOG_WARNING,"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 */
@@ -425,15 +427,14 @@ static int festival_exec(struct ast_channel *chan, void *data)
 			n += read(fd,ack+n,3-n);
 		ack[3] = '\0';
 		if (strcmp(ack,"WV\n") == 0) {         /* receive a waveform */
-			ast_log(LOG_WARNING,"Festival WV command");
+			ast_log(LOG_DEBUG,"Festival WV command\n");
 			waveform = socket_receive_file_to_buff(fd,&filesize);
-			send_waveform_to_channel(chan,waveform,filesize);
+			res = send_waveform_to_channel(chan,waveform,filesize, intstr);
 			free(waveform);
-			res=0;
 			break;
 		}
 		else if (strcmp(ack,"LP\n") == 0) {   /* receive an s-expr */
-			ast_log(LOG_WARNING,"Festival LP command");
+			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);
@@ -445,6 +446,7 @@ static int festival_exec(struct ast_channel *chan, void *data)
     		}
 	} while (strcmp(ack,"OK\n") != 0);
 	close(fd);
+	ast_destroy(cfg);
 	LOCAL_USER_REMOVE(u);                                                                                
 	return res;
 

apps/app_flash.c

@@ -21,11 +21,14 @@
 #include <asterisk/image.h>
 #include <asterisk/options.h>
 #include <sys/ioctl.h>
+#ifdef __linux__
 #include <linux/zaptel.h>
+#else
+#include <zaptel.h>
+#endif /* __linux__ */
 #include <string.h>
 #include <errno.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 static char *tdesc = "Flash zap trunk application";
 

apps/app_forkcdr.c

@@ -0,0 +1,90 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Fork CDR application
+ * Copyright Anthony Minessale anthmct@yahoo.com
+ * Development of this app Sponsered/Funded  by TAAN Softworks Corp
+ * 
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/cdr.h>
+#include <asterisk/module.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <pthread.h>
+
+static char *tdesc = "Fork The CDR into 2 separate entities.";
+static char *app = "ForkCDR";
+static char *synopsis = 
+"Forks the Call Data Record";
+static char *descrip = 
+"  ForkCDR():  Causes the Call Data Record to fork an additional\n"
+	"cdr record starting from the time of the fork call\n";
+
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+
+static void ast_cdr_clone(struct ast_cdr *cdr) {
+	struct ast_cdr *newcdr = ast_cdr_alloc();
+	memcpy(newcdr,cdr,sizeof(struct ast_cdr));
+	ast_cdr_append(cdr,newcdr);
+	gettimeofday(&newcdr->start, NULL);
+	memset(&newcdr->answer, 0, sizeof(newcdr->answer));
+	ast_cdr_add_flag(cdr,AST_CDR_FLAG_CHILD|AST_CDR_FLAG_LOCKED);
+}
+
+static void ast_cdr_fork(struct ast_channel *chan) {
+	if(chan && chan->cdr) {
+		ast_cdr_clone(chan->cdr);
+	}
+}
+
+static int forkcdr_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	LOCAL_USER_ADD(u);
+
+	ast_cdr_fork(chan);
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, forkcdr_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_getcpeid.c

@@ -3,9 +3,9 @@
  *
  * Execute arbitrary system commands
  * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999-2005, Digium
  *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
  *
  * This program is free software, distributed under the terms of
  * the GNU General Public License
@@ -24,9 +24,6 @@
 #include <string.h>
 #include <stdlib.h>
 
-#include <pthread.h>
-
-
 static char *tdesc = "Get ADSI CPE ID";
 
 static char *app = "GetCPEID";
@@ -34,9 +31,9 @@ static char *app = "GetCPEID";
 static char *synopsis = "Get ADSI CPE ID";
 
 static char *descrip =
-"  GetCPEID: Obtains and displays CPE ID and other information in order to\n"
-"properly setup zapata.conf for on-hook operations.  Returns -1 on hanup\n"
-"only.";
+"  GetCPEID: Obtains and displays ADSI CPE ID and other information in order\n"
+"to properly setup zapata.conf for on-hook operations.\n"
+"Returns -1 on hangup only.\n";
 
 STANDARD_LOCAL_USER;
 
@@ -70,9 +67,9 @@ static int cpeid_exec(struct ast_channel *chan, void *idata)
 	stuff[2] = data[2];
 	stuff[3] = data[3];
 	memset(data, 0, sizeof(data));
-	strcpy(stuff[0], "** CPE Info **");
-	strcpy(stuff[1], "Identifying CPE...");
-	strcpy(stuff[2], "Please wait...");
+	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, stuff, 0);
@@ -83,8 +80,8 @@ static int cpeid_exec(struct ast_channel *chan, void *idata)
 				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(stuff[1], "Measuring CPE...");
-			strcpy(stuff[2], "Please wait...");
+			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) {
@@ -95,14 +92,14 @@ static int cpeid_exec(struct ast_channel *chan, void *idata)
 		}
 		if (res > -1) {
 			if (gotcpeid)
-				sprintf(stuff[1], "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(stuff[1], "CPEID Unknown");
+				strncpy(stuff[1], "CPEID Unknown", sizeof(data[1]) - 1);
 			if (gotgeometry) 
-				sprintf(stuff[2], "Geom: %dx%d, %d buttons", width, height, buttons);
+				snprintf(stuff[2], sizeof(data[2]), "Geom: %dx%d, %d buttons", width, height, buttons);
 			else
-				strcpy(stuff[2], "Geometry unknown");
-			strcpy(stuff[3], "Press # to exit");
+				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);

apps/app_groupcount.c

@@ -0,0 +1,224 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Group Manipulation Applications
+ *
+ * Copyright (c) 2004 Digium
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/utils.h>
+
+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 *group_count_synopsis = "GetGroupCount([groupname][@category])";
+static char *group_set_synopsis = "SetGroup(groupname[@category])";
+static char *group_check_synopsis = "CheckGroup(max[@category])";
+
+static char *group_count_descrip =
+"GetGroupCount([group][@category])\n"
+"  Calculates the group count for the specified group, or uses\n"
+"the current channel's group if not specifed (and non-empty).\n"
+"Stores result in GROUPCOUNT.  Always returns 0.\n";
+
+static char *group_set_descrip =
+"SetGroup(group)\n"
+"  Sets the channel group to the specified value.  Equivalent to\n"
+"SetVar(GROUP=group).  Always returns 0.\n";
+
+static char *group_check_descrip =
+"CheckGroup(max)\n"
+"  Checks that the current number of total channels in the\n"
+"current channel's group does not exceed 'max'.  If the number\n"
+"does not exceed 'max', we continue to the next step. If the\n"
+"number does in fact exceed max, if priority n+101 exists, then\n"
+"execution continues at that step, otherwise -1 is returned.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+#define DEFAULT_CATEGORY "GROUP"
+
+static int group_get_count(char *group, char *category)
+{
+	struct ast_channel *chan;
+	int count = 0;
+	char *test;
+	if (group && !ast_strlen_zero(group)) {
+		chan = ast_channel_walk_locked(NULL);
+		while(chan) {
+			test = pbx_builtin_getvar_helper(chan, category);
+			if (test && !strcasecmp(test, group))
+				count++;
+			ast_mutex_unlock(&chan->lock);
+			chan = ast_channel_walk_locked(chan);
+		}
+	}
+	return count;
+}
+
+static int group_count_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	int count;
+	struct localuser *u;
+	char *group=NULL;
+	char *cat = NULL;
+	char ret[80]="";
+	char tmp[256]="";
+
+	LOCAL_USER_ADD(u);
+
+	/* Check and parse arguments */
+	if (data && !ast_strlen_zero(data)) {
+		strncpy(tmp, data, sizeof(tmp) - 1);
+		group = tmp;
+		cat = strchr(tmp, '@');
+		if (cat) {
+			*cat = '\0';
+			cat++;
+		}
+	}
+	if (cat)
+		snprintf(ret, sizeof(ret), "GROUP_%s", cat);
+	else
+		strncpy(ret, DEFAULT_CATEGORY, sizeof(ret) - 1);
+
+	if (!group || ast_strlen_zero(group)) {
+		group = pbx_builtin_getvar_helper(chan, ret);
+	}
+	count = group_get_count(group, ret);
+	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;
+	char ret[80] = "";
+	char tmp[256] = "";
+	char *cat=NULL, *group=NULL;
+
+	LOCAL_USER_ADD(u);
+
+	/* Check and parse arguments */
+	if (data && !ast_strlen_zero(data)) {
+		strncpy(tmp, data, sizeof(tmp) - 1);
+		group = tmp;
+		cat = strchr(tmp, '@');
+		if (cat) {
+			*cat = '\0';
+			cat++;
+		}
+	}
+	if (cat)
+		snprintf(ret, sizeof(ret), "GROUP_%s", cat);
+	else
+		strncpy(ret, DEFAULT_CATEGORY, sizeof(ret) - 1);
+
+	if (group && !ast_strlen_zero(group)) {
+		pbx_builtin_setvar_helper(chan, ret, group);
+	} else
+		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 ret[80] = "";
+	char tmp[256] = "";
+	char *cat, *group;
+
+	LOCAL_USER_ADD(u);
+
+	if (data && !ast_strlen_zero(data)) {
+		strncpy(tmp, data, sizeof(tmp) - 1);
+		group = tmp;
+		cat = strchr(tmp, '@');
+		if (cat) {
+			*cat = '\0';
+			cat++;
+		}
+	 	if ((sscanf((char *)tmp, "%i", &max) == 1) && (max > -1)) {
+			if (cat)
+				snprintf(ret, sizeof(ret), "GROUP_%s", cat);
+			else
+				strncpy(ret, DEFAULT_CATEGORY, sizeof(ret) - 1);
+			
+			count = group_get_count(pbx_builtin_getvar_helper(chan, ret), ret);
+			if (count > max) {
+				if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+					chan->priority += 100;
+				else
+					res = -1;
+			}
+		} else
+			ast_log(LOG_WARNING, "CheckGroup requires a positive integer argument (max)\n");
+	} else
+		ast_log(LOG_WARNING, "CheckGroup requires an argument(max)\n");
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+	STANDARD_HANGUP_LOCALUSERS;
+	res = ast_unregister_application(app_group_count);
+	res |= ast_unregister_application(app_group_set);
+	res |= ast_unregister_application(app_group_check);
+	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);
+	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 +1,169 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * HasVoicemail application
+ * Changes Copyright (c) 2004 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>
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR
+ * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO
+ * EVENT SHALL THE CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include <sys/types.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/lock.h>
+#include <asterisk/utils.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+#include <dirent.h>
+
+#include "../astconf.h"
+
+static char *tdesc = "Indicator for whether a voice mailbox has messages in a given folder.";
+static char *app_hasvoicemail = "HasVoicemail";
+static char *hasvoicemail_synopsis = "Conditionally branches to priority + 101";
+static char *hasvoicemail_descrip =
+"HasVoicemail(vmbox[@context][:folder][|varname])\n"
+"  Branches to priority + 101, if there is voicemail in folder indicated."
+"  Optionally sets <varname> to the number of messages in that folder."
+"  Assumes folder of INBOX if not specified.\n";
+
+static char *app_hasnewvoicemail = "HasNewVoicemail";
+static char *hasnewvoicemail_synopsis = "Conditionally branches to priority + 101";
+static char *hasnewvoicemail_descrip =
+"HasNewVoicemail(vmbox[@context][|varname])\n"
+"  Branches to priority + 101, if there is voicemail in folder INBOX."
+"  Optionally sets <varname> to the number of messages in that folder.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int hasvoicemail_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char vmpath[256], *temps, *input, *varname = NULL, *vmbox, *vmfolder = "INBOX", *context = "default";
+	DIR *vmdir;
+	struct dirent *vment;
+	int vmcount = 0;
+
+	if (!data) {
+		ast_log(LOG_WARNING, "HasVoicemail requires an argument (vm-box[@context][:folder]|varname)\n");
+		return -1;
+	}
+	LOCAL_USER_ADD(u);
+
+	input = ast_strdupa((char *)data);
+	if (input) {
+		temps = input;
+		if ((temps = strsep(&input, "|"))) {
+			if (input && !ast_strlen_zero(input))
+				varname = input;
+			input = temps;
+		}
+		if ((temps = strsep(&input, ":"))) {
+			if (input && !ast_strlen_zero(input))
+				vmfolder = input;
+			input = temps;
+		}
+		if ((vmbox = strsep(&input, "@")))
+			if (input && !ast_strlen_zero(input))
+				context = input;
+		if (!vmbox)
+			vmbox = input;
+
+		snprintf(vmpath,sizeof(vmpath), "%s/voicemail/%s/%s/%s", (char *)ast_config_AST_SPOOL_DIR, context, vmbox, vmfolder);
+		if (!(vmdir = opendir(vmpath))) {
+			ast_log(LOG_NOTICE, "Voice mailbox %s at %s does not exist\n", vmbox, vmpath);
+		} else {
+
+			/* No matter what the format of VM, there will always be a .txt file for each message. */
+			while ((vment = readdir(vmdir)))
+				if (!strncmp(vment->d_name + 7,".txt",4))
+					vmcount++;
+			closedir(vmdir);
+		}
+		/* Set the count in the channel variable */
+		if (varname) {
+			char tmp[12];
+			snprintf(tmp, sizeof(tmp), "%d", vmcount);
+			pbx_builtin_setvar_helper(chan, varname, tmp);
+		}
+
+		if (vmcount > 0) {
+			/* Branch to the next extension */
+			if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid)) {
+				chan->priority += 100;
+			} else
+				ast_log(LOG_WARNING, "VM box %s@%s has new voicemail, but extension %s, priority %d doesn't exist\n", vmbox, context, chan->exten, chan->priority + 101);
+		}
+	} else {
+		ast_log(LOG_ERROR, "Out of memory error\n");
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+	STANDARD_HANGUP_LOCALUSERS;
+	res = ast_unregister_application(app_hasvoicemail);
+	res |= ast_unregister_application(app_hasnewvoicemail);
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+	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

@@ -0,0 +1,196 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Stream to an icecast server via ICES (see contrib/asterisk-ices.xml)
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/frame.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <string.h>
+#include <stdio.h>
+#include <signal.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/time.h>
+#include <errno.h>
+#include "../astconf.h"
+
+#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";
+
+static char *synopsis = "Encode and stream using 'ices'";
+
+static char *descrip = 
+"  ICES(config.xml) Streams to an icecast server using ices\n"
+"(available separately).  A configuration file must be supplied\n"
+"for ices (see examples/asterisk-ices.conf).  Returns  -1  on\n"
+"hangup or 0 otherwise.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int icesencode(char *filename, int fd)
+{
+	int res;
+	int x;
+	res = fork();
+	if (res < 0) 
+		ast_log(LOG_WARNING, "Fork failed\n");
+	if (res)
+		return res;
+	dup2(fd, STDIN_FILENO);
+	for (x=STDERR_FILENO + 1;x<256;x++) {
+		if ((x != STDIN_FILENO) && (x != STDOUT_FILENO))
+			close(x);
+	}
+	/* 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\n");
+	return -1;
+}
+
+static int ices_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	int fds[2];
+	int ms = -1;
+	int pid = -1;
+	int flags;
+	int oreadformat;
+	struct timeval last;
+	struct ast_frame *f;
+	char filename[256]="";
+	char *c;
+	last.tv_usec = 0;
+	last.tv_sec = 0;
+	if (!data || !strlen(data)) {
+		ast_log(LOG_WARNING, "ICES requires an argument (configfile.xml)\n");
+		return -1;
+	}
+	if (pipe(fds)) {
+		ast_log(LOG_WARNING, "Unable to create pipe\n");
+		return -1;
+	}
+	flags = fcntl(fds[1], F_GETFL);
+	fcntl(fds[1], F_SETFL, flags | O_NONBLOCK);
+	
+	LOCAL_USER_ADD(u);
+	ast_stopstream(chan);
+
+	if (chan->_state != AST_STATE_UP)
+		res = ast_answer(chan);
+		
+	if (res) {
+		close(fds[0]);
+		close(fds[1]);
+		ast_log(LOG_WARNING, "Answer failed!\n");
+		return -1;
+	}
+
+	oreadformat = chan->readformat;
+	res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
+	if (res < 0) {
+		close(fds[0]);
+		close(fds[1]);
+		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+		return -1;
+	}
+	if (((char *)data)[0] == '/')
+		strncpy(filename, (char *)data, sizeof(filename) - 1);
+	else
+		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_log(LOG_DEBUG, "Hangup detected\n");
+				res = -1;
+				break;
+			}
+			f = ast_read(chan);
+			if (!f) {
+				ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
+				res = -1;
+				break;
+			}
+			if (f->frametype == AST_FRAME_VOICE) {
+				res = write(fds[1], f->data, f->datalen);
+				if (res < 0) {
+					if (errno != EAGAIN) {
+						ast_log(LOG_WARNING, "Write failed to pipe: %s\n", strerror(errno));
+						res = -1;
+						break;
+					}
+				}
+			}
+			ast_frfree(f);
+		}
+	}
+	close(fds[1]);
+	LOCAL_USER_REMOVE(u);
+	if (pid > -1)
+		kill(pid, SIGKILL);
+	if (!res && oreadformat)
+		ast_set_read_format(chan, oreadformat);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, ices_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_image.c

@@ -21,7 +21,6 @@
 #include <asterisk/image.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 static char *tdesc = "Image Transmission Application";
 

apps/app_intercom.c

@@ -24,12 +24,22 @@
 #include <sys/ioctl.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 #include <sys/time.h>
-#include <linux/soundcard.h>
 #include <netinet/in.h>
 
+#if defined(__linux__)
+#include <linux/soundcard.h>
+#elif defined(__FreeBSD__)
+#include <sys/soundcard.h>
+#else
+#include <soundcard.h>
+#endif
+
+#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
@@ -48,26 +58,26 @@ STANDARD_LOCAL_USER;
 
 LOCAL_USER_DECL;
 
-static pthread_mutex_t sound_lock = AST_MUTEX_INITIALIZER;
+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_pthread_mutex_lock(&sound_lock);
+	ast_mutex_lock(&sound_lock);
 	if (sound < 0) {
 		ast_log(LOG_WARNING, "Sound device closed?\n");
-		ast_pthread_mutex_unlock(&sound_lock);
+		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_pthread_mutex_unlock(&sound_lock);
+		ast_mutex_unlock(&sound_lock);
         return -1;
     }
 	res = write(sound, data, len);
-	ast_pthread_mutex_unlock(&sound_lock);
+	ast_mutex_unlock(&sound_lock);
 	return res;
 }
 

apps/app_lookupblacklist.c

@@ -24,7 +24,6 @@
 #include <asterisk/astdb.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 static char *tdesc = "Look up Caller*ID name/number from blacklist database";
 

apps/app_lookupcidname.c

@@ -24,7 +24,6 @@
 #include <asterisk/astdb.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 static char *tdesc = "Look up CallerID Name from local database";
 

apps/app_macro.c

@@ -11,19 +11,20 @@
  * the GNU General Public License
  */
 
+#include <sys/types.h>
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
 #include <asterisk/options.h>
+#include <asterisk/utils.h>
+#include <asterisk/lock.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <stdlib.h>
 
-#include <pthread.h>
-
 #define MAX_ARGS 80
 
 static char *tdesc = "Extension Macros";
@@ -58,7 +59,7 @@ static int macro_exec(struct ast_channel *chan, void *data)
   char varname[80];
   char *oldargs[MAX_ARGS + 1] = { NULL, };
   int argc, x;
-  int res;
+  int res=0;
   char oldexten[256]="";
   int oldpriority;
   char pc[80];
@@ -72,14 +73,14 @@ static int macro_exec(struct ast_channel *chan, void *data)
   char *save_macro_priority;
   char *save_macro_offset;
   
-  if (!data || !strlen(data)) {
+  if (!data || ast_strlen_zero(data)) {
     ast_log(LOG_WARNING, "Invalid Macro incantation\n");
     return 0;
   }
   strncpy(tmp, data, sizeof(tmp) - 1);
   rest = tmp;
   macro = strsep(&rest, "|");
-  if (!macro || !strlen(macro)) {
+  if (!macro || ast_strlen_zero(macro)) {
   	ast_log(LOG_WARNING, "Invalid macro name specified\n");
 	return 0;
   }
@@ -95,8 +96,10 @@ static int macro_exec(struct ast_channel *chan, void *data)
   oldpriority = chan->priority;
   strncpy(oldexten, chan->exten, sizeof(oldexten) - 1);
   strncpy(oldcontext, chan->context, sizeof(oldcontext) - 1);
-  if (!strlen(chan->macrocontext)) {
+  if (ast_strlen_zero(chan->macrocontext)) {
 	strncpy(chan->macrocontext, chan->context, sizeof(chan->macrocontext) - 1);
+	strncpy(chan->macroexten, chan->exten, sizeof(chan->macroexten) - 1);
+	chan->macropriority = chan->priority;
 	setmacrocontext=1;
   }
   argc = 1;
@@ -119,24 +122,26 @@ static int macro_exec(struct ast_channel *chan, void *data)
   pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", NULL);
 
   /* Setup environment for new run */
-  strcpy(chan->exten, "s");
-  strncpy(chan->context, fullmacro, sizeof(chan->context));
+  chan->exten[0] = 's';
+  chan->exten[1] = '\0';
+  strncpy(chan->context, fullmacro, sizeof(chan->context) - 1);
   chan->priority = 1;
 
   while((cur = strsep(&rest, "|")) && (argc < MAX_ARGS)) {
   	/* Save copy of old arguments if we're overwriting some, otherwise
 	   let them pass through to the other macro */
+  	snprintf(varname, sizeof(varname), "ARG%d", argc);
 	oldargs[argc] = pbx_builtin_getvar_helper(chan, varname);
 	if (oldargs[argc])
 		oldargs[argc] = strdup(oldargs[argc]);
-  	snprintf(varname, sizeof(varname), "ARG%d", argc);
 	pbx_builtin_setvar_helper(chan, varname, cur);
 	argc++;
   }
   while(ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->callerid)) {
 	if ((res = ast_spawn_extension(chan, chan->context, chan->exten, chan->priority, chan->callerid))) {
 		/* Something bad happened, or a hangup has been requested. */
-		if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F'))) {
+		if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F')) ||
+		    (res == '*') || (res == '#')) {
 			/* Just return result as to the previous application as if it had been dialed */
 			ast_log(LOG_DEBUG, "Oooh, got something to jump out with ('%c')!\n", res);
 			break;
@@ -162,7 +167,8 @@ static int macro_exec(struct ast_channel *chan, void *data)
 			ast_verbose(VERBOSE_PREFIX_2 "Channel '%s' jumping out of macro '%s'\n", chan->name, macro);
 		break;
 	}
-	if (chan->_softhangup) {
+	/* don't stop executing extensions when we're in "h" */
+	if (chan->_softhangup && strcasecmp(oldexten,"h")) {
 		ast_log(LOG_DEBUG, "Extension %s, priority %d returned normally even though call was hung up\n",
 			chan->exten, chan->priority);
 		goto out;
@@ -188,14 +194,17 @@ out:
   if (save_macro_context) free(save_macro_context);
   pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
   if (save_macro_priority) free(save_macro_priority);
-  if (setmacrocontext)
-  	strcpy(chan->macrocontext, "");
+  if (setmacrocontext) {
+	chan->macrocontext[0] = '\0';
+	chan->macroexten[0] = '\0';
+	chan->macropriority = 0;
+  }
 
   if (!strcasecmp(chan->context, fullmacro)) {
   	/* If we're leaving the macro normally, restore original information */
 	chan->priority = oldpriority;
 	strncpy(chan->context, oldcontext, sizeof(chan->context) - 1);
-	if (!chan->_softhangup) {
+	if (!(chan->_softhangup & AST_SOFTHANGUP_ASYNCGOTO)) {
 		/* Copy the extension, so long as we're not in softhangup, where we could be given an asyncgoto */
 		strncpy(chan->exten, oldexten, sizeof(chan->exten) - 1);
 		if ((offsets = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"))) {

apps/app_milliwatt.c

@@ -23,8 +23,6 @@
 #include <stdlib.h>
 #include <errno.h>
 
-#include <pthread.h>
-
 static char *tdesc = "Digital Milliwatt (mu-law) Test Application";
 
 static char *app = "Milliwatt";
@@ -55,7 +53,7 @@ static void milliwatt_release(struct ast_channel *chan, void *data)
 	return;
 }
 
-static int milliwatt_generate(struct ast_channel *chan, void *data, int len)
+static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int samples)
 {
 	struct ast_frame wf;
 	unsigned char waste[AST_FRIENDLY_OFFSET];
@@ -64,7 +62,7 @@ static int milliwatt_generate(struct ast_channel *chan, void *data, int len)
 
 	if (len > sizeof(buf))
 	{
-		ast_log(LOG_WARNING,"Only doing %d bytes (%d bytes requested)\n",sizeof(buf),len);
+		ast_log(LOG_WARNING,"Only doing %d bytes (%d bytes requested)\n",(int)sizeof(buf),len);
 		len = sizeof(buf);
 	}
 	waste[0] = 0; /* make compiler happy */
@@ -76,6 +74,8 @@ static int milliwatt_generate(struct ast_channel *chan, void *data, int len)
 	wf.datalen = len;
 	wf.samples = wf.datalen;
 	wf.src = "app_milliwatt";
+	wf.delivery.tv_sec = 0;
+	wf.delivery.tv_usec = 0;
 	/* create a buffer containing the digital milliwatt pattern */
 	for(i = 0; i < len; i++)
 	{

apps/app_mp3.c

@@ -3,9 +3,9 @@
  *
  * Silly application to play an MP3 file -- uses mpg123
  * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999-2004, Digium, Inc.
  *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
  *
  * This program is free software, distributed under the terms of
  * the GNU General Public License
@@ -25,9 +25,9 @@
 #include <stdlib.h>
 #include <unistd.h>
 #include <fcntl.h>
-#include <pthread.h>
 #include <sys/time.h>
 
+#define LOCAL_MPG_123 "/usr/local/bin/mpg123"
 #define MPG_123 "/usr/bin/mpg123"
 
 static char *tdesc = "Silly MP3 Application";
@@ -60,14 +60,41 @@ static int mp3play(char *filename, int fd)
 			close(x);
 	}
 	/* Execute mpg123, but buffer if it's a net connection */
-	if (strncmp(filename, "http://", 7)) 
-	    execl(MPG_123, MPG_123, "-q", "-s", "-b", "1024", "--mono", "-r", "8000", filename, NULL);
-	else
-	    execl(MPG_123, MPG_123, "-q", "-s", "--mono", "-r", "8000", filename, NULL);
+	if (!strncasecmp(filename, "http://", 7)) {
+		/* Most commonly installed in /usr/local/bin */
+	    execl(LOCAL_MPG_123, "mpg123", "-q", "-s", "-b", "1024", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
+		/* But many places has it in /usr/bin */
+	    execl(MPG_123, "mpg123", "-q", "-s", "-b", "1024","-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
+		/* As a last-ditch effort, try to use PATH */
+	    execlp("mpg123", "mpg123", "-q", "-s", "-b", "1024",  "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
+	}
+	else {
+		/* Most commonly installed in /usr/local/bin */
+	    execl(MPG_123, "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
+		/* But many places has it in /usr/bin */
+	    execl(LOCAL_MPG_123, "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
+		/* As a last-ditch effort, try to use PATH */
+	    execlp("mpg123", "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
+	}
 	ast_log(LOG_WARNING, "Execute of mpg123 failed\n");
 	return -1;
 }
 
+static int timed_read(int fd, void *data, int datalen, int timeout)
+{
+	int res;
+	struct pollfd fds[1];
+	fds[0].fd = fd;
+	fds[0].events = POLLIN;
+	res = poll(fds, 1, timeout);
+	if (res < 1) {
+		ast_log(LOG_NOTICE, "Poll timed out/errored out with %d\n", res);
+		return -1;
+	}
+	return read(fd, data, datalen);
+	
+}
+
 static int mp3_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
@@ -76,15 +103,14 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 	int ms = -1;
 	int pid = -1;
 	int owriteformat;
-	struct timeval last;
+	int timeout = 2000;
+	struct timeval now, next;
 	struct ast_frame *f;
 	struct myframe {
 		struct ast_frame f;
 		char offset[AST_FRIENDLY_OFFSET];
 		short frdata[160];
 	} myf;
-	last.tv_usec = 0;
-	last.tv_sec = 0;
 	if (!data) {
 		ast_log(LOG_WARNING, "MP3 Playback requires an argument (filename)\n");
 		return -1;
@@ -103,36 +129,36 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 	
+	gettimeofday(&now, NULL);
 	res = mp3play((char *)data, fds[1]);
+	if (!strncasecmp((char *)data, "http://", 7)) {
+		timeout = 10000;
+	}
 	/* Wait 1000 ms first */
-	ms = 1000;
+	next = now;
+	next.tv_sec += 1;
 	if (res >= 0) {
 		pid = res;
 		/* Order is important -- there's almost always going to be mp3...  we want to prioritize the
 		   user */
 		for (;;) {
-			ms = ast_waitfor(chan, ms);
-			if (ms < 0) {
-				ast_log(LOG_DEBUG, "Hangup detected\n");
-				res = -1;
-				break;
-			}
-			if (ms) {
-				f = ast_read(chan);
-				if (!f) {
-					ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
-					res = -1;
-					break;
+			gettimeofday(&now, NULL);
+			ms = (next.tv_sec - now.tv_sec) * 1000;
+			ms += (next.tv_usec - now.tv_usec) / 1000;
+#if 0
+			printf("ms: %d\n", ms);
+#endif			
+			if (ms <= 0) {
+#if 0
+				{
+					static struct timeval last;
+					struct timeval tv;
+					gettimeofday(&tv, NULL);
+					printf("Since last: %ld\n", (tv.tv_sec - last.tv_sec) * 1000 + (tv.tv_usec - last.tv_usec) / 1000);
+					last = tv;
 				}
-				if (f->frametype == AST_FRAME_DTMF) {
-					ast_log(LOG_DEBUG, "User pressed a key\n");
-					ast_frfree(f);
-					res = 0;
-					break;
-				}
-				ast_frfree(f);
-			} else  {
-				res = read(fds[0], myf.frdata, sizeof(myf.frdata));
+#endif
+				res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata), timeout);
 				if (res > 0) {
 					myf.f.frametype = AST_FRAME_VOICE;
 					myf.f.subclass = AST_FORMAT_SLINEAR;
@@ -141,6 +167,8 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 					myf.f.mallocd = 0;
 					myf.f.offset = AST_FRIENDLY_OFFSET;
 					myf.f.src = __PRETTY_FUNCTION__;
+					myf.f.delivery.tv_sec = 0;
+					myf.f.delivery.tv_usec = 0;
 					myf.f.data = myf.frdata;
 					if (ast_write(chan, &myf.f) < 0) {
 						res = -1;
@@ -151,7 +179,36 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 					res = 0;
 					break;
 				}
-				ms = res / 16;
+				next.tv_usec += res / 2 * 125;
+				if (next.tv_usec >= 1000000) {
+					next.tv_usec -= 1000000;
+					next.tv_sec++;
+				}
+#if 0
+				printf("Next: %d\n", ms);
+#endif				
+			} else {
+				ms = ast_waitfor(chan, ms);
+				if (ms < 0) {
+					ast_log(LOG_DEBUG, "Hangup detected\n");
+					res = -1;
+					break;
+				}
+				if (ms) {
+					f = ast_read(chan);
+					if (!f) {
+						ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
+						res = -1;
+						break;
+					}
+					if (f->frametype == AST_FRAME_DTMF) {
+						ast_log(LOG_DEBUG, "User pressed a key\n");
+						ast_frfree(f);
+						res = 0;
+						break;
+					}
+					ast_frfree(f);
+				} 
 			}
 		}
 	}

apps/app_nbscat.c

@@ -0,0 +1,229 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Silly application to play an NBScat file -- uses nbscat8k
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/frame.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <string.h>
+#include <stdio.h>
+#include <signal.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/time.h>
+#include <sys/socket.h>
+
+#define LOCAL_NBSCAT "/usr/local/bin/nbscat8k"
+#define NBSCAT "/usr/bin/nbscat8k"
+
+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"
+"Returns  -1  on\n hangup or 0 otherwise. User can exit by \n"
+"pressing any key\n.";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int NBScatplay(int fd)
+{
+	int res;
+	int x;
+	res = fork();
+	if (res < 0) 
+		ast_log(LOG_WARNING, "Fork failed\n");
+	if (res)
+		return res;
+	dup2(fd, STDOUT_FILENO);
+	for (x=0;x<256;x++) {
+		if (x != STDOUT_FILENO)
+			close(x);
+	}
+	/* Most commonly installed in /usr/local/bin */
+	execl(NBSCAT, "nbscat8k", "-d", (char *)NULL);
+	execl(LOCAL_NBSCAT, "nbscat8k", "-d", (char *)NULL);
+	ast_log(LOG_WARNING, "Execute of nbscat8k failed\n");
+	return -1;
+}
+
+static int timed_read(int fd, void *data, int datalen)
+{
+	int res;
+	struct pollfd fds[1];
+	fds[0].fd = fd;
+	fds[0].events = POLLIN;
+	res = poll(fds, 1, 2000);
+	if (res < 1) {
+		ast_log(LOG_NOTICE, "Selected timed out/errored out with %d\n", res);
+		return -1;
+	}
+	return read(fd, data, 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;
+	int owriteformat;
+	struct timeval now, next;
+	struct ast_frame *f;
+	struct myframe {
+		struct ast_frame f;
+		char offset[AST_FRIENDLY_OFFSET];
+		short frdata[160];
+	} myf;
+	if (socketpair(AF_LOCAL, SOCK_STREAM, 0, fds)) {
+		ast_log(LOG_WARNING, "Unable to create socketpair\n");
+		return -1;
+	}
+	LOCAL_USER_ADD(u);
+	ast_stopstream(chan);
+
+	owriteformat = chan->writeformat;
+	res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+		return -1;
+	}
+	
+	res = NBScatplay(fds[1]);
+	/* Wait 1000 ms first */
+	next = now;
+	next.tv_sec += 1;
+	if (res >= 0) {
+		pid = res;
+		/* Order is important -- there's almost always going to be mp3...  we want to prioritize the
+		   user */
+		for (;;) {
+			gettimeofday(&now, NULL);
+			ms = (next.tv_sec - now.tv_sec) * 1000;
+			ms += (next.tv_usec - now.tv_usec) / 1000;
+#if 0
+			printf("ms: %d\n", ms);
+#endif			
+			if (ms <= 0) {
+#if 0
+				{
+					static struct timeval last;
+					struct timeval tv;
+					gettimeofday(&tv, NULL);
+					printf("Since last: %ld\n", (tv.tv_sec - last.tv_sec) * 1000 + (tv.tv_usec - last.tv_usec) / 1000);
+					last = tv;
+				}
+#endif
+				res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata));
+				if (res > 0) {
+					myf.f.frametype = AST_FRAME_VOICE;
+					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.delivery.tv_sec = 0;
+					myf.f.delivery.tv_usec = 0;
+					myf.f.data = myf.frdata;
+					if (ast_write(chan, &myf.f) < 0) {
+						res = -1;
+						break;
+					}
+				} else {
+					ast_log(LOG_DEBUG, "No more mp3\n");
+					res = 0;
+					break;
+				}
+				next.tv_usec += res / 2 * 125;
+				if (next.tv_usec >= 1000000) {
+					next.tv_usec -= 1000000;
+					next.tv_sec++;
+				}
+#if 0
+				printf("Next: %d\n", ms);
+#endif				
+			} else {
+				ms = ast_waitfor(chan, ms);
+				if (ms < 0) {
+					ast_log(LOG_DEBUG, "Hangup detected\n");
+					res = -1;
+					break;
+				}
+				if (ms) {
+					f = ast_read(chan);
+					if (!f) {
+						ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
+						res = -1;
+						break;
+					}
+					if (f->frametype == AST_FRAME_DTMF) {
+						ast_log(LOG_DEBUG, "User pressed a key\n");
+						ast_frfree(f);
+						res = 0;
+						break;
+					}
+					ast_frfree(f);
+				} 
+			}
+		}
+	}
+	close(fds[0]);
+	close(fds[1]);
+	LOCAL_USER_REMOVE(u);
+	if (pid > -1)
+		kill(pid, SIGKILL);
+	if (!res && owriteformat)
+		ast_set_write_format(chan, owriteformat);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, NBScat_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_osplookup.c

@@ -0,0 +1,277 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Time of day - Report the time of day
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/options.h>
+#include <asterisk/config.h>
+#include <asterisk/module.h>
+#include <asterisk/utils.h>
+#include <asterisk/causes.h>
+#include <asterisk/astosp.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+#include <ctype.h>
+
+static char *tdesc = "OSP Lookup";
+
+static char *app = "OSPLookup";
+static char *app2 = "OSPNext";
+static char *app3 = "OSPFinish";
+
+static char *synopsis = "Lookup number in OSP";
+static char *synopsis2 = "Lookup next OSP entry";
+static char *synopsis3 = "Record OSP entry";
+
+static char *descrip = 
+"  OSPLookup(exten[|provider[|options]]):  Looks up an extension via OSP and sets\n"
+"the variables, where 'n' is the number of the result beginning with 1:\n"
+" ${OSPTECH}:   The technology to use for the call\n"
+" ${OSPDEST}:   The destination to use for the call\n"
+" ${OSPTOKEN}:  The actual OSP token as a string\n"
+" ${OSPHANDLE}: The OSP Handle for anything remaining\n"
+" ${OSPRESULTS}: The number of OSP results total remaining\n"
+"\n"
+"If the lookup was *not* successful and there exists a priority n + 101,\n"
+"then that priority will be taken next.\n" ;
+
+static char *descrip2 = 
+"  OSPNext:  Looks up the next OSP Destination for ${OSPHANDLE}\n"
+"See OSPLookup for more information\n"
+"\n"
+"If the lookup was *not* successful and there exists a priority n + 101,\n"
+"then that priority will be taken next.\n" ;
+
+static char *descrip3 = 
+"  OSPFinish(status):  Records call state for ${OSPHANDLE}, according to\n"
+"status, which should be one of BUSY, CONGESTION, ANSWER, NOANSWER, or NOCHANAVAIL\n"
+"or coincidentally, just what the Dial application stores in its ${DIALSTATUS}\n"
+"\n"
+"If the finishing was *not* successful and there exists a priority n + 101,\n"
+"then that priority will be taken next.\n" ;
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int str2cause(char *cause)
+{
+	if (!strcasecmp(cause, "BUSY"))
+		return AST_CAUSE_BUSY;
+	if (!strcasecmp(cause, "CONGESTION"))
+		return AST_CAUSE_CONGESTION;
+	if (!strcasecmp(cause, "ANSWER"))
+		return AST_CAUSE_NORMAL;
+	if (!strcasecmp(cause, "CANCEL"))
+		return AST_CAUSE_NORMAL;
+	if (!strcasecmp(cause, "NOANSWER"))
+		return AST_CAUSE_NOANSWER;
+	if (!strcasecmp(cause, "NOCHANAVAIL"))
+		return AST_CAUSE_CONGESTION;
+	ast_log(LOG_WARNING, "Unknown cause '%s', using NORMAL\n", cause);
+	return AST_CAUSE_NORMAL;
+}
+
+static int osplookup_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *temp;
+	char *provider, *opts=NULL;
+	struct ast_osp_result result;
+	if (!data || ast_strlen_zero(data) || !(temp = ast_strdupa(data))) {
+		ast_log(LOG_WARNING, "OSPLookup requires an argument (extension)\n");
+		return -1;
+	}
+	provider = strchr(temp, '|');
+	if (provider) {
+		*provider = '\0';
+		provider++;
+		opts = strchr(provider, '|');
+		if (opts) {
+			*opts = '\0';
+			opts++;
+		}
+	}
+	LOCAL_USER_ADD(u);
+	ast_log(LOG_DEBUG, "Whoo hoo, looking up OSP on '%s' via '%s'\n", temp, provider ? provider : "<default>");
+	if ((res = ast_osp_lookup(chan, provider, temp, chan->callerid, &result)) > 0) {
+		char tmp[80];
+		snprintf(tmp, sizeof(tmp), "%d", result.handle);
+		pbx_builtin_setvar_helper(chan, "OSPHANDLE", tmp);
+		pbx_builtin_setvar_helper(chan, "OSPTECH", result.tech);
+		pbx_builtin_setvar_helper(chan, "OSPDEST", result.dest);
+		pbx_builtin_setvar_helper(chan, "OSPTOKEN", result.token);
+		snprintf(tmp, sizeof(tmp), "%d", result.numresults);
+		pbx_builtin_setvar_helper(chan, "OSPRESULTS", tmp);
+	} else {
+		if (!res)
+			ast_log(LOG_NOTICE, "OSP Lookup failed for '%s' (provider '%s')\n", temp, provider ? provider : "<default>");
+		else
+			ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Lookup for '%s' (provider '%s')!\n", chan->name, temp, provider ? provider : "<default>" );
+	}
+	if (!res) {
+		/* Look for a "busy" place */
+		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+			chan->priority += 100;
+	} else if (res > 0)
+		res = 0;
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static int ospnext_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *temp;
+	int cause;
+	struct ast_osp_result result;
+	if (!data || ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "OSPNext should have an argument (cause)\n");
+	}
+	LOCAL_USER_ADD(u);
+	cause = str2cause((char *)data);
+	temp = pbx_builtin_getvar_helper(chan, "OSPHANDLE");
+	result.handle = -1;
+	if (temp && strlen(temp) && (sscanf(temp, "%i", &result.handle) == 1) && (result.handle > -1)) {
+		if ((res = ast_osp_next(&result, cause)) > 0) {
+			char tmp[80];
+			snprintf(tmp, sizeof(tmp), "%d", result.handle);
+			pbx_builtin_setvar_helper(chan, "OSPHANDLE", tmp);
+			pbx_builtin_setvar_helper(chan, "OSPTECH", result.tech);
+			pbx_builtin_setvar_helper(chan, "OSPDEST", result.dest);
+			pbx_builtin_setvar_helper(chan, "OSPTOKEN", result.token);
+			snprintf(tmp, sizeof(tmp), "%d", result.numresults);
+			pbx_builtin_setvar_helper(chan, "OSPRESULTS", tmp);
+		}
+	} else {
+		if (!res) {
+			if (result.handle < 0)
+				ast_log(LOG_NOTICE, "OSP Lookup Next failed for handle '%d'\n", result.handle);
+			else
+				ast_log(LOG_DEBUG, "No OSP handle specified\n");
+		} else
+			ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Next!\n", chan->name);
+	}
+	if (!res) {
+		/* Look for a "busy" place */
+		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+			chan->priority += 100;
+	} else if (res > 0)
+		res = 0;
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static int ospfinished_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *temp;
+	int cause;
+	time_t start=0, duration=0;
+	struct ast_osp_result result;
+	if (!data || ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "OSPFinish should have an argument (cause)\n");
+	}
+	if (chan->cdr) {
+		start = chan->cdr->answer.tv_sec;
+		if (start)
+			duration = time(NULL) - start;
+		else
+			duration = 0;
+	} else
+		ast_log(LOG_WARNING, "OSPFinish called on channel '%s' with no CDR!\n", chan->name);
+	LOCAL_USER_ADD(u);
+	cause = str2cause((char *)data);
+	temp = pbx_builtin_getvar_helper(chan, "OSPHANDLE");
+	result.handle = -1;
+	if (temp && strlen(temp) && (sscanf(temp, "%i", &result.handle) == 1) && (result.handle > -1)) {
+		if (!ast_osp_terminate(result.handle, cause, start, duration)) {
+			pbx_builtin_setvar_helper(chan, "OSPHANDLE", "");
+			res = 1;
+		}
+	} else {
+		if (!res) {
+			if (result.handle > -1)
+				ast_log(LOG_NOTICE, "OSP Finish failed for handle '%d'\n", result.handle);
+			else
+				ast_log(LOG_DEBUG, "No OSP handle specified\n");
+		} else
+			ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Terminate!\n", chan->name);
+	}
+	if (!res) {
+		/* Look for a "busy" place */
+		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+			chan->priority += 100;
+	} else if (res > 0)
+		res = 0;
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+
+int unload_module(void)
+{
+	int res;
+	STANDARD_HANGUP_LOCALUSERS;
+	res = ast_unregister_application(app3);
+	res |= ast_unregister_application(app2);
+	res |= ast_unregister_application(app);
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+	res = ast_register_application(app, osplookup_exec, synopsis, descrip);
+	if (res)
+		return(res);
+	res = ast_register_application(app2, ospnext_exec, synopsis2, descrip2);
+	if (res)
+		return(res);
+	res = ast_register_application(app3, ospfinished_exec, synopsis3, descrip3);
+	if (res)
+		return(res);
+	return(0);
+}
+
+int reload(void)
+{
+	return 0;
+}
+
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
+

apps/app_parkandannounce.c

@@ -14,24 +14,24 @@
  * the GNU General Public License
  */
 
+#include <sys/types.h>
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
 #include <asterisk/channel_pvt.h>
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
-#include <asterisk/parking.h>
+#include <asterisk/features.h>
 #include <asterisk/options.h>
 #include <asterisk/logger.h>
 #include <asterisk/say.h>
+#include <asterisk/lock.h>
 
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <stdlib.h>
 
-#include <pthread.h>
-
 static char *tdesc = "Call Parking and Announce Application";
 
 static char *app = "ParkAndAnnounce";
@@ -41,7 +41,7 @@ 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 over the console.\n"
-"announce template: colon seperated list of files to announce, the word PARKED\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"
@@ -54,170 +54,171 @@ LOCAL_USER_DECL;
 
 static int parkandannounce_exec(struct ast_channel *chan, void *data)
 {
-  int res=0;
-  char *return_context;
-  int l, lot, timeout, 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;
+	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;
-  int outstate;
+	struct ast_channel *dchan;
+	int outstate;
 
-  struct localuser *u;
-  if (!data) {
-    ast_log(LOG_WARNING, "Park requires an argument (parkinglot)\n");
-    return -1;
-  }
+	struct localuser *u;
+	if(!data || (data && !strlen(data))) {
+		ast_log(LOG_WARNING, "ParkAndAnnounce requires arguments: (announce:template|timeout|dial|[return_context])\n");
+		return -1;
+	}
   
-  l=strlen(data)+2;
-  orig_s=malloc(l);
-  s=orig_s;
-  strncpy(s,data,l);
+	l=strlen(data)+2;
+	orig_s=malloc(l);
+	if(!orig_s) {
+		ast_log(LOG_WARNING, "Out of memory\n");
+		return -1;
+	}
+	s=orig_s;
+	strncpy(s,data,l);
 
-  template=strsep(&s,"|");
-  if (! template) {
-    ast_log(LOG_WARNING, "PARK: An announce template must be defined\n");
-    free(orig_s);
-    return -1;
-  }
+	template=strsep(&s,"|");
+	if(! template) {
+		ast_log(LOG_WARNING, "PARK: An announce template must be defined\n");
+		free(orig_s);
+		return -1;
+	}
   
-  timeout = atoi(strsep(&s, "|"));
-  timeout *= 1000;
+	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);
+		return -1;
+	} else {
+		dialtech=strsep(&dial, "/");
+		dialstr=dial;
+		ast_verbose( VERBOSE_PREFIX_3 "Dial Tech,String: (%s,%s)\n", dialtech,dialstr);
+	}
 
-  dial=strsep(&s, "|");
-  if (! dial) {
-    ast_log(LOG_WARNING, "PARK: A dial resouce must be specified i.e: Console/dsp or Zap/g1/5551212\n");
-    free(orig_s);
-    return -1;
-  }
-  else {
-    dialtech=strsep(&dial, "/");
-    dialstr=dial;
-    ast_verbose( VERBOSE_PREFIX_3 "Dial Tech,String: (%s,%s)\n", dialtech,dialstr);
-  }
-
-  return_context = s;
+	return_context = s;
   
-  if (return_context != NULL) {
-    /* set the return context. Code borrowed from the Goto builtin */
+	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);
-      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++;
-  }
+		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);
+		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->callerid);
-    if (!ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->callerid)) {
-      ast_verbose( VERBOSE_PREFIX_3 "Warning: Return Context Invalid, call will return to default|s\n");
-    }
-  }
+	if(option_verbose > 2) {
+		ast_verbose( VERBOSE_PREFIX_3 "Return Context: (%s,%s,%d) ID: %s\n", chan->context,chan->exten, chan->priority, chan->callerid);
+		if(!ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->callerid)) {
+			ast_verbose( VERBOSE_PREFIX_3 "Warning: Return Context Invalid, call will return to default|s\n");
+		}
+	}
   
-  LOCAL_USER_ADD(u);
+	LOCAL_USER_ADD(u);
 
-  /* we are using masq_park here to protect * from touching the channel once we park it.  If the channel comes out of timeout
-     before we are done announcing and the channel is messed with, Kablooeee.  So we use Masq to prevent this.  */
+	/* 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.  */
 
-  ast_masq_park_call(chan, NULL, timeout, &lot);
+	ast_masq_park_call(chan, NULL, timeout, &lot);
 
-  res=-1; 
+	res=-1; 
 
-  ast_verbose( VERBOSE_PREFIX_3 "Call Parking Called, lot: %d, timeout: %d, context: %s\n", lot, timeout, return_context);
+	ast_verbose( VERBOSE_PREFIX_3 "Call Parking Called, lot: %d, timeout: %d, context: %s\n", lot, timeout, return_context);
 
-  /* Now place the call to the extention */
+	/* Now place the call to the extention */
 
-  dchan = ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, dialstr,30000, &outstate, chan->callerid);
+	dchan = ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, dialstr,30000, &outstate, chan->callerid);
 
-  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 {
-      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; 
-  }
+	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 {
+			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; 
+	}
 
-  ast_stopstream(dchan);
+	ast_stopstream(dchan);
 
-  /* now we have the call placed and are ready to play stuff to it */
+	/* now we have the call placed and are ready to play stuff to it */
 
-  ast_verbose(VERBOSE_PREFIX_4 "Announce Template:%s\n", template);
+	ast_verbose(VERBOSE_PREFIX_4 "Announce Template:%s\n", template);
 
-  tpl_working = template;
-  tpl_current=strsep(&tpl_working, ":");
+	tpl_working = template;
+	tpl_current=strsep(&tpl_working, ":");
 
-  while ( tpl_current && looptemp < sizeof(tmp)) {
-    tmp[looptemp]=tpl_current;
-    looptemp++;
-    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_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)
-	dres = ast_waitstream(dchan, "");
-      else {
-	ast_log(LOG_WARNING, "ast_streamfile of %s failed on %s\n", tmp[i], dchan->name);
-	dres = 0;
-      }
-    }
-  }
+	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) {
+				dres = ast_waitstream(dchan, "");
+			} else {
+				ast_log(LOG_WARNING, "ast_streamfile of %s failed on %s\n", tmp[i], dchan->name);
+				dres = 0;
+			}
+		}
+	}
 
-  ast_stopstream(dchan);  
-  ast_hangup(dchan);
+	ast_stopstream(dchan);  
+	ast_hangup(dchan);
 
-  LOCAL_USER_REMOVE(u);
-  free(orig_s);
-  return res;
+	LOCAL_USER_REMOVE(u);
+	free(orig_s);
+	return res;
 }
 
 
@@ -230,8 +231,8 @@ int unload_module(void)
 
 int load_module(void)
 {
-  /*	return ast_register_application(app, park_exec); */
-  return ast_register_application(app, parkandannounce_exec, synopsis, descrip);
+	/* return ast_register_application(app, park_exec); */
+	return ast_register_application(app, parkandannounce_exec, synopsis, descrip);
 }
 
 char *description(void)

apps/app_playback.c

@@ -18,9 +18,9 @@
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
 #include <asterisk/translate.h>
+#include <asterisk/utils.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 static char *tdesc = "Trivial Playback Application";
 
@@ -30,14 +30,14 @@ static char *synopsis = "Play a file";
 
 static char *descrip = 
 "  Playback(filename[|option]):  Plays  back  a  given  filename (do not put\n"
-"extension). Options may also be  included following a pipe symbol. The only\n"
-"defined option at this time is 'skip',  which  causes  the  playback of the\n"
-"message to  be  skipped  if  the  channel is not in the 'up' state (i.e. it\n"
-"hasn't been  answered  yet. If 'skip' is specified, the application will\n"
-"return immediately should the channel not be off hook.  Otherwise, unless\n"
-"'noanswer' is specified, the channel channel will be answered before the sound\n"
-"is played. Not all channels support playing messages while on hook. Returns -1\n"
-"if the channel was hung up, or if the file does not exist. Returns 0 otherwise.\n";
+"extension). Options may also be  included following a pipe symbol. The 'skip'\n"
+"option causes the playback of the message to  be  skipped  if  the  channel\n"
+"is not in the 'up' state (i.e. it hasn't been  answered  yet. If 'skip' is \n"
+"specified, the application will return immediately should the channel not be\n"
+"off hook.  Otherwise, unless 'noanswer' is specified, the channel channel will\n"
+"be answered before the sound is played. Not all channels support playing\n"
+"messages while still hook. Returns -1 if the channel was hung up, or if the\n"
+"file does not exist. Returns 0 otherwise.\n";
 
 STANDARD_LOCAL_USER;
 
@@ -52,7 +52,7 @@ static int playback_exec(struct ast_channel *chan, void *data)
 	int option_skip=0;
 	int option_noanswer = 0;
 	char *stringp;
-	if (!data || !strlen((char *)data)) {
+	if (!data || ast_strlen_zero((char *)data)) {
 		ast_log(LOG_WARNING, "Playback requires an argument (filename)\n");
 		return -1;
 	}

apps/app_privacy.c

@@ -25,7 +25,6 @@
 #include <asterisk/config.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 #define PRIV_CONFIG "privacy.conf"
 
@@ -38,7 +37,7 @@ static char *synopsis = "Require phone number to be entered, if no CallerID sent
 static char *descrip =
   "  PrivacyManager: If no Caller*ID is sent, PrivacyManager answers the\n"
   "channel and asks the caller to enter their 10 digit phone number.\n"
-  "The caller is given 3 attempts.  If after 3 attempts, they do no enter\n"
+  "The caller is given 3 attempts.  If after 3 attempts, they do not enter\n"
   "their 10 digit phone number, and if there exists a priority n + 101,\n"
   "where 'n' is the priority of the current instance, then  the\n"
   "channel  will  be  setup  to continue at that priority level.\n"

apps/app_qcall.c

@@ -51,6 +51,7 @@
  */
  
 #include <asterisk/lock.h>
+#include <asterisk/utils.h>
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
@@ -62,7 +63,6 @@
 #include <unistd.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 #include <sys/types.h>
 #include <sys/stat.h>
 #include <errno.h>
@@ -73,7 +73,7 @@
 #include <sys/file.h>
 #include "../astconf.h"
 
-char qdir[255];
+static char qdir[255];
 static  char *tdesc = "Call from Queue";
 static  pthread_t qcall_thread;
 static int debug = 0;
@@ -123,7 +123,7 @@ pthread_attr_t attr;
 		while((dp = readdir(dirp)) != NULL)
 		   {
 			if (dp->d_name[0] == '.') continue;
-			sprintf(fname,"%s/%s",qdir,dp->d_name);
+			snprintf(fname, sizeof(fname), "%s/%s", qdir, dp->d_name);
 			if (stat(fname,&mystat) == -1)
 			   {
 				perror("app_qcall:stat");
@@ -157,7 +157,7 @@ pthread_attr_t attr;
 			arg = (void *) strdup(fname);
 		        pthread_attr_init(&attr);
  		        pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
-			if (pthread_create(&dialer_thread,&attr,qcall_do,arg) == -1)
+			if (ast_pthread_create(&dialer_thread,&attr,qcall_do,arg) == -1)
 			   {
 				perror("qcall: Cannot create thread");
 				continue;
@@ -172,15 +172,20 @@ pthread_attr_t attr;
 /* single thread with one file (request) to dial */
 static void *qcall_do(void *arg)
 {
-char fname[300],dialstr[300],extstr[300],ident[300],reqinp[300],buf[300];
-char clid[300],*tele,*context;
-FILE *fp;
-int ms = MAXWAITFORANSWER,maxsecs;
-struct ast_channel *channel;
-time_t	t;
+	char fname[300] = "";
+	char dialstr[300];
+	char extstr[300];
+	char ident[300] = "";
+	char reqinp[300] = "";
+	char buf[300];
+	char clid[300],*tele,*context;
+	FILE *fp;
+	int ms = MAXWAITFORANSWER,maxsecs;
+	struct ast_channel *channel;
+	time_t	t;
 
 	  /* get the filename from the arg */
-	strcpy(fname,(char *)arg);
+	strncpy(fname,(char *)arg, sizeof(fname) - 1);
 	free(arg);
 	time(&t);
 	fp = fopen(fname,"r");
@@ -198,8 +203,11 @@ time_t	t;
 		fprintf(stderr,"%s\n",fname);
 		pthread_exit(NULL);
 	   }
-	strcpy(reqinp,"1");  /* default required input for acknowledgement */
-	strcpy(ident, "");	/* default no ident */
+	/* default required input for acknowledgement */
+	reqinp[0] = '1';
+	reqinp[1] = '\0';
+	/* default no ident */
+	ident[0] = '\0';  /* default no ident */
 	if (fscanf(fp,"%s %s %s %d %s %s",dialstr,clid,
 		extstr,&maxsecs,ident,reqinp) < 4)
 	   {
@@ -308,8 +316,8 @@ time_t	t;
 						ast_verbose(VERBOSE_PREFIX_3 "Qcall got accept, now putting through to %s@%s on %s\n",
 							extstr,context,channel->name);
 					if (strlen(ident)) {
-						strcat(ident,"-ok");
-						/* if file existant, play it */
+						strncat(ident,"-ok", sizeof(ident) - strlen(ident) - 1);
+						/* if file existent, play it */
 						if (!ast_streamfile(channel,ident,0))
 						{
 							ast_waitstream(channel,"");
@@ -326,7 +334,7 @@ time_t	t;
 					channel->amaflags = AMAFLAGS;
 #endif
 #ifdef	ACCTCODE
-					strcpy(channel->accountcode,ACCTCODE);
+					strncpy(channel->accountcode, ACCTCODE, sizeof(chan->accountcode) - 1);
 #else
 					channel->accountcode[0] = 0;
 #endif
@@ -335,10 +343,10 @@ time_t	t;
 						time(&channel->whentohangup);
 						channel->whentohangup += maxsecs;
 					   }
-					strcpy(channel->exten,extstr);
-					strcpy(channel->context,context);
+					strncpy(channel->exten, extstr, sizeof(channel->exten) - 1);
+					strncpy(channel->context, context, sizeof(channel->context) - 1);
 					channel->priority = 1;
-					printf("Caller ID is %s\n", channel->callerid);
+					if(debug) printf("Caller ID is %s\n", channel->callerid);
 					ast_pbx_run(channel);
 					pthread_exit(NULL);
 				}
@@ -362,9 +370,9 @@ int unload_module(void)
 
 int load_module(void)
 {
-	snprintf((char *)qdir,sizeof(qdir)-1,"%s/%s",(char *)ast_config_AST_SPOOL_DIR,"qcall");
-	mkdir(qdir,0660);
-	pthread_create(&qcall_thread,NULL,qcall,NULL);
+	snprintf(qdir, sizeof(qdir), "%s/%s", ast_config_AST_SPOOL_DIR, "qcall");
+	mkdir(qdir,0760);
+	ast_pthread_create(&qcall_thread,NULL,qcall,NULL);
 	return 0;
 }
 

apps/app_random.c

@@ -0,0 +1,127 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Random application
+ * 
+ * Copyright (c) 2003-2004 Tilghman Lesher.  All rights reserved.
+ *
+ * Tilghman Lesher <asterisk__app_random__20040111@the-tilghman.com>
+ *
+ * This code is released by the author with no restrictions on usage or distribution.
+ *
+ */
+
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+
+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 int random_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+
+	char *s;
+	char *exten, *pri, *context;
+	char *prob;
+	int probint, priorityint;
+
+	if (!data) {
+		ast_log(LOG_WARNING, "Random requires an argument ([probability]:[[context|]extension|]priority)\n");
+		return -1;
+	}
+	LOCAL_USER_ADD(u);
+	s = ast_strdupa((void *) data);
+
+	prob = strsep(&s,":");
+	if ((!prob) || (sscanf(prob, "%d", &probint) != 1))
+		probint = 0;
+
+	if ((random() % 100) + probint > 100) {
+		context = strsep(&s, "|");
+		exten = strsep(&s, "|");
+		if (!exten) {
+			/* Only a priority */
+			pri = context;
+			exten = NULL;
+			context = NULL;
+		} else {
+			pri = strsep(&s, "|");
+			if (!pri) {
+				pri = exten;
+				exten = context;
+				context = NULL;
+			}
+		}
+		if (!pri) {
+			ast_log(LOG_WARNING, "No label specified\n");
+			LOCAL_USER_REMOVE(u);
+			return -1;
+		} else if (sscanf(pri, "%d", &priorityint) != 1) {
+			ast_log(LOG_WARNING, "Priority '%s' must be a number > 0\n", pri);
+			LOCAL_USER_REMOVE(u);
+			return -1;
+		}
+		/* At this point we have a priority and */
+		/* maybe an extension and a context     */
+		chan->priority = priorityint - 1;
+		if (exten && strcasecmp(exten, "BYEXTENSION"))
+			strncpy(chan->exten, exten, sizeof(chan->exten)-1);
+		if (context)
+			strncpy(chan->context, context, sizeof(chan->context)-1);
+		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)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app_random);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app_random, random_exec, random_synopsis, random_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_read.c

@@ -0,0 +1,144 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Trivial application to read a variable
+ * 
+ * Copyright (C) 2003, Digium
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/app.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/options.h>
+#include <asterisk/utils.h>
+#include <string.h>
+#include <stdlib.h>
+
+static char *tdesc = "Read Variable Application";
+
+static char *app = "Read";
+
+static char *synopsis = "Read a variable";
+
+static char *descrip = 
+"  Read(variable[|filename][|maxdigits][|option])\n\n"
+"Reads a #-terminated string of digits from the user in to the given variable,\n"
+"optionally playing a given filename first.\n"
+"  maxdigits  -- maximum acceptable number of digits. Stops reading after\n"
+"                maxdigits have been entered (without requiring the user to\n"
+"                press the '#' key).\n"
+"                Defaults to 0 - no limit - wait for the user press the '#' key.\n"
+"                Any value below 0 means the same. Max accepted value is 255.\n"
+"  option     -- may be 'skip' to return immediately if the line is not up,\n"
+"                or 'noanswer' to read digits even if the line is not up.\n\n"
+"Returns -1 on hangup or error and 0 otherwise.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int read_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u;
+	char tmp[256];
+	char argdata[256] = "";
+	char *varname;
+	char *filename;
+	char *stringp;
+	char *maxdigitstr;
+	char *options;
+	int option_skip = 0;
+	int option_noanswer = 0;
+	int maxdigits=255;
+	if (!data || ast_strlen_zero((char *)data)) {
+		ast_log(LOG_WARNING, "Read requires an argument (variable)\n");
+		return -1;
+	}
+	strncpy(argdata, (char *)data, sizeof(argdata)-1);
+	stringp=argdata;
+	varname = strsep(&stringp, "|");
+	filename = strsep(&stringp, "|");
+	maxdigitstr = strsep(&stringp,"|");
+	options = strsep(&stringp, "|");
+	if (options && !strcasecmp(options, "skip"))
+		option_skip = 1;
+	if (options && !strcasecmp(options, "noanswer"))
+		option_noanswer = 1;
+	if (!(filename) || ast_strlen_zero(filename)) 
+		filename = NULL;
+	if (maxdigitstr) {
+	    maxdigits = atoi(maxdigitstr);
+	    if ((maxdigits<1) || (maxdigits>255)) {
+    		maxdigits = 255;
+	    } else
+		ast_verbose(VERBOSE_PREFIX_3 "Accepting a maximum of %i digits.\n", maxdigits);
+	}	
+	if (!(varname) || ast_strlen_zero(varname)) {
+		ast_log(LOG_WARNING, "Read requires an variable name\n");
+		return -1;
+	}
+	LOCAL_USER_ADD(u);
+	if (chan->_state != AST_STATE_UP) {
+		if (option_skip) {
+			/* At the user's option, skip if the line is not up */
+			pbx_builtin_setvar_helper(chan, varname, "\0");
+			LOCAL_USER_REMOVE(u);
+			return 0;
+		} else if (!option_noanswer) {
+			/* Otherwise answer unless we're supposed to read while on-hook */
+			res = ast_answer(chan);
+		}
+	}
+	if (!res) {
+		ast_stopstream(chan);
+		res = ast_app_getdata(chan, filename, tmp, maxdigits, 0);
+		if (res > -1) {
+			pbx_builtin_setvar_helper(chan, varname, tmp);
+			ast_verbose(VERBOSE_PREFIX_3 "User entered '%s'\n", tmp);
+			res = 0;
+		} else {
+			ast_verbose(VERBOSE_PREFIX_3 "User disconnected\n");
+		}
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, read_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_record.c

@@ -18,9 +18,9 @@
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
 #include <asterisk/translate.h>
+#include <asterisk/dsp.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 static char *tdesc = "Trivial Record Application";
 
@@ -29,9 +29,19 @@ static char *app = "Record";
 static char *synopsis = "Record to a file";
 
 static char *descrip = 
-"  Record(filename:extension): Records from the  channel into a given\n"
-"filename. If the file exists it will be overwritten. The 'extension'\n"
-"is the extension of the file type  to  be  recorded (wav, gsm, etc).\n"
+"  Record(filename:format|silence[|maxduration][|option])\n\n"
+"Records from the channel into a given filename. If the file exists it will\n"
+"be overwritten.\n"
+"- 'format' is the format of the file type to be recorded (wav, gsm, etc).\n"
+"- 'silence' is the number of seconds of silence to allow before returning.\n"
+"- 'maxduration' is the maximum recording duration in seconds. If missing\n"
+"or 0 there is no maximum.\n"
+"- 'option' may be 'skip' to return immediately if the line is not up,\n"
+"or 'noanswer' to attempt to record even if the line is not up.\n\n"
+"If filename contains '%d', these characters will be replaced with a number\n"
+"incremented by one each time the file is recorded. \n\n"
+"Formats: g723, g729, gsm, h263, ulaw, alaw, vox, wav, WAV\n\n"
+"User can press '#' to terminate the recording and continue to the next priority.\n\n"
 "Returns -1 when the user hangs up.\n";
 
 STANDARD_LOCAL_USER;
@@ -46,58 +56,142 @@ static int record_exec(struct ast_channel *chan, void *data)
 	char fil[256];
 	char tmp[256];
 	char ext[10];
-	char * vdata;  /* Used so I don't have to typecast every use of *data */
+	char *vdata;
 	int i = 0;
 	int j = 0;
-	
+
 	struct ast_filestream *s = '\0';
 	struct localuser *u;
 	struct ast_frame *f = NULL;
 	
-	vdata = data; /* explained above */
-	
+	struct ast_dsp *sildet = NULL;   	/* silence detector dsp */
+	int totalsilence = 0;
+	int dspsilence = 0;
+	int silence = 0;		/* amount of silence to allow */
+	int gotsilence = 0;		/* did we timeout for silence? */
+	char silencestr[5];
+	char durationstr[8];
+	int maxduration = 0;		/* max duration of recording */
+	int gottimeout = 0;		/* did we timeout for maxduration exceeded? */
+	time_t timeout = 0;
+	char option[16];
+	int option_skip = 0;
+	int option_noanswer = 0;
+	int rfmt = 0;
+	int flags;
+	char *end=NULL;
+	char *p=NULL;
+
+
 
 	/* The next few lines of code parse out the filename and header from the input string */
-	if (!vdata) { /* no data implies no filename or anything is present */
+	if (!data) { /* no data implies no filename or anything is present */
 		ast_log(LOG_WARNING, "Record requires an argument (filename)\n");
 		return -1;
 	}
+	vdata = ast_strdupa(data);
 	
+	p = vdata;
+	while(p && (p=strchr(p,':'))) {
+		end=p;
+		if(!strcasecmp(end,":end")) {
+			*end='\0';
+			end++;
+			break;
+		}
+		p++;
+		end=NULL;
+	}
+	
+
 	for (; vdata[i] && (vdata[i] != ':') && (vdata[i] != '|'); i++ ) {
 		if ((vdata[i] == '%') && (vdata[i+1] == 'd')) {
 			percentflag = 1;                      /* the wildcard is used */
 		}
-		
-		if (i == strlen(vdata) ) {
-			ast_log(LOG_WARNING, "No extension found\n");
-			return -1;
-		}
-		fil[i] = vdata[i];
+		if (j < sizeof(fil) - 1)
+			fil[j++] = vdata[i];
 	}
-	fil[i++] = '\0';
+	fil[j] = '\0';
 
-	for (; j < 10 && i < strlen(data); i++, j++)
-		ext[j] = vdata[i];
+	if (vdata[i] != ':') {
+		ast_log(LOG_WARNING, "No extension found\n");
+		return -1;
+	}
+	i++;
+
+	j = 0;
+	for (; vdata[i] && (vdata[i] != '|'); i++)
+		if (j < sizeof(ext) - 1)
+			ext[j++] = vdata[i];
 	ext[j] = '\0';
+
+	if (vdata[i] == '|')
+		i++;
+
+	j = 0;
+	for (; vdata[i] && (vdata[i] != '|'); i++)
+		if (j < sizeof(silencestr) - 1)
+			silencestr[j++] = vdata[i];
+	silencestr[j] = '\0';
+
+	if (j > 0) {
+		silence = atoi(silencestr);
+		if (silence > 0)
+			silence *= 1000;
+	}
+
+	if (vdata[i] == '|')
+		i++;
+
+	j = 0;
+	for (; vdata[i] && (vdata[i] != '|'); i++)
+		if (j < sizeof(durationstr) - 1)
+			durationstr[j++] = vdata[i];
+	durationstr[j] = '\0';
+
+	if (j > 0)
+		maxduration = atoi(durationstr);
+
+	if (vdata[i] == '|')
+		i++;
+
+	j = 0;
+	for (; vdata[i] && (vdata[i] != '|'); i++)
+		if (j < sizeof(option) - 1)
+			option[j++] = vdata[i];
+	option[j] = '\0';
+
+	if (!strcasecmp(option, "skip"))
+		option_skip = 1;
+	if (!strcasecmp(option, "noanswer"))
+		option_noanswer = 1;
+
 	/* done parsing */
-	
+
 	
 	/* these are to allow the use of the %d in the config file for a wild card of sort to
 	  create a new file with the inputed name scheme */
 	if (percentflag) {
 		do {
-			snprintf(tmp, 256, fil, count);
+			snprintf(tmp, sizeof(tmp), fil, count);
 			count++;
 		} while ( ast_fileexists(tmp, ext, chan->language) != -1 );
+		pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
 	} else
-		strncpy(tmp, fil, 256-1);
+		strncpy(tmp, fil, sizeof(tmp)-1);
 	/* end of routine mentioned */
 
 	LOCAL_USER_ADD(u);
 
 	if (chan->_state != AST_STATE_UP) {
-		res = ast_answer(chan); /* Shouldn't need this, but checking to see if channel is already answered
-					 * Theoretically asterisk should already have answered before running the app */
+		if (option_skip) {
+			/* At the user's option, skip if the line is not up */
+			LOCAL_USER_REMOVE(u);
+			return 0;
+		} else if (!option_noanswer) {
+			/* Otherwise answer unless we're supposed to record while on-hook */
+			res = ast_answer(chan);
+		}
 	}
 	
 	if (!res) {
@@ -109,11 +203,40 @@ static int record_exec(struct ast_channel *chan, void *data)
 			ast_log(LOG_WARNING, "ast_streamfile failed on %s\n", chan->name);
 		}
 		ast_stopstream(chan);
+
 		/* The end of beep code.  Now the recording starts */
-		s = ast_writefile( tmp, ext, NULL, O_CREAT|O_TRUNC|O_WRONLY , 0, 0644);
-	
+
+
+		if (silence > 0) {
+	        	rfmt = chan->readformat;
+        		res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
+        		if (res < 0) {
+                		ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
+        		        return -1;
+        		}
+			sildet = ast_dsp_new();
+        		if (!sildet) {
+                		ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
+                		return -1;
+		       	}
+			ast_dsp_set_threshold(sildet, 256);
+		}
+
+
+		flags = end ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
+		s = ast_writefile( tmp, ext, NULL, flags , 0, 0644);
+
+
 		if (s) {
+			if (maxduration > 0)
+				timeout = time(NULL) + (time_t)maxduration;
+
 			while (ast_waitfor(chan, -1) > -1) {
+				if (maxduration > 0 && time(NULL) > timeout) {
+					gottimeout = 1;
+					break;
+				}
+
 				f = ast_read(chan);
 				if (!f) {
 					res = -1;
@@ -126,6 +249,30 @@ static int record_exec(struct ast_channel *chan, void *data)
 						ast_log(LOG_WARNING, "Problem writing frame\n");
 						break;
 					}
+
+					if (silence > 0) {
+						dspsilence = 0;
+						ast_dsp_silence(sildet, f, &dspsilence);
+						if (dspsilence) {
+							totalsilence = dspsilence;
+						} else {
+							totalsilence = 0;
+						}
+	                                	if (totalsilence > silence) {
+        	                        	        /* Ended happily with silence */
+							ast_frfree(f);
+							gotsilence = 1;
+                	                	        break;
+	                     	        	}
+					}
+				}
+				if (f->frametype == AST_FRAME_VIDEO) {
+					res = ast_writestream(s, f);
+					
+					if (res) {
+						ast_log(LOG_WARNING, "Problem writing frame\n");
+						break;
+					}
 				}
 				if ((f->frametype == AST_FRAME_DTMF) &&
 					(f->subclass == '#')) {
@@ -138,6 +285,15 @@ static int record_exec(struct ast_channel *chan, void *data)
 					ast_log(LOG_DEBUG, "Got hangup\n");
 					res = -1;
 			}
+
+			if (gotsilence) {
+				ast_stream_rewind(s, silence-1000);
+				ast_truncstream(s);
+			} else if (!gottimeout) {
+				/* Strip off the last 1/4 second of it */
+				ast_stream_rewind(s, 250);
+				ast_truncstream(s);
+			}
 			ast_closestream(s);
 		} else			
 			ast_log(LOG_WARNING, "Could not create file %s\n", fil);
@@ -145,6 +301,13 @@ static int record_exec(struct ast_channel *chan, void *data)
 		ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
 
 	LOCAL_USER_REMOVE(u);
+	if ((silence > 0) && rfmt) {
+	        res = ast_set_read_format(chan, rfmt);
+        	if (res)
+        	        ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name);
+		if (sildet)
+			ast_dsp_free(sildet);
+	}
 	return res;
 }
 

apps/app_sayunixtime.c

@@ -0,0 +1,141 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * SayUnixTime application
+ * 
+ * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
+ *
+ * Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com>
+ *
+ * This code is released by the author with no restrictions on usage.
+ *
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/say.h>
+
+
+static char *tdesc = "Say time";
+
+static char *app_sayunixtime = "SayUnixTime";
+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"
+"              defaults to now.\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"
+"              defaults to \"ABdY 'digits/at' IMp\"\n"
+"  Returns 0 or -1 on hangup.\n";
+static char *datetime_descrip =
+"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"
+"              defaults to machine default.\n"
+"  format:   a format the time is to be said in.  See voicemail.conf.\n"
+"              defaults to \"ABdY 'digits/at' IMp\"\n"
+"  Returns 0 or -1 on hangup.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int sayunixtime_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *s,*zone=NULL,*timec;
+	time_t unixtime;
+	char *format = "ABdY 'digits/at' IMp";
+	struct timeval tv;
+
+	LOCAL_USER_ADD(u);
+
+	gettimeofday(&tv,NULL);
+	unixtime = (time_t)tv.tv_sec;
+
+	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) {
+		res = ast_answer(chan);
+	}
+	if (!res)
+		res = ast_say_date_with_format(chan, unixtime, AST_DIGIT_ANY, chan->language, format, zone);
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+	STANDARD_HANGUP_LOCALUSERS;
+	res = ast_unregister_application(app_sayunixtime);
+	if (! res)
+		return ast_unregister_application(app_datetime);
+	else
+		return res;
+}
+
+int load_module(void)
+{
+	int res;
+	res = ast_register_application(app_sayunixtime, sayunixtime_exec, sayunixtime_synopsis, sayunixtime_descrip);
+	if (! res)
+		return ast_register_application(app_datetime, sayunixtime_exec, sayunixtime_synopsis, datetime_descrip);
+	else
+		return res;
+}
+
+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

@@ -19,9 +19,10 @@
 #include <asterisk/module.h>
 #include <asterisk/translate.h>
 #include <asterisk/options.h>
+#include <asterisk/utils.h>
+#include <asterisk/app.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 static char *tdesc = "Send DTMF digits Application";
 
@@ -30,8 +31,9 @@ static char *app = "SendDTMF";
 static char *synopsis = "Sends arbitrary DTMF digits";
 
 static char *descrip = 
-"  SendDTMF(digits): Sends DTMF digits on a channel.  Returns 0 on success"
-"or -1 on a hangup.\n";
+"  SendDTMF(digits): Sends DTMF digits on a channel. \n"
+"  Accepted digits: 0-9, *#abcd\n"
+" Returns 0 on success or -1 on a hangup.\n";
 
 STANDARD_LOCAL_USER;
 
@@ -42,30 +44,13 @@ static int senddtmf_exec(struct ast_channel *chan, void *data)
 	int res = 0;
 	struct localuser *u;
 	char *digits = data;
-	struct ast_frame f;
-	int x;
-	if (!digits || !strlen(digits)) {
-		ast_log(LOG_WARNING, "SendDTMF requires an argument (digits)\n");
+
+	if (!digits || ast_strlen_zero(digits)) {
+		ast_log(LOG_WARNING, "SendDTMF requires an argument (digits or *#aAbBcCdD)\n");
 		return -1;
 	}
 	LOCAL_USER_ADD(u);
-	for (x=0;x<strlen(digits);x++) {
-		memset(&f, 0, sizeof(f));
-		f.frametype = AST_FRAME_DTMF;
-		f.subclass = digits[x];
-		f.src = "app_senddtmf";
-		res = ast_write(chan, &f);
-		if (res)
-			break;
-		/* Wait 250ms */
-		res = ast_safe_sleep(chan, 250);
-		if (res)
-			break;
-	}
-	if (!res)
-		if (option_verbose > 2)
-			ast_verbose(VERBOSE_PREFIX_3 "Sent digit string '%s' on %s\n", digits, chan->name);
-			
+	res = ast_dtmf_stream(chan,NULL,digits,250);
 	LOCAL_USER_REMOVE(u);
 	return res;
 }

apps/app_sendtext.c

@@ -0,0 +1,94 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * App to transmit a text message
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/channel_pvt.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/image.h>
+#include <string.h>
+#include <stdlib.h>
+
+static char *tdesc = "Send Text Applications";
+
+static char *app = "SendText";
+
+static char *synopsis = "Send a Text Message";
+
+static char *descrip = 
+"  SendText(text): Sends text to client.  If the client\n"
+"does not support text transport, and  there  exists  a  step  with\n"
+"priority  n + 101,  then  execution  will  continue  at that step.\n"
+"Otherwise, execution will continue at  the  next  priority  level.\n"
+"SendText only returns 0  if  the  text  was  sent  correctly  or  if\n"
+"the channel  does  not  support text transport,  and -1 otherwise.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int sendtext_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u;
+	if (!data || !strlen((char *)data)) {
+		ast_log(LOG_WARNING, "SendText requires an argument (text)\n");
+		return -1;
+	}
+	LOCAL_USER_ADD(u);
+	ast_mutex_lock(&chan->lock);
+	if (!chan->pvt->send_text) {
+		ast_mutex_unlock(&chan->lock);
+		/* Does not support transport */
+		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+			chan->priority += 100;
+		LOCAL_USER_REMOVE(u);
+		return 0;
+	}
+	ast_mutex_unlock(&chan->lock);
+	res = ast_sendtext(chan, (char *)data);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, sendtext_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_setcallerid.c

@@ -19,9 +19,82 @@
 #include <asterisk/module.h>
 #include <asterisk/translate.h>
 #include <asterisk/image.h>
+#include <asterisk/callerid.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
+
+static char *app2 = "SetCallerPres";
+
+static char *synopsis2 = "Set CallerID Presentation";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static struct {
+	int val;
+	char *name;
+} preses[] = {
+	{  AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED, "allowed_not_screened" },
+	{  AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN, "allowed_passed_screen" },
+	{  AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN, "allowed_failed_screen" },
+	{  AST_PRES_ALLOWED_NETWORK_NUMBER, "allowed" },
+	{  AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED	, "prohib_not_screened" },
+	{  AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN, "prohib_passed_screen" },
+	{  AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN, "prohib_failed_screen" },
+	{  AST_PRES_PROHIB_NETWORK_NUMBER, "prohib" },
+	{  AST_PRES_NUMBER_NOT_AVAILABLE, "unavailable" },
+};
+
+static char *descrip2 = 
+"  SetCallerPres(presentation): Set Caller*ID presentation on\n"
+"a call to a new value.  Sets ANI as well if a flag is used.\n"
+"Always returns 0.  Valid presentations are:\n"
+"\n"
+"      allowed_not_screened    : Presentation Allowed, Not Screened\n"
+"      allowed_passed_screen   : Presentation Allowed, Passed Screen\n" 
+"      allowed_failed_screen   : Presentation Allowed, Failed Screen\n" 
+"      allowed                 : Presentation Allowed, Network Number\n"
+"      prohib_not_screened     : Presentation Prohibited, Not Screened\n" 
+"      prohib_passed_screen    : Presentation Prohibited, Passed Screen\n"
+"      prohib_failed_screen    : Presentation Prohibited, Failed Screen\n"
+"      prohib                  : Presentation Prohibited, Network Number\n"
+"      unavailable             : Number Unavailable\n"
+"\n"
+;
+
+static int setcallerid_pres_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	char tmp[256] = "";
+	struct localuser *u;
+	int x;
+	char *opts;
+	int pres = -1;
+	if (data)
+		strncpy(tmp, (char *)data, sizeof(tmp) - 1);
+	opts = strchr(tmp, '|');
+	if (opts) {
+		*opts = '\0';
+		opts++;
+	}
+	for (x=0;x<sizeof(preses) / sizeof(preses[0]);x++) {
+		if (!strcasecmp(preses[x].name, tmp)) {
+			pres = preses[x].val;
+			break;
+		}
+	}
+	if (pres < 0) {
+		ast_log(LOG_WARNING, "'%s' is not a valid presentation (see 'show application SetCallerPres')\n", tmp);
+		return 0;
+	}
+	LOCAL_USER_ADD(u);
+	chan->callingpres = pres;
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+
 
 static char *tdesc = "Set CallerID Application";
 
@@ -33,10 +106,6 @@ static char *descrip =
 "  SetCallerID(clid[|a]): Set Caller*ID on a call to a new\n"
 "value.  Sets ANI as well if a flag is used.  Always returns 0\n";
 
-STANDARD_LOCAL_USER;
-
-LOCAL_USER_DECL;
-
 static int setcallerid_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
@@ -62,11 +131,13 @@ static int setcallerid_exec(struct ast_channel *chan, void *data)
 int unload_module(void)
 {
 	STANDARD_HANGUP_LOCALUSERS;
+	ast_unregister_application(app2);
 	return ast_unregister_application(app);
 }
 
 int load_module(void)
 {
+	ast_register_application(app2, setcallerid_pres_exec, synopsis2, descrip2);
 	return ast_register_application(app, setcallerid_exec, synopsis, descrip);
 }
 

apps/app_setcdruserfield.c

@@ -0,0 +1,165 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Applictions connected with CDR engine
+ * 
+ * Copyright (C) 2003, Digium
+ *
+ * Justin Huff <jjhuff@mspin.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <sys/types.h>
+#include <asterisk/channel.h>
+#include <asterisk/cdr.h>
+#include <asterisk/module.h>
+#include <asterisk/pbx.h>
+#include <asterisk/logger.h>
+#include <asterisk/config.h>
+#include <asterisk/manager.h>
+#include <asterisk/utils.h>
+#include <stdlib.h>
+#include <string.h>
+
+
+static char *tdesc = "CDR user field apps";
+
+static char *setcdruserfield_descrip = 
+               "[Synopsis]\n"
+               "SetCDRUserField(value)\n\n"
+               "[Description]\n"
+               "SetCDRUserField(value): Set the CDR 'user field' to value\n"
+               "       The Call Data Record (CDR) user field is an extra field you\n"
+               "       can use for data not stored anywhere else in the record.\n"
+               "       CDR records can be used for billing or storing other arbitrary data\n"
+               "       (I.E. telephone survey responses)\n"
+               "       Also see AppendCDRUserField().\n"
+               "       Always returns 0\n";
+
+		
+static char *setcdruserfield_app = "SetCDRUserField";
+static char *setcdruserfield_synopsis = "Set the CDR user field";
+
+static char *appendcdruserfield_descrip = 
+               "[Synopsis]\n"
+               "AppendCDRUserField(value)\n\n"
+               "[Description]\n"
+               "AppendCDRUserField(value): Append value to the CDR user field\n"
+               "       The Call Data Record (CDR) user field is an extra field you\n"
+               "       can use for data not stored anywhere else in the record.\n"
+               "       CDR records can be used for billing or storing other arbitrary data\n"
+               "       (I.E. telephone survey responses)\n"
+               "       Also see SetCDRUserField().\n"
+               "       Always returns 0\n";
+		
+static char *appendcdruserfield_app = "AppendCDRUserField";
+static char *appendcdruserfield_synopsis = "Append to the CDR user field";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int action_setcdruserfield(struct mansession *s, struct message *m)
+{
+	struct ast_channel *c = NULL;
+	char *userfield = astman_get_header(m, "UserField");
+	char *channel = astman_get_header(m, "Channel");
+	char *append = astman_get_header(m, "Append");
+
+	if (ast_strlen_zero(channel)) {
+		astman_send_error(s, m, "No Channel specified");
+		return 0;
+	}
+	if (ast_strlen_zero(userfield)) {
+		astman_send_error(s, m, "No UserField specified");
+		return 0;
+	}
+	c = ast_channel_walk_locked(NULL);
+	while (c) {
+		if (!strcasecmp(c->name, channel))
+			break;
+		ast_mutex_unlock(&c->lock);
+		c = ast_channel_walk_locked(c);
+	}
+	if (!c) {
+		astman_send_error(s, m, "No such channel");
+		return 0;
+	}
+	if (ast_true(append))
+		ast_cdr_appenduserfield(c, userfield);
+	else
+		ast_cdr_setuserfield(c, userfield);
+	ast_mutex_unlock(&c->lock);
+	astman_send_ack(s, m, "CDR Userfield Set");
+	return 0;
+}
+
+static int setcdruserfield_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	int res = 0;
+	
+	LOCAL_USER_ADD(u)
+	if (chan->cdr && data) 
+	{
+		ast_cdr_setuserfield(chan, (char*)data);
+	}
+
+	LOCAL_USER_REMOVE(u);
+	
+	return res;
+}
+
+static int appendcdruserfield_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	int res = 0;
+	
+	LOCAL_USER_ADD(u)
+	if (chan->cdr && data) 
+	{
+		ast_cdr_appenduserfield(chan, (char*)data);
+	}
+
+	LOCAL_USER_REMOVE(u);
+	
+	return res;
+}
+
+int unload_module(void)
+{
+	int res;
+	STANDARD_HANGUP_LOCALUSERS;
+	res = ast_unregister_application(setcdruserfield_app);
+	res |= ast_unregister_application(appendcdruserfield_app);
+	ast_manager_unregister("SetCDRUserField");
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+	res = ast_register_application(setcdruserfield_app, setcdruserfield_exec, setcdruserfield_synopsis, setcdruserfield_descrip);
+	res |= ast_register_application(appendcdruserfield_app, appendcdruserfield_exec, appendcdruserfield_synopsis, appendcdruserfield_descrip);
+	ast_manager_register("SetCDRUserField", EVENT_FLAG_CALL, action_setcdruserfield, "Set the CDR UserField");
+	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_setcidname.c

@@ -20,9 +20,9 @@
 #include <asterisk/translate.h>
 #include <asterisk/image.h>
 #include <asterisk/callerid.h>
+#include <asterisk/utils.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 static char *tdesc = "Set CallerID Name";
 
@@ -63,17 +63,17 @@ static int setcallerid_exec(struct ast_channel *chan, void *data)
 		strncpy(oldcid, chan->callerid, sizeof(oldcid) - 1);
 		ast_callerid_parse(oldcid, &n, &l);
 		n = tmp;
-		if (strlen(n)) {
-			if (l && strlen(l))
+		if (!ast_strlen_zero(n)) {
+			if (l && !ast_strlen_zero(l))
 				snprintf(newcid, sizeof(newcid), "\"%s\" <%s>", n, l);
 			else
 				strncpy(newcid, tmp, sizeof(newcid) - 1);
-		} else if (l && strlen(l)) {
+		} else if (l && !ast_strlen_zero(l)) {
 			strncpy(newcid, l, sizeof(newcid) - 1);
 		}
 	} else
-		strncpy(newcid, tmp, sizeof(newcid));
-	ast_set_callerid(chan, strlen(newcid) ? newcid : NULL, anitoo);
+		strncpy(newcid, tmp, sizeof(newcid) - 1);
+	ast_set_callerid(chan, !ast_strlen_zero(newcid) ? newcid : NULL, anitoo);
 	LOCAL_USER_REMOVE(u);
 	return res;
 }

apps/app_setcidnum.c

@@ -0,0 +1,108 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * App to set callerid
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ * Oliver Daudey <traveler@xs4all.nl>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/image.h>
+#include <asterisk/callerid.h>
+#include <asterisk/utils.h>
+#include <string.h>
+#include <stdlib.h>
+
+static char *tdesc = "Set CallerID Number";
+
+static char *app = "SetCIDNum";
+
+static char *synopsis = "Set CallerID Number";
+
+static char *descrip = 
+"  SetCIDNum(cnum[|a]): Set Caller*ID Number on a call to a new\n"
+"value, while preserving the original Caller*ID name.  This is\n"
+"useful for providing additional information to the called\n"
+"party. Sets ANI as well if a flag is used.  Always returns 0\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int setcallerid_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	char tmp[256] = "";
+	char oldcid[256] = "", *l, *n;
+	char newcid[256] = "";
+	struct localuser *u;
+	char *opt;
+	int anitoo = 0;
+	if (data)
+		strncpy(tmp, (char *)data, sizeof(tmp) - 1);
+	opt = strchr(tmp, '|');
+	if (opt) {
+		*opt = '\0';
+		opt++;
+		if (*opt == 'a')
+			anitoo = 1;
+	}
+	LOCAL_USER_ADD(u);
+	if (chan->callerid) {
+		strncpy(oldcid, chan->callerid, sizeof(oldcid) - 1);
+		ast_callerid_parse(oldcid, &n, &l);
+		l = tmp;
+		if (!ast_strlen_zero(l)) {
+			if (n && !ast_strlen_zero(n))
+				snprintf(newcid, sizeof(newcid), "\"%s\" <%s>", n, l);
+			else
+				strncpy(newcid, tmp, sizeof(newcid) - 1);
+		} else if (n && !ast_strlen_zero(n)) {
+			strncpy(newcid, n, sizeof(newcid) - 1);
+		}
+	} else
+		strncpy(newcid, tmp, sizeof(newcid) - 1);
+	ast_set_callerid(chan, !ast_strlen_zero(newcid) ? newcid : NULL, anitoo);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, setcallerid_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_skel.c

@@ -16,17 +16,16 @@
 #include <asterisk/channel.h>
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
+#include <asterisk/lock.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
-#include <stdlib.h>
-
-#include <pthread.h>
-
 
 static char *tdesc = "Trivial skeleton Application";
-
 static char *app = "skel";
+static char *synopsis = 
+"  This is a skeleton application that shows you the basic structure to create your\n"
+"own asterisk applications.\n";
 
 STANDARD_LOCAL_USER;
 
@@ -54,7 +53,7 @@ int unload_module(void)
 
 int load_module(void)
 {
-	return ast_register_application(app, skel_exec);
+	return ast_register_application(app, skel_exec, synopsis, tdesc);
 }
 
 char *description(void)

apps/app_softhangup.c

@@ -11,19 +11,18 @@
  * the GNU General Public License
  */
 
+#include <sys/types.h>
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
+#include <asterisk/lock.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <stdlib.h>
 
-#include <pthread.h>
-
-
 static char *synopsis = "Soft Hangup Application";
 
 static char *tdesc = "Hangs up the requested channel";
@@ -46,13 +45,15 @@ static int softhangup_exec(struct ast_channel *chan, void *data)
 		return 0;
 	}
 	LOCAL_USER_ADD(u);
-	c = ast_channel_walk(NULL);
+	c = ast_channel_walk_locked(NULL);
 	while (c) {
 		if (!strcasecmp(c->name, data)) {
 			ast_softhangup(c, AST_SOFTHANGUP_EXPLICIT);
+			ast_mutex_unlock(&c->lock);
 			break;
 		}
-		c = ast_channel_walk(c);
+		ast_mutex_unlock(&c->lock);
+		c = ast_channel_walk_locked(c);
 	}
 	LOCAL_USER_REMOVE(u);
 

apps/app_sql_postgres.c

@@ -5,7 +5,7 @@
  * 
  * Copyright (C) 2002, Christos Ricudis
  *
- * Christos Ricudis <ricudis@paiko.gr>
+ * Christos Ricudis <ricudis@itc.auth.gr>
  *
  * This program is free software, distributed under the terms of
  * the GNU General Public License
@@ -26,10 +26,9 @@
 #include <sys/types.h>
 #include <stdio.h>
 #include <unistd.h>
-#include <pthread.h>
 #include "libpq-fe.h"
 
-
+#define EXTRA_LOG 0
 
 
 static char *tdesc = "Simple PostgreSQL Interface";
@@ -39,57 +38,67 @@ static char *app = "PGSQL";
 static char *synopsis = "Do several SQLy things";
 
 static char *descrip = 
-"  PGSQL():  Do several SQLy things\n";
+"PGSQL():  Do several SQLy things\n"
+"Syntax:\n"
+"  PGSQL(Connect var option-string)\n"
+"    Connects to a database.  Option string contains standard PostgreSQL\n"
+"    parameters like host=, dbname=, user=.  Connection identifer returned\n"
+"    in ${var}\n"
+"  PGSQL(Query var ${connection_identifier} query-string)\n"
+"    Executes standard SQL query contained in query-string using established\n"
+"    connection identified by ${connection_identifier}. Reseult of query is\n"
+"    is stored in ${var}.\n"
+"  PGSQL(Fetch statusvar ${result_identifier} var1 var2 ... varn)\n"
+"    Fetches a single row from a result set contained in ${result_identifier}.\n"
+"    Assigns returned fields to ${var1} ... ${varn}.  ${statusvar} is set TRUE\n"
+"    if additional rows exist in reseult set.\n"
+"  PGSQL(Clear ${result_identifier})\n"
+"    Frees memory and datastructures associated with result set.\n" 
+"  PGSQL(Disconnect ${connection_identifier})\n"
+"    Disconnects from named connection to PostgreSQL.\n" ;
 
 /*
 
 Syntax of SQL commands : 
 
-	Connect #var option-string
+	Connect var option-string
 	
 	Connects to a database using the option-string and stores the 
-	connection identifier in $var
+	connection identifier in ${var}
 	
 	
-	Query var connection-identifier query-string
+	Query var ${connection_identifier} query-string
 	
 	Submits query-string to database backend and stores the result
 	identifier in ${var}
 	
 	
-	Fetch statusvar result-identifier var1 var2 var3 ... varn
+	Fetch statusvar ${result_identifier} var1 var2 var3 ... varn
 	
 	Fetches a row from the query and stores end-of-table status in 
 	${statusvar} and columns in ${var1}..${varn}
 	
 	
-	Clear result-identifier
+	Clear ${result_identifier}
 
-	Clears data structures associated with result-identifier
+	Clears data structures associated with ${result_identifier}
 	
 	
-	Disconnect connection-identifier
+	Disconnect ${connection_identifier}
 	
 	Disconnects from named connection
 	
 	
 EXAMPLES OF USE : 
 
-(
- $2 = Connection Identifier
- $3 = Result Identifier
- $4 = Fetch Status Identifier (0 = no more rows)
- $5, $6 = Data variables
-)
-
-exten => s,2,PGSQL,"Connect connid host=localhost user=asterisk dbname=credit";
-exten => s,3,PGSQL,"Query resultid ${connid} SELECT username,credit FROM credit WHERE callerid=${callerid}";
-exten => s,4,PGSQL,"Fetch fetchid ${resultid} datavar1 datavar2";
-exten => s,5,GotoIf,"${fetchid}=1?s|6:s|8";
-exten => s,6,blablabla ${datavar1} ${datavar2}	(does blablabla, datavar1 = username, datavar2 = credit);
-exten => s,7,Goto,s|4
-exten => s,8,PGSQL,"Clear ${resultid}";
-exten => s,9,PGSQL,"Disconnect ${connid}";
+exten => s,2,PGSQL(Connect connid host=localhost user=asterisk dbname=credit)
+exten => s,3,PGSQL(Query resultid ${connid} SELECT username,credit FROM credit WHERE callerid=${CALLERIDNUM})
+exten => s,4,PGSQL(Fetch fetchid ${resultid} datavar1 datavar2)
+exten => s,5,GotoIf(${fetchid}?6:8)
+exten => s,6,Festival("User ${datavar1} currently has credit balance of ${datavar2} dollars.")	
+exten => s,7,Goto(s,4)
+exten => s,8,PGSQL(Clear ${resultid})
+exten => s,9,PGSQL(Disconnect ${connid})
 
 */
 
@@ -97,8 +106,6 @@ STANDARD_LOCAL_USER;
 
 LOCAL_USER_DECL;
 
-extern void pbx_builtin_setvar_helper(struct ast_channel *chan, char *name, char *value); 
-
 #define AST_PGSQL_ID_DUMMY 0
 #define AST_PGSQL_ID_CONNID 1
 #define AST_PGSQL_ID_RESID 2
@@ -201,8 +208,8 @@ static int del_identifier(int identifier,int identifier_type) {
 
 static int aPGSQL_connect(struct ast_channel *chan, void *data) {
 	
-	char *s1,*s4;
-	char s[100];
+	char *s1;
+	char s[100] = "";
 	char *optionstring;
 	char *var;
 	int l;
@@ -215,7 +222,7 @@ static int aPGSQL_connect(struct ast_channel *chan, void *data) {
 	res=0;
 	l=strlen(data)+2;
 	s1=malloc(l);
-	strncpy(s1,data,l);
+	strncpy(s1, data, l -1);
 	stringp=s1;
 	strsep(&stringp," "); // eat the first token, we already know it :P 
 	var=strsep(&stringp," ");
@@ -229,8 +236,7 @@ static int aPGSQL_connect(struct ast_channel *chan, void *data) {
         } else {
         	ast_log(LOG_WARNING,"adding identifier\n");
 		id=add_identifier(AST_PGSQL_ID_CONNID,karoto);
-		s4=&s[0];
-		sprintf(s4,"%d",id);
+		snprintf(s, sizeof(s), "%d", id);
 		pbx_builtin_setvar_helper(chan,var,s);
 	}
  	
@@ -240,8 +246,9 @@ static int aPGSQL_connect(struct ast_channel *chan, void *data) {
 
 static int aPGSQL_query(struct ast_channel *chan, void *data) {
 	
-	char *s1,*s2,*s3,*s4,*s5;
-	char s[100];
+
+	char *s1,*s2,*s3,*s4;
+	char s[100] = "";
 	char *querystring;
 	char *var;
 	int l;
@@ -256,7 +263,7 @@ static int aPGSQL_query(struct ast_channel *chan, void *data) {
 	l=strlen(data)+2;
 	s1=malloc(l);
 	s2=malloc(l);
-	strcpy(s1,data);
+	strncpy(s1, data, l - 1);
 	stringp=s1;
 	strsep(&stringp," "); // eat the first token, we already know it :P 
 	s3=strsep(&stringp," ");
@@ -285,21 +292,21 @@ static int aPGSQL_query(struct ast_channel *chan, void *data) {
 		}
 		nres=PQnfields(PGSQLres); 
 		id1=add_identifier(AST_PGSQL_ID_RESID,PGSQLres);
-		s5=&s[0];
-		sprintf(s5,"%d",id1);
+		snprintf(s, sizeof(s), "%d", id1);
 		pbx_builtin_setvar_helper(chan,var,s);
 	 	break;
 	}
 	
 	free(s1);
 	free(s2);
+
 	return(res);
 }
 
 
 static int aPGSQL_fetch(struct ast_channel *chan, void *data) {
 	
-	char *s1,*s2,*s3,*s4,*s5,*s6,*s7;
+	char *s1,*s2,*fetchid_var,*s4,*s5,*s6,*s7;
 	char s[100];
 	char *var;
 	int l;
@@ -319,16 +326,16 @@ static int aPGSQL_fetch(struct ast_channel *chan, void *data) {
 	s7=NULL;
 	s1=malloc(l);
 	s2=malloc(l);
-	strcpy(s1,data);
+	strncpy(s1, data, l - 1);
 	stringp=s1;
 	strsep(&stringp," "); // eat the first token, we already know it :P 
-	s3=strsep(&stringp," ");
+	fetchid_var=strsep(&stringp," ");
 	while (1) {	// ugly trick to make branches with break;
-		var=s3; // fetchid
+	  var=fetchid_var; // fetchid
 		fnd=0;
 		
 		AST_LIST_TRAVERSE(headp,variables,entries) {
-			if (strncasecmp(ast_var_name(variables),s3,strlen(s3))==0) {
+	    if (strncasecmp(ast_var_name(variables),fetchid_var,strlen(fetchid_var))==0) {
 	                        s7=ast_var_value(variables);
 	                        fnd=1;
                                 break;
@@ -337,7 +344,7 @@ static int aPGSQL_fetch(struct ast_channel *chan, void *data) {
 		
 		if (fnd==0) { 
 			s7="0";
-			pbx_builtin_setvar_helper(chan,s3,s7);
+	    pbx_builtin_setvar_helper(chan,fetchid_var,s7);
 		}
 
 		s4=strsep(&stringp," ");
@@ -349,12 +356,14 @@ static int aPGSQL_fetch(struct ast_channel *chan, void *data) {
 		}
 		id=atoi(s7); //fetchid
 		if ((lalares=find_identifier(id,AST_PGSQL_ID_FETCHID))==NULL) {
-			i=0;
+	    i=0;       // fetching the very first row
 		} else {
 			i=*lalares;
 			free(lalares);
-			del_identifier(id,AST_PGSQL_ID_FETCHID);
+	    del_identifier(id,AST_PGSQL_ID_FETCHID); // will re-add it a bit later
 		}
+
+	  if (i<PQntuples(PGSQLres)) {
 		nres=PQnfields(PGSQLres); 
 		ast_log(LOG_WARNING,"ast_PGSQL_fetch : nres = %d i = %d ;\n",nres,i);
 		for (j=0;j<nres;j++) {
@@ -363,7 +372,6 @@ static int aPGSQL_fetch(struct ast_channel *chan, void *data) {
 				ast_log(LOG_WARNING,"ast_PGSQL_fetch : More tuples (%d) than variables (%d)\n",nres,j);
 				break;
 			}
-			
 			s6=PQgetvalue(PGSQLres,i,j);
 			if (s6==NULL) { 
 				ast_log(LOG_WARNING,"PWgetvalue(res,%d,%d) returned NULL in ast_PGSQL_fetch\n",i,j);
@@ -372,18 +380,16 @@ static int aPGSQL_fetch(struct ast_channel *chan, void *data) {
 			ast_log(LOG_WARNING,"===setting variable '%s' to '%s'\n",s5,s6);
 			pbx_builtin_setvar_helper(chan,s5,s6);
 		}
-		i++;
-		if (i<PQntuples(PGSQLres)) {
 			lalares=malloc(sizeof(int));
-			*lalares=i;
-			id1=add_identifier(AST_PGSQL_ID_FETCHID,lalares);
+	    *lalares = ++i; // advance to the next row
+	    id1 = add_identifier(AST_PGSQL_ID_FETCHID,lalares);
 		} else {
-			id1=0;
+	    ast_log(LOG_WARNING,"ast_PGSQL_fetch : EOF\n");
+	    id1 = 0; // no more rows
 		}
-		s5=&s[0];
-		sprintf(s5,"%d",id1);
-		ast_log(LOG_WARNING,"Setting var '%s' to value '%s'\n",s3,s);
-		pbx_builtin_setvar_helper(chan,s3,s);
+		snprintf(s, sizeof(s), "%d", id1);
+	  ast_log(LOG_WARNING,"Setting var '%s' to value '%s'\n",fetchid_var,s);
+	  pbx_builtin_setvar_helper(chan,fetchid_var,s);
 	 	break;
 	}
 	
@@ -403,7 +409,7 @@ static int aPGSQL_reset(struct ast_channel *chan, void *data) {
 	
 	l=strlen(data)+2;
 	s1=malloc(l);
-	strcpy(s1,data);
+	strncpy(s1, data, l - 1);
 	stringp=s1;
 	strsep(&stringp," "); // eat the first token, we already know it :P 
 	s3=strsep(&stringp," ");
@@ -429,7 +435,7 @@ static int aPGSQL_clear(struct ast_channel *chan, void *data) {
 	
 	l=strlen(data)+2;
 	s1=malloc(l);
-	strcpy(s1,data);
+	strncpy(s1, data, l - 1);
 	stringp=s1;
 	strsep(&stringp," "); // eat the first token, we already know it :P 
 	s3=strsep(&stringp," ");
@@ -459,7 +465,7 @@ static int aPGSQL_disconnect(struct ast_channel *chan, void *data) {
 	
 	l=strlen(data)+2;
 	s1=malloc(l);
-	strcpy(s1,data);
+	strncpy(s1, data, l - 1);
 	stringp=s1;
 	strsep(&stringp," "); // eat the first token, we already know it :P 
 	s3=strsep(&stringp," ");
@@ -487,6 +493,10 @@ static int PGSQL_exec(struct ast_channel *chan, void *data)
 	struct localuser *u;
 	int result;
 
+#if EXTRA_LOG
+	printf("PRSQL_exec: data=%s\n",(char*)data);
+#endif
+
 	if (!data) {
 		ast_log(LOG_WARNING, "APP_PGSQL requires an argument (see manual)\n");
 		return -1;

apps/app_striplsd.c

@@ -11,19 +11,18 @@
  * the GNU General Public License
  */
 
+#include <sys/types.h>
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
+#include <asterisk/lock.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <stdlib.h>
 
-#include <pthread.h>
-
-
 static char *tdesc = "Strip trailing digits";
 
 static char *descrip =
@@ -46,16 +45,27 @@ LOCAL_USER_DECL;
 
 static int striplsd_exec(struct ast_channel *chan, void *data)
 {
-  char newexten[AST_MAX_EXTENSION] = "";
-  if (!data || !atoi(data)) {
-    ast_log(LOG_DEBUG, "Ignoring, since number of digits to strip is 0\n");
-    return 0;
-  }
-  if (strlen(chan->exten) > atoi(data)) {
-    strncpy(newexten, chan->exten, strlen(chan->exten)-atoi(data));
-  }
-  strncpy(chan->exten, newexten, sizeof(chan->exten)-1);
-  return 0;
+	char newexten[AST_MAX_EXTENSION] = "";
+	int maxbytes = 0;
+	int stripcount = 0;
+	int extlen = strlen(chan->exten);
+
+	maxbytes = sizeof(newexten) - 1;
+	if (data) {
+		stripcount = atoi(data);
+	}
+	if (!stripcount) {
+		ast_log(LOG_DEBUG, "Ignoring, since number of digits to strip is 0\n");
+		return 0;
+	}
+	if (extlen > stripcount) {
+		if (extlen - stripcount <= maxbytes) {
+			maxbytes = extlen - stripcount;
+		}
+		strncpy(newexten, chan->exten, maxbytes);
+	}
+	strncpy(chan->exten, newexten, sizeof(chan->exten)-1);
+	return 0;
 }
 
 int unload_module(void)

apps/app_substring.c

@@ -11,40 +11,44 @@
  * the GNU General Public License
  */
 
+#include <sys/types.h>
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
 #include <asterisk/pbx.h>
+#include <asterisk/lock.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <stdlib.h>
 
-#include <pthread.h>
-
-
-static char *tdesc = "Save substring digits in a given variable";
+static char *tdesc = "(Deprecated) Save substring digits in a given variable";
 
 static char *descrip =
-"  StripLSD(variable=string_of_digits|count1|count2): Assigns the substring\n"
+"  (Deprecated, use ${variable:a:b} instead)\n"
+"\n"
+"  SubString(variable=string_of_digits|count1|count2): Assigns the substring\n"
 "of string_of_digits to a given variable. Parameter count1 may be positive\n"
 "or negative. If it's positive then we skip the first count1 digits from the\n"
 "left. If it's negative, we move count1 digits counting from the end of\n"
-"the string to the left. Parameter count2 may be only positive and implies\n"
-"how many digits we're taking from the point that count1 placed us.\n"
+"the string to the left. Parameter count2 implies how many digits we are\n"
+"taking from the point that count1 placed us. If count2 is negative, then\n"
+"that many digits are omitted from the end.\n"
 "For example:\n"
 "exten => _NXXXXXX,1,SubString,test=2564286161|0|3\n"
 "assigns the area code (3 first digits) to variable test.\n"
 "exten => _NXXXXXX,1,SubString,test=2564286161|-7|7\n"
 "assigns the last 7 digits to variable test.\n"
+"exten => _NXXXXXX,1,SubString,test=2564286161|0|-4\n" 
+"assigns all but the last 4 digits to variable test.\n" 
 "If there are no parameters it'll return with -1.\n"
 "If there wrong parameters it go on and return with 0\n";
 
 static char *app = "SubString";
 
-static char *synopsis = "Save substring digits in a given variable";
+static char *synopsis = "(Deprecated) Save substring digits in a given variable";
 
 STANDARD_LOCAL_USER;
 
@@ -55,9 +59,10 @@ static int substring_exec(struct ast_channel *chan, void *data)
   char newexten[AST_MAX_EXTENSION] = "";
   char *count1, *count2;
   char *first, *second, *stringp;
+
   stringp=alloca(strlen(data)+1);
   ast_log(LOG_WARNING, "The use of Substring application is deprecated. Please use ${variable:a:b} instead\n");
-  strncpy(stringp,data,strlen(data)+1);
+  strncpy(stringp,data,strlen(data));
   if (strchr(stringp,'|')&&strchr(stringp,'=')) {
     int icount1,icount2;
     first=strsep(&stringp,"=");
@@ -70,9 +75,8 @@ static int substring_exec(struct ast_channel *chan, void *data)
     }
     icount1=atoi(count1);
     icount2=atoi(count2);
-    if (icount2<=0) {
-      ast_log(LOG_DEBUG, "Exiting, wrong parameter count2\n");
-      return -1;
+	if (icount2<0) {
+	  icount2 = icount2 + strlen(second);
     }
     if (abs(icount1)>strlen(second)) {
       ast_log(LOG_WARNING, "Limiting count1 parameter because it exceeds the length of the string\n");

apps/app_system.c

@@ -3,9 +3,9 @@
  *
  * Execute arbitrary system commands
  * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999-2004, Digium, Inc.
  *
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
  *
  * This program is free software, distributed under the terms of
  * the GNU General Public License
@@ -17,20 +17,23 @@
 #include <asterisk/channel.h>
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
+#include <asterisk/app.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <stdlib.h>
-
-#include <pthread.h>
-
+#include <errno.h>
 
 static char *tdesc = "Generic System() application";
 
 static char *app = "System";
 
+static char *app2 = "TrySystem";
+
 static char *synopsis = "Execute a system command";
 
+static char *synopsis2 = "Try executing a system command";
+
 static char *descrip =
 "  System(command): Executes a command  by  using  system(). Returns -1 on\n"
 "failure to execute the specified command. If  the command itself executes\n"
@@ -38,28 +41,37 @@ static char *descrip =
 "priority of the current instance, then  the  channel  will  be  setup  to\n"
 "continue at that priority level.  Otherwise, System returns 0.\n";
 
+static char *descrip2 =
+"  TrySystem(command): Executes a command  by  using  system(). Returns 0\n"
+"on any situation. If  the command itself executes but is in error, and if\n"
+"there exists a priority n + 101, where 'n' is the priority of the current\n"
+"instance, then  the  channel  will  be  setup  to continue at that\n"
+"priority level.  Otherwise, System returns 0.\n";
+
 STANDARD_LOCAL_USER;
 
 LOCAL_USER_DECL;
 
-static int skel_exec(struct ast_channel *chan, void *data)
+static int system_exec_helper(struct ast_channel *chan, void *data, int failmode)
 {
 	int res=0;
 	struct localuser *u;
 	if (!data) {
 		ast_log(LOG_WARNING, "System requires an argument(command)\n");
-		return -1;
+		return failmode;
 	}
 	LOCAL_USER_ADD(u);
 	/* Do our thing here */
-	res = system((char *)data);
-	if (res < 0) {
+	res = ast_safe_system((char *)data);
+	if ((res < 0) && (errno != ECHILD)) {
 		ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
-		res = -1;
+		res = failmode;
 	} else if (res == 127) {
 		ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
-		res = -1;
+		res = failmode;
 	} else {
+		if (res < 0)
+			res = 0;
 		if (res && ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid)) 
 			chan->priority+=100;
 		res = 0;
@@ -68,15 +80,27 @@ static int skel_exec(struct ast_channel *chan, void *data)
 	return res;
 }
 
+static int system_exec(struct ast_channel *chan, void *data)
+{
+	return system_exec_helper(chan, data, -1);
+}
+
+static int trysystem_exec(struct ast_channel *chan, void *data)
+{
+	return system_exec_helper(chan, data, 0);
+}
+
 int unload_module(void)
 {
 	STANDARD_HANGUP_LOCALUSERS;
+	ast_unregister_application(app2);
 	return ast_unregister_application(app);
 }
 
 int load_module(void)
 {
-	return ast_register_application(app, skel_exec, synopsis, descrip);
+	ast_register_application(app2, trysystem_exec, synopsis2, descrip2);
+	return ast_register_application(app, system_exec, synopsis, descrip);
 }
 
 char *description(void)

apps/app_talkdetect.c

@@ -0,0 +1,218 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Playback a file with audio detect
+ * 
+ * Copyright (C) 2004, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/utils.h>
+#include <asterisk/dsp.h>
+#include <string.h>
+#include <stdlib.h>
+
+static char *tdesc = "Playback with Talk Detection";
+
+static char *app = "BackgroundDetect";
+
+static char *synopsis = "Background a file with talk detect";
+
+static char *descrip = 
+"  BackgroundDetect(filename[|sil[|min|[max]]]):  Plays  back  a  given\n"
+"filename, waiting for interruption from a given digit (the digit must\n"
+"start the beginning of a valid extension, or it will be ignored).\n"
+"During the playback of the file, audio is monitored in the receive\n"
+"direction, and if a period of non-silence which is greater than 'min' ms\n"
+"yet less than 'max' ms is followed by silence for at least 'sil' ms then\n"
+"the audio playback is aborted and processing jumps to the 'talk' extension\n"
+"if available.  If unspecified, sil, min, and max default to 1000, 100, and\n"
+"infinity respectively.  Returns -1 on hangup, and 0 on successful playback\n"
+"completion with no exit conditions.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int background_detect_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u;
+	char tmp[256];
+	char *options;
+	char *stringp;
+	struct ast_frame *fr;
+	int notsilent=0;
+	struct timeval start = { 0, 0}, end = {0, 0};
+	int sil = 1000;
+	int min = 100;
+	int max = -1;
+	int x;
+	int origrformat=0;
+	struct ast_dsp *dsp;
+	if (!data || ast_strlen_zero((char *)data)) {
+		ast_log(LOG_WARNING, "BackgroundDetect requires an argument (filename)\n");
+		return -1;
+	}
+	strncpy(tmp, (char *)data, sizeof(tmp)-1);
+	stringp=tmp;
+	strsep(&stringp, "|");
+	options = strsep(&stringp, "|");
+	if (options) {
+		if ((sscanf(options, "%d", &x) == 1) && (x > 0))
+			sil = x;
+		options = strsep(&stringp, "|");
+		if (options) {
+			if ((sscanf(options, "%d", &x) == 1) && (x > 0))
+				min = x;
+			options = strsep(&stringp, "|");
+			if (options) {
+				if ((sscanf(options, "%d", &x) == 1) && (x > 0))
+					max = x;
+			}
+		}
+	}
+	ast_log(LOG_DEBUG, "Preparing detect of '%s', sil=%d,min=%d,max=%d\n", 
+						tmp, sil, min, max);
+	LOCAL_USER_ADD(u);
+	if (chan->_state != AST_STATE_UP) {
+		/* Otherwise answer unless we're supposed to send this while on-hook */
+		res = ast_answer(chan);
+	}
+	if (!res) {
+		origrformat = chan->readformat;
+		if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR))) 
+			ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
+	}
+	if (!(dsp = ast_dsp_new())) {
+		ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
+		res = -1;
+	}
+	if (!res) {
+		ast_stopstream(chan);
+		res = ast_streamfile(chan, tmp, chan->language);
+		if (!res) {
+			while(chan->stream) {
+				res = ast_sched_wait(chan->sched);
+				if ((res < 0) && !chan->timingfunc) {
+					res = 0;
+					break;
+				}
+				if (res < 0)
+					res = 1000;
+				res = ast_waitfor(chan, res);
+				if (res < 0) {
+					ast_log(LOG_WARNING, "Waitfor failed on %s\n", chan->name);
+					break;
+				} else if (res > 0) {
+					fr = ast_read(chan);
+					if (!fr) {
+						res = -1;
+						break;
+					} else if (fr->frametype == AST_FRAME_DTMF) {
+						char t[2];
+						t[0] = fr->subclass;
+						t[1] = '\0';
+						if (ast_canmatch_extension(chan, chan->context, t, 1, chan->callerid)) {
+							/* They entered a valid  extension, or might be anyhow */
+							res = fr->subclass;
+							ast_frfree(fr);
+							break;
+						}
+					} else if ((fr->frametype == AST_FRAME_VOICE) && (fr->subclass == AST_FORMAT_SLINEAR)) {
+						int totalsilence;
+						int ms;
+						res = ast_dsp_silence(dsp, fr, &totalsilence);
+						if (res && (totalsilence > sil)) {
+							/* We've been quiet a little while */
+							if (notsilent) {
+								/* We had heard some talking */
+								gettimeofday(&end, NULL);
+								ms = (end.tv_sec - start.tv_sec) * 1000;
+								ms += (end.tv_usec - start.tv_usec) / 1000;
+								ms -= sil;
+								if (ms < 0)
+									ms = 0;
+								if ((ms > min) && ((max < 0) || (ms < max))) {
+									ast_log(LOG_DEBUG, "Found qualified token of %d ms\n", ms);
+									if (ast_exists_extension(chan, chan->context, "talk", 1, chan->callerid)) {
+										strncpy(chan->exten, "talk", sizeof(chan->exten) -1 );
+										chan->priority = 0;
+									}
+									res = 0;
+									ast_frfree(fr);
+									break;
+								} else
+									ast_log(LOG_DEBUG, "Found unqualified token of %d ms\n", ms);
+								notsilent = 0;
+							}
+						} else {
+							if (!notsilent) {
+								/* Heard some audio, mark the begining of the token */
+								gettimeofday(&start, NULL);
+								ast_log(LOG_DEBUG, "Start of voice token!\n");
+								notsilent = 1;
+							}
+						}
+						
+					}
+					ast_frfree(fr);
+				}
+				ast_sched_runq(chan->sched);
+			}
+			ast_stopstream(chan);
+		} else {
+			ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
+			res = 0;
+		}
+	}
+	if (res > -1) {
+		if (origrformat && ast_set_read_format(chan, origrformat)) {
+			ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n", 
+				chan->name, ast_getformatname(origrformat));
+		}
+	}
+	if (dsp)
+		ast_dsp_free(dsp);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, background_detect_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_test.c

@@ -0,0 +1,473 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Applications to test connection and produce report in text file
+ * 
+ * Copyright (C) 2004, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ * Russell Bryant <russelb@clemson.edu>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <sys/types.h>
+#include <asterisk/channel.h>
+#include <asterisk/options.h>
+#include <asterisk/module.h>
+#include <asterisk/logger.h>
+#include <asterisk/lock.h>
+#include <asterisk/app.h>
+#include <asterisk/pbx.h>
+#include <asterisk/utils.h>
+#include <string.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include "../astconf.h"
+
+
+static char *tdesc = "Interface Test Application";
+
+static char *tests_descrip = 
+	 "TestServer(): Perform test server function and write call report.\n"
+	 "Results stored in /var/log/asterisk/testreports/<testid>-server.txt";
+static char *tests_app = "TestServer";
+static char *tests_synopsis = "Execute Interface Test Server";
+
+static char *testc_descrip = 
+	 "TestClient(testid): Executes test client with given testid.\n"
+	 "Results stored in /var/log/asterisk/testreports/<testid>-client.txt";
+
+static char *testc_app = "TestClient";
+static char *testc_synopsis = "Execute Interface Test Client";
+
+static int measurenoise(struct ast_channel *chan, int ms, char *who)
+{
+	int res=0;
+	int mssofar;
+	int noise=0;
+	int samples=0;
+	int x;
+	short *foo;
+	struct timeval start, tv;
+	struct ast_frame *f;
+	int rformat;
+	rformat = chan->readformat;
+	if (ast_set_read_format(chan, AST_FORMAT_SLINEAR)) {
+		ast_log(LOG_NOTICE, "Unable to set to linear mode!\n");
+		return -1;
+	}
+	gettimeofday(&start, NULL);
+	for(;;) {
+		gettimeofday(&tv, NULL);
+		mssofar = (tv.tv_sec - start.tv_sec) * 1000;
+		mssofar += (tv.tv_usec - start.tv_usec) / 1000;
+		if (mssofar > ms)
+			break;
+		res = ast_waitfor(chan, ms - mssofar);
+		if (res < 1)
+			break;
+		f = ast_read(chan);
+		if (!f) {
+			res = -1;
+			break;
+		}
+		if ((f->frametype == AST_FRAME_VOICE) && (f->subclass == AST_FORMAT_SLINEAR)) {
+			foo = (short *)f->data;
+			for (x=0;x<f->samples;x++) {
+				noise += abs(foo[x]);
+				samples++;
+			}
+		}
+	}
+
+	if (rformat) {
+		if (ast_set_read_format(chan, rformat)) {
+			ast_log(LOG_NOTICE, "Unable to restore original format!\n");
+			return -1;
+		}
+	}
+	if (res < 0)
+		return res;
+	if (!samples) {
+		ast_log(LOG_NOTICE, "No samples were received from the other side!\n");
+		return -1;
+	}
+	ast_log(LOG_DEBUG, "%s: Noise: %d, samples: %d, avg: %d\n", who, noise, samples, noise / samples);
+	return (noise / samples);
+}
+
+static int sendnoise(struct ast_channel *chan, int ms) 
+{
+	int res;
+	res = ast_tonepair_start(chan, 1537, 2195, ms, 8192);
+	if (!res) {
+		res = ast_waitfordigit(chan, ms);
+		ast_tonepair_stop(chan);
+	}
+	return res;	
+}
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int testclient_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	int res = 0;
+	char *testid=data;
+	char fn[80];
+	char serverver[80];
+	FILE *f;
+	LOCAL_USER_ADD(u);
+	
+	/* Check for test id */
+	if (!testid || ast_strlen_zero(testid)) {
+		ast_log(LOG_WARNING, "TestClient requires an argument - the test id\n");
+		return -1;
+	}
+	
+	if (chan->_state != AST_STATE_UP)
+		res = ast_answer(chan);
+	
+	/* Wait a few just to be sure things get started */
+	res = ast_safe_sleep(chan, 3000);
+	/* Transmit client version */
+	if (!res)
+		res = ast_dtmf_stream(chan, NULL, "8378*1#", 0);
+	ast_log(LOG_DEBUG, "Transmit client version\n");
+	
+	/* Read server version */
+	ast_log(LOG_DEBUG, "Read server version\n");
+	if (!res) 
+		res = ast_app_getdata(chan, NULL, serverver, sizeof(serverver) - 1, 0);
+	if (res > 0)
+		res = 0;
+	ast_log(LOG_DEBUG, "server version: %s\n", serverver);
+		
+	if (res > 0)
+		res = 0;
+
+	if (!res)
+		res = ast_safe_sleep(chan, 1000);
+	/* Send test id */
+	if (!res) 
+		res = ast_dtmf_stream(chan, NULL, testid, 0);		
+	if (!res) 
+		res = ast_dtmf_stream(chan, NULL, "#", 0);		
+	ast_log(LOG_DEBUG, "send test identifier: %s\n", testid);
+
+	if ((res >=0) && (!ast_strlen_zero(testid))) {
+		/* Make the directory to hold the test results in case it's not there */
+		snprintf(fn, sizeof(fn), "%s/testresults", ast_config_AST_LOG_DIR);
+		mkdir(fn, 0777);
+		snprintf(fn, sizeof(fn), "%s/testresults/%s-client.txt", ast_config_AST_LOG_DIR, testid);
+		if ((f = fopen(fn, "w+"))) {
+			setlinebuf(f);
+			fprintf(f, "CLIENTCHAN:    %s\n", chan->name);
+			fprintf(f, "CLIENTTEST ID: %s\n", testid);
+			fprintf(f, "ANSWER:        PASS\n");
+			res = 0;
+			
+			if (!res) {
+				/* Step 1: Wait for "1" */
+				ast_log(LOG_DEBUG, "TestClient: 2.  Wait DTMF 1\n");
+				res = ast_waitfordigit(chan, 3000);
+				fprintf(f, "WAIT DTMF 1:   %s\n", (res != '1') ? "FAIL" : "PASS");
+				if (res == '1')
+					res = 0;
+				else
+					res = -1;
+			}
+			if (!res)
+				res = ast_safe_sleep(chan, 1000);
+			if (!res) {
+				/* Step 2: Send "2" */
+				ast_log(LOG_DEBUG, "TestClient: 2.  Send DTMF 2\n");
+				res = ast_dtmf_stream(chan, NULL, "2", 0);
+				fprintf(f, "SEND DTMF 2:   %s\n", (res < 0) ? "FAIL" : "PASS");
+				if (res > 0)
+					res = 0;
+			}
+			if (!res) {
+				/* Step 3: Wait one second */
+				ast_log(LOG_DEBUG, "TestClient: 3.  Wait one second\n");
+				res = ast_safe_sleep(chan, 1000);
+				fprintf(f, "WAIT 1 SEC:    %s\n", (res < 0) ? "FAIL" : "PASS");
+				if (res > 0)
+					res = 0;
+			}			
+			if (!res) {
+				/* Step 4: Measure noise */
+				ast_log(LOG_DEBUG, "TestClient: 4.  Measure noise\n");
+				res = measurenoise(chan, 5000, "TestClient");
+				fprintf(f, "MEASURENOISE:  %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
+				if (res > 0)
+					res = 0;
+			}
+			if (!res) {
+				/* Step 5: Wait for "4" */
+				ast_log(LOG_DEBUG, "TestClient: 5.  Wait DTMF 4\n");
+				res = ast_waitfordigit(chan, 3000);
+				fprintf(f, "WAIT DTMF 4:   %s\n", (res != '4') ? "FAIL" : "PASS");
+				if (res == '4')
+					res = 0;
+				else
+					res = -1;
+			}
+			if (!res) {
+				/* Step 6: Transmit tone noise */
+				ast_log(LOG_DEBUG, "TestClient: 6.  Transmit tone\n");
+				res = sendnoise(chan, 6000);
+				fprintf(f, "SENDTONE:      %s\n", (res < 0) ? "FAIL" : "PASS");
+			}
+			if (!res || (res == '5')) {
+				/* Step 7: Wait for "5" */
+				ast_log(LOG_DEBUG, "TestClient: 7.  Wait DTMF 5\n");
+				if (!res)
+					res = ast_waitfordigit(chan, 3000);
+				fprintf(f, "WAIT DTMF 5:   %s\n", (res != '5') ? "FAIL" : "PASS");
+				if (res == '5')
+					res = 0;
+				else
+					res = -1;
+			}
+			if (!res) {
+				/* Step 8: Wait one second */
+				ast_log(LOG_DEBUG, "TestClient: 8.  Wait one second\n");
+				res = ast_safe_sleep(chan, 1000);
+				fprintf(f, "WAIT 1 SEC:    %s\n", (res < 0) ? "FAIL" : "PASS");
+				if (res > 0)
+					res = 0;
+			}
+			if (!res) {
+				/* Step 9: Measure noise */
+				ast_log(LOG_DEBUG, "TestClient: 6.  Measure tone\n");
+				res = measurenoise(chan, 4000, "TestClient");
+				fprintf(f, "MEASURETONE:   %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
+				if (res > 0)
+					res = 0;
+			}
+			if (!res) {
+				/* Step 10: Send "7" */
+				ast_log(LOG_DEBUG, "TestClient: 7.  Send DTMF 7\n");
+				res = ast_dtmf_stream(chan, NULL, "7", 0);
+				fprintf(f, "SEND DTMF 7:   %s\n", (res < 0) ? "FAIL" : "PASS");
+				if (res > 0)
+					res =0;
+			}
+			if (!res) {
+				/* Step 11: Wait for "8" */
+				ast_log(LOG_DEBUG, "TestClient: 11.  Wait DTMF 8\n");
+				res = ast_waitfordigit(chan, 3000);
+				fprintf(f, "WAIT DTMF 8:   %s\n", (res != '8') ? "FAIL" : "PASS");
+				if (res == '8')
+					res = 0;
+				else
+					res = -1;
+			}
+			if (!res) {
+				/* Step 12: Hangup! */
+				ast_log(LOG_DEBUG, "TestClient: 12.  Hangup\n");
+			}
+
+			ast_log(LOG_DEBUG, "-- TEST COMPLETE--\n");
+			fprintf(f, "-- END TEST--\n");
+			fclose(f);
+			res = -1;
+		} else
+			res = -1;
+	} else {
+		ast_log(LOG_NOTICE, "Did not read a test ID on '%s'\n", chan->name);
+		res = -1;
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+static int testserver_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	int res = 0;
+	char testid[80]="";
+	char fn[80];
+	FILE *f;
+	LOCAL_USER_ADD(u);
+	if (chan->_state != AST_STATE_UP)
+		res = ast_answer(chan);
+	/* Read version */
+	ast_log(LOG_DEBUG, "Read client version\n");
+	if (!res) 
+		res = ast_app_getdata(chan, NULL, testid, sizeof(testid) - 1, 0);
+	if (res > 0)
+		res = 0;
+	ast_log(LOG_DEBUG, "client version: %s\n", testid);
+	ast_log(LOG_DEBUG, "Transmit server version\n");
+	res = ast_safe_sleep(chan, 1000);
+	if (!res)
+		res = ast_dtmf_stream(chan, NULL, "8378*1#", 0);
+	if (res > 0)
+		res = 0;
+
+	if (!res) 
+		res = ast_app_getdata(chan, NULL, testid, sizeof(testid) - 1, 0);		
+	ast_log(LOG_DEBUG, "read test identifier: %s\n", testid);
+	/* Check for sneakyness */
+	if (strchr(testid, '/'))
+		res = -1;
+	if ((res >=0) && (!ast_strlen_zero(testid))) {
+		/* Got a Test ID!  Whoo hoo! */
+		/* Make the directory to hold the test results in case it's not there */
+		snprintf(fn, sizeof(fn), "%s/testresults", ast_config_AST_LOG_DIR);
+		mkdir(fn, 0777);
+		snprintf(fn, sizeof(fn), "%s/testresults/%s-server.txt", ast_config_AST_LOG_DIR, testid);
+		if ((f = fopen(fn, "w+"))) {
+			setlinebuf(f);
+			fprintf(f, "SERVERCHAN:    %s\n", chan->name);
+			fprintf(f, "SERVERTEST ID: %s\n", testid);
+			fprintf(f, "ANSWER:        PASS\n");
+			ast_log(LOG_DEBUG, "Processing Test ID '%s'\n", testid);
+			res = ast_safe_sleep(chan, 1000);
+			if (!res) {
+				/* Step 1: Send "1" */
+				ast_log(LOG_DEBUG, "TestServer: 1.  Send DTMF 1\n");
+				res = ast_dtmf_stream(chan, NULL, "1", 0);
+				fprintf(f, "SEND DTMF 1:   %s\n", (res < 0) ? "FAIL" : "PASS");
+				if (res > 0)
+					res = 0;
+			}
+			if (!res) {
+				/* Step 2: Wait for "2" */
+				ast_log(LOG_DEBUG, "TestServer: 2.  Wait DTMF 2\n");
+				res = ast_waitfordigit(chan, 3000);
+				fprintf(f, "WAIT DTMF 2:   %s\n", (res != '2') ? "FAIL" : "PASS");
+				if (res == '2')
+					res = 0;
+				else
+					res = -1;
+			}
+			if (!res) {
+				/* Step 3: Measure noise */
+				ast_log(LOG_DEBUG, "TestServer: 3.  Measure noise\n");
+				res = measurenoise(chan, 6000, "TestServer");
+				fprintf(f, "MEASURENOISE:  %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
+				if (res > 0)
+					res = 0;
+			}
+			if (!res) {
+				/* Step 4: Send "4" */
+				ast_log(LOG_DEBUG, "TestServer: 4.  Send DTMF 4\n");
+				res = ast_dtmf_stream(chan, NULL, "4", 0);
+				fprintf(f, "SEND DTMF 4:   %s\n", (res < 0) ? "FAIL" : "PASS");
+				if (res > 0)
+					res = 0;
+			}
+		
+			if (!res) {
+				/* Step 5: Wait one second */
+				ast_log(LOG_DEBUG, "TestServer: 5.  Wait one second\n");
+				res = ast_safe_sleep(chan, 1000);
+				fprintf(f, "WAIT 1 SEC:    %s\n", (res < 0) ? "FAIL" : "PASS");
+				if (res > 0)
+					res = 0;
+			}
+		
+			if (!res) {
+				/* Step 6: Measure noise */
+				ast_log(LOG_DEBUG, "TestServer: 6.  Measure tone\n");
+				res = measurenoise(chan, 4000, "TestServer");
+				fprintf(f, "MEASURETONE:   %s (%d)\n", (res < 0) ? "FAIL" : "PASS", res);
+				if (res > 0)
+					res = 0;
+			}
+
+			if (!res) {
+				/* Step 7: Send "5" */
+				ast_log(LOG_DEBUG, "TestServer: 7.  Send DTMF 5\n");
+				res = ast_dtmf_stream(chan, NULL, "5", 0);
+				fprintf(f, "SEND DTMF 5:   %s\n", (res < 0) ? "FAIL" : "PASS");
+				if (res > 0)
+					res = 0;
+			}
+
+			if (!res) {
+				/* Step 8: Transmit tone noise */
+				ast_log(LOG_DEBUG, "TestServer: 8.  Transmit tone\n");
+				res = sendnoise(chan, 6000);
+				fprintf(f, "SENDTONE:      %s\n", (res < 0) ? "FAIL" : "PASS");
+			}
+		
+			if (!res || (res == '7')) {
+				/* Step 9: Wait for "7" */
+				ast_log(LOG_DEBUG, "TestServer: 9.  Wait DTMF 7\n");
+				if (!res)
+					res = ast_waitfordigit(chan, 3000);
+				fprintf(f, "WAIT DTMF 7:   %s\n", (res != '7') ? "FAIL" : "PASS");
+				if (res == '7')
+					res = 0;
+				else
+					res = -1;
+			}
+			if (!res)
+				res = ast_safe_sleep(chan, 1000);
+			if (!res) {
+				/* Step 10: Send "8" */
+				ast_log(LOG_DEBUG, "TestServer: 10.  Send DTMF 8\n");
+				res = ast_dtmf_stream(chan, NULL, "8", 0);
+				fprintf(f, "SEND DTMF 8:   %s\n", (res < 0) ? "FAIL" : "PASS");
+				if (res > 0)
+					res = 0;
+			}
+			if (!res) {
+				/* Step 11: Wait for hangup to arrive! */
+				ast_log(LOG_DEBUG, "TestServer: 11.  Waiting for hangup\n");
+				res = ast_safe_sleep(chan, 10000);
+				fprintf(f, "WAIT HANGUP:   %s\n", (res < 0) ? "PASS" : "FAIL");
+			}
+
+			ast_log(LOG_DEBUG, "-- TEST COMPLETE--\n");
+			fprintf(f, "-- END TEST--\n");
+			fclose(f);
+			res = -1;
+		} else
+			res = -1;
+	} else {
+		ast_log(LOG_NOTICE, "Did not read a test ID on '%s'\n", chan->name);
+		res = -1;
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	ast_unregister_application(testc_app);
+	return ast_unregister_application(tests_app);
+}
+
+int load_module(void)
+{
+	ast_register_application(testc_app, testclient_exec, testc_synopsis, testc_descrip);
+	return ast_register_application(tests_app, testserver_exec, tests_synopsis, tests_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key(void)
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_transfer.c

@@ -0,0 +1,91 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Transfer a caller
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+
+static char *tdesc = "Transfer";
+
+static char *app = "Transfer";
+
+static char *synopsis = "Transfer caller to remote extension";
+
+static char *descrip = 
+"  Transfer(exten):  Requests the remote caller be transferred to\n"
+"a given extension. Returns -1 on hangup, or 0 on completion\n"
+"regardless of whether the transfer was successful.  If the transfer\n"
+"was *not* supported or successful and there exists a priority n + 101,\n"
+"then that priority will be taken next.\n" ;
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int transfer_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	if (!data || !strlen(data)) {
+		ast_log(LOG_WARNING, "Transfer requires an argument (destination)\n");
+		res = 1;
+	}
+	LOCAL_USER_ADD(u);
+	if (!res) {
+		res = ast_transfer(chan, data);
+	}
+	if (!res) {
+		/* Look for a "busy" place */
+		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+			chan->priority += 100;
+	}
+	if (res > 0)
+		res = 0;
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, transfer_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_txtcidname.c

@@ -0,0 +1,145 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Caller*id name lookup - Look up the caller's name via DNS
+ * 
+ * Copyright (C) 1999-2004, Digium
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/options.h>
+#include <asterisk/config.h>
+#include <asterisk/module.h>
+#include <asterisk/enum.h>
+#include <asterisk/utils.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+#include <ctype.h>
+
+static char *tdesc = "TXTCIDName";
+
+static char *app = "TXTCIDName";
+
+static char *synopsis = "Lookup caller name from TXT record";
+
+static char *descrip = 
+"  TXTLookup(CallerID):  Looks up a Caller Name via DNS and sets\n"
+"the variable 'TXTCIDNAME'. TXTCIDName will either be blank\n"
+"or return the value found in the TXT record in DNS.\n" ;
+
+#define ENUM_CONFIG "enum.conf"
+
+static char h323driver[80] = "";
+#define H323DRIVERDEFAULT "H323"
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int txtcidname_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	char tech[80];
+	char txt[256] = "";
+	char dest[80];
+
+	struct localuser *u;
+	if (!data || !strlen(data)) {
+		ast_log(LOG_WARNING, "TXTCIDName requires an argument (extension)\n");
+		res = 1;
+	}
+	LOCAL_USER_ADD(u);
+	if (!res) {
+		res = ast_get_txt(chan, data, dest, sizeof(dest), tech, sizeof(tech), txt, sizeof(txt));
+	}
+	LOCAL_USER_REMOVE(u);
+	/* Parse it out */
+	if (res > 0) {
+		if (!ast_strlen_zero(txt)) {
+			pbx_builtin_setvar_helper(chan, "TXTCIDNAME", txt);
+#if 0
+			ast_log(LOG_DEBUG, "TXTCIDNAME got '%s'\n", txt);
+#endif
+		}
+	}
+	if (!res) {
+		/* Look for a "busy" place */
+		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+			chan->priority += 100;
+	} else if (res > 0)
+		res = 0;
+	return res;
+}
+
+static int load_config(void)
+{
+	struct ast_config *cfg;
+	char *s;
+
+	cfg = ast_load(ENUM_CONFIG);
+	if (cfg) {
+		if (!(s=ast_variable_retrieve(cfg, "general", "h323driver"))) {
+			strncpy(h323driver, H323DRIVERDEFAULT, sizeof(h323driver) - 1);
+		} else {
+			strncpy(h323driver, s, sizeof(h323driver) - 1);
+		}
+		ast_destroy(cfg);
+		return 0;
+	}
+	ast_log(LOG_NOTICE, "No ENUM Config file, using defaults\n");
+	return 0;
+}
+
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	int res;
+	res = ast_register_application(app, txtcidname_exec, synopsis, descrip);
+	if (res)
+		return(res);
+	if ((res=load_config())) {
+		return(res);
+	}
+	return(0);
+}
+
+int reload(void)
+{
+	return(load_config());
+}
+
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
+

apps/app_url.c

@@ -21,7 +21,6 @@
 #include <asterisk/image.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 static char *tdesc = "Send URL Applications";
 

apps/app_userevent.c

@@ -0,0 +1,106 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * UserEvent application -- send manager event
+ * 
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/manager.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+
+static char *tdesc = "Custom User Event Application";
+
+static char *app = "UserEvent";
+
+static char *synopsis = "Send an arbitrary event to the manager interface";
+
+static char *descrip = 
+"  UserEvent(eventname[|body]): Sends an arbitrary event to the\n"
+"manager interface, with an optional body representing additional\n"
+"arguments.  The format of the event will be:\n"
+"    Event: UserEvent<specified event name>\n"
+"    Channel: <channel name>\n"
+"    Uniqueid: <call uniqueid>\n"
+"    [body]\n"
+"If the body is not specified, only Event, Channel, and Uniqueid fields\n"
+"will be present.  Returns 0.";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int userevent_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	char info[512];
+    char eventname[512];
+	char *eventbody;
+
+	if (!data || !strlen(data)) {
+		ast_log(LOG_WARNING, "UserEvent requires an argument (eventname|optional event body)\n");
+		return -1;
+	}
+
+	strncpy(info, (char *)data, strlen((char *)data) + AST_MAX_EXTENSION-1);
+	snprintf(eventname, sizeof(eventname), "UserEvent%s", info);
+	eventbody = strchr(eventname, '|');
+	if (eventbody) {
+		*eventbody = '\0';
+		eventbody++;
+	}
+	LOCAL_USER_ADD(u);
+
+	if(eventbody) {
+            ast_log(LOG_DEBUG, "Sending user event: %s, %s\n", eventname, eventbody);
+            manager_event(EVENT_FLAG_USER, eventname, 
+			"Channel: %s\r\nUniqueid: %s\r\n%s\r\n",
+			chan->name, chan->uniqueid, eventbody);
+	} else {
+            ast_log(LOG_DEBUG, "Sending user event: %s\n", eventname);
+            manager_event(EVENT_FLAG_USER, eventname, 
+			"Channel: %s\r\nUniqueid: %s\r\n", chan->name, chan->uniqueid);
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return 0;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, userevent_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_verbose.c

@@ -0,0 +1,110 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Verbose application
+ * 
+ * Copyright (c) 2004 Tilghman Lesher.  All rights reserved.
+ *
+ * Tilghman Lesher <app_verbose_v001@the-tilghman.com>
+ *
+ * This code is released by the author with no restrictions on usage.
+ *
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <asterisk/options.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+
+
+static char *tdesc = "Send verbose output";
+
+static char *app_verbose = "Verbose";
+
+static char *verbose_synopsis = "Send arbitrary text to verbose output";
+
+static char *verbose_descrip =
+"Verbose([<level>|]<message>)\n"
+"  level must be an integer value.  If not specified, defaults to 0."
+"  Always returns 0.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int verbose_exec(struct ast_channel *chan, void *data)
+{
+	char *vtext;
+	int vsize;
+
+	if (data) {
+		vtext = ast_strdupa((char *)data);
+		if (vtext) {
+			char *tmp = strsep(&vtext, "|,");
+			if (vtext) {
+				if (sscanf(tmp, "%d", &vsize) != 1) {
+					vsize = 0;
+					ast_log(LOG_WARNING, "'%s' is not a verboser number\n", vtext);
+				}
+			} else {
+				vtext = tmp;
+				vsize = 0;
+			}
+			if (option_verbose >= vsize) {
+				switch (vsize) {
+				case 0:
+					ast_verbose("%s\n", vtext);
+					break;
+				case 1:
+					ast_verbose(VERBOSE_PREFIX_1 "%s\n", vtext);
+					break;
+				case 2:
+					ast_verbose(VERBOSE_PREFIX_2 "%s\n", vtext);
+					break;
+				case 3:
+					ast_verbose(VERBOSE_PREFIX_3 "%s\n", vtext);
+					break;
+				default:
+					ast_verbose(VERBOSE_PREFIX_4 "%s\n", vtext);
+				}
+			}
+		} else {
+			ast_log(LOG_ERROR, "Out of memory\n");
+		}
+	}
+
+	return 0;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app_verbose);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app_verbose, verbose_exec, verbose_synopsis, verbose_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_waitforring.c

@@ -11,20 +11,19 @@
  * the GNU General Public License
  */
 
+#include <sys/types.h>
 #include <asterisk/file.h>
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
 #include <asterisk/pbx.h>
 #include <asterisk/module.h>
 #include <asterisk/options.h>
+#include <asterisk/lock.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
 #include <stdlib.h>
 
-#include <pthread.h>
-
-
 static char *synopsis = "Wait for Ring Application";
 
 static char *tdesc = "Waits until first ring after time";
@@ -47,7 +46,7 @@ static int waitforring_exec(struct ast_channel *chan, void *data)
 	int res = 0;
 	int ms;
 	if (!data || (sscanf(data, "%d", &ms) != 1)) {
-                ast_log(LOG_WARNING, "SoftHangup requires an argument (minimum seconds)\n");
+                ast_log(LOG_WARNING, "WaitForRing requires an argument (minimum seconds)\n");
 		return 0;
 	}
 	ms *= 1000;

apps/app_zapateller.c

@@ -20,7 +20,6 @@
 #include <asterisk/translate.h>
 #include <string.h>
 #include <stdlib.h>
-#include <pthread.h>
 
 static char *tdesc = "Block Telemarketers with Special Information Tone";
 

apps/app_zapbarge.c

@@ -25,6 +25,7 @@
 #include <asterisk/options.h>
 #include <asterisk/cli.h>
 #include <asterisk/say.h>
+#include <asterisk/utils.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <errno.h>
@@ -32,8 +33,12 @@
 #include <stdlib.h>
 #include <sys/ioctl.h>
 
-#include <pthread.h>
+#ifdef __linux__
 #include <linux/zaptel.h>
+#else
+#include <zaptel.h>
+#endif /* __linux__ */
+
 static char *tdesc = "Barge in on Zap channel application";
 
 static char *app = "ZapBarge";
@@ -244,9 +249,9 @@ static int conf_exec(struct ast_channel *chan, void *data)
 	int retrycnt = 0;
 	int confflags = 0;
 	int confno = 0;
-	char confstr[80];
+	char confstr[80] = "";
 
-	if (data && strlen(data)) {
+	if (data && !ast_strlen_zero(data)) {
 		if ((sscanf(data, "Zap/%d", &confno) != 1) &&
 		    (sscanf(data, "%d", &confno) != 1)) {
 			ast_log(LOG_WARNING, "ZapBarge Argument (if specified) must be a channel number, not '%s'\n", (char *)data);
@@ -259,7 +264,7 @@ static int conf_exec(struct ast_channel *chan, void *data)
 
 	while(!confno && (++retrycnt < 4)) {
 		/* Prompt user for conference number */
-		strcpy(confstr, "");
+		confstr[0] = '\0';
 		res = ast_app_getdata(chan, "conf-getchannel",confstr, sizeof(confstr) - 1, 0);
 		if (res <0) goto out;
 		if (sscanf(confstr, "%d", &confno) != 1)

apps/app_zapras.c

@@ -20,7 +20,11 @@
 #include <asterisk/options.h>
 #include <sys/ioctl.h>
 #include <sys/wait.h>
+#ifdef __linux__
 #include <sys/signal.h>
+#else
+#include <signal.h>
+#endif /* __linux__ */
 
 #include <stdlib.h>
 #include <unistd.h>
@@ -30,10 +34,12 @@
 #include <stdio.h>
 #include <fcntl.h>
 
-#include <pthread.h>
-
 /* Need some zaptel help here */
+#ifdef __linux__
 #include <linux/zaptel.h>
+#else
+#include <zaptel.h>
+#endif /* __linux__ */
 
 static char *tdesc = "Zap RAS Application";
 
@@ -44,7 +50,7 @@ static char *synopsis = "Executes Zaptel ISDN RAS application";
 static char *descrip =
 "  ZapRAS(args): Executes a RAS server using pppd on the given channel.\n"
 "The channel must be a clear channel (i.e. PRI source) and a Zaptel\n"
-"channel to be able to use this function (No modem emulcation is included).\n"
+"channel to be able to use this function (No modem emulation is included).\n"
 "Your pppd must be patched to be zaptel aware. Arguments should be\n"
 "separated by | characters.  Always returns -1.\n";
 
@@ -119,8 +125,14 @@ static void run_ras(struct ast_channel *chan, char *args)
 	int status;
 	int res;
 	int signalled = 0;
-	struct zt_bufferinfo bi;
+	struct zt_bufferinfo savebi;
 	int x;
+	
+	res = ioctl(chan->fds[0], ZT_GET_BUFINFO, &savebi);
+	if(res) {
+		ast_log(LOG_WARNING, "Unable to check buffer policy on channel %s\n", chan->name);
+		return;
+	}
 
 	pid = spawn_ras(chan, args);
 	if (pid < 0) {
@@ -156,20 +168,11 @@ static void run_ras(struct ast_channel *chan, char *args)
 			x = 1;
 			ioctl(chan->fds[0], ZT_AUDIOMODE, &x);
 
-			/* Double check buffering too */
-			res = ioctl(chan->fds[0], ZT_GET_BUFINFO, &bi);
-			if (!res) {
-				/* XXX This is ZAP_BLOCKSIZE XXX */
-				bi.bufsize = 204;
-				bi.txbufpolicy = ZT_POLICY_IMMEDIATE;
-				bi.rxbufpolicy = ZT_POLICY_IMMEDIATE;
-				bi.numbufs = 4;
-				res = ioctl(chan->fds[0], ZT_SET_BUFINFO, &bi);
-				if (res < 0) {
-					ast_log(LOG_WARNING, "Unable to set buffer policy on channel %s\n", chan->name);
-				}
-			} else
-				ast_log(LOG_WARNING, "Unable to check buffer policy on channel %s\n", chan->name);
+			/* Restore saved values */
+			res = ioctl(chan->fds[0], ZT_SET_BUFINFO, &savebi);
+			if (res < 0) {
+				ast_log(LOG_WARNING, "Unable to set buffer policy on channel %s\n", chan->name);
+			}
 			break;
 		}
 	}