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/1.0.6@212958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

.cleancount

@@ -1 +0,0 @@
-36

BUGS

@@ -1,22 +1,23 @@
+Asterisk 1.0-RC-1 Known Major Bugs
+==================================
+* Some people still have issues with H.323
+* QuickNet driver still not entirely stable
+
 Asterisk Bug Tracking Information
 =================================
 
-To learn about and report Asterisk bugs, please visit 
-the official Asterisk Bug Tracker at:
+To learn about and report Asterisk bugs or make feature
+requests, please visit the official Asterisk Bug Tracker
+at:
 
-	https://issues.asterisk.org
+	http://bugs.digium.com
 
 For more information on using the bug tracker, or to 
 learn how you can contribute by acting as a bug marshal
 please see:
 
-	http://www.asterisk.org/developers/bug-guidelines
-
-If you would like to submit a feature request, please
-resist the temptation to post it to the bug tracker.
-Feature requests should be posted to the asterisk-dev
-mailing list, located at:
-
-	http://lists.digium.com 
+	http://www.digium.com/bugtracker.html
 
 Thank you!
+
+Mark

COPYING

@@ -1,341 +0,0 @@
-
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CREDITS

@@ -1,6 +1,6 @@
 
 === DEVELOPMENT SUPPORT ===
-We'd like to thank the following companies for helping fund development of
+I'd like to thank the following companies for helping fund development of
 Asterisk:
 
 Pilosoft, Inc. - for supporting ADSI development in Asterisk
@@ -13,13 +13,8 @@ Telesthetic - for supporting SIP development
 
 Christos Ricudis - for substantial code contributions
 
-nic.at - ENUM support in Asterisk
-
 Paul Bagyenda, Digital Solutions - for initial Voicetronix driver development
 
-John Todd, TalkPlus, Inc.  and JR Richardson, Ntegrated Solutions. - for funding
-    the development of SIP Session Timers support.
-
 === WISHLIST CONTRIBUTERS ===
 Jeremy McNamara - SpeeX support
 Nick Seraphin - RDNIS support
@@ -30,185 +25,51 @@ Wasim - Hangup detect
 * Thanks to QuickNet Technologies for their donation of an Internet
 PhoneJack and Linejack card to the project.  (http://www.quicknet.net)
 
-* Thanks to VoipSupply for their donation of Sipura ATAs to the project for
-T.38 testing. (http://www.voipsupply.com)
-
-* Thanks to Grandstream for their donation of ATAs to the project for
-T.38 testing. (http://www.grandstream.com)
-
 === MISCELLANEOUS PATCHES ===
-Jim Dixon - Zapata Telephony and app_rpt
-	http://www.zapatatelephony.org/app_rpt.html
-
-Russell Bryant - Asterisk release manager and countless enhancements and bug
-	fixes.
-	russell(AT)digium.com
-
-Anthony Minessale II - Countless big and small fixes, and relentless forward
-	push. ChanSpy, ForkCDR, ControlPlayback, While/EndWhile, DumpChan, Dictate,
-	MacroIf, ExecIf, ExecIfTime, RetryDial, MixMonitor applications; many
-	realtime concepts and implementation pieces, including res_config_odbc;
-	format_slin; cdr_custom; several features in Dial including L(), G() and
-	enhancements to M() and D(); several CDR enhancements including CDR
-	variables; attended transfer; one touch record; native MOH; manager
-	eventmask; command line '-t' flag to allow recording/voicemail on nfs
-	shares; #exec command and multiline comments in config files; setvar in iax
-	and sip configs.
-	anthmct(AT)yahoo.com              http://www.asterlink.com
-
-James Golovich - Innumerable contributions, including SIP TCP and TLS support.
-	You can find him and asterisk-perl at http://asterisk.gnuinter.net
-
+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
-
+Oliver Daudey - ISDN4Linux fixes
+Pauline Middelink - ISDN4Linux patches and some general patches.
+       She can be found at http://www.polyware.nl/~middelink/En/
 Jean-Denis Girard - Various contributions from the South Pacific Islands
-	jd-girard(AT)esoft.pf             http://www.esoft.pf
-
-William Jordan / Vonage - MySQL enhancements to Voicemail
-	wjordan(AT)vonage.com
-
+       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(AT)basesys.com
-
+       critch@basesys.com
 Jefferson Noxon - app_lookupcidname, app_db, and various other contributions
-
 Klaus-Peter Junghanns - in-band DTMF on SIP and MGCP
-
 Ross Finlayson - Dynamic RTP payload support
-
 Mahmut Fettahlioglu - Audio recording, music-on-hold changes, alaw file
-	format, and various fixes. Can be contacted at mahmut(AT)oa.com.au
-
+       format, and various fixes. Can be contacted at mahmut@oa.com.au
 James Dennis - Cisco SIP compatibility patches to work with SIP service
-	providers. Can be contacted at asterisk(AT)jdennis.net
-
-Tilghman Lesher - ast_localtime(); ast_say_date_with_format(); 
-	GotoIfTime, SayUnixTime, HasNewVoicemail applications;
-	CUT, SORT, EVAL, CURL, FIELDQTY, STRFTIME, some QUEUE* functions;
-	func_odbc, cdr_adaptive_odbc, and other innumerable bug fixes.
-	tilghman(AT)digium.com            http://asterisk.drunkcoder.com/
-
+       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(AT)computingedge.net
-
+	jvantuyl@computingedge.net
 Thorsten Lockert - OpenBSD, FreeBSD ports, making MacOS X port run on 10.3,
-	dialplan include verification, route lookup on OpenBSD, SNMP agent
-	support (res_snmp), various other bugs. tholo(AT)sigmasoft.com
-
-Josh Roberson - chan_zap reload support, Advanced Voicemail Features, & other
-	misc. patches. - josh(AT)asteriasgi.com, http://www.asteriasgi.com
-
-William Waites - syslog support, SIP NAT traversal for SIP-UA. ww(AT)styx.org
-
+	dialplan include verification, route lookup on OpenBSD, various other
+	bugs. tholo@sigmasoft.com
+Brian West - ODBC support and Bug Marshaling  
+William Waites - syslog support, SIP NAT traversal for SIP-UA. ww@styx.org
 Rich Murphey - Porting to FreeBSD, NetBSD, OpenBSD, and Darwin.
-	rich(AT)whiteoaklabs.com  http://whiteoaklabs.com
-
-Simon Lockhart - Porting to Solaris (based on work of Logan ???)	
-	simon(AT)slimey.org
-
-Olle E. Johansson - SIP RFC compliance, documentation and testing, testing,
-	SIP outbound proxy support, Manager 1.1 update, SIP transfer support,
-	SIP presence support, SIP call state updates (dialog-info), 
-	MiniVM - the small voicemail system, many documentation
-	updates/corrections, and many bug fixes.
-	oej(AT)edvina.net, http://edvina.net
-
-Steve Kann - new jitter buffer for IAX2
-	stevek(AT)stevek.com
-
-Constantine Filin - major contributions to the Asterisk Realtime Architecture
-
-Steve Murphy - privacy support, $[ ] parser upgrade, AEL2 parser upgrade.
-	murf(AT)digium.com
-
-Claude Patry - bug fixes, feature enhancements, and bug marshalling
-	cpatry(AT)gmail.com
-
-Miroslav Nachev, miro(AT)space-comm.com COSMOS Software Enterprises, Ltd.
-	- for Variable for No Answer Timeout for Attended Transfer
-
-Slav Klenov & Vanheuverzwijn Joachim - development of the generic jitterbuffer
-	Securax Ltd. info(AT)securax.be
-
-Roy Sigurd Karlsbakk - providing funding for generic jitterbuffer development
-	roy(AT)karlsbakk.net, Briiz Telecom AS
-
-Voop AS, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - funding for rewrite
-	of SIP transfers
-
-Philippe Sultan - RADIUS CDR module, many fixes to res_jabber and gtalk/jingle
-	channel drivers.
-	INRIA, http://www.inria.fr/
-
-John Martin, Aupix - Improved video support in the SIP channel
-	T.140 text support in RTP/SIP
-
-Steve Underwood - Provided T.38 pass through support.
-
-George Konstantoulakis - Support for Greek in voicemail added by InAccess
-	Networks (work funded by HOL, www.hol.gr) gkon(AT)inaccessnetworks.com
-
-Daniel Nylander - Support for Swedish and Norwegian languages in voicemail.
-	http://www.danielnylander.se/
-
-Stojan Sljivic - An option for maximum number of messsages per mailbox in
-	voicemail.  Also an issue with voicemail synchronization has been fixed.
-	GDS Partners www.gdspartners.com .  stojan.sljivic(AT)gdspartners.com
-
-Bartosz Supczinski - Support for Polish added by DIR (www.dir.pl)
-	Bartosz.Supczinski(AT)dir.pl
-
-James Rothenberger - Support for IMAP storage integration added by
-	OneBizTone LLC Work funded by University of Pennsylvania jar(AT)onebiztone.com
-
-Paul Cadach - Bringing chan_h323 up to date, bug fixes, and more!
-
-Voop AS - Financial support for a lot of work with the SIP driver and the IAX
-	trunk MTU patch
-
-Cedric Hans - Development of chan_unistim
-  cedric.hans(AT)mlkj.net
-
-Sergio Fadda - console_video: video support for chan_oss and chan_alsa
-
-Marta Carbone - console_video and the astobj2 framework
-
-Luigi Rizzo - astobj2, console_video, windows build, chan_oss cleanup,
-	and a bunch of infrastructure work (loader, new_cli, ...)
-
-Brett Bryant - digit option for musiconhold selection, ENUMQUERY and ENUMRESULT functions,
-	feature group configuration for features.conf, per-file CLI debug and verbose settings,
-	TCP and TLS support for SIP, and various bug fixes.
-	brettbryant(AT)gmail.com
-
-Sergey Tamkovich - Realtime support for MusicOnHold, store and destroy realtime methods and
-	implementations for odbc, sqlite, and pgsql realtime drivers, attended transfer updates,
-	multiple speeds for ControlPlayback, and multiple bug fixes
-	- See http://voip-info.org/users/view/sergee
-	serg(AT)voipsolutions.ru
+	rich@whiteoaklabs.com  http://whiteoaklabs.com
 
 === OTHER CONTRIBUTIONS ===
 John Todd - Monkey sounds and associated teletorture prompt
-Michael Jerris - bug marshaling
-Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
-	available under a Creative Commons License at http://www.asteriskdocs.org
-Brian M. Clapper - poll.c emulation
-	This product includes software developed by Brian M. Clapper <bmc(AT)clapper.org>
 
 === HOLD MUSIC ===
 Music provided by www.opsound.org
 
 === OTHER SOURCE CODE IN ASTERISK ===
-Asterisk uses libedit, the lightweight readline replacement from NetBSD.
-The cdr_radius module uses libradiusclient-ng, which is also from NetBSD.
-They are BSD-licensed and require the following statement:
 
-      This product includes software developed by the NetBSD
-      Foundation, Inc. and its contributors.
-
-Digium did not implement the codecs in Asterisk.  Here is the copyright on the
+I did not implement the codecs in asterisk.  Here is the copyright on the
 GSM source:
 
 Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
@@ -233,7 +94,7 @@ And the copyright on the ADPCM source:
 Copyright 1992 by Stichting Mathematisch Centrum, Amsterdam, The
 Netherlands.
 
-	                 All Rights Reserved
+                        All Rights Reserved
 
 Permission to use, copy, modify, and distribute this software and its
 documentation for any purpose and without fee is hereby granted,

HARDWARE

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

LICENSE

@@ -1,62 +1,341 @@
-Asterisk is distributed under the GNU General Public License version 2
-and is also available under alternative licenses negotiated directly
-with Digium, Inc. If you obtained Asterisk under the GPL, then the GPL
-applies to all loadable Asterisk modules used on your system as well,
-except as defined below. The GPL (version 2) is included in this
-source tree in the file COPYING.
 
-This package also includes various components that are not part of
-Asterisk itself; these components are in the 'contrib' directory
-and its subdirectories. These components are also distributed under the
-GPL version 2 as well.
+		    GNU GENERAL PUBLIC LICENSE
+		       Version 2, June 1991
 
-Digium, Inc. (formerly Linux Support Services) holds copyright
-and/or sufficient licenses to all components of the Asterisk
-package, and therefore can grant, at its sole discretion, the ability
-for companies, individuals, or organizations to create proprietary or
-Open Source (even if not GPL) modules which may be dynamically linked at
-runtime with the portions of Asterisk which fall under our
-copyright/license umbrella, or are distributed under more flexible
-licenses than GPL.
+ Copyright (C) 1989, 1991 Free Software Foundation, Inc.
+                       59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
 
-If you wish to use our code in other GPL programs, don't worry --
-there is no requirement that you provide the same exception in your
-GPL'd products (although if you've written a module for Asterisk we
-would strongly encourage you to make the same exception that we do).
+			    Preamble
 
-Specific permission is also granted to link Asterisk with OpenSSL, OpenH323
-and/or the UW IMAP Toolkit and distribute the resulting binary files.
+  The licenses for most software are designed to take away your
+freedom to share and change it.  By contrast, the GNU General Public
+License is intended to guarantee your freedom to share and change free
+software--to make sure the software is free for all its users.  This
+General Public License applies to most of the Free Software
+Foundation's software and to any other program whose authors commit to
+using it.  (Some other Free Software Foundation software is covered by
+the GNU Library General Public License instead.)  You can apply it to
+your programs, too.
 
-In addition, Asterisk implements two management/control protocols: the
-Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface
-(AGI). It is our belief that applications using these protocols to
-manage or control an Asterisk instance do not have to be licensed
-under the GPL or a compatible license, as we believe these protocols
-do not create a 'derivative work' as referred to in the GPL. However,
-should any court or other judiciary body find that these protocols do
-fall under the terms of the GPL, then we hereby grant you a license to
-use these protocols in combination with Asterisk in external
-applications licensed under any license you wish.
+  When we speak of free software, we are referring to freedom, not
+price.  Our General Public Licenses are designed to make sure that you
+have the freedom to distribute copies of free software (and charge for
+this service if you wish), that you receive source code or can get it
+if you want it, that you can change the software or use pieces of it
+in new free programs; and that you know you can do these things.
 
-The 'Asterisk' name and logos are trademarks owned by Digium, Inc.,
-and use of them is subject to our trademark licensing policies. If you
-wish to use these trademarks for purposes other than simple
-redistribution of Asterisk source code obtained from Digium, you
-should contact our licensing department to determine the necessary
-steps you must take. For more information on this policy, please read:
+  To protect your rights, we need to make restrictions that forbid
+anyone to deny you these rights or to ask you to surrender the rights.
+These restrictions translate to certain responsibilities for you if you
+distribute copies of the software, or if you modify it.
 
-http://www.digium.com/en/company/profile/trademarkpolicy.php
+  For example, if you distribute copies of such a program, whether
+gratis or for a fee, you must give the recipients all the rights that
+you have.  You must make sure that they, too, receive or can get the
+source code.  And you must show them these terms so they know their
+rights.
 
-If you have any questions regarding our licensing policy, please
-contact us:
+  We protect your rights with two steps: (1) copyright the software, and
+(2) offer you this license which gives you legal permission to copy,
+distribute and/or modify the software.
 
-+1.877.344.4861 (via telephone in the USA)
-+1.256.428.6000 (via telephone outside the USA)
-+1.256.864.0464 (via FAX inside or outside the USA)
-IAX2/pbx.digium.com (via IAX2)
-licensing@digium.com (via email)
+  Also, for each author's protection and ours, we want to make certain
+that everyone understands that there is no warranty for this free
+software.  If the software is modified by someone else and passed on, we
+want its recipients to know that what they have is not the original, so
+that any problems introduced by others will not reflect on the original
+authors' reputations.
 
-Digium, Inc.
-445 Jan Davis Drive
-Huntsville, AL  35806
-USA
+  Finally, any free program is threatened constantly by software
+patents.  We wish to avoid the danger that redistributors of a free
+program will individually obtain patent licenses, in effect making the
+program proprietary.  To prevent this, we have made it clear that any
+patent must be licensed for everyone's free use or not licensed at all.
+
+  The precise terms and conditions for copying, distribution and
+modification follow.
+
+		    GNU GENERAL PUBLIC LICENSE
+   TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
+
+  0. This License applies to any program or other work which contains
+a notice placed by the copyright holder saying it may be distributed
+under the terms of this General Public License.  The "Program", below,
+refers to any such program or work, and a "work based on the Program"
+means either the Program or any derivative work under copyright law:
+that is to say, a work containing the Program or a portion of it,
+either verbatim or with modifications and/or translated into another
+language.  (Hereinafter, translation is included without limitation in
+the term "modification".)  Each licensee is addressed as "you".
+
+Activities other than copying, distribution and modification are not
+covered by this License; they are outside its scope.  The act of
+running the Program is not restricted, and the output from the Program
+is covered only if its contents constitute a work based on the
+Program (independent of having been made by running the Program).
+Whether that is true depends on what the Program does.
+
+  1. You may copy and distribute verbatim copies of the Program's
+source code as you receive it, in any medium, provided that you
+conspicuously and appropriately publish on each copy an appropriate
+copyright notice and disclaimer of warranty; keep intact all the
+notices that refer to this License and to the absence of any warranty;
+and give any other recipients of the Program a copy of this License
+along with the Program.
+
+You may charge a fee for the physical act of transferring a copy, and
+you may at your option offer warranty protection in exchange for a fee.
+
+  2. You may modify your copy or copies of the Program or any portion
+of it, thus forming a work based on the Program, and copy and
+distribute such modifications or work under the terms of Section 1
+above, provided that you also meet all of these conditions:
+
+    a) You must cause the modified files to carry prominent notices
+    stating that you changed the files and the date of any change.
+
+    b) You must cause any work that you distribute or publish, that in
+    whole or in part contains or is derived from the Program or any
+    part thereof, to be licensed as a whole at no charge to all third
+    parties under the terms of this License.
+
+    c) If the modified program normally reads commands interactively
+    when run, you must cause it, when started running for such
+    interactive use in the most ordinary way, to print or display an
+    announcement including an appropriate copyright notice and a
+    notice that there is no warranty (or else, saying that you provide
+    a warranty) and that users may redistribute the program under
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+    License.  (Exception: if the Program itself is interactive but
+    does not normally print such an announcement, your work based on
+    the Program is not required to print an announcement.)
+
+These requirements apply to the modified work as a whole.  If
+identifiable sections of that work are not derived from the Program,
+and can be reasonably considered independent and separate works in
+themselves, then this License, and its terms, do not apply to those
+sections when you distribute them as separate works.  But when you
+distribute the same sections as part of a whole which is a work based
+on the Program, the distribution of the whole must be on the terms of
+this License, whose permissions for other licensees extend to the
+entire whole, and thus to each and every part regardless of who wrote it.
+
+Thus, it is not the intent of this section to claim rights or contest
+your rights to work written entirely by you; rather, the intent is to
+exercise the right to control the distribution of derivative or
+collective works based on the Program.
+
+In addition, mere aggregation of another work not based on the Program
+with the Program (or with a work based on the Program) on a volume of
+a storage or distribution medium does not bring the other work under
+the scope of this License.
+
+  3. You may copy and distribute the Program (or a work based on it,
+under Section 2) in object code or executable form under the terms of
+Sections 1 and 2 above provided that you also do one of the following:
+
+    a) Accompany it with the complete corresponding machine-readable
+    source code, which must be distributed under the terms of Sections
+    1 and 2 above on a medium customarily used for software interchange; or,
+
+    b) Accompany it with a written offer, valid for at least three
+    years, to give any third party, for a charge no more than your
+    cost of physically performing source distribution, a complete
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+    customarily used for software interchange; or,
+
+    c) Accompany it with the information you received as to the offer
+    to distribute corresponding source code.  (This alternative is
+    allowed only for noncommercial distribution and only if you
+    received the program in object code or executable form with such
+    an offer, in accord with Subsection b above.)
+
+The source code for a work means the preferred form of the work for
+making modifications to it.  For an executable work, complete source
+code means all the source code for all modules it contains, plus any
+associated interface definition files, plus the scripts used to
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+operating system on which the executable runs, unless that component
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+
+If distribution of executable or object code is made by offering
+access to copy from a designated place, then offering equivalent
+access to copy the source code from the same place counts as
+distribution of the source code, even though third parties are not
+compelled to copy the source along with the object code.
+
+  4. You may not copy, modify, sublicense, or distribute the Program
+except as expressly provided under this License.  Any attempt
+otherwise to copy, modify, sublicense or distribute the Program is
+void, and will automatically terminate your rights under this License.
+However, parties who have received copies, or rights, from you under
+this License will not have their licenses terminated so long as such
+parties remain in full compliance.
+
+  5. You are not required to accept this License, since you have not
+signed it.  However, nothing else grants you permission to modify or
+distribute the Program or its derivative works.  These actions are
+prohibited by law if you do not accept this License.  Therefore, by
+modifying or distributing the Program (or any work based on the
+Program), you indicate your acceptance of this License to do so, and
+all its terms and conditions for copying, distributing or modifying
+the Program or works based on it.
+
+  6. Each time you redistribute the Program (or any work based on the
+Program), the recipient automatically receives a license from the
+original licensor to copy, distribute or modify the Program subject to
+these terms and conditions.  You may not impose any further
+restrictions on the recipients' exercise of the rights granted herein.
+You are not responsible for enforcing compliance by third parties to
+this License.
+
+  7. If, as a consequence of a court judgment or allegation of patent
+infringement or for any other reason (not limited to patent issues),
+conditions are imposed on you (whether by court order, agreement or
+otherwise) that contradict the conditions of this License, they do not
+excuse you from the conditions of this License.  If you cannot
+distribute so as to satisfy simultaneously your obligations under this
+License and any other pertinent obligations, then as a consequence you
+may not distribute the Program at all.  For example, if a patent
+license would not permit royalty-free redistribution of the Program by
+all those who receive copies directly or indirectly through you, then
+the only way you could satisfy both it and this License would be to
+refrain entirely from distribution of the Program.
+
+If any portion of this section is held invalid or unenforceable under
+any particular circumstance, the balance of the section is intended to
+apply and the section as a whole is intended to apply in other
+circumstances.
+
+It is not the purpose of this section to induce you to infringe any
+patents or other property right claims or to contest validity of any
+such claims; this section has the sole purpose of protecting the
+integrity of the free software distribution system, which is
+implemented by public license practices.  Many people have made
+generous contributions to the wide range of software distributed
+through that system in reliance on consistent application of that
+system; it is up to the author/donor to decide if he or she is willing
+to distribute software through any other system and a licensee cannot
+impose that choice.
+
+This section is intended to make thoroughly clear what is believed to
+be a consequence of the rest of this License.
+
+  8. If the distribution and/or use of the Program is restricted in
+certain countries either by patents or by copyrighted interfaces, the
+original copyright holder who places the Program under this License
+may add an explicit geographical distribution limitation excluding
+those countries, so that distribution is permitted only in or among
+countries not thus excluded.  In such case, this License incorporates
+the limitation as if written in the body of this License.
+
+  9. The Free Software Foundation may publish revised and/or new versions
+of the General Public License from time to time.  Such new versions will
+be similar in spirit to the present version, but may differ in detail to
+address new problems or concerns.
+
+Each version is given a distinguishing version number.  If the Program
+specifies a version number of this License which applies to it and "any
+later version", you have the option of following the terms and conditions
+either of that version or of any later version published by the Free
+Software Foundation.  If the Program does not specify a version number of
+this License, you may choose any version ever published by the Free Software
+Foundation.
+
+  10. If you wish to incorporate parts of the Program into other free
+programs whose distribution conditions are different, write to the author
+to ask for permission.  For software which is copyrighted by the Free
+Software Foundation, write to the Free Software Foundation; we sometimes
+make exceptions for this.  Our decision will be guided by the two goals
+of preserving the free status of all derivatives of our free software and
+of promoting the sharing and reuse of software generally.
+
+			    NO WARRANTY
+
+  11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
+FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW.  EXCEPT WHEN
+OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
+PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED
+OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE.  THE ENTIRE RISK AS
+TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU.  SHOULD THE
+PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,
+REPAIR OR CORRECTION.
+
+  12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
+WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
+REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,
+INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING
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+TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY
+YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER
+PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGES.
+
+		     END OF TERMS AND CONDITIONS
+
+	    How to Apply These Terms to Your New Programs
+
+  If you develop a new program, and you want it to be of the greatest
+possible use to the public, the best way to achieve this is to make it
+free software which everyone can redistribute and change under these terms.
+
+  To do so, attach the following notices to the program.  It is safest
+to attach them to the start of each source file to most effectively
+convey the exclusion of warranty; and each file should have at least
+the "copyright" line and a pointer to where the full notice is found.
+
+    <one line to give the program's name and a brief idea of what it does.>
+    Copyright (C) 19yy  <name of author>
+
+    This program is free software; you can redistribute it and/or modify
+    it under the terms of the GNU General Public License as published by
+    the Free Software Foundation; either version 2 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License
+    along with this program; if not, write to the Free Software
+    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+
+
+Also add information on how to contact you by electronic and paper mail.
+
+If the program is interactive, make it output a short notice like this
+when it starts in an interactive mode:
+
+    Gnomovision version 69, Copyright (C) 19yy name of author
+    Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
+    This is free software, and you are welcome to redistribute it
+    under certain conditions; type `show c' for details.
+
+The hypothetical commands `show w' and `show c' should show the appropriate
+parts of the General Public License.  Of course, the commands you use may
+be called something other than `show w' and `show c'; they could even be
+mouse-clicks or menu items--whatever suits your program.
+
+You should also get your employer (if you work as a programmer) or your
+school, if any, to sign a "copyright disclaimer" for the program, if
+necessary.  Here is a sample; alter the names:
+
+  Yoyodyne, Inc., hereby disclaims all copyright interest in the program
+  `Gnomovision' (which makes passes at compilers) written by James Hacker.
+
+  <signature of Ty Coon>, 1 April 1989
+  Ty Coon, President of Vice
+
+This General Public License does not permit incorporating your program into
+proprietary programs.  If your program is a subroutine library, you may
+consider it more useful to permit linking proprietary applications with the
+library.  If this is what you want to do, use the GNU Library General
+Public License instead of this License.

Makefile.moddir_rules

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

Makefile.rules

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

README

@@ -1,171 +1,108 @@
-===============================================================================
-===                     The Asterisk(R) Open Source PBX
-===
-===                   by Mark Spencer <markster@digium.com>
-===                  and the Asterisk.org developer community
-===
-===                    Copyright (C) 2001-2008 Digium, Inc.
-===                       and other copyright holders.
-===============================================================================
-
--------------------------------------------------------------------------------
---- SECURITY ------------------------------------------------------------------
-
+The Asterisk Open Source PBX
+by Mark Spencer <markster@digium.com>
+Copyright (C) 2001-2004 Digium
+================================================================
+* SECURITY
   It is imperative that you read and fully understand the contents of
-the security information document before you attempt to configure and run
-an Asterisk server.
-
-  If you downloaded Asterisk as a tarball, see the security section in the PDF
-version of the documentation in doc/tex/asterisk.pdf.  Alternatively, pull up
-the HTML version of the documentation in doc/tex/asterisk/index.html.  The
-source for the security document is available in doc/tex/security.tex.
--------------------------------------------------------------------------------
-
--------------------------------------------------------------------------------
---- WHAT IS ASTERISK ? --------------------------------------------------------
+  the SECURITY file before you attempt to configure an Asterisk server.
 
+* WHAT IS ASTERISK
   Asterisk is an Open Source PBX and telephony toolkit.  It is, in a
 sense, middleware between Internet and telephony channels on the bottom,
-and Internet and telephony applications at the top.  However, Asterisk supports
-more telephony interfaces than just Internet telephony.  Asterisk also has a
-vast amount of support for traditional PSTN telephony, as well.  For more
-information on the project itself, please visit the Asterisk home page at:
+and Internet and telephony applications at the top.  For more information
+on the project itself, please visit the Asterisk home page at:
 
            http://www.asterisk.org
 
-  In addition you'll find lots of information compiled by the Asterisk
+In addition you'll find lot's of information compiled by the Asterisk
 community on this Wiki:
 
            http://www.voip-info.org/wiki-Asterisk
 
-  There is a book on Asterisk published by O'Reilly under the Creative Commons
-License. It is available in book stores as well as in a downloadable version on
-the http://www.asteriskdocs.org web site.
--------------------------------------------------------------------------------
+* LICENSING
+  Asterisk is distributed under GNU General Public License.  The GPL also
+must apply to all loadable modules as well, except as defined below.
 
--------------------------------------------------------------------------------
---- SUPPORTED OPERATING SYSTEMS -----------------------------------------------
+  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.  
 
---- Linux
-  The Asterisk Open Source PBX is developed and tested primarily on the
-GNU/Linux operating system, and is supported on every major GNU/Linux
-distribution.
 
---- Others
-  Asterisk has also been 'ported' and reportedly runs properly on other
-operating systems as well, including Sun Solaris, Apple's Mac OS X, Cygwin,
-and the BSD variants.
--------------------------------------------------------------------------------
+  If you wish to use our code in other GPL programs, don't worry -- there
+is no requirement that you provide the same exemption in your GPL'd
+products (although if you've written a module for Asterisk we would
+strongly encourage you to make the same exemption that we do).
 
--------------------------------------------------------------------------------
---- GETTING STARTED -----------------------------------------------------------
+  Specific permission is also granted to OpenSSL and OpenH323 to link to
+Asterisk.
 
-  First, be sure you've got supported hardware (but note that you don't need
-ANY special hardware, not even a sound card) to install and run Asterisk.
+  If you have any questions, whatsoever, regarding our licensing policy,
+please contact us.
 
-  Supported telephony hardware includes:
+  Modules that are GPL-licensed and not available under Digium's 
+licensing scheme are added to the Asterisk-addons CVS module.
+  
+* REQUIRED COMPONENTS
 
-	* All Analog and Digital Interface cards from Digium (www.digium.com)
+== Linux ==
+  Currently, the Asterisk Open Source PBX is only known to run on the
+Linux OS, although it may be portable to other UNIX-like operating systems
+(like FreeBSD) as well. 
+
+
+* GETTING STARTED
+
+First, be sure you've got supported hardware (but note that you don't need ANY hardware, not even a soundcard) to install and run Asterisk. Supported are:
+
+	* All Wildcard (tm) products from Digium (www.digium.com)
 	* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
-	* any full duplex sound card supported by ALSA, OSS, or PortAudio
-	* any ISDN card supported by mISDN on Linux (BRI)
-	* The Xorcom AstriBank channel bank
-	* VoiceTronix OpenLine products
+	* Full Duplex Sound Card supported by Linux
+	* Adtran Atlas 800 Plus
+	* ISDN4Linux compatible ISDN card
+	* Tormenta Dual T1 card (www.bsdtelephony.com.mx)
 
--------------------------------------------------------------------------------
+Hint: CAPI compatible ISDN cards can be run using the add-on channel chan_capi.
 
--------------------------------------------------------------------------------
---- UPGRADING FROM AN EARLIER VERSION -----------------------------------------
+So let's proceed:
 
-  If you are updating from a previous version of Asterisk, make sure you
-read the UPGRADE.txt file in the source directory. There are some files
-and configuration options that you will have to change, even though we
-made every effort possible to maintain backwards compatibility.
+1) Run "make"
+2) Run "make install"
 
-  In order to discover new features to use, please check the configuration
-examples in the /configs directory of the source code distribution.  For a
-list of new features in this version of Asterisk, see the CHANGES file.
--------------------------------------------------------------------------------
-
--------------------------------------------------------------------------------
---- NEW INSTALLATIONS ---------------------------------------------------------
-
-  Ensure that your system contains a compatible compiler and development
-libraries.  Asterisk requires either the GNU Compiler Collection (GCC) version
-3.0 or higher, or a compiler that supports the C99 specification and some of
-the gcc language extensions.  In addition, your system needs to have the C
-library headers available, and the headers and libraries for ncurses.
-
-  There are many modules that have additional dependencies.  To see what
-libraries are being looked for, see ./configure --help, or run
-"make menuselect" to view the dependencies for specific modules.
-
-  On many distributions, these dependencies are installed by packages with names
-like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' 
-or similar.
-
-  So, let's proceed:
-
-1) Read this README file.
-
-  There are more documents than this one in the doc/ directory.  You may also
-want to check the configuration files that contain examples and reference
-guides. They are all in the configs/ directory.
-
-2) Run "./configure"
-
-  Execute the configure script to guess values for system-dependent
-variables used during compilation.
-
-3) Run "make menuselect" [optional]
-
-  This is needed if you want to select the modules that will be compiled and to
-check dependencies for various optional modules.
-
-4) Run "make"
-
-  Assuming the build completes successfully:
-
-5) Run "make install"
-
-  If this is your first time working with Asterisk, you may wish to install
+If this is your first time working with Asterisk, you may wish to install
 the sample PBX, with demonstration extensions, etc.  If so, run:
 
-6) "make samples"
+	"make samples"
 
-  Doing so will overwrite any existing configuration files you have installed.
+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 in the foreground mode (not a daemon) with:
+Finally, you can launch Asterisk with:
 
-# asterisk -vvvc
+	./asterisk -vvvc
 
-  You'll see a bunch of verbose messages fly by your screen as Asterisk
+You'll see a bunch of verbose messages fly by your screen as Asterisk
 initializes (that's the "very very verbose" mode).  When it's ready, if
 you specified the "c" then you'll get a command line console, that looks
 like this:
 
 *CLI>
 
-  You can type "help" at any time to get help with the system.  For help
+You can type "help" at any time to get help with the system.  For help
 with a specific command, type "help <command>".  To start the PBX using
-your sound card, you can type "console dial" to dial the PBX.  Then you can use
-"console answer", "console hangup", and "console dial" to simulate the actions
-of a telephone.  Remember that if you don't have a full duplex sound card
-(and Asterisk will tell you somewhere in its verbose messages if you do/don't)
-then it won't work right (not yet).
+your sound card, you can type "dial" to dial the PBX.  Then you can use
+"answer", "hangup", and "dial" to simulate the actions of a telephone.
+Remember that if you don't have a full duplex sound card (And asterisk
+will tell you somewhere in its verbose messages if you do/don't) than it
+won't work right (not yet).
 
-  "man asterisk" at the Unix/Linux command prompt will give you detailed
-information on how to start and stop Asterisk, as well as all the command
-line options for starting Asterisk.
+Feel free to look over the configuration files in /etc/asterisk, where
+you'll find a lot of information about what you can do with Asterisk.
 
-  Feel free to look over the configuration files in /etc/asterisk, where you
-will find a lot of information about what you can do with Asterisk.
--------------------------------------------------------------------------------
+* ABOUT CONFIGURATION FILES
 
--------------------------------------------------------------------------------
---- ABOUT CONFIGURATION FILES -------------------------------------------------
-
-  All Asterisk configuration files share a common format.  Comments are
+All Asterisk configuration files share a common format.  Comments are
 delimited by ';' (since '#' of course, being a DTMF digit, may occur in
 many places).  A configuration file is divided into sections whose names
 appear in []'s.  Each section typically contains two types of statements,
@@ -174,12 +111,12 @@ parameters'.  Internally the use of '=' and '=>' is exactly the same, so
 they're used only to help make the configuration file easier to
 understand, and do not affect how it is actually parsed.
 
-  Entries of the form 'variable=value' set the value of some parameter in
-asterisk.  For example, in dahdi.conf, one might specify:
+Entries of the form 'variable=value' set the value of some parameter in
+asterisk.  For example, in tormenta.conf, one might specify:
 
 	switchtype=national
 
-  In order to indicate to Asterisk that the switch they are connecting to is
+In order to indicate to Asterisk that the switch they are connecting to is
 of the type "national".  In general, the parameter will apply to
 instantiations which occur below its specification.  For example, if the
 configuration file read:
@@ -190,103 +127,26 @@ configuration file read:
 	switchtype = dms100
 	channel => 25-47
 
-  The "national" switchtype would be applied to channels one through
+Then, the "national" switchtype would be applied to channels one through
 four and channels 10 through 12, whereas the "dms100" switchtype would
 apply to channels 25 through 47.
   
-  The "object => parameters" instantiates an object with the given
+The "object => parameters" instantiates an object with the given
 parameters.  For example, the line "channel => 25-47" creates objects for
-the channels 25 through 47 of the card, obtaining the settings
+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 that Asterisk is sensitive to
-large jumps in time.  Manually changing the system time using date(1)
-(or other similar commands) may cause SIP registrations and other
-internal processes to fail.  If your system cannot keep accurate time
-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.
+* MORE INFORMATION
 
-  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.
+See the doc directory for more documentation.
 
-  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 approximately 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 on various features. Again,
-please read all the configuration samples that include documentation on
-the configuration options.
-
-  If this release of Asterisk was downloaded from a tarball, then some
-additional documentation should have been included.
-     * doc/tex/asterisk.pdf --- PDF version of the documentation
-	 * doc/tex/asterisk/index.html --- HTML version of the documentation
-
-  Finally, you may wish to visit the web site and join the mailing list if
+Finally, you may wish to visit the web site and join the mailing list if
 you're interested in getting more information.
 
-   http://www.asterisk.org/support
+   http://www.asterisk.org/index.php?menu=support
 
-  Welcome to the growing worldwide community of Asterisk users!
--------------------------------------------------------------------------------
+Welcome to the growing worldwide community of Asterisk users!
+
+Mark Spencer
 
---- Mark Spencer, and the Asterisk.org development community
 
--------------------------------------------------------------------------------
-Asterisk is a trademark of Digium, Inc.

README-SERIOUSLY.bestpractices.txt

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

README.opsound

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

SECURITY

@@ -0,0 +1,41 @@
+==== Security Notes with Asterisk ====
+
+PLEASE READ THE FOLLOWING IMPORTANT SECURITY RELATED INFORMATION.  
+IMPROPER CONFIGURATION OF ASTERISK COULD ALLOW UNAUTHORIZED USE OF YOUR
+FACILITIES, POTENTIALLY INCURRING SUBSTANTIAL CHARGES.
+
+First and foremost remember this:
+
+USE THE EXTENSION CONTEXTS TO ISOLATE OUTGOING OR TOLL SERVICES FROM ANY
+INCOMING CONNECTIONS.
+
+You should consider that if any channel, incoming line, etc can enter an
+extension context that it has the capability of accessing any extension
+within that context. 
+
+Therefore, you should NOT allow access to outgoing or toll services in
+contexts that are accessible (especially without a password) from incoming
+channels, be they IAX channels, FX or other trunks, or even untrusted
+stations within you network.  In particular, never ever put outgoing toll
+services in the "default" context.  To make things easier, you can include
+the "default" context within other private contexts by using:
+
+	include => default
+
+in the appropriate section.  A well designed PBX might look like this:
+
+[longdistance]
+exten => _91NXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1})
+include => local
+
+[local]
+exten => _9NXXNXXX,1,Dial(Zap/g2/${EXTEN:1})
+include => default
+
+[default]
+exten => 6123,Dial(Zap/1)
+
+
+DON'T FORGET TO TAKE THE DEMO CONTEXT OUT OF YOUR DEFAULT CONTEXT.  There
+isn't really a security reason, it just will keep people from wanting to 
+play with your asterisk setup remotely.

UPGRADE-1.2.txt

@@ -1,218 +0,0 @@
-=========================================================
-===
-=== Information for upgrading from Asterisk 1.0 to 1.2
-===
-=== This file documents all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also includes advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=========================================================
-
-Compiling:
-
-* The Asterisk 1.2 source code now uses C language features
-  supported only by 'modern' C compilers.  Generally, this means GCC
-  version 3.0 or higher, although some GCC 2.96 releases will also
-  work.  Some non-GCC compilers that support C99 and the common GCC
-  extensions (including anonymous structures and unions) will also
-  work.  All releases of GCC 2.95 do _not_ have the requisite feature
-  support; systems using that compiler will need to be upgraded to
-  a more recent compiler release.
-
-Dialplan Expressions:
-
-* The dialplan expression parser (which handles $[ ... ] constructs)
-  has gone through a major upgrade, but has one incompatible change:
-  spaces are no longer required around expression operators, including
-  string comparisons. However, you can now use quoting to keep strings
-  together for comparison. For more details, please read the
-  doc/README.variables file, and check over your dialplan for possible
-  problems.
-
-Agents:
-
-* The default for ackcall has been changed to "no" instead of "yes" 
-  because of a bug which caused the "yes" behavior to generally act like
-  "no".  You may need to adjust the value if your agents behave 
-  differently than you expect with respect to acknowledgement.
-
-* The AgentCallBackLogin application now requires a second '|' before
-  specifying an extension@context.  This is to distinguish the options
-  string from the extension, so that they do not conflict.  See
-  'show application AgentCallbackLogin' for more details.
-
-Parking:
-
-* Parking behavior has changed slightly; when a parked call times out,
-  Asterisk will attempt to deliver the call back to the extension that
-  parked it, rather than the 's' extension. If that extension is busy
-  or unavailable, the parked call will be lost.
-
-Dialing:
-
-* The Caller*ID of the outbound leg is now the extension that was 
-  called, rather than the Caller*ID of the inbound leg of the call.  The 
-  "o" flag for Dial can be used to restore the original behavior if 
-  desired.  Note that if you are looking for the originating callerid
-  from the manager event, there is a new manager event "Dial" which 
-  provides the source and destination channels and callerid.
-
-IAX: 
-
-* The naming convention for IAX channels has changed in two ways: 
-   1. The call number follows a "-" rather than a "/" character.
-   2. The name of the channel has been simplified to IAX2/peer-callno,
-   rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.
-
-SIP:
-
-* The global option "port" in 1.0.X that is used to set which port to
-  bind to has been changed to "bindport" to be more consistent with
-  the other channel drivers and to avoid confusion with the "port"
-  option for users/peers.
-
-* The "Registry" event now uses "Username" rather than "User" for 
-  consistency with IAX.
-
-Applications:
-
-* With the addition of dialplan functions (which operate similarly
-  to variables), the SetVar application has been renamed to Set.
-
-* The CallerPres application has been removed.  Use SetCallerPres 
-  instead.  It accepts both numeric and symbolic names.
-
-* The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
-  CheckGroup have been deprecated in favor of functions.  Here is a
-  table of their replacements:
-
-  GetGroupCount([groupname][@category]	       GROUP_COUNT([groupname][@category])	Set(GROUPCOUNT=${GROUP_COUNT()})
-  GroupMatchCount(groupmatch[@category])       GROUP_MATCH_COUNT(groupmatch[@category])	Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
-  SetGroup(groupname[@category])	       GROUP([category])=groupname		Set(GROUP()=test)
-  CheckGroup(max[@category])		       N/A					GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)
-
-  Note that CheckGroup does not have a direct replacement.  There is
-  also a new function called GROUP_LIST() which will return a space
-  separated list of all of the groups set on a channel.  The GROUP()
-  function can also return the name of the group set on a channel when
-  used in a read environment.
-
-* The applications DBGet and DBPut have been deprecated in favor of
-  functions.  Here is a table of their replacements:
-
-  DBGet(foo=family/key)        Set(foo=${DB(family/key)})
-  DBPut(family/key=${foo})     Set(DB(family/key)=${foo})
-
-* The application SetLanguage has been deprecated in favor of the
-  function LANGUAGE().
-
-  SetLanguage(fr)		Set(LANGUAGE()=fr)
-
-  The LANGUAGE function can also return the currently set language:
-
-  Set(MYLANG=${LANGUAGE()})
-
-* The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
-  have been deprecated in favor of the function TIMEOUT(timeouttype):
-
-  AbsoluteTimeout(300)		Set(TIMEOUT(absolute)=300)
-  DigitTimeout(15)		Set(TIMEOUT(digit)=15)
-  ResponseTimeout(15)		Set(TIMEOUT(response)=15)
-
-  The TIMEOUT() function can also return the currently set timeouts:
-
-  Set(DTIMEOUT=${TIMEOUT(digit)})
-
-* The applications SetCIDName, SetCIDNum, and SetRDNIS have been
-  deprecated in favor of the CALLERID(datatype) function:
-
-  SetCIDName(Joe Cool)		Set(CALLERID(name)=Joe Cool)
-  SetCIDNum(2025551212)		Set(CALLERID(number)=2025551212)
-  SetRDNIS(2024561414)		Set(CALLERID(RDNIS)=2024561414)
-
-* The application Record now uses the period to separate the filename
-  from the format, rather than the colon.
-
-* The application VoiceMail now supports a 'temporary' greeting for each
-  mailbox. This greeting can be recorded by using option 4 in the
-  'mailbox options' menu, and 'change your password' option has been
-  moved to option 5.
-
-* The application VoiceMailMain now only matches the 'default' context if
-  none is specified in the arguments.  (This was the previously 
-  documented behavior, however, we didn't follow that behavior.)  The old
-  behavior can be restored by setting searchcontexts=yes in voicemail.conf.
-
-Queues:
-
-* A queue is now considered empty not only if there are no members but if
-  none of the members are available (e.g. agents not logged on).  To
-  restore the original behavior, use "leavewhenempty=strict" or 
-  "joinwhenempty=strict" instead of "=yes" for those options.
-
-* It is now possible to use multi-digit extensions in the exit context
-  for a queue (although you should not have overlapping extensions,
-  as there is no digit timeout). This means that the EXITWITHKEY event
-  in queue_log can now contain a key field with more than a single
-  character in it.
-
-Extensions:
-
-* By default, there is a new option called "autofallthrough" in
-  extensions.conf that is set to yes.  Asterisk 1.0 (and earlier) 
-  behavior was to wait for an extension to be dialed after there were no 
-  more extensions to execute.  "autofallthrough" changes this behavior
-  so that the call will immediately be terminated with BUSY,
-  CONGESTION, or HANGUP based on Asterisk's best guess.  If you are
-  writing an extension for IVR, you must use the WaitExten application
-  if "autofallthrough" is set to yes.
-
-AGI:
-
-* AGI scripts did not always get SIGHUP at the end, previously.  That 
-  behavior has been fixed.  If you do not want your script to terminate 
-  at the end of AGI being called (e.g. on a hangup) then set SIGHUP to 
-  be ignored within your application.
-
-* CallerID is reported with agi_callerid and agi_calleridname instead
-  of a single parameter holding both.
-
-Music On Hold:
-
-* The preferred format for musiconhold.conf has changed; please see the
-  sample configuration file for the new format. The existing format
-  is still supported but will generate warnings when the module is loaded.
-
-chan_modem:
-
-* All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
-  in this release, and will be removed in the next major Asterisk release.
-  Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
-  path for aopen and bestdata modem users.
-
-MeetMe:
-
-* The conference application now allows users to increase/decrease their
-  speaking volume and listening volume (independently of each other and 
-  other users); the 'admin' and 'user' menus have changed, and new sound 
-  files are included with this release. However, if a user calling in 
-  over a Zaptel channel that does NOT have hardware DTMF detection 
-  increases their speaking volume, it is likely they will no longer be 
-  able to enter/exit the menu or make any further adjustments, as the  
-  software DTMF detector will not be able to recognize the DTMF coming 
-  from their device.
-
-GetVar Manager Action:
-
-* Previously, the behavior of the GetVar manager action reported the value
-  of a variable in the following manner:
-   > name: value
-  This has been changed to a manner similar to the SetVar action and is now
-   > Variable: name
-   > Value: value

UPGRADE-1.4.txt

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

UPGRADE-1.6.txt

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

UPGRADE.txt

@@ -1,88 +0,0 @@
-===========================================================
-===
-=== Information for upgrading between Asterisk 1.6 versions
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also includes advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
-===
-===========================================================
-
-As of 1.6.0.16:
-
-* The firmware for the IAXy has been removed from Asterisk.  It can be
-  downloaded from http://downloads.digium.com/pub/iaxy/.  To have Asterisk
-  install the firmware into its proper location, place the firmware in the
-  contrib/firmware/iax/ directory in the Asterisk source tree before running
-  "make install".
-
-* T.38 FAX error correction mode can no longer be configured in udptl.conf;
-  instead, it is configured on a per-peer (or global) basis in sip.conf, with
-  the same default as was present in udptl.conf.sample.
-
-* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
-  instead, it is either supplied by the application servicing the T.38 channel
-  (for a FAX send or receive) or calculated from the bridged endpoint's
-  maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
-  allows for overriding the value supplied by a remote endpoint, which is useful
-  when T.38 connections are made to gateways that supply incorrectly-calculated
-  maximum datagram sizes.
-
-As of 1.6.0.15:
-
-* There have been some changes to the IAX2 protocol to address the security
-  concerns documented in the security advisory AST-2009-006.  Please see the
-  IAX2 security document, doc/IAX2-security.pdf, for information regarding
-  backwards compatibility with versions of Asterisk that do not contain these
-  changes to IAX2.
-
-From 1.6.0.10 to 1.6.0.11:
-
-* Beginning with this release, Asterisk's internal methods of
-  negotiating T.38 (FAX over IP) sessions changed in
-  non-backwards-compatible ways. Any applications that previously used
-  AST_CONTROL_T38 control frames will have to be upgraded to use
-  AST_CONTROL_T38_PARAMETERS control frames instead; app_fax.c is a good
-  example of how to generate and respond to these frames. These changes
-  were made to solve significant T.38 interoperability problems between
-  Asterisk and various SIP/T.38 endpoints identified by many users of
-  Asterisk.
-
-From 1.6.0.9 to 1.6.0.10:
-
-* Support for Taiwanese was incorrectly supported with the "tw" language code.
-  In reality, the "tw" language code is reserved for the Twi language, native
-  to Ghana.  If you were previously using the "tw" language code, you should
-  switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
-  specific localizations.  Additionally, "mx" should be changed to "es_MX",
-  Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
-  "cs", not "cz".
-
-From 1.6.0.1 to 1.6.0.2:
-
-* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
-  API calls were added in 1.6.0, so that modules that provide multiple
-  AGI commands could register/unregister them all with a single
-  step. However, these API calls were not implemented properly, and did
-  not allow the caller to know whether registration or unregistration
-  succeeded or failed. They have been redefined to now return success
-  or failure, but this means any code using these functions will need
-  be recompiled after upgrading to a version of Asterisk containing
-  these changes. In addition, the source code using these functions
-  should be reviewed to ensure it can properly react to failure
-  of registration or unregistration of its API commands.
-
-* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
-  to better match what it really does, and the argument order has been
-  changed to be consistent with other API calls that perform similar
-  operations.

Zaptel-to-DAHDI.txt

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

acl.c

@@ -0,0 +1,409 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Various sorts of access control
+ * 
+ * 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 <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/time.h>
+#include <signal.h>
+#include <errno.h>
+#include <unistd.h>
+#include <asterisk/acl.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/utils.h>
+#include <asterisk/lock.h>
+#include <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
+
+
+struct ast_ha {
+	/* Host access rule */
+	struct in_addr netaddr;
+	struct in_addr netmask;
+	int sense;
+	struct ast_ha *next;
+};
+
+/* Default IP - if not otherwise set, don't breathe garbage */
+static struct in_addr __ourip = { 0x00000000 };
+
+struct my_ifreq {
+	char ifrn_name[IFNAMSIZ];	/* Interface name, e.g. "eth0", "ppp0", etc.  */
+	struct sockaddr_in ifru_addr;
+};
+
+/* Free HA structure */
+void ast_free_ha(struct ast_ha *ha)
+{
+	struct ast_ha *hal;
+	while(ha) {
+		hal = ha;
+		ha = ha->next;
+		free(hal);
+	}
+}
+
+/* Copy HA structure */
+static void ast_copy_ha(struct ast_ha *from, struct ast_ha *to)
+{
+	memcpy(&to->netaddr, &from->netaddr, sizeof(from->netaddr));
+	memcpy(&to->netmask, &from->netmask, sizeof(from->netmask));
+	to->sense = from->sense;
+}
+
+/* Create duplicate of ha structure */
+static struct ast_ha *ast_duplicate_ha(struct ast_ha *original)
+{
+	struct ast_ha *new_ha = malloc(sizeof(struct ast_ha));
+
+	/* Copy from original to new object */
+	ast_copy_ha(original, new_ha); 
+
+	return(new_ha);
+
+}
+
+/* Create duplicate HA link list */
+/*  Used in chan_sip2 templates */
+struct ast_ha *ast_duplicate_ha_list(struct ast_ha *original)
+{
+	struct ast_ha *start=original;
+	struct ast_ha *ret = NULL;
+	struct ast_ha *link,*prev=NULL;
+
+	while(start) {
+		link = ast_duplicate_ha(start);  /* Create copy of this object */
+		if (prev)
+			prev->next = link;		/* Link previous to this object */
+
+		if (!ret) 
+			ret = link;		/* Save starting point */
+
+		start = start->next;		/* Go to next object */
+		prev = link;			/* Save pointer to this object */
+	}
+	return (ret);    			/* Return start of list */
+}
+
+struct ast_ha *ast_append_ha(char *sense, char *stuff, struct ast_ha *path)
+{
+	struct ast_ha *ha = malloc(sizeof(struct ast_ha));
+	char *nm="255.255.255.255";
+	char tmp[256] = "";
+	struct ast_ha *prev = NULL;
+	struct ast_ha *ret;
+	int x,z;
+	unsigned int y;
+	ret = path;
+	while(path) {
+		prev = path;
+		path = path->next;
+	}
+	if (ha) {
+		strncpy(tmp, stuff, sizeof(tmp) - 1);
+		nm = strchr(tmp, '/');
+		if (!nm)
+			nm = "255.255.255.255";
+		else {
+			*nm = '\0';
+			nm++;
+		}
+		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 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)) {
+			ha->sense = AST_SENSE_ALLOW;
+		} else {
+			ha->sense = AST_SENSE_DENY;
+		}
+		ha->next = NULL;
+		if (prev)
+			prev->next = ha;
+		else
+			ret = ha;
+	}
+	ast_log(LOG_DEBUG, "%s/%s appended to acl for peer\n",stuff, nm);
+	return ret;
+}
+
+int ast_apply_ha(struct ast_ha *ha, struct sockaddr_in *sin)
+{
+	/* Start optimistic */
+	int res = AST_SENSE_ALLOW;
+	while(ha) {
+		char iabuf[INET_ADDRSTRLEN];
+		char iabuf2[INET_ADDRSTRLEN];
+		/* DEBUG */
+		ast_log(LOG_DEBUG,
+			"##### Testing %s with %s\n",
+			ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr),
+			ast_inet_ntoa(iabuf2, sizeof(iabuf2), ha->netaddr));
+		/* For each rule, if this address and the netmask = the net address
+		   apply the current rule */
+		if ((sin->sin_addr.s_addr & ha->netmask.s_addr) == (ha->netaddr.s_addr))
+			res = ha->sense;
+		ha = ha->next;
+	}
+	return res;
+}
+
+int ast_get_ip(struct sockaddr_in *sin, char *value)
+{
+	struct hostent *hp;
+	struct ast_hostent ahp;
+	hp = ast_gethostbyname(value, &ahp);
+	if (hp) {
+		memcpy(&sin->sin_addr, hp->h_addr, sizeof(sin->sin_addr));
+	} else {
+		ast_log(LOG_WARNING, "Unable to lookup '%s'\n", value);
+		return -1;
+	}
+	return 0;
+}
+
+/* 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[256];
+	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))) {
+		ast_log(LOG_WARNING, "Unable to get hostname\n");
+	} else {
+		hp = ast_gethostbyname(ourhost, &ahp);
+		if (hp) {
+			memcpy(ourip, hp->h_addr, sizeof(*ourip));
+			return 0;
+		}
+	}
+	/* A.ROOT-SERVERS.NET. */
+	if (inet_aton("198.41.0.4", &saddr) && !ast_ouraddrfor(&saddr, ourip))
+		return 0;
+	return -1;
+}
+

aescrypt.c

@@ -30,27 +30,19 @@
  ---------------------------------------------------------------------------
  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
 */
 
-/*! \file
- *
- * \brief  This file contains the code for implementing encryption and decryption
- * for AES (Rijndael) for block and key sizes of 16, 24 and 32 bytes. It
- * can optionally be replaced by code written in assembler using NASM. For
- * further details see the file aesopt.h
- *
- * \author Dr Brian Gladman <brg@gladman.me.uk>
- */
+#include "aesopt.h"
 
 #if defined(__cplusplus)
 extern "C"
 {
 #endif
 
-#ifndef HAVE_CRYPTO
-
-#include "aesopt.h"
-
 #define si(y,x,k,c) (s(y,c) = word_in(x, c) ^ (k)[c])
 #define so(y,x,c)   word_out(y, c, s(x,c))
 
@@ -314,8 +306,6 @@ aes_rval aes_decrypt(const void *in_blk, void *out_blk, const aes_decrypt_ctx cx
 
 #endif
 
-#endif /* !HAVE_CRYPTO */
-
 #if defined(__cplusplus)
 }
 #endif

aeskey.c

@@ -30,26 +30,18 @@
  ---------------------------------------------------------------------------
  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.
 */
 
-/*! \file
- *
- *  \brief  This file contains the code for implementing the key schedule for AES
- *  (Rijndael) for block and key sizes of 16, 24, and 32 bytes. See aesopt.h
- *  for further details including optimisation.
- *
- * \author Dr Brian Gladman <brg@gladman.me.uk>
- */
+#include "aesopt.h"
 
 #if defined(__cplusplus)
 extern "C"
 {
 #endif
 
-#ifndef HAVE_CRYPTO
-
-#include "aesopt.h"
-
 /* Initialise the key schedule from the user supplied key. The key
    length can be specified in bytes, with legal values of 16, 24
    and 32, or in bits, with legal values of 128, 192 and 256. These
@@ -466,8 +458,6 @@ aes_rval aes_decrypt_key(const void *in_key, int key_len, aes_decrypt_ctx cx[1])
 
 #endif
 
-#endif /* !HAVE_CRYPTO */
-
 #if defined(__cplusplus)
 }
 #endif

aesopt.h

@@ -135,8 +135,7 @@
 #ifndef _AESOPT_H
 #define _AESOPT_H
 
-#include "asterisk/aes.h"
-#include "asterisk/endian.h"
+#include <asterisk/aes.h>
 
 /*  CONFIGURATION - USE OF DEFINES
 
@@ -147,6 +146,25 @@
     #if clauses.
 */
 
+/*  PLATFORM SPECIFIC INCLUDES */
+
+#if defined( __OpenBSD__ )
+#  include <machine/types.h>
+#  include <sys/endian.h>
+#elif defined( __FreeBSD__ ) || defined( __NetBSD__ )
+#  include <sys/types.h>
+#  include <sys/endian.h>
+#elif defined( BSD ) && ( BSD >= 199103 ) || defined(__APPLE__)
+#  include <machine/endian.h>
+#elif defined( __GNUC__ ) || defined( __GNU_LIBRARY__ )
+#  include <endian.h>
+#if !defined(__APPLE__)
+#  include <byteswap.h>
+#endif
+#elif defined( linux )
+#  include <endian.h>
+#endif
+
 /*  BYTE ORDER IN 32-BIT WORDS
 
     To obtain the highest speed on processors with 32-bit words, this code

aestab.c

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

agi/Makefile

@@ -1,53 +1,43 @@
 #
 # Asterisk -- A telephony toolkit for Linux.
 # 
-# Makefile for AGI-related stuff
+# Makefile for PBX frontends (dynamically loaded)
 #
-# Copyright (C) 1999-2006, Digium
+# Copyright (C) 1999, Mark Spencer
 #
-# Mark Spencer <markster@digium.com>
+# Mark Spencer <markster@linux-support.net>
 #
 # This program is free software, distributed under the terms of
 # the GNU General Public License
 #
 
-.PHONY: clean all uninstall
+AGIS=agi-test.agi eagi-test eagi-sphinx-test
 
-AGIS=agi-test.agi eagi-test eagi-sphinx-test jukebox.agi
+CFLAGS+=
 
-ifeq ($(OSARCH),SunOS)
-  LIBS+=-lsocket -lnsl
-endif
-
-ifeq ($(OSARCH),mingw32)
-    AGIS:=
-endif
-
-include $(ASTTOPDIR)/Makefile.rules
-
-_ASTCFLAGS+=-DSTANDALONE
-
-all: $(AGIS)
-
-strcompat.c: ../main/strcompat.c
-	@cp $< $@
-
-eagi-test: eagi-test.o strcompat.o
-
-eagi-sphinx-test: eagi-sphinx-test.o
+all: depend $(AGIS)
 
 install: all
 	mkdir -p $(DESTDIR)$(AGI_DIR)
 	for x in $(AGIS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(AGI_DIR) ; done
 
-uninstall:
-	for x in $(AGIS); do rm -f $(DESTDIR)$(AGI_DIR)/$$x ; done
+eagi-test: eagi-test.o
+	$(CC) $(CFLAGS) -o eagi-test eagi-test.o
+
+eagi-sphinx-test: eagi-sphinx-test.o
+	$(CC) $(CFLAGS) -o eagi-sphinx-test eagi-sphinx-test.o
 
 clean:
-	rm -f *.so *.o look eagi-test eagi-sphinx-test
-	rm -f .*.d *.s *.i
-	rm -f strcompat.c
+	rm -f *.so *.o look .depend eagi-test eagi-sphinx-test
 
-ifneq ($(wildcard .*.d),)
-   include .*.d
+%.so : %.o
+	$(CC) -shared -Xlinker -x -o $@ $<
+
+ifneq ($(wildcard .depend),)
+include .depend
 endif
+
+depend: .depend
+
+.depend:
+	../mkdep $(CFLAGS) `ls *.c`

agi/eagi-sphinx-test.c

@@ -6,15 +6,6 @@
  *
  */
 
-/*! \file
- * Extended AGI test application
- *
- * This code is released into public domain
- * without any warranty of any kind.
- *
- *	\ingroup agi
- */
-
 #include <stdio.h>
 #include <unistd.h>
 #include <stdlib.h>
@@ -27,10 +18,6 @@
 #include <arpa/inet.h>
 #include <netdb.h>
 
-#include "asterisk.h"
-
-#include "asterisk/compat.h"
-
 #define AUDIO_FILENO (STDERR_FILENO + 1)
 
 #define SPHINX_HOST "192.168.1.108"
@@ -79,9 +66,7 @@ static int read_environment(void)
 	char *val;
 	/* Read environment */
 	for(;;) {
-		if (!fgets(buf, sizeof(buf), stdin)) {
-			return -1;
-		}
+		fgets(buf, sizeof(buf), stdin);
 		if (feof(stdin))
 			return -1;
 		buf[strlen(buf) - 1] = '\0';
@@ -132,9 +117,7 @@ static char *wait_result(void)
 			return NULL;
 		}
 		if (FD_ISSET(STDIN_FILENO, &fds)) {
-			if (!fgets(astresp, sizeof(astresp), stdin)) {
-				return NULL;
-			}
+			fgets(astresp, sizeof(astresp), stdin);
 			if (feof(stdin)) {
 				fprintf(stderr, "Got hungup on apparently\n");
 				return NULL;
@@ -145,10 +128,9 @@ static char *wait_result(void)
 		}
 		if (FD_ISSET(AUDIO_FILENO, &fds)) {
 			res = read(AUDIO_FILENO, audiobuf, sizeof(audiobuf));
-			if ((res > 0) && (sphinx_sock > -1)) {
-				if (write(sphinx_sock, audiobuf, res) < 0) {
-					fprintf(stderr, "write() failed: %s\n", strerror(errno));
-				}
+			if (res > 0) {
+				if (sphinx_sock > -1) 
+					write(sphinx_sock, audiobuf, res);
 			}
 		}
 		if ((sphinx_sock > -1) && FD_ISSET(sphinx_sock, &fds)) {
@@ -223,7 +205,7 @@ int main(int argc, char *argv[])
 	connect_sphinx();
 	tmp = getenv("agi_enhanced");
 	if (tmp) {
-		if (sscanf(tmp, "%30d.%30d", &ver, &subver) != 2)
+		if (sscanf(tmp, "%d.%d", &ver, &subver) != 2)
 			ver = 0;
 	}
 	if (ver < 1) {

agi/eagi-test.c

@@ -5,28 +5,22 @@
  * with no warranty of any kind
  */
 
-#include "asterisk.h"
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <errno.h>
+#include <string.h>
+#include <sys/select.h>
 
 #define AUDIO_FILENO (STDERR_FILENO + 1)
 
-/*! \file
- * Extended AGI test application
- *
- * This code is released into the public domain
- * with no warranty of any kind
- *
- * \ingroup agi
- */
-
 static int read_environment(void)
 {
 	char buf[256];
 	char *val;
 	/* Read environment */
 	for(;;) {
-		if (!fgets(buf, sizeof(buf), stdin)) {
-			return -1;
-		}
+		fgets(buf, sizeof(buf), stdin);
 		if (feof(stdin))
 			return -1;
 		buf[strlen(buf) - 1] = '\0';
@@ -70,9 +64,7 @@ static char *wait_result(void)
 			return NULL;
 		}
 		if (FD_ISSET(STDIN_FILENO, &fds)) {
-			if (!fgets(astresp, sizeof(astresp), stdin)) {
-				return NULL;
-			}
+			fgets(astresp, sizeof(astresp), stdin);
 			if (feof(stdin)) {
 				fprintf(stderr, "Got hungup on apparently\n");
 				return NULL;
@@ -156,7 +148,7 @@ int main(int argc, char *argv[])
 	}
 	tmp = getenv("agi_enhanced");
 	if (tmp) {
-		if (sscanf(tmp, "%30d.%30d", &ver, &subver) != 2)
+		if (sscanf(tmp, "%d.%d", &ver, &subver) != 2)
 			ver = 0;
 	}
 	if (ver < 1) {

agi/jukebox.agi

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

alaw.c

@@ -0,0 +1,85 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * u-Law to Signed linear conversion
+ * 
+ * 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/alaw.h>
+
+#define AMI_MASK 0x55
+
+static inline unsigned char linear2alaw (short int linear)
+{
+    int mask;
+    int seg;
+    int pcm_val;
+    static int seg_end[8] =
+    {
+         0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF, 0x1FFF, 0x3FFF, 0x7FFF
+    };
+    
+    pcm_val = linear;
+    if (pcm_val >= 0)
+    {
+        /* Sign (7th) bit = 1 */
+        mask = AMI_MASK | 0x80;
+    }
+    else
+    {
+        /* Sign bit = 0 */
+        mask = AMI_MASK;
+        pcm_val = -pcm_val;
+    }
+
+    /* Convert the scaled magnitude to segment number. */
+    for (seg = 0;  seg < 8;  seg++)
+    {
+        if (pcm_val <= seg_end[seg])
+	    break;
+    }
+    /* Combine the sign, segment, and quantization bits. */
+    return  ((seg << 4) | ((pcm_val >> ((seg)  ?  (seg + 3)  :  4)) & 0x0F)) ^ mask;
+}
+/*- End of function --------------------------------------------------------*/
+
+static inline short int alaw2linear (unsigned char alaw)
+{
+    int i;
+    int seg;
+
+    alaw ^= AMI_MASK;
+    i = ((alaw & 0x0F) << 4);
+    seg = (((int) alaw & 0x70) >> 4);
+    if (seg)
+        i = (i + 0x100) << (seg - 1);
+    return (short int) ((alaw & 0x80)  ?  i  :  -i);
+}
+
+unsigned char __ast_lin2a[8192];
+short __ast_alaw[256];
+
+void ast_alaw_init(void)
+{
+	int i;
+	/* 
+	 *  Set up mu-law conversion table
+	 */
+	for(i = 0;i < 256;i++)
+	   {
+	        __ast_alaw[i] = alaw2linear(i);
+	   }
+	  /* set up the reverse (mu-law) conversion table */
+	for(i = -32768; i < 32768; i++)
+	   {
+		__ast_lin2a[((unsigned short)i) >> 3] = linear2alaw(i);
+	   }
+
+}
+

apps/Makefile

@@ -1,35 +1,104 @@
 #
 # Asterisk -- A telephony toolkit for Linux.
 # 
-# Makefile for PBX applications
+# Makefile for PBX frontends (dynamically loaded)
 #
-# Copyright (C) 1999-2006, Digium, Inc.
+# 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 $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
+USE_MYSQL_VM_INTERFACE=0
+USE_POSTGRES_VM_INTERFACE=0
 
-MODULE_PREFIX=app
-MENUSELECT_CATEGORY=APPS
-MENUSELECT_DESCRIPTION=Applications
+#APPS=app_dial.so app_playback.so app_directory.so app_intercom.so app_mp3.so 
+APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
+     app_system.so app_echo.so app_record.so app_image.so app_url.so app_disa.so \
+     app_qcall.so app_adsiprog.so app_getcpeid.so app_milliwatt.so \
+     app_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_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
 
-MENUSELECT_OPTS_app_directory:=$(MENUSELECT_OPTS_app_voicemail)
-ifneq ($(findstring ODBC_STORAGE,$(MENUSELECT_OPTS_app_voicemail)),)
-  MENUSELECT_DEPENDS_app_voicemail+=$(MENUSELECT_DEPENDS_ODBC_STORAGE)
-  MENUSELECT_DEPENDS_app_directory+=$(MENUSELECT_DEPENDS_ODBC_STORAGE)
-endif
-ifneq ($(findstring IMAP_STORAGE,$(MENUSELECT_OPTS_app_voicemail)),)
-  MENUSELECT_DEPENDS_app_voicemail+=$(MENUSELECT_DEPENDS_IMAP_STORAGE)
+ifneq (${OSARCH},Darwin)
+APPS+=app_intercom.so
 endif
 
-all: _all
+#APPS+=app_sql_postgres.so
+#APPS+=app_sql_odbc.so
+#APPS+=app_rpt.so
 
-include $(ASTTOPDIR)/Makefile.moddir_rules
+APPS+=$(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)
 
-ifneq ($(findstring $(OSARCH), mingw32 cygwin ),)
-  LIBS+= -lres_features.so -lres_ael_share.so -lres_monitor.so -lres_speech.so
-  LIBS+= -lres_smdi.so
+CFLAGS+=-fPIC
+
+ifeq ($(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 .depend
+
+%.so : %.o
+	$(CC) $(SOLINK) -o $@ $<
+
+app_rpt.so : app_rpt.o
+	$(CC) $(SOLINK) -o $@ $< -ltonezone
+
+install: all
+	for x in $(APPS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done
+	rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
+
+app_voicemail.so : app_voicemail.o
+ifeq ($(USE_MYSQL_VM_INTERFACE),1)
+	$(CC) $(SOLINK) -o $@ $(MLFLAGS) $< -L/usr/lib/mysql -lmysqlclient -lz
+else
+ifeq ($(USE_POSTGRES_VM_INTERFACE),1)
+	$(CC) $(SOLINK) -o $@ $(MLFLAGS) $< -lpq
+else
+	$(CC) $(SOLINK) -o $@ $(MLFLAGS) $< 
+endif
+endif
+
+app_sql_postgres.o: app_sql_postgres.c
+	$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres.o app_sql_postgres.c
+
+app_sql_postgres.so: app_sql_postgres.o
+	$(CC) $(SOLINK) -o $@ $< -L/usr/local/pgsql/lib -lpq
+
+app_sql_odbc.so: app_sql_odbc.o
+	$(CC) $(SOLINK) -o $@ $< -lodbc
+
+look:	look.c
+	$(CC) -pipe -O6 -g look.c -o look -lncurses
+
+ifneq ($(wildcard .depend),)
+include .depend
+endif
+
+depend: .depend
+
+.depend:
+	../mkdep $(CFLAGS) `ls *.c`
+
+env:
+	env

apps/app_amd.c

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

apps/app_authenticate.c

@@ -1,226 +1,172 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Execute arbitrary authenticate commands
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
 
-/*! \file
- *
- * \brief Execute arbitrary authenticate commands
- *
- * \author Mark Spencer <markster@digium.com>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/app.h"
-#include "asterisk/astdb.h"
-#include "asterisk/utils.h"
-
-enum {
-	OPT_ACCOUNT = (1 << 0),
-	OPT_DATABASE = (1 << 1),
-	OPT_MULTIPLE = (1 << 3),
-	OPT_REMOVE = (1 << 4),
-} auth_option_flags;
-
-AST_APP_OPTIONS(auth_app_options, {
-	AST_APP_OPTION('a', OPT_ACCOUNT),
-	AST_APP_OPTION('d', OPT_DATABASE),
-	AST_APP_OPTION('m', OPT_MULTIPLE),
-	AST_APP_OPTION('r', OPT_REMOVE),
-});
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/app.h>
+#include <asterisk/astdb.h>
+#include <asterisk/utils.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+#include <stdlib.h>
+#include <stdio.h>
 
+static char *tdesc = "Authentication Application";
 
 static char *app = "Authenticate";
 
 static char *synopsis = "Authenticate a user";
 
 static char *descrip =
-"  Authenticate(password[,options[,maxdigits]]): This application asks the caller\n"
-"to enter a given password in order to continue dialplan execution. If the password\n"
-"begins with the '/' character, it is interpreted as a file which contains a list of\n"
-"valid passwords, listed 1 password per line in the file.\n"
-"  When using a database key, the value associated with the key can be anything.\n"
-"Users have three attempts to authenticate before the channel is hung up.\n"
-"  Options:\n"
-"     a - Set the channels' account code to the password that is entered\n"
-"     d - Interpret the given path as database key, not a literal file\n"
-"     m - Interpret the given path as a file which contains a list of account\n"
-"         codes and password hashes delimited with ':', listed one per line in\n"
-"         the file. When one of the passwords is matched, the channel will have\n"
-"         its account code set to the corresponding account code in the file.\n"
-"     r - Remove the database key upon successful entry (valid with 'd' only)\n"
-"     maxdigits  - maximum acceptable number of digits. Stops reading after\n"
-"         maxdigits have been entered (without requiring the user to\n"
-"         press the '#' key).\n"
-"         Defaults to 0 - no limit - wait for the user press the '#' key.\n"
-;
+"  Authenticate(password[|options]): Requires a user to enter a"
+"given password in order to continue execution.  If the\n"
+"password begins with the '/' character, it is interpreted as\n"
+"a file which contains a list of valid passwords (1 per line).\n"
+"an optional set of opions may be provided by concatenating any\n"
+"of the following letters:\n"
+"     a - Set account code to the password that is entered\n"
+"     d - Interpret path as database key, not literal file\n"
+"     r - Remove database key upon successful entry (valid with 'd' only)\n"
+"\n"
+"When using a database key, the value associated with the key can be\n"
+"anything.\n"
+"Returns 0 if the user enters a valid password within three\n"
+"tries, or -1 otherwise (or on hangup).\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int auth_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0, retries, maxdigits;
-	char passwd[256], *prompt = "agent-pass", *argcopy = NULL;
-	struct ast_flags flags = {0};
-
-	AST_DECLARE_APP_ARGS(arglist,
-		AST_APP_ARG(password);
-		AST_APP_ARG(options);
-		AST_APP_ARG(maxdigits);
-	);
-
-	if (ast_strlen_zero(data)) {
+	int res=0;
+	int retries;
+	struct localuser *u;
+	char password[256]="";
+	char passwd[256];
+	char *opts;
+	char *prompt;
+	if (!data || ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Authenticate requires an argument(password)\n");
 		return -1;
 	}
-
+	LOCAL_USER_ADD(u);
 	if (chan->_state != AST_STATE_UP) {
-		if ((res = ast_answer(chan)))
+		res = ast_answer(chan);
+		if (res) {
+			LOCAL_USER_REMOVE(u);
 			return -1;
+		}
 	}
-
-	argcopy = ast_strdupa(data);
-
-	AST_STANDARD_APP_ARGS(arglist, argcopy);
-
-	if (!ast_strlen_zero(arglist.options))
-		ast_app_parse_options(auth_app_options, &flags, NULL, arglist.options);
-
-	if (!ast_strlen_zero(arglist.maxdigits)) {
-		maxdigits = atoi(arglist.maxdigits);
-		if ((maxdigits<1) || (maxdigits>sizeof(passwd)-2))
-			maxdigits = sizeof(passwd) - 2;
-	} else {
-		maxdigits = sizeof(passwd) - 2;
-	}
-
+	strncpy(password, data, sizeof(password) - 1);
+	opts=strchr(password, '|');
+	if (opts) {
+		*opts = 0;
+		opts++;
+	} else
+		opts = "";
 	/* Start asking for password */
+	prompt = "agent-pass";
 	for (retries = 0; retries < 3; retries++) {
-		if ((res = ast_app_getdata(chan, prompt, passwd, maxdigits, 0)) < 0)
+		res = ast_app_getdata(chan, prompt, passwd, sizeof(passwd) - 2, 0);
+		if (res < 0)
 			break;
-
 		res = 0;
-
-		if (arglist.password[0] != '/') {
-			/* Compare against a fixed password */
-			if (!strcmp(passwd, arglist.password))
-				break;
-		} else if (ast_test_flag(&flags,OPT_DATABASE)) {
-			char tmp[256];
-			/* Compare against a database key */
-			if (!ast_db_get(arglist.password + 1, passwd, tmp, sizeof(tmp))) {
-				/* It's a good password */
-				if (ast_test_flag(&flags,OPT_REMOVE))
-					ast_db_del(arglist.password + 1, passwd);
-				break;
+		if (password[0] == '/') {
+			if (strchr(opts, 'd')) {
+				char tmp[256];
+				/* Compare against a database key */
+				if (!ast_db_get(password + 1, passwd, tmp, sizeof(tmp))) {
+					/* It's a good password */
+					if (strchr(opts, 'r')) {
+						ast_db_del(password + 1, passwd);
+					}
+					break;
+				}
+			} else {
+				/* Compare against a file */
+				FILE *f;
+				f = fopen(password, "r");
+				if (f) {
+					char buf[256] = "";
+					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 file */
-			FILE *f;
-			char buf[256] = "", md5passwd[33] = "", *md5secret = NULL;
-
-			if (!(f = fopen(arglist.password, "r"))) {
-				ast_log(LOG_WARNING, "Unable to open file '%s' for authentication: %s\n", arglist.password, strerror(errno));
-				continue;
-			}
-
-			for (;;) {
-				size_t len;
-
-				if (feof(f))
-					break;
-
-				if (!fgets(buf, sizeof(buf), f)) {
-					continue;
-				}
-
-				if (ast_strlen_zero(buf))
-					continue;
-
-				len = strlen(buf) - 1;
-				if (buf[len] == '\n')
-					buf[len] = '\0';
-
-				if (ast_test_flag(&flags, OPT_MULTIPLE)) {
-					md5secret = buf;
-					strsep(&md5secret, ":");
-					if (!md5secret)
-						continue;
-					ast_md5_hash(md5passwd, passwd);
-					if (!strcmp(md5passwd, md5secret)) {
-						if (ast_test_flag(&flags,OPT_ACCOUNT))
-							ast_cdr_setaccount(chan, buf);
-						break;
-					}
-				} else {
-					if (!strcmp(passwd, buf)) {
-						if (ast_test_flag(&flags, OPT_ACCOUNT))
-							ast_cdr_setaccount(chan, buf);
-						break;
-					}
-				}
-			}
-
-			fclose(f);
-
-			if (!ast_strlen_zero(buf)) {
-				if (ast_test_flag(&flags, OPT_MULTIPLE)) {
-					if (md5secret && !strcmp(md5passwd, md5secret))
-						break;
-				} else {
-					if (!strcmp(passwd, buf))
-						break;
-				}
-			}
+			/* Compare against a fixed password */
+			if (!strcmp(passwd, password)) 
+				break;
 		}
-		prompt = "auth-incorrect";
+		prompt="auth-incorrect";
 	}
-
 	if ((retries < 3) && !res) {
-		if (ast_test_flag(&flags,OPT_ACCOUNT) && !ast_test_flag(&flags,OPT_MULTIPLE))
+		if (strchr(opts, 'a')) 
 			ast_cdr_setaccount(chan, passwd);
-		if (!(res = ast_streamfile(chan, "auth-thankyou", chan->language)))
+		res = ast_streamfile(chan, "auth-thankyou", chan->language);
+		if (!res)
 			res = ast_waitstream(chan, "");
 	} else {
-		if (!ast_streamfile(chan, "vm-goodbye", chan->language))
+		if (!res)
+			res = ast_streamfile(chan, "vm-goodbye", chan->language);
+		if (!res)
 			res = ast_waitstream(chan, "");
 		res = -1;
 	}
-
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
-	if (ast_register_application(app, auth_exec, synopsis, descrip))
-		return AST_MODULE_LOAD_FAILURE;
-	return AST_MODULE_LOAD_SUCCESS;
+	return ast_register_application(app, auth_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Authentication Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_cdr.c

@@ -1,63 +1,66 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Applictions connected with CDR engine
+ * 
+ * Copyright (C) 2003, Digium
  *
  * Martin Pycko <martinp@digium.com>
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
 
-/*! \file
- *
- * \brief Applications connected with CDR engine
- *
- * \author Martin Pycko <martinp@digium.com>
- *
- * \ingroup applications
- */
+#include <sys/types.h>
+#include <asterisk/channel.h>
+#include <asterisk/module.h>
+#include <asterisk/pbx.h>
+#include <stdlib.h>
 
-#include "asterisk.h"
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/channel.h"
-#include "asterisk/module.h"
-
-static char *nocdr_descrip =
-"  NoCDR(): This application will tell Asterisk not to maintain a CDR for the\n"
-"current call.\n";
+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 = "Tell Asterisk to not maintain a CDR for the current call";
+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_set_flag(chan->cdr, AST_CDR_FLAG_POST_DISABLED);
-
+	if (chan->cdr) {
+		ast_cdr_free(chan->cdr);
+		chan->cdr = NULL;
+	}
 	return 0;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(nocdr_app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
-	if (ast_register_application(nocdr_app, nocdr_exec, nocdr_synopsis, nocdr_descrip))
-		return AST_MODULE_LOAD_FAILURE;
-	return AST_MODULE_LOAD_SUCCESS;
+	return ast_register_application(nocdr_app, nocdr_exec, nocdr_synopsis, nocdr_descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Tell Asterisk to not maintain a CDR for the current call");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_chanisavail.c

@@ -1,96 +1,69 @@
 /*
-* Asterisk -- An open source telephony toolkit.
-*
-* Copyright (C) 1999 - 2005, Digium, Inc.
-*
-* Mark Spencer <markster@digium.com>
-* James Golovich <james@gnuinter.net>
-*
-* See http://www.asterisk.org for more information about
-* the Asterisk project. Please do not directly contact
-* any of the maintainers of this project for assistance;
-* the project provides a web site, mailing lists and IRC
-* channels for your use.
-*
-* This program is free software, distributed under the terms of
-* the GNU General Public License Version 2. See the LICENSE file
-* at the top of the source tree.
-*/
-
-/*! \file
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * \brief Check if Channel is Available
+ * Check if Channel is Available
+ * 
+ * Copyright (C) 2003, Digium
+ *
+ * Mark Spencer <markster@digium.com>
+ * James Golovich <james@gnuinter.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
  *
- * \author Mark Spencer <markster@digium.com>
- * \author James Golovich <james@gnuinter.net>
-
- * \ingroup applications
  */
 
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/app.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <errno.h>
+#include <string.h>
+#include <stdlib.h>
 #include <sys/ioctl.h>
 
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/app.h"
-#include "asterisk/devicestate.h"
+static char *tdesc = "Check if channel is available";
 
 static char *app = "ChanIsAvail";
 
-static char *synopsis = "Check channel availability";
+static char *synopsis = "Check if channel is available";
 
-static char *descrip =
-"  ChanIsAvail(Technology/resource[&Technology2/resource2...][,options]): \n"
-"This application will check to see if any of the specified channels are\n"
-"available.\n"
-"  Options:\n"
-"    s - Consider the channel unavailable if the channel is in use at all.\n"
-"    t - Simply checks if specified channels exist in the channel list\n"
-"        (implies option s).\n"
-"Note that the AVAILSTATUS variable is used for both device state\n"
-"and cause code. It is therefore possible for it to give a value that may\n"
-"indicate a device is available when it is not. It is suggested that the\n"
-"AVAILORIGCHAN variable is used instead to see whether a device is available\n"
-"or not.\n"
-"This application sets the following channel variable upon completion:\n"
-"  AVAILCHAN     - the name of the available channel, if one exists\n"
-"  AVAILORIGCHAN - the canonical channel name that was used to create the channel\n"
-"  AVAILSTATUS   - the status code for the available channel\n";
+static char *descrip = 
+"  ChanIsAvail(Technology/resource[&Technology2/resource2...]): \n"
+"Checks is any of the requested channels are available.  If none\n"
+"of the requested channels are available the new priority will be\n"
+"n+101 (unless such a priority does not exist or on error, in which\n"
+"case ChanIsAvail will return -1).\n"
+"If any of the requested channels are available, the next priority will be n+1,\n"
+"the channel variable ${AVAILCHAN} will be set to the name of the available channel\n"
+"and the ChanIsAvail app will return 0.\n"
+"${AVAILORIGCHAN} is the canonical channel name that was used to create the channel.\n"
+"${AVAILSTATUS} is the status code for the channel.\n";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int chanavail_exec(struct ast_channel *chan, void *data)
 {
-	int res=-1, inuse=-1, option_state=0, string_compare=0;
-	int status;
-	char *info, tmp[512], trychan[512], *peers, *tech, *number, *rest, *cur;
+	int res=-1;
+	struct localuser *u;
+	char info[512], tmp[512], *peers, *tech, *number, *rest, *cur;
 	struct ast_channel *tempchan;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(reqchans);
-		AST_APP_ARG(options);
-	);
 
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "ChanIsAvail requires an argument (DAHDI/1&DAHDI/2)\n");
+	if (!data) {
+		ast_log(LOG_WARNING, "ChanIsAvail requires an argument (Zap/1&Zap/2)\n");
 		return -1;
 	}
+	LOCAL_USER_ADD(u);
 
-	info = ast_strdupa(data);
-
-	AST_STANDARD_APP_ARGS(args, info);
-
-	if (args.options) {
-		if (strchr(args.options, 's'))
-			option_state = 1;
-		if (strchr(args.options, 't'))
-			string_compare = 1;
-	}
-	peers = args.reqchans;
+	strncpy(info, (char *)data, sizeof(info)-1);
+	peers = info;
 	if (peers) {
 		cur = peers;
 		do {
@@ -108,55 +81,57 @@ static int chanavail_exec(struct ast_channel *chan, void *data)
 			}
 			*number = '\0';
 			number++;
-			
-			if (string_compare) {
-				/* ast_parse_device_state checks for "SIP/1234" as a channel name.
-				   ast_device_state will ask the SIP driver for the channel state. */
-
-				snprintf(trychan, sizeof(trychan), "%s/%s",cur,number);
-				status = inuse = ast_parse_device_state(trychan);
-			} else if (option_state) {
-				/* If the pbx says in use then don't bother trying further.
-				   This is to permit testing if someone's on a call, even if the
-				   channel can permit more calls (ie callwaiting, sip calls, etc).  */
-
-				snprintf(trychan, sizeof(trychan), "%s/%s",cur,number);
-				status = inuse = ast_device_state(trychan);
-			}
-			if ((inuse <= 1) && (tempchan = ast_request(tech, chan->nativeformats, number, &status))) {
+			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);
-					snprintf(tmp, sizeof(tmp), "%d", status);
-					pbx_builtin_setvar_helper(chan, "AVAILSTATUS", tmp);
 					ast_hangup(tempchan);
 					tempchan = NULL;
 					res = 1;
 					break;
-			} else {
-				snprintf(tmp, sizeof(tmp), "%d", status);
-				pbx_builtin_setvar_helper(chan, "AVAILSTATUS", tmp);
 			}
 			cur = rest;
 		} while (cur);
 	}
+
 	if (res < 1) {
 		pbx_builtin_setvar_helper(chan, "AVAILCHAN", "");
 		pbx_builtin_setvar_helper(chan, "AVAILORIGCHAN", "");
+		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+			chan->priority+=100;
+		else
+			return -1;
 	}
 
+	LOCAL_USER_REMOVE(u);
 	return 0;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, chanavail_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Check channel availability");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_channelredirect.c

@@ -1,98 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2006, Sergey Basmanov
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief ChannelRedirect application
- *
- * \author Sergey Basmanov <sergey_basmanov@mail.ru>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/lock.h"
-#include "asterisk/app.h"
-#include "asterisk/features.h"
-
-static char *app = "ChannelRedirect";
-static char *synopsis = "Redirects given channel to a dialplan target.";
-static char *descrip =
-"ChannelRedirect(channel,[[context,]extension,]priority)\n"
-"  Sends the specified channel to the specified extension priority\n"
-"This application sets the following channel variables upon completion:\n"
-"  CHANNELREDIRECT_STATUS - Are set to the result of the redirection\n"
-"                           either NOCHANNEL or SUCCESS\n";
-
-static int asyncgoto_exec(struct ast_channel *chan, void *data)
-{
-	int res = -1;
-	char *info;
-	struct ast_channel *chan2 = NULL;
-
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(channel);
-		AST_APP_ARG(label);
-	);
-
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "%s requires an argument (channel,[[context,]exten,]priority)\n", app);
-		return -1;
-	}
-
-	info = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, info);
-
-	if (ast_strlen_zero(args.channel) || ast_strlen_zero(args.label)) {
-		ast_log(LOG_WARNING, "%s requires an argument (channel,[[context,]exten,]priority)\n", app);
-		return -1;
-	}
-
-	chan2 = ast_get_channel_by_name_locked(args.channel);
-	if (!chan2) {
-		ast_log(LOG_WARNING, "No such channel: %s\n", args.channel);
-		pbx_builtin_setvar_helper(chan, "CHANNELREDIRECT_STATUS", "NOCHANNEL");
-		return 0;
-	}
-
-	if (chan2->pbx) {
-		ast_set_flag(chan2, AST_FLAG_BRIDGE_HANGUP_DONT); /* don't let the after-bridge code run the h-exten */
-	}
-	res = ast_async_parseable_goto(chan2, args.label);
-	pbx_builtin_setvar_helper(chan, "CHANNELREDIRECT_STATUS", "SUCCESS");
-	ast_channel_unlock(chan2);
-
-	return res;
-}
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app, asyncgoto_exec, synopsis, descrip);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Redirects a given channel to a dialplan target");

apps/app_chanspy.c

@@ -1,965 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2005 Anthony Minessale II (anthmct@yahoo.com)
- * Copyright (C) 2005 - 2008, Digium, Inc.
- *
- * A license has been granted to Digium (via disclaimer) for the use of
- * this code.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief ChanSpy: Listen in on any channel.
- *
- * \author Anthony Minessale II <anthmct@yahoo.com>
- * \author Joshua Colp <jcolp@digium.com>
- * \author Russell Bryant <russell@digium.com>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <ctype.h>
-#include <errno.h>
-
-#include "asterisk/paths.h" /* use ast_config_AST_MONITOR_DIR */
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/audiohook.h"
-#include "asterisk/features.h"
-#include "asterisk/app.h"
-#include "asterisk/utils.h"
-#include "asterisk/say.h"
-#include "asterisk/pbx.h"
-#include "asterisk/translate.h"
-#include "asterisk/module.h"
-#include "asterisk/lock.h"
-#include "asterisk/options.h"
-
-#define AST_NAME_STRLEN 256
-
-static const char *tdesc = "Listen to a channel, and optionally whisper into it";
-static const char *app_chan = "ChanSpy";
-static const char *desc_chan =
-"  ChanSpy([chanprefix][,options]): This application is used to listen to the\n"
-"audio from an Asterisk channel. This includes the audio coming in and\n"
-"out of the channel being spied on. If the 'chanprefix' parameter is specified,\n"
-"only channels beginning with this string will be spied upon.\n"
-"  While spying, the following actions may be performed:\n"
-"    - Dialing # cycles the volume level.\n"
-"    - Dialing * will stop spying and look for another channel to spy on.\n"
-"    - Dialing a series of digits followed by # builds a channel name to append\n"
-"      to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing\n"
-"      the digits '1234#' while spying will begin spying on the channel\n"
-"      'Agent/1234'.\n"
-"  Note: The X option supersedes the three features above in that if a valid\n"
-"        single digit extension exists in the correct context ChanSpy will\n"
-"        exit to it. This also disables choosing a channel based on 'chanprefix'\n"
-"        and a digit sequence.\n"
-"  Options:\n"
-"    b             - Only spy on channels involved in a bridged call.\n"
-"    g(grp)        - Match only channels where their SPYGROUP variable is set to\n"
-"                    contain 'grp' in an optional : delimited list.\n"
-"    q             - Don't play a beep when beginning to spy on a channel, or speak the\n"
-"                    selected channel name.\n"
-"    r[(basename)] - Record the session to the monitor spool directory. An\n"
-"                    optional base for the filename may be specified. The\n"
-"                    default is 'chanspy'.\n"
-"    v([value])    - Adjust the initial volume in the range from -4 to 4. A\n"
-"                    negative value refers to a quieter setting.\n"
-"    w             - Enable 'whisper' mode, so the spying channel can talk to\n"
-"                    the spied-on channel.\n"
-"    W             - Enable 'private whisper' mode, so the spying channel can\n"
-"                    talk to the spied-on channel but cannot listen to that\n"
-"                    channel.\n"
-"    o             - Only listen to audio coming from this channel.\n"
-"    X             - Allow the user to exit ChanSpy to a valid single digit\n"
-"                    numeric extension in the current context or the context\n"
-"                    specified by the SPY_EXIT_CONTEXT channel variable. The\n"
-"                    name of the last channel that was spied on will be stored\n"
-"                    in the SPY_CHANNEL variable.\n"
-"    e(ext)        - Enable 'enforced' mode, so the spying channel can\n"
-"                    only monitor extensions whose name is in the 'ext' : \n"
-"                    delimited list.\n"
-;
-
-static const char *app_ext = "ExtenSpy";
-static const char *desc_ext =
-"  ExtenSpy(exten[@context][,options]): This application is used to listen to the\n"
-"audio from an Asterisk channel. This includes the audio coming in and\n"
-"out of the channel being spied on. Only channels created by outgoing calls for the\n"
-"specified extension will be selected for spying. If the optional context is not\n"
-"supplied, the current channel's context will be used.\n"
-"  While spying, the following actions may be performed:\n"
-"    - Dialing # cycles the volume level.\n"
-"    - Dialing * will stop spying and look for another channel to spy on.\n"
-"  Note: The X option superseeds the two features above in that if a valid\n"
-"        single digit extension exists in the correct context it ChanSpy will\n"
-"        exit to it.\n"
-"  Options:\n"
-"    b             - Only spy on channels involved in a bridged call.\n"
-"    g(grp)        - Match only channels where their ${SPYGROUP} variable is set to\n"
-"                    contain 'grp' in an optional : delimited list.\n"
-"    q             - Don't play a beep when beginning to spy on a channel, or speak the\n"
-"                    selected channel name.\n"
-"    r[(basename)] - Record the session to the monitor spool directory. An\n"
-"                    optional base for the filename may be specified. The\n"
-"                    default is 'chanspy'.\n"
-"    v([value])    - Adjust the initial volume in the range from -4 to 4. A\n"
-"                    negative value refers to a quieter setting.\n"
-"    w             - Enable 'whisper' mode, so the spying channel can talk to\n"
-"                    the spied-on channel.\n"
-"    W             - Enable 'private whisper' mode, so the spying channel can\n"
-"                    talk to the spied-on channel but cannot listen to that\n"
-"                    channel.\n"
-"    o             - Only listen to audio coming from this channel.\n"
-"    X             - Allow the user to exit ChanSpy to a valid single digit\n"
-"                    numeric extension in the current context or the context\n"
-"                    specified by the SPY_EXIT_CONTEXT channel variable. The\n"
-"                    name of the last channel that was spied on will be stored\n"
-"                    in the SPY_CHANNEL variable.\n"
-;
-
-enum {
-	OPTION_QUIET     = (1 << 0),    /* Quiet, no announcement */
-	OPTION_BRIDGED   = (1 << 1),    /* Only look at bridged calls */
-	OPTION_VOLUME    = (1 << 2),    /* Specify initial volume */
-	OPTION_GROUP     = (1 << 3),    /* Only look at channels in group */
-	OPTION_RECORD    = (1 << 4),
-	OPTION_WHISPER   = (1 << 5),
-	OPTION_PRIVATE   = (1 << 6),    /* Private Whisper mode */
-	OPTION_READONLY  = (1 << 7),    /* Don't mix the two channels */
-	OPTION_EXIT      = (1 << 8),    /* Exit to a valid single digit extension */
-	OPTION_ENFORCED  = (1 << 9),    /* Enforced mode */
-} chanspy_opt_flags;
-
-enum {
-	OPT_ARG_VOLUME = 0,
-	OPT_ARG_GROUP,
-	OPT_ARG_RECORD,
-	OPT_ARG_ENFORCED,
-	OPT_ARG_ARRAY_SIZE,
-} chanspy_opt_args;
-
-AST_APP_OPTIONS(spy_opts, {
-	AST_APP_OPTION('q', OPTION_QUIET),
-	AST_APP_OPTION('b', OPTION_BRIDGED),
-	AST_APP_OPTION('w', OPTION_WHISPER),
-	AST_APP_OPTION('W', OPTION_PRIVATE),
-	AST_APP_OPTION_ARG('v', OPTION_VOLUME, OPT_ARG_VOLUME),
-	AST_APP_OPTION_ARG('g', OPTION_GROUP, OPT_ARG_GROUP),
-	AST_APP_OPTION_ARG('r', OPTION_RECORD, OPT_ARG_RECORD),
-	AST_APP_OPTION_ARG('e', OPTION_ENFORCED, OPT_ARG_ENFORCED),
-	AST_APP_OPTION('o', OPTION_READONLY),
-	AST_APP_OPTION('X', OPTION_EXIT),
-});
-
-static int next_unique_id_to_use = 0;
-
-struct chanspy_translation_helper {
-	/* spy data */
-	struct ast_audiohook spy_audiohook;
-	struct ast_audiohook whisper_audiohook;
-	int fd;
-	int volfactor;
-};
-
-static void *spy_alloc(struct ast_channel *chan, void *data)
-{
-	/* just store the data pointer in the channel structure */
-	return data;
-}
-
-static void spy_release(struct ast_channel *chan, void *data)
-{
-	/* nothing to do */
-}
-
-static int spy_generate(struct ast_channel *chan, void *data, int len, int samples)
-{
-	struct chanspy_translation_helper *csth = data;
-	struct ast_frame *f, *cur;
-
-	ast_audiohook_lock(&csth->spy_audiohook);
-	if (csth->spy_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
-		/* Channel is already gone more than likely */
-		ast_audiohook_unlock(&csth->spy_audiohook);
-		return -1;
-	}
-
-	if (ast_test_flag(&csth->spy_audiohook, OPTION_READONLY)) {
-		/* Option 'o' was set, so don't mix channel audio */
-		f = ast_audiohook_read_frame(&csth->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_READ, AST_FORMAT_SLINEAR);
-	} else {
-		f = ast_audiohook_read_frame(&csth->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR);
-	}
-
-	ast_audiohook_unlock(&csth->spy_audiohook);
-
-	if (!f)
-		return 0;
-
-	for (cur = f; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
-		if (ast_write(chan, cur)) {
-			ast_frfree(f);
-			return -1;
-		}
-
-		if (csth->fd) {
-			if (write(csth->fd, cur->data, cur->datalen) < 0) {
-				ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
-			}
-		}
-	}
-
-	ast_frfree(f);
-
-	return 0;
-}
-
-static struct ast_generator spygen = {
-	.alloc = spy_alloc,
-	.release = spy_release,
-	.generate = spy_generate,
-};
-
-static int start_spying(struct ast_channel *chan, const char *spychan_name, struct ast_audiohook *audiohook) 
-{
-	int res = 0;
-	struct ast_channel *peer = NULL;
-
-	ast_log(LOG_NOTICE, "Attaching %s to %s\n", spychan_name, chan->name);
-
-	ast_set_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC | AST_AUDIOHOOK_SMALL_QUEUE);
-	res = ast_audiohook_attach(chan, audiohook);
-
-	if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan))) { 
-		ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);
-	}
-	return res;
-}
-
-struct chanspy_ds {
-	struct ast_channel *chan;
-	char unique_id[20];
-	ast_mutex_t lock;
-};
-
-static int channel_spy(struct ast_channel *chan, struct chanspy_ds *spyee_chanspy_ds, 
-	int *volfactor, int fd, const struct ast_flags *flags, char *exitcontext) 
-{
-	struct chanspy_translation_helper csth;
-	int running = 0, res, x = 0;
-	char inp[24] = {0};
-	char *name;
-	struct ast_frame *f;
-	struct ast_silence_generator *silgen = NULL;
-	struct ast_channel *spyee = NULL;
-	const char *spyer_name;
-
-	ast_channel_lock(chan);
-	spyer_name = ast_strdupa(chan->name);
-	ast_channel_unlock(chan);
-
-	ast_mutex_lock(&spyee_chanspy_ds->lock);
-	while ((spyee = spyee_chanspy_ds->chan) && ast_channel_trylock(spyee)) {
-		/* avoid a deadlock here, just in case spyee is masqueraded and
-		 * chanspy_ds_chan_fixup() is called with the channel locked */
-		DEADLOCK_AVOIDANCE(&spyee_chanspy_ds->lock);
-	}
-	ast_mutex_unlock(&spyee_chanspy_ds->lock);
-
-	if (!spyee)
-		return 0;
-
-	/* We now hold the channel lock on spyee */
-
-	if (ast_check_hangup(chan) || ast_check_hangup(spyee)) {
-		ast_channel_unlock(spyee);
-		return 0;
-	}
-
-	name = ast_strdupa(spyee->name);
-	ast_verb(2, "Spying on channel %s\n", name);
-
-	memset(&csth, 0, sizeof(csth));
-	ast_copy_flags(&csth.spy_audiohook, flags, AST_FLAGS_ALL);
-
-	ast_audiohook_init(&csth.spy_audiohook, AST_AUDIOHOOK_TYPE_SPY, "ChanSpy");
-
-	if (start_spying(spyee, spyer_name, &csth.spy_audiohook)) {
-		ast_audiohook_destroy(&csth.spy_audiohook);
-		ast_channel_unlock(spyee);
-		return 0;
-	}
-
-	if (ast_test_flag(flags, OPTION_WHISPER)) {
-		ast_audiohook_init(&csth.whisper_audiohook, AST_AUDIOHOOK_TYPE_WHISPER, "ChanSpy");
-		start_spying(spyee, spyer_name, &csth.whisper_audiohook);
-	}
-
-	ast_channel_unlock(spyee);
-	spyee = NULL;
-
-	csth.volfactor = *volfactor;
-
-	if (csth.volfactor) {
-		csth.spy_audiohook.options.read_volume = csth.volfactor;
-		csth.spy_audiohook.options.write_volume = csth.volfactor;
-	}
-
-	csth.fd = fd;
-
-	if (ast_test_flag(flags, OPTION_PRIVATE))
-		silgen = ast_channel_start_silence_generator(chan);
-	else
-		ast_activate_generator(chan, &spygen, &csth);
-
-	/* We can no longer rely on 'spyee' being an actual channel;
-	   it can be hung up and freed out from under us. However, the
-	   channel destructor will put NULL into our csth.spy.chan
-	   field when that happens, so that is our signal that the spyee
-	   channel has gone away.
-	*/
-
-	/* Note: it is very important that the ast_waitfor() be the first
-	   condition in this expression, so that if we wait for some period
-	   of time before receiving a frame from our spying channel, we check
-	   for hangup on the spied-on channel _after_ knowing that a frame
-	   has arrived, since the spied-on channel could have gone away while
-	   we were waiting
-	*/
-	while ((res = ast_waitfor(chan, -1) > -1) && csth.spy_audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING) {
-		if (!(f = ast_read(chan)) || ast_check_hangup(chan)) {
-			running = -1;
-			break;
-		}
-
-		if (ast_test_flag(flags, OPTION_WHISPER) && f->frametype == AST_FRAME_VOICE) {
-			ast_audiohook_lock(&csth.whisper_audiohook);
-			ast_audiohook_write_frame(&csth.whisper_audiohook, AST_AUDIOHOOK_DIRECTION_WRITE, f);
-			ast_audiohook_unlock(&csth.whisper_audiohook);
-			ast_frfree(f);
-			continue;
-		}
-		
-		res = (f->frametype == AST_FRAME_DTMF) ? f->subclass : 0;
-		ast_frfree(f);
-		if (!res)
-			continue;
-
-		if (x == sizeof(inp))
-			x = 0;
-
-		if (res < 0) {
-			running = -1;
-			break;
-		}
-
-		if (ast_test_flag(flags, OPTION_EXIT)) {
-			char tmp[2];
-			tmp[0] = res;
-			tmp[1] = '\0';
-			if (!ast_goto_if_exists(chan, exitcontext, tmp, 1)) {
-				ast_debug(1, "Got DTMF %c, goto context %s\n", tmp[0], exitcontext);
-				pbx_builtin_setvar_helper(chan, "SPY_CHANNEL", name);
-				running = -2;
-				break;
-			} else {
-				ast_debug(2, "Exit by single digit did not work in chanspy. Extension %s does not exist in context %s\n", tmp, exitcontext);
-			}
-		} else if (res >= '0' && res <= '9') {
-			inp[x++] = res;
-		}
-
-		if (res == '*') {
-			running = 0;
-			break;
-		} else if (res == '#') {
-			if (!ast_strlen_zero(inp)) {
-				running = atoi(inp);
-				break;
-			}
-
-			(*volfactor)++;
-			if (*volfactor > 4)
-				*volfactor = -4;
-			ast_verb(3, "Setting spy volume on %s to %d\n", chan->name, *volfactor);
-
-			csth.volfactor = *volfactor;
-			csth.spy_audiohook.options.read_volume = csth.volfactor;
-			csth.spy_audiohook.options.write_volume = csth.volfactor;
-		}
-	}
-
-	if (ast_test_flag(flags, OPTION_PRIVATE))
-		ast_channel_stop_silence_generator(chan, silgen);
-	else
-		ast_deactivate_generator(chan);
-
-	if (ast_test_flag(flags, OPTION_WHISPER)) {
-		ast_audiohook_lock(&csth.whisper_audiohook);
-		ast_audiohook_detach(&csth.whisper_audiohook);
-		ast_audiohook_unlock(&csth.whisper_audiohook);
-		ast_audiohook_destroy(&csth.whisper_audiohook);
-	}
-
-	ast_audiohook_lock(&csth.spy_audiohook);
-	ast_audiohook_detach(&csth.spy_audiohook);
-	ast_audiohook_unlock(&csth.spy_audiohook);
-	ast_audiohook_destroy(&csth.spy_audiohook);
-	
-	ast_verb(2, "Done Spying on channel %s\n", name);
-
-	return running;
-}
-
-/*!
- * \note This relies on the embedded lock to be recursive, as it may be called
- * due to a call to chanspy_ds_free with the lock held there.
- */
-static void chanspy_ds_destroy(void *data)
-{
-	struct chanspy_ds *chanspy_ds = data;
-
-	/* Setting chan to be NULL is an atomic operation, but we don't want this
-	 * value to change while this lock is held.  The lock is held elsewhere
-	 * while it performs non-atomic operations with this channel pointer */
-
-	ast_mutex_lock(&chanspy_ds->lock);
-	chanspy_ds->chan = NULL;
-	ast_mutex_unlock(&chanspy_ds->lock);
-}
-
-static void chanspy_ds_chan_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan)
-{
-	struct chanspy_ds *chanspy_ds = data;
-	
-	ast_mutex_lock(&chanspy_ds->lock);
-	chanspy_ds->chan = new_chan;
-	ast_mutex_unlock(&chanspy_ds->lock);
-}
-
-static const struct ast_datastore_info chanspy_ds_info = {
-	.type = "chanspy",
-	.destroy = chanspy_ds_destroy,
-	.chan_fixup = chanspy_ds_chan_fixup,
-};
-
-static struct chanspy_ds *chanspy_ds_free(struct chanspy_ds *chanspy_ds)
-{
-	struct ast_channel *chan;
-
-	if (!chanspy_ds) {
-		return NULL;
-	}
-
-	ast_mutex_lock(&chanspy_ds->lock);
-	while ((chan = chanspy_ds->chan)) {
-		struct ast_datastore *datastore;
-
-		if (ast_channel_trylock(chan)) {
-			DEADLOCK_AVOIDANCE(&chanspy_ds->lock);
-			continue;
-		}
-		if ((datastore = ast_channel_datastore_find(chan, &chanspy_ds_info, chanspy_ds->unique_id))) {
-			ast_channel_datastore_remove(chan, datastore);
-			/* chanspy_ds->chan is NULL after this call */
-			chanspy_ds_destroy(datastore->data);
-			datastore->data = NULL;
-			ast_channel_datastore_free(datastore);
-		}
-		ast_channel_unlock(chan);
-		break;
-	}
-	ast_mutex_unlock(&chanspy_ds->lock);
-
-	return NULL;
-}
-
-/*! \note Returns the channel in the chanspy_ds locked as well as the chanspy_ds locked */
-static struct chanspy_ds *setup_chanspy_ds(struct ast_channel *chan, struct chanspy_ds *chanspy_ds)
-{
-	struct ast_datastore *datastore = NULL;
-
-	ast_mutex_lock(&chanspy_ds->lock);
-
-	if (!(datastore = ast_channel_datastore_alloc(&chanspy_ds_info, chanspy_ds->unique_id))) {
-		ast_mutex_unlock(&chanspy_ds->lock);
-		chanspy_ds = chanspy_ds_free(chanspy_ds);
-		ast_channel_unlock(chan);
-		return NULL;
-	}
-	
-	chanspy_ds->chan = chan;
-	datastore->data = chanspy_ds;
-	ast_channel_datastore_add(chan, datastore);
-
-	return chanspy_ds;
-}
-
-static struct chanspy_ds *next_channel(struct ast_channel *chan,
-	const struct ast_channel *last, const char *spec,
-	const char *exten, const char *context, struct chanspy_ds *chanspy_ds)
-{
-	struct ast_channel *next;
-	const size_t pseudo_len = strlen("DAHDI/pseudo");
-
-redo:
-	if (!ast_strlen_zero(spec))
-		next = ast_walk_channel_by_name_prefix_locked(last, spec, strlen(spec));
-
-	else if (!ast_strlen_zero(exten))
-		next = ast_walk_channel_by_exten_locked(last, exten, context);
-	else
-		next = ast_channel_walk_locked(last);
-
-	if (!next)
-		return NULL;
-
-	if (!strncmp(next->name, "DAHDI/pseudo", pseudo_len)) {
-		last = next;
-		ast_channel_unlock(next);
-		goto redo;
-	} else if (next == chan) {
-		last = next;
-		ast_channel_unlock(next);
-		goto redo;
-	}
-
-	return setup_chanspy_ds(next, chanspy_ds);
-}
-
-static int common_exec(struct ast_channel *chan, const struct ast_flags *flags,
-	int volfactor, const int fd, const char *mygroup, const char *myenforced,
-	const char *spec, const char *exten, const char *context)
-{
-	char nameprefix[AST_NAME_STRLEN];
-	char peer_name[AST_NAME_STRLEN + 5];
-	char exitcontext[AST_MAX_CONTEXT] = "";
-	signed char zero_volume = 0;
-	int waitms;
-	int res;
-	char *ptr;
-	int num;
-	int num_spyed_upon = 1;
-	struct chanspy_ds chanspy_ds = { 0, };
-
-	if (ast_test_flag(flags, OPTION_EXIT)) {
-		const char *c;
-		if ((c = pbx_builtin_getvar_helper(chan, "SPY_EXIT_CONTEXT")))
-			ast_copy_string(exitcontext, c, sizeof(exitcontext));
-		else if (!ast_strlen_zero(chan->macrocontext))
-			ast_copy_string(exitcontext, chan->macrocontext, sizeof(exitcontext));
-		else
-			ast_copy_string(exitcontext, chan->context, sizeof(exitcontext));
-	}
-
-	ast_mutex_init(&chanspy_ds.lock);
-
-	snprintf(chanspy_ds.unique_id, sizeof(chanspy_ds.unique_id), "%d", ast_atomic_fetchadd_int(&next_unique_id_to_use, +1));
-
-	if (chan->_state != AST_STATE_UP)
-		ast_answer(chan);
-
-	ast_set_flag(chan, AST_FLAG_SPYING); /* so nobody can spy on us while we are spying */
-
-	waitms = 100;
-
-	for (;;) {
-		struct chanspy_ds *peer_chanspy_ds = NULL, *next_chanspy_ds = NULL;
-		struct ast_channel *prev = NULL, *peer = NULL;
-
-		if (!ast_test_flag(flags, OPTION_QUIET) && num_spyed_upon) {
-			res = ast_streamfile(chan, "beep", chan->language);
-			if (!res)
-				res = ast_waitstream(chan, "");
-			else if (res < 0) {
-				ast_clear_flag(chan, AST_FLAG_SPYING);
-				break;
-			}
-			if (!ast_strlen_zero(exitcontext)) {
-				char tmp[2];
-				tmp[0] = res;
-				tmp[1] = '\0';
-				if (!ast_goto_if_exists(chan, exitcontext, tmp, 1))
-					goto exit;
-				else
-					ast_debug(2, "Exit by single digit did not work in chanspy. Extension %s does not exist in context %s\n", tmp, exitcontext);
-			}
-		}
-
-		res = ast_waitfordigit(chan, waitms);
-		if (res < 0) {
-			ast_clear_flag(chan, AST_FLAG_SPYING);
-			break;
-		}
-		if (!ast_strlen_zero(exitcontext)) {
-			char tmp[2];
-			tmp[0] = res;
-			tmp[1] = '\0';
-			if (!ast_goto_if_exists(chan, exitcontext, tmp, 1))
-				goto exit;
-			else
-				ast_debug(2, "Exit by single digit did not work in chanspy. Extension %s does not exist in context %s\n", tmp, exitcontext);
-		}
-
-		/* reset for the next loop around, unless overridden later */
-		waitms = 100;
-		num_spyed_upon = 0;
-
-		for (peer_chanspy_ds = next_channel(chan, prev, spec, exten, context, &chanspy_ds);
-		     peer_chanspy_ds;
-			 chanspy_ds_free(peer_chanspy_ds), prev = peer,
-		     peer_chanspy_ds = next_chanspy_ds ? next_chanspy_ds : 
-			 	next_channel(chan, prev, spec, exten, context, &chanspy_ds), next_chanspy_ds = NULL) {
-			int igrp = !mygroup;
-			int ienf = !myenforced;
-			char *s;
-
-			peer = peer_chanspy_ds->chan;
-
-			ast_mutex_unlock(&peer_chanspy_ds->lock);
-
-			if (peer == prev) {
-				ast_channel_unlock(peer);
-				chanspy_ds_free(peer_chanspy_ds);
-				break;
-			}
-
-			if (ast_check_hangup(chan)) {
-				ast_channel_unlock(peer);
-				chanspy_ds_free(peer_chanspy_ds);
-				break;
-			}
-
-			if (ast_test_flag(flags, OPTION_BRIDGED) && !ast_bridged_channel(peer)) {
-				ast_channel_unlock(peer);
-				continue;
-			}
-
-			if (ast_check_hangup(peer) || ast_test_flag(peer, AST_FLAG_SPYING)) {
-				ast_channel_unlock(peer);
-				continue;
-			}
-
-			if (mygroup) {
-				int num_groups = 0;
-				char dup_group[512];
-				char *groups[25];
-				const char *group;
-				int x;
-				if ((group = pbx_builtin_getvar_helper(peer, "SPYGROUP"))) {
-					ast_copy_string(dup_group, group, sizeof(dup_group));
-					num_groups = ast_app_separate_args(dup_group, ':', groups,
-						ARRAY_LEN(groups));
-				}
-
-				for (x = 0; x < num_groups; x++) {
-					if (!strcmp(mygroup, groups[x])) {
-						igrp = 1;
-						break;
-					}
-				}
-			}
-
-			if (!igrp) {
-				ast_channel_unlock(peer);
-				continue;
-			}
-
-			if (myenforced) {
-				char ext[AST_CHANNEL_NAME + 3];
-				char buffer[512];
-				char *end;
-
-				snprintf(buffer, sizeof(buffer) - 1, ":%s:", myenforced);
-
-				ast_copy_string(ext + 1, peer->name, sizeof(ext) - 1);
-				if ((end = strchr(ext, '-'))) {
-					*end++ = ':';
-					*end = '\0';
-				}
-
-				ext[0] = ':';
-
-				if (strcasestr(buffer, ext)) {
-					ienf = 1;
-				}
-			}
-
-			if (!ienf) {
-				continue;
-			}
-
-			strcpy(peer_name, "spy-");
-			strncat(peer_name, peer->name, AST_NAME_STRLEN - 4 - 1);
-			ptr = strchr(peer_name, '/');
-			*ptr++ = '\0';
-
-			for (s = peer_name; s < ptr; s++)
-				*s = tolower(*s);
-			/* We have to unlock the peer channel here to avoid a deadlock.
-			 * So, when we need to dereference it again, we have to lock the 
-			 * datastore and get the pointer from there to see if the channel 
-			 * is still valid. */
-			ast_channel_unlock(peer);
-
-			if (!ast_test_flag(flags, OPTION_QUIET)) {
-				if (ast_fileexists(peer_name, NULL, NULL) != -1) {
-					res = ast_streamfile(chan, peer_name, chan->language);
-					if (!res)
-						res = ast_waitstream(chan, "");
-					if (res) {
-						chanspy_ds_free(peer_chanspy_ds);
-						break;
-					}
-				} else
-					res = ast_say_character_str(chan, peer_name, "", chan->language);
-				if ((num = atoi(ptr)))
-					ast_say_digits(chan, atoi(ptr), "", chan->language);
-			}
-
-			res = channel_spy(chan, peer_chanspy_ds, &volfactor, fd, flags, exitcontext);
-			num_spyed_upon++;	
-
-			if (res == -1) {
-				chanspy_ds_free(peer_chanspy_ds);
-				goto exit;
-			} else if (res == -2) {
-				res = 0;
-				chanspy_ds_free(peer_chanspy_ds);
-				goto exit;
-			} else if (res > 1 && spec) {
-				struct ast_channel *next;
-
-				snprintf(nameprefix, AST_NAME_STRLEN, "%s/%d", spec, res);
-
-				if ((next = ast_get_channel_by_name_prefix_locked(nameprefix, strlen(nameprefix)))) {
-					peer_chanspy_ds = chanspy_ds_free(peer_chanspy_ds);
-					next_chanspy_ds = setup_chanspy_ds(next, &chanspy_ds);
-				} else {
-					/* stay on this channel, if it is still valid */
-
-					ast_mutex_lock(&peer_chanspy_ds->lock);
-					if (peer_chanspy_ds->chan) {
-						ast_channel_lock(peer_chanspy_ds->chan);
-						next_chanspy_ds = peer_chanspy_ds;
-						peer_chanspy_ds = NULL;
-					} else {
-						/* the channel is gone */
-						ast_mutex_unlock(&peer_chanspy_ds->lock);
-						next_chanspy_ds = NULL;
-					}
-				}
-
-				peer = NULL;
-			}
-		}
-		if (res == -1 || ast_check_hangup(chan))
-			break;
-	}
-exit:
-
-	ast_clear_flag(chan, AST_FLAG_SPYING);
-
-	ast_channel_setoption(chan, AST_OPTION_TXGAIN, &zero_volume, sizeof(zero_volume), 0);
-
-	ast_mutex_lock(&chanspy_ds.lock);
-	ast_mutex_unlock(&chanspy_ds.lock);
-	ast_mutex_destroy(&chanspy_ds.lock);
-
-	return res;
-}
-
-static int chanspy_exec(struct ast_channel *chan, void *data)
-{
-	char *myenforced = NULL;
-	char *mygroup = NULL;
-	char *recbase = NULL;
-	int fd = 0;
-	struct ast_flags flags;
-	int oldwf = 0;
-	int volfactor = 0;
-	int res;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(spec);
-		AST_APP_ARG(options);
-	);
-	char *opts[OPT_ARG_ARRAY_SIZE];
-
-	data = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, data);
-
-	if (args.spec && !strcmp(args.spec, "all"))
-		args.spec = NULL;
-
-	if (args.options) {
-		ast_app_parse_options(spy_opts, &flags, opts, args.options);
-		if (ast_test_flag(&flags, OPTION_GROUP))
-			mygroup = opts[OPT_ARG_GROUP];
-
-		if (ast_test_flag(&flags, OPTION_RECORD) &&
-			!(recbase = opts[OPT_ARG_RECORD]))
-			recbase = "chanspy";
-
-		if (ast_test_flag(&flags, OPTION_VOLUME) && opts[OPT_ARG_VOLUME]) {
-			int vol;
-
-			if ((sscanf(opts[OPT_ARG_VOLUME], "%30d", &vol) != 1) || (vol > 4) || (vol < -4))
-				ast_log(LOG_NOTICE, "Volume factor must be a number between -4 and 4\n");
-			else
-				volfactor = vol;
-		}
-
-		if (ast_test_flag(&flags, OPTION_PRIVATE))
-			ast_set_flag(&flags, OPTION_WHISPER);
-
-		if (ast_test_flag(&flags, OPTION_ENFORCED))
-			myenforced = opts[OPT_ARG_ENFORCED];
-
-	} else
-		ast_clear_flag(&flags, AST_FLAGS_ALL);
-
-	oldwf = chan->writeformat;
-	if (ast_set_write_format(chan, AST_FORMAT_SLINEAR) < 0) {
-		ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
-		return -1;
-	}
-
-	if (recbase) {
-		char filename[PATH_MAX];
-
-		snprintf(filename, sizeof(filename), "%s/%s.%d.raw", ast_config_AST_MONITOR_DIR, recbase, (int) time(NULL));
-		if ((fd = open(filename, O_CREAT | O_WRONLY | O_TRUNC, AST_FILE_MODE)) <= 0) {
-			ast_log(LOG_WARNING, "Cannot open '%s' for recording\n", filename);
-			fd = 0;
-		}
-	}
-
-	res = common_exec(chan, &flags, volfactor, fd, mygroup, myenforced, args.spec, NULL, NULL);
-
-	if (fd)
-		close(fd);
-
-	if (oldwf && ast_set_write_format(chan, oldwf) < 0)
-		ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
-
-	return res;
-}
-
-static int extenspy_exec(struct ast_channel *chan, void *data)
-{
-	char *ptr, *exten = NULL;
-	char *mygroup = NULL;
-	char *recbase = NULL;
-	int fd = 0;
-	struct ast_flags flags;
-	int oldwf = 0;
-	int volfactor = 0;
-	int res;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(context);
-		AST_APP_ARG(options);
-	);
-
-	data = ast_strdupa(data);
-
-	AST_STANDARD_APP_ARGS(args, data);
-	if (!ast_strlen_zero(args.context) && (ptr = strchr(args.context, '@'))) {
-		exten = args.context;
-		*ptr++ = '\0';
-		args.context = ptr;
-	}
-
-	if (ast_strlen_zero(args.context))
-		args.context = ast_strdupa(chan->context);
-
-	if (args.options) {
-		char *opts[OPT_ARG_ARRAY_SIZE];
-
-		ast_app_parse_options(spy_opts, &flags, opts, args.options);
-		if (ast_test_flag(&flags, OPTION_GROUP))
-			mygroup = opts[OPT_ARG_GROUP];
-
-		if (ast_test_flag(&flags, OPTION_RECORD) &&
-			!(recbase = opts[OPT_ARG_RECORD]))
-			recbase = "chanspy";
-
-		if (ast_test_flag(&flags, OPTION_VOLUME) && opts[OPT_ARG_VOLUME]) {
-			int vol;
-
-			if ((sscanf(opts[OPT_ARG_VOLUME], "%30d", &vol) != 1) || (vol > 4) || (vol < -4))
-				ast_log(LOG_NOTICE, "Volume factor must be a number between -4 and 4\n");
-			else
-				volfactor = vol;
-		}
-
-		if (ast_test_flag(&flags, OPTION_PRIVATE))
-			ast_set_flag(&flags, OPTION_WHISPER);
-	} else
-		ast_clear_flag(&flags, AST_FLAGS_ALL);
-
-	oldwf = chan->writeformat;
-	if (ast_set_write_format(chan, AST_FORMAT_SLINEAR) < 0) {
-		ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
-		return -1;
-	}
-
-	if (recbase) {
-		char filename[PATH_MAX];
-
-		snprintf(filename, sizeof(filename), "%s/%s.%d.raw", ast_config_AST_MONITOR_DIR, recbase, (int) time(NULL));
-		if ((fd = open(filename, O_CREAT | O_WRONLY | O_TRUNC, AST_FILE_MODE)) <= 0) {
-			ast_log(LOG_WARNING, "Cannot open '%s' for recording\n", filename);
-			fd = 0;
-		}
-	}
-
-
-	res = common_exec(chan, &flags, volfactor, fd, mygroup, NULL, NULL, exten, args.context);
-
-	if (fd)
-		close(fd);
-
-	if (oldwf && ast_set_write_format(chan, oldwf) < 0)
-		ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
-
-	return res;
-}
-
-static int unload_module(void)
-{
-	int res = 0;
-
-	res |= ast_unregister_application(app_chan);
-	res |= ast_unregister_application(app_ext);
-
-	return res;
-}
-
-static int load_module(void)
-{
-	int res = 0;
-
-	res |= ast_register_application(app_chan, chanspy_exec, tdesc, desc_chan);
-	res |= ast_register_application(app_ext, extenspy_exec, tdesc, desc_ext);
-
-	return res;
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Listen to the audio of an active channel");

apps/app_controlplayback.c

@@ -1,191 +1,143 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Trivial application to control playback a sound file
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * 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>
 
-/*! \file
- *
- * \brief Trivial application to control playback of a sound file
- *
- * \author Mark Spencer <markster@digium.com>
- *
- * \ingroup applications
- */
+static char *tdesc = "Control Playback Application";
 
-#include "asterisk.h"
+static char *app = "ControlPlayback";
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+static char *synopsis = "Play a file with fast forward and rewind";
 
-#include "asterisk/pbx.h"
-#include "asterisk/app.h"
-#include "asterisk/module.h"
+static char *descrip = 
+"ControlPlayback(filename[|skipms[|ffchar[|rewchar[|stopchar[|pausechr]]]]]):\n"
+"  Plays  back  a  given  filename (do not put extension). Options may also\n"
+"  be included following a pipe symbol.  You can use * and # to rewind and\n"
+"  fast forward the playback specified. If 'stopchar' is added the file will\n"
+"  terminate playback when 'stopchar' is pressed. Returns -1 if the channel\n"
+"  was hung up, or if the file does not exist. Returns 0 otherwise.\n\n"
+"  Example:  exten => 1234,1,ControlPlayback(file|4000|*|#|1|0)\n\n";
 
-static const char *app = "ControlPlayback";
+STANDARD_LOCAL_USER;
 
-static const char *synopsis = "Play a file with fast forward and rewind";
-
-static const char *descrip =
-"  ControlPlayback(file[,skipms[,ff[,rew[,stop[,pause[,restart,options]]]]]]]):\n"
-"This application will play back the given filename. By default, the '*' key\n"
-"can be used to rewind, and the '#' key can be used to fast-forward.\n"
-"Parameters:\n"
-"  skipms  - This is number of milliseconds to skip when rewinding or\n"
-"            fast-forwarding.\n"
-"  ff      - Fast-forward when this DTMF digit is received.\n"
-"  rew     - Rewind when this DTMF digit is received.\n"
-"  stop    - Stop playback when this DTMF digit is received.\n"
-"  pause   - Pause playback when this DTMF digit is received.\n"
-"  restart - Restart playback when this DTMF digit is received.\n"
-"Options:\n"
-"  o(#) - Start at # ms from the beginning of the file.\n"
-"This application sets the following channel variables upon completion:\n"
-"  CPLAYBACKSTATUS -  This variable contains the status of the attempt as a text\n"
-"                     string, one of: SUCCESS | USERSTOPPED | ERROR\n"
-"  CPLAYBACKOFFSET -  This contains the offset in ms into the file where\n"
-"                     playback was at when it stopped.  -1 is end of file.\n"
-"  CPLAYBACKSTOPKEY - If the playback is stopped by the user this variable contains\n"
-"                     the key that was pressed.\n";
-
-enum {
-	OPT_OFFSET = (1 << 1),
-};
-
-enum {
-	OPT_ARG_OFFSET = 0,
-	/* must stay as the last entry ... */
-	OPT_ARG_ARRAY_LEN,
-};
-
-AST_APP_OPTIONS(cpb_opts, BEGIN_OPTIONS
-	AST_APP_OPTION_ARG('o', OPT_OFFSET, OPT_ARG_OFFSET),
-	END_OPTIONS
-);
+LOCAL_USER_DECL;
 
 static int is_on_phonepad(char key)
 {
-	return key == 35 || key == 42 || (key >= 48 && key <= 57);
-}
-
-static int is_argument(const char *haystack, int needle)
-{
-	if (ast_strlen_zero(haystack))
-		return 0;
-
-	if (strchr(haystack, needle))
-		return -1;
-
-	return 0;
+	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;
-	long offsetms = 0;
-	char offsetbuf[20];
-	char stopkeybuf[2];
-	char *tmp;
-	struct ast_flags opts = { 0, };
-	char *opt_args[OPT_ARG_ARRAY_LEN];
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(filename);
-		AST_APP_ARG(skip);
-		AST_APP_ARG(fwd);
-		AST_APP_ARG(rev);
-		AST_APP_ARG(stop);
-		AST_APP_ARG(pause);
-		AST_APP_ARG(restart);
-		AST_APP_ARG(options);
-	);
+	struct localuser *u;
+	char tmp[256];
+	char *skip = NULL, *fwd = NULL, *rev = NULL, *stop = NULL, *pause = NULL, *file = NULL;
 
-	if (ast_strlen_zero(data)) {
+	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';
+		*skip++;
+		fwd=strchr(skip,'|');
+		if (fwd) {
+			*fwd = '\0';
+			*fwd++;
+			rev = strchr(fwd,'|');
+			if (rev) {
+				*rev = '\0';
+				*rev++;
+				stop = strchr(rev,'|');
+				if (stop) {
+					*stop = '\0';
+					*stop++;
+					pause = strchr(stop,'|');
+					if (pause) {
+						*pause = '\0';
+						*pause++;
+					}
+				}
+			}
+		}
+	}
+
+	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);
 	
-	tmp = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, tmp);
-
-	if (args.argc < 1) {
-		ast_log(LOG_WARNING, "ControlPlayback requires an argument (filename)\n");
-		return -1;
-	}
-
-	skipms = args.skip ? (atoi(args.skip) ? atoi(args.skip) : 3000) : 3000;
-
-	if (!args.fwd || !is_on_phonepad(*args.fwd)) {
-		char *digit = "#";
-		if (!is_argument(args.rev, *digit) && !is_argument(args.stop, *digit) && !is_argument(args.pause, *digit) && !is_argument(args.restart, *digit))
-			args.fwd = digit;
-		else
-			args.fwd = NULL;
-	}
-	if (!args.rev || !is_on_phonepad(*args.rev)) {
-		char *digit = "*";
-		if (!is_argument(args.fwd, *digit) && !is_argument(args.stop, *digit) && !is_argument(args.pause, *digit) && !is_argument(args.restart, *digit))
-			args.rev = digit;
-		else
-			args.rev = NULL;
-	}
-	ast_debug(1, "Forward key = %s, Rewind key = %s\n", args.fwd, args.rev);
-	if (args.stop && !is_on_phonepad(*args.stop))
-		args.stop = NULL;
-	if (args.pause && !is_on_phonepad(*args.pause))
-		args.pause = NULL;
-	if (args.restart && !is_on_phonepad(*args.restart))
-		args.restart = NULL;
-
-	if (args.options) {
-		ast_app_parse_options(cpb_opts, &opts, opt_args, args.options);
-		if (ast_test_flag(&opts, OPT_OFFSET))
-			offsetms = atol(opt_args[OPT_ARG_OFFSET]);
-	}
-
-	res = ast_control_streamfile(chan, args.filename, args.fwd, args.rev, args.stop, args.pause, args.restart, skipms, &offsetms);
-
 	/* If we stopped on one of our stop keys, return 0  */
-	if (res > 0 && args.stop && strchr(args.stop, res)) {
-		pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "USERSTOPPED");
-		snprintf(stopkeybuf, sizeof(stopkeybuf), "%c", res);
-		pbx_builtin_setvar_helper(chan, "CPLAYBACKSTOPKEY", stopkeybuf);
+	if(stop && strchr(stop, res)) 
 		res = 0;
-	} else {
-		if (res < 0) {
-			res = 0;
-			pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "ERROR");
-		} else
-			pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "SUCCESS");
-	}
-
-	snprintf(offsetbuf, sizeof(offsetbuf), "%ld", offsetms);
-	pbx_builtin_setvar_helper(chan, "CPLAYBACKOFFSET", offsetbuf);
-
+	
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
-	int res;
-	res = ast_unregister_application(app);
-	return res;
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, controlplayback_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Control Playback Application");
+char *description(void)
+{
+	return 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 = "Cut(newvar=varname|delimiter|fieldspec)";
+
+static char *cut_descrip =
+"Cut(newvar=varname,delimiter,field)\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_dahdiras.c

@@ -1,253 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2005, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Execute an ISDN RAS
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
- */
-
-/*** MODULEINFO
-	<depend>dahdi</depend>
-	<depend>working_fork</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <sys/ioctl.h>
-#include <sys/wait.h>
-#ifdef __linux__
-#include <sys/signal.h>
-#else
-#include <signal.h>
-#endif /* __linux__ */
-
-#include <dahdi/user.h>
-
-#include <fcntl.h>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
-
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-
-static char *app = "DAHDIRAS";
-
-static char *synopsis = "Executes DAHDI ISDN RAS application";
-
-static char *descrip =
-"  DAHDIRAS(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 DAHDI\n"
-"channel to be able to use this function (No modem emulation is included).\n"
-"Your pppd must be patched to be dahdi aware. Arguments should be\n"
-"separated by , characters.\n";
-
-
-#define PPP_MAX_ARGS	32
-#define PPP_EXEC	"/usr/sbin/pppd"
-
-static pid_t spawn_ras(struct ast_channel *chan, char *args)
-{
-	pid_t pid;
-	int x;	
-	char *c;
-
-	char *argv[PPP_MAX_ARGS];
-	int argc = 0;
-	char *stringp=NULL;
-	sigset_t fullset, oldset;
-#ifdef HAVE_CAP
-	cap_t cap;
-#endif
-
-	sigfillset(&fullset);
-	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
-
-	/* Start by forking */
-	pid = fork();
-	if (pid) {
-		pthread_sigmask(SIG_SETMASK, &oldset, NULL);
-		return pid;
-	}
-
-#ifdef HAVE_CAP
-	cap = cap_from_text("cap_net_admin-eip");
-
-	if (cap_set_proc(cap)) {
-		/* Careful with order! Logging cannot happen after we close FDs */
-		ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-	}
-	cap_free(cap);
-#endif
-
-	/* Restore original signal handlers */
-	for (x=0;x<NSIG;x++)
-		signal(x, SIG_DFL);
-
-	pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
-
-	/* Execute RAS on File handles */
-	dup2(chan->fds[0], STDIN_FILENO);
-
-	/* Drop high priority */
-	if (ast_opt_high_priority)
-		ast_set_priority(0);
-
-	/* Close other file descriptors */
-	for (x=STDERR_FILENO + 1;x<1024;x++) 
-		close(x);
-
-	/* Reset all arguments */
-	memset(argv, 0, sizeof(argv));
-
-	/* First argument is executable, followed by standard
-	   arguments for dahdi PPP */
-	argv[argc++] = PPP_EXEC;
-	argv[argc++] = "nodetach";
-
-	/* And all the other arguments */
-	stringp=args;
-	c = strsep(&stringp, ",");
-	while(c && strlen(c) && (argc < (PPP_MAX_ARGS - 4))) {
-		argv[argc++] = c;
-		c = strsep(&stringp, ",");
-	}
-
-	argv[argc++] = "plugin";
-	argv[argc++] = "dahdi.so";
-	argv[argc++] = "stdin";
-
-	/* Finally launch PPP */
-	execv(PPP_EXEC, argv);
-	fprintf(stderr, "Failed to exec PPPD!\n");
-	exit(1);
-}
-
-static void run_ras(struct ast_channel *chan, char *args)
-{
-	pid_t pid;
-	int status;
-	int res;
-	int signalled = 0;
-	struct dahdi_bufferinfo savebi;
-	int x;
-	
-	res = ioctl(chan->fds[0], DAHDI_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) {
-		ast_log(LOG_WARNING, "Failed to spawn RAS\n");
-	} else {
-		for (;;) {
-			res = wait4(pid, &status, WNOHANG, NULL);
-			if (!res) {
-				/* Check for hangup */
-				if (ast_check_hangup(chan) && !signalled) {
-					ast_debug(1, "Channel '%s' hungup.  Signalling RAS at %d to die...\n", chan->name, pid);
-					kill(pid, SIGTERM);
-					signalled=1;
-				}
-				/* Try again */
-				sleep(1);
-				continue;
-			}
-			if (res < 0) {
-				ast_log(LOG_WARNING, "wait4 returned %d: %s\n", res, strerror(errno));
-			}
-			if (WIFEXITED(status)) {
-				ast_verb(3, "RAS on %s terminated with status %d\n", chan->name, WEXITSTATUS(status));
-			} else if (WIFSIGNALED(status)) {
-				ast_verb(3, "RAS on %s terminated with signal %d\n", 
-					 chan->name, WTERMSIG(status));
-			} else {
-				ast_verb(3, "RAS on %s terminated weirdly.\n", chan->name);
-			}
-			/* Throw back into audio mode */
-			x = 1;
-			ioctl(chan->fds[0], DAHDI_AUDIOMODE, &x);
-
-			/* Restore saved values */
-			res = ioctl(chan->fds[0], DAHDI_SET_BUFINFO, &savebi);
-			if (res < 0) {
-				ast_log(LOG_WARNING, "Unable to set buffer policy on channel %s\n", chan->name);
-			}
-			break;
-		}
-	}
-}
-
-static int dahdiras_exec(struct ast_channel *chan, void *data)
-{
-	int res=-1;
-	char *args;
-	struct dahdi_params dahdip;
-
-	if (!data) 
-		data = "";
-
-	args = ast_strdupa(data);
-	
-	/* Answer the channel if it's not up */
-	if (chan->_state != AST_STATE_UP)
-		ast_answer(chan);
-	if (strcasecmp(chan->tech->type, "DAHDI")) {
-		/* If it's not a DAHDI channel, we're done.  Wait a couple of
-		   seconds and then hangup... */
-		ast_verb(2, "Channel %s is not a DAHDI channel\n", chan->name);
-		sleep(2);
-	} else {
-		memset(&dahdip, 0, sizeof(dahdip));
-		if (ioctl(chan->fds[0], DAHDI_GET_PARAMS, &dahdip)) {
-			ast_log(LOG_WARNING, "Unable to get DAHDI parameters\n");
-		} else if (dahdip.sigtype != DAHDI_SIG_CLEAR) {
-			ast_verb(2, "Channel %s is not a clear channel\n", chan->name);
-		} else {
-			/* Everything should be okay.  Run PPP. */
-			ast_verb(3, "Starting RAS on %s\n", chan->name);
-			/* Execute RAS */
-			run_ras(chan, args);
-		}
-	}
-	return res;
-}
-
-static int unload_module(void) 
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
-{
-	return ((ast_register_application(app, dahdiras_exec, synopsis, descrip)) ? AST_MODULE_LOAD_FAILURE : AST_MODULE_LOAD_SUCCESS);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "DAHDI ISDN Remote Access Server");
-

apps/app_dahdiscan.c

@@ -1,364 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2005, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * Modified from app_dahdibarge by David Troy <dave@toad.net>
- *
- * Special thanks to comphealth.com for sponsoring this
- * GPL application.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief DAHDI Scanner
- *
- * \author Mark Spencer <markster@digium.com>
- *
- * \ingroup applications
- */
-
-/*** MODULEINFO
-	<depend>dahdi</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <sys/ioctl.h>
-#include <dahdi/user.h>
-
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/config.h"
-#include "asterisk/app.h"
-#include "asterisk/utils.h"
-#include "asterisk/cli.h"
-#include "asterisk/say.h"
-
-static char *app = "DAHDIScan";
-
-static char *synopsis = "Scan DAHDI channels to monitor calls";
-
-static char *descrip =
-"  DAHDIScan([group]) allows a call center manager to monitor DAHDI channels in\n"
-"a convenient way.  Use '#' to select the next channel and use '*' to exit\n"
-"Limit scanning to a channel GROUP by setting the option group argument.\n";
-
-
-#define CONF_SIZE 160
-
-static struct ast_channel *get_dahdi_channel_locked(int num) {
-	char name[80];
-	
-	snprintf(name, sizeof(name), "DAHDI/%d-1", num);
-	return ast_get_channel_by_name_locked(name);
-}
-
-static int careful_write(int fd, unsigned char *data, int len)
-{
-	int res;
-	while(len) {
-		res = write(fd, data, len);
-		if (res < 1) {
-			if (errno != EAGAIN) {
-				ast_log(LOG_WARNING, "Failed to write audio data to conference: %s\n", strerror(errno));
-				return -1;
-			} else
-				return 0;
-		}
-		len -= res;
-		data += res;
-	}
-	return 0;
-}
-
-static int conf_run(struct ast_channel *chan, int confno, int confflags)
-{
-	int fd;
-	struct dahdi_confinfo dahdic;
-	struct ast_frame *f;
-	struct ast_channel *c;
-	struct ast_frame fr;
-	int outfd;
-	int ms;
-	int nfds;
-	int res;
-	int flags;
-	int retrydahdi;
-	int origfd;
-	int ret = -1;
-	char input[4];
-	int ic=0;
-	
-	struct dahdi_bufferinfo bi;
-	char __buf[CONF_SIZE + AST_FRIENDLY_OFFSET];
-	char *buf = __buf + AST_FRIENDLY_OFFSET;
-	
-	/* Set it into U-law mode (write) */
-	if (ast_set_write_format(chan, AST_FORMAT_ULAW) < 0) {
-		ast_log(LOG_WARNING, "Unable to set '%s' to write ulaw mode\n", chan->name);
-		goto outrun;
-	}
-	
-	/* Set it into U-law mode (read) */
-	if (ast_set_read_format(chan, AST_FORMAT_ULAW) < 0) {
-		ast_log(LOG_WARNING, "Unable to set '%s' to read ulaw mode\n", chan->name);
-		goto outrun;
-	}
-	ast_indicate(chan, -1);
-	retrydahdi = strcasecmp(chan->tech->type, "DAHDI");
- dahdiretry:
-	origfd = chan->fds[0];
-	if (retrydahdi) {
-		fd = open("/dev/dahdi/pseudo", O_RDWR);
-		if (fd < 0) {
-			ast_log(LOG_WARNING, "Unable to open pseudo channel: %s\n", strerror(errno));
-			goto outrun;
-		}
-		/* Make non-blocking */
-		flags = fcntl(fd, F_GETFL);
-		if (flags < 0) {
-			ast_log(LOG_WARNING, "Unable to get flags: %s\n", strerror(errno));
-			close(fd);
-            goto outrun;
-		}
-		if (fcntl(fd, F_SETFL, flags | O_NONBLOCK)) {
-			ast_log(LOG_WARNING, "Unable to set flags: %s\n", strerror(errno));
-			close(fd);
-			goto outrun;
-		}
-		/* Setup buffering information */
-		memset(&bi, 0, sizeof(bi));
-		bi.bufsize = CONF_SIZE;
-		bi.txbufpolicy = DAHDI_POLICY_IMMEDIATE;
-		bi.rxbufpolicy = DAHDI_POLICY_IMMEDIATE;
-		bi.numbufs = 4;
-		if (ioctl(fd, DAHDI_SET_BUFINFO, &bi)) {
-			ast_log(LOG_WARNING, "Unable to set buffering information: %s\n", strerror(errno));
-			close(fd);
-			goto outrun;
-		}
-        nfds = 1;
-	} else {
-		/* XXX Make sure we're not running on a pseudo channel XXX */
-		fd = chan->fds[0];
-		nfds = 0;
-	}
-	memset(&dahdic, 0, sizeof(dahdic));
-	/* Check to see if we're in a conference... */
-        dahdic.chan = 0;
-        if (ioctl(fd, DAHDI_GETCONF, &dahdic)) {
-			ast_log(LOG_WARNING, "Error getting conference\n");
-			close(fd);
-			goto outrun;
-        }
-        if (dahdic.confmode) {
-			/* Whoa, already in a conference...  Retry... */
-			if (!retrydahdi) {
-				ast_debug(1, "DAHDI channel is in a conference already, retrying with pseudo\n");
-				retrydahdi = 1;
-				goto dahdiretry;
-			}
-        }
-        memset(&dahdic, 0, sizeof(dahdic));
-        /* Add us to the conference */
-        dahdic.chan = 0;
-        dahdic.confno = confno;
-        dahdic.confmode = DAHDI_CONF_MONITORBOTH;
-		
-        if (ioctl(fd, DAHDI_SETCONF, &dahdic)) {
-                ast_log(LOG_WARNING, "Error setting conference\n");
-                close(fd);
-                goto outrun;
-        }
-	ast_debug(1, "Placed channel %s in DAHDI channel %d monitor\n", chan->name, confno);
-		
-        for(;;) {
-			outfd = -1;
-			ms = -1;
-			c = ast_waitfor_nandfds(&chan, 1, &fd, nfds, NULL, &outfd, &ms);
-			if (c) {
-				if (c->fds[0] != origfd) {
-					if (retrydahdi) {
-						/* Kill old pseudo */
-						close(fd);
-					}
-					ast_debug(1, "Ooh, something swapped out under us, starting over\n");
-					retrydahdi = 0;
-                                goto dahdiretry;
-				}
-				f = ast_read(c);
-				if (!f)
-					break;
-				if(f->frametype == AST_FRAME_DTMF) {
-					if(f->subclass == '#') {
-						ret = 0;
-						break;
-					}
-					else if (f->subclass == '*') {
-						ret = -1;
-						break;
-						
-					}
-					else {
-						input[ic++] = f->subclass;
-					}
-					if(ic == 3) {
-						input[ic++] = '\0';
-						ic=0;
-						ret = atoi(input);
-						ast_verb(3, "DAHDIscan: change channel to %d\n",ret);
-						break;
-					}
-				}
-				
-				if (fd != chan->fds[0]) {
-					if (f->frametype == AST_FRAME_VOICE) {
-						if (f->subclass == AST_FORMAT_ULAW) {
-							/* Carefully write */
-                                                careful_write(fd, f->data, f->datalen);
-						} else
-							ast_log(LOG_WARNING, "Huh?  Got a non-ulaw (%d) frame in the conference\n", f->subclass);
-					}
-				}
-				ast_frfree(f);
-			} else if (outfd > -1) {
-				res = read(outfd, buf, CONF_SIZE);
-				if (res > 0) {
-					memset(&fr, 0, sizeof(fr));
-					fr.frametype = AST_FRAME_VOICE;
-					fr.subclass = AST_FORMAT_ULAW;
-					fr.datalen = res;
-					fr.samples = res;
-					fr.data = buf;
-					fr.offset = AST_FRIENDLY_OFFSET;
-					if (ast_write(chan, &fr) < 0) {
-						ast_log(LOG_WARNING, "Unable to write frame to channel: %s\n", strerror(errno));
-						/* break; */
-					}
-				} else
-					ast_log(LOG_WARNING, "Failed to read frame: %s\n", strerror(errno));
-			}
-        }
-	if (f)
-		ast_frfree(f);
-        if (fd != chan->fds[0])
-			close(fd);
-        else {
-			/* Take out of conference */
-			/* Add us to the conference */
-			dahdic.chan = 0;
-			dahdic.confno = 0;
-			dahdic.confmode = 0;
-			if (ioctl(fd, DAHDI_SETCONF, &dahdic)) {
-				ast_log(LOG_WARNING, "Error setting conference\n");
-                }
-        }
-		
- outrun:
-		
-        return ret;
-}
-
-static int conf_exec(struct ast_channel *chan, void *data)
-{
-	int res=-1;
-	int confflags = 0;
-	int confno = 0;
-	char confstr[80] = "", *tmp = NULL;
-	struct ast_channel *tempchan = NULL, *lastchan = NULL,*ichan = NULL;
-	struct ast_frame *f;
-	char *desired_group;
-	int input=0,search_group=0;
-		
-	if (chan->_state != AST_STATE_UP)
-		ast_answer(chan);
-	
-	desired_group = ast_strdupa(data);
-	if(!ast_strlen_zero(desired_group)) {
-		ast_verb(3, "Scanning for group %s\n", desired_group);
-		search_group = 1;
-	}
-
-	for (;;) {
-		if (ast_waitfor(chan, 100) < 0)
-			break;
-		
-		f = ast_read(chan);
-		if (!f)
-			break;
-		if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '*')) {
-			ast_frfree(f);
-			break;
-		}
-		ast_frfree(f);
-		ichan = NULL;
-		if(input) {
-			ichan = get_dahdi_channel_locked(input);
-			input = 0;
-		}
-		
-		tempchan = ichan ? ichan : ast_channel_walk_locked(tempchan);
-		
-		if ( !tempchan && !lastchan )
-			break;
-		
-		if (tempchan && search_group) {
-			const char *mygroup;
-			if((mygroup = pbx_builtin_getvar_helper(tempchan, "GROUP")) && (!strcmp(mygroup, desired_group))) {
-				ast_verb(3, "Found Matching Channel %s in group %s\n", tempchan->name, desired_group);
-			} else {
-				ast_channel_unlock(tempchan);
-				lastchan = tempchan;
-				continue;
-			}
-		}
-		if (tempchan && (!strcmp(tempchan->tech->type, "DAHDI")) && (tempchan != chan) ) {
-			ast_verb(3, "DAHDI channel %s is in-use, monitoring...\n", tempchan->name);
-			ast_copy_string(confstr, tempchan->name, sizeof(confstr));
-			ast_channel_unlock(tempchan);
-			if ((tmp = strchr(confstr,'-'))) {
-				*tmp = '\0';
-			}
-			confno = atoi(strchr(confstr,'/') + 1);
-			ast_stopstream(chan);
-			ast_say_number(chan, confno, AST_DIGIT_ANY, chan->language, (char *) NULL);
-			res = conf_run(chan, confno, confflags);
-			if (res<0) break;
-			input = res;
-		} else if (tempchan)
-			ast_channel_unlock(tempchan);
-		lastchan = tempchan;
-	}
-	return res;
-}
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
-{
-	return ((ast_register_application(app, conf_exec, synopsis, descrip)) ? AST_MODULE_LOAD_FAILURE : AST_MODULE_LOAD_SUCCESS);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Scan DAHDI channels application");
-

apps/app_datetime.c

@@ -0,0 +1,81 @@
+/*
+ * 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/module.h>
+#include <asterisk/say.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+
+static char *tdesc = "Date and Time";
+
+static char *app = "DateTime";
+
+static char *synopsis = "Say the date and time";
+
+static char *descrip = 
+"  DateTime():  Says the current date and time.  Returns -1 on hangup or 0\n"
+"otherwise.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int datetime_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	time_t t;
+	struct localuser *u;
+	LOCAL_USER_ADD(u);
+	time(&t);
+	if (chan->_state != AST_STATE_UP)
+		res = ast_answer(chan);
+	if (!res)
+		res = ast_say_datetime(chan, t, "", chan->language);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, datetime_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_db.c

@@ -1,139 +1,255 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Database access functions
+ *
+ * Copyright (C) 1999, Mark Spencer
  * Copyright (C) 2003, Jefferson Noxon
  *
- * Mark Spencer <markster@digium.com>
+ * Mark Spencer <markster@linux-support.net>
  * Jefferson Noxon <jeff@debian.org>
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
 
-/*! \file
- *
- * \brief Database access functions
- *
- * \author Mark Spencer <markster@digium.com>
- * \author Jefferson Noxon <jeff@debian.org>
- *
- * \ingroup applications
- */
+#include <sys/types.h>
+#include <asterisk/options.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/astdb.h>
+#include <asterisk/lock.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
 
-#include "asterisk.h"
+static char *tdesc = "Database access functions for Asterisk extension logic";
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+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";
 
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/astdb.h"
-#include "asterisk/lock.h"
+static char *p_descrip =
+	"  DBput(family/key=value): Stores the given value in the Asterisk\n"
+	"database.  Always returns 0.\n";
 
-/*! \todo XXX Remove this application after 1.4 is relased */
 static char *d_descrip =
-"  DBdel(family/key): This application will delete a key from the Asterisk\n"
-"database.\n"
-"  This application has been DEPRECATED in favor of the DB_DELETE function.\n";
+	"  DBdel(family/key): Deletes a key from the Asterisk database.  Always\n"
+	"returns 0.\n";
 
 static char *dt_descrip =
-"  DBdeltree(family[/keytree]): This application will delete a family or keytree\n"
-"from the Asterisk database\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";
 static char *d_app = "DBdel";
 static char *dt_app = "DBdeltree";
 
+static char *g_synopsis = "Retrieve a value from the database";
+static char *p_synopsis = "Store a value in the database";
 static char *d_synopsis = "Delete a key from the database";
 static char *dt_synopsis = "Delete a family or keytree from the database";
 
+STANDARD_LOCAL_USER;
 
-static int deltree_exec(struct ast_channel *chan, void *data)
+LOCAL_USER_DECL;
+
+static int deltree_exec (struct ast_channel *chan, void *data)
 {
+	int arglen;
 	char *argv, *family, *keytree;
 
-	argv = ast_strdupa(data);
+	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_debug(1, "Ignoring; Syntax error in argument\n");
-			return 0;
-		}
-		if (ast_strlen_zero(keytree))
+	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;
 	}
 
-	if (keytree)
-		ast_verb(3, "DBdeltree: family=%s, keytree=%s\n", family, keytree);
-	else
-		ast_verb(3, "DBdeltree: family=%s\n", family);
+	if (option_verbose > 2)	{
+		if (keytree)
+			ast_verbose (VERBOSE_PREFIX_3 "DBdeltree: family=%s, keytree=%s\n", family, keytree);
+		else
+			ast_verbose (VERBOSE_PREFIX_3 "DBdeltree: family=%s\n", family);
+	}
 
-	if (ast_db_deltree(family, keytree))
-		ast_verb(3, "DBdeltree: Error deleting key from database.\n");
+	if (ast_db_deltree (family, keytree)) {
+		if (option_verbose > 2)
+			ast_verbose (VERBOSE_PREFIX_3 "DBdeltree: Error deleting key from database.\n");
+	}
 
 	return 0;
 }
 
-static int del_exec(struct ast_channel *chan, void *data)
+static int del_exec (struct ast_channel *chan, void *data)
 {
+	int arglen;
 	char *argv, *family, *key;
-	static int deprecation_warning = 0;
 
-	if (!deprecation_warning) {
-		deprecation_warning = 1;
-		ast_log(LOG_WARNING, "The DBdel application has been deprecated in favor of the DB_DELETE dialplan function!\n");
+	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);
 
-	argv = ast_strdupa(data);
-
-	if (strchr(argv, '/')) {
-		family = strsep(&argv, "/");
-		key = strsep(&argv, "\0");
+	if (strchr (argv, '/')) {
+		family = strsep (&argv, "/");
+		key = strsep (&argv, "\0");
 		if (!family || !key) {
-			ast_debug(1, "Ignoring; Syntax error in argument\n");
+			ast_log (LOG_DEBUG, "Ignoring; Syntax error in argument\n");
 			return 0;
 		}
-		ast_verb(3, "DBdel: family=%s, key=%s\n", family, key);
-		if (ast_db_del(family, key))
-			ast_verb(3, "DBdel: Error deleting key from database.\n");
+		if (option_verbose > 2)
+			ast_verbose (VERBOSE_PREFIX_3 "DBdel: family=%s, key=%s\n", family, key);
+		if (ast_db_del (family, key)) {
+			if (option_verbose > 2)
+				ast_verbose (VERBOSE_PREFIX_3 "DBdel: Error deleting key from database.\n");
+		}
 	} else {
-		ast_debug(1, "Ignoring, no parameters\n");
+		ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
 	}
-
 	return 0;
 }
 
-static int unload_module(void)
+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");
+		}
+
+	} 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;
+		}
+
+	} else {
+		ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
+	}
+	return 0;
+}
+
+int unload_module (void)
 {
 	int retval;
 
-	retval = ast_unregister_application(dt_app);
-	retval |= ast_unregister_application(d_app);
+	STANDARD_HANGUP_LOCALUSERS;
+	retval = ast_unregister_application (dt_app);
+	retval |= ast_unregister_application (d_app);
+	retval |= ast_unregister_application (p_app);
+	retval |= ast_unregister_application (g_app);
 
 	return retval;
 }
 
-static int load_module(void)
+int load_module (void)
 {
 	int retval;
 
-	retval = ast_register_application(d_app, del_exec, d_synopsis, d_descrip);
-	retval |= ast_register_application(dt_app, deltree_exec, dt_synopsis, dt_descrip);
-
+	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;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Database Access Functions");
+char *description (void)
+{
+	return tdesc;
+}
+
+int usecount (void)
+{
+	int res;
+	STANDARD_USECOUNT (res);
+	return res;
+}
+
+char *key ()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_dictate.c

@@ -1,340 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2005, Anthony Minessale II
- *
- * Anthony Minessale II <anthmct@yahoo.com>
- *
- * Donated by Sangoma Technologies <http://www.samgoma.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Virtual Dictation Machine Application For Asterisk
- *
- * \author Anthony Minessale II <anthmct@yahoo.com>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <sys/stat.h>
-
-#include "asterisk/paths.h" /* use ast_config_AST_SPOOL_DIR */
-#include "asterisk/file.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/say.h"
-#include "asterisk/app.h"
-
-static char *app = "Dictate";
-static char *synopsis = "Virtual Dictation Machine";
-static char *desc = "  Dictate([<base_dir>[,<filename>]])\n"
-"Start dictation machine using optional base dir for files.\n";
-
-
-typedef enum {
-	DFLAG_RECORD = (1 << 0),
-	DFLAG_PLAY = (1 << 1),
-	DFLAG_TRUNC = (1 << 2),
-	DFLAG_PAUSE = (1 << 3),
-} dflags;
-
-typedef enum {
-	DMODE_INIT,
-	DMODE_RECORD,
-	DMODE_PLAY
-} dmodes;
-
-#define ast_toggle_flag(it,flag) if(ast_test_flag(it, flag)) ast_clear_flag(it, flag); else ast_set_flag(it, flag)
-
-static int play_and_wait(struct ast_channel *chan, char *file, char *digits)
-{
-	int res = -1;
-	if (!ast_streamfile(chan, file, chan->language)) {
-		res = ast_waitstream(chan, digits);
-	}
-	return res;
-}
-
-static int dictate_exec(struct ast_channel *chan, void *data)
-{
-	char *path = NULL, filein[256], *filename = "";
-	char *parse;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(base);
-		AST_APP_ARG(filename);
-	);
-	char dftbase[256];
-	char *base;
-	struct ast_flags flags = {0};
-	struct ast_filestream *fs;
-	struct ast_frame *f = NULL;
-	int ffactor = 320 * 80,
-		res = 0,
-		done = 0,
-		oldr = 0,
-		lastop = 0,
-		samples = 0,
-		speed = 1,
-		digit = 0,
-		len = 0,
-		maxlen = 0,
-		mode = 0;
-
-	snprintf(dftbase, sizeof(dftbase), "%s/dictate", ast_config_AST_SPOOL_DIR);
-	if (!ast_strlen_zero(data)) {
-		parse = ast_strdupa(data);
-		AST_STANDARD_APP_ARGS(args, parse);
-	} else
-		args.argc = 0;
-
-	if (args.argc && !ast_strlen_zero(args.base)) {
-		base = args.base;
-	} else {
-		base = dftbase;
-	}
-	if (args.argc > 1 && args.filename) {
-		filename = args.filename;
-	}
-	oldr = chan->readformat;
-	if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR)) < 0) {
-		ast_log(LOG_WARNING, "Unable to set to linear mode.\n");
-		return -1;
-	}
-
-	if (chan->_state != AST_STATE_UP) {
-		ast_answer(chan);
-	}
-	ast_safe_sleep(chan, 200);
-	for (res = 0; !res;) {
-		if (ast_strlen_zero(filename)) {
-			if (ast_app_getdata(chan, "dictate/enter_filename", filein, sizeof(filein), 0) ||
-				ast_strlen_zero(filein)) {
-				res = -1;
-				break;
-			}
-		} else {
-			ast_copy_string(filein, filename, sizeof(filein));
-			filename = "";
-		}
-		ast_mkdir(base, 0755);
-		len = strlen(base) + strlen(filein) + 2;
-		if (!path || len > maxlen) {
-			path = alloca(len);
-			memset(path, 0, len);
-			maxlen = len;
-		} else {
-			memset(path, 0, maxlen);
-		}
-
-		snprintf(path, len, "%s/%s", base, filein);
-		fs = ast_writefile(path, "raw", NULL, O_CREAT|O_APPEND, 0, AST_FILE_MODE);
-		mode = DMODE_PLAY;
-		memset(&flags, 0, sizeof(flags));
-		ast_set_flag(&flags, DFLAG_PAUSE);
-		digit = play_and_wait(chan, "dictate/forhelp", AST_DIGIT_ANY);
-		done = 0;
-		speed = 1;
-		res = 0;
-		lastop = 0;
-		samples = 0;
-		while (!done && ((res = ast_waitfor(chan, -1)) > -1) && fs && (f = ast_read(chan))) {
-			if (digit) {
-				struct ast_frame fr = {AST_FRAME_DTMF, digit};
-				ast_queue_frame(chan, &fr);
-				digit = 0;
-			}
-			if ((f->frametype == AST_FRAME_DTMF)) {
-				int got = 1;
-				switch(mode) {
-				case DMODE_PLAY:
-					switch(f->subclass) {
-					case '1':
-						ast_set_flag(&flags, DFLAG_PAUSE);
-						mode = DMODE_RECORD;
-						break;
-					case '2':
-						speed++;
-						if (speed > 4) {
-							speed = 1;
-						}
-						res = ast_say_number(chan, speed, AST_DIGIT_ANY, chan->language, NULL);
-						break;
-					case '7':
-						samples -= ffactor;
-						if(samples < 0) {
-							samples = 0;
-						}
-						ast_seekstream(fs, samples, SEEK_SET);
-						break;
-					case '8':
-						samples += ffactor;
-						ast_seekstream(fs, samples, SEEK_SET);
-						break;
-						
-					default:
-						got = 0;
-					}
-					break;
-				case DMODE_RECORD:
-					switch(f->subclass) {
-					case '1':
-						ast_set_flag(&flags, DFLAG_PAUSE);
-						mode = DMODE_PLAY;
-						break;
-					case '8':
-						ast_toggle_flag(&flags, DFLAG_TRUNC);
-						lastop = 0;
-						break;
-					default:
-						got = 0;
-					}
-					break;
-				default:
-					got = 0;
-				}
-				if (!got) {
-					switch(f->subclass) {
-					case '#':
-						done = 1;
-						continue;
-						break;
-					case '*':
-						ast_toggle_flag(&flags, DFLAG_PAUSE);
-						if (ast_test_flag(&flags, DFLAG_PAUSE)) {
-							digit = play_and_wait(chan, "dictate/pause", AST_DIGIT_ANY);
-						} else {
-							digit = play_and_wait(chan, mode == DMODE_PLAY ? "dictate/playback" : "dictate/record", AST_DIGIT_ANY);
-						}
-						break;
-					case '0':
-						ast_set_flag(&flags, DFLAG_PAUSE);
-						digit = play_and_wait(chan, "dictate/paused", AST_DIGIT_ANY);
-						switch(mode) {
-						case DMODE_PLAY:
-							digit = play_and_wait(chan, "dictate/play_help", AST_DIGIT_ANY);
-							break;
-						case DMODE_RECORD:
-							digit = play_and_wait(chan, "dictate/record_help", AST_DIGIT_ANY);
-							break;
-						}
-						if (digit == 0) {
-							digit = play_and_wait(chan, "dictate/both_help", AST_DIGIT_ANY);
-						} else if (digit < 0) {
-							done = 1;
-							break;
-						}
-						break;
-					}
-				}
-				
-			} else if (f->frametype == AST_FRAME_VOICE) {
-				switch(mode) {
-					struct ast_frame *fr;
-					int x;
-				case DMODE_PLAY:
-					if (lastop != DMODE_PLAY) {
-						if (ast_test_flag(&flags, DFLAG_PAUSE)) {
-							digit = play_and_wait(chan, "dictate/playback_mode", AST_DIGIT_ANY);
-							if (digit == 0) {
-								digit = play_and_wait(chan, "dictate/paused", AST_DIGIT_ANY);
-							} else if (digit < 0) {
-								break;
-							}
-						}
-						if (lastop != DFLAG_PLAY) {
-							lastop = DFLAG_PLAY;
-							ast_closestream(fs);
-							if (!(fs = ast_openstream(chan, path, chan->language)))
-								break;
-							ast_seekstream(fs, samples, SEEK_SET);
-							chan->stream = NULL;
-						}
-						lastop = DMODE_PLAY;
-					}
-
-					if (!ast_test_flag(&flags, DFLAG_PAUSE)) {
-						for (x = 0; x < speed; x++) {
-							if ((fr = ast_readframe(fs))) {
-								ast_write(chan, fr);
-								samples += fr->samples;
-								ast_frfree(fr);
-								fr = NULL;
-							} else {
-								samples = 0;
-								ast_seekstream(fs, 0, SEEK_SET);
-							}
-						}
-					}
-					break;
-				case DMODE_RECORD:
-					if (lastop != DMODE_RECORD) {
-						int oflags = O_CREAT | O_WRONLY;
-						if (ast_test_flag(&flags, DFLAG_PAUSE)) {
-							digit = play_and_wait(chan, "dictate/record_mode", AST_DIGIT_ANY);
-							if (digit == 0) {
-								digit = play_and_wait(chan, "dictate/paused", AST_DIGIT_ANY);
-							} else if (digit < 0) {
-								break;
-							}
-						}
-						lastop = DMODE_RECORD;
-						ast_closestream(fs);
-						if ( ast_test_flag(&flags, DFLAG_TRUNC)) {
-							oflags |= O_TRUNC;
-							digit = play_and_wait(chan, "dictate/truncating_audio", AST_DIGIT_ANY);
-						} else {
-							oflags |= O_APPEND;
-						}
-						fs = ast_writefile(path, "raw", NULL, oflags, 0, AST_FILE_MODE);
-						if (ast_test_flag(&flags, DFLAG_TRUNC)) {
-							ast_seekstream(fs, 0, SEEK_SET);
-							ast_clear_flag(&flags, DFLAG_TRUNC);
-						} else {
-							ast_seekstream(fs, 0, SEEK_END);
-						}
-					}
-					if (!ast_test_flag(&flags, DFLAG_PAUSE)) {
-						res = ast_writestream(fs, f);
-					}
-					break;
-				}
-				
-			}
-
-			ast_frfree(f);
-		}
-	}
-	if (oldr) {
-		ast_set_read_format(chan, oldr);
-	}
-	return 0;
-}
-
-static int unload_module(void)
-{
-	int res;
-	res = ast_unregister_application(app);
-	return res;
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app, dictate_exec, synopsis, desc);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Virtual Dictation Machine");

apps/app_directed_pickup.c

@@ -1,172 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2005, Joshua Colp
- *
- * Joshua Colp <jcolp@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Directed Call Pickup Support
- *
- * \author Joshua Colp <jcolp@digium.com>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/lock.h"
-#include "asterisk/app.h"
-#include "asterisk/features.h"
-
-#define PICKUPMARK "PICKUPMARK"
-
-static const char *app = "Pickup";
-static const char *synopsis = "Directed Call Pickup";
-static const char *descrip =
-"  Pickup([extension[@context][&extension2@[context]...]]):  This application can\n"
-"pickup any ringing channel that is calling the specified extension.  If no\n"
-"context is specified, the current context will be used. If you use the special\n"
-"string \"PICKUPMARK\" for the context parameter, for example 10@PICKUPMARK,\n"
-"this application tries to find a channel which has defined a ${PICKUPMARK}\n"
-"channel variable with the same value as \"extension\" (in this example, \"10\").\n"
-"When no parameter is specified, the application will pickup a channel matching\n"
-"the pickup group of the active channel.";
-
-/* Perform actual pickup between two channels */
-static int pickup_do(struct ast_channel *chan, struct ast_channel *target)
-{
-	int res = 0;
-
-	ast_debug(1, "Call pickup on '%s' by '%s'\n", target->name, chan->name);
-
-	if ((res = ast_answer(chan))) {
-		ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan->name);
-		return -1;
-	}
-
-	if ((res = ast_queue_control(chan, AST_CONTROL_ANSWER))) {
-		ast_log(LOG_WARNING, "Unable to queue answer on '%s'\n", chan->name);
-		return -1;
-	}
-
-	if ((res = ast_channel_masquerade(target, chan))) {
-		ast_log(LOG_WARNING, "Unable to masquerade '%s' into '%s'\n", chan->name, target->name);
-		return -1;
-	}
-
-	return res;
-}
-
-/* Helper function that determines whether a channel is capable of being picked up */
-static int can_pickup(struct ast_channel *chan)
-{
-	if (!chan->pbx && (chan->_state == AST_STATE_RINGING || chan->_state == AST_STATE_RING || chan->_state == AST_STATE_DOWN))
-		return 1;
-	else
-		return 0;
-}
-
-/* Attempt to pick up specified extension with context */
-static int pickup_by_exten(struct ast_channel *chan, const char *exten, const char *context)
-{
-	int res = -1;
-	struct ast_channel *target = NULL;
-
-	while ((target = ast_channel_walk_locked(target))) {
-		if ((!strcasecmp(target->macroexten, exten) || !strcasecmp(target->exten, exten)) &&
-			!strcasecmp(target->dialcontext, context) &&
-			(chan != target) && can_pickup(target)) {
-			res = pickup_do(chan, target);
-			ast_channel_unlock(target);
-			break;
-		}
-		ast_channel_unlock(target);
-	}
-
-	return res;
-}
-
-/* Attempt to pick up specified mark */
-static int pickup_by_mark(struct ast_channel *chan, const char *mark)
-{
-	int res = -1;
-	const char *tmp = NULL;
-	struct ast_channel *target = NULL;
-
-	while ((target = ast_channel_walk_locked(target))) {
-		if ((tmp = pbx_builtin_getvar_helper(target, PICKUPMARK)) &&
-			!strcasecmp(tmp, mark) &&
-			can_pickup(target)) {
-			res = pickup_do(chan, target);
-			ast_channel_unlock(target);
-			break;
-		}
-		ast_channel_unlock(target);
-	}
-
-	return res;
-}
-
-/* Main application entry point */
-static int pickup_exec(struct ast_channel *chan, void *data)
-{
-	int res = 0;
-	char *tmp = ast_strdupa(data);
-	char *exten = NULL, *context = NULL;
-
-	if (ast_strlen_zero(data)) {
-		res = ast_pickup_call(chan);
-		return res;
-	}
-	
-	/* Parse extension (and context if there) */
-	while (!ast_strlen_zero(tmp) && (exten = strsep(&tmp, "&"))) {
-		if ((context = strchr(exten, '@')))
-			*context++ = '\0';
-		if (!ast_strlen_zero(context) && !strcasecmp(context, PICKUPMARK)) {
-			if (!pickup_by_mark(chan, exten))
-				break;
-		} else {
-			if (!pickup_by_exten(chan, exten, !ast_strlen_zero(context) ? context : chan->context))
-				break;
-		}
-		ast_log(LOG_NOTICE, "No target channel found for %s.\n", exten);
-	}
-
-	return res;
-}
-
-static int unload_module(void)
-{
-	int res;
-
-	res = ast_unregister_application(app);
-
-	return res;
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app, pickup_exec, synopsis, descrip);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Directed Call Pickup Application");

apps/app_disa.c

@@ -1,112 +1,101 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * DISA -- Direct Inward System Access Application  6/20/2001
+ * 
+ * Copyright (C) 2001, Linux Support Services, Inc.
  *
+ * Jim Dixon <jim@lambdatel.com>
  *
  * Made only slightly more sane by Mark Spencer <markster@digium.com>
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
-
-/*! \file
- *
- * \brief DISA -- Direct Inward System Access Application
- *
- * \author Jim Dixon <jim@lambdatel.com>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
+ 
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/indications.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/ulaw.h>
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
 #include <math.h>
 #include <sys/time.h>
 
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/app.h"
-#include "asterisk/indications.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/translate.h"
-#include "asterisk/ulaw.h"
-#include "asterisk/callerid.h"
-#include "asterisk/stringfields.h"
+
+static char *tdesc = "DISA (Direct Inward System Access) Application";
 
 static char *app = "DISA";
 
 static char *synopsis = "DISA (Direct Inward System Access)";
 
-static char *descrip =
-"DISA(<numeric passcode>[,<context>[,<cid>[,mailbox[,options]]]]) or\n"
-"DISA(<filename>[,,,,options])\n"
-"The DISA, Direct Inward System Access, application allows someone from \n"
-"outside the telephone switch (PBX) to obtain an \"internal\" system \n"
-"dialtone and to place calls from it as if they were placing a call from \n"
-"within the switch.\n"
-"DISA plays a dialtone. The user enters their numeric passcode, followed by\n"
-"the pound sign (#). If the passcode is correct, the user is then given\n"
-"system dialtone within <context> on which a call may be placed. If the user\n"
-"enters an invalid extension and extension \"i\" exists in the specified\n"
-"context, it will be used.\n"
-"\n"
-"If you need to present a DISA dialtone without entering a password, simply\n"
-"set <passcode> to \"no-password\".\n"
-"\n"
-"Be aware that using this may compromise the security of your PBX.\n"
-"\n"
-"The arguments to this application (in extensions.conf) allow either\n"
-"specification of a single global passcode (that everyone uses), or\n"
-"individual passcodes contained in a file.\n"
-"\n"
-"The file that contains the passcodes (if used) allows a complete\n"
-"specification of all of the same arguments available on the command\n"
-"line, with the sole exception of the options. The file may contain blank\n"
-"lines, or comments starting with \"#\" or \";\".\n"
-"\n"
-"<context> specifies the dialplan context in which the user-entered extension\n"
-"will be matched. If no context is specified, the DISA application defaults\n"
-"the context to \"disa\". Presumably a normal system will have a special\n"
-"context set up for DISA use with some or a lot of restrictions.\n"
-"\n"
-"<cid> specifies a new (different) callerid to be used for this call.\n"
-"\n"
-"<mailbox[@context]> will cause a stutter-dialtone (indication \"dialrecall\")\n"
-"to be used, if the specified mailbox contains any new messages.\n"
-"\n"
-"The following options are available:\n"
-"  n - the DISA application will not answer initially.\n"
-"  p - the extension entered will be considered complete when a '#' is entered.\n";
+static char *descrip = 
+	"DISA (Direct Inward System Access) -- Allows someone from outside\n"
+	"the telephone switch (PBX) to obtain an \"internal\" system dialtone\n"
+	"and to place calls from it as if they were placing a call from within\n"
+	"the switch. A user calls a number that connects to the DISA application\n"
+	"and is given dialtone. The user enters their passcode, followed by the\n"
+	"pound sign (#). If the passcode is correct, the user is then given\n"
+	"system dialtone on which a call may be placed. Obviously, this type\n"
+	"of access has SERIOUS security implications, and GREAT care must be\n"
+	"taken NOT to compromise your security.\n\n"
+	"There is a possibility of accessing DISA without password. Simply\n"
+	"exchange your password with no-password.\n\n"
+	"  Example: exten => s,1,DISA,no-password|local\n\n"
+	"but be aware of using this for your security compromising.\n\n"
+	"The arguments to this application (in extensions.conf) allow either\n"
+	"specification of a single global password (that everyone uses), or\n"
+	"individual passwords contained in a file. It also allow specification\n"
+	"of the context on which the user will be dialing. If no context is\n"
+	"specified, the DISA application defaults the context to \"disa\"\n"
+	"presumably that a normal system will have a special context set up\n"
+	"for DISA use with some or a lot of restrictions. The arguments are\n"
+	"one of the following:\n\n"
+	"    numeric-passcode\n"
+	"    numeric-passcode|context\n"
+	"    full-pathname-of-file-that-contains-passcodes\n\n"
+	"The file that contains the passcodes (if used) allows specification\n"
+	"of either just a passcode (defaulting to the \"disa\" context, or\n"
+	"passcode|context on each line of the file. The file may contain blank\n"
+	"lines, or comments starting with \"#\" or \";\". In addition, the\n"
+	"above arguments may have |new-callerid-string appended to them, to\n"
+	"specify a new (different) callerid to be used for this call, for\n"
+	"example: numeric-passcode|context|\"My Phone\" <(234) 123-4567> or \n"
+	"full-pathname-of-passcode-file|\"My Phone\" <(234) 123-4567>. Note that\n"
+	"in the case of specifying the numeric-passcode, the context must be\n"
+	"specified if the callerid is specified also.\n\n"
+	"If login is successful, the application parses the dialed number in\n"
+	"the specified (or default) context, and returns 0 with the new extension\n"
+	"context filled-in and the priority set to 1, so that the PBX may\n"
+	"re-apply the routing tables to it and complete the call normally.";
 
-enum {
-	NOANSWER_FLAG = (1 << 0),
-	POUND_TO_END_FLAG = (1 << 1),
-} option_flags;
 
-AST_APP_OPTIONS(app_opts, {
-	AST_APP_OPTION('n', NOANSWER_FLAG),
-	AST_APP_OPTION('p', POUND_TO_END_FLAG),
-});
+STANDARD_LOCAL_USER;
 
-static void play_dialtone(struct ast_channel *chan, char *mailbox)
+LOCAL_USER_DECL;
+
+static int firstdigittimeout = 20000; /* 20 seconds first digit timeout */
+static int digittimeout = 10000; /* 10 seconds subsequent digit timeout */
+
+static int ms_diff(struct timeval *tv1, struct timeval *tv2)
+{
+int	ms;
+	
+	ms = (tv1->tv_sec - tv2->tv_sec) * 1000;
+	ms += (tv1->tv_usec - tv2->tv_usec) / 1000;
+	return(ms);
+}
+
+static void play_dialtone(struct ast_channel *chan)
 {
 	const struct tone_zone_sound *ts = NULL;
-	if(ast_app_has_voicemail(mailbox, NULL))
-		ts = ast_get_indication_tone(chan->zone, "dialrecall");
-	else
-		ts = ast_get_indication_tone(chan->zone, "dial");
+	ts = ast_get_indication_tone(chan->zone, "dial");
 	if (ts)
 		ast_playtones_start(chan, 0, ts->data, 0);
 	else
@@ -115,194 +104,187 @@ static void play_dialtone(struct ast_channel *chan, char *mailbox)
 
 static int disa_exec(struct ast_channel *chan, void *data)
 {
-	int i = 0, j, k = 0, did_ignore = 0, special_noanswer = 0;
-	int firstdigittimeout = (chan->pbx ? chan->pbx->rtimeout * 1000 : 20000);
-	int digittimeout = (chan->pbx ? chan->pbx->dtimeout * 1000 : 10000);
-	struct ast_flags flags;
-	char *tmp, exten[AST_MAX_EXTENSION] = "", acctcode[20]="";
-	char pwline[256];
-	char ourcidname[256],ourcidnum[256];
+	int i,j,k,x,did_ignore;
+	struct localuser *u;
+	char tmp[256],arg2[256]="",exten[AST_MAX_EXTENSION],acctcode[20]="";
+	char *ourcontext,*ourcallerid;
 	struct ast_frame *f;
-	struct timeval lastdigittime;
+	struct timeval lastout, now, lastdigittime;
 	int res;
+	time_t rstart;
 	FILE *fp;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(passcode);
-		AST_APP_ARG(context);
-		AST_APP_ARG(cid);
-		AST_APP_ARG(mailbox);
-		AST_APP_ARG(options);
-	);
+	char *stringp=NULL;
 
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "DISA requires an argument (passcode/passcode file)\n");
+	if (ast_set_write_format(chan,AST_FORMAT_ULAW))
+	{
+		ast_log(LOG_WARNING, "Unable to set write format to Mu-law on %s\n",chan->name);
 		return -1;
 	}
-
-	ast_debug(1, "Digittimeout: %d\n", digittimeout);
-	ast_debug(1, "Responsetimeout: %d\n", firstdigittimeout);
-
-	tmp = ast_strdupa(data);
-
-	AST_STANDARD_APP_ARGS(args, tmp);
-
-	if (ast_strlen_zero(args.context))
-		args.context = "disa";
-	if (ast_strlen_zero(args.mailbox))
-		args.mailbox = "";
-	if (!ast_strlen_zero(args.options))
-		ast_app_parse_options(app_opts, &flags, NULL, args.options);
-
-	ast_debug(1, "Mailbox: %s\n",args.mailbox);
-
-	if (!ast_test_flag(&flags, NOANSWER_FLAG)) {
-		if (chan->_state != AST_STATE_UP) {
-			/* answer */
-			ast_answer(chan);
-		}
-	} else special_noanswer = 1;
-
-	ast_debug(1, "Context: %s\n",args.context);
-
-	if (!strcasecmp(args.passcode, "no-password")) {
-		k |= 1; /* We have the password */
-		ast_debug(1, "DISA no-password login success\n");
+	if (ast_set_read_format(chan,AST_FORMAT_ULAW))
+	{
+		ast_log(LOG_WARNING, "Unable to set read format to Mu-law on %s\n",chan->name);
+		return -1;
 	}
+	lastout.tv_sec = lastout.tv_usec = 0;
+	if (!data || !strlen((char *)data)) {
+		ast_log(LOG_WARNING, "disa requires an argument (passcode/passcode file)\n");
+		return -1;
+	}
+	strncpy(tmp, (char *)data, sizeof(tmp)-1);
+	stringp=tmp;
+	strsep(&stringp, "|");
+	ourcontext = strsep(&stringp, "|");
+	/* if context specified, save 2nd arg and parse third */
+	if (ourcontext) {
+		strncpy(arg2,ourcontext, sizeof(arg2) - 1);
+		ourcallerid = strsep(&stringp,"|");
+	}
+	  /* if context not specified, use "disa" */
+	else {
+		arg2[0] = 0;
+		ourcallerid = NULL;
+		ourcontext = "disa";
+	}
+	LOCAL_USER_ADD(u);
+	if (chan->_state != AST_STATE_UP)
+	{
+		/* answer */
+		ast_answer(chan);
+	}
+	i = k = x = 0; /* k is 0 for pswd entry, 1 for ext entry */
+	did_ignore = 0;
+	exten[0] = 0;
+	acctcode[0] = 0;
+	/* can we access DISA without password? */ 
 
-	lastdigittime = ast_tvnow();
+	ast_log(LOG_DEBUG, "Context: %s\n",ourcontext);
 
-	play_dialtone(chan, args.mailbox);
+	if (!strcasecmp(tmp, "no-password"))
+	{;
+		k = 1;
+		ast_log(LOG_DEBUG, "DISA no-password login success\n");
+	}
+	gettimeofday(&lastdigittime,NULL);
 
-	ast_set_flag(chan, AST_FLAG_END_DTMF_ONLY);
+	play_dialtone(chan);
 
-	for (;;) {
+	for(;;)
+	{
+		gettimeofday(&now,NULL);
 		  /* if outa time, give em reorder */
-		if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) > ((k&2) ? digittimeout : firstdigittimeout)) {
-			ast_debug(1,"DISA %s entry timeout on chan %s\n",
-				((k&1) ? "extension" : "password"),chan->name);
+		if (ms_diff(&now,&lastdigittime) > 
+		    ((k) ? digittimeout : firstdigittimeout))
+		{
+			ast_log(LOG_DEBUG,"DISA %s entry timeout on chan %s\n",
+				((k) ? "extension" : "password"),chan->name);
 			break;
 		}
-
 		if ((res = ast_waitfor(chan, -1) < 0)) {
-			ast_debug(1, "Waitfor returned %d\n", res);
+			ast_log(LOG_DEBUG, "Waitfor returned %d\n", res);
 			continue;
 		}
-
-		if (!(f = ast_read(chan))) {
-			ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
+			
+		f = ast_read(chan);
+		if (f == NULL) 
+		{
+			LOCAL_USER_REMOVE(u);
 			return -1;
 		}
-
-		if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP)) {
+		if ((f->frametype == AST_FRAME_CONTROL) &&
+		    (f->subclass == AST_CONTROL_HANGUP))
+		{
 			ast_frfree(f);
-			ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
+			LOCAL_USER_REMOVE(u);
 			return -1;
 		}
-
-		/* If the frame coming in is not DTMF, just drop it and continue */
-		if (f->frametype != AST_FRAME_DTMF) {
+		if (f->frametype == AST_FRAME_VOICE) {
+			ast_frfree(f);
+			continue;
+		}
+		  /* if not DTMF, just do it again */
+		if (f->frametype != AST_FRAME_DTMF) 
+		{
 			ast_frfree(f);
 			continue;
 		}
 
 		j = f->subclass;  /* save digit */
 		ast_frfree(f);
-
-		if (!i) {
-			k |= 2; /* We have the first digit */
+		if (i == 0) 
 			ast_playtones_stop(chan);
-		}
 
-		lastdigittime = ast_tvnow();
-
-		/* got a DTMF tone */
-		if (i < AST_MAX_EXTENSION) { /* if still valid number of digits */
-			if (!(k&1)) { /* if in password state */
-				if (j == '#') { /* end of password */
+		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 (j == '#') /* end of password */
+				{
 					  /* see if this is an integer */
-					if (sscanf(args.passcode,"%30d",&j) < 1) { /* nope, it must be a filename */
-						fp = fopen(args.passcode,"r");
-						if (!fp) {
-							ast_log(LOG_WARNING,"DISA password file %s not found on chan %s\n",args.passcode,chan->name);
-							ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
+					if (sscanf(tmp,"%d",&j) < 1)
+					   { /* nope, it must be a filename */
+						fp = fopen(tmp,"r");
+						if (!fp)
+						   {
+							ast_log(LOG_WARNING,"DISA password file %s not found on chan %s\n",tmp,chan->name);
+							LOCAL_USER_REMOVE(u);
 							return -1;
-						}
-						pwline[0] = 0;
-						while(fgets(pwline,sizeof(pwline) - 1,fp)) {
-							if (!pwline[0])
-								continue;
-							if (pwline[strlen(pwline) - 1] == '\n')
-								pwline[strlen(pwline) - 1] = 0;
-							if (!pwline[0])
-								continue;
-							 /* skip comments */
-							if (pwline[0] == '#')
-								continue;
-							if (pwline[0] == ';')
-								continue;
-
-							AST_STANDARD_APP_ARGS(args, pwline);
-
-							ast_debug(1, "Mailbox: %s\n",args.mailbox);
-
-							/* password must be in valid format (numeric) */
-							if (sscanf(args.passcode,"%30d", &j) < 1)
-								continue;
-							 /* if we got it */
-							if (!strcmp(exten,args.passcode)) {
-								if (ast_strlen_zero(args.context))
-									args.context = "disa";
-								if (ast_strlen_zero(args.mailbox))
-									args.mailbox = "";
-								break;
+						   }
+						tmp[0] = 0;
+						while(fgets(tmp,sizeof(tmp) - 1,fp))
+						   {
+							char *stringp=NULL,*stringp2;
+							if (!tmp[0]) continue;
+							if (tmp[strlen(tmp) - 1] == '\n') 
+								tmp[strlen(tmp) - 1] = 0;
+							if (!tmp[0]) continue;
+							  /* skip comments */
+							if (tmp[0] == '#') continue;
+							if (tmp[0] == ';') continue;
+							stringp=tmp;
+							strsep(&stringp, "|");
+							stringp2=strsep(&stringp, "|");
+							if (stringp2) {
+								ourcontext=stringp2;
+								stringp2=strsep(&stringp, "|");
+								if (stringp2) ourcallerid=stringp2;
 							}
-						}
+							  /* password must be in valid format (numeric) */
+							if (sscanf(tmp,"%d",&j) < 1) continue;
+							  /* if we got it */
+							if (!strcmp(exten,tmp)) break;
+						   }
 						fclose(fp);
-					}
-					/* compare the two */
-					if (strcmp(exten,args.passcode)) {
+					   }
+					  /* compare the two */
+					if (strcmp(exten,tmp))
+					{
 						ast_log(LOG_WARNING,"DISA on chan %s got bad password %s\n",chan->name,exten);
 						goto reorder;
 
 					}
 					 /* password good, set to dial state */
-					ast_debug(1,"DISA on chan %s password is good\n",chan->name);
-					play_dialtone(chan, args.mailbox);
+					ast_log(LOG_DEBUG,"DISA on chan %s password is good\n",chan->name);
+					play_dialtone(chan);
 
-					k|=1; /* In number mode */
+					k = 1;
 					i = 0;  /* re-set buffer pointer */
 					exten[sizeof(acctcode)] = 0;
-					ast_copy_string(acctcode, exten, sizeof(acctcode));
+					strncpy(acctcode,exten, sizeof(acctcode) - 1);
 					exten[0] = 0;
-					ast_debug(1,"Successful DISA log-in on chan %s\n", chan->name);
+					ast_log(LOG_DEBUG,"Successful DISA log-in on chan %s\n",chan->name);
 					continue;
 				}
-			} else {
-				if (j == '#') { /* end of extension .. maybe */
-					if (i == 0 && 
-							(ast_matchmore_extension(chan, args.context, "#", 1, chan->cid.cid_num) ||
-							 ast_exists_extension(chan, args.context, "#", 1, chan->cid.cid_num)) ) {
-						/* Let the # be the part of, or the entire extension */
-					} else {
-						break;
-					}
-				}
 			}
 
 			exten[i++] = j;  /* save digit */
 			exten[i] = 0;
-			if (!(k&1))
-				continue; /* if getting password, continue doing it */
-			/* if this exists */
+			if (!k) continue; /* if getting password, continue doing it */
+			  /* if this exists */
 
-			/* user wants end of number, remove # */
-			if (ast_test_flag(&flags, POUND_TO_END_FLAG) && j == '#') {
-				exten[--i] = 0;
-				break;
-			}
-
-			if (ast_ignore_pattern(args.context, exten)) {
-				play_dialtone(chan, "");
+			if (ast_ignore_pattern(ourcontext, exten)) {
+				play_dialtone(chan);
 				did_ignore = 1;
 			} else
 				if (did_ignore) {
@@ -310,63 +292,74 @@ static int disa_exec(struct ast_channel *chan, void *data)
 					did_ignore = 0;
 				}
 
-			/* if can do some more, do it */
-			if (!ast_matchmore_extension(chan,args.context,exten,1, chan->cid.cid_num)) {
+			  /* if can do some more, do it */
+			if (!ast_matchmore_extension(chan,ourcontext,exten,1, chan->callerid)) {
 				break;
 			}
 		}
 	}
 
-	ast_clear_flag(chan, AST_FLAG_END_DTMF_ONLY);
-
-	if (k == 3) {
-		int recheck = 0;
-		struct ast_flags flags = { AST_CDR_FLAG_POSTED };
-
-		if (!ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
-			pbx_builtin_setvar_helper(chan, "INVALID_EXTEN", exten);
-			exten[0] = 'i';
-			exten[1] = '\0';
-			recheck = 1;
-		}
-		if (!recheck || ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
-			ast_playtones_stop(chan);
-			/* We're authenticated and have a target extension */
-			if (!ast_strlen_zero(args.cid)) {
-				ast_callerid_split(args.cid, ourcidname, sizeof(ourcidname), ourcidnum, sizeof(ourcidnum));
-				ast_set_callerid(chan, ourcidnum, ourcidname, ourcidnum);
-			}
-
-			if (!ast_strlen_zero(acctcode))
-				ast_string_field_set(chan, accountcode, acctcode);
-
-			if (special_noanswer) flags.flags = 0;
-			ast_cdr_reset(chan->cdr, &flags);
-			ast_explicit_goto(chan, args.context, exten, 1);
-			return 0;
+	if (k && 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);
+		strncpy(chan->accountcode, acctcode, sizeof(chan->accountcode) - 1);
+		chan->priority = 0;
+		ast_cdr_init(chan->cdr,chan);
+		LOCAL_USER_REMOVE(u);
+		return 0;
 	}
 
-	/* Received invalid, but no "i" extension exists in the given context */
-
 reorder:
-	/* Play congestion for a bit */
-	ast_indicate(chan, AST_CONTROL_CONGESTION);
-	ast_safe_sleep(chan, 10*1000);
 
+	ast_indicate(chan,AST_CONTROL_CONGESTION);
+	/* something is invalid, give em reorder for several seconds */
+	time(&rstart);
+	while(time(NULL) < rstart + 10)
+	{
+		if (ast_waitfor(chan, -1) < 0)
+			break;
+		f = ast_read(chan);
+		if (!f)
+			break;
+		ast_frfree(f);
+	}
 	ast_playtones_stop(chan);
-
+	LOCAL_USER_REMOVE(u);
 	return -1;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, disa_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "DISA (Direct Inward System Access) Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key(void)
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_dumpchan.c

@@ -1,160 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2004 - 2005, Anthony Minessale II.
- *
- * Anthony Minessale <anthmct@yahoo.com>
- *
- * A license has been granted to Digium (via disclaimer) for the use of
- * this code.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Application to dump channel variables
- *
- * \author Anthony Minessale <anthmct@yahoo.com>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/channel.h"
-
-static char *app = "DumpChan";
-static char *synopsis = "Dump Info About The Calling Channel";
-static char *desc =
-	"   DumpChan([<min_verbose_level>])\n"
-	"Displays information on channel and listing of all channel\n"
-	"variables. If min_verbose_level is specified, output is only\n"
-	"displayed when the verbose level is currently set to that number\n"
-	"or greater. \n";
-
-
-static int serialize_showchan(struct ast_channel *c, char *buf, size_t size)
-{
-	struct timeval now;
-	long elapsed_seconds = 0;
-	int hour = 0, min = 0, sec = 0;
-	char cgrp[BUFSIZ/2];
-	char pgrp[BUFSIZ/2];
-	char formatbuf[BUFSIZ/2];
-
-	now = ast_tvnow();
-	memset(buf, 0, size);
-	if (!c)
-		return 0;
-
-	if (c->cdr) {
-		elapsed_seconds = now.tv_sec - c->cdr->start.tv_sec;
-		hour = elapsed_seconds / 3600;
-		min = (elapsed_seconds % 3600) / 60;
-		sec = elapsed_seconds % 60;
-	}
-
-	snprintf(buf,size,
-			"Name=               %s\n"
-			"Type=               %s\n"
-			"UniqueID=           %s\n"
-			"CallerIDNum=        %s\n"
-			"CallerIDName=       %s\n"
-			"DNIDDigits=         %s\n"
-			"RDNIS=              %s\n"
-			"Language=           %s\n"
-			"State=              %s (%d)\n"
-			"Rings=              %d\n"
-			"NativeFormat=       %s\n"
-			"WriteFormat=        %s\n"
-			"ReadFormat=         %s\n"
-			"RawWriteFormat=     %s\n"
-			"RawReadFormat=      %s\n"
-			"1stFileDescriptor=  %d\n"
-			"Framesin=           %d %s\n"
-			"Framesout=          %d %s\n"
-			"TimetoHangup=       %ld\n"
-			"ElapsedTime=        %dh%dm%ds\n"
-			"Context=            %s\n"
-			"Extension=          %s\n"
-			"Priority=           %d\n"
-			"CallGroup=          %s\n"
-			"PickupGroup=        %s\n"
-			"Application=        %s\n"
-			"Data=               %s\n"
-			"Blocking_in=        %s\n",
-			c->name,
-			c->tech->type,
-			c->uniqueid,
-			S_OR(c->cid.cid_num, "(N/A)"),
-			S_OR(c->cid.cid_name, "(N/A)"),
-			S_OR(c->cid.cid_dnid, "(N/A)"),
-			S_OR(c->cid.cid_rdnis, "(N/A)"),
-			c->language,
-			ast_state2str(c->_state),
-			c->_state,
-			c->rings,
-			ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->nativeformats),
-			ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->writeformat),
-			ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->readformat),
-			ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->rawwriteformat),
-			ast_getformatname_multiple(formatbuf, sizeof(formatbuf), c->rawreadformat),
-			c->fds[0], c->fin & ~DEBUGCHAN_FLAG, (c->fin & DEBUGCHAN_FLAG) ? " (DEBUGGED)" : "",
-			c->fout & ~DEBUGCHAN_FLAG, (c->fout & DEBUGCHAN_FLAG) ? " (DEBUGGED)" : "", (long)c->whentohangup,
-			hour,
-			min,
-			sec,
-			c->context,
-			c->exten,
-			c->priority,
-			ast_print_group(cgrp, sizeof(cgrp), c->callgroup),
-			ast_print_group(pgrp, sizeof(pgrp), c->pickupgroup),
-			( c->appl ? c->appl : "(N/A)" ),
-			( c-> data ? S_OR(c->data, "(Empty)") : "(None)"),
-			(ast_test_flag(c, AST_FLAG_BLOCKING) ? c->blockproc : "(Not Blocking)"));
-
-	return 0;
-}
-
-static int dumpchan_exec(struct ast_channel *chan, void *data)
-{
-	struct ast_str *vars = ast_str_alloca(BUFSIZ * 4); /* XXX very large! */
-	char info[1024];
-	int level = 0;
-	static char *line = "================================================================================";
-
-	if (!ast_strlen_zero(data))
-		level = atoi(data);
-
-	pbx_builtin_serialize_variables(chan, &vars);
-	serialize_showchan(chan, info, sizeof(info));
-	if (option_verbose >= level)
-		ast_verbose("\nDumping Info For Channel: %s:\n%s\nInfo:\n%s\nVariables:\n%s%s\n", chan->name, line, info, vars->str, line);
-
-	return 0;
-}
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app, dumpchan_exec, synopsis, desc);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dump Info About The Calling Channel");

apps/app_echo.c

@@ -1,87 +1,100 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Echo application -- play back what you hear to evaluate latency
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
 
-/*! \file
- *
- * \brief Echo application -- play back what you hear to evaluate latency
- *
- * \author Mark Spencer <markster@digium.com>
- *
- * \ingroup applications
- */
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
 
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/module.h"
-#include "asterisk/channel.h"
+static char *tdesc = "Simple Echo Application";
 
 static char *app = "Echo";
 
-static char *synopsis = "Echo audio, video, or DTMF back to the calling party";
+static char *synopsis = "Echo audio read back to the user";
 
-static char *descrip =
-"  Echo(): This application will echo any audio, video, or DTMF frames read from\n"
-"the calling channel back to itself. If the DTMF digit '#' is received, the\n"
-"application will exit.\n";
+static char *descrip = 
+"  Echo():  Echo audio read from channel back to the channel. Returns 0\n"
+"if the user exits with the '#' key, or -1 if the user hangs up.\n";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int echo_exec(struct ast_channel *chan, void *data)
 {
-	int res = -1;
-	int format;
-
-	format = ast_best_codec(chan->nativeformats);
-	ast_set_write_format(chan, format);
-	ast_set_read_format(chan, format);
-
-	while (ast_waitfor(chan, -1) > -1) {
-		struct ast_frame *f = ast_read(chan);
-		if (!f) {
+	int res=-1;
+	struct localuser *u;
+	struct ast_frame *f;
+	LOCAL_USER_ADD(u);
+	ast_set_write_format(chan, ast_best_codec(chan->nativeformats));
+	ast_set_read_format(chan, ast_best_codec(chan->nativeformats));
+	/* Do our thing here */
+	while(ast_waitfor(chan, -1) > -1) {
+		f = ast_read(chan);
+		if (!f)
 			break;
-		}
 		f->delivery.tv_sec = 0;
 		f->delivery.tv_usec = 0;
-		if (ast_write(chan, f)) {
-			ast_frfree(f);
-			goto end;
-		}
-		if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '#')) {
-			res = 0;
-			ast_frfree(f);
-			goto end;
+		if (f->frametype == AST_FRAME_VOICE) {
+			if (ast_write(chan, f)) 
+				break;
+		} else if (f->frametype == AST_FRAME_VIDEO) {
+			if (ast_write(chan, f)) 
+				break;
+		} else if (f->frametype == AST_FRAME_DTMF) {
+			if (f->subclass == '#') {
+				res = 0;
+				break;
+			} else
+				if (ast_write(chan, f))
+					break;
 		}
 		ast_frfree(f);
 	}
-end:
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, echo_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Simple Echo Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_enumlookup.c

@@ -0,0 +1,203 @@
+/*
+ * 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 = "Eval(newvar=somestring)";
+
+static char *eval_descrip =
+"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

@@ -1,246 +1,113 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (c) 2004 - 2005, Tilghman Lesher.  All rights reserved.
- * Portions copyright (c) 2006, Philipp Dunkel.
+ * Exec application
  *
- * Tilghman Lesher <app_exec__v002@the-tilghman.com>
+ * 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.
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
  */
 
-/*! \file
- *
- * \brief Exec application
- *
- * \author Tilghman Lesher <app_exec__v002@the-tilghman.com>
- * \author Philipp Dunkel <philipp.dunkel@ebox.at>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/app.h"
+#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
+#define MAXRESULT	1024
 
-/*! Note
- *
- * The key difference between these two apps is exit status.  In a
- * nutshell, Exec tries to be transparent as possible, behaving
- * in exactly the same way as if the application it calls was
- * directly invoked from the dialplan.
- *
- * TryExec, on the other hand, provides a way to execute applications
- * and catch any possible fatal error without actually fatally
- * affecting the dialplan.
- */
+static char *tdesc = "Executes applications";
 
 static char *app_exec = "Exec";
-static char *exec_synopsis = "Executes dialplan application";
+
+static char *exec_synopsis = "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.  If the underlying application\n"
-"terminates the dialplan, or if the application cannot be found,\n"
-"Exec will terminate the dialplan.\n"
-"  To invoke external applications, see the application System.\n"
-"  If you would like to catch any error instead, see TryExec.\n";
+"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";
 
-static char *app_tryexec = "TryExec";
-static char *tryexec_synopsis = "Executes dialplan application, always returning";
-static char *tryexec_descrip =
-"  TryExec(appname(arguments)):\n"
-"Allows an arbitrary application to be invoked even when not\n"
-"hardcoded into the dialplan. To invoke external applications\n"
-"see the application System.  Always returns to the dialplan.\n"
-"The channel variable TRYSTATUS will be set to one of:\n"
-"    SUCCESS   if the application returned zero\n"
-"    FAILED    if the application returned non-zero\n"
-"    NOAPP     if the application was not found or was not specified\n";
+STANDARD_LOCAL_USER;
 
-static char *app_execif = "ExecIf";
-static char *execif_synopsis = "Executes dialplan application, conditionally";
-static char *execif_descrip =
-"  ExecIF (<expr>?<appiftrue>(<args>)[:<appiffalse>(<args>)])\n"
-"If <expr> is true, execute and return the result of <appiftrue>(<args>).\n"
-"If <expr> is true, but <appiftrue> is not found, then the application\n"
-"will return a non-zero value.\n";
+LOCAL_USER_DECL;
 
 static int exec_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
+	int res=0;
+	struct localuser *u;
 	char *s, *appname, *endargs, args[MAXRESULT];
 	struct ast_app *app;
 
-	if (ast_strlen_zero(data))
-		return 0;
+	LOCAL_USER_ADD(u);
 
-	s = ast_strdupa(data);
-	args[0] = 0;
-	appname = strsep(&s, "(");
-	if (s) {
-		endargs = strrchr(s, ')');
-		if (endargs)
-			*endargs = '\0';
-		pbx_substitute_variables_helper(chan, s, args, MAXRESULT - 1);
-	}
-	if (appname) {
-		app = pbx_findapp(appname);
-		if (app) {
-			res = pbx_exec(chan, app, args);
+	memset(args, 0, MAXRESULT);
+
+	/* Check and parse arguments */
+	if (data) {
+		s = ast_strdupa((char *)data);
+		if (s) {
+			appname = strsep(&s, "(");
+			if (s) {
+				endargs = strrchr(s, ')');
+				if (endargs)
+					*endargs = '\0';
+				pbx_substitute_variables_helper(chan, s, args, MAXRESULT - 1);
+			}
+			if (appname) {
+				app = pbx_findapp(appname);
+				if (app) {
+					res = pbx_exec(chan, app, args, 1);
+				} else {
+					ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
+					res = -2;
+				}
+			}
 		} else {
-			ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
+			ast_log(LOG_ERROR, "Out of memory\n");
 			res = -1;
 		}
 	}
 
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int tryexec_exec(struct ast_channel *chan, void *data)
+int unload_module(void)
 {
-	int res = 0;
-	char *s, *appname, *endargs, args[MAXRESULT];
-	struct ast_app *app;
-
-	if (ast_strlen_zero(data))
-		return 0;
-
-	s = ast_strdupa(data);
-	args[0] = 0;
-	appname = strsep(&s, "(");
-	if (s) {
-		endargs = strrchr(s, ')');
-		if (endargs)
-			*endargs = '\0';
-		pbx_substitute_variables_helper(chan, s, args, MAXRESULT - 1);
-	}
-	if (appname) {
-		app = pbx_findapp(appname);
-		if (app) {
-			res = pbx_exec(chan, app, args);
-			pbx_builtin_setvar_helper(chan, "TRYSTATUS", res ? "FAILED" : "SUCCESS");
-		} else {
-			ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
-			pbx_builtin_setvar_helper(chan, "TRYSTATUS", "NOAPP");
-		}
-	}
-
-	return 0;
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app_exec);
 }
 
-static int execif_exec(struct ast_channel *chan, void *data)
+int load_module(void)
 {
-	int res = 0;
-	char *truedata = NULL, *falsedata = NULL, *end, *firstcomma, *firstquestion;
-	struct ast_app *app = NULL;
-	AST_DECLARE_APP_ARGS(expr,
-		AST_APP_ARG(expr);
-		AST_APP_ARG(remainder);
-	);
-	AST_DECLARE_APP_ARGS(apps,
-		AST_APP_ARG(t);
-		AST_APP_ARG(f);
-	);
-	char *parse = ast_strdupa(data);
-
-	firstcomma = strchr(parse, ',');
-	firstquestion = strchr(parse, '?');
-
-	if ((firstcomma != NULL && firstquestion != NULL && firstcomma < firstquestion) || (firstquestion == NULL)) {
-		/* Deprecated syntax */
-		AST_DECLARE_APP_ARGS(depr,
-			AST_APP_ARG(expr);
-			AST_APP_ARG(appname);
-			AST_APP_ARG(appargs);
-		);
-		AST_STANDARD_APP_ARGS(depr, parse);
-
-		ast_log(LOG_WARNING, "Deprecated syntax found.  Please upgrade to using ExecIf(<expr>?%s(%s))\n", depr.appname, depr.appargs);
-
-		/* Make the two syntaxes look the same */
-		expr.expr = depr.expr;
-		apps.t = depr.appname;
-		apps.f = NULL;
-		truedata = depr.appargs;
-	} else {
-		/* Preferred syntax */
-
-		AST_NONSTANDARD_RAW_ARGS(expr, parse, '?');
-		if (ast_strlen_zero(expr.remainder)) {
-			ast_log(LOG_ERROR, "Usage: ExecIf(<expr>?<appiftrue>(<args>)[:<appiffalse>(<args)])\n");
-			return -1;
-		}
-
-		AST_NONSTANDARD_RAW_ARGS(apps, expr.remainder, ':');
-
-		if (apps.t && (truedata = strchr(apps.t, '('))) {
-			*truedata++ = '\0';
-			if ((end = strrchr(truedata, ')'))) {
-				*end = '\0';
-			}
-		}
-
-		if (apps.f && (falsedata = strchr(apps.f, '('))) {
-			*falsedata++ = '\0';
-			if ((end = strrchr(falsedata, ')'))) {
-				*end = '\0';
-			}
-		}
-	}
-
-	if (pbx_checkcondition(expr.expr)) {
-		if (!ast_strlen_zero(apps.t) && (app = pbx_findapp(apps.t))) {
-			res = pbx_exec(chan, app, S_OR(truedata, ""));
-		} else {
-			ast_log(LOG_WARNING, "Could not find application! (%s)\n", apps.t);
-			res = -1;
-		}
-	} else if (!ast_strlen_zero(apps.f)) {
-		if ((app = pbx_findapp(apps.f))) {
-			res = pbx_exec(chan, app, S_OR(falsedata, ""));
-		} else {
-			ast_log(LOG_WARNING, "Could not find application! (%s)\n", apps.f);
-			res = -1;
-		}
-	}
-
-	return res;
+	return ast_register_application(app_exec, exec_exec, exec_synopsis, exec_descrip);
 }
 
-static int unload_module(void)
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
 {
 	int res;
-
-	res = ast_unregister_application(app_exec);
-	res |= ast_unregister_application(app_tryexec);
-	res |= ast_unregister_application(app_execif);
-
+	STANDARD_USECOUNT(res);
 	return res;
 }
 
-static int load_module(void)
+char *key()
 {
-	int res = ast_register_application(app_exec, exec_exec, exec_synopsis, exec_descrip);
-	res |= ast_register_application(app_tryexec, tryexec_exec, tryexec_synopsis, tryexec_descrip);
-	res |= ast_register_application(app_execif, execif_exec, execif_synopsis, execif_descrip);
-	return res;
+	return ASTERISK_GPL_KEY;
 }
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Executes dialplan applications");

apps/app_externalivr.c

@@ -1,671 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2005, Digium, Inc.
- *
- * Kevin P. Fleming <kpfleming@digium.com>
- *
- * Portions taken from the file-based music-on-hold work
- * created by Anthony Minessale II in res_musiconhold.c
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief External IVR application interface
- *
- * \author Kevin P. Fleming <kpfleming@digium.com>
- *
- * \note Portions taken from the file-based music-on-hold work
- * created by Anthony Minessale II in res_musiconhold.c
- *
- * \ingroup applications
- */
-
-/*** MODULEINFO
-	<depend>working_fork</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <signal.h>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
-
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/linkedlists.h"
-#include "asterisk/app.h"
-#include "asterisk/utils.h"
-
-static const char *app = "ExternalIVR";
-
-static const char *synopsis = "Interfaces with an external IVR application";
-
-static const char *descrip =
-"  ExternalIVR(command[,arg[,arg...]]): Forks a process to run the supplied command,\n"
-"and starts a generator on the channel. The generator's play list is\n"
-"controlled by the external application, which can add and clear entries\n"
-"via simple commands issued over its stdout. The external application\n"
-"will receive all DTMF events received on the channel, and notification\n"
-"if the channel is hung up. The application will not be forcibly terminated\n"
-"when the channel is hung up.\n"
-"See doc/externalivr.txt for a protocol specification.\n";
-
-/* XXX the parser in gcc 2.95 gets confused if you don't put a space between 'name' and the comma */
-#define ast_chan_log(level, channel, format, ...) ast_log(level, "%s: " format, channel->name , ## __VA_ARGS__)
-
-struct playlist_entry {
-	AST_LIST_ENTRY(playlist_entry) list;
-	char filename[1];
-};
-
-struct ivr_localuser {
-	struct ast_channel *chan;
-	AST_LIST_HEAD(playlist, playlist_entry) playlist;
-	AST_LIST_HEAD(finishlist, playlist_entry) finishlist;
-	int abort_current_sound;
-	int playing_silence;
-	int option_autoclear;
-};
-
-
-struct gen_state {
-	struct ivr_localuser *u;
-	struct ast_filestream *stream;
-	struct playlist_entry *current;
-	int sample_queue;
-};
-
-static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u, 
-              int *eivr_events_fd, int *eivr_commands_fd, int *eivr_errors_fd, 
-              const char *args);
-
-static void send_eivr_event(FILE *handle, const char event, const char *data,
-	const struct ast_channel *chan)
-{
-	char tmp[256];
-
-	if (!data) {
-		snprintf(tmp, sizeof(tmp), "%c,%10d", event, (int)time(NULL));
-	} else {
-		snprintf(tmp, sizeof(tmp), "%c,%10d,%s", event, (int)time(NULL), data);
-	}
-
-	fprintf(handle, "%s\n", tmp);
-	if (option_debug)
-		ast_chan_log(LOG_DEBUG, chan, "sent '%s'\n", tmp);
-}
-
-static void *gen_alloc(struct ast_channel *chan, void *params)
-{
-	struct ivr_localuser *u = params;
-	struct gen_state *state;
-
-	if (!(state = ast_calloc(1, sizeof(*state))))
-		return NULL;
-
-	state->u = u;
-
-	return state;
-}
-
-static void gen_closestream(struct gen_state *state)
-{
-	if (!state->stream)
-		return;
-
-	ast_closestream(state->stream);
-	state->u->chan->stream = NULL;
-	state->stream = NULL;
-}
-
-static void gen_release(struct ast_channel *chan, void *data)
-{
-	struct gen_state *state = data;
-
-	gen_closestream(state);
-	ast_free(data);
-}
-
-/* caller has the playlist locked */
-static int gen_nextfile(struct gen_state *state)
-{
-	struct ivr_localuser *u = state->u;
-	char *file_to_stream;
-
-	u->abort_current_sound = 0;
-	u->playing_silence = 0;
-	gen_closestream(state);
-
-	while (!state->stream) {
-		state->current = AST_LIST_REMOVE_HEAD(&u->playlist, list);
-		if (state->current) {
-			file_to_stream = state->current->filename;
-		} else {
-			file_to_stream = "silence/10";
-			u->playing_silence = 1;
-		}
-
-		if (!(state->stream = ast_openstream_full(u->chan, file_to_stream, u->chan->language, 1))) {
-			ast_chan_log(LOG_WARNING, u->chan, "File '%s' could not be opened: %s\n", file_to_stream, strerror(errno));
-			if (!u->playing_silence) {
-				continue;
-			} else {
-				break;
-			}
-		}
-	}
-
-	return (!state->stream);
-}
-
-static struct ast_frame *gen_readframe(struct gen_state *state)
-{
-	struct ast_frame *f = NULL;
-	struct ivr_localuser *u = state->u;
-
-	if (u->abort_current_sound ||
-		(u->playing_silence && AST_LIST_FIRST(&u->playlist))) {
-		gen_closestream(state);
-		AST_LIST_LOCK(&u->playlist);
-		gen_nextfile(state);
-		AST_LIST_UNLOCK(&u->playlist);
-	}
-
-	if (!(state->stream && (f = ast_readframe(state->stream)))) {
-		if (state->current) {
-			AST_LIST_LOCK(&u->finishlist);
-			AST_LIST_INSERT_TAIL(&u->finishlist, state->current, list);
-			AST_LIST_UNLOCK(&u->finishlist);
-			state->current = NULL;
-		}
-		if (!gen_nextfile(state))
-			f = ast_readframe(state->stream);
-	}
-
-	return f;
-}
-
-static int gen_generate(struct ast_channel *chan, void *data, int len, int samples)
-{
-	struct gen_state *state = data;
-	struct ast_frame *f = NULL;
-	int res = 0;
-
-	state->sample_queue += samples;
-
-	while (state->sample_queue > 0) {
-		if (!(f = gen_readframe(state)))
-			return -1;
-
-		res = ast_write(chan, f);
-		ast_frfree(f);
-		if (res < 0) {
-			ast_chan_log(LOG_WARNING, chan, "Failed to write frame: %s\n", strerror(errno));
-			return -1;
-		}
-		state->sample_queue -= f->samples;
-	}
-
-	return res;
-}
-
-static struct ast_generator gen =
-{
-	alloc: gen_alloc,
-	release: gen_release,
-	generate: gen_generate,
-};
-
-static void ast_eivr_getvariable(struct ast_channel *chan, char *data, char *outbuf, int outbuflen)
-{
-	/* original input data: "G,var1,var2," */
-	/* data passed as "data":  "var1,var2" */
-
-	char *inbuf, *variable;
-	const char *value;
-	int j;
-	struct ast_str *newstring = ast_str_alloca(outbuflen); 
-
-	outbuf[0] = '\0';
-
-	for (j = 1, inbuf = data; ; j++) {
-		variable = strsep(&inbuf, ",");
-		if (variable == NULL) {
-			int outstrlen = strlen(outbuf);
-			if(outstrlen && outbuf[outstrlen - 1] == ',') {
-				outbuf[outstrlen - 1] = 0;
-			}
-			break;
-		}
-		
-		value = pbx_builtin_getvar_helper(chan, variable);
-		if(!value)
-			value = "";
-		ast_str_append(&newstring, 0, "%s=%s,", variable, value);
-		ast_copy_string(outbuf, newstring->str, outbuflen);
-	}
-};
-
-static void ast_eivr_setvariable(struct ast_channel *chan, char *data)
-{
-	char *value;
-
-	char *inbuf = ast_strdupa(data), *variable;
-
-	for (variable = strsep(&inbuf, ","); variable; variable = strsep(&inbuf, ",")) {
-		ast_debug(1, "Setting up a variable: %s\n", variable);
-		/* variable contains "varname=value" */
-		value = strchr(variable, '=');
-		if (!value) {
-			value = "";
-		} else {
-			*value++ = '\0';
-		}
-		pbx_builtin_setvar_helper(chan, variable, value);
-	}
-};
-
-static struct playlist_entry *make_entry(const char *filename)
-{
-	struct playlist_entry *entry;
-
-	if (!(entry = ast_calloc(1, sizeof(*entry) + strlen(filename) + 10))) /* XXX why 10 ? */
-		return NULL;
-
-	strcpy(entry->filename, filename);
-
-	return entry;
-}
-
-static int app_exec(struct ast_channel *chan, void *data)
-{
-	struct playlist_entry *entry;
-	int child_stdin[2] = { -1, -1 };
-	int child_stdout[2] = { -1, -1 };
-	int child_stderr[2] = { -1, -1 };
-	int res = -1;
-	int gen_active = 0;
-	int pid;
-	char *buf, *comma_delim_argbuf;
-	struct ivr_localuser foo = {
-		.playlist = AST_LIST_HEAD_INIT_VALUE,
-		.finishlist = AST_LIST_HEAD_INIT_VALUE,
-	};
-	struct ivr_localuser *u = &foo;
-	sigset_t fullset, oldset;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(cmd)[32];
-	);
-
-	sigfillset(&fullset);
-	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
-
-	u->abort_current_sound = 0;
-	u->chan = chan;
-
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "ExternalIVR requires a command to execute\n");
-		return -1;
-	}
-
-	buf = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, buf);
-
-	/* copy args and replace commas with pipes */
-	comma_delim_argbuf = ast_strdupa(data);
-	
-	if (pipe(child_stdin)) {
-		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child input: %s\n", strerror(errno));
-		goto exit;
-	}
-	if (pipe(child_stdout)) {
-		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child output: %s\n", strerror(errno));
-		goto exit;
-	}
-	if (pipe(child_stderr)) {
-		ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child errors: %s\n", strerror(errno));
-		goto exit;
-	}
-	if (chan->_state != AST_STATE_UP) {
-		ast_answer(chan);
-	}
-	if (ast_activate_generator(chan, &gen, u) < 0) {
-		ast_chan_log(LOG_WARNING, chan, "Failed to activate generator\n");
-		goto exit;
-	} else
-		gen_active = 1;
-
-	pid = fork();
-	if (pid < 0) {
-		ast_log(LOG_WARNING, "Failed to fork(): %s\n", strerror(errno));
-		goto exit;
-	}
-
-	if (!pid) {
-		/* child process */
-		int i;
-#ifdef HAVE_CAP
-		cap_t cap = cap_from_text("cap_net_admin-eip");
-
-		if (cap_set_proc(cap)) {
-			/* Careful with order! Logging cannot happen after we close FDs */
-			ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-		}
-		cap_free(cap);
-#endif
-
-		signal(SIGPIPE, SIG_DFL);
-		pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
-
-		if (ast_opt_high_priority)
-			ast_set_priority(0);
-
-		dup2(child_stdin[0], STDIN_FILENO);
-		dup2(child_stdout[1], STDOUT_FILENO);
-		dup2(child_stderr[1], STDERR_FILENO);
-		for (i = STDERR_FILENO + 1; i < 1024; i++)
-			close(i);
-		execv(args.cmd[0], args.cmd);
-		fprintf(stderr, "Failed to execute '%s': %s\n", args.cmd[0], strerror(errno));
-		_exit(1);
-	} else {
-		/* parent process */
-
-		close(child_stdin[0]);
-		child_stdin[0] = -1;
-		close(child_stdout[1]);
-		child_stdout[1] = -1;
-		close(child_stderr[1]);
-		child_stderr[1] = -1;
-		res = eivr_comm(chan, u, &child_stdin[1], &child_stdout[0], &child_stderr[0], comma_delim_argbuf);
-	}
-
-	exit:
-	if (gen_active) {
-		ast_deactivate_generator(chan);
-	}
-	if (child_stdin[0] > -1) {
-		close(child_stdin[0]);
-	}
-	if (child_stdin[1] > -1) {
-		close(child_stdin[1]);
-	}
-	if (child_stdout[0] > -1) {
-		close(child_stdout[0]);
-	}
-	if (child_stdout[1] > -1) {
-		close(child_stdout[1]);
-	}
-	if (child_stderr[0] > -1) {
-		close(child_stderr[0]);
-	}
-	if (child_stderr[1] > -1) {
-		close(child_stderr[1]);
-	}
-	while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
-		ast_free(entry);
-	}
-	return res;
-}
-
-static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u, 
- 				int *eivr_events_fd, int *eivr_commands_fd, int *eivr_errors_fd, 
- 				const char *args)
-{
-	struct playlist_entry *entry;
-	struct ast_frame *f;
-	int ms;
- 	int exception;
- 	int ready_fd;
- 	int waitfds[2] = { *eivr_commands_fd, (eivr_errors_fd) ? *eivr_errors_fd : -1 };
- 	struct ast_channel *rchan;
- 	char *command;
- 	int res = -1;
-	int test_available_fd = -1;
-  
- 	FILE *eivr_commands = NULL;
- 	FILE *eivr_errors = NULL;
- 	FILE *eivr_events = NULL;
- 
- 	if (!(eivr_events = fdopen(*eivr_events_fd, "w"))) {
- 		ast_chan_log(LOG_WARNING, chan, "Could not open stream to send events\n");
- 		goto exit;
- 	}
- 	if (!(eivr_commands = fdopen(*eivr_commands_fd, "r"))) {
- 		ast_chan_log(LOG_WARNING, chan, "Could not open stream to receive commands\n");
- 		goto exit;
- 	}
- 	if (eivr_errors_fd) {  /*if opening a socket connection, error stream will not be used*/
- 		if (!(eivr_errors = fdopen(*eivr_errors_fd, "r"))) {
- 			ast_chan_log(LOG_WARNING, chan, "Could not open stream to receive errors\n");
- 			goto exit;
- 		}
- 	}
-
-	test_available_fd = open("/dev/null", O_RDONLY);
- 
- 	setvbuf(eivr_events, NULL, _IONBF, 0);
- 	setvbuf(eivr_commands, NULL, _IONBF, 0);
- 	if(eivr_errors)
-		setvbuf(eivr_errors, NULL, _IONBF, 0);
-
-	res = 0;
- 
- 	while (1) {
- 		if (ast_test_flag(chan, AST_FLAG_ZOMBIE)) {
- 			ast_chan_log(LOG_NOTICE, chan, "Is a zombie\n");
- 			res = -1;
- 			break;
- 		}
- 		if (ast_check_hangup(chan)) {
- 			ast_chan_log(LOG_NOTICE, chan, "Got check_hangup\n");
- 			send_eivr_event(eivr_events, 'H', NULL, chan);
- 			res = -1;
- 			break;
- 		}
- 
- 		ready_fd = 0;
- 		ms = 100;
- 		errno = 0;
- 		exception = 0;
- 
- 		rchan = ast_waitfor_nandfds(&chan, 1, waitfds, (eivr_errors_fd) ? 2 : 1, &exception, &ready_fd, &ms);
- 
- 		if (!AST_LIST_EMPTY(&u->finishlist)) {
- 			AST_LIST_LOCK(&u->finishlist);
- 			while ((entry = AST_LIST_REMOVE_HEAD(&u->finishlist, list))) {
- 				send_eivr_event(eivr_events, 'F', entry->filename, chan);
- 				ast_free(entry);
- 			}
- 			AST_LIST_UNLOCK(&u->finishlist);
- 		}
- 
- 		if (rchan) {
- 			/* the channel has something */
- 			f = ast_read(chan);
- 			if (!f) {
- 				ast_chan_log(LOG_NOTICE, chan, "Returned no frame\n");
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
- 				res = -1;
- 				break;
- 			}
- 			if (f->frametype == AST_FRAME_DTMF) {
- 				send_eivr_event(eivr_events, f->subclass, NULL, chan);
- 				if (u->option_autoclear) {
-  					if (!u->abort_current_sound && !u->playing_silence)
- 						send_eivr_event(eivr_events, 'T', NULL, chan);
-  					AST_LIST_LOCK(&u->playlist);
-  					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
- 						send_eivr_event(eivr_events, 'D', entry->filename, chan);
-  						ast_free(entry);
-  					}
-  					if (!u->playing_silence)
-  						u->abort_current_sound = 1;
-  					AST_LIST_UNLOCK(&u->playlist);
-  				}
- 			} else if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP)) {
- 				ast_chan_log(LOG_NOTICE, chan, "Got AST_CONTROL_HANGUP\n");
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
- 				ast_frfree(f);
- 				res = -1;
- 				break;
- 			}
- 			ast_frfree(f);
- 		} else if (ready_fd == *eivr_commands_fd) {
- 			char input[1024];
- 
- 			if (exception || (dup2(*eivr_commands_fd, test_available_fd) == -1) || feof(eivr_commands)) {
- 				ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
- 				res = -1;
-  				break;
-  			}
-  
- 			if (!fgets(input, sizeof(input), eivr_commands))
- 				continue;
- 
- 			command = ast_strip(input);
-  
- 			if (option_debug)
- 				ast_chan_log(LOG_DEBUG, chan, "got command '%s'\n", input);
-  
- 			if (strlen(input) < 4)
- 				continue;
-  
-			if (input[0] == 'P') {
- 				send_eivr_event(eivr_events, 'P', args, chan);
- 
- 			} else if (input[0] == 'S') {
- 				if (ast_fileexists(&input[2], NULL, u->chan->language) == -1) {
- 					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
- 					send_eivr_event(eivr_events, 'Z', NULL, chan);
- 					strcpy(&input[2], "exception");
- 				}
- 				if (!u->abort_current_sound && !u->playing_silence)
- 					send_eivr_event(eivr_events, 'T', NULL, chan);
- 				AST_LIST_LOCK(&u->playlist);
- 				while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
- 					send_eivr_event(eivr_events, 'D', entry->filename, chan);
- 					ast_free(entry);
- 				}
- 				if (!u->playing_silence)
- 					u->abort_current_sound = 1;
- 				entry = make_entry(&input[2]);
- 				if (entry)
- 					AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
- 				AST_LIST_UNLOCK(&u->playlist);
- 			} else if (input[0] == 'A') {
- 				if (ast_fileexists(&input[2], NULL, u->chan->language) == -1) {
- 					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
- 					send_eivr_event(eivr_events, 'Z', NULL, chan);
- 					strcpy(&input[2], "exception");
- 				}
- 				entry = make_entry(&input[2]);
- 				if (entry) {
- 					AST_LIST_LOCK(&u->playlist);
- 					AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
- 					AST_LIST_UNLOCK(&u->playlist);
- 				}
- 			} else if (input[0] == 'G') {
- 				/* A get variable message:  "G,variable1,variable2,..." */
- 				char response[2048];
-
- 				ast_chan_log(LOG_NOTICE, chan, "Getting a Variable out of the channel: %s\n", &input[2]);
- 				ast_eivr_getvariable(chan, &input[2], response, sizeof(response));
- 				send_eivr_event(eivr_events, 'G', response, chan);
- 			} else if (input[0] == 'V') {
- 				/* A set variable message:  "V,variablename=foo" */
- 				ast_chan_log(LOG_NOTICE, chan, "Setting a Variable up: %s\n", &input[2]);
- 				ast_eivr_setvariable(chan, &input[2]);
- 			} else if (input[0] == 'L') {
- 				ast_chan_log(LOG_NOTICE, chan, "Log message from EIVR: %s\n", &input[2]);
- 			} else if (input[0] == 'X') {
- 				ast_chan_log(LOG_NOTICE, chan, "Exiting ExternalIVR: %s\n", &input[2]);
- 				/*! \todo add deprecation debug message for X command here */
- 				res = 0;
- 				break;
-			} else if (input[0] == 'E') {
- 				ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
- 				send_eivr_event(eivr_events, 'E', NULL, chan);
- 				res = 0;
- 				break;
- 			} else if (input[0] == 'H') {
- 				ast_chan_log(LOG_NOTICE, chan, "Hanging up: %s\n", &input[2]);
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
- 				res = -1;
- 				break;
- 			} else if (input[0] == 'O') {
- 				if (!strcasecmp(&input[2], "autoclear"))
- 					u->option_autoclear = 1;
- 				else if (!strcasecmp(&input[2], "noautoclear"))
- 					u->option_autoclear = 0;
- 				else
- 					ast_chan_log(LOG_WARNING, chan, "Unknown option requested '%s'\n", &input[2]);
- 			}
- 		} else if (eivr_errors_fd && (ready_fd == *eivr_errors_fd)) {
- 			char input[1024];
-  
- 			if (exception || feof(eivr_errors)) {
- 				ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
- 				res = -1;
- 				break;
- 			}
- 			if (fgets(input, sizeof(input), eivr_errors)) {
- 				command = ast_strip(input);
- 				ast_chan_log(LOG_NOTICE, chan, "stderr: %s\n", command);
- 			}
- 		} else if ((ready_fd < 0) && ms) { 
- 			if (errno == 0 || errno == EINTR)
- 				continue;
- 
- 			ast_chan_log(LOG_WARNING, chan, "Wait failed (%s)\n", strerror(errno));
- 			break;
- 		}
- 	}
-
-	exit:
-	if (test_available_fd > -1) {
-		close(test_available_fd);
-	}
-	if (eivr_events) {
- 		fclose(eivr_events);
- 		*eivr_events_fd = -1;
-	}
- 	if (eivr_commands) {
-		fclose(eivr_commands);
-		*eivr_commands_fd = -1;
-	}
- 	if (eivr_errors) {
- 		fclose(eivr_errors);
-  		*eivr_errors_fd = -1;
-	}
-	return res;
- }
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app, app_exec, synopsis, descrip);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "External IVR Interface Application");

apps/app_fax.c

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

apps/app_festival.c

@@ -1,186 +1,146 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
+ * Connect to festival
+ * 
  * Copyright (C) 2002, Christos Ricudis
  *
  * Christos Ricudis <ricudis@itc.auth.gr>
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
 
-/*! \file
- *
- * \brief Connect to festival
- *
- * \author Christos Ricudis <ricudis@itc.auth.gr>
- *
- * \extref  The Festival Speech Synthesis System - http://www.cstr.ed.ac.uk/projects/festival/
- * 
- * \ingroup applications
- */
-
-/*** MODULEINFO
-	<depend>working_fork</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
+#include <sys/types.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/md5.h>
+#include <asterisk/config.h>
+#include <asterisk/utils.h>
+#include <asterisk/lock.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+#include <sys/types.h>
 #include <sys/socket.h>
 #include <netdb.h>
 #include <netinet/in.h>
 #include <arpa/inet.h>
+#include <stdio.h>
 #include <signal.h>
+#include <stdlib.h>
+#include <unistd.h>
 #include <fcntl.h>
 #include <ctype.h>
-#include <errno.h>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
 
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/md5.h"
-#include "asterisk/config.h"
-#include "asterisk/utils.h"
-#include "asterisk/lock.h"
-#include "asterisk/app.h"
 
 #define FESTIVAL_CONFIG "festival.conf"
-#define MAXLEN 180
-#define MAXFESTLEN 2048
+
+static char *tdesc = "Simple Festival Interface";
 
 static char *app = "Festival";
 
 static char *synopsis = "Say text to the user";
 
 static char *descrip = 
-"  Festival(text[,intkeys]):  Connect to Festival, send the argument, get back the waveform,\n"
+"  Festival(text[|intkeys]):  Connect to Festival, send the argument, get back the waveform,"
 "play it to the user, allowing any given interrupt keys to immediately terminate and return\n"
 "the value, or 'any' to allow any number back (useful in dialplan)\n";
 
+STANDARD_LOCAL_USER;
 
-static char *socket_receive_file_to_buff(int fd, int *size)
+LOCAL_USER_DECL;
+
+static char *socket_receive_file_to_buff(int fd,int *size)
 {
-	/* Receive file (probably a waveform file) from socket using
-	 * Festival key stuff technique, but long winded I know, sorry
-	 * but will receive any file without closing the stream or
-	 * using OOB data
-	 */
-	static char *file_stuff_key = "ft_StUfF_key"; /* must == Festival's key */
-	char *buff, *tmp;
-	int bufflen;
-	int n,k,i;
-	char c;
+    /* Receive file (probably a waveform file) from socket using   */
+    /* Festival key stuff technique, but long winded I know, sorry */
+    /* but will receive any file without closeing the stream or    */
+    /* using OOB data                                              */
+    static char *file_stuff_key = "ft_StUfF_key"; /* must == Festival's key */
+    char *buff;
+    int bufflen;
+    int n,k,i;
+    char c;
 
-	bufflen = 1024;
-	if (!(buff = ast_malloc(bufflen)))
-		return NULL;
-	*size = 0;
+    bufflen = 1024;
+    buff = (char *)malloc(bufflen);
+    *size=0;
 
-	for (k = 0; file_stuff_key[k] != '\0';) {
-		n = read(fd, &c, 1);
-		if (n == 0)
-			break;  /* hit stream eof before end of file */
-		if ((*size) + k + 1 >= bufflen) {
-			/* +1 so you can add a terminating NULL if you want */
-			bufflen += bufflen / 4;
-			if (!(tmp = ast_realloc(buff, bufflen))) {
-				ast_free(buff);
-				return NULL;
-			}
-			buff = tmp;
-		}
-		if (file_stuff_key[k] == c)
-			k++;
-		else if ((c == 'X') && (file_stuff_key[k+1] == '\0')) {
-			/* It looked like the key but wasn't */
-			for (i = 0; i < k; i++, (*size)++)
-				buff[*size] = file_stuff_key[i];
-			k = 0;
-			/* omit the stuffed 'X' */
-		} else {
-			for (i = 0; i < k; i++, (*size)++)
-				buff[*size] = file_stuff_key[i];
-			k = 0;
-			buff[*size] = c;
-			(*size)++;
-		}
-	}
+    for (k=0; file_stuff_key[k] != '\0';)
+    {
+        n = read(fd,&c,1);
+        if (n==0) break;  /* hit stream eof before end of file */
+        if ((*size)+k+1 >= bufflen)
+        {   /* +1 so you can add a NULL if you want */
+            bufflen += bufflen/4;
+            buff = (char *)realloc(buff,bufflen);
+        }
+        if (file_stuff_key[k] == c)
+            k++;
+        else if ((c == 'X') && (file_stuff_key[k+1] == '\0'))
+        {   /* It looked like the key but wasn't */
+            for (i=0; i < k; i++,(*size)++)
+                buff[*size] = file_stuff_key[i];
+            k=0;
+            /* omit the stuffed 'X' */
+        }
+        else
+        {
+            for (i=0; i < k; i++,(*size)++)
+                buff[*size] = file_stuff_key[i];
+            k=0;
+            buff[*size] = c;
+            (*size)++;
+        }
 
-	return buff;
+    }
+
+    return buff;
 }
 
-static int send_waveform_to_fd(char *waveform, int length, int fd)
-{
-	int res;
-	int x;
+static int send_waveform_to_fd(char *waveform, int length, int fd) {
+
+        int res;
+        int x;
 #ifdef __PPC__ 
 	char c;
 #endif
-	sigset_t fullset, oldset;
-#ifdef HAVE_CAP
-	cap_t cap;
-#endif
 
-	sigfillset(&fullset);
-	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
-
-	res = fork();
-	if (res < 0)
-		ast_log(LOG_WARNING, "Fork failed\n");
-	if (res) {
-		pthread_sigmask(SIG_SETMASK, &oldset, NULL);
-		return res;
-	}
-#ifdef HAVE_CAP
-	cap = cap_from_text("cap_net_admin-eip");
-
-	if (cap_set_proc(cap)) {
-		/* Careful with order! Logging cannot happen after we close FDs */
-		ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-	}
-	cap_free(cap);
-#endif
-	for (x = 0; x < 256; x++) {
-		if (x != fd)
-			close(x);
-	}
-	if (ast_opt_high_priority)
-		ast_set_priority(0);
-	signal(SIGPIPE, SIG_DFL);
-	pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
+        res = fork();
+        if (res < 0)
+                ast_log(LOG_WARNING, "Fork failed\n");
+        if (res)
+                return res;
+        for (x=0;x<256;x++) {
+                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;
+	for( x=0; x<length; x+=2)
+	{
+		c = *(waveform+x+1);
+		*(waveform+x+1)=*(waveform+x);
+		*(waveform+x)=c;
 	}
 #endif
 	
-	if (write(fd, waveform, length) < 0) {
-		ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
-	}
-
+	write(fd,waveform,length);
+	write(fd,"a",1);
 	close(fd);
 	exit(0);
 }
 
-static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, int length, char *intkeys)
-{
-	int res = 0;
+
+static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, int length, char *intkeys) {
+	int res=0;
 	int fds[2];
+	int ms = -1;
 	int pid = -1;
 	int needed = 0;
 	int owriteformat;
@@ -189,21 +149,18 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 		struct ast_frame f;
 		char offset[AST_FRIENDLY_OFFSET];
 		char frdata[2048];
-	} myf = {
-		.f = { 0, },
-	};
-
-	if (pipe(fds)) {
-		ast_log(LOG_WARNING, "Unable to create pipe\n");
-		return -1;
-	}
-
+	} myf;
+	
+        if (pipe(fds)) {
+                 ast_log(LOG_WARNING, "Unable to create pipe\n");
+        	return -1;
+        }
+	                                                
 	/* Answer if it's not already going */
 	if (chan->_state != AST_STATE_UP)
 		ast_answer(chan);
 	ast_stopstream(chan);
-	ast_indicate(chan, -1);
-	
+
 	owriteformat = chan->writeformat;
 	res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
 	if (res < 0) {
@@ -211,13 +168,14 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 		return -1;
 	}
 	
-	res = send_waveform_to_fd(waveform, length, fds[1]);
+	res=send_waveform_to_fd(waveform,length,fds[1]);
 	if (res >= 0) {
 		pid = res;
 		/* Order is important -- there's almost always going to be mp3...  we want to prioritize the
 		   user */
 		for (;;) {
-			res = ast_waitfor(chan, 1000);
+			ms = 1000;
+			res = ast_waitfor(chan, ms);
 			if (res < 1) {
 				res = -1;
 				break;
@@ -229,7 +187,7 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 				break;
 			}
 			if (f->frametype == AST_FRAME_DTMF) {
-				ast_debug(1, "User pressed a key\n");
+				ast_log(LOG_DEBUG, "User pressed a key\n");
 				if (intkeys && strchr(intkeys, f->subclass)) {
 					res = f->subclass;
 					ast_frfree(f);
@@ -250,22 +208,21 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 					myf.f.subclass = AST_FORMAT_SLINEAR;
 					myf.f.datalen = res;
 					myf.f.samples = res / 2;
+					myf.f.mallocd = 0;
 					myf.f.offset = AST_FRIENDLY_OFFSET;
 					myf.f.src = __PRETTY_FUNCTION__;
 					myf.f.data = myf.frdata;
 					if (ast_write(chan, &myf.f) < 0) {
 						res = -1;
-						ast_frfree(f);
 						break;
 					}
-					if (res < needed) { /* last frame */
-						ast_debug(1, "Last frame\n");
-						res = 0;
-						ast_frfree(f);
+					if (res < needed) { // last frame
+						ast_log(LOG_DEBUG, "Last frame\n");
+						res=0;
 						break;
 					}
 				} else {
-					ast_debug(1, "No more waveform\n");
+					ast_log(LOG_DEBUG, "No more waveform\n");
 					res = 0;
 				}
 			}
@@ -274,30 +231,34 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 	}
 	close(fds[0]);
 	close(fds[1]);
-
-#if 0
-	if (pid > -1)
-		kill(pid, SIGKILL);
-#endif
+//	if (pid > -1)
+//		kill(pid, SIGKILL);
 	if (!res && owriteformat)
 		ast_set_write_format(chan, owriteformat);
 	return res;
 }
 
+#define MAXLEN 180
+#define MAXFESTLEN 2048
+
+
+
+
 static int festival_exec(struct ast_channel *chan, void *vdata)
 {
 	int usecache;
-	int res = 0;
-	struct sockaddr_in serv_addr;
+	int res=0;
+	struct localuser *u;
+ 	struct sockaddr_in serv_addr;
 	struct hostent *serverhost;
 	struct ast_hostent ahp;
 	int fd;
 	FILE *fs;
-	const char *host;
-	const char *cachedir;
-	const char *temp;
-	const char *festivalcommand;
-	int port = 1314;
+	char *host;
+	char *cachedir;
+	char *temp;
+	char *festivalcommand;
+	int port=1314;
 	int n;
 	char ack[4];
 	char *waveform;
@@ -310,27 +271,17 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
 	char MD5Hex[33] = "";
 	char koko[4] = "";
 	char cachefile[MAXFESTLEN]="";
-	int readcache = 0;
-	int writecache = 0;
+	int readcache=0;
+	int writecache=0;
 	int strln;
 	int fdesc = -1;
 	char buffer[16384];
 	int seekpos = 0;	
-	char *data;	
+	char data[256] = "";
+	char *intstr;
+	
 	struct ast_config *cfg;
-	char *newfestivalcommand;
-	struct ast_flags config_flags = { 0 };
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(text);
-		AST_APP_ARG(interrupt);
-	);
-
-	if (ast_strlen_zero(vdata)) {
-		ast_log(LOG_WARNING, "festival requires an argument (text)\n");
-		return -1;
-	}
-
-	cfg = ast_config_load(FESTIVAL_CONFIG, config_flags);
+	cfg = ast_load(FESTIVAL_CONFIG);
 	if (!cfg) {
 		ast_log(LOG_WARNING, "No such configuration file %s\n", FESTIVAL_CONFIG);
 		return -1;
@@ -344,226 +295,189 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
 		port = atoi(temp);
 	}
 	if (!(temp = ast_variable_retrieve(cfg, "general", "usecache"))) {
-		usecache = 0;
+		usecache=0;
 	} else {
 		usecache = ast_true(temp);
 	}
 	if (!(cachedir = ast_variable_retrieve(cfg, "general", "cachedir"))) {
 		cachedir = "/tmp/";
 	}
-
-	data = ast_strdupa(vdata);
-	AST_STANDARD_APP_ARGS(args, data);
-
 	if (!(festivalcommand = ast_variable_retrieve(cfg, "general", "festivalcommand"))) {
-		const char *startcmd = "(tts_textasterisk \"";
-		const char *endcmd = "\" 'file)(quit)\n";
-
-		strln = strlen(startcmd) + strlen(args.text) + strlen(endcmd) + 1;
-		newfestivalcommand = alloca(strln);
-		snprintf(newfestivalcommand, strln, "%s%s%s", startcmd, args.text, endcmd);
-		festivalcommand = newfestivalcommand;
-	} else { /* This else parses the festivalcommand that we're sent from the config file for \n's, etc */
-		int i, j;
-		newfestivalcommand = alloca(strlen(festivalcommand) + strlen(args.text) + 1);
-
-		for (i = 0, j = 0; i < strlen(festivalcommand); i++) {
-			if (festivalcommand[i] == '\\' && festivalcommand[i + 1] == 'n') {
-				newfestivalcommand[j++] = '\n';
-				i++;
-			} else if (festivalcommand[i] == '\\') {
-				newfestivalcommand[j++] = festivalcommand[i + 1];
-				i++;
-			} else if (festivalcommand[i] == '%' && festivalcommand[i + 1] == 's') {
-				sprintf(&newfestivalcommand[j], "%s", args.text); /* we know it is big enough */
-				j += strlen(args.text);
-				i++;
-			} else
-				newfestivalcommand[j++] = festivalcommand[i];
-		}
-		newfestivalcommand[j] = '\0';
-		festivalcommand = newfestivalcommand;
+		festivalcommand = "(tts_textasterisk \"%s\" 'file)(quit)\n";
 	}
-	
-	if (args.interrupt && !strcasecmp(args.interrupt, "any"))
-		args.interrupt = AST_DIGIT_ANY;
-
-	ast_debug(1, "Text passed to festival server : %s\n", args.text);
-	/* Connect to local festival server */
-	
-	fd = socket(AF_INET, SOCK_STREAM, IPPROTO_TCP);
-
-	if (fd < 0) {
-		ast_log(LOG_WARNING, "festival_client: can't get socket\n");
-		ast_config_destroy(cfg);
+	if (!vdata || ast_strlen_zero(vdata)) {
+		ast_log(LOG_WARNING, "festival requires an argument (text)\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 = ast_gethostbyname(host, &ahp);
-
-		if (serverhost == NULL) {
-			ast_log(LOG_WARNING, "festival_client: gethostbyname failed\n");
-			ast_config_destroy(cfg);
-			return -1;
-		}
-		memmove(&serv_addr.sin_addr, serverhost->h_addr, serverhost->h_length);
+	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_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 = 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);
+    	}
 	serv_addr.sin_family = AF_INET;
 	serv_addr.sin_port = htons(port);
 
 	if (connect(fd, (struct sockaddr *)&serv_addr, sizeof(serv_addr)) != 0) {
-		ast_log(LOG_WARNING, "festival_client: connect to server failed\n");
-		ast_config_destroy(cfg);
-		return -1;
+		ast_log(LOG_WARNING,"festival_client: connect to server failed\n");
+		ast_destroy(cfg);
+        	return -1;
+    	}
+    	
+    	/* Compute MD5 sum of string */
+    	MD5Init(&md5ctx);
+    	MD5Update(&md5ctx,(unsigned char const *)data,strlen(data));
+    	MD5Final(MD5Res,&md5ctx);
+		MD5Hex[0] = '\0';
+    	
+    	/* Convert to HEX and look if there is any matching file in the cache 
+    		directory */
+    	for (i=0;i<16;i++) {
+    		snprintf(koko, sizeof(koko), "%X",MD5Res[i]);
+    		strncat(MD5Hex, koko, sizeof(MD5Hex) - strlen(MD5Hex) - 1);
+    	}
+    	readcache=0;
+    	writecache=0;
+    	if (strlen(cachedir)+strlen(MD5Hex)+1<=MAXFESTLEN && (usecache==-1)) {
+    		snprintf(cachefile, sizeof(cachefile), "%s/%s", cachedir, MD5Hex);
+    		fdesc=open(cachefile,O_RDWR);
+    		if (fdesc==-1) {
+    			fdesc=open(cachefile,O_CREAT|O_RDWR,0777);
+    			if (fdesc!=-1) {
+    				writecache=1;
+    				strln=strlen((char *)data);
+    				ast_log(LOG_DEBUG,"line length : %d\n",strln);
+    				write(fdesc,&strln,sizeof(int));
+    				write(fdesc,data,strln);
+				seekpos=lseek(fdesc,0,SEEK_CUR);
+				ast_log(LOG_DEBUG,"Seek position : %d\n",seekpos);
+    			}
+    		} else {
+    			read(fdesc,&strln,sizeof(int));
+    			ast_log(LOG_DEBUG,"Cache file exists, strln=%d, strlen=%d\n",strln,(int)strlen((char *)data));
+    			if (strlen((char *)data)==strln) {
+    				ast_log(LOG_DEBUG,"Size OK\n");
+    				read(fdesc,&bigstring,strln);
+	    			bigstring[strln] = 0;
+				if (strcmp(bigstring,data)==0) { 
+	    				readcache=1;
+	    			} else {
+	    				ast_log(LOG_WARNING,"Strings do not match\n");
+	    			}
+	    		} else {
+	    			ast_log(LOG_WARNING,"Size mismatch\n");
+	    		}
+	    	}
 	}
-
-	/* Compute MD5 sum of string */
-	MD5Init(&md5ctx);
-	MD5Update(&md5ctx, (unsigned char *)args.text, strlen(args.text));
-	MD5Final(MD5Res, &md5ctx);
-	MD5Hex[0] = '\0';
-
-	/* Convert to HEX and look if there is any matching file in the cache 
-		directory */
-	for (i = 0; i < 16; i++) {
-		snprintf(koko, sizeof(koko), "%X", MD5Res[i]);
-		strncat(MD5Hex, koko, sizeof(MD5Hex) - strlen(MD5Hex) - 1);
-	}
-	readcache = 0;
-	writecache = 0;
-	if (strlen(cachedir) + strlen(MD5Hex) + 1 <= MAXFESTLEN && (usecache == -1)) {
-		snprintf(cachefile, sizeof(cachefile), "%s/%s", cachedir, MD5Hex);
-		fdesc = open(cachefile, O_RDWR);
-		if (fdesc == -1) {
-			fdesc = open(cachefile, O_CREAT | O_RDWR, AST_FILE_MODE);
-			if (fdesc != -1) {
-				writecache = 1;
-				strln = strlen(args.text);
-				ast_debug(1, "line length : %d\n", strln);
-    				if (write(fdesc,&strln,sizeof(int)) < 0) {
-					ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
-				}
-    				if (write(fdesc,data,strln) < 0) {
-					ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
-				}
-				seekpos = lseek(fdesc, 0, SEEK_CUR);
-				ast_debug(1, "Seek position : %d\n", seekpos);
-			}
-		} else {
-    			if (read(fdesc,&strln,sizeof(int)) != sizeof(int)) {
-				ast_log(LOG_WARNING, "read() failed: %s\n", strerror(errno));
-			}
-			ast_debug(1, "Cache file exists, strln=%d, strlen=%d\n", strln, (int)strlen(args.text));
-			if (strlen(args.text) == strln) {
-				ast_debug(1, "Size OK\n");
-    				if (read(fdesc,&bigstring,strln) != strln) {
-					ast_log(LOG_WARNING, "read() failed: %s\n", strerror(errno));
-				}
-				bigstring[strln] = 0;
-				if (strcmp(bigstring, args.text) == 0) { 
-					readcache = 1;
-				} else {
-					ast_log(LOG_WARNING, "Strings do not match\n");
-				}
-			} else {
-				ast_log(LOG_WARNING, "Size mismatch\n");
-			}
-		}
-	}
-
-	if (readcache == 1) {
+    			
+	if (readcache==1) {
 		close(fd);
-		fd = fdesc;
-		ast_debug(1, "Reading from cache...\n");
+		fd=fdesc;
+		ast_log(LOG_DEBUG,"Reading from cache...\n");
 	} else {
-		ast_debug(1, "Passing text to festival...\n");
-		fs = fdopen(dup(fd), "wb");
-
-		fprintf(fs, "%s", festivalcommand);
+		ast_log(LOG_DEBUG,"Passing text to festival...\n");
+    		fs=fdopen(dup(fd),"wb");
+		fprintf(fs,festivalcommand,(char *)data);
 		fflush(fs);
 		fclose(fs);
 	}
 	
 	/* Write to cache and then pass it down */
-	if (writecache == 1) {
-		ast_debug(1, "Writing result to cache...\n");
-		while ((strln = read(fd, buffer, 16384)) != 0) {
-			if (write(fdesc,buffer,strln) < 0) {
-				ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
-			}
+	if (writecache==1) {
+		ast_log(LOG_DEBUG,"Writing result to cache...\n");
+		while ((strln=read(fd,buffer,16384))!=0) {
+			write(fdesc,buffer,strln);
 		}
 		close(fd);
 		close(fdesc);
-		fd = open(cachefile, O_RDWR);
-		lseek(fd, seekpos, SEEK_SET);
+		fd=open(cachefile,O_RDWR);
+		lseek(fd,seekpos,SEEK_SET);
 	}
 	
-	ast_debug(1, "Passing data to channel...\n");
-
+	ast_log(LOG_DEBUG,"Passing data to channel...\n");
+	
 	/* Read back info from server */
 	/* This assumes only one waveform will come back, also LP is unlikely */
 	wave = 0;
 	do {
-		int read_data;
-		for (n = 0; n < 3; ) {
-			read_data = read(fd, ack + n, 3 - n);
-			/* this avoids falling in infinite loop
-			 * in case that festival server goes down
-			 */
-			if (read_data == -1) {
-				ast_log(LOG_WARNING, "Unable to read from cache/festival fd\n");
-				close(fd);
-				ast_config_destroy(cfg);
-				return -1;
-			}
-			n += read_data;
-		}
+		for (n=0; n < 3; )
+			n += read(fd,ack+n,3-n);
 		ack[3] = '\0';
-		if (strcmp(ack, "WV\n") == 0) {         /* receive a waveform */
-			ast_debug(1, "Festival WV command\n");
-			if ((waveform = socket_receive_file_to_buff(fd, &filesize))) {
-				res = send_waveform_to_channel(chan, waveform, filesize, args.interrupt);
-				ast_free(waveform);
-			}
-			break;
-		} else if (strcmp(ack, "LP\n") == 0) {   /* receive an s-expr */
-			ast_debug(1, "Festival LP command\n");
-			if ((waveform = socket_receive_file_to_buff(fd, &filesize))) {
-				waveform[filesize] = '\0';
-				ast_log(LOG_WARNING, "Festival returned LP : %s\n", waveform);
-				ast_free(waveform);
-			}
-		} else if (strcmp(ack, "ER\n") == 0) {    /* server got an error */
-			ast_log(LOG_WARNING, "Festival returned ER\n");
-			res = -1;
+		if (strcmp(ack,"WV\n") == 0) {         /* receive a waveform */
+			ast_log(LOG_DEBUG,"Festival WV command\n");
+			waveform = socket_receive_file_to_buff(fd,&filesize);
+			res = send_waveform_to_channel(chan,waveform,filesize, intstr);
+			free(waveform);
 			break;
 		}
-	} while (strcmp(ack, "OK\n") != 0);
+		else if (strcmp(ack,"LP\n") == 0) {   /* receive an s-expr */
+			ast_log(LOG_DEBUG,"Festival LP command\n");
+			waveform = socket_receive_file_to_buff(fd,&filesize);
+			waveform[filesize]='\0';
+			ast_log(LOG_WARNING,"Festival returned LP : %s\n",waveform);
+			free(waveform);
+		} else if (strcmp(ack,"ER\n") == 0) {    /* server got an error */
+			ast_log(LOG_WARNING,"Festival returned ER\n");
+			res=-1;
+			break;
+    		}
+	} while (strcmp(ack,"OK\n") != 0);
 	close(fd);
-	ast_config_destroy(cfg);
+	ast_destroy(cfg);
+	LOCAL_USER_REMOVE(u);                                                                                
 	return res;
+
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
-	struct ast_flags config_flags = { 0 };
-	struct ast_config *cfg = ast_config_load(FESTIVAL_CONFIG, config_flags);
-	if (!cfg) {
-		ast_log(LOG_WARNING, "No such configuration file %s\n", FESTIVAL_CONFIG);
-		return AST_MODULE_LOAD_DECLINE;
-	}
-	ast_config_destroy(cfg);
+	
 	return ast_register_application(app, festival_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Simple Festival Interface");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_flash.c

@@ -1,67 +1,58 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * App to flash a zap trunk
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief App to flash a DAHDI trunk
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
+ * the GNU General Public License
  */
  
-/*** MODULEINFO
-	<depend>dahdi</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/image.h>
+#include <asterisk/options.h>
 #include <sys/ioctl.h>
-#include <dahdi/user.h>
+#ifdef __linux__
+#include <linux/zaptel.h>
+#else
+#include <zaptel.h>
+#endif /* __linux__ */
+#include <string.h>
+#include <errno.h>
+#include <stdlib.h>
 
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/translate.h"
-#include "asterisk/image.h"
+static char *tdesc = "Flash zap trunk application";
 
 static char *app = "Flash";
 
-static char *synopsis = "Flashes a DAHDI Trunk";
+static char *synopsis = "Flashes a Zap Trunk";
 
 static char *descrip = 
-"Performs a flash on a DAHDI trunk.  This can be used\n"
-"to access features provided on an incoming analogue circuit\n"
-"such as conference and call waiting. Use with SendDTMF() to\n"
-"perform external transfers\n";
+"  Flash(): Sends a flash on a zap trunk.  This is only a hack for\n"
+"people who want to perform transfers and such via AGI and is generally\n"
+"quite useless otherwise.  Returns 0 on success or -1 if this is not\n"
+"a zap trunk\n";
 
+STANDARD_LOCAL_USER;
 
-static inline int dahdi_wait_event(int fd)
+LOCAL_USER_DECL;
+
+static inline int zt_wait_event(int fd)
 {
-	/* Avoid the silly dahdi_waitevent which ignores a bunch of events */
+	/* Avoid the silly zt_waitevent which ignores a bunch of events */
 	int i,j=0;
-	i = DAHDI_IOMUX_SIGEVENT;
-	if (ioctl(fd, DAHDI_IOMUX, &i) == -1) return -1;
-	if (ioctl(fd, DAHDI_GETEVENT, &j) == -1) return -1;
+	i = ZT_IOMUX_SIGEVENT;
+	if (ioctl(fd, ZT_IOMUX, &i) == -1) return -1;
+	if (ioctl(fd, ZT_GETEVENT, &j) == -1) return -1;
 	return j;
 }
 
@@ -69,45 +60,60 @@ static int flash_exec(struct ast_channel *chan, void *data)
 {
 	int res = -1;
 	int x;
-	struct dahdi_params dahdip;
-
-	if (strcasecmp(chan->tech->type, "DAHDI")) {
-		ast_log(LOG_WARNING, "%s is not a DAHDI channel\n", chan->name);
-		return -1;
-	}
-	
-	memset(&dahdip, 0, sizeof(dahdip));
-	res = ioctl(chan->fds[0], DAHDI_GET_PARAMS, &dahdip);
-	if (!res) {
-		if (dahdip.sigtype & __DAHDI_SIG_FXS) {
-			x = DAHDI_FLASH;
-			res = ioctl(chan->fds[0], DAHDI_HOOK, &x);
-			if (!res || (errno == EINPROGRESS)) {
-				if (res) {
-					/* Wait for the event to finish */
-					dahdi_wait_event(chan->fds[0]);
-				}
-				res = ast_safe_sleep(chan, 1000);
-				ast_verb(3, "Flashed channel %s\n", chan->name);
+	struct localuser *u;
+	struct zt_params ztp;
+	LOCAL_USER_ADD(u);
+	if (!strcasecmp(chan->type, "Zap")) {
+		memset(&ztp, 0, sizeof(ztp));
+		res = ioctl(chan->fds[0], ZT_GET_PARAMS, &ztp);
+		if (!res) {
+			if (ztp.sigtype & __ZT_SIG_FXS) {
+				x = ZT_FLASH;
+				res = ioctl(chan->fds[0], ZT_HOOK, &x);
+				if (!res || (errno == EINPROGRESS)) {
+					if (res) {
+						/* Wait for the event to finish */
+						zt_wait_event(chan->fds[0]);
+					}
+					res = ast_safe_sleep(chan, 1000);
+					if (option_verbose > 2)
+						ast_verbose(VERBOSE_PREFIX_3 "Flashed channel %s\n", chan->name);
+				} else
+					ast_log(LOG_WARNING, "Unable to flash channel %s: %s\n", chan->name, strerror(errno));
 			} else
-				ast_log(LOG_WARNING, "Unable to flash channel %s: %s\n", chan->name, strerror(errno));
+				ast_log(LOG_WARNING, "%s is not an FXO Channel\n", chan->name);
 		} else
-			ast_log(LOG_WARNING, "%s is not an FXO Channel\n", chan->name);
+			ast_log(LOG_WARNING, "Unable to get parameters of %s: %s\n", chan->name, strerror(errno));
 	} else
-		ast_log(LOG_WARNING, "Unable to get parameters of %s: %s\n", chan->name, strerror(errno));
-
+		ast_log(LOG_WARNING, "%s is not a Zap channel\n", chan->name);
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, flash_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Flash channel application");
+char *description(void)
+{
+	return tdesc;
+}
 
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_forkcdr.c

@@ -1,251 +1,90 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Anthony Minessale anthmct@yahoo.com
+ * Fork CDR application
+ * Copyright Anthony Minessale anthmct@yahoo.com
  * Development of this app Sponsered/Funded  by TAAN Softworks Corp
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Fork CDR application
- *
- * \author Anthony Minessale anthmct@yahoo.com
- *
- * \note Development of this app Sponsored/Funded by TAAN Softworks Corp
  * 
- * \ingroup applications
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
  */
 
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/cdr.h"
-#include "asterisk/app.h"
-#include "asterisk/module.h"
+#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([options]):  Causes the Call Data Record to fork an additional\n"
-"cdr record starting from the time of the fork call. This new cdr record will\n"
-"be linked to end of the list of cdr records attached to the channel. The original CDR\n"
-"has a LOCKED flag set, which forces most cdr operations to skip it, except\n"
-"for the functions that set the answer and end times, which ignore the LOCKED\n"
-"flag. This allows all the cdr records in the channel to be 'ended' together\n"
-"when the channel is closed.\n"
-"The CDR() func (when setting CDR values) normally ignores the LOCKED flag also,\n"
-"but has options to vary its behavior. The 'T' option (described below), can\n"
-"override this behavior, but beware the risks.\n"
-"\n"
-"Detailed Behavior Description:\n"
-"First, this app finds the last cdr record in the list, and makes\n"
-"a copy of it. This new copy will be the newly forked cdr record.\n"
-"Next, this new record is linked to the end of the cdr record list.\n"
-"Next, The new cdr record is RESET (unless you use an option to prevent this)\n"
-"This means that:\n"
-"   1. All flags are unset on the cdr record\n"
-"   2. the start, end, and answer times are all set to zero.\n"
-"   3. the billsec and duration fields are set to zero.\n"
-"   4. the start time is set to the current time.\n"
-"   5. the disposition is set to NULL.\n"
-"Next, unless you specified the 'v' option, all variables will be\n"
-"removed from the original cdr record. Thus, the 'v' option allows\n"
-"any CDR variables to be replicated to all new forked cdr records.\n"
-"Without the 'v' option, the variables on the original are effectively\n"
-"moved to the new forked cdr record.\n"
-"Next, if the 's' option is set, the provided variable and value\n"
-"are set on the original cdr record.\n"
-"Next, if the 'a' option is given, and the original cdr record has an\n"
-"answer time set, then the new forked cdr record will have its answer\n"
-"time set to its start time. If the old answer time were carried forward,\n"
-"the answer time would be earlier than the start time, giving strange\n"
-"duration and billsec times.\n"
-"Next, if the 'd' option was specified, the disposition is copied from\n"
-"the original cdr record to the new forked cdr.\n"
-"Next, if the 'D' option was specified, the destination channel field\n"
-"in the new forked CDR is erased.\n"
-"Next, if the 'e' option was specified, the 'end' time for the original\n"
-"cdr record is set to the current time. Future hang-up or ending events\n"
-"will not override this time stamp.\n"
-"Next, If the 'A' option is specified, the original cdr record will have\n"
-"it ANS_LOCKED flag set, which prevent future answer events\n"
-"from updating the original cdr record's disposition. Normally, an\n"
-"'ANSWERED' event would mark all cdr records in the chain as 'ANSWERED'.\n"
-"Next, if the 'T' option is specified, the original cdr record will have\n"
-"its 'DONT_TOUCH' flag set, which will force the cdr_answer, cdr_end, and\n"
-"cdr_setvar functions to leave that cdr record alone.\n"
-"And, last but not least, the original cdr record has its LOCKED flag\n"
-"set. Almost all internal CDR functions (except for the funcs that set\n"
-"the end, and answer times, and set a variable) will honor this flag\n"
-"and leave a LOCKED cdr record alone.\n"
-"This means that the newly created forked cdr record will be affected\n"
-"by events transpiring within Asterisk, with the previously noted\n"
-"exceptions.\n"
-"  Options:\n"
-"    a - update the answer time on the NEW CDR just after it's been inited.\n"
-"         The new CDR may have been answered already. The reset that forkcdr\n"
-"         does will erase the answer time. This will bring it back, but\n"
-"         the answer time will be a copy of the fork/start time. It will\n"
-"         only do this if the initial cdr was indeed already answered.\n"
-"    A - Lock the original CDR against the answer time being updated.\n"
-"         This will allow the disposition on the original CDR to remain the same.\n"
-"    d - Copy the disposition forward from the old cdr, after the \n"
-"         init.\n"
-"    D - Clear the dstchannel on the new CDR after reset.\n"
-"    e - end the original CDR. Do this after all the neccessary data\n"
-"         is copied from the original CDR to the new forked CDR.\n"
-"    R -  do NOT reset the new cdr.\n"
-"    s(name=val) - Set the CDR var 'name' in the original CDR, with value\n"
-"                  'val'.\n"
-"    T -  Mark the original CDR with a DONT_TOUCH flag. setvar, answer, and end\n"
-"          cdr funcs will obey this flag; normally they don't honor the LOCKED\n"
-"          flag set on the original CDR record.\n"
-"          Beware-- using this flag may cause CDR's not to have their end times\n"
-"          updated! It is suggested that if you specify this flag, you might\n"
-"          wish to use the 'e' flag as well!\n"
-"    v  - When the new CDR is forked, it gets a copy of the vars attached\n"
-"         to the current CDR. The vars attached to the original CDR are removed\n"
-"         unless this option is specified.\n";
+"  ForkCDR():  Causes the Call Data Record to fork an additional\n"
+	"cdr record starting from the time of the fork call\n";
 
 
-enum {
-	OPT_SETANS =            (1 << 0),
-	OPT_SETDISP =           (1 << 1),
-	OPT_RESETDEST =         (1 << 2),
-	OPT_ENDCDR =            (1 << 3),
-	OPT_NORESET =           (1 << 4),
-	OPT_KEEPVARS =          (1 << 5),
-	OPT_VARSET =            (1 << 6),
-	OPT_ANSLOCK =           (1 << 7),
-	OPT_DONTOUCH =          (1 << 8),
-};
+STANDARD_LOCAL_USER;
 
-enum {
-	OPT_ARG_VARSET = 0,
-	/* note: this entry _MUST_ be the last one in the enum */
-	OPT_ARG_ARRAY_SIZE,
-};
+LOCAL_USER_DECL;
 
-AST_APP_OPTIONS(forkcdr_exec_options, {
-	AST_APP_OPTION('a', OPT_SETANS),
-	AST_APP_OPTION('A', OPT_ANSLOCK),
-	AST_APP_OPTION('d', OPT_SETDISP),
-	AST_APP_OPTION('D', OPT_RESETDEST),
-	AST_APP_OPTION('e', OPT_ENDCDR),
-	AST_APP_OPTION('R', OPT_NORESET),
-	AST_APP_OPTION_ARG('s', OPT_VARSET, OPT_ARG_VARSET),
-	AST_APP_OPTION('T', OPT_DONTOUCH),
-	AST_APP_OPTION('v', OPT_KEEPVARS),
-});
 
-static void ast_cdr_fork(struct ast_channel *chan, struct ast_flags optflags, char *set) 
-{
-	struct ast_cdr *cdr;
-	struct ast_cdr *newcdr;
-	struct ast_flags flags = { AST_CDR_FLAG_KEEP_VARS };
+static void ast_cdr_clone(struct ast_cdr *cdr) {
+	struct ast_cdr *newcdr = ast_cdr_alloc();
+	memcpy(newcdr,cdr,sizeof(struct ast_cdr));
+	ast_cdr_append(cdr,newcdr);
+	gettimeofday(&newcdr->start, NULL);
+	memset(&newcdr->answer, 0, sizeof(newcdr->answer));
+	ast_cdr_add_flag(cdr,AST_CDR_FLAG_CHILD|AST_CDR_FLAG_LOCKED);
+}
 
-	cdr = chan->cdr;
-
-	while (cdr->next)
-		cdr = cdr->next;
-	
-	if (!(newcdr = ast_cdr_dup(cdr)))
-		return;
-	
-	ast_cdr_append(cdr, newcdr);
-
-	if (!ast_test_flag(&optflags, OPT_NORESET))
-		ast_cdr_reset(newcdr, &flags);
-		
-	if (!ast_test_flag(cdr, AST_CDR_FLAG_KEEP_VARS))
-		ast_cdr_free_vars(cdr, 0);
-	
-	if (!ast_strlen_zero(set)) {
-		char *varname = ast_strdupa(set), *varval;
-		varval = strchr(varname,'=');
-		if (varval) {
-			*varval = 0;
-			varval++;
-			ast_cdr_setvar(cdr, varname, varval, 0);
-		}
+static void ast_cdr_fork(struct ast_channel *chan) {
+	if(chan && chan->cdr) {
+		ast_cdr_clone(chan->cdr);
 	}
-	
-	if (ast_test_flag(&optflags, OPT_SETANS) && !ast_tvzero(cdr->answer))
-		newcdr->answer = newcdr->start;
-
-	if (ast_test_flag(&optflags, OPT_SETDISP))
-		newcdr->disposition = cdr->disposition;
-	
-	if (ast_test_flag(&optflags, OPT_RESETDEST))
-		newcdr->dstchannel[0] = 0;
-	
-	if (ast_test_flag(&optflags, OPT_ENDCDR))
-		ast_cdr_end(cdr);
-
-	if (ast_test_flag(&optflags, OPT_ANSLOCK))
-		ast_set_flag(cdr, AST_CDR_FLAG_ANSLOCKED);
-	
-	if (ast_test_flag(&optflags, OPT_DONTOUCH))
-		ast_set_flag(cdr, AST_CDR_FLAG_DONT_TOUCH);
-		
-	ast_set_flag(cdr, AST_CDR_FLAG_CHILD | AST_CDR_FLAG_LOCKED);
 }
 
 static int forkcdr_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
-	char *argcopy = NULL;
-	struct ast_flags flags = {0};
-	char *opts[OPT_ARG_ARRAY_SIZE];
-	AST_DECLARE_APP_ARGS(arglist,
-		AST_APP_ARG(options);
-	);
+	int res=0;
+	struct localuser *u;
+	LOCAL_USER_ADD(u);
 
-	if (!chan->cdr) {
-		ast_log(LOG_WARNING, "Channel does not have a CDR\n");
-		return 0;
-	}
-
-	argcopy = ast_strdupa(data);
-
-	AST_STANDARD_APP_ARGS(arglist, argcopy);
-
-	opts[OPT_ARG_VARSET] = 0;
-
-	if (!ast_strlen_zero(arglist.options))
-		ast_app_parse_options(forkcdr_exec_options, &flags, opts, arglist.options);
-	
-	if (!ast_strlen_zero(data))
-		ast_set2_flag(chan->cdr, ast_test_flag(&flags, OPT_KEEPVARS), AST_CDR_FLAG_KEEP_VARS);
-	
-	ast_cdr_fork(chan, flags, opts[OPT_ARG_VARSET]);
+	ast_cdr_fork(chan);
 
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, forkcdr_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Fork The CDR into 2 separate entities");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_getcpeid.c

@@ -1,49 +1,43 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Execute arbitrary system commands
+ * 
+ * Copyright (C) 1999-2005, Digium
  *
  * Mark Spencer <markster@digium.com>
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
 
-/*! \file
- *
- * \brief Get ADSI CPE ID
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
- */
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/adsi.h>
+#include <asterisk/options.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
 
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/adsi.h"
+static char *tdesc = "Get ADSI CPE ID";
 
 static char *app = "GetCPEID";
 
 static char *synopsis = "Get ADSI CPE ID";
 
 static char *descrip =
-"  GetCPEID(): Obtains and displays ADSI CPE ID and other information in order\n"
-"to properly setup dahdi.conf for on-hook operations.\n";
+"  GetCPEID: Obtains and displays ADSI CPE ID and other information in order\n"
+"to properly setup zapata.conf for on-hook operations.\n"
+"Returns -1 on hangup only.\n";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int cpeid_setstatus(struct ast_channel *chan, char *stuff[], int voice)
 {
@@ -53,54 +47,60 @@ static int cpeid_setstatus(struct ast_channel *chan, char *stuff[], int voice)
 	for (x=0;x<4;x++)
 		tmp[x] = stuff[x];
 	tmp[4] = NULL;
-	return ast_adsi_print(chan, tmp, justify, voice);
+	return adsi_print(chan, tmp, justify, voice);
 }
 
 static int cpeid_exec(struct ast_channel *chan, void *idata)
 {
 	int res=0;
+	struct localuser *u;
 	unsigned char cpeid[4];
 	int gotgeometry = 0;
 	int gotcpeid = 0;
 	int width, height, buttons;
-	char *data[4];
-	unsigned int x;
+	char data[4][80];
+	char *stuff[4];
 
-	for (x = 0; x < 4; x++)
-		data[x] = alloca(80);
-
-	strcpy(data[0], "** CPE Info **");
-	strcpy(data[1], "Identifying CPE...");
-	strcpy(data[2], "Please wait...");
-	res = ast_adsi_load_session(chan, NULL, 0, 1);
+	LOCAL_USER_ADD(u);
+	stuff[0] = data[0];
+	stuff[1] = data[1];
+	stuff[2] = data[2];
+	stuff[3] = data[3];
+	memset(data, 0, sizeof(data));
+	strncpy(stuff[0], "** CPE Info **", sizeof(data[0]) - 1);
+	strncpy(stuff[1], "Identifying CPE...", sizeof(data[1]) - 1);
+	strncpy(stuff[2], "Please wait...", sizeof(data[2]) - 1);
+	res = adsi_load_session(chan, NULL, 0, 1);
 	if (res > 0) {
-		cpeid_setstatus(chan, data, 0);
-		res = ast_adsi_get_cpeid(chan, cpeid, 0);
+		cpeid_setstatus(chan, stuff, 0);
+		res = adsi_get_cpeid(chan, cpeid, 0);
 		if (res > 0) {
 			gotcpeid = 1;
-			ast_verb(3, "Got CPEID of '%02x:%02x:%02x:%02x' on '%s'\n", cpeid[0], cpeid[1], cpeid[2], cpeid[3], chan->name);
+			if (option_verbose > 2)
+				ast_verbose(VERBOSE_PREFIX_3 "Got CPEID of '%02x:%02x:%02x:%02x' on '%s'\n", cpeid[0], cpeid[1], cpeid[2], cpeid[3], chan->name);
 		}
 		if (res > -1) {
-			strcpy(data[1], "Measuring CPE...");
-			strcpy(data[2], "Please wait...");
-			cpeid_setstatus(chan, data, 0);
-			res = ast_adsi_get_cpeinfo(chan, &width, &height, &buttons, 0);
+			strncpy(stuff[1], "Measuring CPE...", sizeof(data[1]) - 1);
+			strncpy(stuff[2], "Please wait...", sizeof(data[2]) - 1);
+			cpeid_setstatus(chan, stuff, 0);
+			res = adsi_get_cpeinfo(chan, &width, &height, &buttons, 0);
 			if (res > -1) {
-				ast_verb(3, "CPE has %d lines, %d columns, and %d buttons on '%s'\n", height, width, buttons, chan->name);
+				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "CPE has %d lines, %d columns, and %d buttons on '%s'\n", height, width, buttons, chan->name);
 				gotgeometry = 1;
 			}
 		}
 		if (res > -1) {
 			if (gotcpeid)
-				snprintf(data[1], 80, "CPEID: %02x:%02x:%02x:%02x", cpeid[0], cpeid[1], cpeid[2], cpeid[3]);
+				snprintf(stuff[1], sizeof(data[1]), "CPEID: %02x:%02x:%02x:%02x", cpeid[0], cpeid[1], cpeid[2], cpeid[3]);
 			else
-				strcpy(data[1], "CPEID Unknown");
+				strncpy(stuff[1], "CPEID Unknown", sizeof(data[1]) - 1);
 			if (gotgeometry) 
-				snprintf(data[2], 80, "Geom: %dx%d, %d buttons", width, height, buttons);
+				snprintf(stuff[2], sizeof(data[2]), "Geom: %dx%d, %d buttons", width, height, buttons);
 			else
-				strcpy(data[2], "Geometry unknown");
-			strcpy(data[3], "Press # to exit");
-			cpeid_setstatus(chan, data, 1);
+				strncpy(stuff[2], "Geometry unknown", sizeof(data[2]) - 1);
+			strncpy(stuff[3], "Press # to exit", sizeof(data[3]) - 1);
+			cpeid_setstatus(chan, stuff, 1);
 			for(;;) {
 				res = ast_waitfordigit(chan, 1000);
 				if (res < 0)
@@ -110,21 +110,37 @@ static int cpeid_exec(struct ast_channel *chan, void *idata)
 					break;
 				}
 			}
-			ast_adsi_unload_session(chan);
+			adsi_unload_session(chan);
 		}
 	}
-
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, cpeid_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Get ADSI CPE ID");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_groupcount.c

@@ -0,0 +1,224 @@
+/*
+ * 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

@@ -1,58 +1,38 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Stream to an icecast server via ICES (see contrib/asterisk-ices.xml)
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Stream to an icecast server via ICES (see contrib/asterisk-ices.xml)
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \extref ICES - http://www.icecast.org/ices.php
- *
- * \ingroup applications
+ * the GNU General Public License
  */
  
-/*** MODULEINFO
-	<depend>working_fork</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/frame.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <string.h>
+#include <stdio.h>
 #include <signal.h>
+#include <stdlib.h>
+#include <unistd.h>
 #include <fcntl.h>
 #include <sys/time.h>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
+#include <errno.h>
+#include "../astconf.h"
 
-#include "asterisk/paths.h"	/* use ast_config_AST_CONFIG_DIR */
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/frame.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/translate.h"
+#define ICES "/usr/bin/ices"
+#define LOCAL_ICES "/usr/local/bin/ices"
 
-#define path_BIN "/usr/bin/"
-#define path_LOCAL "/usr/local/bin/"
+static char *tdesc = "Encode and Stream via icecast and ices";
 
 static char *app = "ICES";
 
@@ -61,75 +41,41 @@ static char *synopsis = "Encode and stream using 'ices'";
 static char *descrip = 
 "  ICES(config.xml) Streams to an icecast server using ices\n"
 "(available separately).  A configuration file must be supplied\n"
-"for ices (see contrib/asterisk-ices.xml). \n"
-"\n"
-"- ICES version 2 cient and server required.\n";
+"for ices (see examples/asterisk-ices.conf).  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;
-	sigset_t fullset, oldset;
-#ifdef HAVE_CAP
-	cap_t cap;
-#endif
-
-	sigfillset(&fullset);
-	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
-
 	res = fork();
 	if (res < 0) 
 		ast_log(LOG_WARNING, "Fork failed\n");
-	if (res) {
-		pthread_sigmask(SIG_SETMASK, &oldset, NULL);
+	if (res)
 		return res;
-	}
-
-	/* Stop ignoring PIPE */
-	signal(SIGPIPE, SIG_DFL);
-	pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
-
-#ifdef HAVE_CAP
-	cap = cap_from_text("cap_net_admin-eip");
-
-	if (cap_set_proc(cap)) {
-		/* Careful with order! Logging cannot happen after we close FDs */
-		ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-	}
-	cap_free(cap);
-#endif
-
-	if (ast_opt_high_priority)
-		ast_set_priority(0);
 	dup2(fd, STDIN_FILENO);
-	for (x=STDERR_FILENO + 1;x<1024;x++) {
+	for (x=STDERR_FILENO + 1;x<256;x++) {
 		if ((x != STDIN_FILENO) && (x != STDOUT_FILENO))
 			close(x);
 	}
-
-	/* Most commonly installed in /usr/local/bin 
-	 * But many places has it in /usr/bin 
-	 * As a last-ditch effort, try to use PATH
-	 */
-	execl(path_LOCAL "ices2", "ices", filename, (char *)NULL);
-	execl(path_BIN "ices2", "ices", filename, (char *)NULL);
-	execlp("ices2", "ices", filename, (char *)NULL);
-
-	ast_debug(1, "Couldn't find ices version 2, attempting to use ices version 1.");
-
-	execl(path_LOCAL "ices", "ices", filename, (char *)NULL);
-	execl(path_BIN "ices", "ices", filename, (char *)NULL);
+	/* Most commonly installed in /usr/local/bin */
+	execl(ICES, "ices", filename, (char *)NULL);
+	/* But many places has it in /usr/bin */
+	execl(LOCAL_ICES, "ices", filename, (char *)NULL);
+	/* As a last-ditch effort, try to use PATH */
 	execlp("ices", "ices", filename, (char *)NULL);
-
-	ast_log(LOG_WARNING, "Execute of ices failed, could not find command.\n");
-	close(fd);
-	_exit(0);
+	ast_log(LOG_WARNING, "Execute of ices failed\n");
+	return -1;
 }
 
 static int ices_exec(struct ast_channel *chan, void *data)
 {
-	int res = 0;
+	int res=0;
+	struct localuser *u;
 	int fds[2];
 	int ms = -1;
 	int pid = -1;
@@ -139,14 +85,12 @@ static int ices_exec(struct ast_channel *chan, void *data)
 	struct ast_frame *f;
 	char filename[256]="";
 	char *c;
-
-	if (ast_strlen_zero(data)) {
+	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;
 	}
-	
-	last = ast_tv(0, 0);
-	
 	if (pipe(fds)) {
 		ast_log(LOG_WARNING, "Unable to create pipe\n");
 		return -1;
@@ -154,6 +98,7 @@ static int ices_exec(struct ast_channel *chan, void *data)
 	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)
@@ -175,27 +120,28 @@ static int ices_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 	if (((char *)data)[0] == '/')
-		ast_copy_string(filename, (char *) data, sizeof(filename));
+		strncpy(filename, (char *)data, sizeof(filename) - 1);
 	else
-		snprintf(filename, sizeof(filename), "%s/%s", ast_config_AST_CONFIG_DIR, (char *)data);
+		snprintf(filename, sizeof(filename), "%s/%s", (char *)ast_config_AST_CONFIG_DIR, (char *)data);
 	/* Placeholder for options */		
 	c = strchr(filename, '|');
 	if (c)
 		*c = '\0';	
 	res = icesencode(filename, fds[0]);
+	close(fds[0]);
 	if (res >= 0) {
 		pid = res;
 		for (;;) {
 			/* Wait for audio, and stream */
 			ms = ast_waitfor(chan, -1);
 			if (ms < 0) {
-				ast_debug(1, "Hangup detected\n");
+				ast_log(LOG_DEBUG, "Hangup detected\n");
 				res = -1;
 				break;
 			}
 			f = ast_read(chan);
 			if (!f) {
-				ast_debug(1, "Null frame == hangup() detected\n");
+				ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
 				res = -1;
 				break;
 			}
@@ -205,7 +151,6 @@ static int ices_exec(struct ast_channel *chan, void *data)
 					if (errno != EAGAIN) {
 						ast_log(LOG_WARNING, "Write failed to pipe: %s\n", strerror(errno));
 						res = -1;
-						ast_frfree(f);
 						break;
 					}
 				}
@@ -213,25 +158,39 @@ static int ices_exec(struct ast_channel *chan, void *data)
 			ast_frfree(f);
 		}
 	}
-	close(fds[0]);
 	close(fds[1]);
-	
+	LOCAL_USER_REMOVE(u);
 	if (pid > -1)
 		kill(pid, SIGKILL);
 	if (!res && oreadformat)
 		ast_set_read_format(chan, oreadformat);
-
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, ices_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Encode and Stream via icecast and ices");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_image.c

@@ -1,80 +1,89 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * App to transmit an image
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief App to transmit an image
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
+ * the GNU General Public License
  */
  
-#include "asterisk.h"
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/image.h>
+#include <string.h>
+#include <stdlib.h>
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/image.h"
+static char *tdesc = "Image Transmission Application";
 
 static char *app = "SendImage";
 
 static char *synopsis = "Send an image file";
 
 static char *descrip = 
-"  SendImage(filename): Sends an image on a channel.\n"
-"If the channel supports image transport but the image send fails, the channel\n"
-"will be hung up.  Otherwise, the dialplan continues execution.  This\n"
-"application sets the following channel variable upon completion:\n"
-"   SENDIMAGESTATUS  The status is the result of the attempt, one of:\n"
-"                    OK | NOSUPPORT \n";			
+"  SendImage(filename): Sends an image on a channel. If the channel\n"
+"does not support  image transport, and there exists  a  step  with\n"
+"priority n + 101, then  execution  will  continue  at  that  step.\n"
+"Otherwise,  execution  will continue at  the  next priority level.\n"
+"SendImage only  returns  0 if  the  image was sent correctly or if\n"
+"the channel does not support image transport, and -1 otherwise.\n";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int sendimage_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
-
-	if (ast_strlen_zero(data)) {
+	struct localuser *u;
+	if (!data || !strlen((char *)data)) {
 		ast_log(LOG_WARNING, "SendImage requires an argument (filename)\n");
 		return -1;
 	}
-
+	LOCAL_USER_ADD(u);
 	if (!ast_supports_images(chan)) {
 		/* Does not support transport */
-		pbx_builtin_setvar_helper(chan, "SENDIMAGESTATUS", "NOSUPPORT");
+		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+			chan->priority += 100;
 		return 0;
 	}
-
-	if (!(res = ast_send_image(chan, data)))
-		pbx_builtin_setvar_helper(chan, "SENDIMAGESTATUS", "OK");
-		
+	res = ast_send_image(chan, data);
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, sendimage_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Image Transmission Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_intercom.c

@@ -0,0 +1,207 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Use /dev/dsp as an intercom.
+ * 
+ * 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/frame.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <unistd.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+#include <string.h>
+#include <stdlib.h>
+#include <sys/time.h>
+#include <netinet/in.h>
+
+#if defined(__linux__)
+#include <linux/soundcard.h>
+#elif defined(__FreeBSD__)
+#include <sys/soundcard.h>
+#else
+#include <soundcard.h>
+#endif
+
+#ifdef __OpenBSD__
+#define DEV_DSP "/dev/audio"
+#else
+#define DEV_DSP "/dev/dsp"
+#endif
+
+/* Number of 32 byte buffers -- each buffer is 2 ms */
+#define BUFFER_SIZE 32
+
+static char *tdesc = "Intercom using /dev/dsp for output";
+
+static char *app = "Intercom";
+
+static char *synopsis = "(Obsolete) Send to Intercom";
+static char *descrip = 
+"  Intercom(): Sends the user to the intercom (i.e. /dev/dsp).  This program\n"
+"is generally considered  obselete by the chan_oss module.  Returns 0 if the\n"
+"user exits with a DTMF tone, or -1 if they hangup.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+AST_MUTEX_DEFINE_STATIC(sound_lock);
+static int sound = -1;
+
+static int write_audio(short *data, int len)
+{
+	int res;
+	struct audio_buf_info info;
+	ast_mutex_lock(&sound_lock);
+	if (sound < 0) {
+		ast_log(LOG_WARNING, "Sound device closed?\n");
+		ast_mutex_unlock(&sound_lock);
+		return -1;
+	}
+    if (ioctl(sound, SNDCTL_DSP_GETOSPACE, &info)) {
+		ast_log(LOG_WARNING, "Unable to read output space\n");
+		ast_mutex_unlock(&sound_lock);
+        return -1;
+    }
+	res = write(sound, data, len);
+	ast_mutex_unlock(&sound_lock);
+	return res;
+}
+
+static int create_audio(void)
+{
+	int fmt, desired, res, fd;
+	fd = open(DEV_DSP, O_WRONLY);
+	if (fd < 0) {
+		ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
+		close(fd);
+		return -1;
+	}
+	fmt = AFMT_S16_LE;
+	res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
+		close(fd);
+		return -1;
+	}
+	fmt = 0;
+	res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+		close(fd);
+		return -1;
+	}
+	/* 8000 Hz desired */
+	desired = 8000;
+	fmt = desired;
+	res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+		close(fd);
+		return -1;
+	}
+	if (fmt != desired) {
+		ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
+	}
+#if 1
+	/* 2 bytes * 15 units of 2^5 = 32 bytes per buffer */
+	fmt = ((BUFFER_SIZE) << 16) | (0x0005);
+	res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
+	}
+#endif
+	sound = fd;
+	return 0;
+}
+
+static int intercom_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u;
+	struct ast_frame *f;
+	int oreadformat;
+	LOCAL_USER_ADD(u);
+	/* Remember original read format */
+	oreadformat = chan->readformat;
+	/* Set mode to signed linear */
+	res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set format to signed linear on channel %s\n", chan->name);
+		return -1;
+	}
+	/* Read packets from the channel */
+	while(!res) {
+		res = ast_waitfor(chan, -1);
+		if (res > 0) {
+			res = 0;
+			f = ast_read(chan);
+			if (f) {
+				if (f->frametype == AST_FRAME_DTMF) {
+					ast_frfree(f);
+					break;
+				} else {
+					if (f->frametype == AST_FRAME_VOICE) {
+						if (f->subclass == AST_FORMAT_SLINEAR) {
+							res = write_audio(f->data, f->datalen);
+							if (res > 0)
+								res = 0;
+						} else
+							ast_log(LOG_DEBUG, "Unable to handle non-signed linear frame (%d)\n", f->subclass);
+					} 
+				}
+				ast_frfree(f);
+			} else
+				res = -1;
+		}
+	}
+	LOCAL_USER_REMOVE(u);
+	if (!res)
+		ast_set_read_format(chan, oreadformat);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	if (sound > -1)
+		close(sound);
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	if (create_audio())
+		return -1;
+	return ast_register_application(app, intercom_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_ivrdemo.c

@@ -1,115 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2005, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief IVR Demo application
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
- */
-
-/*** MODULEINFO
-	<defaultenabled>no</defaultenabled>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/lock.h"
-#include "asterisk/app.h"
-
-static char *tdesc = "IVR Demo Application";
-static char *app = "IVRDemo";
-static char *synopsis = 
-"  This is a skeleton application that shows you the basic structure to create your\n"
-"own asterisk applications and demonstrates the IVR demo.\n";
-
-static int ivr_demo_func(struct ast_channel *chan, void *data)
-{
-	ast_verbose("IVR Demo, data is %s!\n", (char *)data);
-	return 0;
-}
-
-AST_IVR_DECLARE_MENU(ivr_submenu, "IVR Demo Sub Menu", 0, 
-{
-	{ "s", AST_ACTION_BACKGROUND, "demo-abouttotry" },
-	{ "s", AST_ACTION_WAITOPTION },
-	{ "1", AST_ACTION_PLAYBACK, "digits/1" },
-	{ "1", AST_ACTION_PLAYBACK, "digits/1" },
-	{ "1", AST_ACTION_RESTART },
-	{ "2", AST_ACTION_PLAYLIST, "digits/2;digits/3" },
-	{ "3", AST_ACTION_CALLBACK, ivr_demo_func },
-	{ "4", AST_ACTION_TRANSFER, "demo|s|1" },
-	{ "*", AST_ACTION_REPEAT },
-	{ "#", AST_ACTION_UPONE  },
-	{ NULL }
-});
-
-AST_IVR_DECLARE_MENU(ivr_demo, "IVR Demo Main Menu", 0, 
-{
-	{ "s", AST_ACTION_BACKGROUND, "demo-congrats" },
-	{ "g", AST_ACTION_BACKGROUND, "demo-instruct" },
-	{ "g", AST_ACTION_WAITOPTION },
-	{ "1", AST_ACTION_PLAYBACK, "digits/1" },
-	{ "1", AST_ACTION_RESTART },
-	{ "2", AST_ACTION_MENU, &ivr_submenu },
-	{ "2", AST_ACTION_RESTART },
-	{ "i", AST_ACTION_PLAYBACK, "invalid" },
-	{ "i", AST_ACTION_REPEAT, (void *)(unsigned long)2 },
-	{ "#", AST_ACTION_EXIT },
-	{ NULL },
-});
-
-
-static int skel_exec(struct ast_channel *chan, void *data)
-{
-	int res=0;
-	
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "skel requires an argument (filename)\n");
-		return -1;
-	}
-	
-	/* Do our thing here */
-
-	if (chan->_state != AST_STATE_UP)
-		res = ast_answer(chan);
-	if (!res)
-		res = ast_ivr_menu_run(chan, &ivr_demo, data);
-	
-	return res;
-}
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app);
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app, skel_exec, tdesc, synopsis);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "IVR Demo Application");

apps/app_jack.c

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

apps/app_lookupblacklist.c

@@ -0,0 +1,114 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * App to lookup the callerid number, and see if it is blacklisted
+ * 
+ * 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/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/image.h>
+#include <asterisk/callerid.h>
+#include <asterisk/astdb.h>
+#include <string.h>
+#include <stdlib.h>
+
+static char *tdesc = "Look up Caller*ID name/number from blacklist database";
+
+static char *app = "LookupBlacklist";
+
+static char *synopsis = "Look up Caller*ID name/number from blacklist database";
+
+static char *descrip =
+  "  LookupBlacklist: Looks up the Caller*ID number on the active\n"
+  "channel in the Asterisk database (family 'blacklist').  If the\n"
+  "number is found, and if there exists a priority n + 101,\n"
+  "where 'n' is the priority of the current instance, then  the\n"
+  "channel  will  be  setup  to continue at that priority level.\n"
+  "Otherwise, it returns 0.  Does nothing if no Caller*ID was received on the\n"
+  "channel.\n"
+  "Example: database put blacklist <name/number> 1\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int
+lookupblacklist_exec (struct ast_channel *chan, void *data)
+{
+	char old_cid[144] = "", *num, *name;
+	char blacklist[1];
+	char shrunknum[64] = "";
+	struct localuser *u;
+	int bl = 0;
+
+	LOCAL_USER_ADD (u);
+	if (chan->callerid)
+	{
+		strncpy (old_cid, chan->callerid, sizeof (old_cid) - 1);
+		ast_callerid_parse (old_cid, &name, &num);
+		if (num)
+			strncpy (shrunknum, num, sizeof (shrunknum) - 1);
+		else
+			num = shrunknum;
+		
+		ast_shrink_phone_number (shrunknum);
+		if (!ast_db_get ("blacklist", shrunknum, blacklist, sizeof (blacklist)))
+		{
+			if (option_verbose > 2)
+				ast_log(LOG_NOTICE, "Blacklisted number %s found\n",shrunknum);
+			bl = 1;
+		}
+		else if (!ast_db_get ("blacklist", name, blacklist, sizeof (blacklist)))
+		{
+			if (option_verbose > 2)
+				ast_log (LOG_NOTICE,"Blacklisted name \"%s\" found\n",name);
+			bl = 1;
+		}
+	}
+	
+	if (bl && ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+		chan->priority+=100;
+	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, lookupblacklist_exec, synopsis,descrip);
+}
+
+char *description (void)
+{
+	return tdesc;
+}
+
+int usecount (void)
+{
+	int res;
+	STANDARD_USECOUNT (res);
+	return res;
+}
+
+char *key ()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_lookupcidname.c

@@ -0,0 +1,111 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * App to set callerid name from database, based on directory number
+ * 
+ * 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/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/image.h>
+#include <asterisk/callerid.h>
+#include <asterisk/astdb.h>
+#include <string.h>
+#include <stdlib.h>
+
+static char *tdesc = "Look up CallerID Name from local database";
+
+static char *app = "LookupCIDName";
+
+static char *synopsis = "Look up CallerID Name from local database";
+
+static char *descrip =
+  "  LookupCIDName: Looks up the Caller*ID number on the active\n"
+  "channel in the Asterisk database (family 'cidname') and sets the\n"
+  "Caller*ID name.  Does nothing if no Caller*ID was received on the\n"
+  "channel.  This is useful if you do not subscribe to Caller*ID\n"
+  "name delivery, or if you want to change the names on some incoming\n"
+  "calls.  Always returns 0.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int
+lookupcidname_exec (struct ast_channel *chan, void *data)
+{
+  char old_cid[144] = "", *num, *name;
+  char new_cid[144];
+  char dbname[64];
+  char shrunknum[64] = "";
+  struct localuser *u;
+
+  LOCAL_USER_ADD (u);
+  if (chan->callerid)
+    {
+      strncpy (old_cid, chan->callerid, sizeof (old_cid) - 1);
+      ast_callerid_parse (old_cid, &name, &num);	/* this destroys the original string */
+      if (num)			/* It's possible to get an empty number */
+	strncpy (shrunknum, num, sizeof (shrunknum) - 1);
+      else
+	num = shrunknum;
+      ast_shrink_phone_number (shrunknum);
+      if (!ast_db_get ("cidname", shrunknum, dbname, sizeof (dbname)))
+	{
+	  snprintf (new_cid, sizeof (new_cid), "\"%s\" <%s>", dbname, num);
+	  ast_set_callerid (chan, new_cid, 0);
+	  if (option_verbose > 2)
+	    ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID to %s\n",
+			 new_cid);
+	}
+
+    }
+  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, lookupcidname_exec, synopsis,
+				   descrip);
+}
+
+char *
+description (void)
+{
+  return tdesc;
+}
+
+int
+usecount (void)
+{
+  int res;
+  STANDARD_USECOUNT (res);
+  return res;
+}
+
+char *
+key ()
+{
+  return ASTERISK_GPL_KEY;
+}

apps/app_macro.c

@@ -1,561 +1,253 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Macro Implementation
+ * 
+ * Copyright (C) 2003, Digium
  *
  * Mark Spencer <markster@digium.com>
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
 
-/*! \file
- *
- * \brief Dial plan macro Implementation
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/config.h"
-#include "asterisk/utils.h"
-#include "asterisk/lock.h"
+#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>
 
 #define MAX_ARGS 80
 
-/* special result value used to force macro exit */
-#define MACRO_EXIT_RESULT 1024
-
-#define WAITEXTENWARNING "Use of the application WaitExten within a macro will not function as expected.\n" \
-	"Please use the Read application in order to read DTMF from a channel currently\n" \
-	"executing a macro.\n"
+static char *tdesc = "Extension Macros";
 
 static char *descrip =
-"  Macro(macroname,arg1,arg2...): Executes a macro using the context\n"
+"  Macro(macroname|arg1|arg2...): Executes a macro using the context\n"
 "'macro-<macroname>', jumping to the 's' extension of that context and\n"
-"executing each step, then returning when the steps end. \n"
-"The calling extension, context, and priority are stored in ${MACRO_EXTEN},\n"
+"executing each step, then returning when the steps end.  The calling\n"
+"extension, context, and priority are stored in ${MACRO_EXTEN}, \n"
 "${MACRO_CONTEXT} and ${MACRO_PRIORITY} respectively.  Arguments become\n"
-"${ARG1}, ${ARG2}, etc in the macro context.\n"
-"If you Goto out of the Macro context, the Macro will terminate and control\n"
-"will be returned at the location of the Goto.\n"
-"If ${MACRO_OFFSET} is set at termination, Macro will attempt to continue\n"
-"at priority MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.\n"
-"WARNING: Because of the way Macro is implemented (it executes the priorities\n"
-"         contained within it via sub-engine), and a fixed per-thread memory\n"
-"         stack allowance, macros are limited to 7 levels of nesting (macro\n"
-"         calling macro calling macro, etc.). It may be possible that\n"
-"         stack-intensive applications in deeply nested macros could cause\n"
-"         Asterisk to crash earlier than this limit. It is advised that if you\n"
-"         need to deeply nest macro calls, that you use the Gosub application\n"
-"         (now allows arguments like a Macro) with explict Return() calls\n"
-"         instead.\n"
-WAITEXTENWARNING;
-
-static char *if_descrip =
-"  MacroIf(<expr>?macroname_a[,arg1][:macroname_b[,arg1]])\n"
-"Executes macro defined in <macroname_a> if <expr> is true\n"
-"(otherwise <macroname_b> if provided)\n"
-"Arguments and return values as in application Macro()\n"
-WAITEXTENWARNING;
-
-static char *exclusive_descrip =
-"  MacroExclusive(macroname,arg1,arg2...):\n"
-"Executes macro defined in the context 'macro-macroname'\n"
-"Only one call at a time may run the macro.\n"
-"(we'll wait if another call is busy executing in the Macro)\n"
-"Arguments and return values as in application Macro()\n"
-WAITEXTENWARNING;
-
-static char *exit_descrip =
-"  MacroExit():\n"
-"Causes the currently running macro to exit as if it had\n"
-"ended normally by running out of priorities to execute.\n"
-"If used outside a macro, will likely cause unexpected\n"
-"behavior.\n";
+"${ARG1}, ${ARG2}, etc in the macro context.  Macro returns -1 if\n"
+"any step in the macro returns -1, and 0 otherwise.  If you Goto out\n"
+"of the Macro context, the Macro will terminate and control will be return\n"
+"at the location of the Goto.  Otherwise if ${MACRO_OFFSET} is set at\n"
+"termination, Macro will attempt to continue at priority\n"
+"MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.\n";
 
 static char *app = "Macro";
-static char *if_app = "MacroIf";
-static char *exclusive_app = "MacroExclusive";
-static char *exit_app = "MacroExit";
 
 static char *synopsis = "Macro Implementation";
-static char *if_synopsis = "Conditional Macro Implementation";
-static char *exclusive_synopsis = "Exclusive Macro Implementation";
-static char *exit_synopsis = "Exit From Macro";
 
-static void macro_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan);
+STANDARD_LOCAL_USER;
 
-struct ast_datastore_info macro_ds_info = {
-	.type = "MACRO",
-	.chan_fixup = macro_fixup,
-};
-
-static void macro_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan)
-{
-	int i;
-	char varname[10];
-	pbx_builtin_setvar_helper(new_chan, "MACRO_DEPTH", "0");
-	pbx_builtin_setvar_helper(new_chan, "MACRO_CONTEXT", NULL);
-	pbx_builtin_setvar_helper(new_chan, "MACRO_EXTEN", NULL);
-	pbx_builtin_setvar_helper(new_chan, "MACRO_PRIORITY", NULL);
-	pbx_builtin_setvar_helper(new_chan, "MACRO_OFFSET", NULL);
-	for (i = 1; i < 100; i++) {
-		snprintf(varname, sizeof(varname), "ARG%d", i);
-		while (pbx_builtin_getvar_helper(new_chan, varname)) {
-			/* Kill all levels of arguments */
-			pbx_builtin_setvar_helper(new_chan, varname, NULL);
-		}
-	}
-}
-
-static struct ast_exten *find_matching_priority(struct ast_context *c, const char *exten, int priority, const char *callerid)
-{
-	struct ast_exten *e;
-	struct ast_include *i;
-	struct ast_context *c2;
-
-	for (e=ast_walk_context_extensions(c, NULL); e; e=ast_walk_context_extensions(c, e)) {
-		if (ast_extension_match(ast_get_extension_name(e), exten)) {
-			int needmatch = ast_get_extension_matchcid(e);
-			if ((needmatch && ast_extension_match(ast_get_extension_cidmatch(e), callerid)) ||
-				(!needmatch)) {
-				/* This is the matching extension we want */
-				struct ast_exten *p;
-				for (p=ast_walk_extension_priorities(e, NULL); p; p=ast_walk_extension_priorities(e, p)) {
-					if (priority != ast_get_extension_priority(p))
-						continue;
-					return p;
-				}
-			}
-		}
-	}
-
-	/* No match; run through includes */
-	for (i=ast_walk_context_includes(c, NULL); i; i=ast_walk_context_includes(c, i)) {
-		for (c2=ast_walk_contexts(NULL); c2; c2=ast_walk_contexts(c2)) {
-			if (!strcmp(ast_get_context_name(c2), ast_get_include_name(i))) {
-				e = find_matching_priority(c2, exten, priority, callerid);
-				if (e)
-					return e;
-			}
-		}
-	}
-	return NULL;
-}
-
-static int _macro_exec(struct ast_channel *chan, void *data, int exclusive)
-{
-	const char *s;
-	char *tmp;
-	char *cur, *rest;
-	char *macro;
-	char fullmacro[80];
-	char varname[80];
-	char runningapp[80], runningdata[1024];
-	char *oldargs[MAX_ARGS + 1] = { NULL, };
-	int argc, x;
-	int res=0;
-	char oldexten[256]="";
-	int oldpriority, gosub_level = 0;
-	char pc[80], depthc[12];
-	char oldcontext[AST_MAX_CONTEXT] = "";
-	const char *inhangupc;
-	int offset, depth = 0, maxdepth = 7;
-	int setmacrocontext=0;
-	int autoloopflag, inhangup = 0;
-  
-	char *save_macro_exten;
-	char *save_macro_context;
-	char *save_macro_priority;
-	char *save_macro_offset;
-	struct ast_datastore *macro_store = ast_channel_datastore_find(chan, &macro_ds_info, NULL);
- 
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "Macro() requires arguments. See \"core show application macro\" for help.\n");
-		return -1;
-	}
-
-	do {
-		if (macro_store) {
-			break;
-		}
-		if (!(macro_store = ast_channel_datastore_alloc(&macro_ds_info, NULL))) {
-			ast_log(LOG_WARNING, "Unable to allocate new datastore.\n");
-			break;
-		}
-		/* Just the existence of this datastore is enough. */
-		macro_store->inheritance = DATASTORE_INHERIT_FOREVER;
-		ast_channel_datastore_add(chan, macro_store);
-	} while (0);
-
-	/* does the user want a deeper rabbit hole? */
-	s = pbx_builtin_getvar_helper(chan, "MACRO_RECURSION");
-	if (s)
-		sscanf(s, "%30d", &maxdepth);
-
-	/* Count how many levels deep the rabbit hole goes */
-	s = pbx_builtin_getvar_helper(chan, "MACRO_DEPTH");
-	if (s)
-		sscanf(s, "%30d", &depth);
-	/* Used for detecting whether to return when a Macro is called from another Macro after hangup */
-	if (strcmp(chan->exten, "h") == 0)
-		pbx_builtin_setvar_helper(chan, "MACRO_IN_HANGUP", "1");
-	inhangupc = pbx_builtin_getvar_helper(chan, "MACRO_IN_HANGUP");
-	if (!ast_strlen_zero(inhangupc))
-		sscanf(inhangupc, "%30d", &inhangup);
-
-	if (depth >= maxdepth) {
-		ast_log(LOG_ERROR, "Macro():  possible infinite loop detected.  Returning early.\n");
-		return 0;
-	}
-	snprintf(depthc, sizeof(depthc), "%d", depth + 1);
-	pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
-
-	tmp = ast_strdupa(data);
-	rest = tmp;
-	macro = strsep(&rest, ",");
-	if (ast_strlen_zero(macro)) {
-		ast_log(LOG_WARNING, "Invalid macro name specified\n");
-		return 0;
-	}
-
-	snprintf(fullmacro, sizeof(fullmacro), "macro-%s", macro);
-	if (!ast_exists_extension(chan, fullmacro, "s", 1, chan->cid.cid_num)) {
-		if (!ast_context_find(fullmacro)) 
-			ast_log(LOG_WARNING, "No such context '%s' for macro '%s'\n", fullmacro, macro);
-		else
-			ast_log(LOG_WARNING, "Context '%s' for macro '%s' lacks 's' extension, priority 1\n", fullmacro, macro);
-		return 0;
-	}
-
-	/* If we are to run the macro exclusively, take the mutex */
-	if (exclusive) {
-		ast_debug(1, "Locking macrolock for '%s'\n", fullmacro);
-		ast_autoservice_start(chan);
-		if (ast_context_lockmacro(fullmacro)) {
-			ast_log(LOG_WARNING, "Failed to lock macro '%s' as in-use\n", fullmacro);
-			ast_autoservice_stop(chan);
-			return 0;
-		}
-		ast_autoservice_stop(chan);
-	}
-	
-	/* Save old info */
-	oldpriority = chan->priority;
-	ast_copy_string(oldexten, chan->exten, sizeof(oldexten));
-	ast_copy_string(oldcontext, chan->context, sizeof(oldcontext));
-	if (ast_strlen_zero(chan->macrocontext)) {
-		ast_copy_string(chan->macrocontext, chan->context, sizeof(chan->macrocontext));
-		ast_copy_string(chan->macroexten, chan->exten, sizeof(chan->macroexten));
-		chan->macropriority = chan->priority;
-		setmacrocontext=1;
-	}
-	argc = 1;
-	/* Save old macro variables */
-	save_macro_exten = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_EXTEN"));
-	pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", oldexten);
-
-	save_macro_context = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_CONTEXT"));
-	pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", oldcontext);
-
-	save_macro_priority = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_PRIORITY"));
-	snprintf(pc, sizeof(pc), "%d", oldpriority);
-	pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", pc);
-  
-	save_macro_offset = ast_strdup(pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"));
-	pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", NULL);
-
-	/* Setup environment for new run */
-	chan->exten[0] = 's';
-	chan->exten[1] = '\0';
-	ast_copy_string(chan->context, fullmacro, sizeof(chan->context));
-	chan->priority = 1;
-
-	while((cur = strsep(&rest, ",")) && (argc < MAX_ARGS)) {
-		const char *s;
-  		/* Save copy of old arguments if we're overwriting some, otherwise
-	   	let them pass through to the other macro */
-  		snprintf(varname, sizeof(varname), "ARG%d", argc);
-		s = pbx_builtin_getvar_helper(chan, varname);
-		if (s)
-			oldargs[argc] = ast_strdup(s);
-		pbx_builtin_setvar_helper(chan, varname, cur);
-		argc++;
-	}
-	autoloopflag = ast_test_flag(chan, AST_FLAG_IN_AUTOLOOP);
-	ast_set_flag(chan, AST_FLAG_IN_AUTOLOOP);
-	while(ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
-		struct ast_context *c;
-		struct ast_exten *e;
-		int foundx;
-		runningapp[0] = '\0';
-		runningdata[0] = '\0';
-
-		/* What application will execute? */
-		if (ast_rdlock_contexts()) {
-			ast_log(LOG_WARNING, "Failed to lock contexts list\n");
-		} else {
-			for (c = ast_walk_contexts(NULL), e = NULL; c; c = ast_walk_contexts(c)) {
-				if (!strcmp(ast_get_context_name(c), chan->context)) {
-					if (ast_rdlock_context(c)) {
-						ast_log(LOG_WARNING, "Unable to lock context?\n");
-					} else {
-						e = find_matching_priority(c, chan->exten, chan->priority, chan->cid.cid_num);
-						if (e) { /* This will only be undefined for pbx_realtime, which is majorly broken. */
-							ast_copy_string(runningapp, ast_get_extension_app(e), sizeof(runningapp));
-							ast_copy_string(runningdata, ast_get_extension_app_data(e), sizeof(runningdata));
-						}
-						ast_unlock_context(c);
-					}
-					break;
-				}
-			}
-		}
-		ast_unlock_contexts();
-
-		/* Reset the macro depth, if it was changed in the last iteration */
-		pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
-
-		if ((res = ast_spawn_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num, &foundx,1))) {
-			/* Something bad happened, or a hangup has been requested. */
-			if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F')) ||
-		    	(res == '*') || (res == '#')) {
-				/* Just return result as to the previous application as if it had been dialed */
-				ast_debug(1, "Oooh, got something to jump out with ('%c')!\n", res);
-				break;
-			}
-			switch(res) {
-			case MACRO_EXIT_RESULT:
-				res = 0;
-				goto out;
-			default:
-				ast_debug(2, "Spawn extension (%s,%s,%d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
-				ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
-				goto out;
-			}
-		}
-
-		ast_debug(1, "Executed application: %s\n", runningapp);
-
-		if (!strcasecmp(runningapp, "GOSUB")) {
-			gosub_level++;
-			ast_debug(1, "Incrementing gosub_level\n");
-		} else if (!strcasecmp(runningapp, "GOSUBIF")) {
-			char tmp2[1024], *cond, *app, *app2 = tmp2;
-			pbx_substitute_variables_helper(chan, runningdata, tmp2, sizeof(tmp2) - 1);
-			cond = strsep(&app2, "?");
-			app = strsep(&app2, ":");
-			if (pbx_checkcondition(cond)) {
-				if (!ast_strlen_zero(app)) {
-					gosub_level++;
-					ast_debug(1, "Incrementing gosub_level\n");
-				}
-			} else {
-				if (!ast_strlen_zero(app2)) {
-					gosub_level++;
-					ast_debug(1, "Incrementing gosub_level\n");
-				}
-			}
-		} else if (!strcasecmp(runningapp, "RETURN")) {
-			gosub_level--;
-			ast_debug(1, "Decrementing gosub_level\n");
-		} else if (!strcasecmp(runningapp, "STACKPOP")) {
-			gosub_level--;
-			ast_debug(1, "Decrementing gosub_level\n");
-		} else if (!strncasecmp(runningapp, "EXEC", 4)) {
-			/* Must evaluate args to find actual app */
-			char tmp2[1024], *tmp3 = NULL;
-			pbx_substitute_variables_helper(chan, runningdata, tmp2, sizeof(tmp2) - 1);
-			if (!strcasecmp(runningapp, "EXECIF")) {
-				tmp3 = strchr(tmp2, '|');
-				if (tmp3)
-					*tmp3++ = '\0';
-				if (!pbx_checkcondition(tmp2))
-					tmp3 = NULL;
-			} else
-				tmp3 = tmp2;
-
-			if (tmp3)
-				ast_debug(1, "Last app: %s\n", tmp3);
-
-			if (tmp3 && !strncasecmp(tmp3, "GOSUB", 5)) {
-				gosub_level++;
-				ast_debug(1, "Incrementing gosub_level\n");
-			} else if (tmp3 && !strncasecmp(tmp3, "RETURN", 6)) {
-				gosub_level--;
-				ast_debug(1, "Decrementing gosub_level\n");
-			} else if (tmp3 && !strncasecmp(tmp3, "STACKPOP", 8)) {
-				gosub_level--;
-				ast_debug(1, "Decrementing gosub_level\n");
-			}
-		}
-
-		if (gosub_level == 0 && strcasecmp(chan->context, fullmacro)) {
-			ast_verb(2, "Channel '%s' jumping out of macro '%s'\n", chan->name, macro);
-			break;
-		}
-
-		/* don't stop executing extensions when we're in "h" */
-		if (ast_check_hangup(chan) && !inhangup) {
-			ast_debug(1, "Extension %s, macroexten %s, priority %d returned normally even though call was hung up\n", chan->exten, chan->macroexten, chan->priority);
-			goto out;
-		}
-		chan->priority++;
-  	}
-	out:
-
-	/* Don't let the channel change now. */
-	ast_channel_lock(chan);
-
-	/* Reset the depth back to what it was when the routine was entered (like if we called Macro recursively) */
-	snprintf(depthc, sizeof(depthc), "%d", depth);
-	pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
-	ast_set2_flag(chan, autoloopflag, AST_FLAG_IN_AUTOLOOP);
-
-  	for (x = 1; x < argc; x++) {
-  		/* Restore old arguments and delete ours */
-		snprintf(varname, sizeof(varname), "ARG%d", x);
-  		if (oldargs[x]) {
-			pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
-			ast_free(oldargs[x]);
-		} else {
-			pbx_builtin_setvar_helper(chan, varname, NULL);
-		}
-  	}
-
-	/* Restore macro variables */
-	pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
-	pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", save_macro_context);
-	pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
-	if (save_macro_exten)
-		ast_free(save_macro_exten);
-	if (save_macro_context)
-		ast_free(save_macro_context);
-	if (save_macro_priority)
-		ast_free(save_macro_priority);
-
-	if (setmacrocontext) {
-		chan->macrocontext[0] = '\0';
-		chan->macroexten[0] = '\0';
-		chan->macropriority = 0;
-	}
-
-	if (!strcasecmp(chan->context, fullmacro)) {
-  		/* If we're leaving the macro normally, restore original information */
-		chan->priority = oldpriority;
-		ast_copy_string(chan->context, oldcontext, sizeof(chan->context));
-		if (!(ast_check_hangup(chan) & AST_SOFTHANGUP_ASYNCGOTO)) {
-			/* Copy the extension, so long as we're not in softhangup, where we could be given an asyncgoto */
-			const char *offsets;
-			ast_copy_string(chan->exten, oldexten, sizeof(chan->exten));
-			if ((offsets = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"))) {
-				/* Handle macro offset if it's set by checking the availability of step n + offset + 1, otherwise continue
-			   	normally if there is any problem */
-				if (sscanf(offsets, "%30d", &offset) == 1) {
-					if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + offset + 1, chan->cid.cid_num)) {
-						chan->priority += offset;
-					}
-				}
-			}
-		}
-	}
-
-	pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
-	if (save_macro_offset)
-		ast_free(save_macro_offset);
-
-	/* Unlock the macro */
-	if (exclusive) {
-		ast_debug(1, "Unlocking macrolock for '%s'\n", fullmacro);
-		if (ast_context_unlockmacro(fullmacro)) {
-			ast_log(LOG_ERROR, "Failed to unlock macro '%s' - that isn't good\n", fullmacro);
-			res = 0;
-		}
-	}
-	ast_channel_unlock(chan);
-
-	return res;
-}
+LOCAL_USER_DECL;
 
 static int macro_exec(struct ast_channel *chan, void *data)
 {
-	return _macro_exec(chan, data, 0);
-}
+  char tmp[256] = "";
+  char *cur, *rest;
+  char *macro;
+  char fullmacro[80];
+  char varname[80];
+  char *oldargs[MAX_ARGS + 1] = { NULL, };
+  int argc, x;
+  int res=0;
+  char oldexten[256]="";
+  int oldpriority;
+  char pc[80];
+  char oldcontext[256] = "";
+  char *offsets;
+  int offset;
+  int setmacrocontext=0;
+  
+  char *save_macro_exten;
+  char *save_macro_context;
+  char *save_macro_priority;
+  char *save_macro_offset;
+  
+  if (!data || ast_strlen_zero(data)) {
+    ast_log(LOG_WARNING, "Invalid Macro incantation\n");
+    return 0;
+  }
+  strncpy(tmp, data, sizeof(tmp) - 1);
+  rest = tmp;
+  macro = strsep(&rest, "|");
+  if (!macro || ast_strlen_zero(macro)) {
+  	ast_log(LOG_WARNING, "Invalid macro name specified\n");
+	return 0;
+  }
+  snprintf(fullmacro, sizeof(fullmacro), "macro-%s", macro);
+  if (!ast_exists_extension(chan, fullmacro, "s", 1, chan->callerid)) {
+  	if (!ast_context_find(fullmacro)) 
+		ast_log(LOG_WARNING, "No such context '%s' for macro '%s'\n", fullmacro, macro);
+	else
+	  	ast_log(LOG_WARNING, "Context '%s' for macro '%s' lacks 's' extension, priority 1\n", fullmacro, macro);
+	return 0;
+  }
+  /* Save old info */
+  oldpriority = chan->priority;
+  strncpy(oldexten, chan->exten, sizeof(oldexten) - 1);
+  strncpy(oldcontext, chan->context, sizeof(oldcontext) - 1);
+  if (ast_strlen_zero(chan->macrocontext)) {
+	strncpy(chan->macrocontext, chan->context, sizeof(chan->macrocontext) - 1);
+	strncpy(chan->macroexten, chan->exten, sizeof(chan->macroexten) - 1);
+	chan->macropriority = chan->priority;
+	setmacrocontext=1;
+  }
+  argc = 1;
+  /* Save old macro variables */
+  save_macro_exten = pbx_builtin_getvar_helper(chan, "MACRO_EXTEN");
+  if (save_macro_exten) save_macro_exten = strdup(save_macro_exten);
+  pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", oldexten);
 
-static int macroexclusive_exec(struct ast_channel *chan, void *data)
-{
-	return _macro_exec(chan, data, 1);
-}
+  save_macro_context = pbx_builtin_getvar_helper(chan, "MACRO_CONTEXT");
+  if (save_macro_context) save_macro_context = strdup(save_macro_context);
+  pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", oldcontext);
 
-static int macroif_exec(struct ast_channel *chan, void *data) 
-{
-	char *expr = NULL, *label_a = NULL, *label_b = NULL;
-	int res = 0;
+  save_macro_priority = pbx_builtin_getvar_helper(chan, "MACRO_PRIORITY");
+  if (save_macro_priority) save_macro_priority = strdup(save_macro_priority);
+  snprintf(pc, sizeof(pc), "%d", oldpriority);
+  pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", pc);
+  
+  save_macro_offset = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET");
+  if (save_macro_offset) save_macro_offset = strdup(save_macro_offset);
+  pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", NULL);
 
-	if (!(expr = ast_strdupa(data)))
-		return -1;
+  /* Setup environment for new run */
+  chan->exten[0] = 's';
+  chan->exten[1] = '\0';
+  strncpy(chan->context, fullmacro, sizeof(chan->context) - 1);
+  chan->priority = 1;
 
-	if ((label_a = strchr(expr, '?'))) {
-		*label_a = '\0';
-		label_a++;
-		if ((label_b = strchr(label_a, ':'))) {
-			*label_b = '\0';
-			label_b++;
+  while((cur = strsep(&rest, "|")) && (argc < MAX_ARGS)) {
+  	/* Save copy of old arguments if we're overwriting some, otherwise
+	   let them pass through to the other macro */
+  	snprintf(varname, sizeof(varname), "ARG%d", argc);
+	oldargs[argc] = pbx_builtin_getvar_helper(chan, varname);
+	if (oldargs[argc])
+		oldargs[argc] = strdup(oldargs[argc]);
+	pbx_builtin_setvar_helper(chan, varname, cur);
+	argc++;
+  }
+  while(ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->callerid)) {
+	if ((res = ast_spawn_extension(chan, chan->context, chan->exten, chan->priority, chan->callerid))) {
+		/* Something bad happened, or a hangup has been requested. */
+		if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F')) ||
+		    (res == '*') || (res == '#')) {
+			/* Just return result as to the previous application as if it had been dialed */
+			ast_log(LOG_DEBUG, "Oooh, got something to jump out with ('%c')!\n", res);
+			break;
 		}
-		if (pbx_checkcondition(expr))
-			res = macro_exec(chan, label_a);
-		else if (label_b) 
-			res = macro_exec(chan, label_b);
-	} else
-		ast_log(LOG_WARNING, "Invalid Syntax.\n");
+		switch(res) {
+		case AST_PBX_KEEPALIVE:
+			if (option_debug)
+				ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited KEEPALIVE in macro %s on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
+			else if (option_verbose > 1)
+				ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited KEEPALIVE in macro '%s' on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
+			goto out;
+			break;
+		default:
+			if (option_debug)
+				ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
+			else if (option_verbose > 1)
+				ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
+			goto out;
+		}
+	}
+	if (strcasecmp(chan->context, fullmacro)) {
+		if (option_verbose > 1)
+			ast_verbose(VERBOSE_PREFIX_2 "Channel '%s' jumping out of macro '%s'\n", chan->name, macro);
+		break;
+	}
+	/* don't stop executing extensions when we're in "h" */
+	if (chan->_softhangup && strcasecmp(oldexten,"h")) {
+		ast_log(LOG_DEBUG, "Extension %s, priority %d returned normally even though call was hung up\n",
+			chan->exten, chan->priority);
+		goto out;
+	}
+	chan->priority++;
+  }
+out:
+  for (x=1;x<argc;x++) {
+  	/* Restore old arguments and delete ours */
+	snprintf(varname, sizeof(varname), "ARG%d", x);
+  	if (oldargs[x]) {
+		pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
+		free(oldargs[x]);
+	} else {
+		pbx_builtin_setvar_helper(chan, varname, NULL);
+	}
+  }
 
-	return res;
+  /* Restore macro variables */
+  pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
+  if (save_macro_exten) free(save_macro_exten);
+  pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", save_macro_context);
+  if (save_macro_context) free(save_macro_context);
+  pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
+  if (save_macro_priority) free(save_macro_priority);
+  if (setmacrocontext) {
+	chan->macrocontext[0] = '\0';
+	chan->macroexten[0] = '\0';
+	chan->macropriority = 0;
+  }
+
+  if (!strcasecmp(chan->context, fullmacro)) {
+  	/* If we're leaving the macro normally, restore original information */
+	chan->priority = oldpriority;
+	strncpy(chan->context, oldcontext, sizeof(chan->context) - 1);
+	if (!(chan->_softhangup & AST_SOFTHANGUP_ASYNCGOTO)) {
+		/* Copy the extension, so long as we're not in softhangup, where we could be given an asyncgoto */
+		strncpy(chan->exten, oldexten, sizeof(chan->exten) - 1);
+		if ((offsets = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"))) {
+			/* Handle macro offset if it's set by checking the availability of step n + offset + 1, otherwise continue
+			   normally if there is any problem */
+			if (sscanf(offsets, "%d", &offset) == 1) {
+				if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + offset + 1, chan->callerid)) {
+					chan->priority += offset;
+				}
+			}
+		}
+	}
+  }
+
+  pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
+  if (save_macro_offset) free(save_macro_offset);
+  return res;
 }
-			
-static int macro_exit_exec(struct ast_channel *chan, void *data)
+
+int unload_module(void)
 {
-	return MACRO_EXIT_RESULT;
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
 }
 
-static int unload_module(void)
+int load_module(void)
+{
+	return ast_register_application(app, macro_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
 {
 	int res;
-
-	res = ast_unregister_application(if_app);
-	res |= ast_unregister_application(exit_app);
-	res |= ast_unregister_application(app);
-	res |= ast_unregister_application(exclusive_app);
-
+	STANDARD_USECOUNT(res);
 	return res;
 }
 
-static int load_module(void)
+char *key()
 {
-	int res;
-
-	res = ast_register_application(exit_app, macro_exit_exec, exit_synopsis, exit_descrip);
-	res |= ast_register_application(if_app, macroif_exec, if_synopsis, if_descrip);
-	res |= ast_register_application(exclusive_app, macroexclusive_exec, exclusive_synopsis, exclusive_descrip);
-	res |= ast_register_application(app, macro_exec, synopsis, descrip);
-
-	return res;
+	return ASTERISK_GPL_KEY;
 }
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Extension Macros");

apps/app_milliwatt.c

@@ -1,162 +1,148 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Digital Milliwatt Test
+ * 
+ * Copyright (C) 2002, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
 
-/*! \file
- *
- * \brief Digital Milliwatt Test
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
- */
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+#include <errno.h>
 
-/*** MODULEINFO
-	<depend>res_indications</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/module.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/indications.h"
+static char *tdesc = "Digital Milliwatt (mu-law) Test Application";
 
 static char *app = "Milliwatt";
 
-static char *synopsis = "Generate a Constant 1004Hz tone at 0dbm (mu-law)";
+static char *synopsis = "Generate a Constant 1000Hz tone at 0dbm (mu-law)";
 
 static char *descrip = 
-"   Milliwatt([options]): Generate a Constant 1004Hz tone at 0dbm.\n"
-"Previous versions of this application generated the tone at 1000Hz.  If for\n"
-"some reason you would prefer that behavior, supply the 'o' option to get the\n"
-"old behavior.\n"
-"";
+"Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law)\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static char digital_milliwatt[] = {0x1e,0x0b,0x0b,0x1e,0x9e,0x8b,0x8b,0x9e} ;
 
 static void *milliwatt_alloc(struct ast_channel *chan, void *params)
 {
-	return ast_calloc(1, sizeof(int));
+int	*indexp;
+	indexp = malloc(sizeof(int));
+	if (indexp == NULL) return(NULL);
+	*indexp = 0;
+	return(indexp);
 }
 
 static void milliwatt_release(struct ast_channel *chan, void *data)
 {
-	ast_free(data);
+	free(data);
 	return;
 }
 
 static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int samples)
 {
-	unsigned char buf[AST_FRIENDLY_OFFSET + 640];
-	const int maxsamples = ARRAY_LEN(buf);
-	int i, *indexp = (int *) data;
-	struct ast_frame wf = {
-		.frametype = AST_FRAME_VOICE,
-		.subclass = AST_FORMAT_ULAW,
-		.offset = AST_FRIENDLY_OFFSET,
-		.data = buf + AST_FRIENDLY_OFFSET,
-		.src = __FUNCTION__,
-	};
+	struct ast_frame wf;
+	unsigned char waste[AST_FRIENDLY_OFFSET];
+	unsigned char buf[640];
+	int i,*indexp = (int *) data;
 
-	/* Instead of len, use samples, because channel.c generator_force
-	* generate(chan, tmp, 0, 160) ignores len. In any case, len is
-	* a multiple of samples, given by number of samples times bytes per
-	* sample. In the case of ulaw, len = samples. for signed linear
-	* len = 2 * samples */
-	if (samples > maxsamples) {
-		ast_log(LOG_WARNING, "Only doing %d samples (%d requested)\n", maxsamples, samples);
-		samples = maxsamples;
+	if (len > sizeof(buf))
+	{
+		ast_log(LOG_WARNING,"Only doing %d bytes (%d bytes requested)\n",(int)sizeof(buf),len);
+		len = sizeof(buf);
 	}
-
-	len = samples * sizeof (buf[0]);
+	waste[0] = 0; /* make compiler happy */
+	wf.frametype = AST_FRAME_VOICE;
+	wf.subclass = AST_FORMAT_ULAW;
+	wf.offset = AST_FRIENDLY_OFFSET;
+	wf.mallocd = 0;
+	wf.data = buf;
 	wf.datalen = len;
-	wf.samples = samples;
-
+	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++) {
-		buf[AST_FRIENDLY_OFFSET + i] = digital_milliwatt[(*indexp)++];
+	for(i = 0; i < len; i++)
+	{
+		buf[i] = digital_milliwatt[(*indexp)++];
 		*indexp &= 7;
 	}
-
-	if (ast_write(chan,&wf) < 0) {
+	if (ast_write(chan,&wf) < 0)
+	{
 		ast_log(LOG_WARNING,"Failed to write frame to '%s': %s\n",chan->name,strerror(errno));
 		return -1;
 	}
-
 	return 0;
 }
 
-static struct ast_generator milliwattgen = {
+static struct ast_generator milliwattgen = 
+{
 	alloc: milliwatt_alloc,
 	release: milliwatt_release,
 	generate: milliwatt_generate,
-};
-
-static int old_milliwatt_exec(struct ast_channel *chan)
-{
-	ast_set_write_format(chan, AST_FORMAT_ULAW);
-	ast_set_read_format(chan, AST_FORMAT_ULAW);
-
-	if (chan->_state != AST_STATE_UP) {
-		ast_answer(chan);
-	}
-
-	if (ast_activate_generator(chan,&milliwattgen,"milliwatt") < 0) {
-		ast_log(LOG_WARNING,"Failed to activate generator on '%s'\n",chan->name);
-		return -1;
-	}
-
-	while (!ast_safe_sleep(chan, 10000))
-		;
-
-	ast_deactivate_generator(chan);
-
-	return -1;
-}
+} ;
 
 static int milliwatt_exec(struct ast_channel *chan, void *data)
 {
-	const char *options = data;
-	int res = -1;
 
-	if (!ast_strlen_zero(options) && strchr(options, 'o')) {
-		return old_milliwatt_exec(chan);
+	struct localuser *u;
+	LOCAL_USER_ADD(u);
+	ast_set_write_format(chan, AST_FORMAT_ULAW);
+	ast_set_read_format(chan, AST_FORMAT_ULAW);
+	if (chan->_state != AST_STATE_UP)
+	{
+		ast_answer(chan);
 	}
-
-	res = ast_playtones_start(chan, 23255, "1004/1000", 0);
-
-	while (!res) {
-		res = ast_safe_sleep(chan, 10000);
+	if (ast_activate_generator(chan,&milliwattgen,"milliwatt") < 0)
+	{
+		ast_log(LOG_WARNING,"Failed to activate generator on '%s'\n",chan->name);
+		return -1;
 	}
-
-	return res;
+	while(!ast_safe_sleep(chan, 10000));
+	ast_deactivate_generator(chan);
+	LOCAL_USER_REMOVE(u);
+	return -1;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, milliwatt_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Digital Milliwatt (mu-law) Test Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_mixmonitor.c

@@ -1,600 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2005, Anthony Minessale II
- * Copyright (C) 2005 - 2006, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- * Kevin P. Fleming <kpfleming@digium.com>
- *
- * Based on app_muxmon.c provided by
- * Anthony Minessale II <anthmct@yahoo.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief MixMonitor() - Record a call and mix the audio during the recording
- * \ingroup applications
- *
- * \author Mark Spencer <markster@digium.com>
- * \author Kevin P. Fleming <kpfleming@digium.com>
- *
- * \note Based on app_muxmon.c provided by
- * Anthony Minessale II <anthmct@yahoo.com>
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/paths.h"	/* use ast_config_AST_MONITOR_DIR */
-#include "asterisk/file.h"
-#include "asterisk/audiohook.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/cli.h"
-#include "asterisk/app.h"
-#include "asterisk/channel.h"
-
-#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
-
-static const char *app = "MixMonitor";
-static const char *synopsis = "Record a call and mix the audio during the recording";
-static const char *desc = ""
-"  MixMonitor(<file>.<ext>[,<options>[,<command>]]):\n"
-"Records the audio on the current channel to the specified file.\n"
-"If the filename is an absolute path, uses that path, otherwise\n"
-"creates the file in the configured monitoring directory from\n"
-"asterisk.conf.  Use of StopMixMonitor is required to guarantee\n"
-"the audio file is available for processing during dialplan execution.\n\n"
-"Valid options:\n"
-" a      - Append to the file instead of overwriting it.\n"
-" b      - Only save audio to the file while the channel is bridged.\n"
-"          Note: Does not include conferences or sounds played to each bridged\n"
-"                party.\n"
-"          Note: If you utilize this option inside a Local channel, you must\n"
-"                 make sure the Local channel is not optimized away. To do this,\n"
-"                 be sure to call your Local channel with the '/n' option.\n"
-"                 For example: Dial(Local/start@mycontext/n)\n"
-" v(<x>) - Adjust the heard volume by a factor of <x> (range -4 to 4)\n"	
-" V(<x>) - Adjust the spoken volume by a factor of <x> (range -4 to 4)\n"	
-" W(<x>) - Adjust the both heard and spoken volumes by a factor of <x>\n"
-"         (range -4 to 4)\n\n"	
-"<command> will be executed when the recording is over\n"
-"Any strings matching ^{X} will be unescaped to ${X}.\n"
-"All variables will be evaluated at the time MixMonitor is called.\n"
-"The variable MIXMONITOR_FILENAME will contain the filename used to record.\n"
-"";
-
-static const char *stop_app = "StopMixMonitor";
-static const char *stop_synopsis = "Stop recording a call through MixMonitor";
-static const char *stop_desc = ""
-"  StopMixMonitor():\n"
-"Stop recording a call through MixMonitor, and free the recording's file handle.\n"
-"";
-
-struct module_symbols *me;
-
-static const char *mixmonitor_spy_type = "MixMonitor";
-
-struct mixmonitor {
-	struct ast_audiohook audiohook;
-	char *filename;
-	char *post_process;
-	char *name;
-	unsigned int flags;
-	struct mixmonitor_ds *mixmonitor_ds;
-};
-
-enum {
-	MUXFLAG_APPEND = (1 << 1),
-	MUXFLAG_BRIDGED = (1 << 2),
-	MUXFLAG_VOLUME = (1 << 3),
-	MUXFLAG_READVOLUME = (1 << 4),
-	MUXFLAG_WRITEVOLUME = (1 << 5),
-} mixmonitor_flags;
-
-enum {
-	OPT_ARG_READVOLUME = 0,
-	OPT_ARG_WRITEVOLUME,
-	OPT_ARG_VOLUME,
-	OPT_ARG_ARRAY_SIZE,
-} mixmonitor_args;
-
-AST_APP_OPTIONS(mixmonitor_opts, {
-	AST_APP_OPTION('a', MUXFLAG_APPEND),
-	AST_APP_OPTION('b', MUXFLAG_BRIDGED),
-	AST_APP_OPTION_ARG('v', MUXFLAG_READVOLUME, OPT_ARG_READVOLUME),
-	AST_APP_OPTION_ARG('V', MUXFLAG_WRITEVOLUME, OPT_ARG_WRITEVOLUME),
-	AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
-});
-
-/* This structure is used as a means of making sure that our pointer to
- * the channel we are monitoring remains valid. This is very similar to 
- * what is used in app_chanspy.c.
- */
-struct mixmonitor_ds {
-	struct ast_channel *chan;
-	/* These condition variables are used to be sure that the channel
-	 * hangup code completes before the mixmonitor thread attempts to
-	 * free this structure. The combination of a bookean flag and a
-	 * ast_cond_t ensure that no matter what order the threads run in,
-	 * we are guaranteed to never have the waiting thread block forever
-	 * in the case that the signaling thread runs first.
-	 */
-	unsigned int destruction_ok;
-	ast_cond_t destruction_condition;
-	ast_mutex_t lock;
-
-	/* The filestream is held in the datastore so it can be stopped
-	 * immediately during stop_mixmonitor or channel destruction. */
-	int fs_quit;
-	struct ast_filestream *fs;
-	struct ast_audiohook *audiohook;
-};
-
- /*!
-  * \internal
-  * \pre mixmonitor_ds must be locked before calling this function
-  */
-static void mixmonitor_ds_close_fs(struct mixmonitor_ds *mixmonitor_ds)
-{
-	if (mixmonitor_ds->fs) {
-		ast_closestream(mixmonitor_ds->fs);
-		mixmonitor_ds->fs = NULL;
-		mixmonitor_ds->fs_quit = 1;
-		ast_verb(2, "MixMonitor close filestream\n");
-	}
-}
-
-static void mixmonitor_ds_destroy(void *data)
-{
-	struct mixmonitor_ds *mixmonitor_ds = data;
-
-	ast_mutex_lock(&mixmonitor_ds->lock);
-	mixmonitor_ds->chan = NULL;
-	mixmonitor_ds->audiohook = NULL;
-	mixmonitor_ds->destruction_ok = 1;
-	ast_cond_signal(&mixmonitor_ds->destruction_condition);
-	ast_mutex_unlock(&mixmonitor_ds->lock);
-}
-
-static void mixmonitor_ds_chan_fixup(void *data, struct ast_channel *old_chan, struct ast_channel *new_chan)
-{
-	struct mixmonitor_ds *mixmonitor_ds = data;
-
-	ast_mutex_lock(&mixmonitor_ds->lock);
-	mixmonitor_ds->chan = new_chan;
-	ast_mutex_unlock(&mixmonitor_ds->lock);
-}
-
-static struct ast_datastore_info mixmonitor_ds_info = {
-	.type = "mixmonitor",
-	.destroy = mixmonitor_ds_destroy,
-	.chan_fixup = mixmonitor_ds_chan_fixup,
-};
-
-static void destroy_monitor_audiohook(struct mixmonitor *mixmonitor)
-{
-	if (mixmonitor->mixmonitor_ds) {
-		ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
-		mixmonitor->mixmonitor_ds->audiohook = NULL;
-		ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
-	}
-	/* kill the audiohook.*/
-	ast_audiohook_lock(&mixmonitor->audiohook);
-	ast_audiohook_detach(&mixmonitor->audiohook);
-	ast_audiohook_unlock(&mixmonitor->audiohook);
-	ast_audiohook_destroy(&mixmonitor->audiohook);
-}
-
-static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook) 
-{
-	struct ast_channel *peer = NULL;
-	int res = 0;
-
-	if (!chan)
-		return -1;
-
-	ast_audiohook_attach(chan, audiohook);
-
-	if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan)))
-		ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);	
-
-	return res;
-}
-
-#define SAMPLES_PER_FRAME 160
-
-static void mixmonitor_free(struct mixmonitor *mixmonitor)
-{
-	if (mixmonitor) {
-		if (mixmonitor->mixmonitor_ds) {
-			ast_mutex_destroy(&mixmonitor->mixmonitor_ds->lock);
-			ast_cond_destroy(&mixmonitor->mixmonitor_ds->destruction_condition);
-			ast_free(mixmonitor->mixmonitor_ds);
-		}
-		ast_free(mixmonitor);
-	}
-}
-
-static void *mixmonitor_thread(void *obj) 
-{
-	struct mixmonitor *mixmonitor = obj;
-	struct ast_filestream **fs = NULL;
-	unsigned int oflags;
-	char *ext;
-	int errflag = 0;
-
-	ast_verb(2, "Begin MixMonitor Recording %s\n", mixmonitor->name);
-
-	fs = &mixmonitor->mixmonitor_ds->fs;
-
-	/* The audiohook must enter and exit the loop locked */
-	ast_audiohook_lock(&mixmonitor->audiohook);
-	while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING && !mixmonitor->mixmonitor_ds->fs_quit) {
-		struct ast_frame *fr = NULL;
-
-		if (!(fr = ast_audiohook_read_frame(&mixmonitor->audiohook, SAMPLES_PER_FRAME, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR))) {
-			ast_audiohook_trigger_wait(&mixmonitor->audiohook);
-
-			if (mixmonitor->audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
-				break;
-			}
-			continue;
-		}
-
-		/* audiohook lock is not required for the next block.
-		 * Unlock it, but remember to lock it before looping or exiting */
-		ast_audiohook_unlock(&mixmonitor->audiohook);
-
-		ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
-		if (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) || (mixmonitor->mixmonitor_ds->chan && ast_bridged_channel(mixmonitor->mixmonitor_ds->chan))) {
-			/* Initialize the file if not already done so */
-			if (!*fs && !errflag && !mixmonitor->mixmonitor_ds->fs_quit) {
-				oflags = O_CREAT | O_WRONLY;
-				oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
-
-				if ((ext = strrchr(mixmonitor->filename, '.')))
-					*(ext++) = '\0';
-				else
-					ext = "raw";
-
-				if (!(*fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) {
-					ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
-					errflag = 1;
-				}
-			}
-
-			/* Write out the frame(s) */
-			if (*fs) {
-				struct ast_frame *cur;
-
-				for (cur = fr; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
-					ast_writestream(*fs, cur);
-				}
-			}
-		}
-		ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
-
-		/* All done! free it. */
-		ast_frame_free(fr, 0);
-		ast_audiohook_lock(&mixmonitor->audiohook);
-	}
-	ast_audiohook_unlock(&mixmonitor->audiohook);
-
-	/* Datastore cleanup.  close the filestream and wait for ds destruction */
-	ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
-	mixmonitor_ds_close_fs(mixmonitor->mixmonitor_ds);
-	if (!mixmonitor->mixmonitor_ds->destruction_ok) {
-		ast_cond_wait(&mixmonitor->mixmonitor_ds->destruction_condition, &mixmonitor->mixmonitor_ds->lock);
-	}
-	ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
-
-	/* kill the audiohook */
-	destroy_monitor_audiohook(mixmonitor);
-
-	if (mixmonitor->post_process) {
-		ast_verb(2, "Executing [%s]\n", mixmonitor->post_process);
-		ast_safe_system(mixmonitor->post_process);
-	}
-
-	ast_verb(2, "End MixMonitor Recording %s\n", mixmonitor->name);
-	mixmonitor_free(mixmonitor);
-	return NULL;
-}
-
-static int setup_mixmonitor_ds(struct mixmonitor *mixmonitor, struct ast_channel *chan)
-{
-	struct ast_datastore *datastore = NULL;
-	struct mixmonitor_ds *mixmonitor_ds;
-
-	if (!(mixmonitor_ds = ast_calloc(1, sizeof(*mixmonitor_ds)))) {
-		return -1;
-	}
-	
-	ast_mutex_init(&mixmonitor_ds->lock);
-	ast_cond_init(&mixmonitor_ds->destruction_condition, NULL);
-
-	if (!(datastore = ast_channel_datastore_alloc(&mixmonitor_ds_info, NULL))) {
-		ast_mutex_destroy(&mixmonitor_ds->lock);
-		ast_cond_destroy(&mixmonitor_ds->destruction_condition);
-		ast_free(mixmonitor_ds);
-		return -1;
-	}
-
-	/* No need to lock mixmonitor_ds since this is still operating in the channel's thread */
-	mixmonitor_ds->chan = chan;
-	mixmonitor_ds->audiohook = &mixmonitor->audiohook;
-	datastore->data = mixmonitor_ds;
-
-	ast_channel_lock(chan);
-	ast_channel_datastore_add(chan, datastore);
-	ast_channel_unlock(chan);
-
-	mixmonitor->mixmonitor_ds = mixmonitor_ds;
-	return 0;
-}
-
-static void launch_monitor_thread(struct ast_channel *chan, const char *filename, unsigned int flags,
-				  int readvol, int writevol, const char *post_process) 
-{
-	pthread_t thread;
-	struct mixmonitor *mixmonitor;
-	char postprocess2[1024] = "";
-	size_t len;
-
-	len = sizeof(*mixmonitor) + strlen(chan->name) + strlen(filename) + 2;
-
-	postprocess2[0] = 0;
-	/* If a post process system command is given attach it to the structure */
-	if (!ast_strlen_zero(post_process)) {
-		char *p1, *p2;
-
-		p1 = ast_strdupa(post_process);
-		for (p2 = p1; *p2 ; p2++) {
-			if (*p2 == '^' && *(p2+1) == '{') {
-				*p2 = '$';
-			}
-		}
-		pbx_substitute_variables_helper(chan, p1, postprocess2, sizeof(postprocess2) - 1);
-		if (!ast_strlen_zero(postprocess2))
-			len += strlen(postprocess2) + 1;
-	}
-
-	/* Pre-allocate mixmonitor structure and spy */
-	if (!(mixmonitor = ast_calloc(1, len))) {
-		return;
-	}
-
-	/* Setup the actual spy before creating our thread */
-	if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type)) {
-		mixmonitor_free(mixmonitor);
-		return;
-	}
-
-	/* Copy over flags and channel name */
-	mixmonitor->flags = flags;
-	if (setup_mixmonitor_ds(mixmonitor, chan)) {
-		mixmonitor_free(mixmonitor);
-		return;
-	}
-	mixmonitor->name = (char *) mixmonitor + sizeof(*mixmonitor);
-	strcpy(mixmonitor->name, chan->name);
-	if (!ast_strlen_zero(postprocess2)) {
-		mixmonitor->post_process = mixmonitor->name + strlen(mixmonitor->name) + strlen(filename) + 2;
-		strcpy(mixmonitor->post_process, postprocess2);
-	}
-
-	mixmonitor->filename = (char *) mixmonitor + sizeof(*mixmonitor) + strlen(chan->name) + 1;
-	strcpy(mixmonitor->filename, filename);
-
-	ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_SYNC);
-
-	if (readvol)
-		mixmonitor->audiohook.options.read_volume = readvol;
-	if (writevol)
-		mixmonitor->audiohook.options.write_volume = writevol;
-
-	if (startmon(chan, &mixmonitor->audiohook)) {
-		ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
-			mixmonitor_spy_type, chan->name);
-		ast_audiohook_destroy(&mixmonitor->audiohook);
-		mixmonitor_free(mixmonitor);
-		return;
-	}
-
-	ast_pthread_create_detached_background(&thread, NULL, mixmonitor_thread, mixmonitor);
-}
-
-static int mixmonitor_exec(struct ast_channel *chan, void *data)
-{
-	int x, readvol = 0, writevol = 0;
-	struct ast_flags flags = {0};
-	char *parse, *tmp, *slash;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(filename);
-		AST_APP_ARG(options);
-		AST_APP_ARG(post_process);
-	);
-	
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
-		return -1;
-	}
-
-	parse = ast_strdupa(data);
-
-	AST_STANDARD_APP_ARGS(args, parse);
-	
-	if (ast_strlen_zero(args.filename)) {
-		ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
-		return -1;
-	}
-
-	if (args.options) {
-		char *opts[OPT_ARG_ARRAY_SIZE] = { NULL, };
-
-		ast_app_parse_options(mixmonitor_opts, &flags, opts, args.options);
-
-		if (ast_test_flag(&flags, MUXFLAG_READVOLUME)) {
-			if (ast_strlen_zero(opts[OPT_ARG_READVOLUME])) {
-				ast_log(LOG_WARNING, "No volume level was provided for the heard volume ('v') option.\n");
-			} else if ((sscanf(opts[OPT_ARG_READVOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
-				ast_log(LOG_NOTICE, "Heard volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_READVOLUME]);
-			} else {
-				readvol = get_volfactor(x);
-			}
-		}
-		
-		if (ast_test_flag(&flags, MUXFLAG_WRITEVOLUME)) {
-			if (ast_strlen_zero(opts[OPT_ARG_WRITEVOLUME])) {
-				ast_log(LOG_WARNING, "No volume level was provided for the spoken volume ('V') option.\n");
-			} else if ((sscanf(opts[OPT_ARG_WRITEVOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
-				ast_log(LOG_NOTICE, "Spoken volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_WRITEVOLUME]);
-			} else {
-				writevol = get_volfactor(x);
-			}
-		}
-		
-		if (ast_test_flag(&flags, MUXFLAG_VOLUME)) {
-			if (ast_strlen_zero(opts[OPT_ARG_VOLUME])) {
-				ast_log(LOG_WARNING, "No volume level was provided for the combined volume ('W') option.\n");
-			} else if ((sscanf(opts[OPT_ARG_VOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
-				ast_log(LOG_NOTICE, "Combined volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_VOLUME]);
-			} else {
-				readvol = writevol = get_volfactor(x);
-			}
-		}
-	}
-
-	/* if not provided an absolute path, use the system-configured monitoring directory */
-	if (args.filename[0] != '/') {
-		char *build;
-
-		build = alloca(strlen(ast_config_AST_MONITOR_DIR) + strlen(args.filename) + 3);
-		sprintf(build, "%s/%s", ast_config_AST_MONITOR_DIR, args.filename);
-		args.filename = build;
-	}
-
-	tmp = ast_strdupa(args.filename);
-	if ((slash = strrchr(tmp, '/')))
-		*slash = '\0';
-	ast_mkdir(tmp, 0777);
-
-	pbx_builtin_setvar_helper(chan, "MIXMONITOR_FILENAME", args.filename);
-	launch_monitor_thread(chan, args.filename, flags.flags, readvol, writevol, args.post_process);
-
-	return 0;
-}
-
-static int stop_mixmonitor_exec(struct ast_channel *chan, void *data)
-{
-	struct ast_datastore *datastore = NULL;
-
-	ast_channel_lock(chan);
-	ast_audiohook_detach_source(chan, mixmonitor_spy_type);
-	if ((datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info, NULL))) {
-		struct mixmonitor_ds *mixmonitor_ds = datastore->data;
-
-		ast_mutex_lock(&mixmonitor_ds->lock);
-
-		/* closing the filestream here guarantees the file is avaliable to the dialplan
-	 	 * after calling StopMixMonitor */
-		mixmonitor_ds_close_fs(mixmonitor_ds);
-
-		/* The mixmonitor thread may be waiting on the audiohook trigger.
-		 * In order to exit from the mixmonitor loop before waiting on channel
-		 * destruction, poke the audiohook trigger. */
-		if (mixmonitor_ds->audiohook) {
-			ast_audiohook_lock(mixmonitor_ds->audiohook);
-			ast_cond_signal(&mixmonitor_ds->audiohook->trigger);
-			ast_audiohook_unlock(mixmonitor_ds->audiohook);
-			mixmonitor_ds->audiohook = NULL;
-		}
-
-		ast_mutex_unlock(&mixmonitor_ds->lock);
-
-		/* Remove the datastore so the monitor thread can exit */
-		if (!ast_channel_datastore_remove(chan, datastore)) {
-			ast_channel_datastore_free(datastore);
-		}
-	}
-	ast_channel_unlock(chan);
-
-	return 0;
-}
-
-static char *handle_cli_mixmonitor(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
-	struct ast_channel *chan;
-
-	switch (cmd) {
-	case CLI_INIT:
-		e->command = "mixmonitor [start|stop]";
-		e->usage =
-			"Usage: mixmonitor <start|stop> <chan_name> [args]\n"
-			"       The optional arguments are passed to the MixMonitor\n"
-			"       application when the 'start' command is used.\n";
-		return NULL;
-	case CLI_GENERATE:
-		return ast_complete_channels(a->line, a->word, a->pos, a->n, 2);
-	}
-
-	if (a->argc < 3)
-		return CLI_SHOWUSAGE;
-
-	if (!(chan = ast_get_channel_by_name_prefix_locked(a->argv[2], strlen(a->argv[2])))) {
-		ast_cli(a->fd, "No channel matching '%s' found.\n", a->argv[2]);
-		/* Technically this is a failure, but we don't want 2 errors printing out */
-		return CLI_SUCCESS;
-	}
-
-	if (!strcasecmp(a->argv[1], "start")) {
-		mixmonitor_exec(chan, a->argv[3]);
-		ast_channel_unlock(chan);
-	} else {
-		ast_channel_unlock(chan);
-		ast_audiohook_detach_source(chan, mixmonitor_spy_type);
-	}
-
-	return CLI_SUCCESS;
-}
-
-static struct ast_cli_entry cli_mixmonitor[] = {
-	AST_CLI_DEFINE(handle_cli_mixmonitor, "Execute a MixMonitor command")
-};
-
-static int unload_module(void)
-{
-	int res;
-
-	ast_cli_unregister_multiple(cli_mixmonitor, sizeof(cli_mixmonitor) / sizeof(struct ast_cli_entry));
-	res = ast_unregister_application(stop_app);
-	res |= ast_unregister_application(app);
-	
-	return res;
-}
-
-static int load_module(void)
-{
-	int res;
-
-	ast_cli_register_multiple(cli_mixmonitor, sizeof(cli_mixmonitor) / sizeof(struct ast_cli_entry));
-	res = ast_register_application(app, mixmonitor_exec, synopsis, desc);
-	res |= ast_register_application(stop_app, stop_mixmonitor_exec, stop_synopsis, stop_desc);
-
-	return res;
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mixed Audio Monitoring Application");

apps/app_morsecode.c

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

apps/app_mp3.c

@@ -1,101 +1,63 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Silly application to play an MP3 file -- uses mpg123
+ * 
+ * Copyright (C) 1999-2004, Digium, Inc.
  *
  * Mark Spencer <markster@digium.com>
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Silly application to play an MP3 file -- uses mpg123
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
+ * the GNU General Public License
  */
  
-/*** MODULEINFO
-	<depend>working_fork</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <sys/time.h>
+#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>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
-
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/frame.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/translate.h"
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.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";
+
 static char *app = "MP3Player";
 
 static char *synopsis = "Play an MP3 file or stream";
 
 static char *descrip = 
-"  MP3Player(location): Executes mpg123 to play the given location,\n"
-"which typically would be a filename or a URL. User can exit by pressing\n"
-"any key on the dialpad, or by hanging up."; 
+"  MP3Player(location) Executes mpg123 to play the given location\n"
+"which typically would be a  filename  or  a URL. Returns  -1  on\n"
+"hangup or 0 otherwise. User can exit by pressing any key\n.";
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int mp3play(char *filename, int fd)
 {
 	int res;
 	int x;
-	sigset_t fullset, oldset;
-#ifdef HAVE_CAP
-	cap_t cap;
-#endif
-
-	sigfillset(&fullset);
-	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
-
 	res = fork();
 	if (res < 0) 
 		ast_log(LOG_WARNING, "Fork failed\n");
-	if (res) {
-		pthread_sigmask(SIG_SETMASK, &oldset, NULL);
+	if (res)
 		return res;
-	}
-#ifdef HAVE_CAP
-	cap = cap_from_text("cap_net_admin-eip");
-
-	if (cap_set_proc(cap)) {
-		/* Careful with order! Logging cannot happen after we close FDs */
-		ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-	}
-	cap_free(cap);
-#endif
-	if (ast_opt_high_priority)
-		ast_set_priority(0);
-	signal(SIGPIPE, SIG_DFL);
-	pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
-
 	dup2(fd, STDOUT_FILENO);
-	for (x=STDERR_FILENO + 1;x<256;x++) {
-		close(x);
+	for (x=0;x<256;x++) {
+		if (x != STDOUT_FILENO)
+			close(x);
 	}
 	/* Execute mpg123, but buffer if it's a net connection */
 	if (!strncasecmp(filename, "http://", 7)) {
@@ -115,7 +77,7 @@ static int mp3play(char *filename, int fd)
 	    execlp("mpg123", "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
 	}
 	ast_log(LOG_WARNING, "Execute of mpg123 failed\n");
-	_exit(0);
+	return -1;
 }
 
 static int timed_read(int fd, void *data, int datalen, int timeout)
@@ -124,7 +86,7 @@ static int timed_read(int fd, void *data, int datalen, int timeout)
 	struct pollfd fds[1];
 	fds[0].fd = fd;
 	fds[0].events = POLLIN;
-	res = ast_poll(fds, 1, timeout);
+	res = poll(fds, 1, timeout);
 	if (res < 1) {
 		ast_log(LOG_NOTICE, "Poll timed out/errored out with %d\n", res);
 		return -1;
@@ -136,31 +98,28 @@ static int timed_read(int fd, void *data, int datalen, int timeout)
 static int mp3_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
+	struct localuser *u;
 	int fds[2];
 	int ms = -1;
 	int pid = -1;
 	int owriteformat;
 	int timeout = 2000;
-	struct timeval next;
+	struct timeval now, next;
 	struct ast_frame *f;
 	struct myframe {
 		struct ast_frame f;
 		char offset[AST_FRIENDLY_OFFSET];
 		short frdata[160];
-	} myf = {
-		.f = { 0, },
-	};
-
-	if (ast_strlen_zero(data)) {
+	} myf;
+	if (!data) {
 		ast_log(LOG_WARNING, "MP3 Playback requires an argument (filename)\n");
 		return -1;
 	}
-
 	if (pipe(fds)) {
 		ast_log(LOG_WARNING, "Unable to create pipe\n");
 		return -1;
 	}
-	
+	LOCAL_USER_ADD(u);
 	ast_stopstream(chan);
 
 	owriteformat = chan->writeformat;
@@ -170,20 +129,35 @@ 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 */
-	next = ast_tvnow();
+	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_tvdiff_ms(next, ast_tvnow());
+			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), timeout);
 				if (res > 0) {
 					myf.f.frametype = AST_FRAME_VOICE;
@@ -201,27 +175,34 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 						break;
 					}
 				} else {
-					ast_debug(1, "No more mp3\n");
+					ast_log(LOG_DEBUG, "No more mp3\n");
 					res = 0;
 					break;
 				}
-				next = ast_tvadd(next, ast_samp2tv(myf.f.samples, 8000));
+				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_debug(1, "Hangup detected\n");
+					ast_log(LOG_DEBUG, "Hangup detected\n");
 					res = -1;
 					break;
 				}
 				if (ms) {
 					f = ast_read(chan);
 					if (!f) {
-						ast_debug(1, "Null frame == hangup() detected\n");
+						ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
 						res = -1;
 						break;
 					}
 					if (f->frametype == AST_FRAME_DTMF) {
-						ast_debug(1, "User pressed a key\n");
+						ast_log(LOG_DEBUG, "User pressed a key\n");
 						ast_frfree(f);
 						res = 0;
 						break;
@@ -233,23 +214,38 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 	}
 	close(fds[0]);
 	close(fds[1]);
-	
+	LOCAL_USER_REMOVE(u);
 	if (pid > -1)
 		kill(pid, SIGKILL);
 	if (!res && owriteformat)
 		ast_set_write_format(chan, owriteformat);
-	
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, mp3_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Silly MP3 Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_nbscat.c

@@ -1,106 +1,62 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Silly application to play an NBScat file -- uses nbscat8k
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Silly application to play an NBScat file -- uses nbscat8k
- *
- * \author Mark Spencer <markster@digium.com>
- *  
- * \ingroup applications
+ * the GNU General Public License
  */
  
-/*** MODULEINFO
-	<depend>working_fork</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/frame.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <string.h>
+#include <stdio.h>
+#include <signal.h>
+#include <stdlib.h>
+#include <unistd.h>
 #include <fcntl.h>
 #include <sys/time.h>
 #include <sys/socket.h>
-#include <signal.h>
-#ifdef HAVE_CAP
-#include <sys/capability.h>
-#endif /* HAVE_CAP */
-
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/frame.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/translate.h"
 
 #define LOCAL_NBSCAT "/usr/local/bin/nbscat8k"
 #define NBSCAT "/usr/bin/nbscat8k"
 
-#ifndef AF_LOCAL
-#define AF_LOCAL AF_UNIX
-#endif
+static char *tdesc = "Silly NBS Stream Application";
 
 static char *app = "NBScat";
 
 static char *synopsis = "Play an NBS local stream";
 
 static char *descrip = 
-"  NBScat(): Executes nbscat to listen to the local NBS stream.\n"
-"User can exit by pressing any key.\n";
+"  NBScat: Executes nbscat to listen to the local NBS stream.\n"
+"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;
-	sigset_t fullset, oldset;
-#ifdef HAVE_CAP
-	cap_t cap;
-#endif
-
-	sigfillset(&fullset);
-	pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
-
 	res = fork();
 	if (res < 0) 
 		ast_log(LOG_WARNING, "Fork failed\n");
-	if (res) {
-		pthread_sigmask(SIG_SETMASK, &oldset, NULL);
+	if (res)
 		return res;
-	}
-	signal(SIGPIPE, SIG_DFL);
-	pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
-
-#ifdef HAVE_CAP
-	cap = cap_from_text("cap_net_admin-eip");
-
-	if (cap_set_proc(cap)) {
-		/* Careful with order! Logging cannot happen after we close FDs */
-		ast_log(LOG_WARNING, "Unable to remove capabilities.\n");
-	}
-	cap_free(cap);
-#endif
-	if (ast_opt_high_priority)
-		ast_set_priority(0);
-
 	dup2(fd, STDOUT_FILENO);
-	for (x = STDERR_FILENO + 1; x < 1024; x++) {
+	for (x=0;x<256;x++) {
 		if (x != STDOUT_FILENO)
 			close(x);
 	}
@@ -108,7 +64,7 @@ static int NBScatplay(int fd)
 	execl(NBSCAT, "nbscat8k", "-d", (char *)NULL);
 	execl(LOCAL_NBSCAT, "nbscat8k", "-d", (char *)NULL);
 	ast_log(LOG_WARNING, "Execute of nbscat8k failed\n");
-	_exit(0);
+	return -1;
 }
 
 static int timed_read(int fd, void *data, int datalen)
@@ -117,7 +73,7 @@ static int timed_read(int fd, void *data, int datalen)
 	struct pollfd fds[1];
 	fds[0].fd = fd;
 	fds[0].events = POLLIN;
-	res = ast_poll(fds, 1, 2000);
+	res = poll(fds, 1, 2000);
 	if (res < 1) {
 		ast_log(LOG_NOTICE, "Selected timed out/errored out with %d\n", res);
 		return -1;
@@ -129,23 +85,23 @@ static int timed_read(int fd, void *data, int datalen)
 static int NBScat_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
+	struct localuser *u;
 	int fds[2];
 	int ms = -1;
 	int pid = -1;
 	int owriteformat;
-	struct timeval next;
+	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;
@@ -157,15 +113,29 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 	
 	res = NBScatplay(fds[1]);
 	/* Wait 1000 ms first */
-	next = ast_tvnow();
+	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_tvdiff_ms(next, ast_tvnow());
+			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;
@@ -183,27 +153,34 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 						break;
 					}
 				} else {
-					ast_debug(1, "No more mp3\n");
+					ast_log(LOG_DEBUG, "No more mp3\n");
 					res = 0;
 					break;
 				}
-				next = ast_tvadd(next, ast_samp2tv(myf.f.samples, 8000));
+				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_debug(1, "Hangup detected\n");
+					ast_log(LOG_DEBUG, "Hangup detected\n");
 					res = -1;
 					break;
 				}
 				if (ms) {
 					f = ast_read(chan);
 					if (!f) {
-						ast_debug(1, "Null frame == hangup() detected\n");
+						ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
 						res = -1;
 						break;
 					}
 					if (f->frametype == AST_FRAME_DTMF) {
-						ast_debug(1, "User pressed a key\n");
+						ast_log(LOG_DEBUG, "User pressed a key\n");
 						ast_frfree(f);
 						res = 0;
 						break;
@@ -215,23 +192,38 @@ static int NBScat_exec(struct ast_channel *chan, void *data)
 	}
 	close(fds[0]);
 	close(fds[1]);
-	
+	LOCAL_USER_REMOVE(u);
 	if (pid > -1)
 		kill(pid, SIGKILL);
 	if (!res && owriteformat)
 		ast_set_write_format(chan, owriteformat);
-
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, NBScat_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Silly NBS Stream Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_page.c

@@ -1,213 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (c) 2004 - 2006 Digium, Inc.  All rights reserved.
- *
- * Mark Spencer <markster@digium.com>
- *
- * This code is released under the GNU General Public License
- * version 2.0.  See LICENSE for more information.
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- */
-
-/*! \file
- *
- * \brief page() - Paging application
- *
- * \author Mark Spencer <markster@digium.com>
- *
- * \ingroup applications
- */
-
-/*** MODULEINFO
-	<depend>dahdi</depend>
-	<depend>app_meetme</depend>
- ***/
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/file.h"
-#include "asterisk/app.h"
-#include "asterisk/chanvars.h"
-#include "asterisk/utils.h"
-#include "asterisk/devicestate.h"
-#include "asterisk/dial.h"
-
-static const char *app_page= "Page";
-
-static const char *page_synopsis = "Pages phones";
-
-static const char *page_descrip =
-"Page(Technology/Resource&Technology2/Resource2[,options])\n"
-"  Places outbound calls to the given technology / resource and dumps\n"
-"them into a conference bridge as muted participants.  The original\n"
-"caller is dumped into the conference as a speaker and the room is\n"
-"destroyed when the original caller leaves.  Valid options are:\n"
-"        d - full duplex audio\n"
-"        q - quiet, do not play beep to caller\n"
-"        r - record the page into a file (see 'r' for app_meetme)\n"
-"        s - only dial channel if devicestate says it is not in use\n";
-
-enum {
-	PAGE_DUPLEX = (1 << 0),
-	PAGE_QUIET = (1 << 1),
-	PAGE_RECORD = (1 << 2),
-	PAGE_SKIP = (1 << 3),
-} page_opt_flags;
-
-AST_APP_OPTIONS(page_opts, {
-	AST_APP_OPTION('d', PAGE_DUPLEX),
-	AST_APP_OPTION('q', PAGE_QUIET),
-	AST_APP_OPTION('r', PAGE_RECORD),
-	AST_APP_OPTION('s', PAGE_SKIP),
-});
-
-
-static int page_exec(struct ast_channel *chan, void *data)
-{
-	char *options, *tech, *resource, *tmp, *tmp2;
-	char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0];
-	struct ast_flags flags = { 0 };
-	unsigned int confid = ast_random();
-	struct ast_app *app;
-	int res = 0, pos = 0, i = 0;
-	struct ast_dial **dial_list;
-	unsigned int num_dials;
-
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
-		return -1;
-	}
-
-	if (!(app = pbx_findapp("MeetMe"))) {
-		ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
-		return -1;
-	};
-
-	options = ast_strdupa(data);
-
-	ast_copy_string(originator, chan->name, sizeof(originator));
-	if ((tmp = strchr(originator, '-')))
-		*tmp = '\0';
-
-	tmp = strsep(&options, ",");
-	if (options)
-		ast_app_parse_options(page_opts, &flags, opts, options);
-
-	snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
-		(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
-
-	/* Count number of extensions in list by number of ampersands + 1 */
-	num_dials = 1;
-	tmp2 = tmp;
-	while (*tmp2) {
-		if (*tmp2 == '&') {
-			num_dials++;
-		}
-		tmp2++;
-	}
-
-	if (!(dial_list = ast_calloc(num_dials, sizeof(struct ast_dial *)))) {
-		ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(struct ast_dial *) * num_dials));
-		return -1;
-	}
-
-	/* Go through parsing/calling each device */
-	while ((tech = strsep(&tmp, "&"))) {
-		int state = 0;
-		struct ast_dial *dial = NULL;
-
-		/* don't call the originating device */
-		if (!strcasecmp(tech, originator))
-			continue;
-
-		/* If no resource is available, continue on */
-		if (!(resource = strchr(tech, '/'))) {
-			ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
-			continue;
-		}
-
-		/* Ensure device is not in use if skip option is enabled */
-		if (ast_test_flag(&flags, PAGE_SKIP)) {
-			state = ast_device_state(tech);
-			if (state == AST_DEVICE_UNKNOWN) {
-				ast_log(LOG_WARNING, "Destination '%s' has device state '%s'. Paging anyway.\n", tech, devstate2str(state));
-			} else if (state != AST_DEVICE_NOT_INUSE) {
-				ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, devstate2str(state));
-				continue;
-			}
-		}
-
-		*resource++ = '\0';
-
-		/* Create a dialing structure */
-		if (!(dial = ast_dial_create())) {
-			ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
-			continue;
-		}
-
-		/* Append technology and resource */
-		ast_dial_append(dial, tech, resource);
-
-		/* Set ANSWER_EXEC as global option */
-		ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
-
-		/* Run this dial in async mode */
-		ast_dial_run(dial, chan, 1);
-
-		/* Put in our dialing array */
-		dial_list[pos++] = dial;
-	}
-
-	if (!ast_test_flag(&flags, PAGE_QUIET)) {
-		res = ast_streamfile(chan, "beep", chan->language);
-		if (!res)
-			res = ast_waitstream(chan, "");
-	}
-
-	if (!res) {
-		snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
-			(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
-		pbx_exec(chan, app, meetmeopts);
-	}
-
-	/* Go through each dial attempt cancelling, joining, and destroying */
-	for (i = 0; i < pos; i++) {
-		struct ast_dial *dial = dial_list[i];
-
-		/* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
-		ast_dial_join(dial);
-
-		/* Hangup all channels */
-		ast_dial_hangup(dial);
-
-		/* Destroy dialing structure */
-		ast_dial_destroy(dial);
-	}
-
-	return -1;
-}
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app_page);
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
-

apps/app_parkandannounce.c

@@ -1,144 +1,185 @@
 /*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2006, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
+ * Asterisk -- A telephony toolkit for Linux.
  *
+ * ParkAndAnnounce application for Asterisk
  * Author: Ben Miller <bgmiller@dccinc.com>
  *    With TONS of help from Mark!
+ * 
+ * Asterisk is Copyrighted as follows
+ * Copyright (C) 1999, Mark Spencer
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
 
-/*! \file
- *
- * \brief ParkAndAnnounce application for Asterisk
- *
- * \author Ben Miller <bgmiller@dccinc.com>
- * \arg With TONS of help from Mark!
- *
- * \ingroup applications
- */
+#include <sys/types.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/channel_pvt.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/features.h>
+#include <asterisk/options.h>
+#include <asterisk/logger.h>
+#include <asterisk/say.h>
+#include <asterisk/lock.h>
 
-#include "asterisk.h"
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/features.h"
-#include "asterisk/say.h"
-#include "asterisk/lock.h"
-#include "asterisk/utils.h"
-#include "asterisk/app.h"
+static char *tdesc = "Call Parking and Announce Application";
 
 static char *app = "ParkAndAnnounce";
 
 static char *synopsis = "Park and Announce";
 
 static char *descrip =
-"  ParkAndAnnounce(announce:template,timeout,dial[,return_context]):\n"
-"Park a call into the parkinglot and announce the call to another channel.\n"
-"\n"
-"announce template: Colon-separated list of files to announce.  The word PARKED\n"
-"                   will be replaced by a say_digits of the extension in which\n"
-"                   the call is parked.\n"
-"timeout:           Time in seconds before the call returns into the return\n"
-"                   context.\n"
-"dial:              The app_dial style resource to call to make the\n"
-"                   announcement.  Console/dsp calls the console.\n"
-"return_context:    The goto-style label to jump the call back into after\n"
-"                   timeout.  Default <priority+1>.\n"
-"\n"
-"The variable ${PARKEDAT} will contain the parking extension into which the\n"
-"call was placed.  Use with the Local channel to allow the dialplan to make\n"
-"use of this information.\n";
+"  ParkAndAnnounce(announce:template|timeout|dial|[return_context]):\n"
+"Park a call into the parkinglot and announce the call over the console.\n"
+"announce template: colon separated list of files to announce, the word PARKED\n"
+"                   will be replaced by a say_digits of the ext the call is parked in\n"
+"timeout: time in seconds before the call returns into the return context.\n"
+"dial: The app_dial style resource to call to make the announcement. Console/dsp calls the console.\n"
+"return_context: the goto style label to jump the call back into after timeout. default=prio+1\n";
 
 
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
 static int parkandannounce_exec(struct ast_channel *chan, void *data)
 {
-	int res = -1;
-	int lot, timeout = 0, dres;
-	char *dialtech, *tmp[100], buf[13];
-	int looptemp, i;
-	char *s;
+	int res=0;
+	char *return_context;
+	int l, lot, timeout = 0, dres;
+	char *working, *context, *exten, *priority, *dial, *dialtech, *dialstr;
+	char *template, *tpl_working, *tpl_current;
+	char *tmp[100];
+	int looptemp=0,i=0;
+	char *s,*orig_s;
 
 	struct ast_channel *dchan;
-	struct outgoing_helper oh = { 0, };
 	int outstate;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(template);
-		AST_APP_ARG(timeout);
-		AST_APP_ARG(dial);
-		AST_APP_ARG(return_context);
-	);
-	if (ast_strlen_zero(data)) {
+
+	struct localuser *u;
+	if(!data || (data && !strlen(data))) {
 		ast_log(LOG_WARNING, "ParkAndAnnounce requires arguments: (announce:template|timeout|dial|[return_context])\n");
 		return -1;
 	}
   
-	s = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, s);
-
-	if (args.timeout)
-		timeout = atoi(args.timeout) * 1000;
-
-	if (ast_strlen_zero(args.dial)) {
-		ast_log(LOG_WARNING, "PARK: A dial resource must be specified i.e: Console/dsp or DAHDI/g1/5551212\n");
+	l=strlen(data)+2;
+	orig_s=malloc(l);
+	if(!orig_s) {
+		ast_log(LOG_WARNING, "Out of memory\n");
 		return -1;
 	}
+	s=orig_s;
+	strncpy(s,data,l);
 
-	dialtech = strsep(&args.dial, "/");
-	ast_verb(3, "Dial Tech,String: (%s,%s)\n", dialtech, args.dial);
-
-	if (!ast_strlen_zero(args.return_context)) {
-		ast_clear_flag(chan, AST_FLAG_IN_AUTOLOOP);
-		ast_parseable_goto(chan, args.return_context);
+	template=strsep(&s,"|");
+	if(! template) {
+		ast_log(LOG_WARNING, "PARK: An announce template must be defined\n");
+		free(orig_s);
+		return -1;
+	}
+  
+	if(s) {
+		timeout = atoi(strsep(&s, "|"));
+		timeout *= 1000;
+	}
+	dial=strsep(&s, "|");
+	if(!dial) {
+		ast_log(LOG_WARNING, "PARK: A dial resource must be specified i.e: Console/dsp or Zap/g1/5551212\n");
+		free(orig_s);
+		return -1;
+	} else {
+		dialtech=strsep(&dial, "/");
+		dialstr=dial;
+		ast_verbose( VERBOSE_PREFIX_3 "Dial Tech,String: (%s,%s)\n", dialtech,dialstr);
 	}
 
-	ast_verb(3, "Return Context: (%s,%s,%d) ID: %s\n", chan->context, chan->exten, chan->priority, chan->cid.cid_num);
-		if (!ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
-		ast_verb(3, "Warning: Return Context Invalid, call will return to default|s\n");
+	return_context = s;
+  
+	if(return_context != NULL) {
+		/* set the return context. Code borrowed from the Goto builtin */
+    
+		working = return_context;
+		context = strsep(&working, "|");
+		exten = strsep(&working, "|");
+		if(!exten) {
+			/* Only a priority in this one */
+			priority = context;
+			exten = NULL;
+			context = NULL;
+		} else {
+			priority = strsep(&working, "|");
+			if(!priority) {
+				/* Only an extension and priority in this one */
+				priority = exten;
+				exten = context;
+				context = NULL;
 		}
+	}
+	if(atoi(priority) < 0) {
+		ast_log(LOG_WARNING, "Priority '%s' must be a number > 0\n", priority);
+		free(orig_s);
+		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");
+		}
+	}
+  
+	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.  */
 
-	res = ast_masq_park_call(chan, NULL, timeout, &lot);
-	if (res == -1)
-		return res;
+	ast_masq_park_call(chan, NULL, timeout, &lot);
 
-	ast_verb(3, "Call Parking Called, lot: %d, timeout: %d, context: %s\n", lot, timeout, args.return_context);
+	res=-1; 
 
-	/* Now place the call to the extension */
+	ast_verbose( VERBOSE_PREFIX_3 "Call Parking Called, lot: %d, timeout: %d, context: %s\n", lot, timeout, return_context);
 
-	snprintf(buf, sizeof(buf), "%d", lot);
-	oh.parent_channel = chan;
-	oh.vars = ast_variable_new("_PARKEDAT", buf, "");
-	dchan = __ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, args.dial, 30000, &outstate, chan->cid.cid_num, chan->cid.cid_name, &oh);
+	/* Now place the call to the extention */
 
-	if (dchan) {
-		if (dchan->_state == AST_STATE_UP) {
-			ast_verb(4, "Channel %s was answered.\n", dchan->name);
+	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 {
-			ast_verb(4, "Channel %s was never answered.\n", dchan->name);
+			if(option_verbose > 3)
+				ast_verbose(VERBOSE_PREFIX_4 "Channel %s was never answered.\n", dchan->name);
         			ast_log(LOG_WARNING, "PARK: Channel %s was never answered for the announce.\n", dchan->name);
 			ast_hangup(dchan);
+			free(orig_s);
+			LOCAL_USER_REMOVE(u);
 			return -1;
 		}
 	} else {
 		ast_log(LOG_WARNING, "PARK: Unable to allocate announce channel.\n");
+		free(orig_s);
+		LOCAL_USER_REMOVE(u);
 		return -1; 
 	}
 
@@ -146,22 +187,24 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
 
 	/* now we have the call placed and are ready to play stuff to it */
 
-	ast_verb(4, "Announce Template:%s\n", args.template);
+	ast_verbose(VERBOSE_PREFIX_4 "Announce Template:%s\n", template);
 
-	for (looptemp = 0; looptemp < ARRAY_LEN(tmp); looptemp++) {
-		if ((tmp[looptemp] = strsep(&args.template, ":")) != NULL)
-			continue;
-		else
-			break;
+	tpl_working = template;
+	tpl_current=strsep(&tpl_working, ":");
+
+	while(tpl_current && looptemp < sizeof(tmp)) {
+		tmp[looptemp]=tpl_current;
+		looptemp++;
+		tpl_current=strsep(&tpl_working,":");
 	}
 
-	for (i = 0; i < looptemp; i++) {
-		ast_verb(4, "Announce:%s\n", tmp[i]);
-		if (!strcmp(tmp[i], "PARKED")) {
+	for(i=0; i<looptemp; i++) {
+		ast_verbose(VERBOSE_PREFIX_4 "Announce:%s\n", tmp[i]);
+		if(!strcmp(tmp[i], "PARKED")) {
 			ast_say_digits(dchan, lot, "", dchan->language);
 		} else {
 			dres = ast_streamfile(dchan, tmp[i], dchan->language);
-			if (!dres) {
+			if(!dres) {
 				dres = ast_waitstream(dchan, "");
 			} else {
 				ast_log(LOG_WARNING, "ast_streamfile of %s failed on %s\n", tmp[i], dchan->name);
@@ -172,19 +215,39 @@ static int parkandannounce_exec(struct ast_channel *chan, void *data)
 
 	ast_stopstream(dchan);  
 	ast_hangup(dchan);
-	
+
+	LOCAL_USER_REMOVE(u);
+	free(orig_s);
 	return res;
 }
 
-static int unload_module(void)
+
+
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	/* return ast_register_application(app, park_exec); */
 	return ast_register_application(app, parkandannounce_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Call Parking and Announce Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_pickupchan.c

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

apps/app_playback.c

@@ -1,527 +1,118 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Trivial application to playback a sound file
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Trivial application to playback a sound file
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
+ * the GNU General Public License
  */
  
-#include "asterisk.h"
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/utils.h>
+#include <string.h>
+#include <stdlib.h>
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/app.h"
-/* This file provides config-file based 'say' functions, and implenents
- * some CLI commands.
- */
-#include "asterisk/say.h"	/* provides config-file based 'say' functions */
-#include "asterisk/cli.h"
+static char *tdesc = "Trivial Playback Application";
 
 static char *app = "Playback";
 
 static char *synopsis = "Play a file";
 
 static char *descrip = 
-"  Playback(filename[&filename2...][,option]):  Plays back given filenames (do not put\n"
-"extension). Options may also be included following a comma.\n"
-"The 'skip' option causes the playback of the message to be skipped if the channel\n"
-"is not in the 'up' state (i.e. it hasn't been  answered  yet). If 'skip' is \n"
+"  Playback(filename[|option]):  Plays  back  a  given  filename (do not put\n"
+"extension). Options may also be  included following a pipe symbol. The 'skip'\n"
+"option causes the playback of the message to  be  skipped  if  the  channel\n"
+"is not in the 'up' state (i.e. it hasn't been  answered  yet. If 'skip' is \n"
 "specified, the application will return immediately should the channel not be\n"
-"off hook.  Otherwise, unless 'noanswer' is specified, the channel will\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 on hook.\n"
-"This application sets the following channel variable upon completion:\n"
-" PLAYBACKSTATUS    The status of the playback attempt as a text string, one of\n"
-"               SUCCESS | FAILED\n"
-"See Also: Background (application) -- for playing soundfiles that are interruptible\n"
-"          WaitExten (application) -- wait for digits from caller, optionally play music on hold\n"
-;
+"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;
 
-static struct ast_config *say_cfg = NULL;
-/* save the say' api calls.
- * The first entry is NULL if we have the standard source,
- * otherwise we are sourcing from here.
- * 'say load [new|old]' will enable the new or old method, or report status
- */
-static const void *say_api_buf[40];
-static const char *say_old = "old";
-static const char *say_new = "new";
-
-static void save_say_mode(const void *arg)
-{
-	int i = 0;
-	say_api_buf[i++] = arg;
-
-        say_api_buf[i++] = ast_say_number_full;
-        say_api_buf[i++] = ast_say_enumeration_full;
-        say_api_buf[i++] = ast_say_digit_str_full;
-        say_api_buf[i++] = ast_say_character_str_full;
-        say_api_buf[i++] = ast_say_phonetic_str_full;
-        say_api_buf[i++] = ast_say_datetime;
-        say_api_buf[i++] = ast_say_time;
-        say_api_buf[i++] = ast_say_date;
-        say_api_buf[i++] = ast_say_datetime_from_now;
-        say_api_buf[i++] = ast_say_date_with_format;
-}
-
-static void restore_say_mode(void *arg)
-{
-	int i = 0;
-	say_api_buf[i++] = arg;
-
-        ast_say_number_full = say_api_buf[i++];
-        ast_say_enumeration_full = say_api_buf[i++];
-        ast_say_digit_str_full = say_api_buf[i++];
-        ast_say_character_str_full = say_api_buf[i++];
-        ast_say_phonetic_str_full = say_api_buf[i++];
-        ast_say_datetime = say_api_buf[i++];
-        ast_say_time = say_api_buf[i++];
-        ast_say_date = say_api_buf[i++];
-        ast_say_datetime_from_now = say_api_buf[i++];
-        ast_say_date_with_format = say_api_buf[i++];
-}
-
-/* 
- * Typical 'say' arguments in addition to the date or number or string
- * to say. We do not include 'options' because they may be different
- * in recursive calls, and so they are better left as an external
- * parameter.
- */
-typedef struct {
-        struct ast_channel *chan;
-        const char *ints;
-        const char *language;
-        int audiofd;
-        int ctrlfd;
-} say_args_t;
-
-static int s_streamwait3(const say_args_t *a, const char *fn)
-{
-        int res = ast_streamfile(a->chan, fn, a->language);
-        if (res) {
-                ast_log(LOG_WARNING, "Unable to play message %s\n", fn);
-                return res;
-        }
-        res = (a->audiofd  > -1 && a->ctrlfd > -1) ?
-                ast_waitstream_full(a->chan, a->ints, a->audiofd, a->ctrlfd) :
-                ast_waitstream(a->chan, a->ints);
-        ast_stopstream(a->chan);
-        return res;  
-}
-
-/*
- * the string is 'prefix:data' or prefix:fmt:data'
- * with ':' being invalid in strings.
- */
-static int do_say(say_args_t *a, const char *s, const char *options, int depth)
-{
-	struct ast_variable *v;
-	char *lang, *x, *rule = NULL;
-	int ret = 0;   
-	struct varshead head = { .first = NULL, .last = NULL };
-	struct ast_var_t *n;
-
-	ast_debug(2, "string <%s> depth <%d>\n", s, depth);
-	if (depth++ > 10) {
-		ast_log(LOG_WARNING, "recursion too deep, exiting\n");
-		return -1;
-	} else if (!say_cfg) {
-		ast_log(LOG_WARNING, "no say.conf, cannot spell '%s'\n", s);
-		return -1;
-	}
-
-	/* scan languages same as in file.c */
-	if (a->language == NULL)
-		a->language = "en";     /* default */
-	ast_debug(2, "try <%s> in <%s>\n", s, a->language);
-	lang = ast_strdupa(a->language);
-	for (;;) {
-		for (v = ast_variable_browse(say_cfg, lang); v ; v = v->next) {
-			if (ast_extension_match(v->name, s)) {
-				rule = ast_strdupa(v->value);
-				break;
-			}
-		}
-		if (rule)
-			break;
-		if ( (x = strchr(lang, '_')) )
-			*x = '\0';      /* try without suffix */
-		else if (strcmp(lang, "en"))
-			lang = "en";    /* last resort, try 'en' if not done yet */
-		else
-			break;
-	}
-	if (!rule)
-		return 0;
-
-	/* skip up to two prefixes to get the value */
-	if ( (x = strchr(s, ':')) )
-		s = x + 1;
-	if ( (x = strchr(s, ':')) )
-		s = x + 1;
-	ast_debug(2, "value is <%s>\n", s);
-	n = ast_var_assign("SAY", s);
-	AST_LIST_INSERT_HEAD(&head, n, entries);
-
-	/* scan the body, one piece at a time */
-	while ( !ret && (x = strsep(&rule, ",")) ) { /* exit on key */
-		char fn[128];
-		const char *p, *fmt, *data; /* format and data pointers */
-
-		/* prepare a decent file name */
-		x = ast_skip_blanks(x);
-		ast_trim_blanks(x);
-
-		/* replace variables */
-		pbx_substitute_variables_varshead(&head, x, fn, sizeof(fn));
-		ast_debug(2, "doing [%s]\n", fn);
-
-		/* locate prefix and data, if any */
-		fmt = strchr(fn, ':');
-		if (!fmt || fmt == fn)	{	/* regular filename */
-			ret = s_streamwait3(a, fn);
-			continue;
-		}
-		fmt++;
-		data = strchr(fmt, ':');	/* colon before data */
-		if (!data || data == fmt) {	/* simple prefix-fmt */
-			ret = do_say(a, fn, options, depth);
-			continue;
-		}
-		/* prefix:fmt:data */
-		for (p = fmt; p < data && ret <= 0; p++) {
-			char fn2[sizeof(fn)];
-			if (*p == ' ' || *p == '\t')	/* skip blanks */
-				continue;
-			if (*p == '\'') {/* file name - we trim them */
-				char *y;
-				strcpy(fn2, ast_skip_blanks(p+1));	/* make a full copy */
-				y = strchr(fn2, '\'');
-				if (!y) {
-					p = data;	/* invalid. prepare to end */
-					break;
-				}
-				*y = '\0';
-				ast_trim_blanks(fn2);
-				p = strchr(p+1, '\'');
-				ret = s_streamwait3(a, fn2);
-			} else {
-				int l = fmt-fn;
-				strcpy(fn2, fn); /* copy everything */
-				/* after prefix, append the format */
-				fn2[l++] = *p;
-				strcpy(fn2 + l, data);
-				ret = do_say(a, fn2, options, depth);
-			}
-			
-			if (ret) {
-				break;
-			}
-		}
-	}
-	ast_var_delete(n);
-	return ret;
-}
-
-static int say_full(struct ast_channel *chan, const char *string,
-        const char *ints, const char *lang, const char *options,
-        int audiofd, int ctrlfd)
-{
-        say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
-        return do_say(&a, string, options, 0);
-}
-
-static int say_number_full(struct ast_channel *chan, int num,
-	const char *ints, const char *lang, const char *options,
-	int audiofd, int ctrlfd)
-{
-	char buf[64];
-        say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
-	snprintf(buf, sizeof(buf), "num:%d", num);
-        return do_say(&a, buf, options, 0);
-}
-
-static int say_enumeration_full(struct ast_channel *chan, int num,
-	const char *ints, const char *lang, const char *options,
-	int audiofd, int ctrlfd)
-{
-	char buf[64];
-        say_args_t a = { chan, ints, lang, audiofd, ctrlfd };
-	snprintf(buf, sizeof(buf), "enum:%d", num);
-        return do_say(&a, buf, options, 0);
-}
-
-static int say_date_generic(struct ast_channel *chan, time_t t,
-	const char *ints, const char *lang, const char *format, const char *timezone, const char *prefix)
-{
-	char buf[128];
-	struct ast_tm tm;
-	struct timeval tv = { t, 0 };
-        say_args_t a = { chan, ints, lang, -1, -1 };
-	if (format == NULL)
-		format = "";
-
-	ast_localtime(&tv, &tm, NULL);
-	snprintf(buf, sizeof(buf), "%s:%s:%04d%02d%02d%02d%02d.%02d-%d-%3d",
-		prefix,
-		format,
-		tm.tm_year+1900,
-		tm.tm_mon+1,
-		tm.tm_mday,
-		tm.tm_hour,
-		tm.tm_min,
-		tm.tm_sec,
-		tm.tm_wday,
-		tm.tm_yday);
-        return do_say(&a, buf, NULL, 0);
-}
-
-static int say_date_with_format(struct ast_channel *chan, time_t t,
-	const char *ints, const char *lang, const char *format, const char *timezone)
-{
-	return say_date_generic(chan, t, ints, lang, format, timezone, "datetime");
-}
-
-static int say_date(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
-{
-	return say_date_generic(chan, t, ints, lang, "", NULL, "date");
-}
-
-static int say_time(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
-{
-	return say_date_generic(chan, t, ints, lang, "", NULL, "time");
-}
-
-static int say_datetime(struct ast_channel *chan, time_t t, const char *ints, const char *lang)
-{
-	return say_date_generic(chan, t, ints, lang, "", NULL, "datetime");
-}
-
-/*
- * remap the 'say' functions to use those in this file
- */
-static int say_init_mode(const char *mode) {
-	if (!strcmp(mode, say_new)) {
-		if (say_cfg == NULL) {
-			ast_log(LOG_ERROR, "There is no say.conf file to use new mode\n");
-			return -1;
-		}
-		save_say_mode(say_new);
-		ast_say_number_full = say_number_full;
-
-		ast_say_enumeration_full = say_enumeration_full;
-#if 0
-		ast_say_digits_full = say_digits_full;
-		ast_say_digit_str_full = say_digit_str_full;
-		ast_say_character_str_full = say_character_str_full;
-		ast_say_phonetic_str_full = say_phonetic_str_full;
-		ast_say_datetime_from_now = say_datetime_from_now;
-#endif
-		ast_say_datetime = say_datetime;
-		ast_say_time = say_time;
-		ast_say_date = say_date;
-		ast_say_date_with_format = say_date_with_format;
-	} else if (!strcmp(mode, say_old) && say_api_buf[0] == say_new) {
-		restore_say_mode(NULL);
-	} else if (strcmp(mode, say_old)) {
-		ast_log(LOG_WARNING, "unrecognized mode %s\n", mode);
-		return -1;
-	}
-	
-	return 0;
-}
-
-static char *__say_cli_init(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
-	const char *old_mode = say_api_buf[0] ? say_new : say_old;
-	char *mode;
-	switch (cmd) {
-	case CLI_INIT:
-		e->command = "say load [new|old]";
-		e->usage = 
-			"Usage: say load [new|old]\n"
-			"       say load\n"
-			"           Report status of current say mode\n"
-			"       say load new\n"
-			"           Set say method, configured in say.conf\n"
-			"       say load old\n"
-			"           Set old say method, coded in asterisk core\n";
-		return NULL;
-	case CLI_GENERATE:
-		return NULL;
-	}
-	if (a->argc == 2) {
-		ast_cli(a->fd, "say mode is [%s]\n", old_mode);
-		return CLI_SUCCESS;
-	} else if (a->argc != e->args)
-		return CLI_SHOWUSAGE;
-	mode = a->argv[2];
-	if (!strcmp(mode, old_mode))
-		ast_cli(a->fd, "say mode is %s already\n", mode);
-	else
-		if (say_init_mode(mode) == 0)
-			ast_cli(a->fd, "setting say mode from %s to %s\n", old_mode, mode);
-
-	return CLI_SUCCESS;
-}
-
-static struct ast_cli_entry cli_playback[] = {
-	AST_CLI_DEFINE(__say_cli_init, "Set or show the say mode"),
-};
+LOCAL_USER_DECL;
 
 static int playback_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
-	int mres = 0;
-	char *tmp;
+	struct localuser *u;
+	char tmp[256];
+	char *options;
 	int option_skip=0;
-	int option_say=0;
 	int option_noanswer = 0;
-
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(filenames);
-		AST_APP_ARG(options);
-	);
-	
-	if (ast_strlen_zero(data)) {
+	char *stringp;
+	if (!data || ast_strlen_zero((char *)data)) {
 		ast_log(LOG_WARNING, "Playback requires an argument (filename)\n");
 		return -1;
 	}
-
-	tmp = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, tmp);
-
-	if (args.options) {
-		if (strcasestr(args.options, "skip"))
-			option_skip = 1;
-		if (strcasestr(args.options, "say"))
-			option_say = 1;
-		if (strcasestr(args.options, "noanswer"))
-			option_noanswer = 1;
-	} 
+	strncpy(tmp, (char *)data, sizeof(tmp)-1);
+	stringp=tmp;
+	strsep(&stringp, "|");
+	options = strsep(&stringp, "|");
+	if (options && !strcasecmp(options, "skip"))
+		option_skip = 1;
+	if (options && !strcasecmp(options, "noanswer"))
+		option_noanswer = 1;
+	LOCAL_USER_ADD(u);
 	if (chan->_state != AST_STATE_UP) {
 		if (option_skip) {
 			/* At the user's option, skip if the line is not up */
-			goto done;
-		} else if (!option_noanswer) {
+			LOCAL_USER_REMOVE(u);
+			return 0;
+		} else if (!option_noanswer)
 			/* Otherwise answer unless we're supposed to send this while on-hook */
 			res = ast_answer(chan);
-		}
 	}
 	if (!res) {
-		char *back = args.filenames;
-		char *front;
-
 		ast_stopstream(chan);
-		while (!res && (front = strsep(&back, "&"))) {
-			if (option_say)
-				res = say_full(chan, front, "", chan->language, NULL, -1, -1);
-			else
-				res = ast_streamfile(chan, front, chan->language);
-			if (!res) { 
-				res = ast_waitstream(chan, "");	
-				ast_stopstream(chan);
-			} else {
-				ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
-				res = 0;
-				mres = 1;
-			}
+		res = ast_streamfile(chan, tmp, chan->language);
+		if (!res) 
+			res = ast_waitstream(chan, "");
+		else {
+			ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
+			res = 0;
 		}
+		ast_stopstream(chan);
 	}
-done:
-	pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", mres ? "FAILED" : "SUCCESS");
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int reload(void)
+int unload_module(void)
 {
-	struct ast_variable *v;
-	struct ast_flags config_flags = { CONFIG_FLAG_FILEUNCHANGED };
-	struct ast_config *newcfg;
-
-	if ((newcfg = ast_config_load("say.conf", config_flags)) == CONFIG_STATUS_FILEUNCHANGED)
-		return 0;
-
-	if (say_cfg) {
-		ast_config_destroy(say_cfg);
-		ast_log(LOG_NOTICE, "Reloading say.conf\n");
-		say_cfg = newcfg;
-	}
-
-	if (say_cfg) {
-		for (v = ast_variable_browse(say_cfg, "general"); v ; v = v->next) {
-    			if (ast_extension_match(v->name, "mode")) {
-				say_init_mode(v->value);
-				break;
-			}
-		}
-	}
-	
-	/*
-	 * XXX here we should sort rules according to the same order
-	 * we have in pbx.c so we have the same matching behaviour.
-	 */
-	return 0;
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
 }
 
-static int unload_module(void)
+int load_module(void)
 {
-	int res;
-
-	res = ast_unregister_application(app);
-
-	ast_cli_unregister_multiple(cli_playback, sizeof(cli_playback) / sizeof(struct ast_cli_entry));
-
-	if (say_cfg)
-		ast_config_destroy(say_cfg);
-
-	return res;	
-}
-
-static int load_module(void)
-{
-	struct ast_variable *v;
-	struct ast_flags config_flags = { 0 };
-
-	say_cfg = ast_config_load("say.conf", config_flags);
-	if (say_cfg) {
-		for (v = ast_variable_browse(say_cfg, "general"); v ; v = v->next) {
-    			if (ast_extension_match(v->name, "mode")) {
-				say_init_mode(v->value);
-				break;
-			}
-		}
-	}
-
-        ast_cli_register_multiple(cli_playback, sizeof(cli_playback) / sizeof(struct ast_cli_entry));
 	return ast_register_application(app, playback_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Sound File Playback Application",
-		.load = load_module,
-		.unload = unload_module,
-		.reload = reload,
-	       );
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_privacy.c

@@ -1,131 +1,110 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Block all calls without Caller*ID, require phone # to be entered
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
+ * the GNU General Public License
  */
 
-/*! \file
- *
- * \brief Block all calls without Caller*ID, require phone # to be entered
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
- */
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/image.h>
+#include <asterisk/callerid.h>
+#include <asterisk/app.h>
+#include <asterisk/config.h>
+#include <string.h>
+#include <stdlib.h>
 
-#include "asterisk.h"
+#define PRIV_CONFIG "privacy.conf"
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/lock.h"
-#include "asterisk/file.h"
-#include "asterisk/utils.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/translate.h"
-#include "asterisk/image.h"
-#include "asterisk/callerid.h"
-#include "asterisk/app.h"
-#include "asterisk/config.h"
+static char *tdesc = "Require phone number to be entered, if no CallerID sent";
 
 static char *app = "PrivacyManager";
 
 static char *synopsis = "Require phone number to be entered, if no CallerID sent";
 
 static char *descrip =
-  "  PrivacyManager([maxretries][,minlength]): If no Caller*ID \n"
-  "is sent, PrivacyManager answers the channel and asks the caller to\n"
-  "enter their phone number. The caller is given 'maxretries' attempts to do so.\n"
-  "The application does nothing if Caller*ID was received on the channel.\n"
-  "   maxretries  default 3  -maximum number of attempts the caller is allowed \n"
-  "               to input a callerid.\n"
-  "   minlength   default 10 -minimum allowable digits in the input callerid number.\n"
-  "The application sets the following channel variable upon completion: \n"
-  "PRIVACYMGRSTATUS  The status of the privacy manager's attempt to collect \n"
-  "                  a phone number from the user. A text string that is either:\n" 
-  "          SUCCESS | FAILED \n"
-;
+  "  PrivacyManager: If no Caller*ID is sent, PrivacyManager answers the\n"
+  "channel and asks the caller to enter their 10 digit phone number.\n"
+  "The caller is given 3 attempts.  If after 3 attempts, they do not enter\n"
+  "their 10 digit phone number, and if there exists a priority n + 101,\n"
+  "where 'n' is the priority of the current instance, then  the\n"
+  "channel  will  be  setup  to continue at that priority level.\n"
+  "Otherwise, it returns 0.  Does nothing if Caller*ID was received on the\n"
+  "channel.\n";
 
+STANDARD_LOCAL_USER;
 
-static int privacy_exec (struct ast_channel *chan, void *data)
+LOCAL_USER_DECL;
+
+static int
+privacy_exec (struct ast_channel *chan, void *data)
 {
 	int res=0;
 	int retries;
 	int maxretries = 3;
-	int minlength = 10;
-	int x = 0;
-	char phone[30];
-	char *parse = NULL;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(maxretries);
-		AST_APP_ARG(minlength);
-		AST_APP_ARG(options);
-	);
+	int x;
+	char *s;
+	char phone[10];
+	char new_cid[144];
+	struct localuser *u;
+	struct ast_config *cfg;
 
-	if (!ast_strlen_zero(chan->cid.cid_num)) {
-		ast_verb(3, "CallerID Present: Skipping\n");
-	} else {
+	LOCAL_USER_ADD (u);
+	if (chan->callerid)
+	{
+		if (option_verbose > 2)
+			ast_verbose (VERBOSE_PREFIX_3 "CallerID Present: Skipping\n");
+	}
+	else
+	{
 		/*Answer the channel if it is not already*/
 		if (chan->_state != AST_STATE_UP) {
-			if ((res = ast_answer(chan)))
+			res = ast_answer(chan);
+			if (res) {
+				LOCAL_USER_REMOVE(u);
 				return -1;
+			}
 		}
-
-		if (!ast_strlen_zero(data)) {
-			parse = ast_strdupa(data);
-			
-			AST_STANDARD_APP_ARGS(args, parse);
-
-			if (args.maxretries) {
-				if (sscanf(args.maxretries, "%30d", &x) == 1)
-					maxretries = x;
-				else
-					ast_log(LOG_WARNING, "Invalid max retries argument\n");
-			}
-			if (args.minlength) {
-				if (sscanf(args.minlength, "%30d", &x) == 1)
-					minlength = x;
-				else
-					ast_log(LOG_WARNING, "Invalid min length argument\n");
-			}
-
-		}		
-
-		/* Play unidentified call */
+		/*Read in the config file*/
+		cfg = ast_load(PRIV_CONFIG);
+		
+		
+		/*Play unidentified call*/
 		res = ast_safe_sleep(chan, 1000);
 		if (!res)
 			res = ast_streamfile(chan, "privacy-unident", chan->language);
 		if (!res)
 			res = ast_waitstream(chan, "");
 
-		/* Ask for 10 digit number, give 3 attempts */
+        if (cfg && (s = ast_variable_retrieve(cfg, "general", "maxretries"))) {
+                if (sscanf(s, "%d", &x) == 1) {
+                        maxretries = x;
+                } else {
+                        ast_log(LOG_WARNING, "Invalid max retries argument\n");
+                }
+        }
+			
+		/*Ask for 10 digit number, give 3 attempts*/
 		for (retries = 0; retries < maxretries; retries++) {
 			if (!res)
-				res = ast_streamfile(chan, "privacy-prompt", chan->language);
-			if (!res)
-				res = ast_waitstream(chan, "");
-
-			if (!res ) 
-				res = ast_readstring(chan, phone, sizeof(phone) - 1, /* digit timeout ms */ 3200, /* first digit timeout */ 5000, "#");
-
+				res = ast_app_getdata(chan, "privacy-prompt", phone, sizeof(phone), 0);
 			if (res < 0)
 				break;
 
-			/* Make sure we get at least digits */
-			if (strlen(phone) >= minlength ) 
+			/*Make sure we get 10 digits*/
+			if (strlen(phone) == 10) 
 				break;
 			else {
 				res = ast_streamfile(chan, "privacy-incorrect", chan->language);
@@ -134,38 +113,58 @@ static int privacy_exec (struct ast_channel *chan, void *data)
 			}
 		}
 		
-		/* Got a number, play sounds and send them on their way */
-		if ((retries < maxretries) && res >= 0 ) {
+		/*Got a number, play sounds and send them on their way*/
+		if ((retries < maxretries) && !res) {
 			res = ast_streamfile(chan, "privacy-thankyou", chan->language);
 			if (!res)
 				res = ast_waitstream(chan, "");
-
-			ast_set_callerid (chan, phone, "Privacy Manager", NULL); 
-
-			/* Clear the unavailable presence bit so if it came in on PRI
-			 * the caller id will now be passed out to other channels
-			 */
-			chan->cid.cid_pres &= (AST_PRES_UNAVAILABLE ^ 0xFF);
-
-			ast_verb(3, "Changed Caller*ID to %s, callerpres to %d\n",phone,chan->cid.cid_pres);
-
-			pbx_builtin_setvar_helper(chan, "PRIVACYMGRSTATUS", "SUCCESS");
+			snprintf (new_cid, sizeof (new_cid), "\"%s\" <%s>", "Privacy Manager", phone);
+			ast_set_callerid (chan, new_cid, 0);
+			if (option_verbose > 2)
+				ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID to %s\n",new_cid);
 		} else {
-			pbx_builtin_setvar_helper(chan, "PRIVACYMGRSTATUS", "FAILED");
+			/*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 (cfg) 
+			ast_destroy(cfg);
 	}
 
-	return 0;
+  LOCAL_USER_REMOVE (u);
+  return 0;
 }
 
-static int unload_module(void)
+int
+unload_module (void)
 {
-	return ast_unregister_application (app);
+  STANDARD_HANGUP_LOCALUSERS;
+  return ast_unregister_application (app);
 }
 
-static int load_module(void)
+int
+load_module (void)
 {
-	return ast_register_application(app, privacy_exec, synopsis, descrip);
+  return ast_register_application (app, privacy_exec, synopsis,
+				   descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Require phone number to be entered, if no CallerID sent");
+char *
+description (void)
+{
+  return tdesc;
+}
+
+int
+usecount (void)
+{
+  int res;
+  STANDARD_USECOUNT (res);
+  return res;
+}
+
+char *
+key ()
+{
+  return ASTERISK_GPL_KEY;
+}

apps/app_qcall.c

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

apps/app_random.c

@@ -0,0 +1,128 @@
+/*
+ * 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)
+{
+	srandom((unsigned int)getpid() + (unsigned int)time(NULL));
+	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

@@ -1,231 +1,144 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Trivial application to read a variable
+ * 
+ * Copyright (C) 2003, Digium
  *
  * Mark Spencer <markster@digium.com>
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Trivial application to read a variable
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
+ * the GNU General Public License
  */
  
-#include "asterisk.h"
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/app.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/options.h>
+#include <asterisk/utils.h>
+#include <string.h>
+#include <stdlib.h>
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/pbx.h"
-#include "asterisk/channel.h"
-#include "asterisk/app.h"
-#include "asterisk/module.h"
-#include "asterisk/indications.h"
-
-enum {
-	OPT_SKIP = (1 << 0),
-	OPT_INDICATION = (1 << 1),
-	OPT_NOANSWER = (1 << 2),
-} read_option_flags;
-
-AST_APP_OPTIONS(read_app_options, {
-	AST_APP_OPTION('s', OPT_SKIP),
-	AST_APP_OPTION('i', OPT_INDICATION),
-	AST_APP_OPTION('n', OPT_NOANSWER),
-});
+static char *tdesc = "Read Variable Application";
 
 static char *app = "Read";
 
 static char *synopsis = "Read a variable";
 
 static char *descrip = 
-"  Read(variable[,filename[&filename2...]][,maxdigits][,option][,attempts][,timeout])\n\n"
-"Reads a #-terminated string of digits a certain number of times from the\n"
-"user in to the given variable.\n"
-"  filename   -- file(s) to play before reading digits or tone with option i\n"
+"  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     -- options are 's' , 'i', 'n'\n"
-"                's' to return immediately if the line is not up,\n"
-"                'i' to play  filename as an indication tone from your indications.conf\n"
-"                'n' to read digits even if the line is not up.\n"
-"  attempts   -- if greater than 1, that many attempts will be made in the \n"
-"                event no data is entered.\n"
-"  timeout    -- The number of seconds to wait for a digit response. If greater\n"
-"                than 0, that value will override the default timeout. Can be floating point.\n"
-"This application sets the following channel variable upon completion:\n"
-"    READSTATUS - This is the status of the read operation.\n"
-"                 Possible values are:\n"
-"                 OK | ERROR | HANGUP | INTERRUPTED | SKIPPED | TIMEOUT\n";
+"  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;
 
-#define ast_next_data(instr,ptr,delim) if((ptr=strchr(instr,delim))) { *(ptr) = '\0' ; ptr++;}
+LOCAL_USER_DECL;
 
 static int read_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
-	char tmp[256] = "";
-	int maxdigits = 255;
-	int tries = 1, to = 0, x = 0;
-	double tosec;
-	char *argcopy = NULL;
-	struct tone_zone_sound *ts = NULL;
-	struct ast_flags flags = {0};
-	const char *status = "ERROR";
-
-	AST_DECLARE_APP_ARGS(arglist,
-		AST_APP_ARG(variable);
-		AST_APP_ARG(filename);
-		AST_APP_ARG(maxdigits);
-		AST_APP_ARG(options);
-		AST_APP_ARG(attempts);
-		AST_APP_ARG(timeout);
-	);
-	
-	pbx_builtin_setvar_helper(chan, "READSTATUS", status);
-	if (ast_strlen_zero(data)) {
+	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 0;
+		return -1;
 	}
-	
-	argcopy = ast_strdupa(data);
-
-	AST_STANDARD_APP_ARGS(arglist, argcopy);
-
-	if (!ast_strlen_zero(arglist.options)) {
-		ast_app_parse_options(read_app_options, &flags, NULL, arglist.options);
-	}
-	
-	if (!ast_strlen_zero(arglist.attempts)) {
-		tries = atoi(arglist.attempts);
-		if (tries <= 0)
-			tries = 1;
-	}
-
-	if (!ast_strlen_zero(arglist.timeout)) {
-		tosec = atof(arglist.timeout);
-		if (tosec <= 0)
-			to = 0;
-		else
-			to = tosec * 1000.0;
-	}
-
-	if (ast_strlen_zero(arglist.filename)) {
-		arglist.filename = NULL;
-	}
-	if (!ast_strlen_zero(arglist.maxdigits)) {
-		maxdigits = atoi(arglist.maxdigits);
-		if ((maxdigits < 1) || (maxdigits > 255)) {
-			maxdigits = 255;
-		} else
-			ast_verb(3, "Accepting a maximum of %d digits.\n", maxdigits);
-	}
-	if (ast_strlen_zero(arglist.variable)) {
-		ast_log(LOG_WARNING, "Invalid! Usage: Read(variable[,filename][,maxdigits][,option][,attempts][,timeout])\n\n");
-		return 0;
-	}
-	if (ast_test_flag(&flags, OPT_INDICATION)) {
-		if (! ast_strlen_zero(arglist.filename)) {
-			ts = ast_get_indication_tone(chan->zone, arglist.filename);
-		}
+	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 (ast_test_flag(&flags, OPT_SKIP)) {
+		if (option_skip) {
 			/* At the user's option, skip if the line is not up */
-			pbx_builtin_setvar_helper(chan, arglist.variable, "");
-			pbx_builtin_setvar_helper(chan, "READSTATUS", "SKIPPED");
+			pbx_builtin_setvar_helper(chan, varname, "\0");
+			LOCAL_USER_REMOVE(u);
 			return 0;
-		} else if (!ast_test_flag(&flags, OPT_NOANSWER)) {
+		} else if (!option_noanswer) {
 			/* Otherwise answer unless we're supposed to read while on-hook */
 			res = ast_answer(chan);
 		}
 	}
 	if (!res) {
-		while (tries && !res) {
-			ast_stopstream(chan);
-			if (ts && ts->data[0]) {
-				if (!to)
-					to = chan->pbx ? chan->pbx->rtimeout * 1000 : 6000;
-				res = ast_playtones_start(chan, 0, ts->data, 0);
-				for (x = 0; x < maxdigits; ) {
-					res = ast_waitfordigit(chan, to);
-					ast_playtones_stop(chan);
-					if (res < 1) {
-						if (res == 0)
-							status = "TIMEOUT";
-						tmp[x]='\0';
-						break;
-					}
-					tmp[x++] = res;
-					if (tmp[x-1] == '#') {
-						tmp[x-1] = '\0';
-						status = "OK";
-						break;
-					}
-					if (x >= maxdigits) {
-						status = "OK";
-					}
-				}
-			} else {
-				res = ast_app_getdata(chan, arglist.filename, tmp, maxdigits, to);
-				if (res == AST_GETDATA_COMPLETE || res == AST_GETDATA_EMPTY_END_TERMINATED)
-					status = "OK";
-				else if (res == AST_GETDATA_TIMEOUT)
-					status = "TIMEOUT";
-				else if (res == AST_GETDATA_INTERRUPTED)
-					status = "INTERRUPTED";
-			}
-			if (res > -1) {
-				pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
-				if (!ast_strlen_zero(tmp)) {
-					ast_verb(3, "User entered '%s'\n", tmp);
-					tries = 0;
-				} else {
-					tries--;
-					if (tries)
-						ast_verb(3, "User entered nothing, %d chance%s left\n", tries, (tries != 1) ? "s" : "");
-					else
-						ast_verb(3, "User entered nothing.\n");
-				}
-				res = 0;
-			} else {
-				pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
-				ast_verb(3, "User disconnected\n");
-			}
+		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");
 		}
 	}
-
-	if (ast_check_hangup(chan))
-		status = "HANGUP";
-	pbx_builtin_setvar_helper(chan, "READSTATUS", status);
-	return 0;
+	LOCAL_USER_REMOVE(u);
+	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, read_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Read Variable Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_readexten.c

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

apps/app_readfile.c

@@ -1,107 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2005, Digium, Inc.
- *
- * Matt O'Gorman <mogorman@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief ReadFile application -- Reads in a File for you.
- *
- * \author Matt O'Gorman <mogorman@digium.com>
- *
- * \ingroup applications
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/app.h"
-#include "asterisk/module.h"
-
-static char *app_readfile = "ReadFile";
-
-static char *readfile_synopsis = "Read the contents of a text file into a channel variable";
-
-static char *readfile_descrip =
-"ReadFile(varname=file,length)\n"
-"  varname  - Result stored here.\n"
-"  file     - The name of the file to read.\n"
-"  length   - Maximum number of characters to capture.\n";
-
-
-static int readfile_exec(struct ast_channel *chan, void *data)
-{
-	int res=0;
-	char *s, *varname=NULL, *file=NULL, *length=NULL, *returnvar=NULL;
-	int len=0;
-	static int deprecation_warning = 0;
-
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "ReadFile require an argument!\n");
-		return -1;
-	}
-
-	s = ast_strdupa(data);
-
-	varname = strsep(&s, "=");
-	file = strsep(&s, ",");
-	length = s;
-
-	if (deprecation_warning++ % 10 == 0)
-		ast_log(LOG_WARNING, "ReadFile has been deprecated in favor of Set(%s=${FILE(%s,0,%s)})\n", varname, file, length);
-
-	if (!varname || !file) {
-		ast_log(LOG_ERROR, "No file or variable specified!\n");
-		return -1;
-	}
-
-	if (length) {
-		if ((sscanf(length, "%30d", &len) != 1) || (len < 0)) {
-			ast_log(LOG_WARNING, "%s is not a positive number, defaulting length to max\n", length);
-			len = 0;
-		}
-	}
-
-	if ((returnvar = ast_read_textfile(file))) {
-		if (len > 0) {
-			if (len < strlen(returnvar))
-				returnvar[len]='\0';
-			else
-				ast_log(LOG_WARNING, "%s is longer than %d, and %d \n", file, len, (int)strlen(returnvar));
-		}
-		pbx_builtin_setvar_helper(chan, varname, returnvar);
-		ast_free(returnvar);
-	}
-
-	return res;
-}
-
-
-static int unload_module(void)
-{
-	return ast_unregister_application(app_readfile);
-}
-
-static int load_module(void)
-{
-	return ast_register_application(app_readfile, readfile_exec, readfile_synopsis, readfile_descrip);
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Stores output of file into a variable");

apps/app_record.c

@@ -1,101 +1,67 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Trivial application to record a sound file
+ * 
+ * Copyright (C) 2001, Linux Support Services, Inc.
  *
- * Matthew Fredrickson <creslin@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Matthew Fredrickson <creslin@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Trivial application to record a sound file
- *
- * \author Matthew Fredrickson <creslin@digium.com>
- *
- * \ingroup applications
+ * the GNU General Public License
  */
  
-#include "asterisk.h"
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/dsp.h>
+#include <string.h>
+#include <stdlib.h>
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/app.h"
-#include "asterisk/channel.h"
-#include "asterisk/dsp.h"	/* use dsp routines for silence detection */
+static char *tdesc = "Trivial Record Application";
 
 static char *app = "Record";
 
 static char *synopsis = "Record to a file";
 
 static char *descrip = 
-"  Record(filename.format,silence[,maxduration][,options])\n\n"
+"  Record(filename:format|silence[|maxduration][|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"
-"- 'options' may contain any of the following letters:\n"
-"     'a' : append to existing recording rather than replacing\n"
-"     'k' : keep recorded file upon hangup\n"
-"     'n' : do not answer, but record anyway if line not yet answered\n"
-"     'q' : quiet (do not play a beep tone)\n"
-"     's' : skip recording if the line is not yet answered\n"
-"     't' : use alternate '*' terminator key (DTMF) instead of default '#'\n"
-"     'x' : ignore all terminator keys (DTMF) and keep recording until hangup\n"
-"\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. A channel variable\n"
-"named RECORDED_FILE will also be set, which contains the final filemname.\n\n"
-"Use 'core show file formats' to see the available formats on your system\n\n"
+"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"
-"If the user should hangup during a recording, all data will be lost and the\n"
-"application will teminate. \n";
+"Returns -1 when the user hangs up.\n";
 
-enum {
-	OPTION_APPEND = (1 << 0),
-	OPTION_NOANSWER = (1 << 1),
-	OPTION_QUIET = (1 << 2),
-	OPTION_SKIP = (1 << 3),
-	OPTION_STAR_TERMINATE = (1 << 4),
-	OPTION_IGNORE_TERMINATE = (1 << 5),
-	OPTION_KEEP = (1 << 6),
-	FLAG_HAS_PERCENT = (1 << 7),
-};
+STANDARD_LOCAL_USER;
 
-AST_APP_OPTIONS(app_opts,{
-	AST_APP_OPTION('a', OPTION_APPEND),
-	AST_APP_OPTION('k', OPTION_KEEP),	
-	AST_APP_OPTION('n', OPTION_NOANSWER),
-	AST_APP_OPTION('q', OPTION_QUIET),
-	AST_APP_OPTION('s', OPTION_SKIP),
-	AST_APP_OPTION('t', OPTION_STAR_TERMINATE),
-	AST_APP_OPTION('x', OPTION_IGNORE_TERMINATE),
-});
+LOCAL_USER_DECL;
 
 static int record_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
 	int count = 0;
-	char *ext = NULL, *opts[0];
-	char *parse, *dir, *file;
-	int i = 0;
+	int percentflag = 0;
+	char fil[256];
 	char tmp[256];
+	char ext[10];
+	char *vdata;
+	int i = 0;
+	int j = 0;
 
-	struct ast_filestream *s = NULL;
+	struct ast_filestream *s = '\0';
+	struct localuser *u;
 	struct ast_frame *f = NULL;
 	
 	struct ast_dsp *sildet = NULL;   	/* silence detector dsp */
@@ -103,124 +69,132 @@ static int record_exec(struct ast_channel *chan, void *data)
 	int dspsilence = 0;
 	int silence = 0;		/* amount of silence to allow */
 	int gotsilence = 0;		/* did we timeout for silence? */
-	int maxduration = 0;		/* max duration of recording in milliseconds */
+	char silencestr[5];
+	char durationstr[8];
+	int maxduration = 0;		/* max duration of recording */
 	int gottimeout = 0;		/* did we timeout for maxduration exceeded? */
-	int terminator = '#';
+	time_t timeout = 0;
+	char option[16];
+	int option_skip = 0;
+	int option_noanswer = 0;
 	int rfmt = 0;
-	int ioflags;
-	int waitres;
-	struct ast_silence_generator *silgen = NULL;
-	struct ast_flags flags = { 0, };
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(filename);
-		AST_APP_ARG(silence);
-		AST_APP_ARG(maxduration);
-		AST_APP_ARG(options);
-	);
-	
+	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 (ast_strlen_zero(data)) { /* 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;
 	}
-
-	parse = ast_strdupa(data);
-	AST_STANDARD_APP_ARGS(args, parse);
-	if (args.argc == 4)
-		ast_app_parse_options(app_opts, &flags, opts, args.options);
-
-	if (!ast_strlen_zero(args.filename)) {
-		if (strstr(args.filename, "%d"))
-			ast_set_flag(&flags, FLAG_HAS_PERCENT);
-		ext = strrchr(args.filename, '.'); /* to support filename with a . in the filename, not format */
-		if (!ext)
-			ext = strchr(args.filename, ':');
-		if (ext) {
-			*ext = '\0';
-			ext++;
-		}
-	}
-	if (!ext) {
-		ast_log(LOG_WARNING, "No extension specified to filename!\n");
-		return -1;
-	}
-	if (args.silence) {
-		if ((sscanf(args.silence, "%30d", &i) == 1) && (i > -1)) {
-			silence = i * 1000;
-		} else if (!ast_strlen_zero(args.silence)) {
-			ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", args.silence);
+	vdata = ast_strdupa(data);
+	
+	p = vdata;
+	while(p && (p=strchr(p,':'))) {
+		end=p;
+		if(!strcasecmp(end,":end")) {
+			*end='\0';
+			end++;
+			break;
 		}
+		p++;
+		end=NULL;
 	}
 	
-	if (args.maxduration) {
-		if ((sscanf(args.maxduration, "%30d", &i) == 1) && (i > -1))
-			/* Convert duration to milliseconds */
-			maxduration = i * 1000;
-		else if (!ast_strlen_zero(args.maxduration))
-			ast_log(LOG_WARNING, "'%s' is not a valid maximum duration\n", args.maxduration);
+
+	for (; vdata[i] && (vdata[i] != ':') && (vdata[i] != '|'); i++ ) {
+		if ((vdata[i] == '%') && (vdata[i+1] == 'd')) {
+			percentflag = 1;                      /* the wildcard is used */
+		}
+		if (j < sizeof(fil) - 1)
+			fil[j++] = vdata[i];
+	}
+	fil[j] = '\0';
+
+	if (vdata[i] != ':') {
+		ast_log(LOG_WARNING, "No extension found\n");
+		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 (ast_test_flag(&flags, OPTION_STAR_TERMINATE))
-		terminator = '*';
-	if (ast_test_flag(&flags, OPTION_IGNORE_TERMINATE))
-		terminator = '\0';
+	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 (ast_test_flag(&flags, FLAG_HAS_PERCENT)) {
-		AST_DECLARE_APP_ARGS(fname,
-			AST_APP_ARG(piece)[100];
-		);
-		char *tmp2 = ast_strdupa(args.filename);
-		char countstring[15];
-		int i;
-
-		/* Separate each piece out by the format specifier */
-		AST_NONSTANDARD_APP_ARGS(fname, tmp2, '%');
+	if (percentflag) {
 		do {
-			int tmplen;
-			/* First piece has no leading percent, so it's copied verbatim */
-			ast_copy_string(tmp, fname.piece[0], sizeof(tmp));
-			tmplen = strlen(tmp);
-			for (i = 1; i < fname.argc; i++) {
-				if (fname.piece[i][0] == 'd') {
-					/* Substitute the count */
-					snprintf(countstring, sizeof(countstring), "%d", count);
-					ast_copy_string(tmp + tmplen, countstring, sizeof(tmp) - tmplen);
-					tmplen += strlen(countstring);
-				} else if (tmplen + 2 < sizeof(tmp)) {
-					/* Unknown format specifier - just copy it verbatim */
-					tmp[tmplen++] = '%';
-					tmp[tmplen++] = fname.piece[i][0];
-				}
-				/* Copy the remaining portion of the piece */
-				ast_copy_string(tmp + tmplen, &(fname.piece[i][1]), sizeof(tmp) - tmplen);
-			}
+			snprintf(tmp, sizeof(tmp), fil, count);
 			count++;
-		} while (ast_fileexists(tmp, ext, chan->language) > 0);
+		} while ( ast_fileexists(tmp, ext, chan->language) != -1 );
 		pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
 	} else
-		ast_copy_string(tmp, args.filename, sizeof(tmp));
+		strncpy(tmp, fil, sizeof(tmp)-1);
 	/* end of routine mentioned */
 
+	LOCAL_USER_ADD(u);
+
 	if (chan->_state != AST_STATE_UP) {
-		if (ast_test_flag(&flags, OPTION_SKIP)) {
+		if (option_skip) {
 			/* At the user's option, skip if the line is not up */
+			LOCAL_USER_REMOVE(u);
 			return 0;
-		} else if (!ast_test_flag(&flags, OPTION_NOANSWER)) {
+		} else if (!option_noanswer) {
 			/* Otherwise answer unless we're supposed to record while on-hook */
 			res = ast_answer(chan);
 		}
 	}
-
-	if (res) {
-		ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
-		goto out;
-	}
-
-	if (!ast_test_flag(&flags, OPTION_QUIET)) {
+	
+	if (!res) {
 		/* Some code to play a nice little beep to signify the start of the record operation */
 		res = ast_streamfile(chan, "beep", chan->language);
 		if (!res) {
@@ -229,142 +203,138 @@ 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 */
+		/* The end of beep code.  Now the recording starts */
 
-	if (silence > 0) {
-		rfmt = chan->readformat;
-		res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
-		if (res < 0) {
-			ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
-			return -1;
-		}
-		sildet = ast_dsp_new();
-		if (!sildet) {
-			ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
-			return -1;
-		}
-		ast_dsp_set_threshold(sildet, 256);
-	} 
 
-	/* Create the directory if it does not exist. */
-	dir = ast_strdupa(tmp);
-	if ((file = strrchr(dir, '/')))
-		*file++ = '\0';
-	ast_mkdir (dir, 0777);
-
-	ioflags = ast_test_flag(&flags, OPTION_APPEND) ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
-	s = ast_writefile(tmp, ext, NULL, ioflags, 0, AST_FILE_MODE);
-
-	if (!s) {
-		ast_log(LOG_WARNING, "Could not create file %s\n", args.filename);
-		goto out;
-	}
-
-	if (ast_opt_transmit_silence)
-		silgen = ast_channel_start_silence_generator(chan);
-
-	/* Request a video update */
-	ast_indicate(chan, AST_CONTROL_VIDUPDATE);
-
-	if (maxduration <= 0)
-		maxduration = -1;
-
-	while ((waitres = ast_waitfor(chan, maxduration)) > -1) {
-		if (maxduration > 0) {
-			if (waitres == 0) {
-				gottimeout = 1;
-				break;
-			}
-			maxduration = waitres;
+		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);
 		}
 
-		f = ast_read(chan);
-		if (!f) {
-			res = -1;
-			break;
-		}
-		if (f->frametype == AST_FRAME_VOICE) {
-			res = ast_writestream(s, f);
 
-			if (res) {
-				ast_log(LOG_WARNING, "Problem writing frame\n");
-				ast_frfree(f);
-				break;
-			}
+		flags = end ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
+		s = ast_writefile( tmp, ext, NULL, flags , 0, 0644);
 
-			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;
+
+		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;
 				}
-			}
-		} else if (f->frametype == AST_FRAME_VIDEO) {
-			res = ast_writestream(s, f);
 
-			if (res) {
-				ast_log(LOG_WARNING, "Problem writing frame\n");
+				f = ast_read(chan);
+				if (!f) {
+					res = -1;
+					break;
+				}
+				if (f->frametype == AST_FRAME_VOICE) {
+					res = ast_writestream(s, f);
+					
+					if (res) {
+						ast_log(LOG_WARNING, "Problem writing frame\n");
+						break;
+					}
+
+					if (silence > 0) {
+						dspsilence = 0;
+						ast_dsp_silence(sildet, f, &dspsilence);
+						if (dspsilence) {
+							totalsilence = dspsilence;
+						} else {
+							totalsilence = 0;
+						}
+	                                	if (totalsilence > silence) {
+        	                        	        /* Ended happily with silence */
+							ast_frfree(f);
+							gotsilence = 1;
+                	                	        break;
+	                     	        	}
+					}
+				}
+				if (f->frametype == AST_FRAME_VIDEO) {
+					res = ast_writestream(s, f);
+					
+					if (res) {
+						ast_log(LOG_WARNING, "Problem writing frame\n");
+						break;
+					}
+				}
+				if ((f->frametype == AST_FRAME_DTMF) &&
+					(f->subclass == '#')) {
+					ast_frfree(f);
+					break;
+				}
 				ast_frfree(f);
-				break;
 			}
-		} else if ((f->frametype == AST_FRAME_DTMF) &&
-		    (f->subclass == terminator)) {
-			ast_frfree(f);
-			break;
-		}
-		ast_frfree(f);
-	}
-	if (!f) {
-		ast_debug(1, "Got hangup\n");
-		res = -1;
-		if (!ast_test_flag(&flags, OPTION_KEEP)) {
-			ast_filedelete(args.filename, NULL);
-		}
-	}
+			if (!f) {
+					ast_log(LOG_DEBUG, "Got hangup\n");
+					res = -1;
+			}
 
-	if (gotsilence) {
-		ast_stream_rewind(s, silence - 1000);
-		ast_truncstream(s);
-	} else if (!gottimeout) {
-		/* Strip off the last 1/4 second of it */
-		ast_stream_rewind(s, 250);
-		ast_truncstream(s);
-	}
-	ast_closestream(s);
+			if (gotsilence) {
+				ast_stream_rewind(s, silence-1000);
+				ast_truncstream(s);
+			} else if (!gottimeout) {
+				/* Strip off the last 1/4 second of it */
+				ast_stream_rewind(s, 250);
+				ast_truncstream(s);
+			}
+			ast_closestream(s);
+		} else			
+			ast_log(LOG_WARNING, "Could not create file %s\n", fil);
+	} else
+		ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
 
-	if (silgen)
-		ast_channel_stop_silence_generator(chan, silgen);
-
-out:
+	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);
+	        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;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, record_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Trivial Record Application");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_sayunixtime.c

@@ -1,40 +1,30 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (c) 2003, 2006 Tilghman Lesher.  All rights reserved.
- * Copyright (c) 2006 Digium, Inc.
+ * SayUnixTime application
+ * 
+ * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
  *
  * Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com>
  *
  * This code is released by the author with no restrictions on usage.
  *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
  */
 
-/*! \file
- *
- * \brief SayUnixTime application
- *
- * \author Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com>
- * 
- * \ingroup applications
- */
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/say.h>
 
-#include "asterisk.h"
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/say.h"
-#include "asterisk/app.h"
+static char *tdesc = "Say time";
 
 static char *app_sayunixtime = "SayUnixTime";
 static char *app_datetime = "DateTime";
@@ -42,71 +32,110 @@ static char *app_datetime = "DateTime";
 static char *sayunixtime_synopsis = "Says a specified time in a custom format";
 
 static char *sayunixtime_descrip =
-"SayUnixTime([unixtime][,[timezone][,format]])\n"
-"  unixtime  - time, in seconds since Jan 1, 1970.  May be negative.\n"
+"SayUnixTime([unixtime][|[timezone][|format]])\n"
+"  unixtime: time, in seconds since Jan 1, 1970.  May be negative.\n"
 "              defaults to now.\n"
-"  timezone  - timezone, see /usr/share/zoneinfo for a list.\n"
+"  timezone: timezone, see /usr/share/zoneinfo for a list.\n"
 "              defaults to machine default.\n"
-"  format    - a format the time is to be said in.  See voicemail.conf.\n"
-"              defaults to \"ABdY 'digits/at' IMp\"\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"
+"DateTime([unixtime][|[timezone][|format]])\n"
+"  unixtime: time, in seconds since Jan 1, 1970.  May be negative.\n"
 "              defaults to now.\n"
-"  timezone  - timezone, see /usr/share/zoneinfo for a list.\n"
+"  timezone: timezone, see /usr/share/zoneinfo for a list.\n"
 "              defaults to machine default.\n"
-"  format:   - a format the time is to be said in.  See voicemail.conf.\n"
-"              defaults to \"ABdY 'digits/at' IMp\"\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)
 {
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(timeval);
-		AST_APP_ARG(timezone);
-		AST_APP_ARG(format);
-	);
-	char *parse;
-	int res = 0;
+	int res=0;
+	struct localuser *u;
+	char *s,*zone=NULL,*timec;
 	time_t unixtime;
-	
-	if (!data)
-		return 0;
+	char *format = "ABdY 'digits/at' IMp";
+	struct timeval tv;
 
-	parse = ast_strdupa(data);
+	LOCAL_USER_ADD(u);
 
-	AST_STANDARD_APP_ARGS(args, parse);
+	gettimeofday(&tv,NULL);
+	unixtime = (time_t)tv.tv_sec;
 
-	ast_get_time_t(args.timeval, &unixtime, time(NULL), NULL);
+	if (data) {
+		s = data;
+		s = ast_strdupa(s);
+		if (s) {
+			timec = strsep(&s,"|");
+			if ((timec) && (*timec != '\0')) {
+				long timein;
+				if (sscanf(timec,"%ld",&timein) == 1) {
+					unixtime = (time_t)timein;
+				}
+			}
+			if (s) {
+				zone = strsep(&s,"|");
+				if (zone && (*zone == '\0'))
+					zone = NULL;
+				if (s) {
+					format = s;
+				}
+			}
+		} else {
+			ast_log(LOG_ERROR, "Out of memory error\n");
+		}
+	}
 
-	if (chan->_state != AST_STATE_UP)
+	if (chan->_state != AST_STATE_UP) {
 		res = ast_answer(chan);
-
+	}
 	if (!res)
-		res = ast_say_date_with_format(chan, unixtime, AST_DIGIT_ANY,
-					       chan->language, args.format, args.timezone);
+		res = ast_say_date_with_format(chan, unixtime, AST_DIGIT_ANY, chan->language, format, zone);
 
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
 	int res;
-	
+	STANDARD_HANGUP_LOCALUSERS;
 	res = ast_unregister_application(app_sayunixtime);
-	res |= ast_unregister_application(app_datetime);
-	
-	return res;
+	if (! res)
+		return ast_unregister_application(app_datetime);
+	else
+		return res;
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	int res;
-	
 	res = ast_register_application(app_sayunixtime, sayunixtime_exec, sayunixtime_synopsis, sayunixtime_descrip);
-	res |= ast_register_application(app_datetime, sayunixtime_exec, sayunixtime_synopsis, datetime_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;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Say time");
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_senddtmf.c

@@ -1,129 +1,84 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * App to send DTMF digits
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief App to send DTMF digits
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
+ * the GNU General Public License
  */
  
-#include "asterisk.h"
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/options.h>
+#include <asterisk/utils.h>
+#include <asterisk/app.h>
+#include <string.h>
+#include <stdlib.h>
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/app.h"
-#include "asterisk/manager.h"
-#include "asterisk/channel.h"
+static char *tdesc = "Send DTMF digits Application";
 
 static char *app = "SendDTMF";
 
 static char *synopsis = "Sends arbitrary DTMF digits";
 
 static char *descrip = 
-" SendDTMF(digits[,[timeout_ms][,duration_ms]]): Sends DTMF digits on a channel. \n"
-" Accepted digits: 0-9, *#abcd, (default .25s pause between digits)\n"
-" The application will either pass the assigned digits or terminate if it\n"
-" encounters an error.\n"
-" Optional Params: \n"
-"   timeout_ms: pause between digits.\n"
-"   duration_ms: duration of each digit.\n";
+"  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";
 
-static int senddtmf_exec(struct ast_channel *chan, void *vdata)
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int senddtmf_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
-	char *data;
-	int timeout = 0, duration = 0;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(digits);
-		AST_APP_ARG(timeout);
-		AST_APP_ARG(duration);
-	);
+	struct localuser *u;
+	char *digits = data;
 
-	if (ast_strlen_zero(vdata)) {
+	if (!digits || ast_strlen_zero(digits)) {
 		ast_log(LOG_WARNING, "SendDTMF requires an argument (digits or *#aAbBcCdD)\n");
-		return 0;
+		return -1;
 	}
-
-	data = ast_strdupa(vdata);
-	AST_STANDARD_APP_ARGS(args, data);
-
-	if (!ast_strlen_zero(args.timeout))
-		timeout = atoi(args.timeout);
-	if (!ast_strlen_zero(args.duration))
-		duration = atoi(args.duration);
-	res = ast_dtmf_stream(chan, NULL, args.digits, timeout <= 0 ? 250 : timeout, duration);
-
+	LOCAL_USER_ADD(u);
+	res = ast_dtmf_stream(chan,NULL,digits,250);
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
-static char mandescr_playdtmf[] =
-"Description: Plays a dtmf digit on the specified channel.\n"
-"Variables: (all are required)\n"
-"	Channel: Channel name to send digit to\n"
-"	Digit: The dtmf digit to play\n";
-
-static int manager_play_dtmf(struct mansession *s, const struct message *m)
+int unload_module(void)
 {
-	const char *channel = astman_get_header(m, "Channel");
-	const char *digit = astman_get_header(m, "Digit");
-	struct ast_channel *chan = ast_get_channel_by_name_locked(channel);
-	
-	if (!chan) {
-		astman_send_error(s, m, "Channel not specified");
-		return 0;
-	}
-	if (ast_strlen_zero(digit)) {
-		astman_send_error(s, m, "No digit specified");
-		ast_channel_unlock(chan);
-		return 0;
-	}
-
-	ast_senddigit(chan, *digit, 0);
-
-	ast_channel_unlock(chan);
-	astman_send_ack(s, m, "DTMF successfully queued");
-	
-	return 0;
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
 }
 
-static int unload_module(void)
+int load_module(void)
 {
-	int res;
-
-	res = ast_unregister_application(app);
-	res |= ast_manager_unregister("PlayDTMF");
-
-	return res;	
+	return ast_register_application(app, senddtmf_exec, synopsis, descrip);
 }
 
-static int load_module(void)
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
 {
 	int res;
-
-	res = ast_manager_register2( "PlayDTMF", EVENT_FLAG_CALL, manager_play_dtmf, "Play DTMF signal on a specific channel.", mandescr_playdtmf );
-	res |= ast_register_application(app, senddtmf_exec, synopsis, descrip);
-
+	STANDARD_USECOUNT(res);
 	return res;
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Send DTMF digits Application");
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_sendtext.c

@@ -1,99 +1,94 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * App to transmit a text message
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief App to transmit a text message
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \note Requires support of sending text messages from channel driver
- *
- * \ingroup applications
+ * the GNU General Public License
  */
  
-#include "asterisk.h"
+#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>
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+static char *tdesc = "Send Text Applications";
 
-#include "asterisk/file.h"
-#include "asterisk/channel.h"
-#include "asterisk/pbx.h"
-#include "asterisk/module.h"
-#include "asterisk/app.h"
+static char *app = "SendText";
 
-static const char *app = "SendText";
+static char *synopsis = "Send a Text Message";
 
-static const 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";
 
-static const char *descrip = 
-"  SendText(text): Sends text to current channel (callee).\n"
-"Result of transmission will be stored in the SENDTEXTSTATUS\n"
-"channel variable:\n"
-"      SUCCESS      Transmission succeeded\n"
-"      FAILURE      Transmission failed\n"
-"      UNSUPPORTED  Text transmission not supported by channel\n"
-"\n"
-"At this moment, text is supposed to be 7 bit ASCII in most channels.\n";
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int sendtext_exec(struct ast_channel *chan, void *data)
 {
 	int res = 0;
-	char *status = "UNSUPPORTED";
-	char *parse = NULL;
-	AST_DECLARE_APP_ARGS(args,
-		AST_APP_ARG(text);
-	);
-
-	/* NOT ast_strlen_zero, because some protocols (e.g. SIP) MUST be able to
-	 * send a zero-length message. */
-	if (!data) {
+	struct localuser *u;
+	if (!data || !strlen((char *)data)) {
 		ast_log(LOG_WARNING, "SendText requires an argument (text)\n");
 		return -1;
-	} else
-		parse = ast_strdupa(data);
-	
-	AST_STANDARD_APP_ARGS(args, parse);
-
-	ast_channel_lock(chan);
-	if (!chan->tech->send_text) {
-		ast_channel_unlock(chan);
+	}
+	LOCAL_USER_ADD(u);
+	ast_mutex_lock(&chan->lock);
+	if (!chan->pvt->send_text) {
+		ast_mutex_unlock(&chan->lock);
 		/* Does not support transport */
-		pbx_builtin_setvar_helper(chan, "SENDTEXTSTATUS", status);
+		if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+			chan->priority += 100;
+		LOCAL_USER_REMOVE(u);
 		return 0;
 	}
-	status = "FAILURE";
-	ast_channel_unlock(chan);
-	res = ast_sendtext(chan, args.text);
-	if (!res)
-		status = "SUCCESS";
-	pbx_builtin_setvar_helper(chan, "SENDTEXTSTATUS", status);
-	return 0;
+	ast_mutex_unlock(&chan->lock);
+	res = ast_sendtext(chan, (char *)data);
+	LOCAL_USER_REMOVE(u);
+	return res;
 }
 
-static int unload_module(void)
+int unload_module(void)
 {
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
-static int load_module(void)
+int load_module(void)
 {
 	return ast_register_application(app, sendtext_exec, synopsis, descrip);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Send Text Applications");
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

apps/app_setcallerid.c

@@ -1,51 +1,55 @@
 /*
- * Asterisk -- An open source telephony toolkit.
+ * Asterisk -- A telephony toolkit for Linux.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * App to set callerid
+ * 
+ * Copyright (C) 1999, Mark Spencer
  *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
+ * Mark Spencer <markster@linux-support.net>
  *
  * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief App to set callerid presentation
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
+ * the GNU General Public License
  */
  
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/lock.h"
-#include "asterisk/file.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/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/image.h>
+#include <asterisk/callerid.h>
+#include <string.h>
+#include <stdlib.h>
 
 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 a call.\n"
-"  Valid presentations are:\n"
+"  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" 
@@ -61,37 +65,95 @@ static char *descrip2 =
 
 static int setcallerid_pres_exec(struct ast_channel *chan, void *data)
 {
+	int res = 0;
+	char tmp[256] = "";
+	struct localuser *u;
+	int x;
+	char *opts;
 	int pres = -1;
-	static int deprecated = 0;
-
-	if (!deprecated) {
-		deprecated = 1;
-		ast_log(LOG_WARNING, "SetCallerPres is deprecated.  Please use Set(CALLERPRES()=%s) instead.\n", (char *)data);
+	if (data)
+		strncpy(tmp, (char *)data, sizeof(tmp) - 1);
+	opts = strchr(tmp, '|');
+	if (opts) {
+		*opts = '\0';
+		opts++;
 	}
-
-	/* For interface consistency, permit the argument to be specified as a number */
-	if (sscanf(data, "%30d", &pres) != 1 || pres < 0 || pres > 255 || (pres & 0x9c)) {
-		pres = ast_parse_caller_presentation(data);
+	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",
-			(char *) data);
+		ast_log(LOG_WARNING, "'%s' is not a valid presentation (see 'show application SetCallerPres')\n", tmp);
 		return 0;
 	}
-	
-	chan->cid.cid_pres = pres;
-	return 0;
+	LOCAL_USER_ADD(u);
+	chan->callingpres = pres;
+	LOCAL_USER_REMOVE(u);
+	return res;
 }
 
-static int unload_module(void)
+
+
+static char *tdesc = "Set CallerID Application";
+
+static char *app = "SetCallerID";
+
+static char *synopsis = "Set CallerID";
+
+static char *descrip = 
+"  SetCallerID(clid[|a]): Set Caller*ID on a call to a new\n"
+"value.  Sets ANI as well if a flag is used.  Always returns 0\n";
+
+static int setcallerid_exec(struct ast_channel *chan, void *data)
 {
-	return ast_unregister_application(app2);
+	int res = 0;
+	char tmp[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);
+	ast_set_callerid(chan, strlen(tmp) ? tmp : NULL, anitoo);
+	LOCAL_USER_REMOVE(u);
+	return res;
 }
 
-static int load_module(void)
+int unload_module(void)
 {
-	return ast_register_application(app2, setcallerid_pres_exec, synopsis2, descrip2);
+	STANDARD_HANGUP_LOCALUSERS;
+	ast_unregister_application(app2);
+	return ast_unregister_application(app);
 }
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Set CallerID Presentation Application");
+int load_module(void)
+{
+	ast_register_application(app2, setcallerid_pres_exec, synopsis2, descrip2);
+	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_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

@@ -0,0 +1,107 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * App to set callerid
+ * 
+ * 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 <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 Name";
+
+static char *app = "SetCIDName";
+
+static char *synopsis = "Set CallerID Name";
+
+static char *descrip = 
+"  SetCIDName(cname[|a]): Set Caller*ID Name on a call to a new\n"
+"value, while preserving the original Caller*ID number.  This is\n"
+"useful for providing additional information to the called\n"
+"party. Sets ANI as well if a flag is used.  Always returns 0\n";
+
+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);
+		n = tmp;
+		if (!ast_strlen_zero(n)) {
+			if (l && !ast_strlen_zero(l))
+				snprintf(newcid, sizeof(newcid), "\"%s\" <%s>", n, l);
+			else
+				strncpy(newcid, tmp, sizeof(newcid) - 1);
+		} else if (l && !ast_strlen_zero(l)) {
+			strncpy(newcid, l, sizeof(newcid) - 1);
+		}
+	} else
+		strncpy(newcid, tmp, sizeof(newcid) - 1);
+	ast_set_callerid(chan, !ast_strlen_zero(newcid) ? newcid : NULL, anitoo);
+	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_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;
+}