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Author SHA1 Message Date
Kevin P. Fleming
836e861dde remove extraneous svn:executable properties
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/0.1.3@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29 18:24:39 +00:00
Kevin P. Fleming
04ec7f1429 automatic tag renames
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/0.1.3@7201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-27 17:18:34 +00:00
Admin Commit
35e505b579 This commit was manufactured by cvs2svn to create tag 'v0-1-3'.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/v0-1-3@210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2000-05-02 19:18:58 +00:00
1139 changed files with 28240 additions and 304423 deletions

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1.2.36

29
BUGS
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Asterisk Bug Tracking Information
=================================
* EVERYTHING MARKED WITH "XXX" IN THE SOURCE REPRESENTS A BUG! Sometimes
these bugs are in asterisk, and sometimes they relate to the products
that asterisk uses.
To learn about and report Asterisk bugs, please visit
the official Asterisk Bug Tracker at:
* The translator API may introduce warble in the case of going in both
directions, but I haven't verified that. The trouble should only enter
in the case of mismatched frame lengths.
http://bugs.digium.com
For more information on using the bug tracker, or to
learn how you can contribute by acting as a bug marshall
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
Thank you!
* In general Asterisk is a very new program, and there are liable to be
many bugs yet to be discovered, so if you think you've found one, please
be sure to report it.

163
CHANGES
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Changes since Asterisk 1.2.0-beta2:
* Cygwin build system portability
* Optional generation of outbound silence during channel recording
Changes since Asterisk 1.2.0-beta1:
* Many, many bug fixes
* Documentation and sample configuration updates
* Vastly improved presence/subscription support in the SIP channel driver
* A new (experimental) mISDN channel driver
* A new monitoring application (MixMonitor)
* More portability fixes for non-Linux platforms
* New dialplan functions replacing old applications
* Significant deadlock and performance upgrades for the Manager interface
* An upgrade to the 'new' dialplan expression parser for all users
* New Zaptel echo cancellers with improved performance
* Support for the latest OSP toolkit from TransNexus
* Support user-controlled volume adjustment in MeetMe application
* More dialplan applications now return status variables instead of priority jumping
* Much more powerful ENUM support in the dialplan
* SIP domain support for authentication and virtual hosting
* Many PRI protocol updates and fixes, including more complete Q.SIG support
* New applications: Pickup() and Page()
Changes since Asterisk 1.0:
This list currently only containts changes made from the end of November until
March 26, 2005.
* Add new applications:
-- AgentMonitorOutgoing
-- Curl
-- ExecIf
-- ExecIfTime
-- IAX2Provision
-- MacroExit
-- MacroIf
-- PauseQueueMember
-- ReadFile
-- SetRDNIS
-- SIPAddHeader
-- SIPGetHeader
-- StartMusicOnHold
-- StopMusicOnHold
-- UnpauseQueueMember
-- WaitForSilence
-- While / EndWhile
* app Answer
-- added delay option
* app ChanIsAvail
-- added 's' option
* app Dial
-- add option to specify the class for musiconhold with m option
* app EnumLookup
-- added "reload enum" for configuration
* app Goto
-- added relative priorities
* app GotoIf
-- added relative priorities
* app MeetMe
-- added 'i' option
-- added 'r' option
-- added 'T' option
-- added 'P' option
-- added 'c' option
-- added adminpin to meetme.conf
-- added reload command
* app PrivacyManager
-- add config file privacy.conf
* app queue
-- queues.conf
-- added persistentmembers option to queues.conf
-- changed music option to musiconhold
-- added weight option
-- added note about why agent groups probably shouldn't be used
-- added timeoutrestart option
* app Read
-- added attempts parameter
-- added timeout parameter
* app Record
-- added 'q' option
* app SendDTMF
-- add timeout option
* app SMS
-- document alternative syntax for queueing messages
* app Voicemail
-- add info about VM_CATEGORY
-- voicemail.conf
-- added usedirectory option
-- added VM_CIDNUM and VM_CIDNAME in message config
* chan IAX2
-- new jitterbuffer
-- added setvar option
-- added regex to iax2 show peers/users
-- allow multiple bindaddr lines in iax.conf
-- added reload command
-- added forcejitterbuffer option
-- added note about specifying bindport before bindaddr
-- added trunktimestamps option
* chan Agent
-- added agent logoff CLI command
* chan OSS
-- added Flash CLI command
* chan SIP
-- added setvar option
-- added compactheaders option
-- added usereqphone option
-- added registertimeout option
-- added externhost option
-- added sip notify CLI command
-- added sip_notify.conf
-- added allowguest option
* chan Zap
-- added hanguponplarityswitch option
-- added sendcalleridafter option
-- added priresetinterval option
-- added TON/NPI config options (the ones right above the resetinterval option)
-- added answeronpolarityswitch option
-- added "never" for resetinterval
* extensions
-- allow '*' when including files (#include "sip-*.conf")
-- added eswitch
* General
-- added #exec syntax for including output from a command
-- added show features CLI command
-- added configuration templates for category inheritance
* Asterisk 0.1.3
-- Update VoFR for latest sangoma code
-- Update QuickNet Driver
-- Add text message handling
-- Fix transfers to use "default" if not in current context
-- Add call parking
-- Improve format/content negotiation
-- Added support for multiple languages
-- Bug fixes, as always...
* Asterisk 0.1.2
-- Updated README file with a "Getting Started" section
-- Added sample sounds and configuration files.
-- Added LPC10 very low bandwidth (low quality) compression
-- Enhanced translation selection mechanism.
-- Enhanced IAX jitter buffer, improved reliability
-- Support echo cancelation on PhoneJack
-- Updated PhoneJack driver to std. Telephony interface
-- Added app_echo for evaluating VoIP latency
-- Added app_system to execute arbitrary programs
-- Updated sample configuration files
-- Added OSS channel driver (full duplex only)
-- Added IAX implementation
-- Fixed some deadlocks.
-- A whole bunch of bug fixes
* Asterisk 0.1.1
-- Revised translator, fixed some general race conditions throughout *
-- Made dialer somewhat more aware of incompatible voice channels
-- Added Voice Modem driver and A/Open Modem Driver stub
-- Added MP3 decoder channel
-- Added Microsoft WAV49 support
-- Revised License -- Pure GPL, nothing else
-- Modified Copyright statement since code is still currently owned by author
-- Added RAW GSM headerless data format
-- Innumerable bug fixes
* Asterisk 0.1.0
-- Initial Release

341
COPYING
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GNU GENERAL PUBLIC LICENSE
Version 2, June 1991
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This General Public License does not permit incorporating your program into
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Public License instead of this License.

105
CREDITS
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@@ -1,105 +1,14 @@
=== DEVELOPMENT SUPPORT ===
We'd like to thank the following companies for helping fund development of
Asterisk:
Pilosoft, Inc. - for supporting ADSI development in Asterisk
Asterlink, Inc. - for supporting broad Asterisk development
GFS - for supporting ALSA development
Telesthetic - for supporting SIP development
Christos Ricudis - for substantial code contributions
nic.at - ENUM support in Asterisk
Paul Bagyenda, Digital Solutions - for initial Voicetronix driver development
=== WISHLIST CONTRIBUTERS ===
Jeremy McNamara - SpeeX support
Nick Seraphin - RDNIS support
Gary - Phonejack ADSI (in progress)
Wasim - Hangup detect
=== HARDWARE DONORS ===
* Special thanks to Adtran, Inc, for their donation of an Adtran Atlas,
and TSU 120e to the project. (http://www.adtran.com)
* Thanks to QuickNet Technologies for their donation of an Internet
PhoneJack and Linejack card to the project. (http://www.quicknet.net)
=== MISCELLANEOUS PATCHES ===
Jim Dixon - Zapata Telephony and app_rpt
http://www.zapatatelephony.org/app_rpt.html
Russell Bryant - Asterisk 1.0 maintainer and misc. enhancements
russelb@clemson.edu
Anthony Minessale II - Countless big and small fixes, and relentless forward push
ChanSpy, ForkCDR, ControlPlayback, While/EndWhile, DumpChan, Dictate,
MacroIf, ExecIf, ExecIfTime, RetryDial, MixMonitor applications; res_odbc;
many realtime concepts and implementation pieces, including res_config_odbc;
format_slin; cdr_custom; several features in Dial including L(), G() and
enhancements to M() and D(); several CDR enhancements including CDR variables;
attended transfer; one touch record; native MOH; manager eventmask; command
line '-t' flag to allow recording/voicemail on nfs shares; #exec command and
multiline comments in config files; setvar in iax and sip configs.
anthmct@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@esoft.pf http://www.esoft.pf
William Jordan / Vonage - MySQL enhancements to Voicemail
wjordan@vonage.com
Jac Kersing - Various fixes
Steven Critchfield - Seek and Trunc functions for playback and recording
critch@basesys.com
Jefferson Noxon - app_lookupcidname, app_db, and various other contributions
Klaus-Peter Junghanns - in-band DTMF on SIP and MGCP
Ross Finlayson - Dynamic RTP payload support
Mahmut Fettahlioglu - Audio recording, music-on-hold changes, alaw file
format, and various fixes. Can be contacted at mahmut@oa.com.au
James Dennis - Cisco SIP compatibility patches to work with SIP service
providers. Can be contacted at asterisk@jdennis.net
Tilghman Lesher - ast_localtime(); ast_say_date_with_format();
GotoIfTime, Random, SayUnixTime, HasNewVoicemail applications;
CUT, SORT, EVAL, CURL, FIELDQTY, STRFTIME, QUEUEAGENT* functions;
and other innumerable bug fixes. http://asterisk.drunkcoder.com/
Jayson Vantuyl - Manager protocol changes, various other bugs.
jvantuyl@computingedge.net
Thorsten Lockert - OpenBSD, FreeBSD ports, making MacOS X port run on 10.3,
dialplan include verification, route lookup on OpenBSD, various other
bugs. tholo@sigmasoft.com
Brian West - ODBC support and Bug Marshaling
Josh Roberson - chan_zap reload support, Advanced Voicemail Features, other misc. patches,
and Bug Marshalling. - josh@asteriasgi.com, http://www.asteriasgi.com
William Waites - syslog support, SIP NAT traversal for SIP-UA. ww@styx.org
Rich Murphey - Porting to FreeBSD, NetBSD, OpenBSD, and Darwin.
rich@whiteoaklabs.com http://whiteoaklabs.com
Simon Lockhart - Porting to Solaris (based on work of Logan ???)
simon@slimey.org
Olle E. Johansson - SIP RFC compliance, documentation and testing, testing, testing
oej@edvina.net, http://edvina.net
Steve Kann - new jitter buffer for IAX2
stevek@stevek.com
Constantine Filin - major contributions to the Asterisk Realtime Architecture
Steve Murphy - privacy support
Claude Patry - bug fixes, feature enhancements, and bug marshalling
cpatry@gmail.com
=== OTHER CONTRIBUTIONS ===
John Todd - Monkey sounds and associated teletorture prompt
Michael Jerris - bug marshaling
Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
available under a Creative Commons License at http://www.asteriskdocs.org
=== HOLD MUSIC ===
Music provided by www.opsound.org
PhoneJack card to the project. (http://www.quicknet.net)
=== OTHER SOURCE CODE IN ASTERISK ===
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,
@@ -124,7 +33,7 @@ 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,

6899
ChangeLog

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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

389
LICENSE
View File

@@ -1,68 +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. Most of these components are also
distributed under the GPL version 2 as well, except for the following:
GNU GENERAL PUBLIC LICENSE
Version 2, June 1991
contrib/firmware/iax/iaxy.bin:
This file is Copyright (C) Digium, Inc. and is licensed for
use with Digium IAXy hardware devices only. It can be
distributed freely as long as the distribution is in the
original form present in this package (not reformatted or
modified).
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.
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.
Preamble
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).
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.
Specific permission is also granted to link Asterisk with OpenSSL and
OpenH323 and distribute the resulting binary files.
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.
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.
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.
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 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.
+1.877.546.8963 (via telephone in the USA)
+1.256.428.6000 (via telephone outside the USA)
+1.256.864.0464 (via FAX inside or outside the USA)
IAX2/pbx.digium.com (via IAX2)
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.
150 West Park Loop
Suite 100
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.
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TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
0. This License applies to any program or other work which contains
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END OF TERMS AND CONDITIONS
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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
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<one line to give the program's name and a brief idea of what it does.>
Copyright (C) 19yy <name of author>
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Also add information on how to contact you by electronic and paper mail.
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Gnomovision version 69, Copyright (C) 19yy name of author
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You should also get your employer (if you work as a programmer) or your
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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.

934
Makefile
View File

@@ -3,424 +3,34 @@
#
# Top level Makefile
#
# Copyright (C) 1999-2005, Mark Spencer
# 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
#
.EXPORT_ALL_VARIABLES:
# Create OPTIONS variable
OPTIONS=
# If cross compiling, define these to suit
# CROSS_COMPILE=/opt/montavista/pro/devkit/arm/xscale_be/bin/xscale_be-
# CROSS_COMPILE_BIN=/opt/montavista/pro/devkit/arm/xscale_be/bin/
# CROSS_COMPILE_TARGET=/opt/montavista/pro/devkit/arm/xscale_be/target
CC=$(CROSS_COMPILE)gcc
HOST_CC=gcc
# CROSS_ARCH=Linux
# CROSS_PROC=arm
# SUB_PROC=xscale # or maverick
ifeq ($(CROSS_COMPILE),)
OSARCH=$(shell uname -s)
OSREV=$(shell uname -r)
OSCPU=$(shell uname -m)
else
OSARCH=$(CROSS_ARCH)
OSREV=$(CROSS_REV)
endif
# Remember the MAKELEVEL at the top
MAKETOPLEVEL?=$(MAKELEVEL)
PWD=$(shell pwd)
ifneq ($(findstring dont-optimize,$(MAKECMDGOALS)),dont-optimize)
######### More GSM codec optimization
######### Uncomment to enable MMXTM optimizations for x86 architecture CPU's
######### which support MMX instructions. This should be newer pentiums,
######### ppro's, etc, as well as the AMD K6 and K7.
#K6OPT = -DK6OPT
#Tell gcc to optimize the code
OPTIMIZE+=-O6
endif
#Overwite config files on "make samples"
OVERWRITE=y
#Include debug and macro symbols in the executables (-g) and profiling info (-pg)
DEBUG=-g3 #-pg
#Set NOCRYPTO to yes if you do not want to have crypto support or
#dependencies
#NOCRYPTO=yes
# If you are running a radio application, define RADIO_RELAX so that the DTMF
# will be received more reliably
#OPTIONS += -DRADIO_RELAX
# If you don't have a lot of memory (e.g. embedded Asterisk), define LOW_MEMORY
# to reduce the size of certain static buffers
#ifneq ($(CROSS_COMPILE),)
#OPTIONS += -DLOW_MEMORY
#endif
# Optional debugging parameters
DEBUG_THREADS = #-DDUMP_SCHEDULER #-DDEBUG_SCHEDULER #-DDEBUG_THREADS #-DDO_CRASH #-DDETECT_DEADLOCKS
# Uncomment next one to enable ast_frame tracing (for debugging)
TRACE_FRAMES = #-DTRACE_FRAMES
# Uncomment next one to enable malloc debugging
# You can view malloc debugging with:
# *CLI> show memory allocations [filename]
# *CLI> show memory summary [filename]
#
MALLOC_DEBUG = #-include $(PWD)/include/asterisk/astmm.h
INSTALL=install
# Where to install asterisk after compiling
# Default -> leave empty
INSTALL_PREFIX?=
# Staging directory
# Files are copied here temporarily during the install process
# For example, make DESTDIR=/tmp/asterisk woud put things in
# /tmp/asterisk/etc/asterisk
# XXX watch out, put no spaces or comments after the value
DESTDIR?=
#DESTDIR?=/tmp/asterisk
# Original busydetect routine
BUSYDETECT = #-DBUSYDETECT
# Improved busydetect routine, comment the previous one if you use this one
BUSYDETECT+= #-DBUSYDETECT_MARTIN
# Detect the busy signal looking only at tone lengths
# For example if you have 3 beeps 100ms tone, 100ms silence separated by 500 ms of silence
BUSYDETECT+= #-DBUSYDETECT_TONEONLY
# Inforce the detection of busy singal (get rid of false hangups)
# Don't use together with -DBUSYDETECT_TONEONLY
BUSYDETECT+= #-DBUSYDETECT_COMPARE_TONE_AND_SILENCE
ifneq ($(OSARCH),SunOS)
ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk
ASTVARLIBDIR=$(INSTALL_PREFIX)/var/lib/asterisk
ASTETCDIR=$(INSTALL_PREFIX)/etc/asterisk
ASTSPOOLDIR=$(INSTALL_PREFIX)/var/spool/asterisk
ASTLOGDIR=$(INSTALL_PREFIX)/var/log/asterisk
ASTHEADERDIR=$(INSTALL_PREFIX)/usr/include/asterisk
ASTCONFPATH=$(ASTETCDIR)/asterisk.conf
ASTBINDIR=$(INSTALL_PREFIX)/usr/bin
ASTSBINDIR=$(INSTALL_PREFIX)/usr/sbin
ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run
ASTMANDIR=$(INSTALL_PREFIX)/usr/share/man
MODULES_DIR=$(ASTLIBDIR)/modules
AGI_DIR=$(ASTVARLIBDIR)/agi-bin
else
ASTLIBDIR=$(INSTALL_PREFIX)/opt/asterisk/lib
ASTVARLIBDIR=$(INSTALL_PREFIX)/var/opt/asterisk/lib
ASTETCDIR=$(INSTALL_PREFIX)/etc/opt/asterisk
ASTSPOOLDIR=$(INSTALL_PREFIX)/var/opt/asterisk/spool
ASTLOGDIR=$(INSTALL_PREFIX)/var/opt/asterisk/log
ASTHEADERDIR=$(INSTALL_PREFIX)/opt/asterisk/usr/include/asterisk
ASTCONFPATH=$(ASTETCDIR)/asterisk.conf
ASTBINDIR=$(INSTALL_PREFIX)/opt/asterisk/usr/bin
ASTSBINDIR=$(INSTALL_PREFIX)/opt/asterisk/usr/sbin
ASTVARRUNDIR=$(INSTALL_PREFIX)/var/opt/asterisk/run
ASTMANDIR=$(INSTALL_PREFIX)/opt/asterisk/usr/share/man
MODULES_DIR=$(ASTLIBDIR)/modules
AGI_DIR=$(ASTVARLIBDIR)/agi-bin
endif
ASTCFLAGS=-Wformat-security
MODULES_DIR=/usr/lib/asterisk/modules
# Pentium Pro Optimize
#PROC=i686
# Pentium Optimize
PROC=i586
# Pentium & VIA processors optimize
#PROC=i586
#PROC=k6
#PROC=ppc
#Uncomment this to use the older DSP routines
#ASTCFLAGS+=-DOLD_DSP_ROUTINES
# Determine by a grep 'DocumentRoot' of your httpd.conf file
HTTP_DOCSDIR=/var/www/html
# Determine by a grep 'ScriptAlias' of your httpd.conf file
HTTP_CGIDIR=/var/www/cgi-bin
# If the file .asterisk.makeopts is present in your home directory, you can
# include all of your favorite Makefile options so that every time you download
# a new version of Asterisk, you don't have to edit the makefile to set them.
# The file, /etc/asterisk.makeopts will also be included, but can be overridden
# by the file in your home directory.
ifneq ($(wildcard /etc/asterisk.makeopts),)
include /etc/asterisk.makeopts
endif
ifneq ($(wildcard ~/.asterisk.makeopts),)
include ~/.asterisk.makeopts
endif
ifeq ($(OSARCH),Linux)
ifeq ($(CROSS_COMPILE),)
PROC?=$(shell uname -m)
else
PROC=$(CROSS_PROC)
endif
ifeq ($(PROC),x86_64)
# You must have GCC 3.4 to use k8, otherwise use athlon
PROC=k8
#PROC=athlon
endif
ifeq ($(PROC),sparc64)
#The problem with sparc is the best stuff is in newer versions of gcc (post 3.0) only.
#This works for even old (2.96) versions of gcc and provides a small boost either way.
#A ultrasparc cpu is really v9 but the stock debian stable 3.0 gcc doesn't support it.
#So we go lowest common available by gcc and go a step down, still a step up from
#the default as we now have a better instruction set to work with. - Belgarath
PROC=ultrasparc
OPTIONS+=$(shell if $(CC) -mtune=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-mtune=$(PROC)"; fi)
OPTIONS+=$(shell if $(CC) -mcpu=v8 -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-mcpu=v8"; fi)
OPTIONS+=-fomit-frame-pointer
endif
ifeq ($(PROC),arm)
# The Cirrus logic is the only heavily shipping arm processor with a real floating point unit
ifeq ($(SUB_PROC),maverick)
OPTIONS+=-fsigned-char -mcpu=ep9312
else
ifeq ($(SUB_PROC),xscale)
OPTIONS+=-fsigned-char -mcpu=xscale
else
OPTIONS+=-fsigned-char
endif
endif
endif
MPG123TARG=linux
endif
GREP=grep
LN=ln
ID=id
ifeq ($(OSARCH),SunOS)
GREP=/usr/xpg4/bin/grep
M4=/usr/local/bin/m4
ID=/usr/xpg4/bin/id
LN=/usr/xpg4/bin/ln
INSTALL=ginstall
endif
INCLUDE+=-Iinclude -I../include
ASTCFLAGS+=-pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE #-DMAKE_VALGRIND_HAPPY
ASTCFLAGS+=$(OPTIMIZE)
ASTOBJ=-o asterisk
ifeq ($(findstring BSD,$(OSARCH)),BSD)
PROC=$(shell uname -m)
ASTCFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
ifneq ($(PROC),ultrasparc)
ASTCFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
endif
ifeq ($(PROC),ppc)
ASTCFLAGS+=-fsigned-char
endif
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/osp/osp.h),)
ASTCFLAGS+=-DOSP_SUPPORT -I$(CROSS_COMPILE_TARGET)/usr/local/include/osp
else
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
ASTCFLAGS+=-DOSP_SUPPORT -I$(CROSS_COMPILE_TARGET)/usr/include/osp
endif
endif
ifeq ($(OSARCH),FreeBSD)
BSDVERSION=$(shell make -V OSVERSION -f $(CROSS_COMPILE_TARGET)/usr/share/mk/bsd.port.subdir.mk)
ASTCFLAGS+=$(shell if test $(BSDVERSION) -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
LIBS+=$(shell if test $(BSDVERSION) -lt 502102 ; then echo "-lc_r"; else echo "-pthread"; fi)
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/spandsp),)
ASTCFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include/spandsp
endif
MPG123TARG=freebsd
# XXX FreeBSD paths
PREFIX?=/usr/local
ASTLIBDIR=$(INSTALL_PREFIX)$(PREFIX)/lib/asterisk
ASTVARLIBDIR=$(INSTALL_PREFIX)$(PREFIX)/share/asterisk
ASTETCDIR=$(INSTALL_PREFIX)$(PREFIX)/etc/asterisk
ASTSPOOLDIR=$(INSTALL_PREFIX)/var/spool/asterisk
ASTLOGDIR=$(INSTALL_PREFIX)/var/log/asterisk
ASTHEADERDIR=$(INSTALL_PREFIX)$(PREFIX)/include/asterisk
ASTCONFPATH=$(ASTETCDIR)/asterisk.conf
ASTBINDIR=$(INSTALL_PREFIX)$(PREFIX)/bin
ASTSBINDIR=$(INSTALL_PREFIX)$(PREFIX)/sbin
ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run
ASTMANDIR=$(INSTALL_PREFIX)$(PREFIX)/man
# XXX end FreeBSD paths
endif # FreeBSD
ifeq ($(OSARCH),NetBSD)
ASTCFLAGS+=-pthread
INCLUDE+=-I$(CROSS_COMPILE_TARGET)/usr/pkg/include
MPG123TARG=netbsd
endif
ifeq ($(OSARCH),OpenBSD)
ASTCFLAGS+=-pthread
endif
ifeq ($(OSARCH),SunOS)
ASTCFLAGS+=-Wcast-align -DSOLARIS
INCLUDE+=-Iinclude/solaris-compat -I$(CROSS_COMPILE_TARGET)/opt/ssl/include -I$(CROSS_COMPILE_TARGET)/usr/local/ssl/include
ifeq ($(OSCPU),sun4u)
OPTIMIZE+=-mcpu=v9 -mcpu=ultrasparc
endif
endif
ifeq ($(findstring CYGWIN,$(OSARCH)),CYGWIN)
CYGLOADER=cygwin_a
OSARCH=CYGWIN
ASTOBJ=-shared -o asterisk.dll -Wl,--out-implib=libasterisk.dll.a -Wl,--export-all-symbols
ASTLINK=
LIBS+=-lpthread -lncurses -lm -lresolv
ASTSBINDIR=$(MODULES_DIR)
PROC=$(shell uname -m)
endif
ifndef WITHOUT_ZAPTEL
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/zaptel.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/zaptel.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/pkg/include/zaptel.h),)
ASTCFLAGS+=-DZAPTEL_OPTIMIZATIONS
endif
endif # WITHOUT_ZAPTEL
LIBEDIT=editline/libedit.a
ifneq ($(wildcard .version),)
ASTERISKVERSION:=$(shell cat .version)
ASTERISKVERSIONNUM:=$(shell awk -F. '{printf "%01d%02d%02d", $$1, $$2, $$3}' .version)
RPMVERSION:=$(shell sed 's/[-\/:]/_/g' .version)
else
RPMVERSION=unknown
endif
# CVS mirrors of SVN have .svnrevision files showing
# which SVN revision they are based on, and .svnbranch
# showing the branch they are made from
ifneq ($(wildcard .svnrevision),)
ASTERISKVERSIONNUM=999999
ASTERISKVERSION:=SVN-$(shell cat .svnbranch)-r$(shell cat .svnrevision)
else
ifneq ($(wildcard .svn),)
ASTERISKVERSIONNUM=999999
ASTERISKVERSION=SVN-$(shell build_tools/make_svn_branch_name)
endif
endif
ASTCFLAGS+= $(DEBUG_THREADS)
ASTCFLAGS+= $(TRACE_FRAMES)
ASTCFLAGS+= $(MALLOC_DEBUG)
ASTCFLAGS+= $(BUSYDETECT)
ASTCFLAGS+= $(OPTIONS)
ifneq ($(findstring dont-optimize,$(MAKECMDGOALS)),dont-optimize)
ASTCFLAGS+= -fomit-frame-pointer
endif
SUBDIRS=res channels pbx apps codecs formats agi cdr funcs utils stdtime
OBJS=io.o sched.o logger.o frame.o loader.o config.o channel.o \
translate.o file.o say.o pbx.o cli.o md5.o term.o \
ulaw.o alaw.o callerid.o fskmodem.o image.o app.o \
cdr.o tdd.o acl.o rtp.o manager.o asterisk.o \
dsp.o chanvars.o indications.o autoservice.o db.o privacy.o \
astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o \
utils.o plc.o jitterbuf.o dnsmgr.o devicestate.o \
netsock.o slinfactory.o ast_expr2.o ast_expr2f.o \
cryptostub.o astobj2.o sha1.o
ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/sys/poll.h),)
OBJS+= poll.o
ASTCFLAGS+=-DPOLLCOMPAT
endif
ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/dlfcn.h),)
OBJS+= dlfcn.o
ASTCFLAGS+=-DDLFCNCOMPAT
endif
ifeq ($(OSARCH),Linux)
LIBS+=-ldl -lpthread -lncurses -lm -lresolv #-lnjamd
else
ifeq ($(OSARCH),SunOS)
LIBS+=-lm -lcurses
else
LIBS+=-lncurses -lm
endif
endif
ifeq ($(OSARCH),Darwin)
LIBS+=-lresolv
ASTCFLAGS+=-D__Darwin__
AUDIO_LIBS=-framework CoreAudio
ASTLINK=-Wl,-dynamic
SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
OBJS+= poll.o
ASTCFLAGS+=-DPOLLCOMPAT
else
#These are used for all but Darwin
ASTLINK=-Wl,-E
SOLINK=-shared -Xlinker -x
endif
ifeq ($(OSARCH),FreeBSD)
LIBS+=-lcrypto
endif
ifeq ($(OSARCH),NetBSD)
LIBS+=-lpthread -lcrypto -lm -L$(CROSS_COMPILE_TARGET)/usr/pkg/lib -lncurses
endif
ifeq ($(OSARCH),OpenBSD)
LIBS+=-lcrypto -lpthread -lm -lncurses
endif
ifeq ($(OSARCH),SunOS)
LIBS+=-lpthread -ldl -lnsl -lsocket -lresolv -L$(CROSS_COMPILE_TARGET)/opt/ssl/lib -L$(CROSS_COMPILE_TARGET)/usr/local/ssl/lib
OBJS+=strcompat.o
ASTLINK=
SOLINK=-shared -fpic -L$(CROSS_COMPILE_TARGET)/opt/ssl/lib -L$(CROSS_COMPILE_TARGET)/usr/local/ssl/lib
endif
ifeq ($(MAKETOPLEVEL),$(MAKELEVEL))
CFLAGS+=$(ASTCFLAGS)
endif
# This is used when generating the doxygen documentation
ifneq ($(wildcard /usr/local/bin/dot)$(wildcard /usr/bin/dot),)
HAVEDOT=yes
else
HAVEDOT=no
endif
LIBS+=-lssl
DEBUG=-g #-pg
INCLUDE=-Iinclude -I../include
CFLAGS=-pipe -Wall -Werror -Wmissing-prototypes -Wmissing-declarations -O6 $(DEBUG) $(INCLUDE) -D_REENTRANT
CFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
CFLAGS+=$(shell [ ! -f /usr/include/linux/if_wanpipe.h ] && echo " -DOLD_SANGOMA_API")
SUBDIRS=channels pbx apps codecs formats
LIBS=-ldl -lpthread -lreadline #-lefence
OBJS=io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o asterisk.o
CC=gcc
INSTALL=install
_all: all
@echo " +--------- Asterisk Build Complete ---------+"
@@ -428,498 +38,68 @@ _all: all
@echo " + cannot be run before being installed by +"
@echo " + running: +"
@echo " + +"
@echo " + $(MAKE) install +"
@echo " + make install +"
@echo " + +"
@echo " +-------------------------------------------+"
all: cleantest depend asterisk subdirs
all: asterisk subdirs
#ifneq ($(wildcard tags),)
ctags: tags
#endif
ifneq ($(wildcard TAGS),)
all: TAGS
endif
noclean: depend asterisk subdirs
editline/config.h:
cd editline && unset CFLAGS LIBS && CFLAGS="$(OPTIMIZE)" ./configure ; \
editline/libedit.a: FORCE
cd editline && unset CFLAGS LIBS && test -f config.h || CFLAGS="$(OPTIMIZE)" ./configure
$(MAKE) -C editline libedit.a
db1-ast/libdb1.a: FORCE
@if [ -d db1-ast ]; then \
$(MAKE) -C db1-ast libdb1.a ; \
else \
echo "You need to do a cvs update -d not just cvs update"; \
exit 1; \
fi
ifneq ($(wildcard .depend),)
include .depend
endif
ifneq ($(wildcard .tags-depend),)
include .tags-depend
endif
ast_expr2.c:
bison -d --name-prefix=ast_yy ast_expr2.y -o ast_expr2.c
ast_expr2f.c:
flex --full ast_expr2.fl
testexpr2: ast_expr2f.c ast_expr2.c ast_expr2.h
gcc -g -c -DSTANDALONE ast_expr2f.c
gcc -g -c -DSTANDALONE ast_expr2.c
gcc -g -o testexpr2 ast_expr2f.o ast_expr2.o
rm ast_expr2.o ast_expr2f.o
manpage: asterisk.8
asterisk.8: asterisk.sgml
rm -f asterisk.8
docbook2man asterisk.sgml
mv ./*.8 asterisk.8
asterisk.pdf: asterisk.sgml
docbook2pdf asterisk.sgml
asterisk.ps: asterisk.sgml
docbook2ps asterisk.sgml
asterisk.html: asterisk.sgml
docbook2html asterisk.sgml
mv r1.html asterisk.html
asterisk.txt: asterisk.sgml
docbook2txt asterisk.sgml
defaults.h: FORCE
build_tools/make_defaults_h > $@.tmp
if cmp -s $@.tmp $@ ; then echo ; else \
mv $@.tmp $@ ; \
fi
rm -f $@.tmp
include/asterisk/version.h: FORCE
build_tools/make_version_h > $@.tmp
if cmp -s $@.tmp $@ ; then echo; else \
mv $@.tmp $@ ; \
fi
rm -f $@.tmp
stdtime/libtime.a: FORCE
@if [ -d stdtime ]; then \
$(MAKE) -C stdtime libtime.a ; \
else \
echo "You need to do a cvs update -d not just cvs update"; \
exit 1; \
fi
cygwin_a:
$(MAKE) -C cygwin all
asterisk: $(CYGLOADER) editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a $(OBJS)
build_tools/make_build_h > include/asterisk/build.h.tmp
if cmp -s include/asterisk/build.h.tmp include/asterisk/build.h ; then echo ; else \
mv include/asterisk/build.h.tmp include/asterisk/build.h ; \
fi
rm -f include/asterisk/build.h.tmp
$(CC) -c -o buildinfo.o $(CFLAGS) buildinfo.c
$(CC) $(DEBUG) $(ASTOBJ) $(ASTLINK) $(OBJS) buildinfo.o $(LIBEDIT) db1-ast/libdb1.a stdtime/libtime.a $(LIBS)
muted: muted.o
$(CC) $(AUDIO_LIBS) -o muted muted.o
asterisk: $(OBJS)
gcc -o asterisk -rdynamic $(OBJS) $(LIBS)
subdirs:
for x in $(SUBDIRS); do $(MAKE) -C $$x || exit 1 ; done
clean:
for x in $(SUBDIRS); do $(MAKE) -C $$x clean || exit 1 ; done
rm -f *.o *.so asterisk .depend
rm -f defaults.h
rm -f include/asterisk/build.h
rm -f include/asterisk/version.h
rm -f .tags-depend .tags-sources tags TAGS
@if [ -f editline/Makefile ]; then $(MAKE) -C editline distclean ; fi
@if [ -d mpg123-0.59r ]; then $(MAKE) -C mpg123-0.59r clean; fi
$(MAKE) -C db1-ast clean
$(MAKE) -C stdtime clean
rm -f *.o *.so asterisk
datafiles: all
if [ x`$(ID) -un` = xroot ]; then sh mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/digits
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/silence
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/priv-callerintros
for x in sounds/digits/*.gsm; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds/digits ; \
else \
echo "No description for $$x"; \
exit 1; \
fi; \
mkdir -p /var/lib/asterisk/sounds/digits
for x in sounds/digits/*; do \
install $$x /var/lib/asterisk/sounds/digits ; \
done
for x in sounds/silence/*.gsm; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds/silence ; \
else \
echo "No description for $$x"; \
exit 1; \
fi; \
for x in sounds/vm-* sounds/transfer* ; do \
install $$x /var/lib/asterisk/sounds ; \
done
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/dictate
for x in sounds/dictate/*.gsm; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds/dictate ; \
else \
echo "No description for $$x"; \
exit 1; \
fi; \
done
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/letters
for x in sounds/letters/*.gsm; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds/letters ; \
else \
echo "No description for $$x"; \
exit 1; \
fi; \
done
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/phonetic
for x in sounds/phonetic/*.gsm; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds/phonetic ; \
else \
echo "No description for $$x"; \
exit 1; \
fi; \
done
for x in sounds/demo-* sounds/vm-* sounds/transfer* sounds/pbx-* sounds/ss-* sounds/beep* sounds/dir-* sounds/conf-* sounds/agent-* sounds/invalid* sounds/tt-* sounds/auth-* sounds/privacy-* sounds/queue-* sounds/spy-* sounds/priv-* sounds/screen-* sounds/hello-* sounds/hours* sounds/minute* sounds/second* ; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds ; \
else \
echo "No description for $$x"; \
exit 1; \
fi; \
done
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/mohmp3
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/images
for x in images/*.jpg; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/images ; \
done
mkdir -p $(DESTDIR)$(AGI_DIR)
update:
@if [ -d .svn ]; then \
echo "Updating from Subversion..." ; \
svn update | tee update.out; \
rm -f .version; \
if [ `grep -c ^C update.out` -gt 0 ]; then \
echo ; echo "The following files have conflicts:" ; \
grep ^C update.out | cut -b4- ; \
fi ; \
rm -f update.out; \
elif [ -d CVS ]; then \
echo "Updating from CVS..." ; \
cvs -q -z3 update -Pd | tee update.out; \
rm -f .version; \
if [ `grep -c ^C update.out` -gt 0 ]; then \
echo ; echo "The following files have conflicts:" ; \
grep ^C update.out | cut -d' ' -f2- ; \
fi ; \
rm -f update.out; \
else \
echo "Not under version control"; \
fi
NEWHEADERS=$(notdir $(wildcard include/asterisk/*.h))
OLDHEADERS=$(filter-out $(NEWHEADERS),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
bininstall: all
mkdir -p $(DESTDIR)$(MODULES_DIR)
mkdir -p $(DESTDIR)$(ASTSBINDIR)
mkdir -p $(DESTDIR)$(ASTETCDIR)
mkdir -p $(DESTDIR)$(ASTBINDIR)
mkdir -p $(DESTDIR)$(ASTVARRUNDIR)
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/voicemail
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/dictate
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/system
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/tmp
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/meetme
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/monitor
if [ -f asterisk ]; then $(INSTALL) -m 755 asterisk $(DESTDIR)$(ASTSBINDIR)/; fi
if [ -f cygwin/asterisk.exe ]; then $(INSTALL) -m 755 cygwin/asterisk.exe $(DESTDIR)$(ASTSBINDIR)/; fi
if [ -f asterisk.dll ]; then $(INSTALL) -m 755 asterisk.dll $(DESTDIR)$(ASTSBINDIR)/; fi
$(LN) -sf asterisk $(DESTDIR)$(ASTSBINDIR)/rasterisk
$(INSTALL) -m 755 contrib/scripts/astgenkey $(DESTDIR)$(ASTSBINDIR)/
$(INSTALL) -m 755 contrib/scripts/autosupport $(DESTDIR)$(ASTSBINDIR)/
if [ ! -f $(DESTDIR)$(ASTSBINDIR)/safe_asterisk ]; then \
cat contrib/scripts/safe_asterisk | sed 's|__ASTERISK_SBIN_DIR__|$(ASTSBINDIR)|;s|__ASTERISK_VARRUN_DIR__|$(ASTVARRUNDIR)|;' > $(DESTDIR)$(ASTSBINDIR)/safe_asterisk ;\
chmod 755 $(DESTDIR)$(ASTSBINDIR)/safe_asterisk;\
fi
$(INSTALL) -d $(DESTDIR)$(ASTHEADERDIR)
$(INSTALL) -m 644 include/asterisk/*.h $(DESTDIR)$(ASTHEADERDIR)
if [ -n "$(OLDHEADERS)" ]; then \
rm -f $(addprefix $(DESTDIR)$(ASTHEADERDIR)/,$(OLDHEADERS)) ;\
fi
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds
mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-csv
mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-custom
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/keys
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/firmware
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/firmware/iax
mkdir -p $(DESTDIR)$(ASTMANDIR)/man8
$(INSTALL) -m 644 keys/iaxtel.pub $(DESTDIR)$(ASTVARLIBDIR)/keys
$(INSTALL) -m 644 keys/freeworlddialup.pub $(DESTDIR)$(ASTVARLIBDIR)/keys
$(INSTALL) -m 644 asterisk.8 $(DESTDIR)$(ASTMANDIR)/man8
$(INSTALL) -m 644 contrib/scripts/astgenkey.8 $(DESTDIR)$(ASTMANDIR)/man8
$(INSTALL) -m 644 contrib/scripts/autosupport.8 $(DESTDIR)$(ASTMANDIR)/man8
$(INSTALL) -m 644 contrib/scripts/safe_asterisk.8 $(DESTDIR)$(ASTMANDIR)/man8
if [ -d contrib/firmware/iax ]; then \
$(INSTALL) -m 644 contrib/firmware/iax/iaxy.bin $(DESTDIR)$(ASTVARLIBDIR)/firmware/iax/iaxy.bin; \
else \
echo "You need to do cvs update -d not just cvs update" ; \
fi
if [ -f mpg123-0.59r/mpg123 ]; then $(MAKE) -C mpg123-0.59r install; fi
install-subdirs:
install: all
mkdir -p $(MODULES_DIR)
for x in $(SUBDIRS); do $(MAKE) -C $$x install || exit 1 ; done
NEWMODS=$(notdir $(wildcard */*.so))
OLDMODS=$(filter-out $(NEWMODS),$(notdir $(wildcard $(DESTDIR)$(MODULES_DIR)/*.so)))
oldmodcheck:
@if [ -n "$(OLDMODS)" ]; then \
echo " WARNING WARNING WARNING" ;\
echo "" ;\
echo " Your Asterisk modules directory, located at" ;\
echo " $(DESTDIR)$(MODULES_DIR)" ;\
echo " contains modules that were not installed by this " ;\
echo " version of Asterisk. Please ensure that these" ;\
echo " modules are compatible with this version before" ;\
echo " attempting to run Asterisk." ;\
echo "" ;\
for f in $(OLDMODS); do \
echo " $$f" ;\
done ;\
echo "" ;\
echo " WARNING WARNING WARNING" ;\
fi
install: all datafiles bininstall install-subdirs
@if [ -x /usr/sbin/asterisk-post-install ]; then \
/usr/sbin/asterisk-post-install $(DESTDIR) . ; \
fi
install -d /usr/include/asterisk
install include/asterisk/*.h /usr/include/asterisk
rm -f /var/lib/asterisk/sounds/vm
mkdir -p /var/spool/asterisk/vm
rm -f /usr/lib/asterisk/modules/chan_ixj.so
mkdir -p /var/lib/asterisk/sounds
( cd /var/lib/asterisk/sounds ; ln -s ../../../spool/asterisk/vm . )
@echo " +---- Asterisk Installation Complete -------+"
@echo " + +"
@echo " + YOU MUST READ THE SECURITY DOCUMENT +"
@echo " + +"
@echo " + Asterisk has successfully been installed. +"
@echo " + If you would like to install the sample +"
@echo " + configuration files (overwriting any +"
@echo " + existing config files), run: +"
@echo " + +"
@echo " + $(MAKE) samples +"
@echo " + +"
@echo " +----------------- or ---------------------+"
@echo " + +"
@echo " + You can go ahead and install the asterisk +"
@echo " + program documentation now or later run: +"
@echo " + +"
@echo " + $(MAKE) progdocs +"
@echo " + +"
@echo " + **Note** This requires that you have +"
@echo " + doxygen installed on your local system +"
@echo " +-------------------------------------------+"
@$(MAKE) -s oldmodcheck
upgrade: all bininstall
adsi:
mkdir -p $(DESTDIR)$(ASTETCDIR)
for x in configs/*.adsi; do \
if [ ! -f $(DESTDIR)$(ASTETCDIRX)/$$x ]; then \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTETCDIR)/`basename $$x` ; \
fi ; \
done
samples: adsi
mkdir -p $(DESTDIR)$(ASTETCDIR)
for x in configs/*.sample; do \
if [ -f $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` ]; then \
if [ "$(OVERWRITE)" = "y" ]; then \
if cmp -s $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` $$x ; then \
echo "Config file $$x is unchanged"; \
continue; \
fi ; \
mv -f $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample`.old ; \
else \
echo "Skipping config file $$x"; \
continue; \
fi ;\
fi ; \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` ;\
done
if [ "$(OVERWRITE)" = "y" ] || [ ! -f $(DESTDIR)$(ASTCONFPATH) ]; then \
( \
echo "[directories]" ; \
echo "astetcdir => $(ASTETCDIR)" ; \
echo "astmoddir => $(MODULES_DIR)" ; \
echo "astvarlibdir => $(ASTVARLIBDIR)" ; \
echo "astagidir => $(AGI_DIR)" ; \
echo "astspooldir => $(ASTSPOOLDIR)" ; \
echo "astrundir => $(ASTVARRUNDIR)" ; \
echo "astlogdir => $(ASTLOGDIR)" ; \
echo "" ; \
echo "; Changing the following lines may compromise your security." ; \
echo ";[files]" ; \
echo ";astctlpermissions = 0660" ; \
echo ";astctlowner = root" ; \
echo ";astctlgroup = apache" ; \
echo ";astctl = asterisk.ctl" ; \
) > $(DESTDIR)$(ASTCONFPATH) ; \
else \
echo "Skipping asterisk.conf creation"; \
fi
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds ; \
for x in sounds/demo-*; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds ; \
else \
echo "No description for $$x"; \
exit 1; \
fi; \
done
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/mohmp3 ; \
for x in sounds/moh/*.mp3 sounds/moh/LICENSE; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/mohmp3 ; \
done
rm -f $(DESTDIR)$(ASTVARLIBDIR)/mohmp3/sample-hold.mp3
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/INBOX
:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/unavail.gsm ; \
done
:> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/busy.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isonphone; do \
cat $(DESTDIR)$(ASTVARLIBDIR)/sounds/$$x.gsm >> $(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/busy.gsm ; \
done
webvmail:
@[ -d $(DESTDIR)$(HTTP_DOCSDIR)/ ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
@[ -d $(DESTDIR)$(HTTP_CGIDIR) ] || ( printf "cgi-bin directory not found.\nUpdate assignment of variable HTTP_CGIDIR in Makefile!\n" && exit 1 )
$(INSTALL) -m 4755 -o root -g root contrib/scripts/vmail.cgi $(DESTDIR)$(HTTP_CGIDIR)/vmail.cgi
mkdir -p $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk
for x in images/*.gif; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(HTTP_DOCSDIR)/_asterisk/; \
done
@echo " +--------- Asterisk Web Voicemail ----------+"
@echo " + +"
@echo " + Asterisk Web Voicemail is installed in +"
@echo " + your cgi-bin directory: +"
@echo " + $(DESTDIR)$(HTTP_CGIDIR)"
@echo " + IT USES A SETUID ROOT PERL SCRIPT, SO +"
@echo " + IF YOU DON'T LIKE THAT, UNINSTALL IT! +"
@echo " + +"
@echo " + Other static items have been stored in: +"
@echo " + $(DESTDIR)$(HTTP_DOCSDIR)"
@echo " + +"
@echo " + If these paths do not match your httpd +"
@echo " + installation, correct the definitions +"
@echo " + in your Makefile of HTTP_CGIDIR and +"
@echo " + HTTP_DOCSDIR +"
@echo " + make samples +"
@echo " + +"
@echo " +-------------------------------------------+"
spec:
sed "s/^Version:.*/Version: $(RPMVERSION)/g" redhat/asterisk.spec > asterisk.spec ; \
rpm: __rpm
__rpm: include/asterisk/version.h spec
rm -rf /tmp/asterisk ; \
mkdir -p /tmp/asterisk/redhat/RPMS/i386 ; \
$(MAKE) DESTDIR=/tmp/asterisk install ; \
$(MAKE) DESTDIR=/tmp/asterisk samples ; \
mkdir -p /tmp/asterisk/etc/rc.d/init.d ; \
cp -f contrib/init.d/rc.redhat.asterisk /tmp/asterisk/etc/rc.d/init.d/asterisk ; \
rpmbuild --rcfile /usr/lib/rpm/rpmrc:redhat/rpmrc -bb asterisk.spec
progdocs:
(cat contrib/asterisk-ng-doxygen; echo "HAVE_DOT=$(HAVEDOT)"; \
echo "PROJECT_NUMBER=$(ASTERISKVERSION)") | doxygen -
mpg123:
@wget -V >/dev/null || (echo "You need wget" ; false )
[ -f mpg123-0.59r.tar.gz ] || wget http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz
[ -d mpg123-0.59r ] || tar xfz mpg123-0.59r.tar.gz
$(MAKE) -C mpg123-0.59r $(MPG123TARG)
config:
if [ -d /etc/rc.d/init.d ]; then \
$(INSTALL) -m 755 contrib/init.d/rc.redhat.asterisk /etc/rc.d/init.d/asterisk; \
/sbin/chkconfig --add asterisk; \
elif [ -d /etc/init.d ]; then \
echo "Only distros that use rc.d based init scripts are currently supported."; \
echo "You may be able to copy one of the scripts in contrib/init.d/ into your /etc/init.d/ directory."; \
exit 1; \
fi
dont-optimize: install
valgrind: dont-optimize
depend: include/asterisk/version.h .depend defaults.h
for x in $(SUBDIRS); do $(MAKE) -C $$x depend || exit 1 ; done
.depend: include/asterisk/version.h
build_tools/mkdep $(CFLAGS) $(wildcard *.c)
.tags-depend:
@echo -n ".tags-depend: " > $@
@find . -maxdepth 1 -name \*.c -printf "\t%p \\\\\n" >> $@
@find . -maxdepth 1 -name \*.h -printf "\t%p \\\\\n" >> $@
@find $(SUBDIRS) -name \*.c -printf "\t%p \\\\\n" >> $@
@find $(SUBDIRS) -name \*.h -printf "\t%p \\\\\n" >> $@
@find include -name \*.h -printf "\t%p \\\\\n" >> $@
@echo >> $@
.tags-sources:
@rm -f $@
@find . -maxdepth 1 -name \*.c -print >> $@
@find . -maxdepth 1 -name \*.h -print >> $@
@find $(SUBDIRS) -name \*.c -print >> $@
@find $(SUBDIRS) -name \*.h -print >> $@
@find include -name \*.h -print >> $@
tags: .tags-depend .tags-sources
ctags -L .tags-sources -o $@
ctags: tags
TAGS: .tags-depend .tags-sources
etags -o $@ `cat .tags-sources`
etags: TAGS
FORCE:
%_env:
$(MAKE) -C $(shell echo $@ | sed "s/_env//g") env
env:
env
# If the cleancount has been changed, force a make clean.
# .cleancount is the global clean count, and .lastclean is the
# last clean count we had
# We can avoid this by making noclean
cleantest:
if cmp -s .cleancount .lastclean ; then echo ; else \
$(MAKE) clean; cp -f .cleancount .lastclean;\
fi
samples: all datafiles
mkdir -p /etc/asterisk
for x in configs/*.sample; do \
if [ -f /etc/asterisk/`basename $$x .sample` ]; then \
mv -f /etc/asterisk/`basename $$x .sample` /etc/asterisk/`basename $$x .sample`.old ; \
fi ; \
install $$x /etc/asterisk/`basename $$x .sample` ;\
done
for x in sounds/demo-*; do \
install $$x /var/lib/asterisk/sounds; \
done
mkdir -p /var/spool/asterisk/vm/1234/INBOX
:> /var/lib/asterisk/sounds/vm/1234/unavail.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
cat /var/lib/asterisk/sounds/$$x.gsm >> /var/lib/asterisk/sounds/vm/1234/unavail.gsm ; \
done
:> /var/lib/asterisk/sounds/vm/1234/busy.gsm
for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isonphone; do \
cat /var/lib/asterisk/sounds/$$x.gsm >> /var/lib/asterisk/sounds/vm/1234/busy.gsm ; \
done

239
README
View File

@@ -1,240 +1,81 @@
The Asterisk Open Source PBX
by Mark Spencer <markster@digium.com>
and the Asterisk.org developer community
Copyright (C) 2001-2005 Digium, Inc.
and other copyright holders.
by Mark Spencer <markster@linux-support.net>
Copyright (C) 1999, Mark Spencer
================================================================
* SECURITY
It is imperative that you read and fully understand the contents of
the SECURITY file before you attempt to configure and run an Asterisk
server.
* WHAT IS ASTERISK ?
* WHAT IS ASTERISK
Asterisk is an Open Source PBX and telephony toolkit. It is, in a
sense, middleware between Internet and telephony channels on the bottom,
and Internet and telephony applications at the top. For more information
on the project itself, please visit the Asterisk home page at:
http://www.asterisk.org
http://www.asteriskpbx.com
In addition you'll find lots of information compiled by the Asterisk
community on this Wiki:
http://www.voip-info.org/wiki-Asterisk
There is a book on Asterisk published by O'Reilly under the
Creative Commons License. It is available in book stores as well
as in a downloadable version on the http://www.asteriskdocs.org
web site.
* SUPPORTED OPERATING SYSTEMS
* REQUIRED COMPONENTS
== 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.
Currently, the Asterisk Open Source PBX is only known to run on the
Linux OS, although it may be portable to other UNIX-like operating systems
as well.
== libaudiofile ==
If you want to use format_wav module, then you need a very recent
version of libaudiofile (at least version 0.2.0, or you can apply the
included patch. RPMS for the patched libaudiofile are available at:
ftp://ftp.asteriskpbx.com/pub/asterisk/support
== Others ==
Asterisk has also been 'ported' and reportedly runs properly on other
operating systems as well, including Sun Solaris, Apple's Mac OS X, and
the BSD variants.
* GETTING STARTED
First, be sure you've got supported hardware (but note that you don't need
ANY special hardware, not even a soundcard) to install and run Asterisk.
First, be sure you've installed the required libaudiofile upgrade if
you want to use the non-GSM WAV format. Next, be sure you've got
supported hardware. To use Asterisk right now, you will need one of
the following:
Supported telephony hardware includes:
* Adtran Atlas 800 Plus
* QuickNet Internet PhoneJack
* Full Duplex Sound Card supported by Linux
* 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 or OSS
* VoiceTronix OpenLine products
The are several drivers for ISDN BRI cards available from third party sources.
Check the voip-info.org wiki for more information on chan_capi, chan_misdn and
zaphfc.
* UPGRADING FROM VERSION 1.0
If you are updating from a previous version of Asterisk, make sure you
read the UPGRADE.txt file in the source directory. There are some files
and configuration options that you will have to change, even though we
made every effort possible to maintain backwards compatibility.
In order to discover new features to use, please check the configuration
examples in the /configs directory of the source code distribution.
To discover the major new features of Asterisk 1.2, please visit
http://edvina.net/asterisk1-2/
* NEW INSTALLATIONS
Ensure that your system contains a compatible compiler and development
libraries. Asterisk requires either the GNU Compiler Collection (GCC) version
3.0 or higher, or a compiler that supports the C99 specification and some of
the gcc language extensions. In addition, your system needs to have the C
library headers available, and the headers and libraries for OpenSSL,
ncurses and zlib.
On many distributions, these files are installed by packages with names like
'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
So let's proceed:
Assuming you have one of these (most likely the third) you're ready to
proceed:
1) Run "make"
Assuming the build completes successfully:
2) Run "make install"
Each time you update or checkout from CVS, you are strongly encouraged
to ensure all previous object files are removed to avoid internal
inconsistency in Asterisk. Normally, this is automatically done with
the presence of the file .cleancount, which increments each time a 'make clean'
is required, and the file .lastclean, which contains the last .cleancount used.
If this is your first time working with Asterisk, you may wish to install
If this is your first time working with Asterisk, you may wish to install
the sample PBX, with demonstration extensions, etc. If so, run:
3) "make samples"
"make samples"
Doing so will overwrite any existing config files you have.
Doing so will overwrite any existing config files you have.
Finally, you can launch Asterisk in the foreground mode (not a daemon)
with:
Finally, you can launch Asterisk with:
# asterisk -vvvc
./asterisk -vvvc
You'll see a bunch of verbose messages fly by your screen as Asterisk
If you get an error about unresolved symbols, install the updated
libaudiofile (available at ftp://ftp.asteriskpbx.com/pub/asterisk/support
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:
like this:
*CLI>
You can type "help" at any time to get help with the system. For help
You can type "help" at any time to get help with the system. For help
with a specific command, type "help <command>". To start the PBX using
your sound card, you can type "dial" to dial the PBX. Then you can use
"answer", "hangup", and "dial" to simulate the actions of a telephone.
Remember that if you don't have a full duplex sound card (and Asterisk
will tell you somewhere in its verbose messages if you do/don't) then it
Remember that if you don't have a full duplex sound card (And asterisk
will tell you somewhere in its verbose messages if you do/don't) 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
Feel free to look over the configuration files in /etc/asterisk, where
you'll find a lot of information about what you can do with Asterisk.
* ABOUT CONFIGURATION FILES
All Asterisk configuration files share a common format. Comments are
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,
those of the form 'variable = value', and those of the form 'object =>
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 zapata.conf, one might specify:
switchtype=national
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:
switchtype = national
channel => 1-4
channel => 10-12
switchtype = dms100
channel => 25-47
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
parameters. For example, the line "channel => 25-47" creates objects for
the channels 25 through 47 of the card, obtaining the settings
from the variables specified above.
* SPECIAL NOTE ON TIME
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.
Apparent time changes due to daylight savings time are just that,
apparent. The use of daylight savings time in a Linux system is
purely a user interface issue and does not affect the operation of the
Linux kernel or Asterisk. The system clock on Linux kernels operates
on UTC. UTC does not use daylight savings time.
Also note that this issue is separate from the clocking of TDM
channels, and is known to at least affect SIP registrations.
* FILE DESCRIPTORS
Depending on the size of your system and your configuration,
Asterisk can consume a large number of file descriptors. In UNIX,
file descriptors are used for more than just files on disk. File
descriptors are also used for handling network communication
(e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and
digital trunk hardware). Asterisk accesses many on-disk files for
everything from configuration information to voicemail storage.
Most systems limit the number of file descriptors that Asterisk can
have open at one time. This can limit the number of simultaneous
calls that your system can handle. For example, if the limit is set
at 1024 (a common default value) Asterisk can handle approxiately 150
SIP calls simultaneously. To change the number of file descriptors
follow the instructions for your system below:
== PAM-based Linux System ==
If your system uses PAM (Pluggable Authentication Modules) edit
/etc/security/limits.conf. Add these lines to the bottom of the file:
root soft nofile 4096
root hard nofile 8196
asterisk soft nofile 4096
asterisk hard nofile 8196
(adjust the numbers to taste). You may need to reboot the system for
these changes to take effect.
== Generic UNIX System ==
If there are no instructions specifically adapted to your system
above you can try adding the command "ulimit -n 8192" to the script
that starts Asterisk.
* MORE INFORMATION
See the doc directory for more documentation on various features. Again,
please read all the configuration samples that include documentation on
the configuration options.
Finally, you may wish to visit the web site and join the mailing list if
Finally, you may wish to visit the web site and join the mailing list if
you're interested in getting more information.
http://www.asterisk.org/support
Welcome to the growing worldwide community of Asterisk users!
Mark Spencer
Mark

View File

@@ -1,22 +0,0 @@
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

View File

@@ -1,73 +0,0 @@
==== 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.
Asterisk security involves both network security (encryption, authentication)
as well as dialplan security (authorization - who can access services in
your pbx). If you are setting up Asterisk in production use, please make
sure you understand the issues involved.
* NETWORK SECURITY
If you install Asterisk and use the "make samples" command to install
a demonstration configuration, Asterisk will open a few ports for accepting
VoIP calls. Check the channel configuration files for the ports and IP addresses.
If you enable the manager interface in manager.conf, please make sure that
you access manager in a safe environment or protect it with SSH or other
VPN solutions.
For all TCP/IP connections in Asterisk, you can set ACL lists that
will permit or deny network access to Asterisk services. Please check
the "permit" and "deny" configuration options in manager.conf and
the VoIP channel configurations - i.e. sip.conf and iax.conf.
The IAX2 protocol supports strong RSA key authentication as well as
AES encryption of voice and signalling. The SIP channel does not
support encryption in this version of Asterisk.
* DIALPLAN SECURITY
First and foremost remember this:
USE THE EXTENSION CONTEXTS TO ISOLATE OUTGOING OR TOLL SERVICES FROM ANY
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.
* LOG SECURITY
Please note that the Asterisk log files, as well as information printed to the
Asterisk CLI, may contain sensitive information such as passwords and call
history. Keep this in mind when providing access to these resources.

View File

@@ -1,225 +0,0 @@
Information for Upgrading From Previous Asterisk Releases
=========================================================
IAX2:
* 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.
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
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.

347
acl.c
View File

@@ -1,347 +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 Various sorts of access control
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <sys/time.h>
#include <signal.h>
#include <errno.h>
#include <unistd.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <sys/socket.h>
#include <netdb.h>
#include <net/if.h>
#include <netinet/in.h>
#include <netinet/in_systm.h>
#include <netinet/ip.h>
#include <sys/ioctl.h>
#if defined(__OpenBSD__) || defined(__NetBSD__) || defined(__FreeBSD__)
#include <fcntl.h>
#include <net/route.h>
#endif
#if defined(SOLARIS)
#include <sys/sockio.h>
#endif
/* netinet/ip.h may not define the following (See RFCs 791 and 1349) */
#if !defined(IPTOS_LOWCOST)
#define IPTOS_LOWCOST 0x02
#endif
#if !defined(IPTOS_MINCOST)
#define IPTOS_MINCOST IPTOS_LOWCOST
#endif
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/acl.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/options.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/srv.h"
#include "asterisk/compat.h"
#if defined(__OpenBSD__) || defined(__NetBSD__) || defined(__FreeBSD__)
AST_MUTEX_DEFINE_STATIC(routeseq_lock);
#endif
/* 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 */
void ast_copy_ha(const 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, const char *stuff, struct ast_ha *path)
{
struct ast_ha *ha = malloc(sizeof(struct ast_ha));
char *nm = "255.255.255.255";
char tmp[256];
struct ast_ha *prev = NULL;
struct ast_ha *ret;
int x, z;
unsigned int y;
ret = path;
while (path) {
prev = path;
path = path->next;
}
if (ha) {
ast_copy_string(tmp, stuff, sizeof(tmp));
nm = strchr(tmp, '/');
if (!nm) {
nm = "255.255.255.255";
} else {
*nm = '\0';
nm++;
}
if (!strchr(nm, '.')) {
if ((sscanf(nm, "%30d", &x) == 1) && (x >= 0) && (x <= 32)) {
y = 0;
for (z=0;z<x;z++) {
y >>= 1;
y |= 0x80000000;
}
ha->netmask.s_addr = htonl(y);
}
} else if (!inet_aton(nm, &ha->netmask)) {
ast_log(LOG_WARNING, "%s is not a valid netmask\n", nm);
free(ha);
return ret;
}
if (!inet_aton(tmp, &ha->netaddr)) {
ast_log(LOG_WARNING, "%s is not a valid IP\n", tmp);
free(ha);
return ret;
}
ha->netaddr.s_addr &= ha->netmask.s_addr;
if (!strncasecmp(sense, "p", 1)) {
ha->sense = AST_SENSE_ALLOW;
} else {
ha->sense = AST_SENSE_DENY;
}
ha->next = NULL;
if (prev) {
prev->next = ha;
} else {
ret = ha;
}
}
ast_log(LOG_DEBUG, "%s/%s appended to acl for peer\n", stuff, nm);
return ret;
}
int ast_apply_ha(struct ast_ha *ha, struct sockaddr_in *sin)
{
/* Start optimistic */
int res = AST_SENSE_ALLOW;
while (ha) {
char iabuf[INET_ADDRSTRLEN];
char iabuf2[INET_ADDRSTRLEN];
/* DEBUG */
ast_log(LOG_DEBUG,
"##### Testing %s with %s\n",
ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr),
ast_inet_ntoa(iabuf2, sizeof(iabuf2), ha->netaddr));
/* For each rule, if this address and the netmask = the net address
apply the current rule */
if ((sin->sin_addr.s_addr & ha->netmask.s_addr) == ha->netaddr.s_addr)
res = ha->sense;
ha = ha->next;
}
return res;
}
int ast_get_ip_or_srv(struct sockaddr_in *sin, const char *value, const char *service)
{
struct hostent *hp;
struct ast_hostent ahp;
char srv[256];
char host[256];
int tportno = ntohs(sin->sin_port);
if (inet_aton(value, &sin->sin_addr))
return 0;
if (service) {
snprintf(srv, sizeof(srv), "%s.%s", service, value);
if (ast_get_srv(NULL, host, sizeof(host), &tportno, srv) > 0) {
sin->sin_port = htons(tportno);
value = host;
}
}
hp = ast_gethostbyname(value, &ahp);
if (hp) {
memcpy(&sin->sin_addr, hp->h_addr, sizeof(sin->sin_addr));
} else {
ast_log(LOG_WARNING, "Unable to lookup '%s'\n", value);
return -1;
}
return 0;
}
int ast_str2tos(const char *value, int *tos)
{
int fval;
if (sscanf(value, "%30i", &fval) == 1)
*tos = fval & 0xff;
else if (!strcasecmp(value, "lowdelay"))
*tos = IPTOS_LOWDELAY;
else if (!strcasecmp(value, "throughput"))
*tos = IPTOS_THROUGHPUT;
else if (!strcasecmp(value, "reliability"))
*tos = IPTOS_RELIABILITY;
else if (!strcasecmp(value, "mincost"))
*tos = IPTOS_MINCOST;
else if (!strcasecmp(value, "none"))
*tos = 0;
else
return -1;
return 0;
}
int ast_get_ip(struct sockaddr_in *sin, const char *value)
{
return ast_get_ip_or_srv(sin, value, NULL);
}
/* iface is the interface (e.g. eth0); address is the return value */
int ast_lookup_iface(char *iface, struct in_addr *address)
{
int mysock, res = 0;
struct my_ifreq ifreq;
memset(&ifreq, 0, sizeof(ifreq));
ast_copy_string(ifreq.ifrn_name, iface, sizeof(ifreq.ifrn_name));
mysock = socket(PF_INET, SOCK_DGRAM, IPPROTO_IP);
res = ioctl(mysock, SIOCGIFADDR, &ifreq);
close(mysock);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to get IP of %s: %s\n", iface, strerror(errno));
memcpy((char *)address, (char *)&__ourip, sizeof(__ourip));
return -1;
} else {
memcpy((char *)address, (char *)&ifreq.ifru_addr.sin_addr, sizeof(ifreq.ifru_addr.sin_addr));
return 0;
}
}
int ast_ouraddrfor(struct in_addr *them, struct in_addr *us)
{
int s;
struct sockaddr_in sin;
socklen_t slen;
s = socket(PF_INET, SOCK_DGRAM, 0);
if (s < 0) {
ast_log(LOG_WARNING, "Cannot create socket\n");
return -1;
}
sin.sin_family = AF_INET;
sin.sin_port = 5060;
sin.sin_addr = *them;
if (connect(s, (struct sockaddr *)&sin, sizeof(sin))) {
ast_log(LOG_WARNING, "Cannot connect\n");
close(s);
return -1;
}
slen = sizeof(sin);
if (getsockname(s, (struct sockaddr *)&sin, &slen)) {
ast_log(LOG_WARNING, "Cannot get socket name\n");
close(s);
return -1;
}
close(s);
*us = sin.sin_addr;
return 0;
}
int ast_find_ourip(struct in_addr *ourip, struct sockaddr_in bindaddr)
{
char ourhost[MAXHOSTNAMELEN] = "";
struct ast_hostent ahp;
struct hostent *hp;
struct in_addr saddr;
/* just use the bind address if it is nonzero */
if (ntohl(bindaddr.sin_addr.s_addr)) {
memcpy(ourip, &bindaddr.sin_addr, sizeof(*ourip));
return 0;
}
/* try to use our hostname */
if (gethostname(ourhost, sizeof(ourhost) - 1)) {
ast_log(LOG_WARNING, "Unable to get hostname\n");
} else {
hp = ast_gethostbyname(ourhost, &ahp);
if (hp) {
memcpy(ourip, hp->h_addr, sizeof(*ourip));
return 0;
}
}
/* A.ROOT-SERVERS.NET. */
if (inet_aton("198.41.0.4", &saddr) && !ast_ouraddrfor(&saddr, ourip))
return 0;
return -1;
}

View File

@@ -1,313 +0,0 @@
/*
---------------------------------------------------------------------------
Copyright (c) 2003, Dr Brian Gladman <brg@gladman.me.uk>, Worcester, UK.
All rights reserved.
LICENSE TERMS
The free distribution and use of this software in both source and binary
form is allowed (with or without changes) provided that:
1. distributions of this source code include the above copyright
notice, this list of conditions and the following disclaimer;
2. distributions in binary form include the above copyright
notice, this list of conditions and the following disclaimer
in the documentation and/or other associated materials;
3. the copyright holder's name is not used to endorse products
built using this software without specific written permission.
ALTERNATIVELY, provided that this notice is retained in full, this product
may be distributed under the terms of the GNU General Public License (GPL),
in which case the provisions of the GPL apply INSTEAD OF those given above.
DISCLAIMER
This software is provided 'as is' with no explicit or implied warranties
in respect of its properties, including, but not limited to, correctness
and/or fitness for purpose.
---------------------------------------------------------------------------
Issue Date: 26/08/2003
*/
/*! \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
*/
#include "aesopt.h"
#if defined(__cplusplus)
extern "C"
{
#endif
#define si(y,x,k,c) (s(y,c) = word_in(x, c) ^ (k)[c])
#define so(y,x,c) word_out(y, c, s(x,c))
#if defined(ARRAYS)
#define locals(y,x) x[4],y[4]
#else
#define locals(y,x) x##0,x##1,x##2,x##3,y##0,y##1,y##2,y##3
#endif
#define l_copy(y, x) s(y,0) = s(x,0); s(y,1) = s(x,1); \
s(y,2) = s(x,2); s(y,3) = s(x,3);
#define state_in(y,x,k) si(y,x,k,0); si(y,x,k,1); si(y,x,k,2); si(y,x,k,3)
#define state_out(y,x) so(y,x,0); so(y,x,1); so(y,x,2); so(y,x,3)
#define round(rm,y,x,k) rm(y,x,k,0); rm(y,x,k,1); rm(y,x,k,2); rm(y,x,k,3)
#if defined(ENCRYPTION) && !defined(AES_ASM)
/* Visual C++ .Net v7.1 provides the fastest encryption code when using
Pentium optimiation with small code but this is poor for decryption
so we need to control this with the following VC++ pragmas
*/
#if defined(_MSC_VER)
#pragma optimize( "s", on )
#endif
/* Given the column (c) of the output state variable, the following
macros give the input state variables which are needed in its
computation for each row (r) of the state. All the alternative
macros give the same end values but expand into different ways
of calculating these values. In particular the complex macro
used for dynamically variable block sizes is designed to expand
to a compile time constant whenever possible but will expand to
conditional clauses on some branches (I am grateful to Frank
Yellin for this construction)
*/
#define fwd_var(x,r,c)\
( r == 0 ? ( c == 0 ? s(x,0) : c == 1 ? s(x,1) : c == 2 ? s(x,2) : s(x,3))\
: r == 1 ? ( c == 0 ? s(x,1) : c == 1 ? s(x,2) : c == 2 ? s(x,3) : s(x,0))\
: r == 2 ? ( c == 0 ? s(x,2) : c == 1 ? s(x,3) : c == 2 ? s(x,0) : s(x,1))\
: ( c == 0 ? s(x,3) : c == 1 ? s(x,0) : c == 2 ? s(x,1) : s(x,2)))
#if defined(FT4_SET)
#undef dec_fmvars
#define fwd_rnd(y,x,k,c) (s(y,c) = (k)[c] ^ four_tables(x,t_use(f,n),fwd_var,rf1,c))
#elif defined(FT1_SET)
#undef dec_fmvars
#define fwd_rnd(y,x,k,c) (s(y,c) = (k)[c] ^ one_table(x,upr,t_use(f,n),fwd_var,rf1,c))
#else
#define fwd_rnd(y,x,k,c) (s(y,c) = (k)[c] ^ fwd_mcol(no_table(x,t_use(s,box),fwd_var,rf1,c)))
#endif
#if defined(FL4_SET)
#define fwd_lrnd(y,x,k,c) (s(y,c) = (k)[c] ^ four_tables(x,t_use(f,l),fwd_var,rf1,c))
#elif defined(FL1_SET)
#define fwd_lrnd(y,x,k,c) (s(y,c) = (k)[c] ^ one_table(x,ups,t_use(f,l),fwd_var,rf1,c))
#else
#define fwd_lrnd(y,x,k,c) (s(y,c) = (k)[c] ^ no_table(x,t_use(s,box),fwd_var,rf1,c))
#endif
aes_rval aes_encrypt(const void *in_blk, void *out_blk, const aes_encrypt_ctx cx[1])
{ aes_32t locals(b0, b1);
const aes_32t *kp = cx->ks;
#ifdef dec_fmvars
dec_fmvars; /* declare variables for fwd_mcol() if needed */
#endif
aes_32t nr = (kp[45] ^ kp[52] ^ kp[53] ? kp[52] : 14);
#ifdef AES_ERR_CHK
if( (nr != 10 || !(kp[0] | kp[3] | kp[4]))
&& (nr != 12 || !(kp[0] | kp[5] | kp[6]))
&& (nr != 14 || !(kp[0] | kp[7] | kp[8])) )
return aes_error;
#endif
state_in(b0, in_blk, kp);
#if (ENC_UNROLL == FULL)
switch(nr)
{
case 14:
round(fwd_rnd, b1, b0, kp + 1 * N_COLS);
round(fwd_rnd, b0, b1, kp + 2 * N_COLS);
kp += 2 * N_COLS;
case 12:
round(fwd_rnd, b1, b0, kp + 1 * N_COLS);
round(fwd_rnd, b0, b1, kp + 2 * N_COLS);
kp += 2 * N_COLS;
case 10:
round(fwd_rnd, b1, b0, kp + 1 * N_COLS);
round(fwd_rnd, b0, b1, kp + 2 * N_COLS);
round(fwd_rnd, b1, b0, kp + 3 * N_COLS);
round(fwd_rnd, b0, b1, kp + 4 * N_COLS);
round(fwd_rnd, b1, b0, kp + 5 * N_COLS);
round(fwd_rnd, b0, b1, kp + 6 * N_COLS);
round(fwd_rnd, b1, b0, kp + 7 * N_COLS);
round(fwd_rnd, b0, b1, kp + 8 * N_COLS);
round(fwd_rnd, b1, b0, kp + 9 * N_COLS);
round(fwd_lrnd, b0, b1, kp +10 * N_COLS);
}
#else
#if (ENC_UNROLL == PARTIAL)
{ aes_32t rnd;
for(rnd = 0; rnd < (nr >> 1) - 1; ++rnd)
{
kp += N_COLS;
round(fwd_rnd, b1, b0, kp);
kp += N_COLS;
round(fwd_rnd, b0, b1, kp);
}
kp += N_COLS;
round(fwd_rnd, b1, b0, kp);
#else
{ aes_32t rnd;
for(rnd = 0; rnd < nr - 1; ++rnd)
{
kp += N_COLS;
round(fwd_rnd, b1, b0, kp);
l_copy(b0, b1);
}
#endif
kp += N_COLS;
round(fwd_lrnd, b0, b1, kp);
}
#endif
state_out(out_blk, b0);
#ifdef AES_ERR_CHK
return aes_good;
#endif
}
#endif
#if defined(DECRYPTION) && !defined(AES_ASM)
/* Visual C++ .Net v7.1 provides the fastest encryption code when using
Pentium optimiation with small code but this is poor for decryption
so we need to control this with the following VC++ pragmas
*/
#if defined(_MSC_VER)
#pragma optimize( "t", on )
#endif
/* Given the column (c) of the output state variable, the following
macros give the input state variables which are needed in its
computation for each row (r) of the state. All the alternative
macros give the same end values but expand into different ways
of calculating these values. In particular the complex macro
used for dynamically variable block sizes is designed to expand
to a compile time constant whenever possible but will expand to
conditional clauses on some branches (I am grateful to Frank
Yellin for this construction)
*/
#define inv_var(x,r,c)\
( r == 0 ? ( c == 0 ? s(x,0) : c == 1 ? s(x,1) : c == 2 ? s(x,2) : s(x,3))\
: r == 1 ? ( c == 0 ? s(x,3) : c == 1 ? s(x,0) : c == 2 ? s(x,1) : s(x,2))\
: r == 2 ? ( c == 0 ? s(x,2) : c == 1 ? s(x,3) : c == 2 ? s(x,0) : s(x,1))\
: ( c == 0 ? s(x,1) : c == 1 ? s(x,2) : c == 2 ? s(x,3) : s(x,0)))
#if defined(IT4_SET)
#undef dec_imvars
#define inv_rnd(y,x,k,c) (s(y,c) = (k)[c] ^ four_tables(x,t_use(i,n),inv_var,rf1,c))
#elif defined(IT1_SET)
#undef dec_imvars
#define inv_rnd(y,x,k,c) (s(y,c) = (k)[c] ^ one_table(x,upr,t_use(i,n),inv_var,rf1,c))
#else
#define inv_rnd(y,x,k,c) (s(y,c) = inv_mcol((k)[c] ^ no_table(x,t_use(i,box),inv_var,rf1,c)))
#endif
#if defined(IL4_SET)
#define inv_lrnd(y,x,k,c) (s(y,c) = (k)[c] ^ four_tables(x,t_use(i,l),inv_var,rf1,c))
#elif defined(IL1_SET)
#define inv_lrnd(y,x,k,c) (s(y,c) = (k)[c] ^ one_table(x,ups,t_use(i,l),inv_var,rf1,c))
#else
#define inv_lrnd(y,x,k,c) (s(y,c) = (k)[c] ^ no_table(x,t_use(i,box),inv_var,rf1,c))
#endif
aes_rval aes_decrypt(const void *in_blk, void *out_blk, const aes_decrypt_ctx cx[1])
{ aes_32t locals(b0, b1);
#ifdef dec_imvars
dec_imvars; /* declare variables for inv_mcol() if needed */
#endif
aes_32t nr = (cx->ks[45] ^ cx->ks[52] ^ cx->ks[53] ? cx->ks[52] : 14);
const aes_32t *kp = cx->ks + nr * N_COLS;
#ifdef AES_ERR_CHK
if( (nr != 10 || !(cx->ks[0] | cx->ks[3] | cx->ks[4]))
&& (nr != 12 || !(cx->ks[0] | cx->ks[5] | cx->ks[6]))
&& (nr != 14 || !(cx->ks[0] | cx->ks[7] | cx->ks[8])) )
return aes_error;
#endif
state_in(b0, in_blk, kp);
#if (DEC_UNROLL == FULL)
switch(nr)
{
case 14:
round(inv_rnd, b1, b0, kp - 1 * N_COLS);
round(inv_rnd, b0, b1, kp - 2 * N_COLS);
kp -= 2 * N_COLS;
case 12:
round(inv_rnd, b1, b0, kp - 1 * N_COLS);
round(inv_rnd, b0, b1, kp - 2 * N_COLS);
kp -= 2 * N_COLS;
case 10:
round(inv_rnd, b1, b0, kp - 1 * N_COLS);
round(inv_rnd, b0, b1, kp - 2 * N_COLS);
round(inv_rnd, b1, b0, kp - 3 * N_COLS);
round(inv_rnd, b0, b1, kp - 4 * N_COLS);
round(inv_rnd, b1, b0, kp - 5 * N_COLS);
round(inv_rnd, b0, b1, kp - 6 * N_COLS);
round(inv_rnd, b1, b0, kp - 7 * N_COLS);
round(inv_rnd, b0, b1, kp - 8 * N_COLS);
round(inv_rnd, b1, b0, kp - 9 * N_COLS);
round(inv_lrnd, b0, b1, kp - 10 * N_COLS);
}
#else
#if (DEC_UNROLL == PARTIAL)
{ aes_32t rnd;
for(rnd = 0; rnd < (nr >> 1) - 1; ++rnd)
{
kp -= N_COLS;
round(inv_rnd, b1, b0, kp);
kp -= N_COLS;
round(inv_rnd, b0, b1, kp);
}
kp -= N_COLS;
round(inv_rnd, b1, b0, kp);
#else
{ aes_32t rnd;
for(rnd = 0; rnd < nr - 1; ++rnd)
{
kp -= N_COLS;
round(inv_rnd, b1, b0, kp);
l_copy(b0, b1);
}
#endif
kp -= N_COLS;
round(inv_lrnd, b0, b1, kp);
}
#endif
state_out(out_blk, b0);
#ifdef AES_ERR_CHK
return aes_good;
#endif
}
#endif
#if defined(__cplusplus)
}
#endif

465
aeskey.c
View File

@@ -1,465 +0,0 @@
/*
---------------------------------------------------------------------------
Copyright (c) 2003, Dr Brian Gladman <brg@gladman.me.uk>, Worcester, UK.
All rights reserved.
LICENSE TERMS
The free distribution and use of this software in both source and binary
form is allowed (with or without changes) provided that:
1. distributions of this source code include the above copyright
notice, this list of conditions and the following disclaimer;
2. distributions in binary form include the above copyright
notice, this list of conditions and the following disclaimer
in the documentation and/or other associated materials;
3. the copyright holder's name is not used to endorse products
built using this software without specific written permission.
ALTERNATIVELY, provided that this notice is retained in full, this product
may be distributed under the terms of the GNU General Public License (GPL),
in which case the provisions of the GPL apply INSTEAD OF those given above.
DISCLAIMER
This software is provided 'as is' with no explicit or implied warranties
in respect of its properties, including, but not limited to, correctness
and/or fitness for purpose.
---------------------------------------------------------------------------
Issue Date: 26/08/2003
*/
/*! \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.
*/
#include "aesopt.h"
#if defined(__cplusplus)
extern "C"
{
#endif
/* Initialise the key schedule from the user supplied key. The key
length can be specified in bytes, with legal values of 16, 24
and 32, or in bits, with legal values of 128, 192 and 256. These
values correspond with Nk values of 4, 6 and 8 respectively.
The following macros implement a single cycle in the key
schedule generation process. The number of cycles needed
for each cx->n_col and nk value is:
nk = 4 5 6 7 8
------------------------------
cx->n_col = 4 10 9 8 7 7
cx->n_col = 5 14 11 10 9 9
cx->n_col = 6 19 15 12 11 11
cx->n_col = 7 21 19 16 13 14
cx->n_col = 8 29 23 19 17 14
*/
#define ke4(k,i) \
{ k[4*(i)+4] = ss[0] ^= ls_box(ss[3],3) ^ t_use(r,c)[i]; k[4*(i)+5] = ss[1] ^= ss[0]; \
k[4*(i)+6] = ss[2] ^= ss[1]; k[4*(i)+7] = ss[3] ^= ss[2]; \
}
#define kel4(k,i) \
{ k[4*(i)+4] = ss[0] ^= ls_box(ss[3],3) ^ t_use(r,c)[i]; k[4*(i)+5] = ss[1] ^= ss[0]; \
k[4*(i)+6] = ss[2] ^= ss[1]; k[4*(i)+7] = ss[3] ^= ss[2]; \
}
#define ke6(k,i) \
{ k[6*(i)+ 6] = ss[0] ^= ls_box(ss[5],3) ^ t_use(r,c)[i]; k[6*(i)+ 7] = ss[1] ^= ss[0]; \
k[6*(i)+ 8] = ss[2] ^= ss[1]; k[6*(i)+ 9] = ss[3] ^= ss[2]; \
k[6*(i)+10] = ss[4] ^= ss[3]; k[6*(i)+11] = ss[5] ^= ss[4]; \
}
#define kel6(k,i) \
{ k[6*(i)+ 6] = ss[0] ^= ls_box(ss[5],3) ^ t_use(r,c)[i]; k[6*(i)+ 7] = ss[1] ^= ss[0]; \
k[6*(i)+ 8] = ss[2] ^= ss[1]; k[6*(i)+ 9] = ss[3] ^= ss[2]; \
}
#define ke8(k,i) \
{ k[8*(i)+ 8] = ss[0] ^= ls_box(ss[7],3) ^ t_use(r,c)[i]; k[8*(i)+ 9] = ss[1] ^= ss[0]; \
k[8*(i)+10] = ss[2] ^= ss[1]; k[8*(i)+11] = ss[3] ^= ss[2]; \
k[8*(i)+12] = ss[4] ^= ls_box(ss[3],0); k[8*(i)+13] = ss[5] ^= ss[4]; \
k[8*(i)+14] = ss[6] ^= ss[5]; k[8*(i)+15] = ss[7] ^= ss[6]; \
}
#define kel8(k,i) \
{ k[8*(i)+ 8] = ss[0] ^= ls_box(ss[7],3) ^ t_use(r,c)[i]; k[8*(i)+ 9] = ss[1] ^= ss[0]; \
k[8*(i)+10] = ss[2] ^= ss[1]; k[8*(i)+11] = ss[3] ^= ss[2]; \
}
#if defined(ENCRYPTION_KEY_SCHEDULE)
#if defined(AES_128) || defined(AES_VAR)
aes_rval aes_encrypt_key128(const void *in_key, aes_encrypt_ctx cx[1])
{ aes_32t ss[4];
cx->ks[0] = ss[0] = word_in(in_key, 0);
cx->ks[1] = ss[1] = word_in(in_key, 1);
cx->ks[2] = ss[2] = word_in(in_key, 2);
cx->ks[3] = ss[3] = word_in(in_key, 3);
#if ENC_UNROLL == NONE
{ aes_32t i;
for(i = 0; i < ((11 * N_COLS - 1) / 4); ++i)
ke4(cx->ks, i);
}
#else
ke4(cx->ks, 0); ke4(cx->ks, 1);
ke4(cx->ks, 2); ke4(cx->ks, 3);
ke4(cx->ks, 4); ke4(cx->ks, 5);
ke4(cx->ks, 6); ke4(cx->ks, 7);
ke4(cx->ks, 8); kel4(cx->ks, 9);
#endif
/* cx->ks[45] ^ cx->ks[52] ^ cx->ks[53] is zero for a 256 bit */
/* key and must be non-zero for 128 and 192 bits keys */
cx->ks[53] = cx->ks[45] = 0;
cx->ks[52] = 10;
#ifdef AES_ERR_CHK
return aes_good;
#endif
}
#endif
#if defined(AES_192) || defined(AES_VAR)
aes_rval aes_encrypt_key192(const void *in_key, aes_encrypt_ctx cx[1])
{ aes_32t ss[6];
cx->ks[0] = ss[0] = word_in(in_key, 0);
cx->ks[1] = ss[1] = word_in(in_key, 1);
cx->ks[2] = ss[2] = word_in(in_key, 2);
cx->ks[3] = ss[3] = word_in(in_key, 3);
cx->ks[4] = ss[4] = word_in(in_key, 4);
cx->ks[5] = ss[5] = word_in(in_key, 5);
#if ENC_UNROLL == NONE
{ aes_32t i;
for(i = 0; i < (13 * N_COLS - 1) / 6; ++i)
ke6(cx->ks, i);
}
#else
ke6(cx->ks, 0); ke6(cx->ks, 1);
ke6(cx->ks, 2); ke6(cx->ks, 3);
ke6(cx->ks, 4); ke6(cx->ks, 5);
ke6(cx->ks, 6); kel6(cx->ks, 7);
#endif
/* cx->ks[45] ^ cx->ks[52] ^ cx->ks[53] is zero for a 256 bit */
/* key and must be non-zero for 128 and 192 bits keys */
cx->ks[53] = cx->ks[45];
cx->ks[52] = 12;
#ifdef AES_ERR_CHK
return aes_good;
#endif
}
#endif
#if defined(AES_256) || defined(AES_VAR)
aes_rval aes_encrypt_key256(const void *in_key, aes_encrypt_ctx cx[1])
{ aes_32t ss[8];
cx->ks[0] = ss[0] = word_in(in_key, 0);
cx->ks[1] = ss[1] = word_in(in_key, 1);
cx->ks[2] = ss[2] = word_in(in_key, 2);
cx->ks[3] = ss[3] = word_in(in_key, 3);
cx->ks[4] = ss[4] = word_in(in_key, 4);
cx->ks[5] = ss[5] = word_in(in_key, 5);
cx->ks[6] = ss[6] = word_in(in_key, 6);
cx->ks[7] = ss[7] = word_in(in_key, 7);
#if ENC_UNROLL == NONE
{ aes_32t i;
for(i = 0; i < (15 * N_COLS - 1) / 8; ++i)
ke8(cx->ks, i);
}
#else
ke8(cx->ks, 0); ke8(cx->ks, 1);
ke8(cx->ks, 2); ke8(cx->ks, 3);
ke8(cx->ks, 4); ke8(cx->ks, 5);
kel8(cx->ks, 6);
#endif
#ifdef AES_ERR_CHK
return aes_good;
#endif
}
#endif
#if defined(AES_VAR)
aes_rval aes_encrypt_key(const void *in_key, int key_len, aes_encrypt_ctx cx[1])
{
switch(key_len)
{
#ifdef AES_ERR_CHK
case 16: case 128: return aes_encrypt_key128(in_key, cx);
case 24: case 192: return aes_encrypt_key192(in_key, cx);
case 32: case 256: return aes_encrypt_key256(in_key, cx);
default: return aes_error;
#else
case 16: case 128: aes_encrypt_key128(in_key, cx); return;
case 24: case 192: aes_encrypt_key192(in_key, cx); return;
case 32: case 256: aes_encrypt_key256(in_key, cx); return;
#endif
}
}
#endif
#endif
#if defined(DECRYPTION_KEY_SCHEDULE)
#if DEC_ROUND == NO_TABLES
#define ff(x) (x)
#else
#define ff(x) inv_mcol(x)
#ifdef dec_imvars
#define d_vars dec_imvars
#endif
#endif
#if 1
#define kdf4(k,i) \
{ ss[0] = ss[0] ^ ss[2] ^ ss[1] ^ ss[3]; ss[1] = ss[1] ^ ss[3]; ss[2] = ss[2] ^ ss[3]; ss[3] = ss[3]; \
ss[4] = ls_box(ss[(i+3) % 4], 3) ^ t_use(r,c)[i]; ss[i % 4] ^= ss[4]; \
ss[4] ^= k[4*(i)]; k[4*(i)+4] = ff(ss[4]); ss[4] ^= k[4*(i)+1]; k[4*(i)+5] = ff(ss[4]); \
ss[4] ^= k[4*(i)+2]; k[4*(i)+6] = ff(ss[4]); ss[4] ^= k[4*(i)+3]; k[4*(i)+7] = ff(ss[4]); \
}
#define kd4(k,i) \
{ ss[4] = ls_box(ss[(i+3) % 4], 3) ^ t_use(r,c)[i]; ss[i % 4] ^= ss[4]; ss[4] = ff(ss[4]); \
k[4*(i)+4] = ss[4] ^= k[4*(i)]; k[4*(i)+5] = ss[4] ^= k[4*(i)+1]; \
k[4*(i)+6] = ss[4] ^= k[4*(i)+2]; k[4*(i)+7] = ss[4] ^= k[4*(i)+3]; \
}
#define kdl4(k,i) \
{ ss[4] = ls_box(ss[(i+3) % 4], 3) ^ t_use(r,c)[i]; ss[i % 4] ^= ss[4]; \
k[4*(i)+4] = (ss[0] ^= ss[1]) ^ ss[2] ^ ss[3]; k[4*(i)+5] = ss[1] ^ ss[3]; \
k[4*(i)+6] = ss[0]; k[4*(i)+7] = ss[1]; \
}
#else
#define kdf4(k,i) \
{ ss[0] ^= ls_box(ss[3],3) ^ t_use(r,c)[i]; k[4*(i)+ 4] = ff(ss[0]); ss[1] ^= ss[0]; k[4*(i)+ 5] = ff(ss[1]); \
ss[2] ^= ss[1]; k[4*(i)+ 6] = ff(ss[2]); ss[3] ^= ss[2]; k[4*(i)+ 7] = ff(ss[3]); \
}
#define kd4(k,i) \
{ ss[4] = ls_box(ss[3],3) ^ t_use(r,c)[i]; \
ss[0] ^= ss[4]; ss[4] = ff(ss[4]); k[4*(i)+ 4] = ss[4] ^= k[4*(i)]; \
ss[1] ^= ss[0]; k[4*(i)+ 5] = ss[4] ^= k[4*(i)+ 1]; \
ss[2] ^= ss[1]; k[4*(i)+ 6] = ss[4] ^= k[4*(i)+ 2]; \
ss[3] ^= ss[2]; k[4*(i)+ 7] = ss[4] ^= k[4*(i)+ 3]; \
}
#define kdl4(k,i) \
{ ss[0] ^= ls_box(ss[3],3) ^ t_use(r,c)[i]; k[4*(i)+ 4] = ss[0]; ss[1] ^= ss[0]; k[4*(i)+ 5] = ss[1]; \
ss[2] ^= ss[1]; k[4*(i)+ 6] = ss[2]; ss[3] ^= ss[2]; k[4*(i)+ 7] = ss[3]; \
}
#endif
#define kdf6(k,i) \
{ ss[0] ^= ls_box(ss[5],3) ^ t_use(r,c)[i]; k[6*(i)+ 6] = ff(ss[0]); ss[1] ^= ss[0]; k[6*(i)+ 7] = ff(ss[1]); \
ss[2] ^= ss[1]; k[6*(i)+ 8] = ff(ss[2]); ss[3] ^= ss[2]; k[6*(i)+ 9] = ff(ss[3]); \
ss[4] ^= ss[3]; k[6*(i)+10] = ff(ss[4]); ss[5] ^= ss[4]; k[6*(i)+11] = ff(ss[5]); \
}
#define kd6(k,i) \
{ ss[6] = ls_box(ss[5],3) ^ t_use(r,c)[i]; \
ss[0] ^= ss[6]; ss[6] = ff(ss[6]); k[6*(i)+ 6] = ss[6] ^= k[6*(i)]; \
ss[1] ^= ss[0]; k[6*(i)+ 7] = ss[6] ^= k[6*(i)+ 1]; \
ss[2] ^= ss[1]; k[6*(i)+ 8] = ss[6] ^= k[6*(i)+ 2]; \
ss[3] ^= ss[2]; k[6*(i)+ 9] = ss[6] ^= k[6*(i)+ 3]; \
ss[4] ^= ss[3]; k[6*(i)+10] = ss[6] ^= k[6*(i)+ 4]; \
ss[5] ^= ss[4]; k[6*(i)+11] = ss[6] ^= k[6*(i)+ 5]; \
}
#define kdl6(k,i) \
{ ss[0] ^= ls_box(ss[5],3) ^ t_use(r,c)[i]; k[6*(i)+ 6] = ss[0]; ss[1] ^= ss[0]; k[6*(i)+ 7] = ss[1]; \
ss[2] ^= ss[1]; k[6*(i)+ 8] = ss[2]; ss[3] ^= ss[2]; k[6*(i)+ 9] = ss[3]; \
}
#define kdf8(k,i) \
{ ss[0] ^= ls_box(ss[7],3) ^ t_use(r,c)[i]; k[8*(i)+ 8] = ff(ss[0]); ss[1] ^= ss[0]; k[8*(i)+ 9] = ff(ss[1]); \
ss[2] ^= ss[1]; k[8*(i)+10] = ff(ss[2]); ss[3] ^= ss[2]; k[8*(i)+11] = ff(ss[3]); \
ss[4] ^= ls_box(ss[3],0); k[8*(i)+12] = ff(ss[4]); ss[5] ^= ss[4]; k[8*(i)+13] = ff(ss[5]); \
ss[6] ^= ss[5]; k[8*(i)+14] = ff(ss[6]); ss[7] ^= ss[6]; k[8*(i)+15] = ff(ss[7]); \
}
#define kd8(k,i) \
{ aes_32t g = ls_box(ss[7],3) ^ t_use(r,c)[i]; \
ss[0] ^= g; g = ff(g); k[8*(i)+ 8] = g ^= k[8*(i)]; \
ss[1] ^= ss[0]; k[8*(i)+ 9] = g ^= k[8*(i)+ 1]; \
ss[2] ^= ss[1]; k[8*(i)+10] = g ^= k[8*(i)+ 2]; \
ss[3] ^= ss[2]; k[8*(i)+11] = g ^= k[8*(i)+ 3]; \
g = ls_box(ss[3],0); \
ss[4] ^= g; g = ff(g); k[8*(i)+12] = g ^= k[8*(i)+ 4]; \
ss[5] ^= ss[4]; k[8*(i)+13] = g ^= k[8*(i)+ 5]; \
ss[6] ^= ss[5]; k[8*(i)+14] = g ^= k[8*(i)+ 6]; \
ss[7] ^= ss[6]; k[8*(i)+15] = g ^= k[8*(i)+ 7]; \
}
#define kdl8(k,i) \
{ ss[0] ^= ls_box(ss[7],3) ^ t_use(r,c)[i]; k[8*(i)+ 8] = ss[0]; ss[1] ^= ss[0]; k[8*(i)+ 9] = ss[1]; \
ss[2] ^= ss[1]; k[8*(i)+10] = ss[2]; ss[3] ^= ss[2]; k[8*(i)+11] = ss[3]; \
}
#if defined(AES_128) || defined(AES_VAR)
aes_rval aes_decrypt_key128(const void *in_key, aes_decrypt_ctx cx[1])
{ aes_32t ss[5];
#ifdef d_vars
d_vars;
#endif
cx->ks[0] = ss[0] = word_in(in_key, 0);
cx->ks[1] = ss[1] = word_in(in_key, 1);
cx->ks[2] = ss[2] = word_in(in_key, 2);
cx->ks[3] = ss[3] = word_in(in_key, 3);
#if DEC_UNROLL == NONE
{ aes_32t i;
for(i = 0; i < (11 * N_COLS - 1) / 4; ++i)
ke4(cx->ks, i);
#if !(DEC_ROUND == NO_TABLES)
for(i = N_COLS; i < 10 * N_COLS; ++i)
cx->ks[i] = inv_mcol(cx->ks[i]);
#endif
}
#else
kdf4(cx->ks, 0); kd4(cx->ks, 1);
kd4(cx->ks, 2); kd4(cx->ks, 3);
kd4(cx->ks, 4); kd4(cx->ks, 5);
kd4(cx->ks, 6); kd4(cx->ks, 7);
kd4(cx->ks, 8); kdl4(cx->ks, 9);
#endif
/* cx->ks[45] ^ cx->ks[52] ^ cx->ks[53] is zero for a 256 bit */
/* key and must be non-zero for 128 and 192 bits keys */
cx->ks[53] = cx->ks[45] = 0;
cx->ks[52] = 10;
#ifdef AES_ERR_CHK
return aes_good;
#endif
}
#endif
#if defined(AES_192) || defined(AES_VAR)
aes_rval aes_decrypt_key192(const void *in_key, aes_decrypt_ctx cx[1])
{ aes_32t ss[7];
#ifdef d_vars
d_vars;
#endif
cx->ks[0] = ss[0] = word_in(in_key, 0);
cx->ks[1] = ss[1] = word_in(in_key, 1);
cx->ks[2] = ss[2] = word_in(in_key, 2);
cx->ks[3] = ss[3] = word_in(in_key, 3);
#if DEC_UNROLL == NONE
cx->ks[4] = ss[4] = word_in(in_key, 4);
cx->ks[5] = ss[5] = word_in(in_key, 5);
{ aes_32t i;
for(i = 0; i < (13 * N_COLS - 1) / 6; ++i)
ke6(cx->ks, i);
#if !(DEC_ROUND == NO_TABLES)
for(i = N_COLS; i < 12 * N_COLS; ++i)
cx->ks[i] = inv_mcol(cx->ks[i]);
#endif
}
#else
cx->ks[4] = ff(ss[4] = word_in(in_key, 4));
cx->ks[5] = ff(ss[5] = word_in(in_key, 5));
kdf6(cx->ks, 0); kd6(cx->ks, 1);
kd6(cx->ks, 2); kd6(cx->ks, 3);
kd6(cx->ks, 4); kd6(cx->ks, 5);
kd6(cx->ks, 6); kdl6(cx->ks, 7);
#endif
/* cx->ks[45] ^ cx->ks[52] ^ cx->ks[53] is zero for a 256 bit */
/* key and must be non-zero for 128 and 192 bits keys */
cx->ks[53] = cx->ks[45];
cx->ks[52] = 12;
#ifdef AES_ERR_CHK
return aes_good;
#endif
}
#endif
#if defined(AES_256) || defined(AES_VAR)
aes_rval aes_decrypt_key256(const void *in_key, aes_decrypt_ctx cx[1])
{ aes_32t ss[8];
#ifdef d_vars
d_vars;
#endif
cx->ks[0] = ss[0] = word_in(in_key, 0);
cx->ks[1] = ss[1] = word_in(in_key, 1);
cx->ks[2] = ss[2] = word_in(in_key, 2);
cx->ks[3] = ss[3] = word_in(in_key, 3);
#if DEC_UNROLL == NONE
cx->ks[4] = ss[4] = word_in(in_key, 4);
cx->ks[5] = ss[5] = word_in(in_key, 5);
cx->ks[6] = ss[6] = word_in(in_key, 6);
cx->ks[7] = ss[7] = word_in(in_key, 7);
{ aes_32t i;
for(i = 0; i < (15 * N_COLS - 1) / 8; ++i)
ke8(cx->ks, i);
#if !(DEC_ROUND == NO_TABLES)
for(i = N_COLS; i < 14 * N_COLS; ++i)
cx->ks[i] = inv_mcol(cx->ks[i]);
#endif
}
#else
cx->ks[4] = ff(ss[4] = word_in(in_key, 4));
cx->ks[5] = ff(ss[5] = word_in(in_key, 5));
cx->ks[6] = ff(ss[6] = word_in(in_key, 6));
cx->ks[7] = ff(ss[7] = word_in(in_key, 7));
kdf8(cx->ks, 0); kd8(cx->ks, 1);
kd8(cx->ks, 2); kd8(cx->ks, 3);
kd8(cx->ks, 4); kd8(cx->ks, 5);
kdl8(cx->ks, 6);
#endif
#ifdef AES_ERR_CHK
return aes_good;
#endif
}
#endif
#if defined(AES_VAR)
aes_rval aes_decrypt_key(const void *in_key, int key_len, aes_decrypt_ctx cx[1])
{
switch(key_len)
{
#ifdef AES_ERR_CHK
case 16: case 128: return aes_decrypt_key128(in_key, cx);
case 24: case 192: return aes_decrypt_key192(in_key, cx);
case 32: case 256: return aes_decrypt_key256(in_key, cx);
default: return aes_error;
#else
case 16: case 128: aes_decrypt_key128(in_key, cx); return;
case 24: case 192: aes_decrypt_key192(in_key, cx); return;
case 32: case 256: aes_decrypt_key256(in_key, cx); return;
#endif
}
}
#endif
#endif
#if defined(__cplusplus)
}
#endif

1029
aesopt.h

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

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@@ -1,82 +0,0 @@
#!/usr/bin/perl
#
# Simple AGI application to play mp3's selected by a user both using
# xmms and over the phone itself.
#
$|=1;
while(<STDIN>) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
print STDERR "AGI Environment Dump:\n";
foreach $i (sort keys %AGI) {
print STDERR " -- $i = $AGI{$i}\n";
}
dbmopen(%DIGITS, "/var/lib/asterisk/mp3list", 0644) || die("Unable to open mp3list");;
sub checkresult {
my ($res) = @_;
my $retval;
$tests++;
chomp $res;
if ($res =~ /^200/) {
$res =~ /result=(-?[\w\*\#]+)/;
return $1;
} else {
return -1;
}
}
#print STDERR "1. Playing beep...\n";
#print "STREAM FILE beep \"\"\n";
#$result = <STDIN>;
#checkresult($result);
print STDERR "2. Getting song name...\n";
print "GET DATA demo-enterkeywords\n";
$result = <STDIN>;
$digitstr = checkresult($result);
if ($digitstr < 0) {
exit(1);
}
$digitstr =~ s/\*/ /g;
print STDERR "Resulting songname is $digitstr\n";
@searchwords = split (/\s+/, $digitstr);
print STDERR "Searchwords: " . join(':', @searchwords) . "\n";
foreach $key (sort keys %DIGITS) {
@words = split(/\s+/, $DIGITS{$key});
$match = 1;
foreach $search (@searchwords) {
$match = 0 unless grep(/$search/, @words);
}
if ($match > 0) {
print STDERR "File $key matches\n";
# Play a beep
print "STREAM FILE beep \"\"\n";
system("xmms", $key);
$result = <STDIN>;
if (&checkresult($result) < 0) {
exit 0;
}
print "EXEC MP3Player \"$key\"\n";
# print "WAIT FOR DIGIT 60000\n";
$result = <STDIN>;
if (&checkresult($result) < 0) {
exit 0;
}
print STDERR "Got here...\n";
}
}
print STDERR "4. Testing 'saynumber' of $digitstr...\n";
print "STREAM FILE demo-nomatch\"\"\n";
$result = <STDIN>;
checkresult($result);

View File

@@ -1,52 +0,0 @@
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for AGI-related stuff
#
# Copyright (C) 1999-2005, Digium
#
# Mark Spencer <markster@digium.com>
#
# This program is free software, distributed under the terms of
# the GNU General Public License
#
AGIS=agi-test.agi eagi-test eagi-sphinx-test
CFLAGS+=-DNO_AST_MM
LIBS=
ifeq ($(OSARCH),SunOS)
LIBS=-lsocket -lnsl ../strcompat.o
endif
ifeq ($(findstring BSD,${OSARCH}),BSD)
CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
all: depend $(AGIS)
install: all
mkdir -p $(DESTDIR)$(AGI_DIR)
for x in $(AGIS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(AGI_DIR) ; done
eagi-test: eagi-test.o
$(CC) $(CFLAGS) -o eagi-test eagi-test.o $(LIBS)
eagi-sphinx-test: eagi-sphinx-test.o
$(CC) $(CFLAGS) -o eagi-sphinx-test eagi-sphinx-test.o $(LIBS)
clean:
rm -f *.so *.o look .depend eagi-test eagi-sphinx-test
%.so : %.o
$(CC) -shared -Xlinker -x -o $@ $<
ifneq ($(wildcard .depend),)
include .depend
endif
depend: .depend
.depend:
../build_tools/mkdep $(CFLAGS) `ls *.c`

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@@ -1,79 +0,0 @@
#!/usr/bin/perl
use strict;
$|=1;
# Setup some variables
my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;
while(<STDIN>) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
print STDERR "AGI Environment Dump:\n";
foreach my $i (sort keys %AGI) {
print STDERR " -- $i = $AGI{$i}\n";
}
sub checkresult {
my ($res) = @_;
my $retval;
$tests++;
chomp $res;
if ($res =~ /^200/) {
$res =~ /result=(-?\d+)/;
if (!length($1)) {
print STDERR "FAIL ($res)\n";
$fail++;
} else {
print STDERR "PASS ($1)\n";
$pass++;
}
} else {
print STDERR "FAIL (unexpected result '$res')\n";
$fail++;
}
}
print STDERR "1. Testing 'sendfile'...";
print "STREAM FILE beep \"\"\n";
my $result = <STDIN>;
&checkresult($result);
print STDERR "2. Testing 'sendtext'...";
print "SEND TEXT \"hello world\"\n";
my $result = <STDIN>;
&checkresult($result);
print STDERR "3. Testing 'sendimage'...";
print "SEND IMAGE asterisk-image\n";
my $result = <STDIN>;
&checkresult($result);
print STDERR "4. Testing 'saynumber'...";
print "SAY NUMBER 192837465 \"\"\n";
my $result = <STDIN>;
&checkresult($result);
print STDERR "5. Testing 'waitdtmf'...";
print "WAIT FOR DIGIT 1000\n";
my $result = <STDIN>;
&checkresult($result);
print STDERR "6. Testing 'record'...";
print "RECORD FILE testagi gsm 1234 3000\n";
my $result = <STDIN>;
&checkresult($result);
print STDERR "6a. Testing 'record' playback...";
print "STREAM FILE testagi \"\"\n";
my $result = <STDIN>;
&checkresult($result);
print STDERR "================== Complete ======================\n";
print STDERR "$tests tests completed, $pass passed, $fail failed\n";
print STDERR "==================================================\n";

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@@ -1,219 +0,0 @@
/*
* Extended AGI test application
*
* This code is released into public domain
* without any warranty of any kind.
*
*/
#include <stdio.h>
#include <unistd.h>
#include <stdlib.h>
#include <errno.h>
#include <string.h>
#include <sys/select.h>
#include <fcntl.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <netdb.h>
#include <asterisk/compat.h>
#define AUDIO_FILENO (STDERR_FILENO + 1)
#define SPHINX_HOST "192.168.1.108"
#define SPHINX_PORT 3460
static int sphinx_sock = -1;
static int connect_sphinx(void)
{
struct hostent *hp;
struct sockaddr_in sin;
int res;
hp = gethostbyname(SPHINX_HOST);
if (!hp) {
fprintf(stderr, "Unable to resolve '%s'\n", SPHINX_HOST);
return -1;
}
sphinx_sock = socket(PF_INET, SOCK_STREAM, 0);
if (sphinx_sock < 0) {
fprintf(stderr, "Unable to allocate socket: %s\n", strerror(errno));
return -1;
}
memset(&sin, 0, sizeof(sin));
sin.sin_family = AF_INET;
sin.sin_port = htons(SPHINX_PORT);
memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
if (connect(sphinx_sock, (struct sockaddr *)&sin, sizeof(sin))) {
fprintf(stderr, "Unable to connect on socket: %s\n", strerror(errno));
close(sphinx_sock);
sphinx_sock = -1;
return -1;
}
res = fcntl(sphinx_sock, F_GETFL);
if ((res < 0) || (fcntl(sphinx_sock, F_SETFL, res | O_NONBLOCK) < 0)) {
fprintf(stderr, "Unable to set flags on socket: %s\n", strerror(errno));
close(sphinx_sock);
sphinx_sock = -1;
return -1;
}
return 0;
}
static int read_environment(void)
{
char buf[256];
char *val;
/* Read environment */
for(;;) {
fgets(buf, sizeof(buf), stdin);
if (feof(stdin))
return -1;
buf[strlen(buf) - 1] = '\0';
/* Check for end of environment */
if (!strlen(buf))
return 0;
val = strchr(buf, ':');
if (!val) {
fprintf(stderr, "Invalid environment: '%s'\n", buf);
return -1;
}
*val = '\0';
val++;
val++;
/* Skip space */
fprintf(stderr, "Environment: '%s' is '%s'\n", buf, val);
/* Load into normal environment */
setenv(buf, val, 1);
}
/* Never reached */
return 0;
}
static char *wait_result(void)
{
fd_set fds;
int res;
int max;
static char astresp[256];
static char sphinxresp[256];
char audiobuf[4096];
for (;;) {
FD_ZERO(&fds);
FD_SET(STDIN_FILENO, &fds);
FD_SET(AUDIO_FILENO, &fds);
max = AUDIO_FILENO;
if (sphinx_sock > -1) {
FD_SET(sphinx_sock, &fds);
if (sphinx_sock > max)
max = sphinx_sock;
}
/* Wait for *some* sort of I/O */
res = select(max + 1, &fds, NULL, NULL, NULL);
if (res < 0) {
fprintf(stderr, "Error in select: %s\n", strerror(errno));
return NULL;
}
if (FD_ISSET(STDIN_FILENO, &fds)) {
fgets(astresp, sizeof(astresp), stdin);
if (feof(stdin)) {
fprintf(stderr, "Got hungup on apparently\n");
return NULL;
}
astresp[strlen(astresp) - 1] = '\0';
fprintf(stderr, "Ooh, got a response from Asterisk: '%s'\n", astresp);
return astresp;
}
if (FD_ISSET(AUDIO_FILENO, &fds)) {
res = read(AUDIO_FILENO, audiobuf, sizeof(audiobuf));
if (res > 0) {
if (sphinx_sock > -1)
write(sphinx_sock, audiobuf, res);
}
}
if ((sphinx_sock > -1) && FD_ISSET(sphinx_sock, &fds)) {
res = read(sphinx_sock, sphinxresp, sizeof(sphinxresp));
if (res > 0) {
fprintf(stderr, "Oooh, Sphinx found a token: '%s'\n", sphinxresp);
return sphinxresp;
} else if (res == 0) {
fprintf(stderr, "Hrm, lost sphinx, guess we're on our own\n");
close(sphinx_sock);
sphinx_sock = -1;
}
}
}
}
static char *run_command(char *command)
{
fprintf(stdout, "%s\n", command);
return wait_result();
}
static int run_script(void)
{
char *res;
res = run_command("STREAM FILE demo-enterkeywords 0123456789*#");
if (!res) {
fprintf(stderr, "Failed to execute command\n");
return -1;
}
fprintf(stderr, "1. Result is '%s'\n", res);
res = run_command("STREAM FILE demo-nomatch 0123456789*#");
if (!res) {
fprintf(stderr, "Failed to execute command\n");
return -1;
}
fprintf(stderr, "2. Result is '%s'\n", res);
res = run_command("SAY NUMBER 23452345 0123456789*#");
if (!res) {
fprintf(stderr, "Failed to execute command\n");
return -1;
}
fprintf(stderr, "3. Result is '%s'\n", res);
res = run_command("GET DATA demo-enterkeywords");
if (!res) {
fprintf(stderr, "Failed to execute command\n");
return -1;
}
fprintf(stderr, "4. Result is '%s'\n", res);
res = run_command("STREAM FILE auth-thankyou \"\"");
if (!res) {
fprintf(stderr, "Failed to execute command\n");
return -1;
}
fprintf(stderr, "5. Result is '%s'\n", res);
return 0;
}
int main(int argc, char *argv[])
{
char *tmp;
int ver = 0;
int subver = 0;
/* Setup stdin/stdout for line buffering */
setlinebuf(stdin);
setlinebuf(stdout);
if (read_environment()) {
fprintf(stderr, "Failed to read environment: %s\n", strerror(errno));
exit(1);
}
connect_sphinx();
tmp = getenv("agi_enhanced");
if (tmp) {
if (sscanf(tmp, "%d.%d", &ver, &subver) != 2)
ver = 0;
}
if (ver < 1) {
fprintf(stderr, "No enhanced AGI services available. Use EAGI, not AGI\n");
exit(1);
}
if (run_script())
return -1;
exit(0);
}

View File

@@ -1,162 +0,0 @@
/*
* Extended AGI test application
*
* This code is released into the public domain
* with no warranty of any kind
*/
#include <stdio.h>
#include <unistd.h>
#include <stdlib.h>
#include <errno.h>
#include <string.h>
#include <sys/select.h>
#include <asterisk/compat.h>
#define AUDIO_FILENO (STDERR_FILENO + 1)
static int read_environment(void)
{
char buf[256];
char *val;
/* Read environment */
for(;;) {
fgets(buf, sizeof(buf), stdin);
if (feof(stdin))
return -1;
buf[strlen(buf) - 1] = '\0';
/* Check for end of environment */
if (!strlen(buf))
return 0;
val = strchr(buf, ':');
if (!val) {
fprintf(stderr, "Invalid environment: '%s'\n", buf);
return -1;
}
*val = '\0';
val++;
val++;
/* Skip space */
fprintf(stderr, "Environment: '%s' is '%s'\n", buf, val);
/* Load into normal environment */
setenv(buf, val, 1);
}
/* Never reached */
return 0;
}
static char *wait_result(void)
{
fd_set fds;
int res;
int bytes = 0;
static char astresp[256];
char audiobuf[4096];
for (;;) {
FD_ZERO(&fds);
FD_SET(STDIN_FILENO, &fds);
FD_SET(AUDIO_FILENO, &fds);
/* Wait for *some* sort of I/O */
res = select(AUDIO_FILENO + 1, &fds, NULL, NULL, NULL);
if (res < 0) {
fprintf(stderr, "Error in select: %s\n", strerror(errno));
return NULL;
}
if (FD_ISSET(STDIN_FILENO, &fds)) {
fgets(astresp, sizeof(astresp), stdin);
if (feof(stdin)) {
fprintf(stderr, "Got hungup on apparently\n");
return NULL;
}
astresp[strlen(astresp) - 1] = '\0';
fprintf(stderr, "Ooh, got a response from Asterisk: '%s'\n", astresp);
return astresp;
}
if (FD_ISSET(AUDIO_FILENO, &fds)) {
res = read(AUDIO_FILENO, audiobuf, sizeof(audiobuf));
if (res > 0) {
/* XXX Process the audio with sphinx here XXX */
#if 0
fprintf(stderr, "Got %d/%d bytes of audio\n", res, bytes);
#endif
bytes += res;
/* Prentend we detected some audio after 3 seconds */
if (bytes > 16000 * 3) {
return "Sample Message";
bytes = 0;
}
}
}
}
}
static char *run_command(char *command)
{
fprintf(stdout, "%s\n", command);
return wait_result();
}
static int run_script(void)
{
char *res;
res = run_command("STREAM FILE demo-enterkeywords 0123456789*#");
if (!res) {
fprintf(stderr, "Failed to execute command\n");
return -1;
}
fprintf(stderr, "1. Result is '%s'\n", res);
res = run_command("STREAM FILE demo-nomatch 0123456789*#");
if (!res) {
fprintf(stderr, "Failed to execute command\n");
return -1;
}
fprintf(stderr, "2. Result is '%s'\n", res);
res = run_command("SAY NUMBER 23452345 0123456789*#");
if (!res) {
fprintf(stderr, "Failed to execute command\n");
return -1;
}
fprintf(stderr, "3. Result is '%s'\n", res);
res = run_command("GET DATA demo-enterkeywords");
if (!res) {
fprintf(stderr, "Failed to execute command\n");
return -1;
}
fprintf(stderr, "4. Result is '%s'\n", res);
res = run_command("STREAM FILE auth-thankyou \"\"");
if (!res) {
fprintf(stderr, "Failed to execute command\n");
return -1;
}
fprintf(stderr, "5. Result is '%s'\n", res);
return 0;
}
int main(int argc, char *argv[])
{
char *tmp;
int ver = 0;
int subver = 0;
/* Setup stdin/stdout for line buffering */
setlinebuf(stdin);
setlinebuf(stdout);
if (read_environment()) {
fprintf(stderr, "Failed to read environment: %s\n", strerror(errno));
exit(1);
}
tmp = getenv("agi_enhanced");
if (tmp) {
if (sscanf(tmp, "%d.%d", &ver, &subver) != 2)
ver = 0;
}
if (ver < 1) {
fprintf(stderr, "No enhanced AGI services available. Use EAGI, not AGI\n");
exit(1);
}
if (run_script())
return -1;
exit(0);
}

View File

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

View File

@@ -1,44 +0,0 @@
#!/usr/bin/perl
#
# Build a database linking filenames to their numerical representations
# using a keypad for the DialAnMp3 application
#
$mp3dir="/usr/media/mpeg3";
dbmopen(%DIGITS, "/var/lib/asterisk/mp3list", 0644) || die("Unable to open mp3list");;
sub process_dir {
my ($dir) = @_;
my $file;
my $digits;
my @entries;
opendir(DIR, $dir);
@entries = readdir(DIR);
closedir(DIR);
foreach $_ (@entries) {
if (!/^\./) {
$file = "$dir/$_";
if (-d "$file") {
process_dir("$file");
} else {
$digits = $_;
$digits =~ s/[^ \w]+//g;
$digits =~ s/\_/ /g;
$digits =~ tr/[a-z]/[A-Z]/;
$digits =~ tr/[A-C]/2/;
$digits =~ tr/[D-F]/3/;
$digits =~ tr/[G-I]/4/;
$digits =~ tr/[J-L]/5/;
$digits =~ tr/[M-O]/6/;
$digits =~ tr/[P-S]/7/;
$digits =~ tr/[T-V]/8/;
$digits =~ tr/[W-Z]/9/;
$digits =~ s/\s+/ /;
print "File: $file, digits: $digits\n";
$DIGITS{$file} = $digits;
}
}
}
}
process_dir($mp3dir);

100
alaw.c
View File

@@ -1,100 +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 u-Law to Signed linear conversion
*
*/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/alaw.h"
#define AMI_MASK 0x55
static inline unsigned char linear2alaw (short int linear)
{
int mask;
int seg;
int pcm_val;
static int seg_end[8] =
{
0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF, 0x1FFF, 0x3FFF, 0x7FFF
};
pcm_val = linear;
if (pcm_val >= 0)
{
/* Sign (7th) bit = 1 */
mask = AMI_MASK | 0x80;
}
else
{
/* Sign bit = 0 */
mask = AMI_MASK;
pcm_val = -pcm_val;
}
/* Convert the scaled magnitude to segment number. */
for (seg = 0; seg < 8; seg++)
{
if (pcm_val <= seg_end[seg])
break;
}
/* Combine the sign, segment, and quantization bits. */
return ((seg << 4) | ((pcm_val >> ((seg) ? (seg + 3) : 4)) & 0x0F)) ^ mask;
}
/*- End of function --------------------------------------------------------*/
static inline short int alaw2linear (unsigned char alaw)
{
int i;
int seg;
alaw ^= AMI_MASK;
i = ((alaw & 0x0F) << 4);
seg = (((int) alaw & 0x70) >> 4);
if (seg)
i = (i + 0x100) << (seg - 1);
return (short int) ((alaw & 0x80) ? i : -i);
}
unsigned char __ast_lin2a[8192];
short __ast_alaw[256];
void ast_alaw_init(void)
{
int i;
/*
* Set up mu-law conversion table
*/
for(i = 0;i < 256;i++)
{
__ast_alaw[i] = alaw2linear(i);
}
/* set up the reverse (mu-law) conversion table */
for(i = -32768; i < 32768; i++)
{
__ast_lin2a[((unsigned short)i) >> 3] = linear2alaw(i);
}
}

1633
app.c

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@@ -1,130 +1,27 @@
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for PBX applications
# Makefile for PBX frontends (dynamically loaded)
#
# Copyright (C) 1999-2005, 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
#
APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
app_system.so app_echo.so app_record.so app_image.so app_url.so app_disa.so \
app_adsiprog.so app_getcpeid.so app_milliwatt.so \
app_zapateller.so app_setcallerid.so app_festival.so \
app_queue.so app_senddtmf.so app_parkandannounce.so \
app_setcidname.so app_lookupcidname.so app_macro.so \
app_authenticate.so app_softhangup.so app_lookupblacklist.so \
app_waitforring.so app_privacy.so app_db.so app_chanisavail.so \
app_enumlookup.so app_transfer.so app_setcidnum.so app_cdr.so \
app_hasnewvoicemail.so app_sayunixtime.so app_cut.so app_read.so \
app_setcdruserfield.so app_random.so app_ices.so app_eval.so \
app_nbscat.so app_sendtext.so app_exec.so \
app_groupcount.so app_txtcidname.so app_controlplayback.so \
app_talkdetect.so app_alarmreceiver.so app_userevent.so app_verbose.so \
app_test.so app_forkcdr.so app_math.so app_realtime.so \
app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \
app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so \
app_dictate.so app_externalivr.so app_directed_pickup.so \
app_mixmonitor.so app_stack.so
APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_intercom.so app_mp3.so app_system.so app_echo.so
#
# Obsolete things...
#
#APPS+=app_sql_postgres.so
#APPS+=app_sql_odbc.so
#
# Experimental things
#
#APPS+=app_ivrdemo.so
#APPS+=app_skel.so
#APPS+=app_rpt.so
ifndef WITHOUT_ZAPTEL
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/zaptel.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/zaptel.h),)
APPS+=app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so app_page.so
endif
endif # WITHOUT_ZAPTEL
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/osp/osp.h $(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
APPS+=app_osplookup.so
endif
ifeq ($(findstring BSD,${OSARCH}),BSD)
CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
ifneq ($(shell if [[ 0x`$(CROSS_COMPILE_BIN)curl-config --vernum` -ge 0x70907 ]]; then echo "OK" ; fi),)
ifneq (${CURLLIBS},)
APPS+=app_curl.so
endif
endif
ifeq (${OSARCH},CYGWIN)
CYGSOLINK=-Wl,--out-implib=lib$@.a -Wl,--export-all-symbols
CYGSOLIB=-L.. -L. -L../res -lasterisk.dll -lres_features.so -lres_adsi.so -lres_monitor.so
else
CFLAGS+=-fPIC
APPS+=app_sms.so
endif
#
# If you have UnixODBC you can use ODBC voicemail
# storage
#
# Uncomment to use ODBC storage
#CFLAGS+=-DUSE_ODBC_STORAGE
# Uncomment for extended ODBC voicemail storage
#CFLAGS+=-DEXTENDED_ODBC_STORAGE
# See doc/README.odbcstorage for more information
CFLAGS+=
all: $(APPS)
clean:
rm -f *.so *.o .depend
rm -f *.so *.o
%.so : %.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB}
app_rpt.so : app_rpt.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -ltonezone
$(CC) -shared -Xlinker -x -o $@ $<
install: all
for x in $(APPS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done
rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} $(CURLLIBS)
app_sql_postgres.o: app_sql_postgres.c
$(CC) -pipe -I$(CROSS_COMPILE_TARGET)/usr/local/pgsql/include -I$(CROSS_COMPILE_TARGET)/usr/include/postgresql $(CFLAGS) -c -o app_sql_postgres.o app_sql_postgres.c
app_sql_postgres.so: app_sql_postgres.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -L/usr/local/pgsql/lib -lpq
app_sql_odbc.so: app_sql_odbc.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -lodbc
ifeq (SunOS,$(shell uname))
app_chanspy.so: app_chanspy.o
$(CC) $(SOLINK) -o $@ $< -lrt
endif
ifneq ($(wildcard .depend),)
include .depend
endif
depend: .depend
.depend:
../build_tools/mkdep $(CFLAGS) `ls *.c`
env:
env
for x in $(APPS); do $(INSTALL) -m 755 $$x $(MODULES_DIR) ; done

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@@ -1,868 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2004 - 2005 Steve Rodgers
*
* Steve Rodgers <hwstar@rodgers.sdcoxmail.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 Central Station Alarm receiver for Ademco Contact ID
* \author Steve Rodgers <hwstar@rodgers.sdcoxmail.com>
*
* *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING ***
*
* Use at your own risk. Please consult the GNU GPL license document included with Asterisk details. *
*
* *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING ***
*
* \ingroup applications
*/
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <math.h>
#include <sys/wait.h>
#include <unistd.h>
#include <sys/time.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/ulaw.h"
#include "asterisk/options.h"
#include "asterisk/app.h"
#include "asterisk/dsp.h"
#include "asterisk/config.h"
#include "asterisk/localtime.h"
#include "asterisk/callerid.h"
#include "asterisk/astdb.h"
#define ALMRCV_CONFIG "alarmreceiver.conf"
#define ADEMCO_CONTACT_ID "ADEMCO_CONTACT_ID"
struct event_node{
char data[17];
struct event_node *next;
};
typedef struct event_node event_node_t;
static char *tdesc = "Alarm Receiver for Asterisk";
static char *app = "AlarmReceiver";
static char *synopsis = "Provide support for receving alarm reports from a burglar or fire alarm panel";
static char *descrip =
" AlarmReceiver(): Only 1 signalling format is supported at this time: Ademco\n"
"Contact ID. This application should be called whenever there is an alarm\n"
"panel calling in to dump its events. The application will handshake with the\n"
"alarm panel, and receive events, validate them, handshake them, and store them\n"
"until the panel hangs up. Once the panel hangs up, the application will run the\n"
"system command specified by the eventcmd setting in alarmreceiver.conf and pipe\n"
"the events to the standard input of the application. The configuration file also\n"
"contains settings for DTMF timing, and for the loudness of the acknowledgement\n"
"tones.\n";
/* Config Variables */
static int fdtimeout = 2000;
static int sdtimeout = 200;
static int toneloudness = 4096;
static int log_individual_events = 0;
static char event_spool_dir[128] = {'\0'};
static char event_app[128] = {'\0'};
static char db_family[128] = {'\0'};
static char time_stamp_format[128] = {"%a %b %d, %Y @ %H:%M:%S %Z"};
/* Misc variables */
static char event_file[14] = "/event-XXXXXX";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
/*
* Attempt to access a database variable and increment it,
* provided that the user defined db-family in alarmreceiver.conf
* The alarmreceiver app will write statistics to a few variables
* in this family if it is defined. If the new key doesn't exist in the
* family, then create it and set its value to 1.
*/
static void database_increment( char *key )
{
int res = 0;
unsigned v;
char value[16];
if (ast_strlen_zero(db_family))
return; /* If not defined, don't do anything */
res = ast_db_get(db_family, key, value, sizeof(value) - 1);
if(res){
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Creating database entry %s and setting to 1\n", key);
/* Guess we have to create it */
res = ast_db_put(db_family, key, "1");
return;
}
sscanf(value, "%30u", &v);
v++;
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: New value for %s: %u\n", key, v);
snprintf(value, sizeof(value), "%u", v);
res = ast_db_put(db_family, key, value);
if((res)&&(option_verbose >= 4))
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: database_increment write error\n");
return;
}
/*
* Build a MuLaw data block for a single frequency tone
*/
static void make_tone_burst(unsigned char *data, float freq, float loudness, int len, int *x)
{
int i;
float val;
for(i = 0; i < len; i++){
val = loudness * sin((freq * 2.0 * M_PI * (*x)++)/8000.0);
data[i] = AST_LIN2MU((int)val);
}
/* wrap back around from 8000 */
if (*x >= 8000) *x = 0;
return;
}
/*
* Send a single tone burst for a specifed duration and frequency.
* Returns 0 if successful
*/
static int send_tone_burst(struct ast_channel *chan, float freq, int duration, int tldn)
{
int res = 0;
int i = 0;
int x = 0;
struct ast_frame *f, wf;
struct {
unsigned char offset[AST_FRIENDLY_OFFSET];
unsigned char buf[640];
} tone_block;
for(;;)
{
if (ast_waitfor(chan, -1) < 0){
res = -1;
break;
}
f = ast_read(chan);
if (!f){
res = -1;
break;
}
if (f->frametype == AST_FRAME_VOICE) {
wf.frametype = AST_FRAME_VOICE;
wf.subclass = AST_FORMAT_ULAW;
wf.offset = AST_FRIENDLY_OFFSET;
wf.mallocd = 0;
wf.data = tone_block.buf;
wf.datalen = f->datalen;
wf.samples = wf.datalen;
make_tone_burst(tone_block.buf, freq, (float) tldn, wf.datalen, &x);
i += wf.datalen / 8;
if (i > duration) {
ast_frfree(f);
break;
}
if (ast_write(chan, &wf)){
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Failed to write frame on %s\n", chan->name);
ast_log(LOG_WARNING, "AlarmReceiver Failed to write frame on %s\n",chan->name);
res = -1;
ast_frfree(f);
break;
}
}
ast_frfree(f);
}
return res;
}
/*
* Receive a string of DTMF digits where the length of the digit string is known in advance. Do not give preferential
* treatment to any digit value, and allow separate time out values to be specified for the first digit and all subsequent
* digits.
*
* Returns 0 if all digits successfully received.
* Returns 1 if a digit time out occurred
* Returns -1 if the caller hung up or there was a channel error.
*
*/
static int receive_dtmf_digits(struct ast_channel *chan, char *digit_string, int length, int fdto, int sdto)
{
int res = 0;
int i = 0;
int r;
struct ast_frame *f;
struct timeval lastdigittime;
lastdigittime = ast_tvnow();
for(;;){
/* if outa time, leave */
if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) >
((i > 0) ? sdto : fdto)){
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: DTMF Digit Timeout on %s\n", chan->name);
ast_log(LOG_DEBUG,"AlarmReceiver: DTMF timeout on chan %s\n",chan->name);
res = 1;
break;
}
if ((r = ast_waitfor(chan, -1) < 0)) {
ast_log(LOG_DEBUG, "Waitfor returned %d\n", r);
continue;
}
f = ast_read(chan);
if (f == NULL){
res = -1;
break;
}
/* If they hung up, leave */
if ((f->frametype == AST_FRAME_CONTROL) &&
(f->subclass == AST_CONTROL_HANGUP)){
ast_frfree(f);
res = -1;
break;
}
/* if not DTMF, just do it again */
if (f->frametype != AST_FRAME_DTMF){
ast_frfree(f);
continue;
}
digit_string[i++] = f->subclass; /* save digit */
ast_frfree(f);
/* If we have all the digits we expect, leave */
if(i >= length)
break;
lastdigittime = ast_tvnow();
}
digit_string[i] = '\0'; /* Nul terminate the end of the digit string */
return res;
}
/*
* Write the metadata to the log file
*/
static int write_metadata( FILE *logfile, char *signalling_type, struct ast_channel *chan)
{
int res = 0;
time_t t;
struct tm now;
char *cl,*cn;
char workstring[80];
char timestamp[80];
/* Extract the caller ID location */
if (chan->cid.cid_num)
ast_copy_string(workstring, chan->cid.cid_num, sizeof(workstring));
workstring[sizeof(workstring) - 1] = '\0';
ast_callerid_parse(workstring, &cn, &cl);
if (cl)
ast_shrink_phone_number(cl);
/* Get the current time */
time(&t);
ast_localtime(&t, &now, NULL);
/* Format the time */
strftime(timestamp, sizeof(timestamp), time_stamp_format, &now);
res = fprintf(logfile, "\n\n[metadata]\n\n");
if(res >= 0)
res = fprintf(logfile, "PROTOCOL=%s\n", signalling_type);
if(res >= 0)
res = fprintf(logfile, "CALLINGFROM=%s\n", (!cl) ? "<unknown>" : cl);
if(res >- 0)
res = fprintf(logfile, "CALLERNAME=%s\n", (!cn) ? "<unknown>" : cn);
if(res >= 0)
res = fprintf(logfile, "TIMESTAMP=%s\n\n", timestamp);
if(res >= 0)
res = fprintf(logfile, "[events]\n\n");
if(res < 0){
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: can't write metadata\n");
ast_log(LOG_DEBUG,"AlarmReceiver: can't write metadata\n");
}
else
res = 0;
return res;
}
/*
* Write a single event to the log file
*/
static int write_event( FILE *logfile, event_node_t *event)
{
int res = 0;
if( fprintf(logfile, "%s\n", event->data) < 0)
res = -1;
return res;
}
/*
* If we are configured to log events, do so here.
*
*/
static int log_events(struct ast_channel *chan, char *signalling_type, event_node_t *event)
{
int res = 0;
char workstring[sizeof(event_spool_dir)+sizeof(event_file)] = "";
int fd;
FILE *logfile;
event_node_t *elp = event;
if (!ast_strlen_zero(event_spool_dir)) {
/* Make a template */
ast_copy_string(workstring, event_spool_dir, sizeof(workstring));
strncat(workstring, event_file, sizeof(workstring) - strlen(workstring) - 1);
/* Make the temporary file */
fd = mkstemp(workstring);
if(fd == -1){
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: can't make temporary file\n");
ast_log(LOG_DEBUG,"AlarmReceiver: can't make temporary file\n");
res = -1;
}
if(!res){
logfile = fdopen(fd, "w");
if(logfile){
/* Write the file */
res = write_metadata(logfile, signalling_type, chan);
if(!res)
while((!res) && (elp != NULL)){
res = write_event(logfile, elp);
elp = elp->next;
}
if(!res){
if(fflush(logfile) == EOF)
res = -1;
if(!res){
if(fclose(logfile) == EOF)
res = -1;
}
}
}
else
res = -1;
}
}
return res;
}
/*
* This function implements the logic to receive the Ademco contact ID format.
*
* The function will return 0 when the caller hangs up, else a -1 if there was a problem.
*/
static int receive_ademco_contact_id( struct ast_channel *chan, void *data, int fdto, int sdto, int tldn, event_node_t **ehead)
{
int i,j;
int res = 0;
int checksum;
char event[17];
event_node_t *enew, *elp;
int got_some_digits = 0;
int events_received = 0;
int ack_retries = 0;
static char digit_map[15] = "0123456789*#ABC";
static unsigned char digit_weights[15] = {10,1,2,3,4,5,6,7,8,9,11,12,13,14,15};
database_increment("calls-received");
/* Wait for first event */
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Waiting for first event from panel\n");
while(res >= 0){
if(got_some_digits == 0){
/* Send ACK tone sequence */
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Sending 1400Hz 100ms burst (ACK)\n");
res = send_tone_burst(chan, 1400.0, 100, tldn);
if(!res)
res = ast_safe_sleep(chan, 100);
if(!res){
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Sending 2300Hz 100ms burst (ACK)\n");
res = send_tone_burst(chan, 2300.0, 100, tldn);
}
}
if( res >= 0)
res = receive_dtmf_digits(chan, event, sizeof(event) - 1, fdto, sdto);
if (res < 0){
if(events_received == 0)
/* Hangup with no events received should be logged in the DB */
database_increment("no-events-received");
else{
if(ack_retries){
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: ACK retries during this call: %d\n", ack_retries);
database_increment("ack-retries");
}
}
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: App exiting...\n");
res = -1;
break;
}
if(res != 0){
/* Didn't get all of the digits */
if(option_verbose >= 2)
ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Incomplete string: %s, trying again...\n", event);
if(!got_some_digits){
got_some_digits = (!ast_strlen_zero(event)) ? 1 : 0;
ack_retries++;
}
continue;
}
got_some_digits = 1;
if(option_verbose >= 2)
ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Received Event %s\n", event);
ast_log(LOG_DEBUG, "AlarmReceiver: Received event: %s\n", event);
/* Calculate checksum */
for(j = 0, checksum = 0; j < 16; j++){
for(i = 0 ; i < sizeof(digit_map) ; i++){
if(digit_map[i] == event[j])
break;
}
if(i == 16)
break;
checksum += digit_weights[i];
}
if(i == 16){
if(option_verbose >= 2)
ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Bad DTMF character %c, trying again\n", event[j]);
continue; /* Bad character */
}
/* Checksum is mod(15) of the total */
checksum = checksum % 15;
if(checksum){
database_increment("checksum-errors");
if(option_verbose >= 2)
ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Nonzero checksum\n");
ast_log(LOG_DEBUG, "AlarmReceiver: Nonzero checksum\n");
continue;
}
/* Check the message type for correctness */
if(strncmp(event + 4, "18", 2)){
if(strncmp(event + 4, "98", 2)){
database_increment("format-errors");
if(option_verbose >= 2)
ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Wrong message type\n");
ast_log(LOG_DEBUG, "AlarmReceiver: Wrong message type\n");
continue;
}
}
events_received++;
/* Queue the Event */
if((enew = malloc(sizeof(event_node_t))) == NULL){
if(option_verbose >= 1)
ast_verbose(VERBOSE_PREFIX_1 "AlarmReceiver: Failed to allocate memory\n");
ast_log(LOG_WARNING, "AlarmReceiver Failed to allocate memory\n");
res = -1;
break;
}
memset(enew, 0, sizeof(event_node_t));
enew->next = NULL;
ast_copy_string(enew->data, event, sizeof(enew->data));
/*
* Insert event onto end of list
*/
if(*ehead == NULL){
*ehead = enew;
}
else{
for(elp = *ehead; elp->next != NULL; elp = elp->next)
;
elp->next = enew;
}
if(res > 0)
res = 0;
/* Let the user have the option of logging the single event before sending the kissoff tone */
if((res == 0) && (log_individual_events))
res = log_events(chan, ADEMCO_CONTACT_ID, enew);
/* Wait 200 msec before sending kissoff */
if(res == 0)
res = ast_safe_sleep(chan, 200);
/* Send the kissoff tone */
if(res == 0)
res = send_tone_burst(chan, 1400.0, 900, tldn);
}
return res;
}
/*
* This is the main function called by Asterisk Core whenever the App is invoked in the extension logic.
* This function will always return 0.
*/
static int alarmreceiver_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u;
event_node_t *elp, *efree;
char signalling_type[64] = "";
event_node_t *event_head = NULL;
LOCAL_USER_ADD(u);
/* Set write and read formats to ULAW */
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Setting read and write formats to ULAW\n");
if (ast_set_write_format(chan,AST_FORMAT_ULAW)){
ast_log(LOG_WARNING, "AlarmReceiver: Unable to set write format to Mu-law on %s\n",chan->name);
LOCAL_USER_REMOVE(u);
return -1;
}
if (ast_set_read_format(chan,AST_FORMAT_ULAW)){
ast_log(LOG_WARNING, "AlarmReceiver: Unable to set read format to Mu-law on %s\n",chan->name);
LOCAL_USER_REMOVE(u);
return -1;
}
/* Set default values for this invokation of the application */
ast_copy_string(signalling_type, ADEMCO_CONTACT_ID, sizeof(signalling_type));
/* Answer the channel if it is not already */
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Answering channel\n");
if (chan->_state != AST_STATE_UP) {
res = ast_answer(chan);
if (res) {
LOCAL_USER_REMOVE(u);
return -1;
}
}
/* Wait for the connection to settle post-answer */
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Waiting for connection to stabilize\n");
res = ast_safe_sleep(chan, 1250);
/* Attempt to receive the events */
if(!res){
/* Determine the protocol to receive in advance */
/* Note: Ademco contact is the only one supported at this time */
/* Others may be added later */
if(!strcmp(signalling_type, ADEMCO_CONTACT_ID))
receive_ademco_contact_id(chan, data, fdtimeout, sdtimeout, toneloudness, &event_head);
else
res = -1;
}
/* Events queued by receiver, write them all out here if so configured */
if((!res) && (log_individual_events == 0)){
res = log_events(chan, signalling_type, event_head);
}
/*
* Do we exec a command line at the end?
*/
if((!res) && (!ast_strlen_zero(event_app)) && (event_head)){
ast_log(LOG_DEBUG,"Alarmreceiver: executing: %s\n", event_app);
ast_safe_system(event_app);
}
/*
* Free up the data allocated in our linked list
*/
for(elp = event_head; (elp != NULL);){
efree = elp;
elp = elp->next;
free(efree);
}
LOCAL_USER_REMOVE(u);
return 0;
}
/*
* Load the configuration from the configuration file
*/
static int load_config(void)
{
struct ast_config *cfg;
char *p;
/* Read in the config file */
cfg = ast_config_load(ALMRCV_CONFIG);
if(!cfg){
if(option_verbose >= 4)
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: No config file\n");
}
else{
p = ast_variable_retrieve(cfg, "general", "eventcmd");
if(p){
ast_copy_string(event_app, p, sizeof(event_app));
event_app[sizeof(event_app) - 1] = '\0';
}
p = ast_variable_retrieve(cfg, "general", "loudness");
if(p){
toneloudness = atoi(p);
if(toneloudness < 100)
toneloudness = 100;
if(toneloudness > 8192)
toneloudness = 8192;
}
p = ast_variable_retrieve(cfg, "general", "fdtimeout");
if(p){
fdtimeout = atoi(p);
if(fdtimeout < 1000)
fdtimeout = 1000;
if(fdtimeout > 10000)
fdtimeout = 10000;
}
p = ast_variable_retrieve(cfg, "general", "sdtimeout");
if(p){
sdtimeout = atoi(p);
if(sdtimeout < 110)
sdtimeout = 110;
if(sdtimeout > 4000)
sdtimeout = 4000;
}
p = ast_variable_retrieve(cfg, "general", "logindividualevents");
if(p){
log_individual_events = ast_true(p);
}
p = ast_variable_retrieve(cfg, "general", "eventspooldir");
if(p){
ast_copy_string(event_spool_dir, p, sizeof(event_spool_dir));
event_spool_dir[sizeof(event_spool_dir) - 1] = '\0';
}
p = ast_variable_retrieve(cfg, "general", "timestampformat");
if(p){
ast_copy_string(time_stamp_format, p, sizeof(time_stamp_format));
time_stamp_format[sizeof(time_stamp_format) - 1] = '\0';
}
p = ast_variable_retrieve(cfg, "general", "db-family");
if(p){
ast_copy_string(db_family, p, sizeof(db_family));
db_family[sizeof(db_family) - 1] = '\0';
}
ast_config_destroy(cfg);
}
return 0;
}
/*
* These functions are required to implement an Asterisk App.
*/
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
load_config();
return ast_register_application(app, alarmreceiver_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

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@@ -1,232 +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 arbitrary authenticate commands
*
* \ingroup applications
*/
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <errno.h>
#include <stdio.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/app.h"
#include "asterisk/astdb.h"
#include "asterisk/utils.h"
static char *tdesc = "Authentication Application";
static char *app = "Authenticate";
static char *synopsis = "Authenticate a user";
static char *descrip =
" Authenticate(password[|options]): This application asks the caller to enter a\n"
"given password in order to continue dialplan execution. If the password begins\n"
"with the '/' character, it is interpreted as a file which contains a list of\n"
"valid passwords, listed 1 password per line in the file.\n"
" When using a database key, the value associated with the key can be anything.\n"
"Users have three attempts to authenticate before the channel is hung up. If the\n"
"passsword is invalid, the 'j' option is specified, and priority n+101 exists,\n"
"dialplan execution will continnue at this location.\n"
" Options:\n"
" a - Set the channels' account code to the password that is entered\n"
" d - Interpret the given path as database key, not a literal file\n"
" j - Support jumping to n+101 if authentication fails\n"
" m - Interpret the given path as a file which contains a list of account\n"
" codes and password hashes delimited with ':', listed one per line in\n"
" the file. When one of the passwords is matched, the channel will have\n"
" its account code set to the corresponding account code in the file.\n"
" r - Remove the database key upon successful entry (valid with 'd' only)\n"
;
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int auth_exec(struct ast_channel *chan, void *data)
{
int res=0;
int jump = 0;
int retries;
struct localuser *u;
char password[256]="";
char passwd[256];
char *opts;
char *prompt;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Authenticate requires an argument(password)\n");
return -1;
}
LOCAL_USER_ADD(u);
if (chan->_state != AST_STATE_UP) {
res = ast_answer(chan);
if (res) {
LOCAL_USER_REMOVE(u);
return -1;
}
}
strncpy(password, data, sizeof(password) - 1);
opts=strchr(password, '|');
if (opts) {
*opts = 0;
opts++;
} else
opts = "";
if (strchr(opts, 'j'))
jump = 1;
/* Start asking for password */
prompt = "agent-pass";
for (retries = 0; retries < 3; retries++) {
res = ast_app_getdata(chan, prompt, passwd, sizeof(passwd) - 2, 0);
if (res < 0)
break;
res = 0;
if (password[0] == '/') {
if (strchr(opts, 'd')) {
char tmp[256];
/* Compare against a database key */
if (!ast_db_get(password + 1, passwd, tmp, sizeof(tmp))) {
/* It's a good password */
if (strchr(opts, 'r')) {
ast_db_del(password + 1, passwd);
}
break;
}
} else {
/* Compare against a file */
FILE *f;
f = fopen(password, "r");
if (f) {
char buf[256] = "";
char md5passwd[33] = "";
char *md5secret = NULL;
while (!feof(f)) {
fgets(buf, sizeof(buf), f);
if (!feof(f) && !ast_strlen_zero(buf)) {
buf[strlen(buf) - 1] = '\0';
if (strchr(opts, 'm')) {
md5secret = strchr(buf, ':');
if (md5secret == NULL)
continue;
*md5secret = '\0';
md5secret++;
ast_md5_hash(md5passwd, passwd);
if (!strcmp(md5passwd, md5secret)) {
if (strchr(opts, 'a'))
ast_cdr_setaccount(chan, buf);
break;
}
} else {
if (!strcmp(passwd, buf)) {
if (strchr(opts, 'a'))
ast_cdr_setaccount(chan, buf);
break;
}
}
}
}
fclose(f);
if (!ast_strlen_zero(buf)) {
if (strchr(opts, 'm')) {
if (md5secret && !strcmp(md5passwd, md5secret))
break;
} else {
if (!strcmp(passwd, buf))
break;
}
}
} else
ast_log(LOG_WARNING, "Unable to open file '%s' for authentication: %s\n", password, strerror(errno));
}
} else {
/* Compare against a fixed password */
if (!strcmp(passwd, password))
break;
}
prompt="auth-incorrect";
}
if ((retries < 3) && !res) {
if (strchr(opts, 'a') && !strchr(opts, 'm'))
ast_cdr_setaccount(chan, passwd);
res = ast_streamfile(chan, "auth-thankyou", chan->language);
if (!res)
res = ast_waitstream(chan, "");
} else {
if (jump && ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101) == 0) {
res = 0;
} else {
if (!ast_streamfile(chan, "vm-goodbye", chan->language))
res = ast_waitstream(chan, "");
res = -1;
}
}
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, auth_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

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@@ -1,97 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Martin Pycko <martinp@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
* \brief Applications connected with CDR engine
*
* \ingroup applications
*/
#include <sys/types.h>
#include <stdlib.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
static char *tdesc = "Tell Asterisk to not maintain a CDR for the current call";
static char *nocdr_descrip =
" NoCDR(): This application will tell Asterisk not to maintain a CDR for the\n"
"current call.\n";
static char *nocdr_app = "NoCDR";
static char *nocdr_synopsis = "Tell Asterisk to not maintain a CDR for the current call";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int nocdr_exec(struct ast_channel *chan, void *data)
{
struct localuser *u;
LOCAL_USER_ADD(u);
if (chan->cdr) {
ast_cdr_free(chan->cdr);
chan->cdr = NULL;
}
LOCAL_USER_REMOVE(u);
return 0;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(nocdr_app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(nocdr_app, nocdr_exec, nocdr_synopsis, nocdr_descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

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@@ -1,185 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
* James Golovich <james@gnuinter.net>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
* \brief Check if Channel is Available
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <errno.h>
#include <sys/ioctl.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/app.h"
#include "asterisk/devicestate.h"
#include "asterisk/options.h"
static char *tdesc = "Check channel availability";
static char *app = "ChanIsAvail";
static char *synopsis = "Check channel availability";
static char *descrip =
" ChanIsAvail(Technology/resource[&Technology2/resource2...][|options]): \n"
"This application will check to see if any of the specified channels are\n"
"available. The following variables will be set by this application:\n"
" ${AVAILCHAN} - the name of the available channel, if one exists\n"
" ${AVAILORIGCHAN} - the canonical channel name that was used to create the channel\n"
" ${AVAILSTATUS} - the status code for the available channel\n"
" Options:\n"
" s - Consider the channel unavailable if the channel is in use at all\n"
" j - Support jumping to priority n+101 if no channel is available\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int chanavail_exec(struct ast_channel *chan, void *data)
{
int res=-1, inuse=-1, option_state=0, priority_jump=0;
int status;
struct localuser *u;
char *info, tmp[512], trychan[512], *peers, *tech, *number, *rest, *cur, *options, *stringp;
struct ast_channel *tempchan;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "ChanIsAvail requires an argument (Zap/1&Zap/2)\n");
return -1;
}
LOCAL_USER_ADD(u);
info = ast_strdupa(data);
stringp = info;
strsep(&stringp, "|");
options = strsep(&stringp, "|");
if (options) {
if (strchr(options, 's'))
option_state = 1;
if (strchr(options, 'j'))
priority_jump = 1;
}
peers = info;
if (peers) {
cur = peers;
do {
/* remember where to start next time */
rest = strchr(cur, '&');
if (rest) {
*rest = 0;
rest++;
}
tech = cur;
number = strchr(tech, '/');
if (!number) {
ast_log(LOG_WARNING, "ChanIsAvail argument takes format ([technology]/[device])\n");
LOCAL_USER_REMOVE(u);
return -1;
}
*number = '\0';
number++;
if (option_state) {
/* If the pbx says in use then don't bother trying further.
This is to permit testing if someone's on a call, even if the
channel can permit more calls (ie callwaiting, sip calls, etc). */
snprintf(trychan, sizeof(trychan), "%s/%s",cur,number);
status = inuse = ast_device_state(trychan);
}
if ((inuse <= 1) && (tempchan = ast_request(tech, chan->nativeformats, number, &status))) {
pbx_builtin_setvar_helper(chan, "AVAILCHAN", tempchan->name);
/* Store the originally used channel too */
snprintf(tmp, sizeof(tmp), "%s/%s", tech, number);
pbx_builtin_setvar_helper(chan, "AVAILORIGCHAN", tmp);
snprintf(tmp, sizeof(tmp), "%d", status);
pbx_builtin_setvar_helper(chan, "AVAILSTATUS", tmp);
ast_hangup(tempchan);
tempchan = NULL;
res = 1;
break;
} else {
snprintf(tmp, sizeof(tmp), "%d", status);
pbx_builtin_setvar_helper(chan, "AVAILSTATUS", tmp);
}
cur = rest;
} while (cur);
}
if (res < 1) {
pbx_builtin_setvar_helper(chan, "AVAILCHAN", "");
pbx_builtin_setvar_helper(chan, "AVAILORIGCHAN", "");
if (priority_jump || option_priority_jumping) {
if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
LOCAL_USER_REMOVE(u);
return -1;
}
}
}
LOCAL_USER_REMOVE(u);
return 0;
}
int unload_module(void)
{
int res = 0;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, chanavail_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

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@@ -1,585 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2005 Anthony Minessale II (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 ChanSpy: Listen in on any channel.
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <ctype.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/chanspy.h"
#include "asterisk/features.h"
#include "asterisk/options.h"
#include "asterisk/app.h"
#include "asterisk/utils.h"
#include "asterisk/say.h"
#include "asterisk/pbx.h"
#include "asterisk/translate.h"
#include "asterisk/module.h"
#include "asterisk/lock.h"
AST_MUTEX_DEFINE_STATIC(modlock);
#define AST_NAME_STRLEN 256
#define ALL_DONE(u, ret) LOCAL_USER_REMOVE(u); return ret;
#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
static const char *synopsis = "Listen to the audio of an active channel";
static const char *app = "ChanSpy";
static const char *desc =
" ChanSpy([chanprefix][|options]): This application is used to listen to the\n"
"audio from an active Asterisk channel. This includes the audio coming in and\n"
"out of the channel being spied on. If the 'chanprefix' parameter is specified,\n"
"only channels beginning with this string will be spied upon.\n"
" While Spying, the following actions may be performed:\n"
" - Dialing # cycles the volume level.\n"
" - Dialing * will stop spying and look for another channel to spy on.\n"
" - Dialing a series of digits followed by # builds a channel name to append\n"
" to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing\n"
" the digits '1234#' while spying will begin spying on the channel,\n"
" 'Agent/1234'.\n"
" Options:\n"
" b - Only spy on channels involved in a bridged call.\n"
" g(grp) - Match only channels where their ${SPYGROUP} variable is set to\n"
" 'grp'.\n"
" q - Don't play a beep when beginning to spy on a channel.\n"
" r[(basename)] - Record the session to the monitor spool directory. An\n"
" optional base for the filename may be specified. The\n"
" default is 'chanspy'.\n"
" v([value]) - Adjust the initial volume in the range from -4 to 4. A\n"
" negative value refers to a quieter setting.\n"
;
static const char *chanspy_spy_type = "ChanSpy";
enum {
OPTION_QUIET = (1 << 0), /* Quiet, no announcement */
OPTION_BRIDGED = (1 << 1), /* Only look at bridged calls */
OPTION_VOLUME = (1 << 2), /* Specify initial volume */
OPTION_GROUP = (1 << 3), /* Only look at channels in group */
OPTION_RECORD = (1 << 4), /* Record */
} chanspy_opt_flags;
enum {
OPT_ARG_VOLUME = 0,
OPT_ARG_GROUP,
OPT_ARG_RECORD,
OPT_ARG_ARRAY_SIZE,
} chanspy_opt_args;
AST_APP_OPTIONS(chanspy_opts, {
AST_APP_OPTION('q', OPTION_QUIET),
AST_APP_OPTION('b', OPTION_BRIDGED),
AST_APP_OPTION_ARG('v', OPTION_VOLUME, OPT_ARG_VOLUME),
AST_APP_OPTION_ARG('g', OPTION_GROUP, OPT_ARG_GROUP),
AST_APP_OPTION_ARG('r', OPTION_RECORD, OPT_ARG_RECORD),
});
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
struct chanspy_translation_helper {
/* spy data */
struct ast_channel_spy spy;
int fd;
int volfactor;
};
static struct ast_channel *local_channel_walk(struct ast_channel *chan)
{
struct ast_channel *ret;
ast_mutex_lock(&modlock);
if ((ret = ast_channel_walk_locked(chan))) {
ast_mutex_unlock(&ret->lock);
}
ast_mutex_unlock(&modlock);
return ret;
}
static struct ast_channel *local_get_channel_begin_name(char *name)
{
struct ast_channel *chan, *ret = NULL;
ast_mutex_lock(&modlock);
chan = local_channel_walk(NULL);
while (chan) {
if (!strncmp(chan->name, name, strlen(name)) && strncmp(chan->name, "Zap/pseudo", 10)) {
ret = chan;
break;
}
chan = local_channel_walk(chan);
}
ast_mutex_unlock(&modlock);
return ret;
}
static void *spy_alloc(struct ast_channel *chan, void *data)
{
/* just store the data pointer in the channel structure */
return data;
}
static void spy_release(struct ast_channel *chan, void *data)
{
/* nothing to do */
}
static int spy_generate(struct ast_channel *chan, void *data, int len, int samples)
{
struct chanspy_translation_helper *csth = data;
struct ast_frame *f;
if (csth->spy.status != CHANSPY_RUNNING)
/* Channel is already gone more than likely */
return -1;
ast_mutex_lock(&csth->spy.lock);
f = ast_channel_spy_read_frame(&csth->spy, samples);
ast_mutex_unlock(&csth->spy.lock);
if (!f)
return 0;
if (ast_write(chan, f)) {
ast_frfree(f);
return -1;
}
if (csth->fd)
write(csth->fd, f->data, f->datalen);
ast_frfree(f);
return 0;
}
static struct ast_generator spygen = {
.alloc = spy_alloc,
.release = spy_release,
.generate = spy_generate,
};
static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, struct ast_channel_spy *spy)
{
int res;
struct ast_channel *peer;
ast_log(LOG_NOTICE, "Attaching %s to %s\n", spychan->name, chan->name);
ast_mutex_lock(&chan->lock);
res = ast_channel_spy_add(chan, spy);
ast_mutex_unlock(&chan->lock);
if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan))) {
ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);
}
return res;
}
/* Map 'volume' levels from -4 through +4 into
decibel (dB) settings for channel drivers
*/
static signed char volfactor_map[] = {
-24,
-18,
-12,
-6,
0,
6,
12,
18,
24,
};
/* attempt to set the desired gain adjustment via the channel driver;
if successful, clear it out of the csth structure so the
generator will not attempt to do the adjustment itself
*/
static void set_volume(struct ast_channel *chan, struct chanspy_translation_helper *csth)
{
signed char volume_adjust = volfactor_map[csth->volfactor + 4];
if (!ast_channel_setoption(chan, AST_OPTION_TXGAIN, &volume_adjust, sizeof(volume_adjust), 0))
csth->volfactor = 0;
}
static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int *volfactor, int fd)
{
struct chanspy_translation_helper csth;
int running = 0, res = 0, x = 0;
char inp[24] = "", *name = NULL;
struct ast_frame *f = NULL;
if ((chan && ast_check_hangup(chan)) || (spyee && ast_check_hangup(spyee)))
return 0;
name = ast_strdupa(spyee->name);
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "Spying on channel %s\n", name);
memset(&csth, 0, sizeof(csth));
ast_set_flag(&csth.spy, CHANSPY_FORMAT_AUDIO);
ast_set_flag(&csth.spy, CHANSPY_TRIGGER_NONE);
ast_set_flag(&csth.spy, CHANSPY_MIXAUDIO);
csth.spy.type = chanspy_spy_type;
csth.spy.status = CHANSPY_RUNNING;
csth.spy.read_queue.format = AST_FORMAT_SLINEAR;
csth.spy.write_queue.format = AST_FORMAT_SLINEAR;
ast_mutex_init(&csth.spy.lock);
csth.volfactor = *volfactor;
set_volume(chan, &csth);
if (csth.volfactor) {
ast_set_flag(&csth.spy, CHANSPY_READ_VOLADJUST);
csth.spy.read_vol_adjustment = csth.volfactor;
ast_set_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST);
csth.spy.write_vol_adjustment = csth.volfactor;
}
csth.fd = fd;
if (start_spying(spyee, chan, &csth.spy)) {
ast_channel_spy_free(&csth.spy);
return 0;
}
ast_activate_generator(chan, &spygen, &csth);
while (csth.spy.status == CHANSPY_RUNNING &&
(res = ast_waitfor(chan, -1) > -1)) {
/* Read in frame from channel, break out if no frame */
if (!(f = ast_read(chan)))
break;
/* Now if this is DTMF then we have to handle it as such, otherwise just skip it */
res = 0;
if (f->frametype == AST_FRAME_DTMF)
res = f->subclass;
ast_frfree(f);
if (!res)
continue;
if (x == sizeof(inp))
x = 0;
if (res < 0) {
running = -1;
break;
}
/* Process DTMF digits */
if (res == '#') {
if (!ast_strlen_zero(inp)) {
running = x ? atoi(inp) : -1;
break;
} else {
(*volfactor)++;
if (*volfactor > 4)
*volfactor = -1;
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Setting spy volume on %s to %d\n", chan->name, *volfactor);
csth.volfactor = *volfactor;
set_volume(chan, &csth);
if (csth.volfactor) {
ast_set_flag(&csth.spy, CHANSPY_READ_VOLADJUST);
csth.spy.read_vol_adjustment = csth.volfactor;
ast_set_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST);
csth.spy.write_vol_adjustment = csth.volfactor;
} else {
ast_clear_flag(&csth.spy, CHANSPY_READ_VOLADJUST);
ast_clear_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST);
}
}
} else if (res == '*') {
break;
} else if (res >= 48 && res <= 57) {
inp[x++] = res;
}
}
ast_deactivate_generator(chan);
csth.spy.status = CHANSPY_DONE;
ast_mutex_lock(&csth.spy.lock);
if (csth.spy.chan) {
ast_mutex_lock(&csth.spy.chan->lock);
ast_channel_spy_remove(csth.spy.chan, &csth.spy);
ast_mutex_unlock(&csth.spy.chan->lock);
}
ast_mutex_unlock(&csth.spy.lock);
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "Done Spying on channel %s\n", name);
ast_channel_spy_free(&csth.spy);
return running;
}
static int chanspy_exec(struct ast_channel *chan, void *data)
{
struct localuser *u;
struct ast_channel *peer=NULL, *prev=NULL;
char name[AST_NAME_STRLEN],
peer_name[AST_NAME_STRLEN + 5],
*args,
*ptr = NULL,
*options = NULL,
*spec = NULL,
*argv[5],
*mygroup = NULL,
*recbase = NULL;
int res = -1,
volfactor = 0,
silent = 0,
argc = 0,
bronly = 0,
chosen = 0,
count=0,
waitms = 100,
num = 0,
oldrf = 0,
oldwf = 0,
fd = 0;
struct ast_flags flags;
signed char zero_volume = 0;
if (!(args = ast_strdupa((char *)data))) {
ast_log(LOG_ERROR, "Out of memory!\n");
return -1;
}
LOCAL_USER_ADD(u);
oldrf = chan->readformat;
oldwf = chan->writeformat;
if (ast_set_read_format(chan, AST_FORMAT_SLINEAR) < 0) {
ast_log(LOG_ERROR, "Could Not Set Read Format.\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if (ast_set_write_format(chan, AST_FORMAT_SLINEAR) < 0) {
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
LOCAL_USER_REMOVE(u);
return -1;
}
ast_answer(chan);
ast_set_flag(chan, AST_FLAG_SPYING); /* so nobody can spy on us while we are spying */
if ((argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0])))) {
spec = argv[0];
if ( argc > 1) {
options = argv[1];
}
if (ast_strlen_zero(spec) || !strcmp(spec, "all")) {
spec = NULL;
}
}
if (options) {
char *opts[OPT_ARG_ARRAY_SIZE];
ast_app_parse_options(chanspy_opts, &flags, opts, options);
if (ast_test_flag(&flags, OPTION_GROUP)) {
mygroup = opts[OPT_ARG_GROUP];
}
if (ast_test_flag(&flags, OPTION_RECORD)) {
if (!(recbase = opts[OPT_ARG_RECORD])) {
recbase = "chanspy";
}
}
silent = ast_test_flag(&flags, OPTION_QUIET);
bronly = ast_test_flag(&flags, OPTION_BRIDGED);
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;
}
}
else
ast_clear_flag(&flags, AST_FLAGS_ALL);
if (recbase) {
char filename[512];
snprintf(filename,sizeof(filename),"%s/%s.%d.raw",ast_config_AST_MONITOR_DIR, recbase, (int)time(NULL));
if ((fd = open(filename, O_CREAT | O_WRONLY | O_TRUNC, 0644)) <= 0) {
ast_log(LOG_WARNING, "Cannot open %s for recording\n", filename);
fd = 0;
}
}
for(;;) {
if (!silent) {
res = ast_streamfile(chan, "beep", chan->language);
if (!res)
res = ast_waitstream(chan, "");
if (res < 0) {
ast_clear_flag(chan, AST_FLAG_SPYING);
break;
}
}
count = 0;
res = ast_waitfordigit(chan, waitms);
if (res < 0) {
ast_clear_flag(chan, AST_FLAG_SPYING);
break;
}
peer = local_channel_walk(NULL);
prev=NULL;
while(peer) {
if (peer != chan) {
char *group = NULL;
int igrp = 1;
if (peer == prev && !chosen) {
break;
}
chosen = 0;
group = pbx_builtin_getvar_helper(peer, "SPYGROUP");
if (mygroup) {
if (!group || strcmp(mygroup, group)) {
igrp = 0;
}
}
if (igrp && (!spec || ((strlen(spec) <= strlen(peer->name) &&
!strncasecmp(peer->name, spec, strlen(spec)))))) {
if (peer && (!bronly || ast_bridged_channel(peer)) &&
!ast_check_hangup(peer) && !ast_test_flag(peer, AST_FLAG_SPYING)) {
int x = 0;
strncpy(peer_name, "spy-", 5);
strncpy(peer_name + strlen(peer_name), peer->name, AST_NAME_STRLEN);
ptr = strchr(peer_name, '/');
*ptr = '\0';
ptr++;
for (x = 0 ; x < strlen(peer_name) ; x++) {
if (peer_name[x] == '/') {
break;
}
peer_name[x] = tolower(peer_name[x]);
}
if (!silent) {
if (ast_fileexists(peer_name, NULL, NULL) != -1) {
res = ast_streamfile(chan, peer_name, chan->language);
if (!res)
res = ast_waitstream(chan, "");
if (res)
break;
} else
res = ast_say_character_str(chan, peer_name, "", chan->language);
if ((num=atoi(ptr)))
ast_say_digits(chan, atoi(ptr), "", chan->language);
}
count++;
prev = peer;
res = channel_spy(chan, peer, &volfactor, fd);
if (res == -1) {
break;
} else if (res > 1 && spec) {
snprintf(name, AST_NAME_STRLEN, "%s/%d", spec, res);
if ((peer = local_get_channel_begin_name(name))) {
chosen = 1;
}
continue;
}
}
}
}
if ((peer = local_channel_walk(peer)) == NULL) {
break;
}
}
waitms = count ? 100 : 5000;
}
if (fd > 0) {
close(fd);
}
if (oldrf && ast_set_read_format(chan, oldrf) < 0) {
ast_log(LOG_ERROR, "Could Not Set Read Format.\n");
}
if (oldwf && ast_set_write_format(chan, oldwf) < 0) {
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
}
ast_clear_flag(chan, AST_FLAG_SPYING);
ast_channel_setoption(chan, AST_OPTION_TXGAIN, &zero_volume, sizeof(zero_volume), 0);
ALL_DONE(u, res);
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, chanspy_exec, synopsis, desc);
}
char *description(void)
{
return (char *) synopsis;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,190 +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 Trivial application to control playback of a sound file
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/app.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/utils.h"
#include "asterisk/options.h"
static const char *tdesc = "Control Playback Application";
static const char *app = "ControlPlayback";
static const char *synopsis = "Play a file with fast forward and rewind";
static const char *descrip =
" ControlPlayback(file[|skipms[|ff[|rew[|stop[|pause[|restart|options]]]]]]]):\n"
"This application will play back the given filename. By default, the '*' key\n"
"can be used to rewind, and the '#' key can be used to fast-forward.\n"
"Parameters:\n"
" skipms - This is number of milliseconds to skip when rewinding or\n"
" fast-forwarding.\n"
" ff - Fast-forward when this DTMF digit is received.\n"
" rew - Rewind when this DTMF digit is received.\n"
" stop - Stop playback when this DTMF digit is received.\n"
" pause - Pause playback when this DTMF digit is received.\n"
" restart - Restart playback when this DTMF digit is received.\n"
"Options:\n"
" j - Jump to priority n+101 if the requested file is not found.\n"
"This application sets the following channel variable upon completion:\n"
" CPLAYBACKSTATUS - This variable contains the status of the attempt as a text\n"
" string, one of: SUCCESS | USERSTOPPED | ERROR\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int is_on_phonepad(char key)
{
return key == 35 || key == 42 || (key >= 48 && key <= 57);
}
static int controlplayback_exec(struct ast_channel *chan, void *data)
{
int res = 0, priority_jump = 0;
int skipms = 0;
struct localuser *u;
char *tmp;
int argc;
char *argv[8];
enum arg_ids {
arg_file = 0,
arg_skip = 1,
arg_fwd = 2,
arg_rev = 3,
arg_stop = 4,
arg_pause = 5,
arg_restart = 6,
options = 7,
};
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "ControlPlayback requires an argument (filename)\n");
return -1;
}
LOCAL_USER_ADD(u);
tmp = ast_strdupa(data);
memset(argv, 0, sizeof(argv));
argc = ast_app_separate_args(tmp, '|', argv, sizeof(argv) / sizeof(argv[0]));
if (argc < 1) {
ast_log(LOG_WARNING, "ControlPlayback requires an argument (filename)\n");
LOCAL_USER_REMOVE(u);
return -1;
}
skipms = argv[arg_skip] ? atoi(argv[arg_skip]) : 3000;
if (!skipms)
skipms = 3000;
if (!argv[arg_fwd] || !is_on_phonepad(*argv[arg_fwd]))
argv[arg_fwd] = "#";
if (!argv[arg_rev] || !is_on_phonepad(*argv[arg_rev]))
argv[arg_rev] = "*";
if (argv[arg_stop] && !is_on_phonepad(*argv[arg_stop]))
argv[arg_stop] = NULL;
if (argv[arg_pause] && !is_on_phonepad(*argv[arg_pause]))
argv[arg_pause] = NULL;
if (argv[arg_restart] && !is_on_phonepad(*argv[arg_restart]))
argv[arg_restart] = NULL;
if (argv[options]) {
if (strchr(argv[options], 'j'))
priority_jump = 1;
}
res = ast_control_streamfile(chan, argv[arg_file], argv[arg_fwd], argv[arg_rev], argv[arg_stop], argv[arg_pause], argv[arg_restart], skipms);
/* If we stopped on one of our stop keys, return 0 */
if (argv[arg_stop] && strchr(argv[arg_stop], res)) {
res = 0;
pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "USERSTOPPED");
} else {
if (res < 0) {
if (priority_jump || option_priority_jumping) {
if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
ast_log(LOG_WARNING, "ControlPlayback tried to jump to priority n+101 as requested, but priority didn't exist\n");
}
}
res = 0;
pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "ERROR");
} else
pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "SUCCESS");
}
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, controlplayback_exec, synopsis, descrip);
}
char *description(void)
{
return (char *) tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

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

View File

@@ -1,473 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (c) 2003 Tilghman Lesher. All rights reserved.
*
* Tilghman Lesher <app_cut__v003@the-tilghman.com>
*
* This code is released by the author with no restrictions on usage.
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
*/
/*! \file
* \brief Cut application
*
* \ingroup applications
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/app.h"
/* Maximum length of any variable */
#define MAXRESULT 1024
static char *tdesc = "Cut out information from a string";
static char *app_cut = "Cut";
static char *cut_synopsis = "Splits a variable's contents using the specified delimiter";
static char *cut_descrip =
" Cut(newvar=varname,delimiter,fieldspec): This application will split the\n"
"contents of a variable based on the given delimeter and store the result in\n"
"a new variable.\n"
"Parameters:\n"
" newvar - new variable created from result string\n"
" varname - variable you want cut\n"
" delimiter - defaults to '-'\n"
" fieldspec - number of the field you want (1-based offset)\n"
" may also be specified as a range (with -)\n"
" or group of ranges and fields (with &)\n"
"This application has been deprecated in favor of the CUT function.\n";
static char *app_sort = "Sort";
static char *app_sort_synopsis = "Sorts a list of keywords and values";
static char *app_sort_descrip =
" Sort(newvar=key1:val1[,key2:val2[[...],keyN:valN]]): This application will\n"
"sort the list provided in ascending order. The result will be stored in the\n"
"specified variable name.\n"
" This application has been deprecated in favor of the SORT function.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
struct sortable_keys {
char *key;
float value;
};
static int sort_subroutine(const void *arg1, const void *arg2)
{
const struct sortable_keys *one=arg1, *two=arg2;
if (one->value < two->value) {
return -1;
} else if (one->value == two->value) {
return 0;
} else {
return 1;
}
}
#define ERROR_NOARG (-1)
#define ERROR_NOMEM (-2)
#define ERROR_USAGE (-3)
static int sort_internal(struct ast_channel *chan, char *data, char *buffer, size_t buflen)
{
char *strings, *ptrkey, *ptrvalue;
int count=1, count2, element_count=0;
struct sortable_keys *sortable_keys;
memset(buffer, 0, buflen);
if (!data) {
return ERROR_NOARG;
}
strings = ast_strdupa((char *)data);
if (!strings) {
return ERROR_NOMEM;
}
for (ptrkey = strings; *ptrkey; ptrkey++) {
if (*ptrkey == '|') {
count++;
}
}
sortable_keys = alloca(count * sizeof(struct sortable_keys));
if (!sortable_keys) {
return ERROR_NOMEM;
}
memset(sortable_keys, 0, count * sizeof(struct sortable_keys));
/* Parse each into a struct */
count2 = 0;
while ((ptrkey = strsep(&strings, "|"))) {
ptrvalue = index(ptrkey, ':');
if (!ptrvalue) {
count--;
continue;
}
*ptrvalue = '\0';
ptrvalue++;
sortable_keys[count2].key = ptrkey;
sscanf(ptrvalue, "%30f", &sortable_keys[count2].value);
count2++;
}
/* Sort the structs */
qsort(sortable_keys, count, sizeof(struct sortable_keys), sort_subroutine);
for (count2 = 0; count2 < count; count2++) {
int blen = strlen(buffer);
if (element_count++) {
strncat(buffer + blen, ",", buflen - blen - 1);
blen++;
}
strncat(buffer + blen, sortable_keys[count2].key, buflen - blen - 1);
}
return 0;
}
static int cut_internal(struct ast_channel *chan, char *data, char *buffer, size_t buflen)
{
char *s, *args[3], *varname=NULL, *delimiter=NULL, *field=NULL;
int args_okay = 0;
memset(buffer, 0, buflen);
/* Check and parse arguments */
if (data) {
s = ast_strdupa((char *)data);
if (s) {
ast_app_separate_args(s, '|', args, 3);
varname = args[0];
delimiter = args[1];
field = args[2];
if (field) {
args_okay = 1;
}
} else {
return ERROR_NOMEM;
}
}
if (args_okay) {
char d, ds[2];
char *tmp = alloca(strlen(varname) + 4);
char varvalue[MAXRESULT], *tmp2=varvalue;
if (tmp) {
snprintf(tmp, strlen(varname) + 4, "${%s}", varname);
memset(varvalue, 0, sizeof(varvalue));
} else {
return ERROR_NOMEM;
}
if (delimiter[0])
d = delimiter[0];
else
d = '-';
/* String form of the delimiter, for use with strsep(3) */
snprintf(ds, sizeof(ds), "%c", d);
pbx_substitute_variables_helper(chan, tmp, tmp2, MAXRESULT - 1);
if (tmp2) {
int curfieldnum = 1;
while ((tmp2 != NULL) && (field != NULL)) {
char *nextgroup = strsep(&field, "&");
int num1 = 0, num2 = MAXRESULT;
char trashchar;
if (sscanf(nextgroup, "%30d-%30d", &num1, &num2) == 2) {
/* range with both start and end */
} else if (sscanf(nextgroup, "-%30d", &num2) == 1) {
/* range with end */
num1 = 0;
} else if ((sscanf(nextgroup, "%30d%1c", &num1, &trashchar) == 2) && (trashchar == '-')) {
/* range with start */
num2 = MAXRESULT;
} else if (sscanf(nextgroup, "%30d", &num1) == 1) {
/* single number */
num2 = num1;
} else {
return ERROR_USAGE;
}
/* Get to start, if any */
if (num1 > 0) {
while ((tmp2 != (char *)NULL + 1) && (curfieldnum < num1)) {
tmp2 = index(tmp2, d) + 1;
curfieldnum++;
}
}
/* Most frequent problem is the expectation of reordering fields */
if ((num1 > 0) && (curfieldnum > num1)) {
ast_log(LOG_WARNING, "We're already past the field you wanted?\n");
}
/* Re-null tmp2 if we added 1 to NULL */
if (tmp2 == (char *)NULL + 1)
tmp2 = NULL;
/* Output fields until we either run out of fields or num2 is reached */
while ((tmp2 != NULL) && (curfieldnum <= num2)) {
char *tmp3 = strsep(&tmp2, ds);
int curlen = strlen(buffer);
if (curlen) {
snprintf(buffer + curlen, buflen - curlen, "%c%s", d, tmp3);
} else {
snprintf(buffer, buflen, "%s", tmp3);
}
curfieldnum++;
}
}
}
} else {
return ERROR_NOARG;
}
return 0;
}
static int sort_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
char *varname, *strings, result[512] = "";
static int dep_warning=0;
if (!dep_warning) {
ast_log(LOG_WARNING, "The application Sort is deprecated. Please use the SORT() function instead.\n");
dep_warning=1;
}
if (!data) {
ast_log(LOG_ERROR, "Sort() requires an argument\n");
return 0;
}
LOCAL_USER_ADD(u);
strings = ast_strdupa((char *)data);
if (!strings) {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return 0;
}
varname = strsep(&strings, "=");
switch (sort_internal(chan, strings, result, sizeof(result))) {
case ERROR_NOARG:
ast_log(LOG_ERROR, "Sort() requires an argument\n");
res = 0;
break;
case ERROR_NOMEM:
ast_log(LOG_ERROR, "Out of memory\n");
res = -1;
break;
case 0:
pbx_builtin_setvar_helper(chan, varname, result);
res = 0;
break;
default:
ast_log(LOG_ERROR, "Unknown internal error\n");
res = -1;
}
LOCAL_USER_REMOVE(u);
return res;
}
static int cut_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
char *s, *newvar=NULL, result[512];
static int dep_warning = 0;
LOCAL_USER_ADD(u);
if (!dep_warning) {
ast_log(LOG_WARNING, "The application Cut is deprecated. Please use the CUT() function instead.\n");
dep_warning=1;
}
/* Check and parse arguments */
if (data) {
s = ast_strdupa((char *)data);
if (s) {
newvar = strsep(&s, "=");
} else {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return -1;
}
}
switch (cut_internal(chan, s, result, sizeof(result))) {
case ERROR_NOARG:
ast_log(LOG_ERROR, "Cut() requires an argument\n");
res = 0;
break;
case ERROR_NOMEM:
ast_log(LOG_ERROR, "Out of memory\n");
res = -1;
break;
case ERROR_USAGE:
ast_log(LOG_ERROR, "Usage: %s\n", cut_synopsis);
res = 0;
break;
case 0:
pbx_builtin_setvar_helper(chan, newvar, result);
res = 0;
break;
default:
ast_log(LOG_ERROR, "Unknown internal error\n");
res = -1;
}
LOCAL_USER_REMOVE(u);
return res;
}
static char *acf_sort_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
{
struct localuser *u;
LOCAL_USER_ACF_ADD(u);
switch (sort_internal(chan, data, buf, len)) {
case ERROR_NOARG:
ast_log(LOG_ERROR, "SORT() requires an argument\n");
break;
case ERROR_NOMEM:
ast_log(LOG_ERROR, "Out of memory\n");
break;
case 0:
break;
default:
ast_log(LOG_ERROR, "Unknown internal error\n");
}
LOCAL_USER_REMOVE(u);
return buf;
}
static char *acf_cut_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
{
struct localuser *u;
LOCAL_USER_ACF_ADD(u);
switch (cut_internal(chan, data, buf, len)) {
case ERROR_NOARG:
ast_log(LOG_ERROR, "CUT() requires an argument\n");
break;
case ERROR_NOMEM:
ast_log(LOG_ERROR, "Out of memory\n");
break;
case ERROR_USAGE:
ast_log(LOG_ERROR, "Usage: %s\n", cut_synopsis);
break;
case 0:
break;
default:
ast_log(LOG_ERROR, "Unknown internal error\n");
}
LOCAL_USER_REMOVE(u);
return buf;
}
struct ast_custom_function acf_sort = {
.name = "SORT",
.synopsis = "Sorts a list of key/vals into a list of keys, based upon the vals",
.syntax = "SORT(key1:val1[...][,keyN:valN])",
.desc =
"Takes a comma-separated list of keys and values, each separated by a colon, and returns a\n"
"comma-separated list of the keys, sorted by their values. Values will be evaluated as\n"
"floating-point numbers.\n",
.read = acf_sort_exec,
};
struct ast_custom_function acf_cut = {
.name = "CUT",
.synopsis = "Slices and dices strings, based upon a named delimiter.",
.syntax = "CUT(<varname>,<char-delim>,<range-spec>)",
.desc =
" varname - variable you want cut\n"
" char-delim - defaults to '-'\n"
" range-spec - number of the field you want (1-based offset)\n"
" may also be specified as a range (with -)\n"
" or group of ranges and fields (with &)\n",
.read = acf_cut_exec,
};
int unload_module(void)
{
int res;
res = ast_custom_function_unregister(&acf_cut);
res |= ast_custom_function_unregister(&acf_sort);
res |= ast_unregister_application(app_sort);
res |= ast_unregister_application(app_cut);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
res = ast_custom_function_register(&acf_cut);
res |= ast_custom_function_register(&acf_sort);
res |= ast_register_application(app_sort, sort_exec, app_sort_synopsis, app_sort_descrip);
res |= ast_register_application(app_cut, cut_exec, cut_synopsis, cut_descrip);
return res;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,101 +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 Time of day - Report the time of day
*
* \ingroup applications
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/say.h"
static char *tdesc = "Date and Time";
static char *app = "DateTime";
static char *synopsis = "Say the date and time";
static char *descrip =
" DateTime(): This application will say the current date and time.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int datetime_exec(struct ast_channel *chan, void *data)
{
int res=0;
time_t t;
struct localuser *u;
LOCAL_USER_ADD(u);
time(&t);
if (chan->_state != AST_STATE_UP)
res = ast_answer(chan);
if (!res)
res = ast_say_datetime(chan, t, "", chan->language);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, datetime_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,323 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
* Copyright (C) 2003, Jefferson Noxon
*
* Mark Spencer <markster@digium.com>
* Jefferson Noxon <jeff@debian.org>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Database access functions
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <sys/types.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/options.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/astdb.h"
#include "asterisk/lock.h"
#include "asterisk/options.h"
static char *tdesc = "Database Access Functions";
static char *g_descrip =
" DBget(varname=family/key[|options]): This application will retrieve a value\n"
"from the Asterisk database and store it in the given variable.\n"
" Options:\n"
" j - Jump to priority n+101 if the requested family/key isn't found.\n"
" This application sets the following channel variable upon completion:\n"
" DBGETSTATUS - This variable will contain the status of the attempt\n"
" FOUND | NOTFOUND \n"
" This application has been deprecated in favor of the DB function.\n";
static char *p_descrip =
" DBput(family/key=value): This application will store the given value in the\n"
"specified location in the Asterisk database.\n"
" This application has been deprecated in favor of the DB function.\n";
static char *d_descrip =
" DBdel(family/key): This application will delete a key from the Asterisk\n"
"database.\n";
static char *dt_descrip =
" DBdeltree(family[/keytree]): This application will delete a family or keytree\n"
"from the Asterisk database\n";
static char *g_app = "DBget";
static char *p_app = "DBput";
static char *d_app = "DBdel";
static char *dt_app = "DBdeltree";
static char *g_synopsis = "Retrieve a value from the database";
static char *p_synopsis = "Store a value in the database";
static char *d_synopsis = "Delete a key from the database";
static char *dt_synopsis = "Delete a family or keytree from the database";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int deltree_exec(struct ast_channel *chan, void *data)
{
char *argv, *family, *keytree;
struct localuser *u;
LOCAL_USER_ADD(u);
argv = ast_strdupa(data);
if (!argv) {
ast_log(LOG_ERROR, "Memory allocation failed\n");
LOCAL_USER_REMOVE(u);
return 0;
}
if (strchr(argv, '/')) {
family = strsep(&argv, "/");
keytree = strsep(&argv, "\0");
if (!family || !keytree) {
ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
LOCAL_USER_REMOVE(u);
return 0;
}
if (ast_strlen_zero(keytree))
keytree = 0;
} else {
family = argv;
keytree = 0;
}
if (option_verbose > 2) {
if (keytree)
ast_verbose(VERBOSE_PREFIX_3 "DBdeltree: family=%s, keytree=%s\n", family, keytree);
else
ast_verbose(VERBOSE_PREFIX_3 "DBdeltree: family=%s\n", family);
}
if (ast_db_deltree(family, keytree)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "DBdeltree: Error deleting key from database.\n");
}
LOCAL_USER_REMOVE(u);
return 0;
}
static int del_exec(struct ast_channel *chan, void *data)
{
char *argv, *family, *key;
struct localuser *u;
LOCAL_USER_ADD(u);
argv = ast_strdupa(data);
if (!argv) {
ast_log (LOG_ERROR, "Memory allocation failed\n");
LOCAL_USER_REMOVE(u);
return 0;
}
if (strchr(argv, '/')) {
family = strsep(&argv, "/");
key = strsep(&argv, "\0");
if (!family || !key) {
ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
LOCAL_USER_REMOVE(u);
return 0;
}
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "DBdel: family=%s, key=%s\n", family, key);
if (ast_db_del(family, key)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "DBdel: Error deleting key from database.\n");
}
} else {
ast_log(LOG_DEBUG, "Ignoring, no parameters\n");
}
LOCAL_USER_REMOVE(u);
return 0;
}
static int put_exec(struct ast_channel *chan, void *data)
{
char *argv, *value, *family, *key;
static int dep_warning = 0;
struct localuser *u;
LOCAL_USER_ADD(u);
if (!dep_warning) {
ast_log(LOG_WARNING, "This application has been deprecated, please use the ${DB(family/key)} function instead.\n");
dep_warning = 1;
}
argv = ast_strdupa(data);
if (!argv) {
ast_log(LOG_ERROR, "Memory allocation failed\n");
LOCAL_USER_REMOVE(u);
return 0;
}
if (strchr(argv, '/') && strchr(argv, '=')) {
family = strsep(&argv, "/");
key = strsep(&argv, "=");
value = strsep(&argv, "\0");
if (!value || !family || !key) {
ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
LOCAL_USER_REMOVE(u);
return 0;
}
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "DBput: family=%s, key=%s, value=%s\n", family, key, value);
if (ast_db_put(family, key, value)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "DBput: Error writing value to database.\n");
}
} else {
ast_log (LOG_DEBUG, "Ignoring, no parameters\n");
}
LOCAL_USER_REMOVE(u);
return 0;
}
static int get_exec(struct ast_channel *chan, void *data)
{
char *argv, *varname, *family, *key, *options = NULL;
char dbresult[256];
static int dep_warning = 0;
int priority_jump = 0;
struct localuser *u;
LOCAL_USER_ADD(u);
if (!dep_warning) {
ast_log(LOG_WARNING, "This application has been deprecated, please use the ${DB(family/key)} function instead.\n");
dep_warning = 1;
}
argv = ast_strdupa(data);
if (!argv) {
ast_log(LOG_ERROR, "Memory allocation failed\n");
LOCAL_USER_REMOVE(u);
return 0;
}
if (strchr(argv, '=') && strchr(argv, '/')) {
varname = strsep(&argv, "=");
family = strsep(&argv, "/");
if (strchr((void *)&argv, '|')) {
key = strsep(&argv, "|");
options = strsep(&argv, "\0");
} else
key = strsep(&argv, "\0");
if (!varname || !family || !key) {
ast_log(LOG_DEBUG, "Ignoring; Syntax error in argument\n");
LOCAL_USER_REMOVE(u);
return 0;
}
if (options) {
if (strchr(options, 'j'))
priority_jump = 1;
}
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "DBget: varname=%s, family=%s, key=%s\n", varname, family, key);
if (!ast_db_get(family, key, dbresult, sizeof (dbresult) - 1)) {
pbx_builtin_setvar_helper(chan, varname, dbresult);
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "DBget: set variable %s to %s\n", varname, dbresult);
pbx_builtin_setvar_helper(chan, "DBGETSTATUS", "FOUND");
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "DBget: Value not found in database.\n");
if (priority_jump || option_priority_jumping) {
/* Send the call to n+101 priority, where n is the current priority */
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
}
pbx_builtin_setvar_helper(chan, "DBGETSTATUS", "NOTFOUND");
}
} else {
ast_log(LOG_DEBUG, "Ignoring, no parameters\n");
}
LOCAL_USER_REMOVE(u);
return 0;
}
int unload_module(void)
{
int retval;
retval = ast_unregister_application(dt_app);
retval |= ast_unregister_application(d_app);
retval |= ast_unregister_application(p_app);
retval |= ast_unregister_application(g_app);
STANDARD_HANGUP_LOCALUSERS;
return retval;
}
int load_module(void)
{
int retval;
retval = ast_register_application(g_app, get_exec, g_synopsis, g_descrip);
retval |= ast_register_application(p_app, put_exec, p_synopsis, p_descrip);
retval |= ast_register_application(d_app, del_exec, d_synopsis, d_descrip);
retval |= ast_register_application(dt_app, deltree_exec, dt_synopsis, dt_descrip);
return retval;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

File diff suppressed because it is too large Load Diff

View File

@@ -1,359 +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
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <sys/stat.h> /* for mkdir */
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/say.h"
#include "asterisk/lock.h"
#include "asterisk/app.h"
static char *tdesc = "Virtual Dictation Machine";
static char *app = "Dictate";
static char *synopsis = "Virtual Dictation Machine";
static char *desc = " Dictate([<base_dir>])\n"
"Start dictation machine using optional base dir for files.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
typedef enum {
DFLAG_RECORD = (1 << 0),
DFLAG_PLAY = (1 << 1),
DFLAG_TRUNC = (1 << 2),
DFLAG_PAUSE = (1 << 3),
} dflags;
typedef enum {
DMODE_INIT,
DMODE_RECORD,
DMODE_PLAY
} dmodes;
#define ast_toggle_flag(it,flag) if(ast_test_flag(it, flag)) ast_clear_flag(it, flag); else ast_set_flag(it, flag)
static int play_and_wait(struct ast_channel *chan, char *file, char *digits)
{
int res = -1;
if (!ast_streamfile(chan, file, chan->language)) {
res = ast_waitstream(chan, digits);
}
return res;
}
static int dictate_exec(struct ast_channel *chan, void *data)
{
char *mydata, *argv[2], *path = NULL, filein[256];
char dftbase[256];
char *base;
struct ast_flags flags = {0};
struct ast_filestream *fs;
struct ast_frame *f = NULL;
struct localuser *u;
int ffactor = 320 * 80,
res = 0,
argc = 0,
done = 0,
oldr = 0,
lastop = 0,
samples = 0,
speed = 1,
digit = 0,
len = 0,
maxlen = 0,
mode = 0;
LOCAL_USER_ADD(u);
snprintf(dftbase, sizeof(dftbase), "%s/dictate", ast_config_AST_SPOOL_DIR);
if (!ast_strlen_zero(data) && (mydata = ast_strdupa(data))) {
argc = ast_app_separate_args(mydata, '|', argv, sizeof(argv) / sizeof(argv[0]));
}
if (argc) {
base = argv[0];
} else {
base = dftbase;
}
oldr = chan->readformat;
if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR)) < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode.\n");
LOCAL_USER_REMOVE(u);
return -1;
}
ast_answer(chan);
ast_safe_sleep(chan, 200);
for(res = 0; !res;) {
if (ast_app_getdata(chan, "dictate/enter_filename", filein, sizeof(filein), 0) ||
ast_strlen_zero(filein)) {
res = -1;
break;
}
mkdir(base, 0755);
len = strlen(base) + strlen(filein) + 2;
if (!path || len > maxlen) {
path = alloca(len);
memset(path, 0, len);
maxlen = len;
} else {
memset(path, 0, maxlen);
}
snprintf(path, len, "%s/%s", base, filein);
fs = ast_writefile(path, "raw", NULL, O_CREAT|O_APPEND, 0, 0700);
mode = DMODE_PLAY;
memset(&flags, 0, sizeof(flags));
ast_set_flag(&flags, DFLAG_PAUSE);
digit = play_and_wait(chan, "dictate/forhelp", AST_DIGIT_ANY);
done = 0;
speed = 1;
res = 0;
lastop = 0;
samples = 0;
while (!done && ((res = ast_waitfor(chan, -1)) > -1) && fs && (f = ast_read(chan))) {
if (digit) {
struct ast_frame fr = {AST_FRAME_DTMF, digit};
ast_queue_frame(chan, &fr);
digit = 0;
}
if ((f->frametype == AST_FRAME_DTMF)) {
int got = 1;
switch(mode) {
case DMODE_PLAY:
switch(f->subclass) {
case '1':
ast_set_flag(&flags, DFLAG_PAUSE);
mode = DMODE_RECORD;
break;
case '2':
speed++;
if (speed > 4) {
speed = 1;
}
res = ast_say_number(chan, speed, AST_DIGIT_ANY, chan->language, (char *) NULL);
break;
case '7':
samples -= ffactor;
if(samples < 0) {
samples = 0;
}
ast_seekstream(fs, samples, SEEK_SET);
break;
case '8':
samples += ffactor;
ast_seekstream(fs, samples, SEEK_SET);
break;
default:
got = 0;
}
break;
case DMODE_RECORD:
switch(f->subclass) {
case '1':
ast_set_flag(&flags, DFLAG_PAUSE);
mode = DMODE_PLAY;
break;
case '8':
ast_toggle_flag(&flags, DFLAG_TRUNC);
lastop = 0;
break;
default:
got = 0;
}
break;
default:
got = 0;
}
if (!got) {
switch(f->subclass) {
case '#':
done = 1;
continue;
break;
case '*':
ast_toggle_flag(&flags, DFLAG_PAUSE);
if (ast_test_flag(&flags, DFLAG_PAUSE)) {
digit = play_and_wait(chan, "dictate/pause", AST_DIGIT_ANY);
} else {
digit = play_and_wait(chan, mode == DMODE_PLAY ? "dictate/playback" : "dictate/record", AST_DIGIT_ANY);
}
break;
case '0':
ast_set_flag(&flags, DFLAG_PAUSE);
digit = play_and_wait(chan, "dictate/paused", AST_DIGIT_ANY);
switch(mode) {
case DMODE_PLAY:
digit = play_and_wait(chan, "dictate/play_help", AST_DIGIT_ANY);
break;
case DMODE_RECORD:
digit = play_and_wait(chan, "dictate/record_help", AST_DIGIT_ANY);
break;
}
if (digit == 0) {
digit = play_and_wait(chan, "dictate/both_help", AST_DIGIT_ANY);
} else if (digit < 0) {
done = 1;
break;
}
break;
}
}
} else if (f->frametype == AST_FRAME_VOICE) {
switch(mode) {
struct ast_frame *fr;
int x;
case DMODE_PLAY:
if (lastop != DMODE_PLAY) {
if (ast_test_flag(&flags, DFLAG_PAUSE)) {
digit = play_and_wait(chan, "dictate/playback_mode", AST_DIGIT_ANY);
if (digit == 0) {
digit = play_and_wait(chan, "dictate/paused", AST_DIGIT_ANY);
} else if (digit < 0) {
break;
}
}
if (lastop != DFLAG_PLAY) {
lastop = DFLAG_PLAY;
ast_closestream(fs);
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, 0700);
if (ast_test_flag(&flags, DFLAG_TRUNC)) {
ast_seekstream(fs, 0, SEEK_SET);
ast_clear_flag(&flags, DFLAG_TRUNC);
} else {
ast_seekstream(fs, 0, SEEK_END);
}
}
if (!ast_test_flag(&flags, DFLAG_PAUSE)) {
res = ast_writestream(fs, f);
}
break;
}
}
ast_frfree(f);
}
}
if (oldr) {
ast_set_read_format(chan, oldr);
}
LOCAL_USER_REMOVE(u);
return 0;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, dictate_exec, synopsis, desc);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,169 +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
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/lock.h"
#include "asterisk/app.h"
static const char *tdesc = "Directed Call Pickup Application";
static const char *app = "Pickup";
static const char *synopsis = "Directed Call Pickup";
static const char *descrip =
" Pickup(extension[@context]): This application can pickup any ringing channel\n"
"that is calling the specified extension. If no context is specified, the current\n"
"context will be used.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int pickup_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u = NULL;
struct ast_channel *origin = NULL, *target = NULL;
char *tmp = NULL, *exten = NULL, *context = NULL;
char workspace[256] = "";
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Pickup requires an argument (extension) !\n");
return -1;
}
LOCAL_USER_ADD(u);
/* Get the extension and context if present */
exten = data;
context = strchr(data, '@');
if (context) {
*context = '\0';
context++;
}
/* Find a channel to pickup */
origin = ast_get_channel_by_exten_locked(exten, context);
if (origin && origin->cdr) {
ast_cdr_getvar(origin->cdr, "dstchannel", &tmp, workspace,
sizeof(workspace), 0);
if (tmp) {
/* We have a possible channel... now we need to find it! */
target = ast_get_channel_by_name_locked(tmp);
} else {
ast_log(LOG_DEBUG, "No target channel found.\n");
res = -1;
}
ast_mutex_unlock(&origin->lock);
} else {
if (origin)
ast_mutex_unlock(&origin->lock);
ast_log(LOG_DEBUG, "No originating channel found.\n");
}
if (res)
goto out;
if (target && (!target->pbx) && ((target->_state == AST_STATE_RINGING) || (target->_state == AST_STATE_RING))) {
ast_log(LOG_DEBUG, "Call pickup on chan '%s' by '%s'\n", target->name,
chan->name);
res = ast_answer(chan);
if (res) {
ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan->name);
res = -1;
goto out;
}
res = ast_queue_control(chan, AST_CONTROL_ANSWER);
if (res) {
ast_log(LOG_WARNING, "Unable to queue answer on '%s'\n",
chan->name);
res = -1;
goto out;
}
res = ast_channel_masquerade(target, chan);
if (res) {
ast_log(LOG_WARNING, "Unable to masquerade '%s' into '%s'\n", chan->name, target->name);
res = -1;
goto out;
}
} else {
ast_log(LOG_DEBUG, "No call pickup possible...\n");
res = -1;
}
/* Done */
out:
if (target)
ast_mutex_unlock(&target->lock);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, pickup_exec, synopsis, descrip);
}
char *description(void)
{
return (char *) tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,84 +1,37 @@
/*
* Asterisk -- An open source telephony toolkit.
* Asterisk -- A telephony toolkit for Linux.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
* Provide a directory of extensions
*
* 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 Provide a directory of extensions
*
* \ingroup applications
* the GNU General Public License
*/
#include <asterisk/file.h>
#include <asterisk/logger.h>
#include <asterisk/channel.h>
#include <asterisk/pbx.h>
#include <asterisk/module.h>
#include <asterisk/config.h>
#include <asterisk/say.h>
#include <string.h>
#include <ctype.h>
#include <stdlib.h>
#include <pthread.h>
#include <stdio.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/config.h"
#include "asterisk/say.h"
#include "asterisk/utils.h"
#ifdef USE_ODBC_STORAGE
#include <errno.h>
#include <sys/mman.h>
#include "asterisk/res_odbc.h"
static char odbc_database[80] = "asterisk";
static char odbc_table[80] = "voicemessages";
static char vmfmts[80] = "wav";
#endif
#include "../asterisk.h"
static char *tdesc = "Extension Directory";
static char *app = "Directory";
static char *synopsis = "Provide directory of voicemail extensions";
static char *descrip =
" Directory(vm-context[|dial-context[|options]]): This application will present\n"
"the calling channel with a directory of extensions from which they can search\n"
"by name. The list of names and corresponding extensions is retrieved from the\n"
"voicemail configuration file, voicemail.conf.\n"
" This application will immediately exit if one of the following DTMF digits are\n"
"received and the extension to jump to exists:\n"
" 0 - Jump to the 'o' extension, if it exists.\n"
" * - Jump to the 'a' extension, if it exists.\n\n"
" Parameters:\n"
" vm-context - This is the context within voicemail.conf to use for the\n"
" Directory.\n"
" dial-context - This is the dialplan context to use when looking for an\n"
" extension that the user has selected, or when jumping to the\n"
" 'o' or 'a' extension.\n\n"
" Options:\n"
" f - Allow the caller to enter the first name of a user in the directory\n"
" instead of using the last name.\n";
/* For simplicity, I'm keeping the format compatible with the voicemail config,
but i'm open to suggestions for isolating it */
#define VOICEMAIL_CONFIG "voicemail.conf"
#define DIRECTORY_CONFIG "voicemail.conf"
/* How many digits to read in */
#define NUMDIGITS 3
@@ -87,101 +40,6 @@ STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
#ifdef USE_ODBC_STORAGE
static void retrieve_file(char *dir)
{
int x = 0;
int res;
int fd=-1;
size_t fdlen = 0;
void *fdm = MAP_FAILED;
SQLHSTMT stmt;
char sql[256];
char fmt[80]="", empty[10] = "";
char *c;
SQLLEN colsize;
char full_fn[256];
odbc_obj *obj;
obj = fetch_odbc_obj(odbc_database, 0);
if (obj) {
do {
ast_copy_string(fmt, vmfmts, sizeof(fmt));
c = strchr(fmt, '|');
if (c)
*c = '\0';
if (!strcasecmp(fmt, "wav49"))
strcpy(fmt, "WAV");
snprintf(full_fn, sizeof(full_fn), "%s.%s", dir, fmt);
res = SQLAllocHandle(SQL_HANDLE_STMT, obj->con, &stmt);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Alloc Handle failed!\n");
break;
}
snprintf(sql, sizeof(sql), "SELECT recording FROM %s WHERE dir=? AND msgnum=-1", odbc_table);
res = SQLPrepare(stmt, (unsigned char *)sql, SQL_NTS);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Prepare failed![%s]\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
}
SQLBindParameter(stmt, 1, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(dir), 0, (void *)dir, 0, NULL);
res = odbc_smart_execute(obj, stmt);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Execute error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
}
res = SQLFetch(stmt);
if (res == SQL_NO_DATA) {
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
} else if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Fetch error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
}
fd = open(full_fn, O_RDWR | O_CREAT | O_TRUNC, 0770);
if (fd < 0) {
ast_log(LOG_WARNING, "Failed to write '%s': %s\n", full_fn, strerror(errno));
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
}
res = SQLGetData(stmt, 1, SQL_BINARY, empty, 0, &colsize);
fdlen = colsize;
if (fd > -1) {
char tmp[1]="";
lseek(fd, fdlen - 1, SEEK_SET);
if (write(fd, tmp, 1) != 1) {
close(fd);
fd = -1;
break;
}
if (fd > -1)
fdm = mmap(NULL, fdlen, PROT_READ | PROT_WRITE, MAP_SHARED, fd, 0);
}
if (fdm != MAP_FAILED) {
memset(fdm, 0, fdlen);
res = SQLGetData(stmt, x + 1, SQL_BINARY, fdm, fdlen, &colsize);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Get Data error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
}
}
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
} while (0);
} else
ast_log(LOG_WARNING, "Failed to obtain database object for '%s'!\n", odbc_database);
if (fdm != MAP_FAILED)
munmap(fdm, fdlen);
if (fd > -1)
close(fd);
return;
}
#endif
static char *convert(char *lastname)
{
char *tmp;
@@ -243,6 +101,7 @@ static char *convert(char *lastname)
case 'Z':
tmp[lcount++] = '9';
break;
default:
}
lastname++;
}
@@ -251,206 +110,18 @@ static char *convert(char *lastname)
return tmp;
}
/* play name of mailbox owner.
* returns: -1 for bad or missing extension
* '1' for selected entry from directory
* '*' for skipped entry from directory
*/
static int play_mailbox_owner(struct ast_channel *chan, char *context, char *dialcontext, char *ext, char *name, int fromappvm) {
int res = 0;
int loop = 3;
char fn[256];
char fn2[256];
/* Check for the VoiceMail2 greeting first */
snprintf(fn, sizeof(fn), "%s/voicemail/%s/%s/greet",
(char *)ast_config_AST_SPOOL_DIR, context, ext);
#ifdef USE_ODBC_STORAGE
retrieve_file(fn);
#endif
/* Otherwise, check for an old-style Voicemail greeting */
snprintf(fn2, sizeof(fn2), "%s/vm/%s/greet",
(char *)ast_config_AST_SPOOL_DIR, ext);
#ifdef USE_ODBC_STORAGE
retrieve_file(fn2);
#endif
if (ast_fileexists(fn, NULL, chan->language) > 0) {
res = ast_streamfile(chan, fn, chan->language);
if (!res) {
res = ast_waitstream(chan, AST_DIGIT_ANY);
}
ast_stopstream(chan);
} else if (ast_fileexists(fn2, NULL, chan->language) > 0) {
res = ast_streamfile(chan, fn2, chan->language);
if (!res) {
res = ast_waitstream(chan, AST_DIGIT_ANY);
}
ast_stopstream(chan);
} else {
res = ast_say_character_str(chan, !ast_strlen_zero(name) ? name : ext,
AST_DIGIT_ANY, chan->language);
}
#ifdef USE_ODBC_STORAGE
ast_filedelete(fn, NULL);
ast_filedelete(fn2, NULL);
#endif
while (loop) {
if (!res) {
res = ast_streamfile(chan, "dir-instr", chan->language);
}
if (!res) {
res = ast_waitstream(chan, AST_DIGIT_ANY);
}
if (!res) {
res = ast_waitfordigit(chan, 3000);
}
ast_stopstream(chan);
if (res > -1) {
switch (res) {
case '1':
/* Name selected */
loop = 0;
if (fromappvm) {
/* We still want to set the exten */
ast_copy_string(chan->exten, ext, sizeof(chan->exten));
} else {
if (ast_goto_if_exists(chan, dialcontext, ext, 1)) {
ast_log(LOG_WARNING,
"Can't find extension '%s' in context '%s'. "
"Did you pass the wrong context to Directory?\n",
ext, dialcontext);
res = -1;
}
}
break;
case '*':
/* Skip to next match in list */
loop = 0;
break;
default:
/* Not '1', or '*', so decrement number of tries */
res = 0;
loop--;
break;
} /* end switch */
} /* end if */
else {
/* User hungup, so jump out now */
loop = 0;
}
} /* end while */
return(res);
}
static struct ast_config *realtime_directory(char *context)
{
struct ast_config *cfg;
struct ast_config *rtdata;
struct ast_category *cat;
struct ast_variable *var;
char *mailbox;
char *fullname;
char *hidefromdir;
char tmp[100];
/* Load flat file config. */
cfg = ast_config_load(VOICEMAIL_CONFIG);
if (!cfg) {
/* Loading config failed. */
ast_log(LOG_WARNING, "Loading config failed.\n");
return NULL;
}
/* Get realtime entries, categorized by their mailbox number
and present in the requested context */
rtdata = ast_load_realtime_multientry("voicemail", "mailbox LIKE", "%", "context", context, NULL);
/* if there are no results, just return the entries from the config file */
if (!rtdata)
return cfg;
/* Does the context exist within the config file? If not, make one */
cat = ast_category_get(cfg, context);
if (!cat) {
cat = ast_category_new(context);
if (!cat) {
ast_log(LOG_WARNING, "Out of memory\n");
ast_config_destroy(cfg);
return NULL;
}
ast_category_append(cfg, cat);
}
mailbox = ast_category_browse(rtdata, NULL);
while (mailbox) {
fullname = ast_variable_retrieve(rtdata, mailbox, "fullname");
hidefromdir = ast_variable_retrieve(rtdata, mailbox, "hidefromdir");
snprintf(tmp, sizeof(tmp), "no-password,%s,hidefromdir=%s",
fullname ? fullname : "",
hidefromdir ? hidefromdir : "no");
var = ast_variable_new(mailbox, tmp);
if (var)
ast_variable_append(cat, var);
else
ast_log(LOG_WARNING, "Out of memory adding mailbox '%s'\n", mailbox);
mailbox = ast_category_browse(rtdata, mailbox);
}
ast_config_destroy(rtdata);
return cfg;
}
static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *context, char *dialcontext, char digit, int last, int fromappvm)
static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *context, char digit)
{
/* Read in the first three digits.. "digit" is the first digit, already read */
char ext[NUMDIGITS + 1];
char name[80] = "";
struct ast_variable *v;
int res;
int found=0;
int lastuserchoice = 0;
char *start, *pos, *conv,*stringp=NULL;
if (ast_strlen_zero(context)) {
ast_log(LOG_WARNING,
"Directory must be called with an argument "
"(context in which to interpret extensions)\n");
return -1;
}
if (digit == '0') {
if (!ast_goto_if_exists(chan, dialcontext, "o", 1) ||
(!ast_strlen_zero(chan->macrocontext) &&
!ast_goto_if_exists(chan, chan->macrocontext, "o", 1))) {
return 0;
} else {
ast_log(LOG_WARNING, "Can't find extension 'o' in current context. "
"Not Exiting the Directory!\n");
res = 0;
}
}
if (digit == '*') {
if (!ast_goto_if_exists(chan, dialcontext, "a", 1) ||
(!ast_strlen_zero(chan->macrocontext) &&
!ast_goto_if_exists(chan, chan->macrocontext, "a", 1))) {
return 0;
} else {
ast_log(LOG_WARNING, "Can't find extension 'a' in current context. "
"Not Exiting the Directory!\n");
res = 0;
}
}
char *start, *pos, *conv;
char fn[256];
memset(ext, 0, sizeof(ext));
ext[0] = digit;
res = 0;
if (ast_readstring(chan, ext + 1, NUMDIGITS - 1, 3000, 3000, "#") < 0) res = -1;
res = ast_readstring(chan, ext + 1, NUMDIGITS, 3000, 3000, "#");
if (!res) {
/* Search for all names which start with those digits */
v = ast_variable_browse(cfg, context);
@@ -459,18 +130,16 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
while(v) {
/* Find a candidate extension */
start = strdup(v->value);
if (start && !strcasestr(start, "hidefromdir=yes")) {
stringp=start;
strsep(&stringp, ",");
pos = strsep(&stringp, ",");
if (start) {
strtok(start, ",");
pos = strtok(NULL, ",");
if (pos) {
ast_copy_string(name, pos, sizeof(name));
/* Grab the last name */
if (last && strrchr(pos,' '))
if (strrchr(pos, ' '))
pos = strrchr(pos, ' ') + 1;
conv = convert(pos);
if (conv) {
if (!strncmp(conv, ext, strlen(ext))) {
if (!strcmp(conv, ext)) {
/* Match! */
found++;
free(conv);
@@ -484,46 +153,51 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
}
v = v->next;
}
if (v) {
/* We have a match -- play a greeting if they have it */
res = play_mailbox_owner(chan, context, dialcontext, v->name, name, fromappvm);
switch (res) {
case -1:
/* user pressed '1' but extension does not exist, or
* user hungup
*/
lastuserchoice = 0;
break;
case '1':
/* user pressed '1' and extensions exists;
play_mailbox_owner will already have done
a goto() on the channel
*/
lastuserchoice = res;
break;
case '*':
/* user pressed '*' to skip something found */
lastuserchoice = res;
snprintf(fn, sizeof(fn), "%s/vm/%s/greet", AST_SPOOL_DIR, v->name);
if (ast_fileexists(fn, NULL, chan->language)) {
res = ast_streamfile(chan, fn, chan->language);
if (!res)
res = ast_waitstream(chan, AST_DIGIT_ANY);
ast_stopstream(chan);
} else {
res = ast_say_digit_str(chan, v->name, chan->language);
}
ahem:
if (!res)
res = ast_streamfile(chan, "dir-instr", chan->language);
if (!res)
res = ast_waitstream(chan, AST_DIGIT_ANY);
if (!res)
res = ast_waitfordigit(chan, 3000);
ast_stopstream(chan);
if (res > -1) {
if (res == '1') {
strncpy(chan->exten, v->name, sizeof(chan->exten));
chan->priority = 0;
strncpy(chan->context, context, sizeof(chan->context));
res = 0;
break;
default:
break;
} else if (res == '*') {
res = 0;
v = v->next;
} else {
res = 0;
goto ahem;
}
}
v = v->next;
} else {
if (found)
res = ast_streamfile(chan, "dir-nomore", chan->language);
else
res = ast_streamfile(chan, "dir-nomatch", chan->language);
if (!res)
res = 1;
return res;
}
}
if (lastuserchoice != '1') {
if (found)
res = ast_streamfile(chan, "dir-nomore", chan->language);
else
res = ast_streamfile(chan, "dir-nomatch", chan->language);
if (!res)
res = 1;
return res;
}
return 0;
}
return res;
}
@@ -533,111 +207,48 @@ static int directory_exec(struct ast_channel *chan, void *data)
int res = 0;
struct localuser *u;
struct ast_config *cfg;
int last = 1;
int fromappvm = 0;
char *context, *dialcontext, *dirintro, *options;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Directory requires an argument (context[,dialcontext])\n");
if (!data) {
ast_log(LOG_WARNING, "directory requires an argument (context)\n");
return -1;
}
LOCAL_USER_ADD(u);
context = ast_strdupa(data);
dialcontext = strchr(context, '|');
if (dialcontext) {
*dialcontext = '\0';
dialcontext++;
options = strchr(dialcontext, '|');
if (options) {
*options = '\0';
options++;
if (strchr(options, 'f'))
last = 0;
if (strchr(options, 'v'))
fromappvm = 1;
}
} else
dialcontext = context;
cfg = realtime_directory(context);
cfg = ast_load(DIRECTORY_CONFIG);
if (!cfg) {
LOCAL_USER_REMOVE(u);
ast_log(LOG_WARNING, "Unable to open directory configuration %s\n", DIRECTORY_CONFIG);
return -1;
}
dirintro = ast_variable_retrieve(cfg, context, "directoryintro");
if (ast_strlen_zero(dirintro))
dirintro = ast_variable_retrieve(cfg, "general", "directoryintro");
if (ast_strlen_zero(dirintro)) {
if (last)
dirintro = "dir-intro";
else
dirintro = "dir-intro-fn";
}
if (chan->_state != AST_STATE_UP)
res = ast_answer(chan);
for (;;) {
if (!res)
res = ast_streamfile(chan, dirintro, chan->language);
if (!res)
res = ast_waitstream(chan, AST_DIGIT_ANY);
ast_stopstream(chan);
if (!res)
res = ast_waitfordigit(chan, 5000);
LOCAL_USER_ADD(u);
top:
if (!res)
res = ast_streamfile(chan, "dir-intro", chan->language);
if (!res)
res = ast_waitstream(chan, AST_DIGIT_ANY);
ast_stopstream(chan);
if (!res)
res = ast_waitfordigit(chan, 5000);
if (res > 0) {
res = do_directory(chan, cfg, (char *)data, res);
if (res > 0) {
res = do_directory(chan, cfg, context, dialcontext, res, last, fromappvm);
if (res > 0) {
res = ast_waitstream(chan, AST_DIGIT_ANY);
ast_stopstream(chan);
if (res >= 0) {
continue;
}
res = ast_waitstream(chan, AST_DIGIT_ANY);
ast_stopstream(chan);
if (res >= 0) {
goto top;
}
}
break;
}
ast_config_destroy(cfg);
ast_destroy(cfg);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
return ast_unregister_application(app);
}
int load_module(void)
{
#ifdef USE_ODBC_STORAGE
struct ast_config *cfg = ast_config_load(VOICEMAIL_CONFIG);
char *tmp;
if (cfg) {
if ((tmp = ast_variable_retrieve(cfg, "general", "odbcstorage"))) {
ast_copy_string(odbc_database, tmp, sizeof(odbc_database));
}
if ((tmp = ast_variable_retrieve(cfg, "general", "odbctable"))) {
ast_copy_string(odbc_table, tmp, sizeof(odbc_table));
}
if ((tmp = ast_variable_retrieve(cfg, "general", "format"))) {
ast_copy_string(vmfmts, tmp, sizeof(vmfmts));
}
ast_config_destroy(cfg);
} else
ast_log(LOG_WARNING, "Unable to load " VOICEMAIL_CONFIG " - ODBC defaults will be used\n");
#endif
return ast_register_application(app, directory_exec, synopsis, descrip);
return ast_register_application(app, directory_exec);
}
char *description(void)
@@ -651,8 +262,3 @@ int usecount(void)
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,412 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
*
* 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.
*/
/*! \file
*
* \brief DISA -- Direct Inward System Access Application
*
* \author Jim Dixon <jim@lambdatel.com>
*
* \ingroup applications
*/
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <math.h>
#include <sys/time.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/app.h"
#include "asterisk/indications.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/ulaw.h"
#include "asterisk/callerid.h"
static char *tdesc = "DISA (Direct Inward System Access) Application";
static char *app = "DISA";
static char *synopsis = "DISA (Direct Inward System Access)";
static char *descrip =
"DISA(<numeric passcode>[|<context>]) or disa(<filename>)\n"
"The DISA, Direct Inward System Access, application allows someone from \n"
"outside the telephone switch (PBX) to obtain an \"internal\" system \n"
"dialtone and to place calls from it as if they were placing a call from \n"
"within the switch.\n"
"DISA plays a dialtone. The user enters their numeric passcode, followed by\n"
"the pound sign (#). If the passcode is correct, the user is then given\n"
"system dialtone on which a call may be placed. Obviously, this type\n"
"of access has SERIOUS security implications, and GREAT care must be\n"
"taken NOT to compromise your security.\n\n"
"There is a possibility of accessing DISA without password. Simply\n"
"exchange your password with \"no-password\".\n\n"
" Example: exten => s,1,DISA(no-password|local)\n\n"
"Be aware that using this compromises the security of your PBX.\n\n"
"The arguments to this application (in extensions.conf) allow either\n"
"specification of a single global passcode (that everyone uses), or\n"
"individual passcodes contained in a file. It also allow specification\n"
"of the context on which the user will be dialing. If no context is\n"
"specified, the DISA application defaults the context to \"disa\".\n"
"Presumably a normal system will have a special context set up\n"
"for DISA use with some or a lot of restrictions. \n\n"
"The file that contains the passcodes (if used) allows specification\n"
"of either just a passcode (defaulting to the \"disa\" context, or\n"
"passcode|context on each line of the file. The file may contain blank\n"
"lines, or comments starting with \"#\" or \";\". In addition, the\n"
"above arguments may have |new-callerid-string appended to them, to\n"
"specify a new (different) callerid to be used for this call, for\n"
"example: numeric-passcode|context|\"My Phone\" <(234) 123-4567> or \n"
"full-pathname-of-passcode-file|\"My Phone\" <(234) 123-4567>. Last\n"
"but not least, |mailbox[@context] may be appended, which will cause\n"
"a stutter-dialtone (indication \"dialrecall\") to be used, if the\n"
"specified mailbox contains any new messages, for example:\n"
"numeric-passcode|context||1234 (w/a changing callerid). Note that\n"
"in the case of specifying the numeric-passcode, the context must be\n"
"specified if the callerid is specified also.\n\n"
"If login is successful, the application looks up the dialed number in\n"
"the specified (or default) context, and executes it if found.\n"
"If the user enters an invalid extension and extension \"i\" (invalid) \n"
"exists in the context, it will be used.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static void play_dialtone(struct ast_channel *chan, char *mailbox)
{
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");
if (ts)
ast_playtones_start(chan, 0, ts->data, 0);
else
ast_tonepair_start(chan, 350, 440, 0, 0);
}
static int disa_exec(struct ast_channel *chan, void *data)
{
int i,j,k,x,did_ignore;
int firstdigittimeout = 20000;
int digittimeout = 10000;
struct localuser *u;
char *tmp, exten[AST_MAX_EXTENSION],acctcode[20]="";
char pwline[256];
char ourcidname[256],ourcidnum[256];
struct ast_frame *f;
struct timeval lastdigittime;
int res;
time_t rstart;
FILE *fp;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(passcode);
AST_APP_ARG(context);
AST_APP_ARG(cid);
AST_APP_ARG(mailbox);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "disa requires an argument (passcode/passcode file)\n");
return -1;
}
LOCAL_USER_ADD(u);
if (chan->pbx) {
firstdigittimeout = chan->pbx->rtimeout*1000;
digittimeout = chan->pbx->dtimeout*1000;
}
if (ast_set_write_format(chan,AST_FORMAT_ULAW)) {
ast_log(LOG_WARNING, "Unable to set write format to Mu-law on %s\n",chan->name);
LOCAL_USER_REMOVE(u);
return -1;
}
if (ast_set_read_format(chan,AST_FORMAT_ULAW)) {
ast_log(LOG_WARNING, "Unable to set read format to Mu-law on %s\n",chan->name);
LOCAL_USER_REMOVE(u);
return -1;
}
ast_log(LOG_DEBUG, "Digittimeout: %d\n", digittimeout);
ast_log(LOG_DEBUG, "Responsetimeout: %d\n", firstdigittimeout);
tmp = ast_strdupa(data);
if (!tmp) {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, tmp);
if (ast_strlen_zero(args.context))
args.context = "disa";
if (ast_strlen_zero(args.mailbox))
args.mailbox = "";
ast_log(LOG_DEBUG, "Mailbox: %s\n",args.mailbox);
if (chan->_state != AST_STATE_UP) {
/* answer */
ast_answer(chan);
}
i = k = x = 0; /* k is 0 for pswd entry, 1 for ext entry */
did_ignore = 0;
exten[0] = 0;
acctcode[0] = 0;
/* can we access DISA without password? */
ast_log(LOG_DEBUG, "Context: %s\n",args.context);
if (!strcasecmp(args.passcode, "no-password")) {
k |= 1; /* We have the password */
ast_log(LOG_DEBUG, "DISA no-password login success\n");
}
lastdigittime = ast_tvnow();
play_dialtone(chan, args.mailbox);
for (;;) {
/* if outa time, give em reorder */
if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) >
((k&2) ? digittimeout : firstdigittimeout))
{
ast_log(LOG_DEBUG,"DISA %s entry timeout on chan %s\n",
((k&1) ? "extension" : "password"),chan->name);
break;
}
if ((res = ast_waitfor(chan, -1) < 0)) {
ast_log(LOG_DEBUG, "Waitfor returned %d\n", res);
continue;
}
f = ast_read(chan);
if (f == NULL)
{
LOCAL_USER_REMOVE(u);
return -1;
}
if ((f->frametype == AST_FRAME_CONTROL) &&
(f->subclass == AST_CONTROL_HANGUP))
{
ast_frfree(f);
LOCAL_USER_REMOVE(u);
return -1;
}
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 == 0)
{
k|=2; /* We have the first digit */
ast_playtones_stop(chan);
}
lastdigittime = ast_tvnow();
/* got a DTMF tone */
if (i < AST_MAX_EXTENSION) /* if still valid number of digits */
{
if (!(k&1)) /* if in password state */
{
if (j == '#') /* end of password */
{
/* 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);
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_log(LOG_DEBUG, "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;
}
}
fclose(fp);
}
/* compare the two */
if (strcmp(exten,args.passcode))
{
ast_log(LOG_WARNING,"DISA on chan %s got bad password %s\n",chan->name,exten);
goto reorder;
}
/* password good, set to dial state */
ast_log(LOG_DEBUG,"DISA on chan %s password is good\n",chan->name);
play_dialtone(chan, args.mailbox);
k|=1; /* In number mode */
i = 0; /* re-set buffer pointer */
exten[sizeof(acctcode)] = 0;
ast_copy_string(acctcode, exten, sizeof(acctcode));
exten[0] = 0;
ast_log(LOG_DEBUG,"Successful DISA log-in on chan %s\n",chan->name);
continue;
}
}
exten[i++] = j; /* save digit */
exten[i] = 0;
if (!(k&1)) continue; /* if getting password, continue doing it */
/* if this exists */
if (ast_ignore_pattern(args.context, exten)) {
play_dialtone(chan, "");
did_ignore = 1;
} else
if (did_ignore) {
ast_playtones_stop(chan);
did_ignore = 0;
}
/* if can do some more, do it */
if (!ast_matchmore_extension(chan,args.context,exten,1, chan->cid.cid_num)) {
break;
}
}
}
if (k == 3) {
int recheck = 0;
struct ast_flags flags = { AST_CDR_FLAG_POSTED };
if (!ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
pbx_builtin_setvar_helper(chan, "INVALID_EXTEN", exten);
exten[0] = 'i';
exten[1] = '\0';
recheck = 1;
}
if (!recheck || ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
ast_playtones_stop(chan);
/* We're authenticated and have a target extension */
if (!ast_strlen_zero(args.cid))
{
ast_callerid_split(args.cid, ourcidname, sizeof(ourcidname), ourcidnum, sizeof(ourcidnum));
ast_set_callerid(chan, ourcidnum, ourcidname, ourcidnum);
}
if (!ast_strlen_zero(acctcode))
ast_copy_string(chan->accountcode, acctcode, sizeof(chan->accountcode));
ast_cdr_reset(chan->cdr, &flags);
ast_explicit_goto(chan, args.context, exten, 1);
LOCAL_USER_REMOVE(u);
return 0;
}
}
/* Received invalid, but no "i" extension exists in the given context */
reorder:
ast_indicate(chan,AST_CONTROL_CONGESTION);
/* something is invalid, give em reorder for several seconds */
time(&rstart);
while(time(NULL) < rstart + 10)
{
if (ast_waitfor(chan, -1) < 0)
break;
f = ast_read(chan);
if (!f)
break;
ast_frfree(f);
}
ast_playtones_stop(chan);
LOCAL_USER_REMOVE(u);
return -1;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, disa_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key(void)
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,193 +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
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/options.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/utils.h"
static char *tdesc = "Dump Info About The Calling Channel";
static char *app = "DumpChan";
static char *synopsis = "Dump Info About The Calling Channel";
static char *desc =
" DumpChan([<min_verbose_level>])\n"
"Displays information on channel and listing of all channel\n"
"variables. If min_verbose_level is specified, output is only\n"
"displayed when the verbose level is currently set to that number\n"
"or greater. \n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int ast_serialize_showchan(struct ast_channel *c, char *buf, size_t size)
{
struct timeval now;
long elapsed_seconds=0;
int hour=0, min=0, sec=0;
char cgrp[256];
char pgrp[256];
now = ast_tvnow();
memset(buf,0,size);
if (!c)
return 0;
if (c->cdr) {
elapsed_seconds = now.tv_sec - c->cdr->start.tv_sec;
hour = elapsed_seconds / 3600;
min = (elapsed_seconds % 3600) / 60;
sec = elapsed_seconds % 60;
}
snprintf(buf,size,
"Name= %s\n"
"Type= %s\n"
"UniqueID= %s\n"
"CallerID= %s\n"
"CallerIDName= %s\n"
"DNIDDigits= %s\n"
"State= %s (%d)\n"
"Rings= %d\n"
"NativeFormat= %d\n"
"WriteFormat= %d\n"
"ReadFormat= %d\n"
"1stFileDescriptor= %d\n"
"Framesin= %d %s\n"
"Framesout= %d %s\n"
"TimetoHangup= %ld\n"
"ElapsedTime= %dh%dm%ds\n"
"Context= %s\n"
"Extension= %s\n"
"Priority= %d\n"
"CallGroup= %s\n"
"PickupGroup= %s\n"
"Application= %s\n"
"Data= %s\n"
"Blocking_in= %s\n",
c->name,
c->type,
c->uniqueid,
(c->cid.cid_num ? c->cid.cid_num : "(N/A)"),
(c->cid.cid_name ? c->cid.cid_name : "(N/A)"),
(c->cid.cid_dnid ? c->cid.cid_dnid : "(N/A)" ),
ast_state2str(c->_state),
c->_state,
c->rings,
c->nativeformats,
c->writeformat,
c->readformat,
c->fds[0], c->fin & 0x7fffffff, (c->fin & 0x80000000) ? " (DEBUGGED)" : "",
c->fout & 0x7fffffff, (c->fout & 0x80000000) ? " (DEBUGGED)" : "", (long)c->whentohangup,
hour,
min,
sec,
c->context,
c->exten,
c->priority,
ast_print_group(cgrp, sizeof(cgrp), c->callgroup),
ast_print_group(pgrp, sizeof(pgrp), c->pickupgroup),
( c->appl ? c->appl : "(N/A)" ),
( c-> data ? (!ast_strlen_zero(c->data) ? c->data : "(Empty)") : "(None)"),
(ast_test_flag(c, AST_FLAG_BLOCKING) ? c->blockproc : "(Not Blocking)"));
return 0;
}
static int dumpchan_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
char vars[1024];
char info[1024];
int level = 0;
static char *line = "================================================================================";
LOCAL_USER_ADD(u);
if (!ast_strlen_zero(data)) {
level = atoi(data);
}
pbx_builtin_serialize_variables(chan, vars, sizeof(vars));
ast_serialize_showchan(chan, info, sizeof(info));
if (option_verbose >= level)
ast_verbose("\nDumping Info For Channel: %s:\n%s\nInfo:\n%s\nVariables:\n%s%s\n",chan->name, line, info, vars, line);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, dumpchan_exec, synopsis, desc);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,80 +1,48 @@
/*
* 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.
*/
/*! \file
*
* \brief Echo application -- play back what you hear to evaluate latency
*
* \ingroup applications
* the GNU General Public License
*/
#include <asterisk/file.h>
#include <asterisk/logger.h>
#include <asterisk/channel.h>
#include <asterisk/pbx.h>
#include <asterisk/module.h>
#include <stdlib.h>
#include <stdio.h>
#include <unistd.h>
#include <string.h>
#include <stdlib.h>
#include "asterisk.h"
#include <pthread.h>
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
static char *tdesc = "Simple Echo Application";
static char *app = "Echo";
static char *synopsis = "Echo audio read back to the user";
static char *descrip =
" Echo(): Echo audio read from channel back to the channel. \n"
"User can exit the application by either pressing the '#' key, \n"
"or hanging up.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int echo_exec(struct ast_channel *chan, void *data)
static int skel_exec(struct ast_channel *chan, void *data)
{
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;
while((f = ast_read(chan))) {
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;
@@ -83,7 +51,6 @@ static int echo_exec(struct ast_channel *chan, void *data)
if (ast_write(chan, f))
break;
}
ast_frfree(f);
}
LOCAL_USER_REMOVE(u);
return res;
@@ -91,18 +58,13 @@ static int echo_exec(struct ast_channel *chan, void *data)
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
return ast_unregister_application(app);
}
int load_module(void)
{
return ast_register_application(app, echo_exec, synopsis, descrip);
return ast_register_application(app, skel_exec);
}
char *description(void)
@@ -116,8 +78,3 @@ int usecount(void)
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,270 +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 Enumlookup - lookup entry in ENUM
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <ctype.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/enum.h"
#include "asterisk/utils.h"
#include "asterisk/app.h"
#include "asterisk/options.h"
static char *tdesc = "ENUM Lookup";
static char *app = "EnumLookup";
static char *synopsis = "Lookup number in ENUM";
static char *descrip =
" EnumLookup(exten[|option]): Looks up an extension via ENUM and sets\n"
"the variable 'ENUM'. For VoIP URIs this variable will \n"
"look like 'TECHNOLOGY/URI' with the appropriate technology.\n"
"Currently, the enumservices SIP, H323, IAX, IAX2 and TEL are recognized. \n"
"\nReturns status in the ENUMSTATUS channel variable:\n"
" ERROR Failed to do a lookup\n"
" <tech> Technology of the successful lookup: SIP, H323, IAX, IAX2 or TEL\n"
" BADURI Got URI Asterisk does not understand.\n"
" The option string may contain zero or the following character:\n"
" 'j' -- jump to +101 priority if the lookup isn't successful.\n"
" and jump to +51 priority on a TEL entry.\n";
#define ENUM_CONFIG "enum.conf"
static char h323driver[80] = "";
#define H323DRIVERDEFAULT "H323"
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
/*--- enumlookup_exec: Look up number in ENUM and return result */
static int enumlookup_exec(struct ast_channel *chan, void *data)
{
int res=0,priority_jump=0;
char tech[80];
char dest[80];
char tmp[256];
char *c,*t = NULL;
static int dep_warning=0;
struct localuser *u;
char *parse;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(d);
AST_APP_ARG(o);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "EnumLookup requires an argument (extension)\n");
return -1;
}
if (!dep_warning) {
ast_log(LOG_WARNING, "The application EnumLookup is deprecated. Please use the ENUMLOOKUP() function instead.\n");
dep_warning = 1;
}
LOCAL_USER_ADD(u);
parse = ast_strdupa(data);
if (!parse) {
ast_log(LOG_ERROR, "Out of memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, parse);
tech[0] = '\0';
dest[0] = '\0';
if (args.o) {
if (strchr(args.o, 'j'))
priority_jump = 1;
}
res = ast_get_enum(chan, args.d, dest, sizeof(dest), tech, sizeof(tech), NULL, NULL);
if (!res) { /* Failed to do a lookup */
if (priority_jump || option_priority_jumping) {
/* Look for a "busy" place */
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
}
pbx_builtin_setvar_helper(chan, "ENUMSTATUS", "ERROR");
LOCAL_USER_REMOVE(u);
return 0;
}
pbx_builtin_setvar_helper(chan, "ENUMSTATUS", tech);
/* Parse it out */
if (res > 0) {
if (!strcasecmp(tech, "SIP")) {
c = dest;
if (!strncmp(c, "sip:", 4))
c += 4;
snprintf(tmp, sizeof(tmp), "SIP/%s", c);
pbx_builtin_setvar_helper(chan, "ENUM", tmp);
} else if (!strcasecmp(tech, "h323")) {
c = dest;
if (!strncmp(c, "h323:", 5))
c += 5;
snprintf(tmp, sizeof(tmp), "%s/%s", h323driver, c);
/* do a s!;.*!! on the H323 URI */
t = strchr(c,';');
if (t)
*t = 0;
pbx_builtin_setvar_helper(chan, "ENUM", tmp);
} else if (!strcasecmp(tech, "iax")) {
c = dest;
if (!strncmp(c, "iax:", 4))
c += 4;
snprintf(tmp, sizeof(tmp), "IAX/%s", c);
pbx_builtin_setvar_helper(chan, "ENUM", tmp);
} else if (!strcasecmp(tech, "iax2")) {
c = dest;
if (!strncmp(c, "iax2:", 5))
c += 5;
snprintf(tmp, sizeof(tmp), "IAX2/%s", c);
pbx_builtin_setvar_helper(chan, "ENUM", tmp);
} else if (!strcasecmp(tech, "tel")) {
c = dest;
if (!strncmp(c, "tel:", 4))
c += 4;
if (c[0] != '+') {
ast_log(LOG_NOTICE, "tel: uri must start with a \"+\" (got '%s')\n", c);
res = 0;
} else {
/* now copy over the number, skipping all non-digits and stop at ; or NULL */
t = tmp;
while( *c && (*c != ';') && (t - tmp < (sizeof(tmp) - 1))) {
if (isdigit(*c))
*t++ = *c;
c++;
}
*t = 0;
pbx_builtin_setvar_helper(chan, "ENUM", tmp);
ast_log(LOG_NOTICE, "tel: ENUM set to \"%s\"\n", tmp);
if (priority_jump || option_priority_jumping) {
if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 51))
res = 0;
}
}
} else if (!ast_strlen_zero(tech)) {
ast_log(LOG_NOTICE, "Don't know how to handle technology '%s'\n", tech);
pbx_builtin_setvar_helper(chan, "ENUMSTATUS", "BADURI");
res = 0;
}
}
LOCAL_USER_REMOVE(u);
return 0;
}
/*--- load_config: Load enum.conf and find out how to handle H.323 */
static int load_config(void)
{
struct ast_config *cfg;
char *s;
cfg = ast_config_load(ENUM_CONFIG);
if (cfg) {
if (!(s=ast_variable_retrieve(cfg, "general", "h323driver"))) {
strncpy(h323driver, H323DRIVERDEFAULT, sizeof(h323driver) - 1);
} else {
strncpy(h323driver, s, sizeof(h323driver) - 1);
}
ast_config_destroy(cfg);
return 0;
}
ast_log(LOG_NOTICE, "No ENUM Config file, using defaults\n");
return 0;
}
/*--- unload_module: Unload this application from PBX */
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
/*--- load_module: Load this application into PBX */
int load_module(void)
{
int res;
res = ast_register_application(app, enumlookup_exec, synopsis, descrip);
if (!res)
res = load_config();
return res;
}
/*--- reload: Reload configuration file */
int reload(void)
{
return load_config();
}
/*--- description: Describe module */
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

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

View File

@@ -1,135 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (c) 2004 - 2005, Tilghman Lesher. All rights reserved.
*
* Tilghman Lesher <app_exec__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 Exec application
*
* \author Tilghman Lesher <app_exec__v001@the-tilghman.com>
*
* \ingroup applications
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
/* Maximum length of any variable */
#define MAXRESULT 1024
static char *tdesc = "Executes applications";
static char *app_exec = "Exec";
static char *exec_synopsis = "Executes internal application";
static char *exec_descrip =
"Usage: Exec(appname(arguments))\n"
" Allows an arbitrary application to be invoked even when not\n"
"hardcoded into the dialplan. To invoke external applications\n"
"see the application System. Returns whatever value the\n"
"app returns or a non-zero value if the app cannot be found.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int exec_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
char *s, *appname, *endargs, args[MAXRESULT];
struct ast_app *app;
LOCAL_USER_ADD(u);
memset(args, 0, MAXRESULT);
/* Check and parse arguments */
if (data) {
s = ast_strdupa((char *)data);
if (s) {
appname = strsep(&s, "(");
if (s) {
endargs = strrchr(s, ')');
if (endargs)
*endargs = '\0';
pbx_substitute_variables_helper(chan, s, args, MAXRESULT - 1);
}
if (appname) {
app = pbx_findapp(appname);
if (app) {
res = pbx_exec(chan, app, args, 1);
} else {
ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
res = -1;
}
}
} else {
ast_log(LOG_ERROR, "Out of memory\n");
res = -1;
}
}
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app_exec);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app_exec, exec_exec, exec_synopsis, exec_descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,597 +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
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <errno.h>
#include <signal.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/linkedlists.h"
#include "asterisk/app.h"
#include "asterisk/options.h"
static const char *tdesc = "External IVR Interface Application";
static const char *app = "ExternalIVR";
static const char *synopsis = "Interfaces with an external IVR application";
static const char *descrip =
" ExternalIVR(command[|arg[|arg...]]): Forks an process to run the supplied command,\n"
"and starts a generator on the channel. The generator's play list is\n"
"controlled by the external application, which can add and clear entries\n"
"via simple commands issued over its stdout. The external application\n"
"will receive all DTMF events received on the channel, and notification\n"
"if the channel is hung up. The application will not be forcibly terminated\n"
"when the channel is hung up.\n"
"See doc/README.externalivr for a protocol specification.\n";
/* XXX the parser in gcc 2.95 gets confused if you don't put a space between 'name' and the comma */
#define ast_chan_log(level, channel, format, ...) ast_log(level, "%s: " format, channel->name , ## __VA_ARGS__)
struct playlist_entry {
AST_LIST_ENTRY(playlist_entry) list;
char filename[1];
};
struct localuser {
struct ast_channel *chan;
struct localuser *next;
AST_LIST_HEAD(playlist, playlist_entry) playlist;
AST_LIST_HEAD(finishlist, playlist_entry) finishlist;
int abort_current_sound;
int playing_silence;
int option_autoclear;
};
LOCAL_USER_DECL;
struct gen_state {
struct localuser *u;
struct ast_filestream *stream;
struct playlist_entry *current;
int sample_queue;
};
static void send_child_event(FILE *handle, const char event, const char *data,
const struct ast_channel *chan)
{
char tmp[256];
if (!data) {
snprintf(tmp, sizeof(tmp), "%c,%10d", event, (int)time(NULL));
} else {
snprintf(tmp, sizeof(tmp), "%c,%10d,%s", event, (int)time(NULL), data);
}
fprintf(handle, "%s\n", tmp);
ast_chan_log(LOG_DEBUG, chan, "sent '%s'\n", tmp);
}
static void *gen_alloc(struct ast_channel *chan, void *params)
{
struct localuser *u = params;
struct gen_state *state;
state = calloc(1, sizeof(*state));
if (!state)
return NULL;
state->u = u;
return state;
}
static void gen_closestream(struct gen_state *state)
{
if (!state->stream)
return;
ast_closestream(state->stream);
state->u->chan->stream = NULL;
state->stream = NULL;
}
static void gen_release(struct ast_channel *chan, void *data)
{
struct gen_state *state = data;
gen_closestream(state);
free(data);
}
/* caller has the playlist locked */
static int gen_nextfile(struct gen_state *state)
{
struct localuser *u = state->u;
char *file_to_stream;
u->abort_current_sound = 0;
u->playing_silence = 0;
gen_closestream(state);
while (!state->stream) {
state->current = AST_LIST_REMOVE_HEAD(&u->playlist, list);
if (state->current) {
file_to_stream = state->current->filename;
} else {
file_to_stream = "silence/10";
u->playing_silence = 1;
}
if (!(state->stream = ast_openstream_full(u->chan, file_to_stream, u->chan->language, 1))) {
ast_chan_log(LOG_WARNING, u->chan, "File '%s' could not be opened: %s\n", file_to_stream, strerror(errno));
if (!u->playing_silence) {
continue;
} else {
break;
}
}
}
return (!state->stream);
}
static struct ast_frame *gen_readframe(struct gen_state *state)
{
struct ast_frame *f = NULL;
struct localuser *u = state->u;
if (u->abort_current_sound ||
(u->playing_silence && AST_LIST_FIRST(&u->playlist))) {
gen_closestream(state);
AST_LIST_LOCK(&u->playlist);
gen_nextfile(state);
AST_LIST_UNLOCK(&u->playlist);
}
if (!(state->stream && (f = ast_readframe(state->stream)))) {
if (state->current) {
AST_LIST_LOCK(&u->finishlist);
AST_LIST_INSERT_TAIL(&u->finishlist, state->current, list);
AST_LIST_UNLOCK(&u->finishlist);
state->current = NULL;
}
if (!gen_nextfile(state))
f = ast_readframe(state->stream);
}
return f;
}
static int gen_generate(struct ast_channel *chan, void *data, int len, int samples)
{
struct gen_state *state = data;
struct ast_frame *f = NULL;
int res = 0;
state->sample_queue += samples;
while (state->sample_queue > 0) {
if (!(f = gen_readframe(state)))
return -1;
res = ast_write(chan, f);
ast_frfree(f);
if (res < 0) {
ast_chan_log(LOG_WARNING, chan, "Failed to write frame: %s\n", strerror(errno));
return -1;
}
state->sample_queue -= f->samples;
}
return res;
}
static struct ast_generator gen =
{
alloc: gen_alloc,
release: gen_release,
generate: gen_generate,
};
static struct playlist_entry *make_entry(const char *filename)
{
struct playlist_entry *entry;
entry = calloc(1, sizeof(*entry) + strlen(filename) + 10);
if (!entry)
return NULL;
strcpy(entry->filename, filename);
return entry;
}
static int app_exec(struct ast_channel *chan, void *data)
{
struct localuser *u = NULL;
struct playlist_entry *entry;
const char *args = data;
int child_stdin[2] = { 0,0 };
int child_stdout[2] = { 0,0 };
int child_stderr[2] = { 0,0 };
int res = -1;
int gen_active = 0;
int pid;
char *argv[32];
int argc = 1;
char *buf, *command;
FILE *child_commands = NULL;
FILE *child_errors = NULL;
FILE *child_events = NULL;
sigset_t fullset, oldset;
LOCAL_USER_ADD(u);
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
AST_LIST_HEAD_INIT(&u->playlist);
AST_LIST_HEAD_INIT(&u->finishlist);
u->abort_current_sound = 0;
if (ast_strlen_zero(args)) {
ast_log(LOG_WARNING, "ExternalIVR requires a command to execute\n");
goto exit;
}
buf = ast_strdupa(data);
if (!buf) {
ast_log(LOG_ERROR, "Out of memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
argc = ast_app_separate_args(buf, '|', argv, sizeof(argv) / sizeof(argv[0]));
if (pipe(child_stdin)) {
ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child input: %s\n", strerror(errno));
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;
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
if (option_highpriority)
ast_set_priority(0);
dup2(child_stdin[0], STDIN_FILENO);
dup2(child_stdout[1], STDOUT_FILENO);
dup2(child_stderr[1], STDERR_FILENO);
for (i = STDERR_FILENO + 1; i < 1024; i++)
close(i);
execv(argv[0], argv);
fprintf(stderr, "Failed to execute '%s': %s\n", argv[0], strerror(errno));
_exit(1);
} else {
/* parent process */
int child_events_fd = child_stdin[1];
int child_commands_fd = child_stdout[0];
int child_errors_fd = child_stderr[0];
struct ast_frame *f;
int ms;
int exception;
int ready_fd;
int waitfds[2] = { child_errors_fd, child_commands_fd };
struct ast_channel *rchan;
pthread_sigmask(SIG_SETMASK, &oldset, NULL);
close(child_stdin[0]);
child_stdin[0] = 0;
close(child_stdout[1]);
child_stdout[1] = 0;
close(child_stderr[1]);
child_stderr[1] = 0;
if (!(child_events = fdopen(child_events_fd, "w"))) {
ast_chan_log(LOG_WARNING, chan, "Could not open stream for child events\n");
goto exit;
}
if (!(child_commands = fdopen(child_commands_fd, "r"))) {
ast_chan_log(LOG_WARNING, chan, "Could not open stream for child commands\n");
goto exit;
}
if (!(child_errors = fdopen(child_errors_fd, "r"))) {
ast_chan_log(LOG_WARNING, chan, "Could not open stream for child errors\n");
goto exit;
}
setvbuf(child_events, NULL, _IONBF, 0);
setvbuf(child_commands, NULL, _IONBF, 0);
setvbuf(child_errors, NULL, _IONBF, 0);
res = 0;
while (1) {
if (ast_test_flag(chan, AST_FLAG_ZOMBIE)) {
ast_chan_log(LOG_NOTICE, chan, "Is a zombie\n");
res = -1;
break;
}
if (ast_check_hangup(chan)) {
ast_chan_log(LOG_NOTICE, chan, "Got check_hangup\n");
send_child_event(child_events, 'H', NULL, chan);
res = -1;
break;
}
ready_fd = 0;
ms = 100;
errno = 0;
exception = 0;
rchan = ast_waitfor_nandfds(&chan, 1, waitfds, 2, &exception, &ready_fd, &ms);
if (!AST_LIST_EMPTY(&u->finishlist)) {
AST_LIST_LOCK(&u->finishlist);
while ((entry = AST_LIST_REMOVE_HEAD(&u->finishlist, list))) {
send_child_event(child_events, 'F', entry->filename, chan);
free(entry);
}
AST_LIST_UNLOCK(&u->finishlist);
}
if (rchan) {
/* the channel has something */
f = ast_read(chan);
if (!f) {
ast_chan_log(LOG_NOTICE, chan, "Returned no frame\n");
send_child_event(child_events, 'H', NULL, chan);
res = -1;
break;
}
if (f->frametype == AST_FRAME_DTMF) {
send_child_event(child_events, f->subclass, NULL, chan);
if (u->option_autoclear) {
if (!u->abort_current_sound && !u->playing_silence)
send_child_event(child_events, 'T', NULL, chan);
AST_LIST_LOCK(&u->playlist);
while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
send_child_event(child_events, 'D', entry->filename, chan);
free(entry);
}
if (!u->playing_silence)
u->abort_current_sound = 1;
AST_LIST_UNLOCK(&u->playlist);
}
} else if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP)) {
ast_chan_log(LOG_NOTICE, chan, "Got AST_CONTROL_HANGUP\n");
send_child_event(child_events, 'H', NULL, chan);
ast_frfree(f);
res = -1;
break;
}
ast_frfree(f);
} else if (ready_fd == child_commands_fd) {
char input[1024];
if (exception || feof(child_commands)) {
ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
res = -1;
break;
}
if (!fgets(input, sizeof(input), child_commands))
continue;
command = ast_strip(input);
ast_chan_log(LOG_DEBUG, chan, "got command '%s'\n", input);
if (strlen(input) < 4)
continue;
if (input[0] == 'S') {
if (ast_fileexists(&input[2], NULL, u->chan->language) == -1) {
ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
send_child_event(child_events, 'Z', NULL, chan);
strcpy(&input[2], "exception");
}
if (!u->abort_current_sound && !u->playing_silence)
send_child_event(child_events, 'T', NULL, chan);
AST_LIST_LOCK(&u->playlist);
while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
send_child_event(child_events, 'D', entry->filename, chan);
free(entry);
}
if (!u->playing_silence)
u->abort_current_sound = 1;
entry = make_entry(&input[2]);
if (entry)
AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
AST_LIST_UNLOCK(&u->playlist);
} else if (input[0] == 'A') {
if (ast_fileexists(&input[2], NULL, u->chan->language) == -1) {
ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
send_child_event(child_events, 'Z', NULL, chan);
strcpy(&input[2], "exception");
}
entry = make_entry(&input[2]);
if (entry) {
AST_LIST_LOCK(&u->playlist);
AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
AST_LIST_UNLOCK(&u->playlist);
}
} else if (input[0] == 'H') {
ast_chan_log(LOG_NOTICE, chan, "Hanging up: %s\n", &input[2]);
send_child_event(child_events, 'H', NULL, chan);
break;
} else if (input[0] == 'O') {
if (!strcasecmp(&input[2], "autoclear"))
u->option_autoclear = 1;
else if (!strcasecmp(&input[2], "noautoclear"))
u->option_autoclear = 0;
else
ast_chan_log(LOG_WARNING, chan, "Unknown option requested '%s'\n", &input[2]);
}
} else if (ready_fd == child_errors_fd) {
char input[1024];
if (exception || feof(child_errors)) {
ast_chan_log(LOG_WARNING, chan, "Child process went away\n");
res = -1;
break;
}
if (fgets(input, sizeof(input), child_errors)) {
command = ast_strip(input);
ast_chan_log(LOG_NOTICE, chan, "stderr: %s\n", command);
}
} else if ((ready_fd < 0) && ms) {
if (errno == 0 || errno == EINTR)
continue;
ast_chan_log(LOG_WARNING, chan, "Wait failed (%s)\n", strerror(errno));
break;
}
}
}
exit:
if (gen_active)
ast_deactivate_generator(chan);
if (child_events)
fclose(child_events);
if (child_commands)
fclose(child_commands);
if (child_errors)
fclose(child_errors);
if (child_stdin[0])
close(child_stdin[0]);
if (child_stdin[1])
close(child_stdin[1]);
if (child_stdout[0])
close(child_stdout[0]);
if (child_stdout[1])
close(child_stdout[1]);
if (child_stderr[0])
close(child_stderr[0]);
if (child_stderr[1])
close(child_stderr[1]);
while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list)))
free(entry);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, app_exec, synopsis, descrip);
}
char *description(void)
{
return (char *) tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,548 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* 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.
*/
/*! \file
*
* \brief Connect to festival
*
* \ingroup applications
*/
#include <sys/types.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <stdlib.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netdb.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <stdio.h>
#include <signal.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include <ctype.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "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 "asterisk/options.h"
#define FESTIVAL_CONFIG "festival.conf"
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,"
"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;
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 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;
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 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;
}
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;
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;
}
for (x=0;x<256;x++) {
if (x != fd)
close(x);
}
if (option_highpriority)
ast_set_priority(0);
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
/*IAS */
#ifdef __PPC__
for( x=0; x<length; x+=2)
{
c = *(waveform+x+1);
*(waveform+x+1)=*(waveform+x);
*(waveform+x)=c;
}
#endif
write(fd,waveform,length);
close(fd);
exit(0);
}
static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, int length, char *intkeys) {
int res=0;
int fds[2];
int ms = -1;
int pid = -1;
int needed = 0;
int owriteformat;
struct ast_frame *f;
struct myframe {
struct ast_frame f;
char offset[AST_FRIENDLY_OFFSET];
char frdata[2048];
} 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) {
ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
return -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 (;;) {
ms = 1000;
res = ast_waitfor(chan, ms);
if (res < 1) {
res = -1;
break;
}
f = ast_read(chan);
if (!f) {
ast_log(LOG_WARNING, "Null frame == hangup() detected\n");
res = -1;
break;
}
if (f->frametype == AST_FRAME_DTMF) {
ast_log(LOG_DEBUG, "User pressed a key\n");
if (intkeys && strchr(intkeys, f->subclass)) {
res = f->subclass;
ast_frfree(f);
break;
}
}
if (f->frametype == AST_FRAME_VOICE) {
/* Treat as a generator */
needed = f->samples * 2;
if (needed > sizeof(myf.frdata)) {
ast_log(LOG_WARNING, "Only able to deliver %d of %d requested samples\n",
(int)sizeof(myf.frdata) / 2, needed/2);
needed = sizeof(myf.frdata);
}
res = read(fds[0], myf.frdata, needed);
if (res > 0) {
myf.f.frametype = AST_FRAME_VOICE;
myf.f.subclass = AST_FORMAT_SLINEAR;
myf.f.datalen = res;
myf.f.samples = res / 2;
myf.f.mallocd = 0;
myf.f.offset = AST_FRIENDLY_OFFSET;
myf.f.src = __PRETTY_FUNCTION__;
myf.f.data = myf.frdata;
if (ast_write(chan, &myf.f) < 0) {
res = -1;
ast_frfree(f);
break;
}
if (res < needed) { /* last frame */
ast_log(LOG_DEBUG, "Last frame\n");
res=0;
ast_frfree(f);
break;
}
} else {
ast_log(LOG_DEBUG, "No more waveform\n");
res = 0;
}
}
ast_frfree(f);
}
}
close(fds[0]);
close(fds[1]);
/* 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 localuser *u;
struct sockaddr_in serv_addr;
struct hostent *serverhost;
struct ast_hostent ahp;
int fd;
FILE *fs;
char *host;
char *cachedir;
char *temp;
char *festivalcommand;
int port=1314;
int n;
char ack[4];
char *waveform;
int filesize;
int wave;
char bigstring[MAXFESTLEN];
int i;
struct MD5Context md5ctx;
unsigned char MD5Res[16];
char MD5Hex[33] = "";
char koko[4] = "";
char cachefile[MAXFESTLEN]="";
int readcache=0;
int writecache=0;
int strln;
int fdesc = -1;
char buffer[16384];
int seekpos = 0;
char *data;
char *intstr;
struct ast_config *cfg;
if (ast_strlen_zero(vdata)) {
ast_log(LOG_WARNING, "festival requires an argument (text)\n");
return -1;
}
LOCAL_USER_ADD(u);
cfg = ast_config_load(FESTIVAL_CONFIG);
if (!cfg) {
ast_log(LOG_WARNING, "No such configuration file %s\n", FESTIVAL_CONFIG);
LOCAL_USER_REMOVE(u);
return -1;
}
if (!(host = ast_variable_retrieve(cfg, "general", "host"))) {
host = "localhost";
}
if (!(temp = ast_variable_retrieve(cfg, "general", "port"))) {
port = 1314;
} else {
port = atoi(temp);
}
if (!(temp = ast_variable_retrieve(cfg, "general", "usecache"))) {
usecache=0;
} else {
usecache = ast_true(temp);
}
if (!(cachedir = ast_variable_retrieve(cfg, "general", "cachedir"))) {
cachedir = "/tmp/";
}
if (!(festivalcommand = ast_variable_retrieve(cfg, "general", "festivalcommand"))) {
festivalcommand = "(tts_textasterisk \"%s\" 'file)(quit)\n";
}
data = ast_strdupa(vdata);
if (!data) {
ast_log(LOG_ERROR, "Out of memery\n");
ast_config_destroy(cfg);
LOCAL_USER_REMOVE(u);
return -1;
}
intstr = strchr(data, '|');
if (intstr) {
*intstr = '\0';
intstr++;
if (!strcasecmp(intstr, "any"))
intstr = AST_DIGIT_ANY;
}
ast_log(LOG_DEBUG, "Text passed to festival server : %s\n",(char *)data);
/* Connect to local festival server */
fd = socket(AF_INET, SOCK_STREAM, IPPROTO_TCP);
if (fd < 0) {
ast_log(LOG_WARNING,"festival_client: can't get socket\n");
ast_config_destroy(cfg);
LOCAL_USER_REMOVE(u);
return -1;
}
memset(&serv_addr, 0, sizeof(serv_addr));
if ((serv_addr.sin_addr.s_addr = inet_addr(host)) == -1) {
/* its a name rather than an ipnum */
serverhost = ast_gethostbyname(host, &ahp);
if (serverhost == (struct hostent *)0) {
ast_log(LOG_WARNING,"festival_client: gethostbyname failed\n");
ast_config_destroy(cfg);
LOCAL_USER_REMOVE(u);
return -1;
}
memmove(&serv_addr.sin_addr,serverhost->h_addr, serverhost->h_length);
}
serv_addr.sin_family = AF_INET;
serv_addr.sin_port = htons(port);
if (connect(fd, (struct sockaddr *)&serv_addr, sizeof(serv_addr)) != 0) {
ast_log(LOG_WARNING,"festival_client: connect to server failed\n");
ast_config_destroy(cfg);
LOCAL_USER_REMOVE(u);
return -1;
}
/* Compute MD5 sum of string */
MD5Init(&md5ctx);
MD5Update(&md5ctx,(unsigned char const *)data,strlen(data));
MD5Final(MD5Res,&md5ctx);
MD5Hex[0] = '\0';
/* Convert to HEX and look if there is any matching file in the cache
directory */
for (i=0;i<16;i++) {
snprintf(koko, sizeof(koko), "%X",MD5Res[i]);
strncat(MD5Hex, koko, sizeof(MD5Hex) - strlen(MD5Hex) - 1);
}
readcache=0;
writecache=0;
if (strlen(cachedir)+strlen(MD5Hex)+1<=MAXFESTLEN && (usecache==-1)) {
snprintf(cachefile, sizeof(cachefile), "%s/%s", cachedir, MD5Hex);
fdesc=open(cachefile,O_RDWR);
if (fdesc==-1) {
fdesc=open(cachefile,O_CREAT|O_RDWR,0777);
if (fdesc!=-1) {
writecache=1;
strln=strlen((char *)data);
ast_log(LOG_DEBUG,"line length : %d\n",strln);
write(fdesc,&strln,sizeof(int));
write(fdesc,data,strln);
seekpos=lseek(fdesc,0,SEEK_CUR);
ast_log(LOG_DEBUG,"Seek position : %d\n",seekpos);
}
} else {
read(fdesc,&strln,sizeof(int));
ast_log(LOG_DEBUG,"Cache file exists, strln=%d, strlen=%d\n",strln,(int)strlen((char *)data));
if (strlen((char *)data)==strln) {
ast_log(LOG_DEBUG,"Size OK\n");
read(fdesc,&bigstring,strln);
bigstring[strln] = 0;
if (strcmp(bigstring,data)==0) {
readcache=1;
} else {
ast_log(LOG_WARNING,"Strings do not match\n");
}
} else {
ast_log(LOG_WARNING,"Size mismatch\n");
}
}
}
if (readcache==1) {
close(fd);
fd=fdesc;
ast_log(LOG_DEBUG,"Reading from cache...\n");
} else {
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_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);
}
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);
LOCAL_USER_REMOVE(u);
return -1;
}
n += read_data;
}
ack[3] = '\0';
if (strcmp(ack,"WV\n") == 0) { /* receive a waveform */
ast_log(LOG_DEBUG,"Festival WV command\n");
waveform = socket_receive_file_to_buff(fd,&filesize);
res = send_waveform_to_channel(chan,waveform,filesize, intstr);
free(waveform);
break;
}
else if (strcmp(ack,"LP\n") == 0) { /* receive an s-expr */
ast_log(LOG_DEBUG,"Festival LP command\n");
waveform = socket_receive_file_to_buff(fd,&filesize);
waveform[filesize]='\0';
ast_log(LOG_WARNING,"Festival returned LP : %s\n",waveform);
free(waveform);
} else if (strcmp(ack,"ER\n") == 0) { /* server got an error */
ast_log(LOG_WARNING,"Festival returned ER\n");
res=-1;
break;
}
} while (strcmp(ack,"OK\n") != 0);
close(fd);
ast_config_destroy(cfg);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, festival_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,141 +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 App to flash a zap trunk
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <sys/ioctl.h>
#ifdef __linux__
#include <linux/zaptel.h>
#else
#include <zaptel.h>
#endif /* __linux__ */
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/options.h"
static char *tdesc = "Flash zap trunk application";
static char *app = "Flash";
static char *synopsis = "Flashes a Zap Trunk";
static char *descrip =
" Flash(): Sends a flash on a zap trunk. This is only a hack for\n"
"people who want to perform transfers and such via AGI and is generally\n"
"quite useless oths application will only work on Zap trunks.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static inline int zt_wait_event(int fd)
{
/* Avoid the silly zt_waitevent which ignores a bunch of events */
int i,j=0;
i = ZT_IOMUX_SIGEVENT;
if (ioctl(fd, ZT_IOMUX, &i) == -1) return -1;
if (ioctl(fd, ZT_GETEVENT, &j) == -1) return -1;
return j;
}
static int flash_exec(struct ast_channel *chan, void *data)
{
int res = -1;
int x;
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, "%s is not an FXO Channel\n", chan->name);
} else
ast_log(LOG_WARNING, "Unable to get parameters of %s: %s\n", chan->name, strerror(errno));
} else
ast_log(LOG_WARNING, "%s is not a Zap channel\n", chan->name);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, flash_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,131 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Anthony Minessale anthmct@yahoo.com
* Development of this app Sponsered/Funded by TAAN Softworks Corp
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Fork CDR application
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/cdr.h"
#include "asterisk/module.h"
static char *tdesc = "Fork The CDR into 2 separate entities.";
static char *app = "ForkCDR";
static char *synopsis =
"Forks the Call Data Record";
static char *descrip =
" ForkCDR([options]): Causes the Call Data Record to fork an additional\n"
"cdr record starting from the time of the fork call\n"
"If the option 'v' is passed all cdr variables will be passed along also.\n"
"";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static void ast_cdr_fork(struct ast_channel *chan)
{
struct ast_cdr *cdr;
struct ast_cdr *newcdr;
struct ast_flags flags = { AST_CDR_FLAG_KEEP_VARS };
cdr = chan->cdr;
while (cdr->next)
cdr = cdr->next;
if (!(newcdr = ast_cdr_dup(cdr)))
return;
ast_cdr_append(cdr, newcdr);
ast_cdr_reset(newcdr, &flags);
if (!ast_test_flag(cdr, AST_CDR_FLAG_KEEP_VARS))
ast_cdr_free_vars(cdr, 0);
ast_set_flag(cdr, AST_CDR_FLAG_CHILD | AST_CDR_FLAG_LOCKED);
}
static int forkcdr_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u;
if (!chan->cdr) {
ast_log(LOG_WARNING, "Channel does not have a CDR\n");
return 0;
}
LOCAL_USER_ADD(u);
if (!ast_strlen_zero(data))
ast_set2_flag(chan->cdr, strchr(data, 'v'), AST_CDR_FLAG_KEEP_VARS);
ast_cdr_fork(chan);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, forkcdr_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,167 +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 Get ADSI CPE ID
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <unistd.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/adsi.h"
#include "asterisk/options.h"
static char *tdesc = "Get ADSI CPE ID";
static char *app = "GetCPEID";
static char *synopsis = "Get ADSI CPE ID";
static char *descrip =
" GetCPEID: Obtains and displays ADSI CPE ID and other information in order\n"
"to properly setup zapata.conf for on-hook operations.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int cpeid_setstatus(struct ast_channel *chan, char *stuff[], int voice)
{
int justify[5] = { ADSI_JUST_CENT, ADSI_JUST_LEFT, ADSI_JUST_LEFT, ADSI_JUST_LEFT };
char *tmp[5];
int x;
for (x=0;x<4;x++)
tmp[x] = stuff[x];
tmp[4] = NULL;
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][80];
char *stuff[4];
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, stuff, 0);
res = adsi_get_cpeid(chan, cpeid, 0);
if (res > 0) {
gotcpeid = 1;
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) {
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) {
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(stuff[1], sizeof(data[1]), "CPEID: %02x:%02x:%02x:%02x", cpeid[0], cpeid[1], cpeid[2], cpeid[3]);
else
strncpy(stuff[1], "CPEID Unknown", sizeof(data[1]) - 1);
if (gotgeometry)
snprintf(stuff[2], sizeof(data[2]), "Geom: %dx%d, %d buttons", width, height, buttons);
else
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)
break;
if (res == '#') {
res = 0;
break;
}
}
adsi_unload_session(chan);
}
}
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, cpeid_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,338 +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 Group Manipulation Applications
*
* \ingroup applications
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <sys/types.h>
#include <regex.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/utils.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int group_count_exec(struct ast_channel *chan, void *data)
{
int res = 0;
int count;
struct localuser *u;
char group[80] = "";
char category[80] = "";
char ret[80] = "";
char *grp;
static int deprecation_warning = 0;
LOCAL_USER_ADD(u);
if (!deprecation_warning) {
ast_log(LOG_WARNING, "The GetGroupCount application has been deprecated, please use the GROUP_COUNT function.\n");
deprecation_warning = 1;
}
ast_app_group_split_group(data, group, sizeof(group), category, sizeof(category));
if (ast_strlen_zero(group)) {
grp = pbx_builtin_getvar_helper(chan, category);
if (!ast_strlen_zero(grp))
ast_copy_string(group, grp, sizeof(group));
}
count = ast_app_group_get_count(group, category);
snprintf(ret, sizeof(ret), "%d", count);
pbx_builtin_setvar_helper(chan, "GROUPCOUNT", ret);
LOCAL_USER_REMOVE(u);
return res;
}
static int group_match_count_exec(struct ast_channel *chan, void *data)
{
int res = 0;
int count;
struct localuser *u;
char group[80] = "";
char category[80] = "";
char ret[80] = "";
static int deprecation_warning = 0;
LOCAL_USER_ADD(u);
if (!deprecation_warning) {
ast_log(LOG_WARNING, "The GetGroupMatchCount application has been deprecated, please use the GROUP_MATCH_COUNT function.\n");
deprecation_warning = 1;
}
ast_app_group_split_group(data, group, sizeof(group), category, sizeof(category));
if (!ast_strlen_zero(group)) {
count = ast_app_group_match_get_count(group, category);
snprintf(ret, sizeof(ret), "%d", count);
pbx_builtin_setvar_helper(chan, "GROUPCOUNT", ret);
}
LOCAL_USER_REMOVE(u);
return res;
}
static int group_set_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u;
static int deprecation_warning = 0;
LOCAL_USER_ADD(u);
if (!deprecation_warning) {
ast_log(LOG_WARNING, "The SetGroup application has been deprecated, please use the GROUP() function.\n");
deprecation_warning = 1;
}
if (ast_app_group_set_channel(chan, data))
ast_log(LOG_WARNING, "SetGroup requires an argument (group name)\n");
LOCAL_USER_REMOVE(u);
return res;
}
static int group_check_exec(struct ast_channel *chan, void *data)
{
int res = 0;
int max, count;
struct localuser *u;
char limit[80]="";
char category[80]="";
static int deprecation_warning = 0;
char *parse;
int priority_jump = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(max);
AST_APP_ARG(options);
);
LOCAL_USER_ADD(u);
if (!deprecation_warning) {
ast_log(LOG_WARNING, "The CheckGroup application has been deprecated, please use a combination of the GotoIf application and the GROUP_COUNT() function.\n");
deprecation_warning = 1;
}
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "CheckGroup requires an argument(max[@category][|options])\n");
return 0;
}
if (!(parse = ast_strdupa(data))) {
ast_log(LOG_WARNING, "Memory Error!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, parse);
if (args.options) {
if (strchr(args.options, 'j'))
priority_jump = 1;
}
if (ast_strlen_zero(args.max)) {
ast_log(LOG_WARNING, "CheckGroup requires an argument(max[@category][|options])\n");
return res;
}
ast_app_group_split_group(args.max, limit, sizeof(limit), category, sizeof(category));
if ((sscanf(limit, "%30d", &max) == 1) && (max > -1)) {
count = ast_app_group_get_count(pbx_builtin_getvar_helper(chan, category), category);
if (count > max) {
pbx_builtin_setvar_helper(chan, "CHECKGROUPSTATUS", "OVERMAX");
if (priority_jump || option_priority_jumping) {
if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
res = -1;
}
} else
pbx_builtin_setvar_helper(chan, "CHECKGROUPSTATUS", "OK");
} else
ast_log(LOG_WARNING, "CheckGroup requires a positive integer argument (max)\n");
LOCAL_USER_REMOVE(u);
return res;
}
static int group_show_channels(int fd, int argc, char *argv[])
{
#define FORMAT_STRING "%-25s %-20s %-20s\n"
int numchans = 0;
struct ast_group_info *gi = NULL;
regex_t regexbuf;
int havepattern = 0;
if (argc < 3 || argc > 4)
return RESULT_SHOWUSAGE;
if (argc == 4) {
if (regcomp(&regexbuf, argv[3], REG_EXTENDED | REG_NOSUB))
return RESULT_SHOWUSAGE;
havepattern = 1;
}
ast_cli(fd, FORMAT_STRING, "Channel", "Group", "Category");
ast_app_group_list_lock();
gi = ast_app_group_list_head();
while (gi) {
if (!havepattern || !regexec(&regexbuf, gi->group, 0, NULL, 0)) {
ast_cli(fd, FORMAT_STRING, gi->chan->name, gi->group, (ast_strlen_zero(gi->category) ? "(default)" : gi->category));
numchans++;
}
gi = AST_LIST_NEXT(gi, list);
}
ast_app_group_list_unlock();
if (havepattern)
regfree(&regexbuf);
ast_cli(fd, "%d active channel%s\n", numchans, (numchans != 1) ? "s" : "");
return RESULT_SUCCESS;
#undef FORMAT_STRING
}
static char *tdesc = "Group Management Routines";
static char *app_group_count = "GetGroupCount";
static char *app_group_set = "SetGroup";
static char *app_group_check = "CheckGroup";
static char *app_group_match_count = "GetGroupMatchCount";
static char *group_count_synopsis = "Get the channel count of a group";
static char *group_set_synopsis = "Set the channel's group";
static char *group_check_synopsis = "Check the channel count of a group against a limit";
static char *group_match_count_synopsis = "Get the channel count of all groups that match a pattern";
static char *group_count_descrip =
"Usage: GetGroupCount([groupname][@category])\n"
" Calculates the group count for the specified group, or uses\n"
"the current channel's group if not specifed (and non-empty).\n"
"Stores result in GROUPCOUNT. \n"
"Note: This application has been deprecated, please use the function\n"
"GROUP_COUNT.\n";
static char *group_set_descrip =
"Usage: SetGroup(groupname[@category])\n"
" Sets the channel group to the specified value. Equivalent to\n"
"Set(GROUP=group). Always returns 0.\n";
static char *group_check_descrip =
"Usage: CheckGroup(max[@category][|options])\n"
" Checks that the current number of total channels in the\n"
"current channel's group does not exceed 'max'. If the number\n"
"does not exceed 'max', we continue to the next step. \n"
" The option string may contain zero of the following character:\n"
" 'j' -- jump to n+101 priority if the number does in fact exceed max,\n"
" and priority n+101 exists. Execuation then continues at that\n"
" step, otherwise -1 is returned.\n"
" This application sets the following channel variable upon successful completion:\n"
" CHECKGROUPSTATUS The status of the check that the current channel's\n"
" group does not exceed 'max'. It's value is one of\n"
" OK | OVERMAX \n";
static char *group_match_count_descrip =
"Usage: GetGroupMatchCount(groupmatch[@category])\n"
" Calculates the group count for all groups that match the specified\n"
"pattern. Uses standard regular expression matching (see regex(7)).\n"
"Stores result in GROUPCOUNT. Always returns 0.\n"
"Note: This application has been deprecated, please use the function\n"
"GROUP_MATCH_COUNT.\n";
static char show_channels_usage[] =
"Usage: group show channels [pattern]\n"
" Lists all currently active channels with channel group(s) specified.\n Optional regular expression pattern is matched to group names for each channel.\n";
static struct ast_cli_entry cli_show_channels =
{ { "group", "show", "channels", NULL }, group_show_channels, "Show active channels with group(s)", show_channels_usage};
int unload_module(void)
{
int res;
res = ast_cli_unregister(&cli_show_channels);
res |= ast_unregister_application(app_group_count);
res |= ast_unregister_application(app_group_set);
res |= ast_unregister_application(app_group_check);
res |= ast_unregister_application(app_group_match_count);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
res = ast_register_application(app_group_count, group_count_exec, group_count_synopsis, group_count_descrip);
res |= ast_register_application(app_group_set, group_set_exec, group_set_synopsis, group_set_descrip);
res |= ast_register_application(app_group_check, group_check_exec, group_check_synopsis, group_check_descrip);
res |= ast_register_application(app_group_match_count, group_match_count_exec, group_match_count_synopsis, group_match_count_descrip);
res |= ast_cli_register(&cli_show_channels);
return res;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,363 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Changes Copyright (c) 2004 - 2005 Todd Freeman <freeman@andrews.edu>
*
* 95% based on HasNewVoicemail by:
*
* Copyright (c) 2003 Tilghman Lesher. All rights reserved.
*
* Tilghman Lesher <asterisk-hasnewvoicemail-app@the-tilghman.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief HasVoicemail application
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <dirent.h>
#include <sys/types.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/lock.h"
#include "asterisk/utils.h"
#include "asterisk/app.h"
#include "asterisk/options.h"
#ifdef USE_ODBC_STORAGE
#include "asterisk/res_odbc.h"
static char odbc_database[80];
static char odbc_table[80];
#endif
static char *tdesc = "Indicator for whether a voice mailbox has messages in a given folder.";
static char *app_hasvoicemail = "HasVoicemail";
static char *hasvoicemail_synopsis = "Conditionally branches to priority + 101 with the right options set";
static char *hasvoicemail_descrip =
"HasVoicemail(vmbox[/folder][@context][|varname[|options]])\n"
" Optionally sets <varname> to the number of messages in that folder."
" Assumes folder of INBOX if not specified.\n"
" The option string may contain zero or the following character:\n"
" 'j' -- jump to priority n+101, if there is voicemail in the folder indicated.\n"
" This application sets the following channel variable upon completion:\n"
" HASVMSTATUS The result of the voicemail check returned as a text string as follows\n"
" <# of messages in the folder, 0 for NONE>\n";
static char *app_hasnewvoicemail = "HasNewVoicemail";
static char *hasnewvoicemail_synopsis = "Conditionally branches to priority + 101 with the right options set";
static char *hasnewvoicemail_descrip =
"HasNewVoicemail(vmbox[/folder][@context][|varname[|options]])\n"
"Assumes folder 'INBOX' if folder is not specified. Optionally sets <varname> to the number of messages\n"
"in that folder.\n"
" The option string may contain zero of the following character:\n"
" 'j' -- jump to priority n+101, if there is new voicemail in folder 'folder' or INBOX\n"
" This application sets the following channel variable upon completion:\n"
" HASVMSTATUS The result of the new voicemail check returned as a text string as follows\n"
" <# of messages in the folder, 0 for NONE>\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
#ifdef USE_ODBC_STORAGE
static int hasvoicemail_internal(const char *context, const char *mailbox, const char *folder)
{
int nummsgs = 0;
int res;
SQLHSTMT stmt;
char sql[256];
char rowdata[20];
if (!folder)
folder = "INBOX";
/* If no mailbox, return immediately */
if (ast_strlen_zero(mailbox))
return 0;
if (ast_strlen_zero(context))
context = "default";
odbc_obj *obj;
obj = fetch_odbc_obj(odbc_database, 0);
if (obj) {
res = SQLAllocHandle(SQL_HANDLE_STMT, obj->con, &stmt);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Alloc Handle failed!\n");
goto yuck;
}
snprintf(sql, sizeof(sql), "SELECT COUNT(*) FROM %s WHERE dir = '%s/voicemail/%s/%s/%s'", odbc_table, ast_config_AST_SPOOL_DIR, context, mailbox, folder);
res = SQLPrepare(stmt, sql, SQL_NTS);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Prepare failed![%s]\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
goto yuck;
}
res = odbc_smart_execute(obj, stmt);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Execute error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
goto yuck;
}
res = SQLFetch(stmt);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Fetch error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
goto yuck;
}
res = SQLGetData(stmt, 1, SQL_CHAR, rowdata, sizeof(rowdata), NULL);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Get Data error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
goto yuck;
}
nummsgs = atoi(rowdata);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
} else
ast_log(LOG_WARNING, "Failed to obtain database object for '%s'!\n", odbc_database);
yuck:
return nummsgs;
}
#else
static int hasvoicemail_internal(const char *context, const char *mailbox, const char *folder)
{
DIR *dir;
struct dirent *de;
char fn[256];
int count = 0;
if (ast_strlen_zero(folder))
folder = "INBOX";
if (ast_strlen_zero(context))
context = "default";
/* If no mailbox, return immediately */
if (ast_strlen_zero(mailbox))
return 0;
snprintf(fn, sizeof(fn), "%s/voicemail/%s/%s/%s", ast_config_AST_SPOOL_DIR, context, mailbox, folder);
dir = opendir(fn);
if (!dir)
return 0;
while ((de = readdir(dir))) {
if (!strncasecmp(de->d_name, "msg", 3) && !strcasecmp(de->d_name + 8, "txt"))
count++;
}
closedir(dir);
return count;
}
#endif
static int hasvoicemail_exec(struct ast_channel *chan, void *data)
{
struct localuser *u;
char *input, *varname = NULL, *vmbox, *context = "default";
char *vmfolder;
int vmcount = 0;
static int dep_warning = 0;
int priority_jump = 0;
char tmp[12];
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(vmbox);
AST_APP_ARG(varname);
AST_APP_ARG(options);
);
if (!dep_warning) {
ast_log(LOG_WARNING, "The applications HasVoicemail and HasNewVoicemail have been deprecated. Please use the VMCOUNT() function instead.\n");
dep_warning = 1;
}
if (!data) {
ast_log(LOG_WARNING, "HasVoicemail requires an argument (vm-box[/folder][@context][|varname[|options]])\n");
return -1;
}
LOCAL_USER_ADD(u);
input = ast_strdupa((char *)data);
if (! input) {
ast_log(LOG_ERROR, "Out of memory error\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, input);
if ((vmbox = strsep(&args.vmbox, "@")))
if (!ast_strlen_zero(args.vmbox))
context = args.vmbox;
if (!vmbox)
vmbox = args.vmbox;
vmfolder = strchr(vmbox, '/');
if (vmfolder) {
*vmfolder = '\0';
vmfolder++;
} else {
vmfolder = "INBOX";
}
if (args.options) {
if (strchr(args.options, 'j'))
priority_jump = 1;
}
vmcount = hasvoicemail_internal(context, vmbox, vmfolder);
/* Set the count in the channel variable */
if (varname) {
snprintf(tmp, sizeof(tmp), "%d", vmcount);
pbx_builtin_setvar_helper(chan, varname, tmp);
}
if (vmcount > 0) {
/* Branch to the next extension */
if (priority_jump || option_priority_jumping) {
if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
ast_log(LOG_WARNING, "VM box %s@%s has new voicemail, but extension %s, priority %d doesn't exist\n", vmbox, context, chan->exten, chan->priority + 101);
}
}
snprintf(tmp, sizeof(tmp), "%d", vmcount);
pbx_builtin_setvar_helper(chan, "HASVMSTATUS", tmp);
LOCAL_USER_REMOVE(u);
return 0;
}
static char *acf_vmcount_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
{
struct localuser *u;
char *args, *context, *box, *folder;
LOCAL_USER_ACF_ADD(u);
buf[0] = '\0';
args = ast_strdupa(data);
if (!args) {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return buf;
}
box = strsep(&args, "|");
if (strchr(box, '@')) {
context = box;
box = strsep(&context, "@");
} else {
context = "default";
}
if (args) {
folder = args;
} else {
folder = "INBOX";
}
snprintf(buf, len, "%d", hasvoicemail_internal(context, box, folder));
LOCAL_USER_REMOVE(u);
return buf;
}
struct ast_custom_function acf_vmcount = {
.name = "VMCOUNT",
.synopsis = "Counts the voicemail in a specified mailbox",
.syntax = "VMCOUNT(vmbox[@context][|folder])",
.desc =
" context - defaults to \"default\"\n"
" folder - defaults to \"INBOX\"\n",
.read = acf_vmcount_exec,
};
static int load_config(void)
{
#ifdef USE_ODBC_STORAGE
struct ast_config *cfg;
char *tmp;
cfg = ast_config_load("voicemail.conf");
if (cfg) {
if (! (tmp = ast_variable_retrieve(cfg, "general", "odbcstorage")))
tmp = "asterisk";
ast_copy_string(odbc_database, tmp, sizeof(odbc_database));
if (! (tmp = ast_variable_retrieve(cfg, "general", "odbctable")))
tmp = "voicemessages";
ast_copy_string(odbc_table, tmp, sizeof(odbc_table));
ast_config_destroy(cfg);
}
#endif
return 0;
}
int reload(void)
{
return load_config();
}
int unload_module(void)
{
int res;
res = ast_custom_function_unregister(&acf_vmcount);
res |= ast_unregister_application(app_hasvoicemail);
res |= ast_unregister_application(app_hasnewvoicemail);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
load_config();
res = ast_custom_function_register(&acf_vmcount);
res |= ast_register_application(app_hasvoicemail, hasvoicemail_exec, hasvoicemail_synopsis, hasvoicemail_descrip);
res |= ast_register_application(app_hasnewvoicemail, hasvoicemail_exec, hasnewvoicemail_synopsis, hasnewvoicemail_descrip);
return res;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,241 +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 Stream to an icecast server via ICES (see contrib/asterisk-ices.xml)
*
* \ingroup applications
*/
#include <string.h>
#include <stdio.h>
#include <signal.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/time.h>
#include <errno.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/frame.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/options.h"
#define ICES "/usr/bin/ices"
#define LOCAL_ICES "/usr/local/bin/ices"
static char *tdesc = "Encode and Stream via icecast and ices";
static char *app = "ICES";
static char *synopsis = "Encode and stream using 'ices'";
static char *descrip =
" ICES(config.xml) Streams to an icecast server using ices\n"
"(available separately). A configuration file must be supplied\n"
"for ices (see examples/asterisk-ices.conf). \n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int icesencode(char *filename, int fd)
{
int res;
int x;
sigset_t fullset, oldset;
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;
}
/* Stop ignoring PIPE */
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
if (option_highpriority)
ast_set_priority(0);
dup2(fd, STDIN_FILENO);
for (x=STDERR_FILENO + 1;x<1024;x++) {
if ((x != STDIN_FILENO) && (x != STDOUT_FILENO))
close(x);
}
/* Most commonly installed in /usr/local/bin */
execl(ICES, "ices", filename, (char *)NULL);
/* But many places has it in /usr/bin */
execl(LOCAL_ICES, "ices", filename, (char *)NULL);
/* As a last-ditch effort, try to use PATH */
execlp("ices", "ices", filename, (char *)NULL);
ast_log(LOG_WARNING, "Execute of ices failed\n");
_exit(0);
}
static int ices_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
int fds[2];
int ms = -1;
int pid = -1;
int flags;
int oreadformat;
struct timeval last;
struct ast_frame *f;
char filename[256]="";
char *c;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "ICES requires an argument (configfile.xml)\n");
return -1;
}
LOCAL_USER_ADD(u);
last = ast_tv(0, 0);
if (pipe(fds)) {
ast_log(LOG_WARNING, "Unable to create pipe\n");
LOCAL_USER_REMOVE(u);
return -1;
}
flags = fcntl(fds[1], F_GETFL);
fcntl(fds[1], F_SETFL, flags | O_NONBLOCK);
ast_stopstream(chan);
if (chan->_state != AST_STATE_UP)
res = ast_answer(chan);
if (res) {
close(fds[0]);
close(fds[1]);
ast_log(LOG_WARNING, "Answer failed!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
oreadformat = chan->readformat;
res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
close(fds[0]);
close(fds[1]);
ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if (((char *)data)[0] == '/')
strncpy(filename, (char *)data, sizeof(filename) - 1);
else
snprintf(filename, sizeof(filename), "%s/%s", (char *)ast_config_AST_CONFIG_DIR, (char *)data);
/* Placeholder for options */
c = strchr(filename, '|');
if (c)
*c = '\0';
res = icesencode(filename, fds[0]);
close(fds[0]);
if (res >= 0) {
pid = res;
for (;;) {
/* Wait for audio, and stream */
ms = ast_waitfor(chan, -1);
if (ms < 0) {
ast_log(LOG_DEBUG, "Hangup detected\n");
res = -1;
break;
}
f = ast_read(chan);
if (!f) {
ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
res = -1;
break;
}
if (f->frametype == AST_FRAME_VOICE) {
res = write(fds[1], f->data, f->datalen);
if (res < 0) {
if (errno != EAGAIN) {
ast_log(LOG_WARNING, "Write failed to pipe: %s\n", strerror(errno));
res = -1;
ast_frfree(f);
break;
}
}
}
ast_frfree(f);
}
}
close(fds[1]);
if (pid > -1)
kill(pid, SIGKILL);
if (!res && oreadformat)
ast_set_read_format(chan, oreadformat);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, ices_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,147 +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 App to transmit an image
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/app.h"
#include "asterisk/options.h"
static char *tdesc = "Image Transmission Application";
static char *app = "SendImage";
static char *synopsis = "Send an image file";
static char *descrip =
" SendImage(filename): Sends an image on a channel. \n"
"If the channel supports image transport but the image send\n"
"fails, the channel will be hung up. Otherwise, the dialplan\n"
"continues execution.\n"
"The option string may contain the following character:\n"
" 'j' -- jump to priority n+101 if the channel doesn't support image transport\n"
"This application sets the following channel variable upon completion:\n"
" SENDIMAGESTATUS The status is the result of the attempt as a text string, one of\n"
" OK | NOSUPPORT \n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int sendimage_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u;
char *parse;
int priority_jump = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(filename);
AST_APP_ARG(options);
);
LOCAL_USER_ADD(u);
if (!(parse = ast_strdupa(data))) {
ast_log(LOG_WARNING, "Memory Error!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, parse);
if (ast_strlen_zero(args.filename)) {
ast_log(LOG_WARNING, "SendImage requires an argument (filename[|options])\n");
return -1;
}
if (args.options) {
if (strchr(args.options, 'j'))
priority_jump = 1;
}
if (!ast_supports_images(chan)) {
/* Does not support transport */
if (priority_jump || option_priority_jumping)
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
pbx_builtin_setvar_helper(chan, "SENDIMAGESTATUS", "NOSUPPORT");
LOCAL_USER_REMOVE(u);
return 0;
}
res = ast_send_image(chan, args.filename);
if (!res)
pbx_builtin_setvar_helper(chan, "SENDIMAGESTATUS", "OK");
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, sendimage_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,63 +1,34 @@
/*
* Asterisk -- An open source telephony toolkit.
* Asterisk -- A telephony toolkit for Linux.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
* Use /dev/dsp as an intercom.
*
* 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 Use /dev/dsp as an intercom.
*
* \ingroup applications
* the GNU General Public License
*/
#include <stdio.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 <pthread.h>
#include <sys/time.h>
#include <linux/soundcard.h>
#include <netinet/in.h>
#if defined(__linux__)
#include <linux/soundcard.h>
#elif defined(__FreeBSD__)
#include <sys/soundcard.h>
#else
#include <soundcard.h>
#endif
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/frame.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#ifdef __OpenBSD__
#define DEV_DSP "/dev/audio"
#else
#define DEV_DSP "/dev/dsp"
#endif
/* Number of 32 byte buffers -- each buffer is 2 ms */
#define BUFFER_SIZE 32
@@ -66,39 +37,34 @@ static char *tdesc = "Intercom using /dev/dsp for output";
static char *app = "Intercom";
static char *synopsis = "(Obsolete) Send to Intercom";
static char *descrip =
" Intercom(): Sends the user to the intercom (i.e. /dev/dsp). This program\n"
"is generally considered obselete by the chan_oss module. User can terminate\n"with a DTMF tone, or by hangup.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
AST_MUTEX_DEFINE_STATIC(sound_lock);
static pthread_mutex_t sound_lock = PTHREAD_MUTEX_INITIALIZER;
static int sound = -1;
static int write_audio(short *data, int len)
{
int res;
struct audio_buf_info info;
ast_mutex_lock(&sound_lock);
pthread_mutex_lock(&sound_lock);
if (sound < 0) {
ast_log(LOG_WARNING, "Sound device closed?\n");
ast_mutex_unlock(&sound_lock);
pthread_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);
pthread_mutex_unlock(&sound_lock);
return -1;
}
res = write(sound, data, len);
ast_mutex_unlock(&sound_lock);
pthread_mutex_unlock(&sound_lock);
return res;
}
static int create_audio(void)
static int create_audio()
{
int fmt, desired, res, fd;
fd = open(DEV_DSP, O_WRONLY);
@@ -150,70 +116,64 @@ static int intercom_exec(struct ast_channel *chan, void *data)
int res = 0;
struct localuser *u;
struct ast_frame *f;
int oreadformat;
LOCAL_USER_ADD(u);
/* Remember original read format */
oreadformat = chan->readformat;
/* Set mode to signed linear */
res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set format to signed linear on channel %s\n", chan->name);
LOCAL_USER_REMOVE(u);
struct ast_channel *trans;
if (!data) {
ast_log(LOG_WARNING, "Playback requires an argument (filename)\n");
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) {
LOCAL_USER_ADD(u);
/* See if we need a translator */
if (!(chan->format & AST_FORMAT_SLINEAR))
trans = ast_translator_create(chan, AST_FORMAT_SLINEAR, AST_DIRECTION_IN);
else
trans = chan;
if (trans) {
/* Read packets from the channel */
while(!res) {
res = ast_waitfor(trans, -1);
if (res > 0) {
res = 0;
f = ast_read(trans);
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);
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;
} else
res = -1;
}
}
}
if (!res)
ast_set_read_format(chan, oreadformat);
if (trans != chan)
ast_translator_destroy(trans);
} else
ast_log(LOG_WARNING, "Unable to build translator to signed linear format on '%s'\n", chan->name);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
STANDARD_HANGUP_LOCALUSERS;
if (sound > -1)
close(sound);
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
return ast_unregister_application(app);
}
int load_module(void)
{
if (create_audio())
return -1;
return ast_register_application(app, intercom_exec, synopsis, descrip);
return ast_register_application(app, intercom_exec);
}
char *description(void)
@@ -227,8 +187,3 @@ int usecount(void)
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,144 +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
*
* \ingroup applications
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/lock.h"
#include "asterisk/app.h"
static char *tdesc = "IVR Demo Application";
static char *app = "IVRDemo";
static char *synopsis =
" This is a skeleton application that shows you the basic structure to create your\n"
"own asterisk applications and demonstrates the IVR demo.\n";
static int ivr_demo_func(struct ast_channel *chan, void *data)
{
ast_verbose("IVR Demo, data is %s!\n", (char *)data);
return 0;
}
AST_IVR_DECLARE_MENU(ivr_submenu, "IVR Demo Sub Menu", 0,
{
{ "s", AST_ACTION_BACKGROUND, "demo-abouttotry" },
{ "s", AST_ACTION_WAITOPTION },
{ "1", AST_ACTION_PLAYBACK, "digits/1" },
{ "1", AST_ACTION_PLAYBACK, "digits/1" },
{ "1", AST_ACTION_RESTART },
{ "2", AST_ACTION_PLAYLIST, "digits/2;digits/3" },
{ "3", AST_ACTION_CALLBACK, ivr_demo_func },
{ "4", AST_ACTION_TRANSFER, "demo|s|1" },
{ "*", AST_ACTION_REPEAT },
{ "#", AST_ACTION_UPONE },
{ NULL }
});
AST_IVR_DECLARE_MENU(ivr_demo, "IVR Demo Main Menu", 0,
{
{ "s", AST_ACTION_BACKGROUND, "demo-congrats" },
{ "g", AST_ACTION_BACKGROUND, "demo-instruct" },
{ "g", AST_ACTION_WAITOPTION },
{ "1", AST_ACTION_PLAYBACK, "digits/1" },
{ "1", AST_ACTION_RESTART },
{ "2", AST_ACTION_MENU, &ivr_submenu },
{ "2", AST_ACTION_RESTART },
{ "i", AST_ACTION_PLAYBACK, "invalid" },
{ "i", AST_ACTION_REPEAT, (void *)(unsigned long)2 },
{ "#", AST_ACTION_EXIT },
{ NULL },
});
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int skel_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "skel requires an argument (filename)\n");
return -1;
}
LOCAL_USER_ADD(u);
/* Do our thing here */
if (chan->_state != AST_STATE_UP)
res = ast_answer(chan);
if (!res)
res = ast_ivr_menu_run(chan, &ivr_demo, data);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, skel_exec, tdesc, synopsis);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,141 +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 App to lookup the callerid number, and see if it is blacklisted
*
* \ingroup applications
*
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/callerid.h"
#include "asterisk/astdb.h"
#include "asterisk/options.h"
static char *tdesc = "Look up Caller*ID name/number from blacklist database";
static char *app = "LookupBlacklist";
static char *synopsis = "Look up Caller*ID name/number from blacklist database";
static char *descrip =
" LookupBlacklist(options): Looks up the Caller*ID number on the active\n"
"channel in the Asterisk database (family 'blacklist'). \n"
"The option string may contain the following character:\n"
" 'j' -- jump to n+101 priority if the number/name is found in the blacklist\n"
"This application sets the following channel variable upon completion:\n"
" LOOKUPBLSTATUS The status of the Blacklist lookup as a text string, one of\n"
" FOUND | NOTFOUND\n"
"Example: exten => 1234,1,LookupBlacklist()\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int
lookupblacklist_exec (struct ast_channel *chan, void *data)
{
char blacklist[1];
struct localuser *u;
int bl = 0;
int priority_jump = 0;
LOCAL_USER_ADD(u);
if (!ast_strlen_zero(data)) {
if (strchr(data, 'j'))
priority_jump = 1;
}
if (chan->cid.cid_num) {
if (!ast_db_get("blacklist", chan->cid.cid_num, blacklist, sizeof (blacklist))) {
if (option_verbose > 2)
ast_log(LOG_NOTICE, "Blacklisted number %s found\n",chan->cid.cid_num);
bl = 1;
}
}
if (chan->cid.cid_name) {
if (!ast_db_get("blacklist", chan->cid.cid_name, blacklist, sizeof (blacklist))) {
if (option_verbose > 2)
ast_log (LOG_NOTICE,"Blacklisted name \"%s\" found\n",chan->cid.cid_name);
bl = 1;
}
}
if (bl) {
if (priority_jump || option_priority_jumping)
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
pbx_builtin_setvar_helper(chan, "LOOKUPBLSTATUS", "FOUND");
} else
pbx_builtin_setvar_helper(chan, "LOOKUPBLSTATUS", "NOTFOUND");
LOCAL_USER_REMOVE(u);
return 0;
}
int unload_module (void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module (void)
{
return ast_register_application (app, lookupblacklist_exec, synopsis,descrip);
}
char *description (void)
{
return tdesc;
}
int usecount (void)
{
int res;
STANDARD_USECOUNT (res);
return res;
}
char *key ()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,120 +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 App to set callerid name from database, based on directory number
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/callerid.h"
#include "asterisk/astdb.h"
static char *tdesc = "Look up CallerID Name from local database";
static char *app = "LookupCIDName";
static char *synopsis = "Look up CallerID Name from local database";
static char *descrip =
" LookupCIDName: Looks up the Caller*ID number on the active\n"
"channel in the Asterisk database (family 'cidname') and sets the\n"
"Caller*ID name. Does nothing if no Caller*ID was received on the\n"
"channel. This is useful if you do not subscribe to Caller*ID\n"
"name delivery, or if you want to change the names on some incoming\n"
"calls.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int
lookupcidname_exec (struct ast_channel *chan, void *data)
{
char dbname[64];
struct localuser *u;
LOCAL_USER_ADD (u);
if (chan->cid.cid_num) {
if (!ast_db_get ("cidname", chan->cid.cid_num, dbname, sizeof (dbname))) {
ast_set_callerid (chan, NULL, dbname, NULL);
if (option_verbose > 2)
ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID name to %s\n",
dbname);
}
}
LOCAL_USER_REMOVE (u);
return 0;
}
int
unload_module (void)
{
int res;
res = ast_unregister_application (app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int
load_module (void)
{
return ast_register_application (app, lookupcidname_exec, synopsis,
descrip);
}
char *
description (void)
{
return tdesc;
}
int
usecount (void)
{
int res;
STANDARD_USECOUNT (res);
return res;
}
char *
key ()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,540 +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 Dial plan macro Implementation
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <sys/types.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/options.h"
#include "asterisk/config.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#define MAX_ARGS 80
/* special result value used to force macro exit */
#define MACRO_EXIT_RESULT 1024
static char *tdesc = "Extension Macros";
static char *descrip =
" Macro(macroname|arg1|arg2...): Executes a macro using the context\n"
"'macro-<macroname>', jumping to the 's' extension of that context and\n"
"executing each step, then returning when the steps end. \n"
"The calling extension, context, and priority are stored in ${MACRO_EXTEN}, \n"
"${MACRO_CONTEXT} and ${MACRO_PRIORITY} respectively. Arguments become\n"
"${ARG1}, ${ARG2}, etc in the macro context.\n"
"If you Goto out of the Macro context, the Macro will terminate and control\n"
"will be returned at the location of the Goto.\n"
"If ${MACRO_OFFSET} is set at termination, Macro will attempt to continue\n"
"at priority MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.\n"
"Extensions: While a macro is being executed, it becomes the current context.\n"
" This means that if a hangup occurs, for instance, that the macro\n"
" will be searched for an 'h' extension, NOT the context from which\n"
" the macro was called. So, make sure to define all appropriate\n"
" extensions in your macro! (you can use 'catch' in AEL) \n"
"WARNING: Because of the way Macro is implemented (it executes the priorities\n"
" contained within it via sub-engine), and a fixed per-thread\n"
" memory stack allowance, macros are limited to 7 levels\n"
" of nesting (macro calling macro calling macro, etc.); It\n"
" may be possible that stack-intensive applications in deeply nested\n"
" macros could cause asterisk to crash earlier than this limit.\n";
static char *if_descrip =
" MacroIf(<expr>?macroname_a[|arg1][:macroname_b[|arg1]])\n"
"Executes macro defined in <macroname_a> if <expr> is true\n"
"(otherwise <macroname_b> if provided)\n"
"Arguments and return values as in application macro()\n";
static char *exit_descrip =
" MacroExit():\n"
"Causes the currently running macro to exit as if it had\n"
"ended normally by running out of priorities to execute.\n"
"If used outside a macro, will likely cause unexpected\n"
"behavior.\n";
static char *app = "Macro";
static char *if_app = "MacroIf";
static char *exit_app = "MacroExit";
static char *synopsis = "Macro Implementation";
static char *if_synopsis = "Conditional Macro Implementation";
static char *exit_synopsis = "Exit From Macro";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
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)
{
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] = "";
char *offsets, *s, *inhangupc;
int offset, depth = 0, maxdepth = 7;
int setmacrocontext=0;
int autoloopflag, dead = 0, inhangup = 0;
char *save_macro_exten;
char *save_macro_context;
char *save_macro_priority;
char *save_macro_offset;
struct localuser *u;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Macro() requires arguments. See \"show application macro\" for help.\n");
return -1;
}
LOCAL_USER_ADD(u);
/* does the user want a deeper rabbit hole? */
s = pbx_builtin_getvar_helper(chan, "MACRO_RECURSION");
if (s)
sscanf(s, "%3d", &maxdepth);
/* Count how many levels deep the rabbit hole goes */
tmp = pbx_builtin_getvar_helper(chan, "MACRO_DEPTH");
if (tmp) {
sscanf(tmp, "%3d", &depth);
} else {
depth = 0;
}
/* Used for detecting whether to return when a Macro is called from another Macro after hangup */
if (strcmp(chan->exten, "h") == 0)
pbx_builtin_setvar_helper(chan, "MACRO_IN_HANGUP", "1");
inhangupc = pbx_builtin_getvar_helper(chan, "MACRO_IN_HANGUP");
if (!ast_strlen_zero(inhangupc))
sscanf(inhangupc, "%3d", &inhangup);
if (depth >= maxdepth) {
ast_log(LOG_ERROR, "Macro(): possible infinite loop detected. Returning early.\n");
LOCAL_USER_REMOVE(u);
return 0;
}
snprintf(depthc, sizeof(depthc), "%d", depth + 1);
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");
LOCAL_USER_REMOVE(u);
return 0;
}
snprintf(fullmacro, sizeof(fullmacro), "macro-%s", macro);
if (!ast_exists_extension(chan, fullmacro, "s", 1, chan->cid.cid_num)) {
if (!ast_context_find(fullmacro))
ast_log(LOG_WARNING, "No such context '%s' for macro '%s'\n", fullmacro, macro);
else
ast_log(LOG_WARNING, "Context '%s' for macro '%s' lacks 's' extension, priority 1\n", fullmacro, macro);
LOCAL_USER_REMOVE(u);
return 0;
}
/* Save old info */
oldpriority = chan->priority;
ast_copy_string(oldexten, chan->exten, sizeof(oldexten));
ast_copy_string(oldcontext, chan->context, sizeof(oldcontext));
if (ast_strlen_zero(chan->macrocontext)) {
ast_copy_string(chan->macrocontext, chan->context, sizeof(chan->macrocontext));
ast_copy_string(chan->macroexten, chan->exten, sizeof(chan->macroexten));
chan->macropriority = chan->priority;
setmacrocontext=1;
}
argc = 1;
/* Save old macro variables */
save_macro_exten = pbx_builtin_getvar_helper(chan, "MACRO_EXTEN");
if (save_macro_exten)
save_macro_exten = strdup(save_macro_exten);
pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", oldexten);
save_macro_context = pbx_builtin_getvar_helper(chan, "MACRO_CONTEXT");
if (save_macro_context)
save_macro_context = strdup(save_macro_context);
pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", oldcontext);
save_macro_priority = pbx_builtin_getvar_helper(chan, "MACRO_PRIORITY");
if (save_macro_priority)
save_macro_priority = strdup(save_macro_priority);
snprintf(pc, sizeof(pc), "%d", oldpriority);
pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", pc);
save_macro_offset = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET");
if (save_macro_offset)
save_macro_offset = strdup(save_macro_offset);
pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", NULL);
/* Setup environment for new run */
chan->exten[0] = 's';
chan->exten[1] = '\0';
ast_copy_string(chan->context, fullmacro, sizeof(chan->context));
chan->priority = 1;
while((cur = strsep(&rest, "|")) && (argc < MAX_ARGS)) {
/* Save copy of old arguments if we're overwriting some, otherwise
let them pass through to the other macro */
snprintf(varname, sizeof(varname), "ARG%d", argc);
oldargs[argc] = pbx_builtin_getvar_helper(chan, varname);
if (oldargs[argc])
oldargs[argc] = strdup(oldargs[argc]);
pbx_builtin_setvar_helper(chan, varname, cur);
argc++;
}
autoloopflag = ast_test_flag(chan, AST_FLAG_IN_AUTOLOOP);
ast_set_flag(chan, AST_FLAG_IN_AUTOLOOP);
while(ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
struct ast_context *c;
struct ast_exten *e;
runningapp[0] = '\0';
runningdata[0] = '\0';
/* What application will execute? */
if (ast_lock_contexts()) {
ast_log(LOG_WARNING, "Failed to lock contexts list\n");
} else {
for (c = ast_walk_contexts(NULL), e = NULL; c; c = ast_walk_contexts(c)) {
if (!strcmp(ast_get_context_name(c), chan->context)) {
if (ast_lock_context(c)) {
ast_log(LOG_WARNING, "Unable to lock context?\n");
} else {
e = find_matching_priority(c, chan->exten, chan->priority, chan->cid.cid_num);
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))) {
/* Something bad happened, or a hangup has been requested. */
if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F')) ||
(res == '*') || (res == '#')) {
/* Just return result as to the previous application as if it had been dialed */
ast_log(LOG_DEBUG, "Oooh, got something to jump out with ('%c')!\n", res);
break;
}
switch(res) {
case MACRO_EXIT_RESULT:
res = 0;
goto out;
case AST_PBX_KEEPALIVE:
if (option_debug)
ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited KEEPALIVE in macro %s on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
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);
if (option_verbose > 1)
ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
dead = 1;
goto out;
}
}
ast_log(LOG_DEBUG, "Executed application: %s\n", runningapp);
if (!strcasecmp(runningapp, "GOSUB")) {
gosub_level++;
ast_log(LOG_DEBUG, "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_log(LOG_DEBUG, "Incrementing gosub_level\n");
}
} else {
if (!ast_strlen_zero(app2)) {
gosub_level++;
ast_log(LOG_DEBUG, "Incrementing gosub_level\n");
}
}
} else if (!strcasecmp(runningapp, "RETURN")) {
gosub_level--;
ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
} else if (!strcasecmp(runningapp, "STACKPOP")) {
gosub_level--;
ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
} else if (!strncasecmp(runningapp, "EXEC", 4)) {
/* Must evaluate args to find actual app */
char tmp2[1024] = "", *tmp3 = NULL;
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_log(LOG_DEBUG, "Last app: %s\n", tmp3);
if (tmp3 && !strncasecmp(tmp3, "GOSUB", 5)) {
gosub_level++;
ast_log(LOG_DEBUG, "Incrementing gosub_level\n");
} else if (tmp3 && !strncasecmp(tmp3, "RETURN", 6)) {
gosub_level--;
ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
} else if (tmp3 && !strncasecmp(tmp3, "STACKPOP", 8)) {
gosub_level--;
ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
}
}
if (gosub_level == 0 && 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 && !inhangup) {
ast_log(LOG_DEBUG, "Extension %s, macroexten %s, priority %d returned normally even though call was hung up\n",
chan->exten, chan->macroexten, chan->priority);
goto out;
}
chan->priority++;
}
out:
/* 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);
if (!dead) {
pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
ast_set2_flag(chan, autoloopflag, AST_FLAG_IN_AUTOLOOP);
}
for (x = 1; x < argc; x++) {
/* Restore old arguments and delete ours */
snprintf(varname, sizeof(varname), "ARG%d", x);
if (oldargs[x]) {
if (!dead)
pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
free(oldargs[x]);
} else if (!dead) {
pbx_builtin_setvar_helper(chan, varname, NULL);
}
}
/* Restore macro variables */
if (!dead) {
pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", save_macro_context);
pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
}
if (save_macro_exten)
free(save_macro_exten);
if (save_macro_context)
free(save_macro_context);
if (save_macro_priority)
free(save_macro_priority);
if (!dead && setmacrocontext) {
chan->macrocontext[0] = '\0';
chan->macroexten[0] = '\0';
chan->macropriority = 0;
}
if (!dead && !strcasecmp(chan->context, fullmacro)) {
/* If we're leaving the macro normally, restore original information */
chan->priority = oldpriority;
ast_copy_string(chan->context, oldcontext, sizeof(chan->context));
if (!(chan->_softhangup & AST_SOFTHANGUP_ASYNCGOTO)) {
/* Copy the extension, so long as we're not in softhangup, where we could be given an asyncgoto */
ast_copy_string(chan->exten, oldexten, sizeof(chan->exten));
if ((offsets = pbx_builtin_getvar_helper(chan, "MACRO_OFFSET"))) {
/* Handle macro offset if it's set by checking the availability of step n + offset + 1, otherwise continue
normally if there is any problem */
if (sscanf(offsets, "%30d", &offset) == 1) {
if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + offset + 1, chan->cid.cid_num)) {
chan->priority += offset;
}
}
}
}
}
if (!dead)
pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
if (save_macro_offset)
free(save_macro_offset);
LOCAL_USER_REMOVE(u);
return res;
}
static int macroif_exec(struct ast_channel *chan, void *data)
{
char *expr = NULL, *label_a = NULL, *label_b = NULL;
int res = 0;
struct localuser *u;
LOCAL_USER_ADD(u);
expr = ast_strdupa(data);
if (!expr) {
ast_log(LOG_ERROR, "Out of Memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if ((label_a = strchr(expr, '?'))) {
*label_a = '\0';
label_a++;
if ((label_b = strchr(label_a, ':'))) {
*label_b = '\0';
label_b++;
}
if (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");
LOCAL_USER_REMOVE(u);
return res;
}
static int macro_exit_exec(struct ast_channel *chan, void *data)
{
return MACRO_EXIT_RESULT;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(if_app);
res |= ast_unregister_application(exit_app);
res |= ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
res = ast_register_application(exit_app, macro_exit_exec, exit_synopsis, exit_descrip);
res |= ast_register_application(if_app, macroif_exec, if_synopsis, if_descrip);
res |= ast_register_application(app, macro_exec, synopsis, descrip);
return res;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

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

View File

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

File diff suppressed because it is too large Load Diff

View File

@@ -1,170 +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 Digital Milliwatt Test
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <errno.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
static char *tdesc = "Digital Milliwatt (mu-law) Test Application";
static char *app = "Milliwatt";
static char *synopsis = "Generate a Constant 1000Hz tone at 0dbm (mu-law)";
static char *descrip =
"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)
{
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)
{
free(data);
return;
}
static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int samples)
{
struct ast_frame wf;
unsigned char buf[AST_FRIENDLY_OFFSET + 640];
int i,*indexp = (int *) data;
if (len + AST_FRIENDLY_OFFSET > sizeof(buf))
{
ast_log(LOG_WARNING,"Only doing %d bytes (%d bytes requested)\n",(int)(sizeof(buf) - AST_FRIENDLY_OFFSET),len);
len = sizeof(buf) - AST_FRIENDLY_OFFSET;
}
wf.frametype = AST_FRAME_VOICE;
wf.subclass = AST_FORMAT_ULAW;
wf.offset = AST_FRIENDLY_OFFSET;
wf.mallocd = 0;
wf.data = buf + AST_FRIENDLY_OFFSET;
wf.datalen = len;
wf.samples = wf.datalen;
wf.src = "app_milliwatt";
wf.delivery.tv_sec = 0;
wf.delivery.tv_usec = 0;
wf.prev = wf.next = NULL;
/* create a buffer containing the digital milliwatt pattern */
for(i = 0; i < len; i++)
{
buf[AST_FRIENDLY_OFFSET + i] = digital_milliwatt[(*indexp)++];
*indexp &= 7;
}
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 =
{
alloc: milliwatt_alloc,
release: milliwatt_release,
generate: milliwatt_generate,
} ;
static int milliwatt_exec(struct ast_channel *chan, void *data)
{
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);
}
if (ast_activate_generator(chan,&milliwattgen,"milliwatt") < 0)
{
ast_log(LOG_WARNING,"Failed to activate generator on '%s'\n",chan->name);
LOCAL_USER_REMOVE(u);
return -1;
}
while(!ast_safe_sleep(chan, 10000));
ast_deactivate_generator(chan);
LOCAL_USER_REMOVE(u);
return -1;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, milliwatt_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,454 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2005, Anthony Minessale II
* Copyright (C) 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
* Kevin P. Fleming <kpfleming@digium.com>
*
* Based on app_muxmon.c provided by
* Anthony Minessale II <anthmct@yahoo.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
* \brief MixMonitor() - Record a call and mix the audio during the recording
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/chanspy.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/lock.h"
#include "asterisk/cli.h"
#include "asterisk/options.h"
#include "asterisk/app.h"
#include "asterisk/linkedlists.h"
#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
static const char *tdesc = "Mixed Audio Monitoring Application";
static const char *app = "MixMonitor";
static const char *synopsis = "Record a call and mix the audio during the recording";
static const char *desc = ""
" MixMonitor(<file>.<ext>[|<options>[|<command>]])\n\n"
"Records the audio on the current channel to the specified file.\n"
"If the filename is an absolute path, uses that path, otherwise\n"
"creates the file in the configured monitoring directory from\n"
"asterisk.conf.\n\n"
"Valid options:\n"
" a - Append to the file instead of overwriting it.\n"
" b - Only save audio to the file while the channel is bridged.\n"
" Note: Does not include conferences or sounds played to each bridged\n"
" party.\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"
"";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static const char *mixmonitor_spy_type = "MixMonitor";
struct mixmonitor {
struct ast_channel_spy spy;
char *filename;
char *post_process;
char *name;
unsigned int flags;
};
enum {
MUXFLAG_APPEND = (1 << 1),
MUXFLAG_BRIDGED = (1 << 2),
MUXFLAG_VOLUME = (1 << 3),
MUXFLAG_READVOLUME = (1 << 4),
MUXFLAG_WRITEVOLUME = (1 << 5),
} mixmonitor_flags;
enum {
OPT_ARG_READVOLUME = 0,
OPT_ARG_WRITEVOLUME,
OPT_ARG_VOLUME,
OPT_ARG_ARRAY_SIZE,
} mixmonitor_args;
AST_APP_OPTIONS(mixmonitor_opts, {
AST_APP_OPTION('a', MUXFLAG_APPEND),
AST_APP_OPTION('b', MUXFLAG_BRIDGED),
AST_APP_OPTION_ARG('v', MUXFLAG_READVOLUME, OPT_ARG_READVOLUME),
AST_APP_OPTION_ARG('V', MUXFLAG_WRITEVOLUME, OPT_ARG_WRITEVOLUME),
AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
});
static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy)
{
struct ast_channel *peer;
int res;
if (!chan)
return -1;
ast_mutex_lock(&chan->lock);
res = ast_channel_spy_add(chan, spy);
ast_mutex_unlock(&chan->lock);
if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan)))
ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);
return res;
}
#define SAMPLES_PER_FRAME 160
static void *mixmonitor_thread(void *obj)
{
struct mixmonitor *mixmonitor = obj;
struct ast_frame *f = NULL;
struct ast_filestream *fs = NULL;
unsigned int oflags;
char *ext;
int errflag = 0;
STANDARD_INCREMENT_USECOUNT;
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "Begin MixMonitor Recording %s\n", mixmonitor->name);
ast_mutex_lock(&mixmonitor->spy.lock);
while (mixmonitor->spy.chan) {
struct ast_frame *next;
int write;
ast_channel_spy_trigger_wait(&mixmonitor->spy);
if (!mixmonitor->spy.chan || mixmonitor->spy.status != CHANSPY_RUNNING)
break;
while (1) {
if (!(f = ast_channel_spy_read_frame(&mixmonitor->spy, SAMPLES_PER_FRAME)))
break;
write = (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) ||
ast_bridged_channel(mixmonitor->spy.chan));
/* it is possible for ast_channel_spy_read_frame() to return a chain
of frames if a queue flush was necessary, so process them
*/
for (; f; f = next) {
next = f->next;
if (write && errflag == 0) {
if (!fs) {
/* Determine creation flags and filename plus extension for filestream */
oflags = O_CREAT | O_WRONLY;
oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
if ((ext = strrchr(mixmonitor->filename, '.')))
*(ext++) = '\0';
else
ext = "raw";
/* Move onto actually creating the filestream */
if (!(fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) {
ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
errflag = 1;
}
}
if (fs)
ast_writestream(fs, f);
}
ast_frfree(f);
}
}
}
ast_mutex_unlock(&mixmonitor->spy.lock);
ast_channel_spy_free(&mixmonitor->spy);
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "End MixMonitor Recording %s\n", mixmonitor->name);
if (mixmonitor->post_process) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_2 "Executing [%s]\n", mixmonitor->post_process);
ast_safe_system(mixmonitor->post_process);
}
if (fs)
ast_closestream(fs);
free(mixmonitor);
STANDARD_DECREMENT_USECOUNT;
return NULL;
}
static void launch_monitor_thread(struct ast_channel *chan, const char *filename, unsigned int flags,
int readvol, int writevol, const char *post_process)
{
pthread_attr_t attr;
pthread_t thread;
struct mixmonitor *mixmonitor;
char postprocess2[1024] = "";
size_t len;
len = sizeof(*mixmonitor) + strlen(chan->name) + strlen(filename) + 2;
/* 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 = calloc(1, len))) {
ast_log(LOG_ERROR, "Memory Error!\n");
return;
}
/* Copy over flags and channel name */
mixmonitor->flags = flags;
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);
/* Setup the actual spy before creating our thread */
ast_set_flag(&mixmonitor->spy, CHANSPY_FORMAT_AUDIO);
ast_set_flag(&mixmonitor->spy, CHANSPY_MIXAUDIO);
mixmonitor->spy.type = mixmonitor_spy_type;
mixmonitor->spy.status = CHANSPY_RUNNING;
mixmonitor->spy.read_queue.format = AST_FORMAT_SLINEAR;
mixmonitor->spy.write_queue.format = AST_FORMAT_SLINEAR;
if (readvol) {
ast_set_flag(&mixmonitor->spy, CHANSPY_READ_VOLADJUST);
mixmonitor->spy.read_vol_adjustment = readvol;
}
if (writevol) {
ast_set_flag(&mixmonitor->spy, CHANSPY_WRITE_VOLADJUST);
mixmonitor->spy.write_vol_adjustment = writevol;
}
ast_mutex_init(&mixmonitor->spy.lock);
if (startmon(chan, &mixmonitor->spy)) {
ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
mixmonitor->spy.type, chan->name);
/* Since we couldn't add ourselves - bail out! */
ast_mutex_destroy(&mixmonitor->spy.lock);
free(mixmonitor);
return;
}
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
ast_pthread_create(&thread, &attr, mixmonitor_thread, mixmonitor);
pthread_attr_destroy(&attr);
}
static int mixmonitor_exec(struct ast_channel *chan, void *data)
{
int x, readvol = 0, writevol = 0;
struct localuser *u;
struct ast_flags flags = {0};
char *parse;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(filename);
AST_APP_ARG(options);
AST_APP_ARG(post_process);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
return -1;
}
LOCAL_USER_ADD(u);
if (!(parse = ast_strdupa(data))) {
ast_log(LOG_WARNING, "Memory Error!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, parse);
if (ast_strlen_zero(args.filename)) {
ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if (args.options) {
char *opts[OPT_ARG_ARRAY_SIZE] = { NULL, };
ast_app_parse_options(mixmonitor_opts, &flags, opts, args.options);
if (ast_test_flag(&flags, MUXFLAG_READVOLUME)) {
if (ast_strlen_zero(opts[OPT_ARG_READVOLUME])) {
ast_log(LOG_WARNING, "No volume level was provided for the heard volume ('v') option.\n");
} else if ((sscanf(opts[OPT_ARG_READVOLUME], "%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;
}
pbx_builtin_setvar_helper(chan, "MIXMONITOR_FILENAME", args.filename);
launch_monitor_thread(chan, args.filename, flags.flags, readvol, writevol, args.post_process);
LOCAL_USER_REMOVE(u);
return 0;
}
static int mixmonitor_cli(int fd, int argc, char **argv)
{
struct ast_channel *chan;
if (argc < 3)
return RESULT_SHOWUSAGE;
if (!(chan = ast_get_channel_by_name_prefix_locked(argv[2], strlen(argv[2])))) {
ast_cli(fd, "No channel matching '%s' found.\n", argv[2]);
return RESULT_SUCCESS;
}
if (!strcasecmp(argv[1], "start"))
mixmonitor_exec(chan, argv[3]);
else if (!strcasecmp(argv[1], "stop"))
ast_channel_spy_stop_by_type(chan, mixmonitor_spy_type);
ast_mutex_unlock(&chan->lock);
return RESULT_SUCCESS;
}
static struct ast_cli_entry cli_mixmonitor = {
{ "mixmonitor", NULL, NULL },
mixmonitor_cli,
"Execute a MixMonitor command",
"mixmonitor <start|stop> <chan_name> [<args>]\n"
};
int unload_module(void)
{
int res;
res = ast_cli_unregister(&cli_mixmonitor);
res |= ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
res = ast_cli_register(&cli_mixmonitor);
res |= ast_register_application(app, mixmonitor_exec, synopsis, desc);
return res;
}
char *description(void)
{
return (char *) tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,64 +1,38 @@
/*
* 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, 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 MP3 file -- uses mpg123
*
* \ingroup applications
* the GNU General Public License
*/
#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 <pthread.h>
#include <sys/time.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/frame.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/options.h"
#define LOCAL_MPG_123 "/usr/local/bin/mpg123"
#define MPG_123 "/usr/bin/mpg123"
static char *tdesc = "Silly MP3 Application";
static char *app = "MP3Player";
static char *synopsis = "Play an MP3 file or stream";
static char *descrip =
" MP3Player(location) Executes mpg123 to play the given location,\n"
"which typically would be a filename or a URL. User can exit by pressing\n"
"any key on the dialpad, or by hanging up.";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
@@ -66,150 +40,73 @@ LOCAL_USER_DECL;
static int mp3play(char *filename, int fd)
{
int res;
int x;
sigset_t fullset, oldset;
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;
}
if (option_highpriority)
ast_set_priority(0);
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
dup2(fd, STDOUT_FILENO);
for (x=STDERR_FILENO + 1;x<256;x++) {
if (x != STDOUT_FILENO)
close(x);
}
/* Execute mpg123, but buffer if it's a net connection */
if (!strncasecmp(filename, "http://", 7)) {
/* Most commonly installed in /usr/local/bin */
execl(LOCAL_MPG_123, "mpg123", "-q", "-s", "-b", "1024", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
/* But many places has it in /usr/bin */
execl(MPG_123, "mpg123", "-q", "-s", "-b", "1024","-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
/* As a last-ditch effort, try to use PATH */
execlp("mpg123", "mpg123", "-q", "-s", "-b", "1024", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
}
else {
/* Most commonly installed in /usr/local/bin */
execl(MPG_123, "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
/* But many places has it in /usr/bin */
execl(LOCAL_MPG_123, "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
/* As a last-ditch effort, try to use PATH */
execlp("mpg123", "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
}
if (strncmp(filename, "http://", 7))
execl(MPG_123, MPG_123, "-q", "-s", "-b", "1024", "--mono", "-r", "8000", filename, NULL);
else
execl(MPG_123, MPG_123, "-q", "-s", "--mono", "-r", "8000", filename, NULL);
ast_log(LOG_WARNING, "Execute of mpg123 failed\n");
_exit(0);
}
static int timed_read(int fd, void *data, int datalen, int timeout)
{
int res;
struct pollfd fds[1];
fds[0].fd = fd;
fds[0].events = POLLIN;
res = poll(fds, 1, timeout);
if (res < 1) {
ast_log(LOG_NOTICE, "Poll timed out/errored out with %d\n", res);
return -1;
}
return read(fd, data, datalen);
return -1;
}
static int mp3_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
struct ast_channel *trans;
int fds[2];
int rfds[2];
int ms = -1;
int pid = -1;
int owriteformat;
int timeout = 2000;
struct timeval next;
int pid;
int us;
struct timeval tv;
struct timeval last;
struct ast_frame *f;
struct myframe {
struct ast_frame f;
char offset[AST_FRIENDLY_OFFSET];
short frdata[160];
char frdata[160];
} myf;
if (ast_strlen_zero(data)) {
last.tv_usec = 0;
last.tv_sec = 0;
if (!data) {
ast_log(LOG_WARNING, "MP3 Playback requires an argument (filename)\n");
return -1;
}
LOCAL_USER_ADD(u);
if (pipe(fds)) {
ast_log(LOG_WARNING, "Unable to create pipe\n");
LOCAL_USER_REMOVE(u);
return -1;
}
LOCAL_USER_ADD(u);
ast_stopstream(chan);
owriteformat = chan->writeformat;
res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
LOCAL_USER_REMOVE(u);
return -1;
}
res = mp3play((char *)data, fds[1]);
if (!strncasecmp((char *)data, "http://", 7)) {
timeout = 10000;
}
/* Wait 1000 ms first */
next = ast_tvnow();
next.tv_sec += 1;
if (res >= 0) {
pid = res;
/* Order is important -- there's almost always going to be mp3... we want to prioritize the
user */
for (;;) {
ms = ast_tvdiff_ms(next, ast_tvnow());
if (ms <= 0) {
res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata), timeout);
if (res > 0) {
myf.f.frametype = AST_FRAME_VOICE;
myf.f.subclass = AST_FORMAT_SLINEAR;
myf.f.datalen = res;
myf.f.samples = res / 2;
myf.f.mallocd = 0;
myf.f.offset = AST_FRIENDLY_OFFSET;
myf.f.src = __PRETTY_FUNCTION__;
myf.f.delivery.tv_sec = 0;
myf.f.delivery.tv_usec = 0;
myf.f.data = myf.frdata;
if (ast_write(chan, &myf.f) < 0) {
res = -1;
break;
}
} else {
ast_log(LOG_DEBUG, "No more mp3\n");
res = 0;
break;
}
next = ast_tvadd(next, ast_samp2tv(myf.f.samples, 8000));
} else {
ms = ast_waitfor(chan, ms);
if (ms < 0) {
if (chan->format & AST_FORMAT_SLINEAR)
trans = chan;
else
trans = ast_translator_create(chan, AST_FORMAT_SLINEAR, AST_DIRECTION_OUT);
if (trans) {
res = mp3play((char *)data, fds[1]);
if (res >= 0) {
pid = res;
/* Order is important -- there's almost always going to be mp3... we want to prioritize the
user */
rfds[0] = trans->fd;
rfds[1] = fds[0];
for (;;) {
CHECK_BLOCKING(trans);
res = ast_waitfor_n_fd(rfds, 2, &ms);
trans->blocking = 0;
if (res < 1) {
ast_log(LOG_DEBUG, "Hangup detected\n");
res = -1;
break;
}
if (ms) {
f = ast_read(chan);
} else if (res == trans->fd) {
f = ast_read(trans);
if (!f) {
ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
res = -1;
@@ -222,37 +119,65 @@ static int mp3_exec(struct ast_channel *chan, void *data)
break;
}
ast_frfree(f);
}
} else if (res == fds[0]) {
gettimeofday(&tv, NULL);
if (last.tv_sec || last.tv_usec) {
/* We should wait at least a frame length */
us = sizeof(myf.frdata) / 16 * 1000;
/* Subtract 1,000,000 us for each second late we've passed */
us -= (tv.tv_sec - last.tv_sec) * 1000000;
/* And one for each us late we've passed */
us -= (tv.tv_usec - last.tv_usec);
/* Sleep that long if needed */
if (us > 0)
usleep(us);
}
last = tv;
res = read(fds[0], myf.frdata, sizeof(myf.frdata));
if (res > 0) {
myf.f.frametype = AST_FRAME_VOICE;
myf.f.subclass = AST_FORMAT_SLINEAR;
myf.f.datalen = res;
myf.f.timelen = res / 16;
myf.f.mallocd = 0;
myf.f.offset = AST_FRIENDLY_OFFSET;
myf.f.src = __PRETTY_FUNCTION__;
myf.f.data = myf.frdata;
if (ast_write(trans, &myf.f) < 0) {
res = -1;
break;
}
} else {
ast_log(LOG_DEBUG, "No more mp3\n");
res = 0;
}
} else {
ast_log(LOG_DEBUG, "HuhHHH?\n");
res = -1;
break;
}
}
kill(pid, SIGTERM);
}
}
if (trans != chan)
ast_translator_destroy(trans);
} else
ast_log(LOG_WARNING, "No translator channel available\n");
close(fds[0]);
close(fds[1]);
if (pid > -1)
kill(pid, SIGKILL);
if (!res && owriteformat)
ast_set_write_format(chan, owriteformat);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
return ast_unregister_application(app);
}
int load_module(void)
{
return ast_register_application(app, mp3_exec, synopsis, descrip);
return ast_register_application(app, mp3_exec);
}
char *description(void)
@@ -266,8 +191,3 @@ int usecount(void)
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,255 +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 Silly application to play an NBScat file -- uses nbscat8k
*
* \ingroup applications
*/
#include <string.h>
#include <stdio.h>
#include <signal.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/time.h>
#include <sys/socket.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/frame.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/options.h"
#define LOCAL_NBSCAT "/usr/local/bin/nbscat8k"
#define NBSCAT "/usr/bin/nbscat8k"
#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.";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int NBScatplay(int fd)
{
int res;
int x;
sigset_t fullset, oldset;
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;
}
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
if (option_highpriority)
ast_set_priority(0);
dup2(fd, STDOUT_FILENO);
for (x = STDERR_FILENO + 1; x < 1024; x++) {
if (x != STDOUT_FILENO)
close(x);
}
/* Most commonly installed in /usr/local/bin */
execl(NBSCAT, "nbscat8k", "-d", (char *)NULL);
execl(LOCAL_NBSCAT, "nbscat8k", "-d", (char *)NULL);
ast_log(LOG_WARNING, "Execute of nbscat8k failed\n");
_exit(0);
}
static int timed_read(int fd, void *data, int datalen)
{
int res;
struct pollfd fds[1];
fds[0].fd = fd;
fds[0].events = POLLIN;
res = poll(fds, 1, 2000);
if (res < 1) {
ast_log(LOG_NOTICE, "Selected timed out/errored out with %d\n", res);
return -1;
}
return read(fd, data, datalen);
}
static int NBScat_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
int fds[2];
int ms = -1;
int pid = -1;
int owriteformat;
struct timeval next;
struct ast_frame *f;
struct myframe {
struct ast_frame f;
char offset[AST_FRIENDLY_OFFSET];
short frdata[160];
} myf;
LOCAL_USER_ADD(u);
if (socketpair(AF_LOCAL, SOCK_STREAM, 0, fds)) {
ast_log(LOG_WARNING, "Unable to create socketpair\n");
LOCAL_USER_REMOVE(u);
return -1;
}
ast_stopstream(chan);
owriteformat = chan->writeformat;
res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
LOCAL_USER_REMOVE(u);
return -1;
}
res = NBScatplay(fds[1]);
/* Wait 1000 ms first */
next = ast_tvnow();
next.tv_sec += 1;
if (res >= 0) {
pid = res;
/* Order is important -- there's almost always going to be mp3... we want to prioritize the
user */
for (;;) {
ms = ast_tvdiff_ms(next, ast_tvnow());
if (ms <= 0) {
res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata));
if (res > 0) {
myf.f.frametype = AST_FRAME_VOICE;
myf.f.subclass = AST_FORMAT_SLINEAR;
myf.f.datalen = res;
myf.f.samples = res / 2;
myf.f.mallocd = 0;
myf.f.offset = AST_FRIENDLY_OFFSET;
myf.f.src = __PRETTY_FUNCTION__;
myf.f.delivery.tv_sec = 0;
myf.f.delivery.tv_usec = 0;
myf.f.data = myf.frdata;
if (ast_write(chan, &myf.f) < 0) {
res = -1;
break;
}
} else {
ast_log(LOG_DEBUG, "No more mp3\n");
res = 0;
break;
}
next = ast_tvadd(next, ast_samp2tv(myf.f.samples, 8000));
} else {
ms = ast_waitfor(chan, ms);
if (ms < 0) {
ast_log(LOG_DEBUG, "Hangup detected\n");
res = -1;
break;
}
if (ms) {
f = ast_read(chan);
if (!f) {
ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
res = -1;
break;
}
if (f->frametype == AST_FRAME_DTMF) {
ast_log(LOG_DEBUG, "User pressed a key\n");
ast_frfree(f);
res = 0;
break;
}
ast_frfree(f);
}
}
}
}
close(fds[0]);
close(fds[1]);
if (pid > -1)
kill(pid, SIGKILL);
if (!res && owriteformat)
ast_set_write_format(chan, owriteformat);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, NBScat_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,377 +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 Open Settlement Protocol Lookup
*
* \ingroup applications
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <ctype.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/utils.h"
#include "asterisk/causes.h"
#include "asterisk/astosp.h"
#include "asterisk/app.h"
#include "asterisk/options.h"
static char *tdesc = "OSP Lookup";
static char *app = "OSPLookup";
static char *app2 = "OSPNext";
static char *app3 = "OSPFinish";
static char *synopsis = "Lookup number in OSP";
static char *synopsis2 = "Lookup next OSP entry";
static char *synopsis3 = "Record OSP entry";
static char *descrip =
" OSPLookup(exten[|provider[|options]]): Looks up an extension via OSP and sets\n"
"the variables, where 'n' is the number of the result beginning with 1:\n"
" ${OSPTECH}: The technology to use for the call\n"
" ${OSPDEST}: The destination to use for the call\n"
" ${OSPTOKEN}: The actual OSP token as a string\n"
" ${OSPHANDLE}: The OSP Handle for anything remaining\n"
" ${OSPRESULTS}: The number of OSP results total remaining\n"
"\n"
"The option string may contain the following character:\n"
" 'j' -- jump to n+101 priority if the lookup was NOT successful\n"
"This application sets the following channel variable upon completion:\n"
" OSPLOOKUPSTATUS The status of the OSP Lookup attempt as a text string, one of\n"
" SUCCESS | FAILED \n";
static char *descrip2 =
" OSPNext(cause[|options]): Looks up the next OSP Destination for ${OSPHANDLE}\n"
"See OSPLookup for more information\n"
"\n"
"The option string may contain the following character:\n"
" 'j' -- jump to n+101 priority if the lookup was NOT successful\n"
"This application sets the following channel variable upon completion:\n"
" OSPNEXTSTATUS The status of the OSP Next attempt as a text string, one of\n"
" SUCCESS | FAILED \n";
static char *descrip3 =
" OSPFinish(status[|options]): Records call state for ${OSPHANDLE}, according to\n"
"status, which should be one of BUSY, CONGESTION, ANSWER, NOANSWER, or CHANUNAVAIL\n"
"or coincidentally, just what the Dial application stores in its ${DIALSTATUS}.\n"
"\n"
"The option string may contain the following character:\n"
" 'j' -- jump to n+101 priority if the finish attempt was NOT successful\n"
"This application sets the following channel variable upon completion:\n"
" OSPFINISHSTATUS The status of the OSP Finish attempt as a text string, one of\n"
" SUCCESS | FAILED \n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int str2cause(char *cause)
{
if (!strcasecmp(cause, "BUSY"))
return AST_CAUSE_BUSY;
if (!strcasecmp(cause, "CONGESTION"))
return AST_CAUSE_CONGESTION;
if (!strcasecmp(cause, "ANSWER"))
return AST_CAUSE_NORMAL;
if (!strcasecmp(cause, "CANCEL"))
return AST_CAUSE_NORMAL;
if (!strcasecmp(cause, "NOANSWER"))
return AST_CAUSE_NOANSWER;
if (!strcasecmp(cause, "NOCHANAVAIL"))
return AST_CAUSE_CONGESTION;
ast_log(LOG_WARNING, "Unknown cause '%s', using NORMAL\n", cause);
return AST_CAUSE_NORMAL;
}
static int osplookup_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
char *temp;
struct ast_osp_result result;
int priority_jump = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(extension);
AST_APP_ARG(provider);
AST_APP_ARG(options);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "OSPLookup requires an argument OSPLookup(exten[|provider[|options]])\n");
return -1;
}
LOCAL_USER_ADD(u);
temp = ast_strdupa(data);
if (!temp) {
ast_log(LOG_ERROR, "Out of memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, temp);
if (args.options) {
if (strchr(args.options, 'j'))
priority_jump = 1;
}
ast_log(LOG_DEBUG, "Whoo hoo, looking up OSP on '%s' via '%s'\n", args.extension, args.provider ? args.provider : "<default>");
if ((res = ast_osp_lookup(chan, args.provider, args.extension, chan->cid.cid_num, &result)) > 0) {
char tmp[80];
snprintf(tmp, sizeof(tmp), "%d", result.handle);
pbx_builtin_setvar_helper(chan, "_OSPHANDLE", tmp);
pbx_builtin_setvar_helper(chan, "_OSPTECH", result.tech);
pbx_builtin_setvar_helper(chan, "_OSPDEST", result.dest);
pbx_builtin_setvar_helper(chan, "_OSPTOKEN", result.token);
snprintf(tmp, sizeof(tmp), "%d", result.numresults);
pbx_builtin_setvar_helper(chan, "_OSPRESULTS", tmp);
pbx_builtin_setvar_helper(chan, "OSPLOOKUPSTATUS", "SUCCESS");
} else {
if (!res) {
ast_log(LOG_NOTICE, "OSP Lookup failed for '%s' (provider '%s')\n", args.extension, args.provider ? args.provider : "<default>");
pbx_builtin_setvar_helper(chan, "OSPLOOKUPSTATUS", "FAILED");
} else
ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Lookup for '%s' (provider '%s')!\n", chan->name, args.extension, args.provider ? args.provider : "<default>" );
}
if (!res) {
/* Look for a "busy" place */
if (priority_jump || option_priority_jumping)
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
} else if (res > 0)
res = 0;
LOCAL_USER_REMOVE(u);
return res;
}
static int ospnext_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
char *temp;
int cause;
struct ast_osp_result result;
int priority_jump = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(cause);
AST_APP_ARG(options);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "OSPNext should have an argument (cause[|options])\n");
return -1;
}
LOCAL_USER_ADD(u);
temp = ast_strdupa(data);
if (!temp) {
ast_log(LOG_ERROR, "Out of memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, temp);
if (args.options) {
if (strchr(args.options, 'j'))
priority_jump = 1;
}
cause = str2cause(args.cause);
temp = pbx_builtin_getvar_helper(chan, "OSPHANDLE");
result.handle = -1;
if (ast_strlen_zero(temp) || (sscanf(temp, "%30d", &result.handle) != 1)) {
result.handle = -1;
}
temp = pbx_builtin_getvar_helper(chan, "OSPRESULTS");
if (ast_strlen_zero(temp) || (sscanf(temp, "%30d", &result.numresults) != 1)) {
result.numresults = 0;
}
if ((res = ast_osp_next(&result, cause)) > 0) {
char tmp[80];
snprintf(tmp, sizeof(tmp), "%d", result.handle);
pbx_builtin_setvar_helper(chan, "_OSPHANDLE", tmp);
pbx_builtin_setvar_helper(chan, "_OSPTECH", result.tech);
pbx_builtin_setvar_helper(chan, "_OSPDEST", result.dest);
pbx_builtin_setvar_helper(chan, "_OSPTOKEN", result.token);
snprintf(tmp, sizeof(tmp), "%d", result.numresults);
pbx_builtin_setvar_helper(chan, "_OSPRESULTS", tmp);
pbx_builtin_setvar_helper(chan, "OSPNEXTSTATUS", "SUCCESS");
} else {
if (!res) {
if (result.handle < 0)
ast_log(LOG_NOTICE, "OSP Lookup Next failed for handle '%d'\n", result.handle);
else
ast_log(LOG_DEBUG, "No OSP handle specified\n");
} else
ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Next!\n", chan->name);
pbx_builtin_setvar_helper(chan, "OSPNEXTSTATUS", "FAILED");
}
if (!res) {
/* Look for a "busy" place */
if (priority_jump || option_priority_jumping)
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
} else if (res > 0)
res = 0;
LOCAL_USER_REMOVE(u);
return res;
}
static int ospfinished_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
char *temp;
int cause;
time_t start=0, duration=0;
struct ast_osp_result result;
int priority_jump = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(status);
AST_APP_ARG(options);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "OSPFinish should have an argument (status[|options])\n");
return -1;
}
LOCAL_USER_ADD(u);
temp = ast_strdupa(data);
if (!temp) {
ast_log(LOG_ERROR, "Out of memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, temp);
if (args.options) {
if (strchr(args.options, 'j'))
priority_jump = 1;
}
if (chan->cdr) {
start = chan->cdr->answer.tv_sec;
if (start)
duration = time(NULL) - start;
else
duration = 0;
} else
ast_log(LOG_WARNING, "OSPFinish called on channel '%s' with no CDR!\n", chan->name);
cause = str2cause(args.status);
temp = pbx_builtin_getvar_helper(chan, "OSPHANDLE");
result.handle = -1;
if (!ast_strlen_zero(temp) && (sscanf(temp, "%30d", &result.handle) == 1) && (result.handle > -1)) {
if (!ast_osp_terminate(result.handle, cause, start, duration)) {
pbx_builtin_setvar_helper(chan, "_OSPHANDLE", "");
pbx_builtin_setvar_helper(chan, "OSPFINISHSTATUS", "SUCCESS");
res = 1;
}
} else {
if (!res) {
if (result.handle > -1)
ast_log(LOG_NOTICE, "OSP Finish failed for handle '%d'\n", result.handle);
else
ast_log(LOG_DEBUG, "No OSP handle specified\n");
pbx_builtin_setvar_helper(chan, "OSPFINISHSTATUS", "FAILED");
} else
ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Terminate!\n", chan->name);
}
if (!res) {
/* Look for a "busy" place */
if (priority_jump || option_priority_jumping)
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
} else if (res > 0)
res = 0;
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app3);
res |= ast_unregister_application(app2);
res |= ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
res = ast_register_application(app, osplookup_exec, synopsis, descrip);
res |= ast_register_application(app2, ospnext_exec, synopsis2, descrip2);
res |= ast_register_application(app3, ospfinished_exec, synopsis3, descrip3);
return res;
}
int reload(void)
{
return 0;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,246 +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
*
* \ingroup applications
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <errno.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/options.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/file.h"
#include "asterisk/app.h"
#include "asterisk/chanvars.h"
static const char *tdesc = "Page Multiple Phones";
static const char *app_page= "Page";
static const char *page_synopsis = "Pages phones";
static const char *page_descrip =
"Page(Technology/Resource&Technology2/Resource2[|options])\n"
" Places outbound calls to the given technology / resource and dumps\n"
"them into a conference bridge as muted participants. The original\n"
"caller is dumped into the conference as a speaker and the room is\n"
"destroyed when the original caller leaves. Valid options are:\n"
" d - full duplex audio\n"
" q - quiet, do not play beep to caller\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
enum {
PAGE_DUPLEX = (1 << 0),
PAGE_QUIET = (1 << 1),
} page_opt_flags;
AST_APP_OPTIONS(page_opts, {
AST_APP_OPTION('d', PAGE_DUPLEX),
AST_APP_OPTION('q', PAGE_QUIET),
});
struct calloutdata {
char cidnum[64];
char cidname[64];
char tech[64];
char resource[256];
char meetmeopts[64];
struct ast_variable *variables;
};
static void *page_thread(void *data)
{
struct calloutdata *cd = data;
ast_pbx_outgoing_app(cd->tech, AST_FORMAT_SLINEAR, cd->resource, 30000,
"MeetMe", cd->meetmeopts, NULL, 0, cd->cidnum, cd->cidname, cd->variables, NULL, NULL);
free(cd);
return NULL;
}
static void launch_page(struct ast_channel *chan, const char *meetmeopts, const char *tech, const char *resource)
{
struct calloutdata *cd;
const char *varname;
struct ast_variable *lastvar = NULL;
struct ast_var_t *varptr;
pthread_t t;
pthread_attr_t attr;
cd = malloc(sizeof(struct calloutdata));
if (cd) {
memset(cd, 0, sizeof(struct calloutdata));
ast_copy_string(cd->cidnum, chan->cid.cid_num ? chan->cid.cid_num : "", sizeof(cd->cidnum));
ast_copy_string(cd->cidname, chan->cid.cid_name ? chan->cid.cid_name : "", sizeof(cd->cidname));
ast_copy_string(cd->tech, tech, sizeof(cd->tech));
ast_copy_string(cd->resource, resource, sizeof(cd->resource));
ast_copy_string(cd->meetmeopts, meetmeopts, sizeof(cd->meetmeopts));
AST_LIST_TRAVERSE(&chan->varshead, varptr, entries) {
if (!(varname = ast_var_full_name(varptr)))
continue;
if (varname[0] == '_') {
struct ast_variable *newvar = NULL;
if (varname[1] == '_') {
newvar = ast_variable_new(varname, ast_var_value(varptr));
} else {
newvar = ast_variable_new(&varname[1], ast_var_value(varptr));
}
if (newvar) {
if (lastvar)
lastvar->next = newvar;
else
cd->variables = newvar;
lastvar = newvar;
}
}
}
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
if (ast_pthread_create(&t, &attr, page_thread, cd)) {
ast_log(LOG_WARNING, "Unable to create paging thread: %s\n", strerror(errno));
free(cd);
}
pthread_attr_destroy(&attr);
}
}
static int page_exec(struct ast_channel *chan, void *data)
{
struct localuser *u;
char *options;
char *tech, *resource;
char meetmeopts[80];
struct ast_flags flags = { 0 };
unsigned int confid = rand();
struct ast_app *app;
char *tmp;
int res=0;
char originator[AST_CHANNEL_NAME];
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
return -1;
}
LOCAL_USER_ADD(u);
if (!(app = pbx_findapp("MeetMe"))) {
ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
LOCAL_USER_REMOVE(u);
return -1;
};
options = ast_strdupa(data);
if (!options) {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return -1;
}
ast_copy_string(originator, chan->name, sizeof(originator));
if ((tmp = strchr(originator, '-')))
*tmp = '\0';
tmp = strsep(&options, "|");
if (options)
ast_app_parse_options(page_opts, &flags, NULL, options);
snprintf(meetmeopts, sizeof(meetmeopts), "%ud|%sqxdw(5)", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m");
while ((tech = strsep(&tmp, "&"))) {
/* don't call the originating device */
if (!strcasecmp(tech, originator))
continue;
if ((resource = strchr(tech, '/'))) {
*resource++ = '\0';
launch_page(chan, meetmeopts, tech, resource);
} else {
ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
}
}
if (!ast_test_flag(&flags, PAGE_QUIET)) {
res = ast_streamfile(chan, "beep", chan->language);
if (!res)
res = ast_waitstream(chan, "");
}
if (!res) {
snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%sqxd", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t");
pbx_exec(chan, app, meetmeopts, 1);
}
LOCAL_USER_REMOVE(u);
return -1;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app_page);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
}
char *description(void)
{
return (char *) tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,281 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* Author: Ben Miller <bgmiller@dccinc.com>
* With TONS of help from Mark!
*
* 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 ParkAndAnnounce application for Asterisk
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <sys/types.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/features.h"
#include "asterisk/options.h"
#include "asterisk/logger.h"
#include "asterisk/say.h"
#include "asterisk/lock.h"
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 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=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;
int outstate;
struct localuser *u;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "ParkAndAnnounce requires arguments: (announce:template|timeout|dial|[return_context])\n");
return -1;
}
LOCAL_USER_ADD(u);
l=strlen(data)+2;
orig_s=malloc(l);
if(!orig_s) {
ast_log(LOG_WARNING, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return -1;
}
s=orig_s;
strncpy(s,data,l);
template=strsep(&s,"|");
if(! template) {
ast_log(LOG_WARNING, "PARK: An announce template must be defined\n");
free(orig_s);
LOCAL_USER_REMOVE(u);
return -1;
}
if(s) {
timeout = atoi(strsep(&s, "|"));
timeout *= 1000;
}
dial=strsep(&s, "|");
if(!dial) {
ast_log(LOG_WARNING, "PARK: A dial resource must be specified i.e: Console/dsp or Zap/g1/5551212\n");
free(orig_s);
LOCAL_USER_REMOVE(u);
return -1;
} else {
dialtech=strsep(&dial, "/");
dialstr=dial;
ast_verbose( VERBOSE_PREFIX_3 "Dial Tech,String: (%s,%s)\n", dialtech,dialstr);
}
return_context = s;
if(return_context != NULL) {
/* set the return context. Code borrowed from the Goto builtin */
working = return_context;
context = strsep(&working, "|");
exten = strsep(&working, "|");
if(!exten) {
/* Only a priority in this one */
priority = context;
exten = NULL;
context = NULL;
} else {
priority = strsep(&working, "|");
if(!priority) {
/* Only an extension and priority in this one */
priority = exten;
exten = context;
context = NULL;
}
}
if(atoi(priority) < 0) {
ast_log(LOG_WARNING, "Priority '%s' must be a number > 0\n", priority);
free(orig_s);
LOCAL_USER_REMOVE(u);
return -1;
}
/* At this point we have a priority and maybe an extension and a context */
chan->priority = atoi(priority);
if(exten && strcasecmp(exten, "BYEXTENSION"))
strncpy(chan->exten, exten, sizeof(chan->exten)-1);
if(context)
strncpy(chan->context, context, sizeof(chan->context)-1);
} else { /* increment the priority by default*/
chan->priority++;
}
if(option_verbose > 2) {
ast_verbose( VERBOSE_PREFIX_3 "Return Context: (%s,%s,%d) ID: %s\n", chan->context,chan->exten, chan->priority, chan->cid.cid_num);
if(!ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
ast_verbose( VERBOSE_PREFIX_3 "Warning: Return Context Invalid, call will return to default|s\n");
}
}
/* we are using masq_park here to protect * from touching the channel once we park it. If the channel comes out of timeout
before we are done announcing and the channel is messed with, Kablooeee. So we use Masq to prevent this. */
ast_masq_park_call(chan, NULL, timeout, &lot);
res=-1;
ast_verbose( VERBOSE_PREFIX_3 "Call Parking Called, lot: %d, timeout: %d, context: %s\n", lot, timeout, return_context);
/* Now place the call to the extention */
memset(&oh, 0, sizeof(oh));
oh.parent_channel = chan;
dchan = __ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, dialstr,30000, &outstate, chan->cid.cid_num, chan->cid.cid_name, &oh);
if(dchan) {
if(dchan->_state == AST_STATE_UP) {
if(option_verbose > 3)
ast_verbose(VERBOSE_PREFIX_4 "Channel %s was answered.\n", dchan->name);
} else {
if(option_verbose > 3)
ast_verbose(VERBOSE_PREFIX_4 "Channel %s was never answered.\n", dchan->name);
ast_log(LOG_WARNING, "PARK: Channel %s was never answered for the announce.\n", dchan->name);
ast_hangup(dchan);
free(orig_s);
LOCAL_USER_REMOVE(u);
return -1;
}
} else {
ast_log(LOG_WARNING, "PARK: Unable to allocate announce channel.\n");
free(orig_s);
LOCAL_USER_REMOVE(u);
return -1;
}
ast_stopstream(dchan);
/* now we have the call placed and are ready to play stuff to it */
ast_verbose(VERBOSE_PREFIX_4 "Announce Template:%s\n", template);
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_verbose(VERBOSE_PREFIX_4 "Announce:%s\n", tmp[i]);
if(!strcmp(tmp[i], "PARKED")) {
ast_say_digits(dchan, lot, "", dchan->language);
} else {
dres = ast_streamfile(dchan, tmp[i], dchan->language);
if(!dres) {
dres = ast_waitstream(dchan, "");
} else {
ast_log(LOG_WARNING, "ast_streamfile of %s failed on %s\n", tmp[i], dchan->name);
dres = 0;
}
}
}
ast_stopstream(dchan);
ast_hangup(dchan);
free(orig_s);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
/* return ast_register_application(app, park_exec); */
return ast_register_application(app, parkandannounce_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,166 +1,65 @@
/*
* 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
*
* \ingroup applications
* the GNU General Public License
*/
#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 <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <pthread.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/utils.h"
#include "asterisk/options.h"
#include "asterisk/app.h"
static char *tdesc = "Sound File Playback Application";
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 pipe symbol. The 'skip'\n"
"option causes the playback of the message to be skipped if the channel\n"
"is not in the 'up' state (i.e. it hasn't been answered yet). If 'skip' is \n"
"specified, the application will return immediately should the channel not be\n"
"off hook. Otherwise, unless 'noanswer' is specified, the channel will\n"
"be answered before the sound is played. Not all channels support playing\n"
"messages while still on hook. If 'j' is specified, the application\n"
"will jump to priority n+101 if present when a file specified to be played\n"
"does not exist.\n"
"This application sets the following channel variable upon completion:\n"
" PLAYBACKSTATUS The status of the playback attempt as a text string, one of\n"
" SUCCESS | FAILED\n"
;
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int playback_exec(struct ast_channel *chan, void *data)
{
int res = 0, mres = 0;
int res = 0;
struct localuser *u;
char *tmp = NULL;
int option_skip=0;
int option_noanswer = 0;
char *front = NULL, *back = NULL;
int priority_jump = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(filenames);
AST_APP_ARG(options);
);
if (ast_strlen_zero(data)) {
if (!data) {
ast_log(LOG_WARNING, "Playback requires an argument (filename)\n");
return -1;
}
LOCAL_USER_ADD(u);
tmp = ast_strdupa(data);
if (!tmp) {
ast_log(LOG_ERROR, "Out of memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, tmp);
if (args.options) {
if (strcasestr(args.options, "skip"))
option_skip = 1;
if (strcasestr(args.options, "noanswer"))
option_noanswer = 1;
if (strchr(args.options, 'j'))
priority_jump = 1;
}
if (chan->_state != AST_STATE_UP) {
if (option_skip) {
/* At the user's option, skip if the line is not up */
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "SUCCESS");
LOCAL_USER_REMOVE(u);
return 0;
} else if (!option_noanswer)
/* Otherwise answer unless we're supposed to send this while on-hook */
res = ast_answer(chan);
}
if (chan->state != AST_STATE_UP)
res = ast_answer(chan);
if (!res) {
ast_stopstream(chan);
front = tmp;
while (!res && front) {
if ((back = strchr(front, '&'))) {
*back = '\0';
back++;
}
res = ast_streamfile(chan, front, chan->language);
if (!res) {
res = ast_waitstream(chan, "");
ast_stopstream(chan);
} else {
ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
if (priority_jump || option_priority_jumping)
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
res = 0;
mres = 1;
}
front = back;
}
res = ast_streamfile(chan, (char *)data, chan->language);
if (!res)
res = ast_waitstream(chan, "");
ast_stopstream(chan);
}
if (mres)
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "FAILED");
else
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "SUCCESS");
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
return ast_unregister_application(app);
}
int load_module(void)
{
return ast_register_application(app, playback_exec, synopsis, descrip);
return ast_register_application(app, playback_exec);
}
char *description(void)
@@ -174,8 +73,3 @@ int usecount(void)
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,262 +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 Block all calls without Caller*ID, require phone # to be entered
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/utils.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/callerid.h"
#include "asterisk/app.h"
#include "asterisk/config.h"
#define PRIV_CONFIG "privacy.conf"
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[|options]]]): If no Caller*ID \n"
"is sent, PrivacyManager answers the channel and asks the caller to\n"
"enter their phone number. The caller is given 3 attempts to do so.\n"
"The application does nothing if Caller*ID was received on the channel.\n"
" Configuration file privacy.conf contains two variables:\n"
" maxretries default 3 -maximum number of attempts the caller is allowed \n"
" to input a callerid.\n"
" minlength default 10 -minimum allowable digits in the input callerid number.\n"
"If you don't want to use the config file and have an i/o operation with\n"
"every call, you can also specify maxretries and minlength as application\n"
"parameters. Doing so supercedes any values set in privacy.conf.\n"
"The option string may contain the following character: \n"
" 'j' -- jump to n+101 priority after <maxretries> failed attempts to collect\n"
" the minlength number of digits.\n"
"The application sets the following channel variable upon completion: \n"
"PRIVACYMGRSTATUS The status of the privacy manager's attempt to collect \n"
" a phone number from the user. A text string that is either:\n"
" SUCCESS | FAILED \n"
;
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int privacy_exec (struct ast_channel *chan, void *data)
{
int res=0;
int retries;
int maxretries = 3;
int minlength = 10;
int x = 0;
char *s;
char phone[30];
struct localuser *u;
struct ast_config *cfg = NULL;
char *parse = NULL;
int priority_jump = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(maxretries);
AST_APP_ARG(minlength);
AST_APP_ARG(options);
);
LOCAL_USER_ADD (u);
if (!ast_strlen_zero(chan->cid.cid_num)) {
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) {
res = ast_answer(chan);
if (res) {
LOCAL_USER_REMOVE(u);
return -1;
}
}
if (!ast_strlen_zero((char *)data))
{
parse = ast_strdupa(data);
if (!parse) {
ast_log(LOG_ERROR, "Out of memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, parse);
if (args.maxretries) {
if (sscanf(args.maxretries, "%30d", &x) == 1)
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");
}
if (args.options)
if (strchr(args.options, 'j'))
priority_jump = 1;
}
if (!x)
{
/*Read in the config file*/
cfg = ast_config_load(PRIV_CONFIG);
if (cfg && (s = ast_variable_retrieve(cfg, "general", "maxretries"))) {
if (sscanf(s, "%30d", &x) == 1)
maxretries = x;
else
ast_log(LOG_WARNING, "Invalid max retries argument\n");
}
if (cfg && (s = ast_variable_retrieve(cfg, "general", "minlength"))) {
if (sscanf(s, "%30d", &x) == 1)
minlength = x;
else
ast_log(LOG_WARNING, "Invalid min length argument\n");
}
}
/*Play unidentified call*/
res = ast_safe_sleep(chan, 1000);
if (!res)
res = ast_streamfile(chan, "privacy-unident", chan->language);
if (!res)
res = ast_waitstream(chan, "");
/*Ask for 10 digit number, give 3 attempts*/
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, "#");
if (res < 0)
break;
/*Make sure we get at least digits*/
if (strlen(phone) >= minlength )
break;
else {
res = ast_streamfile(chan, "privacy-incorrect", chan->language);
if (!res)
res = ast_waitstream(chan, "");
}
}
/*Got a number, play sounds and send them on their way*/
if ((retries < maxretries) && res >= 0 ) {
res = ast_streamfile(chan, "privacy-thankyou", chan->language);
if (!res)
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);
if (option_verbose > 2) {
ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID to %s, callerpres to %d\n",phone,chan->cid.cid_pres);
}
pbx_builtin_setvar_helper(chan, "PRIVACYMGRSTATUS", "SUCCESS");
} else {
if (priority_jump || option_priority_jumping)
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
pbx_builtin_setvar_helper(chan, "PRIVACYMGRSTATUS", "FAILED");
}
if (cfg)
ast_config_destroy(cfg);
}
LOCAL_USER_REMOVE (u);
return 0;
}
int
unload_module (void)
{
int res;
res = ast_unregister_application (app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int
load_module (void)
{
return ast_register_application (app, privacy_exec, synopsis,
descrip);
}
char *
description (void)
{
return tdesc;
}
int
usecount (void)
{
int res;
STANDARD_USECOUNT (res);
return res;
}
char *
key ()
{
return ASTERISK_GPL_KEY;
}

File diff suppressed because it is too large Load Diff

View File

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

View File

@@ -1,235 +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 Trivial application to read a variable
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/app.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/options.h"
#include "asterisk/utils.h"
static char *tdesc = "Read Variable Application";
static char *app = "Read";
static char *synopsis = "Read a variable";
static char *descrip =
" Read(variable[|filename][|maxdigits][|option][|attempts][|timeout])\n\n"
"Reads a #-terminated string of digits a certain number of times from the\n"
"user in to the given variable.\n"
" filename -- file to play before reading digits.\n"
" maxdigits -- maximum acceptable number of digits. Stops reading after\n"
" maxdigits have been entered (without requiring the user to\n"
" press the '#' key).\n"
" Defaults to 0 - no limit - wait for the user press the '#' key.\n"
" Any value below 0 means the same. Max accepted value is 255.\n"
" option -- may be 'skip' to return immediately if the line is not up,\n"
" or 'noanswer' to read digits even if the line is not up.\n"
" attempts -- if greater than 1, that many attempts will be made in the \n"
" event no data is entered.\n"
" timeout -- if greater than 0, that value will override the default timeout.\n\n"
"Read should disconnect if the function fails or errors out.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
#define ast_next_data(instr,ptr,delim) if((ptr=strchr(instr,delim))) { *(ptr) = '\0' ; ptr++;}
static int read_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u;
char tmp[256];
char *timeout = NULL;
char *varname = NULL;
char *filename = NULL;
char *loops;
char *maxdigitstr=NULL;
char *options=NULL;
int option_skip = 0;
int option_noanswer = 0;
int maxdigits=255;
int tries = 1;
int to = 0;
int x = 0;
char *argcopy = NULL;
char *args[8];
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Read requires an argument (variable)\n");
return -1;
}
LOCAL_USER_ADD(u);
argcopy = ast_strdupa(data);
if (!argcopy) {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if (ast_app_separate_args(argcopy, '|', args, sizeof(args) / sizeof(args[0])) < 1) {
ast_log(LOG_WARNING, "Cannot Parse Arguments.\n");
LOCAL_USER_REMOVE(u);
return -1;
}
varname = args[x++];
filename = args[x++];
maxdigitstr = args[x++];
options = args[x++];
loops = args[x++];
timeout = args[x++];
if (options) {
if (!strcasecmp(options, "skip"))
option_skip = 1;
else if (!strcasecmp(options, "noanswer"))
option_noanswer = 1;
else {
if (strchr(options, 's'))
option_skip = 1;
if (strchr(options, 'n'))
option_noanswer = 1;
}
}
if(loops) {
tries = atoi(loops);
if(tries <= 0)
tries = 1;
}
if(timeout) {
to = atoi(timeout);
if (to <= 0)
to = 0;
else
to *= 1000;
}
if (ast_strlen_zero(filename))
filename = NULL;
if (maxdigitstr) {
maxdigits = atoi(maxdigitstr);
if ((maxdigits<1) || (maxdigits>255)) {
maxdigits = 255;
} else if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Accepting a maximum of %d digits.\n", maxdigits);
}
if (ast_strlen_zero(varname)) {
ast_log(LOG_WARNING, "Invalid! Usage: Read(variable[|filename][|maxdigits][|option][|attempts][|timeout])\n\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if (chan->_state != AST_STATE_UP) {
if (option_skip) {
/* At the user's option, skip if the line is not up */
pbx_builtin_setvar_helper(chan, varname, "\0");
LOCAL_USER_REMOVE(u);
return 0;
} else if (!option_noanswer) {
/* Otherwise answer unless we're supposed to read while on-hook */
res = ast_answer(chan);
}
}
if (!res) {
while(tries && !res) {
ast_stopstream(chan);
res = ast_app_getdata(chan, filename, tmp, maxdigits, to);
if (res > -1) {
pbx_builtin_setvar_helper(chan, varname, tmp);
if (!ast_strlen_zero(tmp)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "User entered '%s'\n", tmp);
tries = 0;
} else {
tries--;
if (option_verbose > 2) {
if (tries)
ast_verbose(VERBOSE_PREFIX_3 "User entered nothing, %d chance%s left\n", tries, (tries != 1) ? "s" : "");
else
ast_verbose(VERBOSE_PREFIX_3 "User entered nothing.\n");
}
}
res = 0;
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "User disconnected\n");
}
}
}
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, read_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,144 +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.
*
* \ingroup applications
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/app.h"
#include "asterisk/module.h"
static char *tdesc = "Stores output of file into a variable";
static char *app_readfile = "ReadFile";
static char *readfile_synopsis = "ReadFile(varname=file,length)";
static char *readfile_descrip =
"ReadFile(varname=file,length)\n"
" Varname - Result stored here.\n"
" File - The name of the file to read.\n"
" Length - Maximum number of characters to capture.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int readfile_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
char *s, *varname=NULL, *file=NULL, *length=NULL, *returnvar=NULL;
int len=0;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "ReadFile require an argument!\n");
return -1;
}
LOCAL_USER_ADD(u);
s = ast_strdupa(data);
if (!s) {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return -1;
}
varname = strsep(&s, "=");
file = strsep(&s, "|");
length = s;
if (!varname || !file) {
ast_log(LOG_ERROR, "No file or variable specified!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if (length) {
if ((sscanf(length, "%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);
free(returnvar);
}
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app_readfile);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app_readfile, readfile_exec, readfile_synopsis, readfile_descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

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

View File

@@ -1,406 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, 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.
*
* 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
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/dsp.h"
#include "asterisk/utils.h"
#include "asterisk/options.h"
#include "asterisk/app.h"
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"
"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"
" 'n' : do not answer, but record anyway if line not yet answered\n"
" 'q' : quiet (do not play a beep tone)\n"
" 's' : skip recording if the line is not yet answered\n"
" 't' : use alternate '*' terminator key instead of default '#'\n"
"\n"
"If filename contains '%d', these characters will be replaced with a number\n"
"incremented by one each time the file is recorded. A channel variable\n"
"named RECORDED_FILE will also be set, which contains the final filemname.\n\n"
"Use 'show file formats' to see the available formats on your system\n\n"
"User can press '#' to terminate the recording and continue to the next priority.\n\n"
"If the user should hangup during a recording, all data will be lost and the\n"
"application will teminate. \n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int record_exec(struct ast_channel *chan, void *data)
{
int res = 0;
int count = 0;
int percentflag = 0;
char *filename, *ext = NULL, *silstr, *maxstr, *options;
char *vdata, *p;
int i = 0;
char tmp[256];
struct ast_filestream *s = '\0';
struct localuser *u;
struct ast_frame *f = NULL;
struct ast_dsp *sildet = NULL; /* silence detector dsp */
int totalsilence = 0;
int dspsilence = 0;
int silence = 0; /* amount of silence to allow */
int gotsilence = 0; /* did we timeout for silence? */
int maxduration = 0; /* max duration of recording in milliseconds */
int gottimeout = 0; /* did we timeout for maxduration exceeded? */
int option_skip = 0;
int option_noanswer = 0;
int option_append = 0;
int terminator = '#';
int option_quiet = 0;
int rfmt = 0;
int flags;
int waitres;
struct ast_silence_generator *silgen = NULL;
/* The next few lines of code parse out the filename and header from the input string */
if (ast_strlen_zero(data)) { /* no data implies no filename or anything is present */
ast_log(LOG_WARNING, "Record requires an argument (filename)\n");
return -1;
}
LOCAL_USER_ADD(u);
/* Yay for strsep being easy */
vdata = ast_strdupa(data);
if (!vdata) {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return -1;
}
p = vdata;
filename = strsep(&p, "|");
silstr = strsep(&p, "|");
maxstr = strsep(&p, "|");
options = strsep(&p, "|");
if (filename) {
if (strstr(filename, "%d"))
percentflag = 1;
ext = strrchr(filename, '.'); /* to support filename with a . in the filename, not format */
if (!ext)
ext = strchr(filename, ':');
if (ext) {
*ext = '\0';
ext++;
}
}
if (!ext) {
ast_log(LOG_WARNING, "No extension specified to filename!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if (silstr) {
if ((sscanf(silstr, "%30d", &i) == 1) && (i > -1)) {
silence = i * 1000;
} else if (!ast_strlen_zero(silstr)) {
ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", silstr);
}
}
if (maxstr) {
if ((sscanf(maxstr, "%30d", &i) == 1) && (i > -1))
/* Convert duration to milliseconds */
maxduration = i * 1000;
else if (!ast_strlen_zero(maxstr))
ast_log(LOG_WARNING, "'%s' is not a valid maximum duration\n", maxstr);
}
if (options) {
/* Retain backwards compatibility with old style options */
if (!strcasecmp(options, "skip"))
option_skip = 1;
else if (!strcasecmp(options, "noanswer"))
option_noanswer = 1;
else {
if (strchr(options, 's'))
option_skip = 1;
if (strchr(options, 'n'))
option_noanswer = 1;
if (strchr(options, 'a'))
option_append = 1;
if (strchr(options, 't'))
terminator = '*';
if (strchr(options, 'q'))
option_quiet = 1;
}
}
/* done parsing */
/* these are to allow the use of the %d in the config file for a wild card of sort to
create a new file with the inputed name scheme */
if (percentflag) {
AST_DECLARE_APP_ARGS(fname,
AST_APP_ARG(piece)[100];
);
char *tmp2 = ast_strdupa(filename);
char countstring[15];
int i;
/* Separate each piece out by the format specifier */
/* AST_NONSTANDARD_APP_ARGS(fname, tmp2, '%'); */
fname.argc = ast_app_separate_args(tmp2, '%', fname.argv, (sizeof(fname) - sizeof(fname.argc)) / sizeof(fname.argv[0]));
do {
int tmplen;
/* First piece has no leading percent, so it's copied verbatim */
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);
}
count++;
} while ( ast_fileexists(tmp, ext, chan->language) != -1 );
pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
} else
strncpy(tmp, filename, sizeof(tmp)-1);
/* end of routine mentioned */
if (chan->_state != AST_STATE_UP) {
if (option_skip) {
/* At the user's option, skip if the line is not up */
LOCAL_USER_REMOVE(u);
return 0;
} else if (!option_noanswer) {
/* Otherwise answer unless we're supposed to record while on-hook */
res = ast_answer(chan);
}
}
if (res) {
ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
goto out;
}
if (!option_quiet) {
/* Some code to play a nice little beep to signify the start of the record operation */
res = ast_streamfile(chan, "beep", chan->language);
if (!res) {
res = ast_waitstream(chan, "");
} else {
ast_log(LOG_WARNING, "ast_streamfile failed on %s\n", chan->name);
}
ast_stopstream(chan);
}
/* The end of beep code. Now the recording starts */
if (silence > 0) {
rfmt = chan->readformat;
res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
LOCAL_USER_REMOVE(u);
return -1;
}
sildet = ast_dsp_new();
if (!sildet) {
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
LOCAL_USER_REMOVE(u);
return -1;
}
ast_dsp_set_threshold(sildet, 256);
}
flags = option_append ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
s = ast_writefile( tmp, ext, NULL, flags , 0, 0644);
if (!s) {
ast_log(LOG_WARNING, "Could not create file %s\n", filename);
goto out;
}
if (option_transmit_silence_during_record)
silgen = ast_channel_start_silence_generator(chan);
/* Request a video update */
ast_indicate(chan, AST_CONTROL_VIDUPDATE);
if (maxduration <= 0)
maxduration = -1;
while ((waitres = ast_waitfor(chan, maxduration)) > -1) {
if (maxduration > 0) {
if (waitres == 0) {
gottimeout = 1;
break;
}
maxduration = waitres;
}
f = ast_read(chan);
if (!f) {
res = -1;
break;
}
if (f->frametype == AST_FRAME_VOICE) {
res = ast_writestream(s, f);
if (res) {
ast_log(LOG_WARNING, "Problem writing frame\n");
ast_frfree(f);
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;
}
}
} else if (f->frametype == AST_FRAME_VIDEO) {
res = ast_writestream(s, f);
if (res) {
ast_log(LOG_WARNING, "Problem writing frame\n");
ast_frfree(f);
break;
}
} else if ((f->frametype == AST_FRAME_DTMF) &&
(f->subclass == terminator)) {
ast_frfree(f);
break;
}
ast_frfree(f);
}
if (!f) {
ast_log(LOG_DEBUG, "Got hangup\n");
res = -1;
}
if (gotsilence) {
ast_stream_rewind(s, silence-1000);
ast_truncstream(s);
} else if (!gottimeout) {
/* Strip off the last 1/4 second of it */
ast_stream_rewind(s, 250);
ast_truncstream(s);
}
ast_closestream(s);
if (silgen)
ast_channel_stop_silence_generator(chan, silgen);
out:
if ((silence > 0) && rfmt) {
res = ast_set_read_format(chan, rfmt);
if (res)
ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name);
if (sildet)
ast_dsp_free(sildet);
}
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, record_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

File diff suppressed because it is too large Load Diff

View File

@@ -1,163 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (c) 2003 Tilghman Lesher. All rights reserved.
*
* Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com>
*
* This code is released by the author with no restrictions on usage.
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
*/
/*! \file
*
* \brief SayUnixTime application
*
* \ingroup applications
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/say.h"
static char *tdesc = "Say time";
static char *app_sayunixtime = "SayUnixTime";
static char *app_datetime = "DateTime";
static char *sayunixtime_synopsis = "Says a specified time in a custom format";
static char *sayunixtime_descrip =
"SayUnixTime([unixtime][|[timezone][|format]])\n"
" unixtime: time, in seconds since Jan 1, 1970. May be negative.\n"
" defaults to now.\n"
" timezone: timezone, see /usr/share/zoneinfo for a list.\n"
" defaults to machine default.\n"
" format: a format the time is to be said in. See voicemail.conf.\n"
" defaults to \"ABdY 'digits/at' IMp\"\n";
static char *datetime_descrip =
"DateTime([unixtime][|[timezone][|format]])\n"
" unixtime: time, in seconds since Jan 1, 1970. May be negative.\n"
" defaults to now.\n"
" timezone: timezone, see /usr/share/zoneinfo for a list.\n"
" defaults to machine default.\n"
" format: a format the time is to be said in. See voicemail.conf.\n"
" defaults to \"ABdY 'digits/at' IMp\"\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int sayunixtime_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
char *s,*zone=NULL,*timec,*format;
time_t unixtime;
struct timeval tv;
LOCAL_USER_ADD(u);
tv = ast_tvnow();
unixtime = (time_t)tv.tv_sec;
if( !strcasecmp(chan->language, "da" ) ) {
format = "A dBY HMS";
} else if ( !strcasecmp(chan->language, "de" ) ) {
format = "A dBY HMS";
} else {
format = "ABdY 'digits/at' IMp";
}
if (data) {
s = data;
s = ast_strdupa(s);
if (s) {
timec = strsep(&s,"|");
if ((timec) && (*timec != '\0')) {
long timein;
if (sscanf(timec,"%30ld",&timein) == 1) {
unixtime = (time_t)timein;
}
}
if (s) {
zone = strsep(&s,"|");
if (zone && (*zone == '\0'))
zone = NULL;
if (s) {
format = s;
}
}
} else {
ast_log(LOG_ERROR, "Out of memory error\n");
}
}
if (chan->_state != AST_STATE_UP) {
res = ast_answer(chan);
}
if (!res)
res = ast_say_date_with_format(chan, unixtime, AST_DIGIT_ANY, chan->language, format, zone);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app_sayunixtime);
res |= ast_unregister_application(app_datetime);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
res = ast_register_application(app_sayunixtime, sayunixtime_exec, sayunixtime_synopsis, sayunixtime_descrip);
res |= ast_register_application(app_datetime, sayunixtime_exec, sayunixtime_synopsis, datetime_descrip);
return res;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,129 +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 App to send DTMF digits
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/options.h"
#include "asterisk/utils.h"
#include "asterisk/app.h"
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]): Sends DTMF digits on a channel. \n"
" Accepted digits: 0-9, *#abcd, w (.5s pause)\n"
" The application will either pass the assigned digits or terminate if it\n"
" encounters an error.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int senddtmf_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u;
char *digits = NULL, *to = NULL;
int timeout = 250;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "SendDTMF requires an argument (digits or *#aAbBcCdD)\n");
return 0;
}
LOCAL_USER_ADD(u);
digits = ast_strdupa(data);
if (!digits) {
ast_log(LOG_ERROR, "Out of Memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if ((to = strchr(digits,'|'))) {
*to = '\0';
to++;
timeout = atoi(to);
}
if(timeout <= 0)
timeout = 250;
res = ast_dtmf_stream(chan,NULL,digits,timeout);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, senddtmf_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,156 +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 App to transmit a text message
*
* Requires support of sending text messages from channel driver
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/options.h"
#include "asterisk/app.h"
static const char *tdesc = "Send Text Applications";
static const char *app = "SendText";
static const char *synopsis = "Send a Text Message";
static const char *descrip =
" SendText(text[|options]): Sends text to current channel (callee).\n"
"Result of transmission will be stored in the SENDTEXTSTATUS\n"
"channel variable:\n"
" SUCCESS Transmission succeeded\n"
" FAILURE Transmission failed\n"
" UNSUPPORTED Text transmission not supported by channel\n"
"\n"
"At this moment, text is supposed to be 7 bit ASCII in most channels.\n"
"The option string many contain the following character:\n"
"'j' -- jump to n+101 priority if the channel doesn't support\n"
" text transport\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int sendtext_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u;
char *status = "UNSUPPORTED";
char *parse = NULL;
int priority_jump = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(text);
AST_APP_ARG(options);
);
LOCAL_USER_ADD(u);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "SendText requires an argument (text[|options])\n");
LOCAL_USER_REMOVE(u);
return -1;
} else {
parse = ast_strdupa(data);
if (!parse) {
ast_log(LOG_ERROR, "Out of memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
}
AST_STANDARD_APP_ARGS(args, parse);
if (args.options) {
if (strchr(args.options, 'j'))
priority_jump = 1;
}
ast_mutex_lock(&chan->lock);
if (!chan->tech->send_text) {
ast_mutex_unlock(&chan->lock);
/* Does not support transport */
if (priority_jump || option_priority_jumping)
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
LOCAL_USER_REMOVE(u);
return 0;
}
status = "FAILURE";
ast_mutex_unlock(&chan->lock);
res = ast_sendtext(chan, args.text);
if (!res)
status = "SUCCESS";
pbx_builtin_setvar_helper(chan, "SENDTEXTSTATUS", status);
LOCAL_USER_REMOVE(u);
return 0;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, sendtext_exec, synopsis, descrip);
}
char *description(void)
{
return (char *) tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

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@@ -1,178 +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 App to set callerid
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/callerid.h"
static char *app2 = "SetCallerPres";
static char *synopsis2 = "Set CallerID Presentation";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static char *descrip2 =
" SetCallerPres(presentation): Set Caller*ID presentation on a call.\n"
" Valid presentations are:\n"
"\n"
" allowed_not_screened : Presentation Allowed, Not Screened\n"
" allowed_passed_screen : Presentation Allowed, Passed Screen\n"
" allowed_failed_screen : Presentation Allowed, Failed Screen\n"
" allowed : Presentation Allowed, Network Number\n"
" prohib_not_screened : Presentation Prohibited, Not Screened\n"
" prohib_passed_screen : Presentation Prohibited, Passed Screen\n"
" prohib_failed_screen : Presentation Prohibited, Failed Screen\n"
" prohib : Presentation Prohibited, Network Number\n"
" unavailable : Number Unavailable\n"
"\n"
;
static int setcallerid_pres_exec(struct ast_channel *chan, void *data)
{
struct localuser *u;
int pres = -1;
LOCAL_USER_ADD(u);
pres = ast_parse_caller_presentation(data);
if (pres < 0) {
ast_log(LOG_WARNING, "'%s' is not a valid presentation (see 'show application SetCallerPres')\n",
(char *) data);
LOCAL_USER_REMOVE(u);
return 0;
}
chan->cid.cid_pres = pres;
LOCAL_USER_REMOVE(u);
return 0;
}
static char *tdesc = "Set CallerID Application";
static char *app = "SetCallerID";
static char *synopsis = "Set CallerID";
static char *descrip =
" SetCallerID(clid[|a]): Set Caller*ID on a call to a new\n"
"value. Sets ANI as well if a flag is used. \n";
static int setcallerid_exec(struct ast_channel *chan, void *data)
{
int res = 0;
char *tmp = NULL;
char name[256];
char num[256];
struct localuser *u;
char *opt;
int anitoo = 0;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "SetCallerID requires an argument!\n");
return 0;
}
LOCAL_USER_ADD(u);
tmp = ast_strdupa(data);
if (!tmp) {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return -1;
}
opt = strchr(tmp, '|');
if (opt) {
*opt = '\0';
opt++;
if (*opt == 'a')
anitoo = 1;
}
ast_callerid_split(tmp, name, sizeof(name), num, sizeof(num));
ast_set_callerid(chan, num, name, anitoo ? num : NULL);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app2);
res |= ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
res = ast_register_application(app2, setcallerid_pres_exec, synopsis2, descrip2);
res |= ast_register_application(app, setcallerid_exec, synopsis, descrip);
return res;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

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@@ -1,179 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Justin Huff <jjhuff@mspin.net>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Applictions connected with CDR engine
*
* \ingroup applications
*/
#include <sys/types.h>
#include <stdlib.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/channel.h"
#include "asterisk/cdr.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/logger.h"
#include "asterisk/config.h"
#include "asterisk/manager.h"
#include "asterisk/utils.h"
static char *tdesc = "CDR user field apps";
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";
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";
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_get_channel_by_name_locked(channel);
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;
res = ast_unregister_application(setcdruserfield_app);
res |= ast_unregister_application(appendcdruserfield_app);
res |= ast_manager_unregister("SetCDRUserField");
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
res = ast_register_application(setcdruserfield_app, setcdruserfield_exec, setcdruserfield_synopsis, setcdruserfield_descrip);
res |= ast_register_application(appendcdruserfield_app, appendcdruserfield_exec, appendcdruserfield_synopsis, appendcdruserfield_descrip);
res |= 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;
}

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@@ -1,132 +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 App to set callerid
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/callerid.h"
#include "asterisk/utils.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. \n"
"SetCIDName has been deprecated in favor of the function\n"
"CALLERID(name)\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int setcallerid_exec(struct ast_channel *chan, void *data)
{
char *tmp = NULL;
struct localuser *u;
char *opt;
static int deprecation_warning = 0;
if (!deprecation_warning) {
ast_log(LOG_WARNING, "SetCIDName is deprecated, please use Set(CALLERID(name)=value) instead.\n");
deprecation_warning = 1;
}
if (ast_strlen_zero(data)) {
ast_log(LOG_ERROR, "SetCIDName requires an argument!\n");
return 0;
}
LOCAL_USER_ADD(u);
tmp = ast_strdupa(data);
if (!tmp) {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return -1;
}
opt = strchr(tmp, '|');
if (opt) {
*opt = '\0';
}
ast_set_callerid(chan, NULL, tmp, NULL);
LOCAL_USER_REMOVE(u);
return 0;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, 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;
}

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@@ -1,131 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
* Oliver Daudey <traveler@xs4all.nl>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief App to set callerid number
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/callerid.h"
#include "asterisk/utils.h"
static char *tdesc = "Set CallerID Number";
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.\n"
"SetCIDNum has been deprecated in favor of the function\n"
"CALLERID(number)\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int setcallerid_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u;
char *opt;
int anitoo = 0;
char *tmp = NULL;
static int deprecation_warning = 0;
LOCAL_USER_ADD(u);
if (!deprecation_warning) {
ast_log(LOG_WARNING, "SetCIDNum is deprecated, please use Set(CALLERID(number)=value) instead.\n");
deprecation_warning = 1;
}
if (data)
tmp = ast_strdupa(data);
else
tmp = "";
opt = strchr(tmp, '|');
if (opt) {
*opt = '\0';
opt++;
if (*opt == 'a')
anitoo = 1;
}
ast_set_callerid(chan, tmp, NULL, anitoo ? tmp : NULL);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
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;
}

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@@ -1,132 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
* Oliver Daudey <traveler@xs4all.nl>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief App to set rdnis
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/callerid.h"
#include "asterisk/utils.h"
static char *tdesc = "Set RDNIS Number";
static char *app = "SetRDNIS";
static char *synopsis = "Set RDNIS Number";
static char *descrip =
" SetRDNIS(cnum): Set RDNIS Number on a call to a new\n"
"value.\n"
"SetRDNIS has been deprecated in favor of the function\n"
"CALLERID(rdnis)\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int setrdnis_exec(struct ast_channel *chan, void *data)
{
struct localuser *u;
char *opt, *n, *l;
char *tmp = NULL;
static int deprecation_warning = 0;
LOCAL_USER_ADD(u);
if (!deprecation_warning) {
ast_log(LOG_WARNING, "SetRDNIS is deprecated, please use Set(CALLERID(rdnis)=value) instead.\n");
deprecation_warning = 1;
}
if (data)
tmp = ast_strdupa(data);
else
tmp = "";
opt = strchr(tmp, '|');
if (opt)
*opt = '\0';
n = l = NULL;
ast_callerid_parse(tmp, &n, &l);
if (l) {
ast_shrink_phone_number(l);
ast_mutex_lock(&chan->lock);
if (chan->cid.cid_rdnis)
free(chan->cid.cid_rdnis);
chan->cid.cid_rdnis = (l[0]) ? strdup(l) : NULL;
ast_mutex_unlock(&chan->lock);
}
LOCAL_USER_REMOVE(u);
return 0;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, setrdnis_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

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@@ -1,145 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2005, Frank Sautter, levigo holding gmbh, www.levigo.de
*
* Frank Sautter - asterisk+at+sautter+dot+com
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief App to set the ISDN Transfer Capability
*
* \ingroup applications
*/
#include <string.h>
#include <stdlib.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/options.h"
#include "asterisk/transcap.h"
static char *app = "SetTransferCapability";
static char *synopsis = "Set ISDN Transfer Capability";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static struct { int val; char *name; } transcaps[] = {
{ AST_TRANS_CAP_SPEECH, "SPEECH" },
{ AST_TRANS_CAP_DIGITAL, "DIGITAL" },
{ AST_TRANS_CAP_RESTRICTED_DIGITAL, "RESTRICTED_DIGITAL" },
{ AST_TRANS_CAP_3_1K_AUDIO, "3K1AUDIO" },
{ AST_TRANS_CAP_DIGITAL_W_TONES, "DIGITAL_W_TONES" },
{ AST_TRANS_CAP_VIDEO, "VIDEO" },
};
static char *descrip =
" SetTransferCapability(transfercapability): Set the ISDN Transfer \n"
"Capability of a call to a new value.\n"
"Valid Transfer Capabilities are:\n"
"\n"
" SPEECH : 0x00 - Speech (default, voice calls)\n"
" DIGITAL : 0x08 - Unrestricted digital information (data calls)\n"
" RESTRICTED_DIGITAL : 0x09 - Restricted digital information\n"
" 3K1AUDIO : 0x10 - 3.1kHz Audio (fax calls)\n"
" DIGITAL_W_TONES : 0x11 - Unrestricted digital information with tones/announcements\n"
" VIDEO : 0x18 - Video:\n"
"\n"
;
static int settransfercapability_exec(struct ast_channel *chan, void *data)
{
char *tmp = NULL;
struct localuser *u;
int x;
char *opts;
int transfercapability = -1;
LOCAL_USER_ADD(u);
if (data)
tmp = ast_strdupa(data);
else
tmp = "";
opts = strchr(tmp, '|');
if (opts)
*opts = '\0';
for (x = 0; x < (sizeof(transcaps) / sizeof(transcaps[0])); x++) {
if (!strcasecmp(transcaps[x].name, tmp)) {
transfercapability = transcaps[x].val;
break;
}
}
if (transfercapability < 0) {
ast_log(LOG_WARNING, "'%s' is not a valid transfer capability (see 'show application SetTransferCapability')\n", tmp);
LOCAL_USER_REMOVE(u);
return 0;
}
chan->transfercapability = (unsigned short)transfercapability;
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Setting transfer capability to: 0x%.2x - %s.\n", transfercapability, tmp);
LOCAL_USER_REMOVE(u);
return 0;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, settransfercapability_exec, synopsis, descrip);
}
char *description(void)
{
return synopsis;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,134 +1,60 @@
/*
* Asterisk -- An open source telephony toolkit.
* Asterisk -- A telephony toolkit for Linux.
*
* Copyright (C) <Year>, <Your Name Here>
* Skeleton application
*
* Copyright (C) 1999, Mark Spencer
*
* <Your Name Here> <<Your Email Here>>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
* 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 Skeleton application
*
* This is a skeleton for development of an Asterisk application
* \ingroup applications
*/
#include <stdio.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"
#include <pthread.h>
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/lock.h"
#include "asterisk/app.h"
static char *tdesc = "Trivial skeleton Application";
static char *app = "Skel";
static char *synopsis =
"Skeleton application.";
static char *descrip = "This application is a template to build other applications from.\n"
" It shows you the basic structure to create your own Asterisk applications.\n";
#define OPTION_A (1 << 0) /* Option A */
#define OPTION_B (1 << 1) /* Option B(n) */
#define OPTION_C (1 << 2) /* Option C(str) */
#define OPTION_NULL (1 << 3) /* Dummy Termination */
AST_APP_OPTIONS(app_opts,{
['a'] = { OPTION_A },
['b'] = { OPTION_B, 1 },
['c'] = { OPTION_C, 2 }
});
static char *app = "skel";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int app_exec(struct ast_channel *chan, void *data)
static int skel_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct ast_flags flags;
int res=0;
struct localuser *u;
char *options=NULL;
char *dummy = NULL;
char *args;
int argc = 0;
char *opts[2];
char *argv[2];
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "%s requires an argument (dummy|[options])\n",app);
if (!data) {
ast_log(LOG_WARNING, "skel requires an argument (filename)\n");
return -1;
}
LOCAL_USER_ADD(u);
/* Do our thing here */
/* We need to make a copy of the input string if we are going to modify it! */
args = ast_strdupa(data);
if (!args) {
ast_log(LOG_ERROR, "Out of memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if ((argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0])))) {
dummy = argv[0];
options = argv[1];
ast_app_parse_options(app_opts, &flags, opts, options);
}
if (!ast_strlen_zero(dummy))
ast_log(LOG_NOTICE, "Dummy value is : %s\n", dummy);
if (ast_test_flag(&flags, OPTION_A))
ast_log(LOG_NOTICE, "Option A is set\n");
if (ast_test_flag(&flags, OPTION_B))
ast_log(LOG_NOTICE,"Option B is set with : %s\n", opts[0] ? opts[0] : "<unspecified>");
if (ast_test_flag(&flags, OPTION_C))
ast_log(LOG_NOTICE,"Option C is set with : %s\n", opts[1] ? opts[1] : "<unspecified>");
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
return ast_unregister_application(app);
}
int load_module(void)
{
return ast_register_application(app, app_exec, synopsis, descrip);
return ast_register_application(app, skel_exec);
}
char *description(void)
@@ -142,8 +68,3 @@ int usecount(void)
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

File diff suppressed because it is too large Load Diff

View File

@@ -1,139 +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 SoftHangup application
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <sys/types.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/lock.h"
static char *synopsis = "Soft Hangup Application";
static char *tdesc = "Hangs up the requested channel";
static char *desc = " SoftHangup(Technology/resource|options)\n"
"Hangs up the requested channel. If there are no channels to hangup,\n"
"the application will report it.\n"
"- 'options' may contain the following letter:\n"
" 'a' : hang up all channels on a specified device instead of a single resource\n";
static char *app = "SoftHangup";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int softhangup_exec(struct ast_channel *chan, void *data)
{
struct localuser *u;
struct ast_channel *c=NULL;
char *options, *cut, *cdata, *match;
char name[AST_CHANNEL_NAME] = "";
int all = 0;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "SoftHangup requires an argument (Technology/resource)\n");
return 0;
}
LOCAL_USER_ADD(u);
cdata = ast_strdupa(data);
match = strsep(&cdata, "|");
options = strsep(&cdata, "|");
all = options && strchr(options,'a');
c = ast_channel_walk_locked(NULL);
while (c) {
strncpy(name, c->name, sizeof(name)-1);
ast_mutex_unlock(&c->lock);
/* XXX watch out, i think it is wrong to access c-> after unlocking! */
if (all) {
/* CAPI is set up like CAPI[foo/bar]/clcnt */
if (!strcmp(c->type,"CAPI"))
cut = strrchr(name,'/');
/* Basically everything else is Foo/Bar-Z */
else
cut = strchr(name,'-');
/* Get rid of what we've cut */
if (cut)
*cut = 0;
}
if (!strcasecmp(name, match)) {
ast_log(LOG_WARNING, "Soft hanging %s up.\n",c->name);
ast_softhangup(c, AST_SOFTHANGUP_EXPLICIT);
if(!all)
break;
}
c = ast_channel_walk_locked(c);
}
LOCAL_USER_REMOVE(u);
return 0;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, softhangup_exec, synopsis, desc);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,572 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* 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.
*/
/*! \file
*
* \brief Connect to PostgreSQL
*
* \ingroup applications
*/
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <sys/types.h>
#include <stdio.h>
#include <unistd.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/linkedlists.h"
#include "asterisk/chanvars.h"
#include "asterisk/lock.h"
#include "libpq-fe.h"
static char *tdesc = "Simple PostgreSQL Interface";
static char *app = "PGSQL";
static char *synopsis = "Do several SQLy things";
static char *descrip =
"PGSQL(): Do several SQLy things\n"
"Syntax:\n"
" PGSQL(Connect var option-string)\n"
" Connects to a database. Option string contains standard PostgreSQL\n"
" parameters like host=, dbname=, user=. Connection identifer returned\n"
" in ${var}\n"
" PGSQL(Query var ${connection_identifier} query-string)\n"
" Executes standard SQL query contained in query-string using established\n"
" connection identified by ${connection_identifier}. Reseult of query is\n"
" is stored in ${var}.\n"
" PGSQL(Fetch statusvar ${result_identifier} var1 var2 ... varn)\n"
" Fetches a single row from a result set contained in ${result_identifier}.\n"
" Assigns returned fields to ${var1} ... ${varn}. ${statusvar} is set TRUE\n"
" if additional rows exist in reseult set.\n"
" PGSQL(Clear ${result_identifier})\n"
" Frees memory and datastructures associated with result set.\n"
" PGSQL(Disconnect ${connection_identifier})\n"
" Disconnects from named connection to PostgreSQL.\n" ;
/*
Syntax of SQL commands :
Connect var option-string
Connects to a database using the option-string and stores the
connection identifier in ${var}
Query var ${connection_identifier} query-string
Submits query-string to database backend and stores the result
identifier in ${var}
Fetch statusvar ${result_identifier} var1 var2 var3 ... varn
Fetches a row from the query and stores end-of-table status in
${statusvar} and columns in ${var1}..${varn}
Clear ${result_identifier}
Clears data structures associated with ${result_identifier}
Disconnect ${connection_identifier}
Disconnects from named connection
EXAMPLES OF USE :
exten => s,2,PGSQL(Connect connid host=localhost user=asterisk dbname=credit)
exten => s,3,PGSQL(Query resultid ${connid} SELECT username,credit FROM credit WHERE callerid=${CALLERIDNUM})
exten => s,4,PGSQL(Fetch fetchid ${resultid} datavar1 datavar2)
exten => s,5,GotoIf(${fetchid}?6:8)
exten => s,6,Festival("User ${datavar1} currently has credit balance of ${datavar2} dollars.")
exten => s,7,Goto(s,4)
exten => s,8,PGSQL(Clear ${resultid})
exten => s,9,PGSQL(Disconnect ${connid})
*/
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
#define AST_PGSQL_ID_DUMMY 0
#define AST_PGSQL_ID_CONNID 1
#define AST_PGSQL_ID_RESID 2
#define AST_PGSQL_ID_FETCHID 3
struct ast_PGSQL_id {
int identifier_type; /* 0=dummy, 1=connid, 2=resultid */
int identifier;
void *data;
AST_LIST_ENTRY(ast_PGSQL_id) entries;
} *ast_PGSQL_id;
static AST_LIST_HEAD_STATIC(PGSQLidshead,ast_PGSQL_id);
static void *find_identifier(int identifier,int identifier_type) {
struct PGSQLidshead *headp;
struct ast_PGSQL_id *i;
void *res=NULL;
int found=0;
headp=&PGSQLidshead;
if (AST_LIST_LOCK(headp)) {
ast_log(LOG_WARNING,"Unable to lock identifiers list\n");
} else {
AST_LIST_TRAVERSE(headp,i,entries) {
if ((i->identifier==identifier) && (i->identifier_type==identifier_type)) {
found=1;
res=i->data;
break;
}
}
if (!found) {
ast_log(LOG_WARNING,"Identifier %d, identifier_type %d not found in identifier list\n",identifier,identifier_type);
}
AST_LIST_UNLOCK(headp);
}
return(res);
}
static int add_identifier(int identifier_type,void *data) {
struct ast_PGSQL_id *i,*j;
struct PGSQLidshead *headp;
int maxidentifier=0;
headp=&PGSQLidshead;
i=NULL;
j=NULL;
if (AST_LIST_LOCK(headp)) {
ast_log(LOG_WARNING,"Unable to lock identifiers list\n");
return(-1);
} else {
i=malloc(sizeof(struct ast_PGSQL_id));
AST_LIST_TRAVERSE(headp,j,entries) {
if (j->identifier>maxidentifier) {
maxidentifier=j->identifier;
}
}
i->identifier=maxidentifier+1;
i->identifier_type=identifier_type;
i->data=data;
AST_LIST_INSERT_HEAD(headp,i,entries);
AST_LIST_UNLOCK(headp);
}
return(i->identifier);
}
static int del_identifier(int identifier,int identifier_type) {
struct ast_PGSQL_id *i;
struct PGSQLidshead *headp;
int found=0;
headp=&PGSQLidshead;
if (AST_LIST_LOCK(headp)) {
ast_log(LOG_WARNING,"Unable to lock identifiers list\n");
} else {
AST_LIST_TRAVERSE(headp,i,entries) {
if ((i->identifier==identifier) &&
(i->identifier_type==identifier_type)) {
AST_LIST_REMOVE(headp,i,entries);
free(i);
found=1;
break;
}
}
AST_LIST_UNLOCK(headp);
}
if (found==0) {
ast_log(LOG_WARNING,"Could not find identifier %d, identifier_type %d in list to delete\n",identifier,identifier_type);
return(-1);
} else {
return(0);
}
}
static int aPGSQL_connect(struct ast_channel *chan, void *data) {
char *s1;
char s[100] = "";
char *optionstring;
char *var;
int l;
int res;
PGconn *karoto;
int id;
char *stringp=NULL;
res=0;
l=strlen(data)+2;
s1=malloc(l);
strncpy(s1, data, l -1);
stringp=s1;
strsep(&stringp," "); /* eat the first token, we already know it :P */
var=strsep(&stringp," ");
optionstring=strsep(&stringp,"\n");
karoto = PQconnectdb(optionstring);
if (PQstatus(karoto) == CONNECTION_BAD) {
ast_log(LOG_WARNING,"Connection to database using '%s' failed. postgress reports : %s\n", optionstring,
PQerrorMessage(karoto));
res=-1;
} else {
ast_log(LOG_WARNING,"adding identifier\n");
id=add_identifier(AST_PGSQL_ID_CONNID,karoto);
snprintf(s, sizeof(s), "%d", id);
pbx_builtin_setvar_helper(chan,var,s);
}
free(s1);
return res;
}
static int aPGSQL_query(struct ast_channel *chan, void *data) {
char *s1,*s2,*s3,*s4;
char s[100] = "";
char *querystring;
char *var;
int l;
int res,nres;
PGconn *karoto;
PGresult *PGSQLres;
int id,id1;
char *stringp=NULL;
res=0;
l=strlen(data)+2;
s1=malloc(l);
s2=malloc(l);
strncpy(s1, data, l - 1);
stringp=s1;
strsep(&stringp," "); /* eat the first token, we already know it :P */
s3=strsep(&stringp," ");
while (1) { /* ugly trick to make branches with break; */
var=s3;
s4=strsep(&stringp," ");
id=atoi(s4);
querystring=strsep(&stringp,"\n");
if ((karoto=find_identifier(id,AST_PGSQL_ID_CONNID))==NULL) {
ast_log(LOG_WARNING,"Invalid connection identifier %d passed in aPGSQL_query\n",id);
res=-1;
break;
}
PGSQLres=PQexec(karoto,querystring);
if (PGSQLres==NULL) {
ast_log(LOG_WARNING,"aPGSQL_query: Connection Error (connection identifier = %d, error message : %s)\n",id,PQerrorMessage(karoto));
res=-1;
break;
}
if (PQresultStatus(PGSQLres) == PGRES_BAD_RESPONSE ||
PQresultStatus(PGSQLres) == PGRES_NONFATAL_ERROR ||
PQresultStatus(PGSQLres) == PGRES_FATAL_ERROR) {
ast_log(LOG_WARNING,"aPGSQL_query: Query Error (connection identifier : %d, error message : %s)\n",id,PQcmdStatus(PGSQLres));
res=-1;
break;
}
nres=PQnfields(PGSQLres);
id1=add_identifier(AST_PGSQL_ID_RESID,PGSQLres);
snprintf(s, sizeof(s), "%d", id1);
pbx_builtin_setvar_helper(chan,var,s);
break;
}
free(s1);
free(s2);
return(res);
}
static int aPGSQL_fetch(struct ast_channel *chan, void *data) {
char *s1,*s2,*fetchid_var,*s4,*s5,*s6,*s7;
char s[100];
char *var;
int l;
int res;
PGresult *PGSQLres;
int id,id1,i,j,fnd;
int *lalares=NULL;
int nres;
struct ast_var_t *variables;
struct varshead *headp;
char *stringp=NULL;
headp=&chan->varshead;
res=0;
l=strlen(data)+2;
s7=NULL;
s1=malloc(l);
s2=malloc(l);
strncpy(s1, data, l - 1);
stringp=s1;
strsep(&stringp," "); /* eat the first token, we already know it :P */
fetchid_var=strsep(&stringp," ");
while (1) { /* ugly trick to make branches with break; */
var=fetchid_var; /* fetchid */
fnd=0;
AST_LIST_TRAVERSE(headp,variables,entries) {
if (strncasecmp(ast_var_name(variables),fetchid_var,strlen(fetchid_var))==0) {
s7=ast_var_value(variables);
fnd=1;
break;
}
}
if (fnd==0) {
s7="0";
pbx_builtin_setvar_helper(chan,fetchid_var,s7);
}
s4=strsep(&stringp," ");
id=atoi(s4); /* resultid */
if ((PGSQLres=find_identifier(id,AST_PGSQL_ID_RESID))==NULL) {
ast_log(LOG_WARNING,"Invalid result identifier %d passed in aPGSQL_fetch\n",id);
res=-1;
break;
}
id=atoi(s7); /*fetchid */
if ((lalares=find_identifier(id,AST_PGSQL_ID_FETCHID))==NULL) {
i=0; /* fetching the very first row */
} else {
i=*lalares;
free(lalares);
del_identifier(id,AST_PGSQL_ID_FETCHID); /* will re-add it a bit later */
}
if (i<PQntuples(PGSQLres)) {
nres=PQnfields(PGSQLres);
ast_log(LOG_WARNING,"ast_PGSQL_fetch : nres = %d i = %d ;\n",nres,i);
for (j=0;j<nres;j++) {
s5=strsep(&stringp," ");
if (s5==NULL) {
ast_log(LOG_WARNING,"ast_PGSQL_fetch : More tuples (%d) than variables (%d)\n",nres,j);
break;
}
s6=PQgetvalue(PGSQLres,i,j);
if (s6==NULL) {
ast_log(LOG_WARNING,"PWgetvalue(res,%d,%d) returned NULL in ast_PGSQL_fetch\n",i,j);
break;
}
ast_log(LOG_WARNING,"===setting variable '%s' to '%s'\n",s5,s6);
pbx_builtin_setvar_helper(chan,s5,s6);
}
lalares=malloc(sizeof(int));
*lalares = ++i; /* advance to the next row */
id1 = add_identifier(AST_PGSQL_ID_FETCHID,lalares);
} else {
ast_log(LOG_WARNING,"ast_PGSQL_fetch : EOF\n");
id1 = 0; /* no more rows */
}
snprintf(s, sizeof(s), "%d", id1);
ast_log(LOG_WARNING,"Setting var '%s' to value '%s'\n",fetchid_var,s);
pbx_builtin_setvar_helper(chan,fetchid_var,s);
break;
}
free(s1);
free(s2);
return(res);
}
static int aPGSQL_reset(struct ast_channel *chan, void *data) {
char *s1,*s3;
int l;
PGconn *karoto;
int id;
char *stringp=NULL;
l=strlen(data)+2;
s1=malloc(l);
strncpy(s1, data, l - 1);
stringp=s1;
strsep(&stringp," "); /* eat the first token, we already know it :P */
s3=strsep(&stringp," ");
id=atoi(s3);
if ((karoto=find_identifier(id,AST_PGSQL_ID_CONNID))==NULL) {
ast_log(LOG_WARNING,"Invalid connection identifier %d passed in aPGSQL_reset\n",id);
} else {
PQreset(karoto);
}
free(s1);
return(0);
}
static int aPGSQL_clear(struct ast_channel *chan, void *data) {
char *s1,*s3;
int l;
PGresult *karoto;
int id;
char *stringp=NULL;
l=strlen(data)+2;
s1=malloc(l);
strncpy(s1, data, l - 1);
stringp=s1;
strsep(&stringp," "); /* eat the first token, we already know it :P */
s3=strsep(&stringp," ");
id=atoi(s3);
if ((karoto=find_identifier(id,AST_PGSQL_ID_RESID))==NULL) {
ast_log(LOG_WARNING,"Invalid result identifier %d passed in aPGSQL_clear\n",id);
} else {
PQclear(karoto);
del_identifier(id,AST_PGSQL_ID_RESID);
}
free(s1);
return(0);
}
static int aPGSQL_disconnect(struct ast_channel *chan, void *data) {
char *s1,*s3;
int l;
PGconn *karoto;
int id;
char *stringp=NULL;
l=strlen(data)+2;
s1=malloc(l);
strncpy(s1, data, l - 1);
stringp=s1;
strsep(&stringp," "); /* eat the first token, we already know it :P */
s3=strsep(&stringp," ");
id=atoi(s3);
if ((karoto=find_identifier(id,AST_PGSQL_ID_CONNID))==NULL) {
ast_log(LOG_WARNING,"Invalid connection identifier %d passed in aPGSQL_disconnect\n",id);
} else {
PQfinish(karoto);
del_identifier(id,AST_PGSQL_ID_CONNID);
}
free(s1);
return(0);
}
static int aPGSQL_debug(struct ast_channel *chan, void *data) {
ast_log(LOG_WARNING,"Debug : %s\n",(char *)data);
return(0);
}
static int PGSQL_exec(struct ast_channel *chan, void *data)
{
struct localuser *u;
int result;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "APP_PGSQL requires an argument (see manual)\n");
return -1;
}
LOCAL_USER_ADD(u);
result=0;
if (strncasecmp("connect",data,strlen("connect"))==0) {
result=(aPGSQL_connect(chan,data));
} else if (strncasecmp("query",data,strlen("query"))==0) {
result=(aPGSQL_query(chan,data));
} else if (strncasecmp("fetch",data,strlen("fetch"))==0) {
result=(aPGSQL_fetch(chan,data));
} else if (strncasecmp("reset",data,strlen("reset"))==0) {
result=(aPGSQL_reset(chan,data));
} else if (strncasecmp("clear",data,strlen("clear"))==0) {
result=(aPGSQL_clear(chan,data));
} else if (strncasecmp("debug",data,strlen("debug"))==0) {
result=(aPGSQL_debug(chan,data));
} else if (strncasecmp("disconnect",data,strlen("disconnect"))==0) {
result=(aPGSQL_disconnect(chan,data));
} else {
ast_log(LOG_WARNING, "Unknown APP_PGSQL argument : %s\n",(char *)data);
result=-1;
}
LOCAL_USER_REMOVE(u);
return result;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, PGSQL_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,193 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (c) 2004-2005 Tilghman Lesher <app_stack_v002@the-tilghman.com>.
*
* This code is released by the author with no restrictions on usage.
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Stack applications Gosub, Return, etc.
*
* \ingroup applications
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include "asterisk/options.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/chanvars.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/config.h"
#define STACKVAR "~GOSUB~STACK~"
static const char *tdesc = "Stack Routines";
static const char *app_gosub = "Gosub";
static const char *app_gosubif = "GosubIf";
static const char *app_return = "Return";
static const char *app_pop = "StackPop";
static const char *gosub_synopsis = "Jump to label, saving return address";
static const char *gosubif_synopsis = "Jump to label, saving return address";
static const char *return_synopsis = "Return from gosub routine";
static const char *pop_synopsis = "Remove one address from gosub stack";
static const char *gosub_descrip =
"Gosub([[context|]exten|]priority)\n"
" Jumps to the label specified, saving the return address.\n";
static const char *gosubif_descrip =
"GosubIf(condition?labeliftrue[:labeliffalse])\n"
" If the condition is true, then jump to labeliftrue. If false, jumps to\n"
"labeliffalse, if specified. In either case, a jump saves the return point\n"
"in the dialplan, to be returned to with a Return.\n";
static const char *return_descrip =
"Return()\n"
" Jumps to the last label on the stack, removing it.\n";
static const char *pop_descrip =
"StackPop()\n"
" Removes last label on the stack, discarding it.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int pop_exec(struct ast_channel *chan, void *data)
{
pbx_builtin_setvar_helper(chan, STACKVAR, NULL);
return 0;
}
static int return_exec(struct ast_channel *chan, void *data)
{
char *label = pbx_builtin_getvar_helper(chan, STACKVAR);
if (ast_strlen_zero(label)) {
ast_log(LOG_ERROR, "Return without Gosub: stack is empty\n");
return -1;
} else if (ast_parseable_goto(chan, label)) {
ast_log(LOG_WARNING, "No next statement after Gosub?\n");
return -1;
}
pbx_builtin_setvar_helper(chan, STACKVAR, NULL);
return 0;
}
static int gosub_exec(struct ast_channel *chan, void *data)
{
char newlabel[AST_MAX_EXTENSION * 2 + 3 + 11];
struct localuser *u;
if (ast_strlen_zero(data)) {
ast_log(LOG_ERROR, "%s requires an argument: %s([[context|]exten|]priority)\n", app_gosub, app_gosub);
return -1;
}
LOCAL_USER_ADD(u);
snprintf(newlabel, sizeof(newlabel), "%s|%s|%d", chan->context, chan->exten, chan->priority + 1);
if (ast_parseable_goto(chan, data)) {
LOCAL_USER_REMOVE(u);
return -1;
}
pbx_builtin_pushvar_helper(chan, STACKVAR, newlabel);
LOCAL_USER_REMOVE(u);
return 0;
}
static int gosubif_exec(struct ast_channel *chan, void *data)
{
struct localuser *u;
char *condition="", *label1, *label2, *args;
int res=0;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "GosubIf requires an argument\n");
return 0;
}
args = ast_strdupa((char *)data);
if (!args) {
ast_log(LOG_ERROR, "Out of memory\n");
return -1;
}
LOCAL_USER_ADD(u);
condition = strsep(&args, "?");
label1 = strsep(&args, ":");
label2 = args;
if (pbx_checkcondition(condition)) {
if (label1) {
res = gosub_exec(chan, label1);
}
} else if (label2) {
res = gosub_exec(chan, label2);
}
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
ast_unregister_application(app_return);
ast_unregister_application(app_pop);
ast_unregister_application(app_gosubif);
ast_unregister_application(app_gosub);
STANDARD_HANGUP_LOCALUSERS;
return 0;
}
int load_module(void)
{
ast_register_application(app_pop, pop_exec, pop_synopsis, pop_descrip);
ast_register_application(app_return, return_exec, return_synopsis, return_descrip);
ast_register_application(app_gosubif, gosubif_exec, gosubif_synopsis, gosubif_descrip);
ast_register_application(app_gosub, gosub_exec, gosub_synopsis, gosub_descrip);
return 0;
}
char *description(void)
{
return (char *) tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,161 +1,72 @@
/*
* 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, 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 Execute arbitrary system commands
*
* \ingroup applications
* the GNU General Public License
*/
#include <asterisk/file.h>
#include <asterisk/logger.h>
#include <asterisk/channel.h>
#include <asterisk/pbx.h>
#include <asterisk/module.h>
#include <stdlib.h>
#include <stdio.h>
#include <unistd.h>
#include <string.h>
#include <errno.h>
#include <stdlib.h>
#include "asterisk.h"
#include <pthread.h>
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/app.h"
#include "asterisk/options.h"
static char *tdesc = "Generic System() application";
static char *app = "System";
static char *app2 = "TrySystem";
static char *synopsis = "Execute a system command";
static char *synopsis2 = "Try executing a system command";
static char *chanvar = "SYSTEMSTATUS";
static char *descrip =
" System(command): Executes a command by using system(). If the command\n"
"fails, the console should report a fallthrough. \n"
"Result of execution is returned in the SYSTEMSTATUS channel variable:\n"
" FAILURE Could not execute the specified command\n"
" SUCCESS Specified command successfully executed\n"
"\n"
"Old behaviour:\n"
"If the command itself executes but is in error, and if there exists\n"
"a priority n + 101, where 'n' is the priority of the current instance,\n"
"then the channel will be setup to continue at that priority level.\n"
"Note that this jump functionality has been deprecated and will only occur\n"
"if the global priority jumping option is enabled in extensions.conf.\n";
static char *descrip2 =
" TrySystem(command): Executes a command by using system().\n"
"on any situation.\n"
"Result of execution is returned in the SYSTEMSTATUS channel variable:\n"
" FAILURE Could not execute the specified command\n"
" SUCCESS Specified command successfully executed\n"
" APPERROR Specified command successfully executed, but returned error code\n"
"\n"
"Old behaviour:\nIf the command itself executes but is in error, and if\n"
"there exists a priority n + 101, where 'n' is the priority of the current\n"
"instance, then the channel will be setup to continue at that\n"
"priority level. Otherwise, System will terminate.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int system_exec_helper(struct ast_channel *chan, void *data, int failmode)
static int skel_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
if (ast_strlen_zero(data)) {
if (!data) {
ast_log(LOG_WARNING, "System requires an argument(command)\n");
pbx_builtin_setvar_helper(chan, chanvar, "FAILURE");
return failmode;
return -1;
}
LOCAL_USER_ADD(u);
/* Do our thing here */
res = ast_safe_system((char *)data);
if ((res < 0) && (errno != ECHILD)) {
ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
pbx_builtin_setvar_helper(chan, chanvar, "FAILURE");
res = failmode;
res = system((char *)data);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to execute '%s'\n", data);
res = -1;
} else if (res == 127) {
ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
pbx_builtin_setvar_helper(chan, chanvar, "FAILURE");
res = failmode;
ast_log(LOG_WARNING, "Unable to execute '%s'\n", data);
res = -1;
} else {
if (res < 0)
res = 0;
if (option_priority_jumping && res)
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101);
if (res != 0)
pbx_builtin_setvar_helper(chan, chanvar, "APPERROR");
else
pbx_builtin_setvar_helper(chan, chanvar, "SUCCESS");
if (res && ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101))
chan->priority+=100;
res = 0;
}
}
LOCAL_USER_REMOVE(u);
return res;
}
static int system_exec(struct ast_channel *chan, void *data)
{
return system_exec_helper(chan, data, -1);
}
static int trysystem_exec(struct ast_channel *chan, void *data)
{
return system_exec_helper(chan, data, 0);
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
res |= ast_unregister_application(app2);
STANDARD_HANGUP_LOCALUSERS;
return res;
return ast_unregister_application(app);
}
int load_module(void)
{
int res;
res = ast_register_application(app2, trysystem_exec, synopsis2, descrip2);
res |= ast_register_application(app, system_exec, synopsis, descrip);
return res;
return ast_register_application(app, skel_exec);
}
char *description(void)
@@ -169,8 +80,3 @@ int usecount(void)
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

View File

@@ -1,250 +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 Playback a file with audio detect
*
* \ingroup applications
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/utils.h"
#include "asterisk/dsp.h"
static char *tdesc = "Playback with Talk Detection";
static char *app = "BackgroundDetect";
static char *synopsis = "Background a file with talk detect";
static char *descrip =
" BackgroundDetect(filename[|sil[|min|[max]]]): Plays back a given\n"
"filename, waiting for interruption from a given digit (the digit must\n"
"start the beginning of a valid extension, or it will be ignored).\n"
"During the playback of the file, audio is monitored in the receive\n"
"direction, and if a period of non-silence which is greater than 'min' ms\n"
"yet less than 'max' ms is followed by silence for at least 'sil' ms then\n"
"the audio playback is aborted and processing jumps to the 'talk' extension\n"
"if available. If unspecified, sil, min, and max default to 1000, 100, and\n"
"infinity respectively.\n";
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int background_detect_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u;
char *tmp;
char *options;
char *stringp;
struct ast_frame *fr;
int notsilent=0;
struct timeval start = { 0, 0};
int sil = 1000;
int min = 100;
int max = -1;
int x;
int origrformat=0;
struct ast_dsp *dsp;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "BackgroundDetect requires an argument (filename)\n");
return -1;
}
LOCAL_USER_ADD(u);
tmp = ast_strdupa(data);
if (!tmp) {
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return -1;
}
stringp=tmp;
strsep(&stringp, "|");
options = strsep(&stringp, "|");
if (options) {
if ((sscanf(options, "%30d", &x) == 1) && (x > 0))
sil = x;
options = strsep(&stringp, "|");
if (options) {
if ((sscanf(options, "%30d", &x) == 1) && (x > 0))
min = x;
options = strsep(&stringp, "|");
if (options) {
if ((sscanf(options, "%30d", &x) == 1) && (x > 0))
max = x;
}
}
}
ast_log(LOG_DEBUG, "Preparing detect of '%s', sil=%d,min=%d,max=%d\n",
tmp, sil, min, max);
if (chan->_state != AST_STATE_UP) {
/* Otherwise answer unless we're supposed to send this while on-hook */
res = ast_answer(chan);
}
if (!res) {
origrformat = chan->readformat;
if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR)))
ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
}
if (!(dsp = ast_dsp_new())) {
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
res = -1;
}
if (!res) {
ast_stopstream(chan);
res = ast_streamfile(chan, tmp, chan->language);
if (!res) {
while(chan->stream) {
res = ast_sched_wait(chan->sched);
if ((res < 0) && !chan->timingfunc) {
res = 0;
break;
}
if (res < 0)
res = 1000;
res = ast_waitfor(chan, res);
if (res < 0) {
ast_log(LOG_WARNING, "Waitfor failed on %s\n", chan->name);
break;
} else if (res > 0) {
fr = ast_read(chan);
if (!fr) {
res = -1;
break;
} else if (fr->frametype == AST_FRAME_DTMF) {
char t[2];
t[0] = fr->subclass;
t[1] = '\0';
if (ast_canmatch_extension(chan, chan->context, t, 1, chan->cid.cid_num)) {
/* They entered a valid extension, or might be anyhow */
res = fr->subclass;
ast_frfree(fr);
break;
}
} else if ((fr->frametype == AST_FRAME_VOICE) && (fr->subclass == AST_FORMAT_SLINEAR)) {
int totalsilence;
int ms;
res = ast_dsp_silence(dsp, fr, &totalsilence);
if (res && (totalsilence > sil)) {
/* We've been quiet a little while */
if (notsilent) {
/* We had heard some talking */
ms = ast_tvdiff_ms(ast_tvnow(), start);
ms -= sil;
if (ms < 0)
ms = 0;
if ((ms > min) && ((max < 0) || (ms < max))) {
char ms_str[10];
ast_log(LOG_DEBUG, "Found qualified token of %d ms\n", ms);
/* Save detected talk time (in milliseconds) */
sprintf(ms_str, "%d", ms );
pbx_builtin_setvar_helper(chan, "TALK_DETECTED", ms_str);
ast_goto_if_exists(chan, chan->context, "talk", 1);
res = 0;
ast_frfree(fr);
break;
} else
ast_log(LOG_DEBUG, "Found unqualified token of %d ms\n", ms);
notsilent = 0;
}
} else {
if (!notsilent) {
/* Heard some audio, mark the begining of the token */
start = ast_tvnow();
ast_log(LOG_DEBUG, "Start of voice token!\n");
notsilent = 1;
}
}
}
ast_frfree(fr);
}
ast_sched_runq(chan->sched);
}
ast_stopstream(chan);
} else {
ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
res = 0;
}
}
if (res > -1) {
if (origrformat && ast_set_read_format(chan, origrformat)) {
ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n",
chan->name, ast_getformatname(origrformat));
}
}
if (dsp)
ast_dsp_free(dsp);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
return ast_register_application(app, background_detect_exec, synopsis, descrip);
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

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